bluuebunny/crossref_metadata_2025 huggingface dataset:
A six-month post-electrical stimulation effect in speech production
Authors: Abdelli Nassima , Tartter Vivien C. , Economou Alexandra
Published: October 2005

This study documents an unusual effect in cochlear implant patients’ speech 6 months following electrical stimulation. Spectrographic measures were made of 45 phonemically balanced words, recorded before and after electrical stimulation (1 day, 1–6 months, and 1 year). Five Nucleus multichannel cochlear implant patients with different deafness histories participated. A prelingually deafened patient showed a steady decrease in F0 and an increase in duration of several (supra) segmental parameters after electrical stimulation. At 6 months, F0 reached its lowest value and syllable and vowel durations reached their highest values. A postlingually deafened reimplantation patient also showed cusps 6 months post Nucleus: F0 increased for the one and only time, aspiration/frication durations in medial and final syllable positions became abnormally long and then corrected at 1 year, and a long closure was introduced , which diminished at 1 year. The possibility of a critical change in modification of F0 and duration aspects of speech at 6 months is being investigated in other patients. Implications of such a change will be discussed. [Work supported by NIDCD.]
The healing distance of steady-state shock waves in the atmosphere
Authors: Raspet Richard , Bass Henry , Yao Lixin , Wu Wenliang
Published: October 2005

The Anderson algorithm has been used to calculate the time development and rise time of transient shock waves in the atmosphere. This algorithm has been applied to square pulses in order to investigate the healing distance of steady-state shocks perturbed by turbulent scattering. Perturbed 100-Pa shocks require on the order of 1.0-km travel distance to return to within 10% of their steady shock rise time. For 30-Pa shocks the minimum recovery distance increases to 3.0 km. It is unlikely that finite wave effects can remove the longer rise times and irregular features introduced into the sonic boom by turbulent scattering in the planetary boundary layer.
Waveform freezing of sonic booms in the atmosphere
Authors: Cleveland Robin O. , Blackstock David T.
Published: October 2005

Under certain conditions a finite-amplitude wave approaches a frozen state in which no further nonlinear distortion of the waveform takes place even though the wave amplitude would normally be regarded as still finite. Waveform freezing of sonic booms [W. D. Hayes and H. L. Runyan, Jr., J. Acoust. Soc. Am. 51, 695–701 (1971)], which is due to stratification of the atmosphere, is considered in this paper. The physical interpretation proposed is that stratification produces an effective coefficient of nonlinearity βeff that approaches zero with travel distance. Most previous discussions of sonic boom freezing have been based on Hayes’s ray theory computer program. Analytical predictions are reported here for both an isothermal model and a bilinear temperature profile model of the atmosphere. For straight downward propagation, which corresponds to a very high Mach number of the aircraft, the aircraft has to be at an altitude greater than 24 km (80 000 ft) for the boom to be within 5% of its frozen state when it reaches the ground. [Work supported by NASA.]
Real-time implementation of spatial auditory displays
Authors: Foster Scott
Published: October 2005

Recent advances in VLSI hardware have made possible the construction of real-time systems that can ‘‘render’’ acoustics. These systems use tracking mechanisms to monitor the position and orientation of a listener and produce synthetic binaural sound for that listener using headphones. The most sophisticated of these systems utilize supercomputing performance to render complex acoustic scenarios including transmission loss effects, reflection, nonuniform radiation, Doppler effects, and more, resulting in a compelling and natural presentation. With the advances in algorithms and hardware to produce such simulations, there is a need to develop an extensible protocol for virtual audio. Such a protocol will need to encompass all the acoustic models in use today and those expected to be developed in the near future. It is hoped that this protocol will allow virtual reality developers to utilize virtual audio technology, even as that technology makes dramatic improvements in its capabilities.
Applications of sampling musical instruments to simulations and virtual environments
Authors: Massie Dana C.
Published: October 2005

Sampling musical instruments are a powerful resource for virtual environment sound generators. Samplers provide real-time multi-channel playback of digitized audio signals such as individual musical instrument notes or sound effects, under control of real-time user input devices, such as a music keyboard or computer. Samplers also transform the sound through pitch shifting, time-varying amplitude modification (enveloping), filtering, and looping or repeating a section of the sound to create an indefinite sustain from a short representative segment. Interactive simulators can exploit these transformations; pitch shifting can simulate Doppler shift for moving sound sources, and simple models of spatialization can be implemented using amplitude and time delay panning and low-pass filtering. Sampling is analogous to image processing in computer graphics; digitizing and processing of real images (or sounds) may be faster than developing a synthetic model, if the model is even feasible. Sampling and its extensions may thus always offer an advantage over pure model-based approaches.
Perceptual similarity of measured and synthetic HRTF filtered speech stimuli
Authors: Begault Durand R.
Published: October 2005

The relationship between measured and synthetic head-related transfer function (HRTF) headphone localization performance was measured by gathering direct localization estimates of speech material. Thirteen people participated in a ‘‘within subjects’’ design, and evaluated target positions at 30-deg azimuth intervals, all at 0-deg elevation. The measured HRTFs were of a nonparticipating subject. The synthetic HRTFs were derived by two procedures. (1) The magnitude transfer function of the original 512 coefficient impulse response was analyzed via a DFT, and then supplied with a specific weighing scheme to a Parks–McClellan FIR linear phase filter design algorithm, which output a 65-coefficient filter. (2) The interaural delay was implemented by averaging interaural group delay values between 0.2–6 kHz for each target position, resulting in a constant interaural time delay across frequency. Results showed that for each subject, localization judgments were nearly identical between conditions. These results parallel those found for HRTFs synthesized with a principal components analysis model using minimum phase filters [D. J. Kistler and F. L. Wightman, J. Acoust. Soc. Am. 91, 1637–1647 (1989)].
Effects of frequency and masker duration on free-field masking
Authors: Gilkey R. H. , Good Michael D.
Published: October 2005

The design of three-dimensional auditory displays for virtual environments should consider the potential for interference among sound sources. In the present study, free-field masking was measured between sound sources that were separated in azimuth or in elevation. The click train signal and noise masker were filtered to lie within low- (below 1.5 kHz), middle- (1.2–7.0 kHz), or high- (above 3.5 kHz) frequency regions. As expected, based on headphone data and on traditional models of binaural masking, there was a dramatic decrease in masking (as much as 18 kB) when the sound sources were separated within the horizontal plane. Similar decreases in masking (as much as 9 dB) were observed for the high-frequency signal and masker when they were separated within the median plane. Although phenomenologically similar, the masking release observed in the horizontal plane is likely to depend on interaural cues, whereas, these cues should play a minimal role within the median plane. Differences across frequency regions and differences between continuous and pulsed masker presentation are also discussed. [Work supported by AFOSR-91-0289.]
Adaptation to transformed auditory localization cues in a hybrid real/virtual environment
Authors: Shinn-Cunningham Barbara , Durlach Nat , Held Richard
Published: October 2005

The ability of humans to adapt to inter-modal discrepancies is an important factor in the design of virtual environments. In the present study, azimuthal localization cues were altered (to magnify interaural differences) relative to real proprioceptive, visual, and vestibular cues. Subjects were alternately tested and trained in hybrid real/virtual environments where auditory stimuli were synthesized (using a PC, Convolvotron, and electromagnetic head tracker) to be a 1 of 13 discrete positions marked by real lights. Testing consisted of identifying the azimuth of virtual sound sources without correct answer feedback or significant head motion. Preliminary findings on resolution and bias for a variety of different training procedures as well as a number of different transformations of the localization cues will be overviewed. [Work supported by AFOSR, Grant No. 90-200.]
Projection of near-field measurements of scattered pressure to the far field
Authors: Washburn Karl B. , Williams Earl G. , Houston Brian H.
Published: October 2005

NRL has successfully produced the first scattered hologram of an underwater shell, ensonified by a bow-incident plane wave. The total acoustic pressure is measured as a hologram on a cylindrical surface concentric to and in the near field of the shell. The scattered pressure is constructed by subtracting a hologram of the incident field, measured without the shell, from the total field hologram, with the shell in place. Using generalized near-field acoustical holography (GENAH), the scattered hologram is processed to reconstruct the pressure and radial velocity on the surface coincident with the shell exterior. The Helmholtz integral equation (HIE) is then applied to the pressure and velocity to compute the far-field directivity pattern. The shell used in the holographic measurement had its far-field response directly measured as well: the two measurements will be compared. Sources of error, including scattered energy escaping from the ends of the unclosed cylindrical contour used in the HIE, will be examined.
A method for predicting community annoyance due to exposure to sonic booms
Authors: Hubbard Harvey H. , Shepherd Kevin P.
Published: October 2005

A method to predict community annoyance due to sonic booms has been formulated in which the exposure is expressed using the equivalent day–night average sound level metric (Ldn). It is based on the paired comparison results of Kryter’s sonic boom flyover tests [K. D. Kryter, TheEffectsofNoiseonMan (Academic, New York, 1970)] and a synthesis of available community response data by Schultz for aircraft, highway, and railway noise sources [T. J. Schultz, ‘‘Synthesis of Social Surveys on Noise Annoyance,’’ J. Acoust. Soc. Am. 64, 377 (1978)]. The method incorporates average indoor–outdoor responses and includes the effects of sonic boom overpressures and frequency rates of exposure. Good agreement is shown between the predicted percentages of highly annoyed persons and the limited results from several community surveys involving aircraft supersonic overflights. [Work conducted in part by Douglas Aircraft Co. and Wyle Labs. under contract to NASA−Langley Research Center.]
Results from a sonic boom signature distortion algorithm combined with a turbulent boundary layer evolution model
Authors: Gionfriddo Thomas A. , Sparrow Victor W.
Published: October 2005

Recently a new algorithm for quantifying the distortion of sonic boom signatures was presented by Gionfriddo [J. Acoust. Soc. Am. 91, 2453 (A) (1992)]. In that study digitized sonic boom waveforms from aircraft overflights recorded by the Air Force were compared to classic N waves. A statistical relation between the waveform distortion and the amount of turbulence a sonic boom encounters during propagation was sought. That study assumed a turbulent boundary layer of constant height. In the present research the height of the turbulent boundary layer has been allowed to change based on use of the Blackadar (1989) model. In this model meteorological data from rawinsonde, satellite, and ground station sources during the days of the overflights provide a more accurate estimate of the boundary layer size. It has been observed that in some cases, there exists a strong linear correlation between the sonic boom signature distortion and the amount of turbulence through which the wave traveled. In the remaining cases, a good to fair level of correlation is reported. [Work supported by NASA−LaRC.]
A ray theory model of sonic boom propagation through a turbulent ground layer
Authors: Robinson Leick
Published: October 2005

The ZEPHYRUS computer model has been developed to calculate sonic boom propagation through a realistic, nonturbulent atmosphere [L. Robinson, Ph. D. thesis, Univ. of Texas at Austin (December 1991)]. A technique is presented to extend this model to include the effects of a turbulent atmospheric ground layer on the shape of sonic boom waveforms. Ray theory is used to model the effects of refraction and subsequent folding of the wave front by an instantaneous realization of the turbulence. The problem of maintaining wave-front coherence and efficient eigenray identification algorithms are discussed. A Monte Carlo method is used to form a statistical ensemble of predicted sonic boom waveforms. These results are compared with characteristics observed in sonic boom measurements. [Work supported by NASA−Langley.]
Performance of acoustic tomography via matched-field processing
Authors: Tolstoy A.
Published: October 2005

Results will be presented showing the deep ocean performance of the matched-field processing (MFP) approach to acoustic tomography as discussed in Tolstoy etal. [J. Acoust. Soc. Am. 89, 1119 (1991)] and Tolstoy [J. Acoust. Soc. Am. 91, 781 (1992)]. These results are generated for the simulated environment provided by Chiu etal., which was selected and designed for comparison between presently available tomography techniques. The MFP approach requires the presence of at least three vertical arrays plus low-frequency sources, e.g., air-deployed shots, distributed around the region perimeter. The resolution of the final inverse solution is dependent on the number and placement of the sources and arrays, on the quality and availability of additional environmental information, e.g., sound-speed profile data at the arrays, and on our ability to mode filter or time window the acoustic data to eliminate bottom interacting energy.
The effect of acoustic tomographic data in a dynamical model-based assimilation scheme
Authors: Robinson A. R. , Pourkaviani M.
Published: October 2005

Two problems are addressed using historical acoustic data. The first is the assessment of oceanographic content of tomographic data when the inverse method involves assimilation into a realistic general ocean dynamical model. The second is to consider the impact of tomographic data on field estimates when it is assimilated in combination with hydrographic and other data. [Work supported partly by the Office of Naval Research.]
Acoustic thermometers at basins-scales in the northeast Pacific
Authors: Spiesberger John L. , Duda Timothy , Freitag Lee
Published: October 2005

Acoustic multipaths (250-Hz, 64-Hz bandwidth) from basin-scale (1000–4000 km) sections in the northeast Pacific are stable from month-to-month. This stability is a necessary pre-requisite for measuring climatic temperature change in the ocean. A new ray-tracing program [J. Bowlin etal., Woods Hole Oceanogr. Inst. technical report, Woods Hole, MA 02543 (in preparation, 1992)] is used to identify observed multipaths with more reliability than possible with previous ray trace programs.
New theory/algorithms for bubble density measurement using inverse acoustic scattering techniques
Authors: Duraiswami Ramani
Published: October 2005

Acoustical techniques have long been used to estimate the bubble density function. The conventional technique assumes isolated (non-interacting) scatterers, and results in a Fredholm integral equation of the first kind relating the bubble density and the measured scattering is obtained. These ill-posed equations are numerically challenging to solve, especially when the data−obtained from experiments−inevitably contain noise/error. Usually these equations are solved by assuming resonant scattering, or by accounting approximately for off-resonance scattering. Additionally the conventional model used for the bubble scattering, does not properly account for thermal losses in the bubble oscillations. In the present work, a multiphase model for sound propagation through bubbly liquids (due to Caflisch etal.) is combined with Prosperetti’s model for bubble oscillations, to develop twonewequations for determining the bubble population function from measured phase-velocity and attenuation data. The new theory/equations address perceived drawbacks in the conventional technique. The equations are evaluated for their potential for determining the bubble population, by testing them with analytical data with varying artificial noise. Numerical algorithms using new regularization techniques are developed. [Work supported by NSF.]
On ray/mode approaches to the broadband shallow water tomography problem in the coastal Barents Sea acoustic test experiments
Authors: Shmelerv A. Yu. , Kozel’sky A. R. , Migulin A. A. , Lynch James F.
Published: October 2005

In shallow water tomography, use of acoustic sources with 100-Hz bandwidth causes great problems with resolution and identification of the arrivals due to diffraction and dispersion at low frequency. These problems can be resolved in terms of ray theory with beam displacement and mode theory in different ways. This paper discusses the results of numerical calculations of ray/mode resolution and estimates dispersion in shallow water waveguides. Numerical calculations were carried out for an ideal model waveguide with linear and isovelocity sound-speed profiles as well as for the coastal Barents Sea acoustic test experiment conditions. The experiment took place during October 1990 in the coastal region of the Barents Sea (to be published in Sov. Phys. Acoust. in 1992). Presented here are also plans for future coastal Barents tomography experiments.
The effect of turbulence on the loudness of sonic booms
Authors: Plotkin Kenneth J.
Published: October 2005

Turbulence is known to distort sonic booms in two ways: random perturbations occur at and following each shock, and the shock rise time increases. Both types of distortion affect the high-frequency content of a boom, and are therefore important to loudness. It is of particular interest whether this distortion will adversely affect the loudness of minimized sonic boom signatures that are associated with low-boom supersonic transport design. Available scattering theories for the magnitude of perturbations and for shock rise times were extended to provide estimates of the spectral content of the distortions. During this analysis, care was taken to recognize the interacting roles of turbulent scattering, absorption of sound by molecular relaxation, and weak nonlinear steepening. It was found that the perturbations at and following the shocks tend to increase the loudness, while the increased rise times tend to decrease the loudness by a comparable amount. Because turbulent scattering is random and three dimensional, an individual boom may be affected by one of these effects more than by the other. Turbulence can therefore cause significant variation in the loudness of individual booms. The change to the average loudness of a number of booms, which is primary to the assessment of cumulative community impact, will, however, be small. [Work supported by NASA−Langley Res. Ctr.]
Across frequency grouping in the binaural masking release of complex sounds
Authors: Culling John F. , Summerfield Quentin
Published: October 2005

Common periodicity has been proposed as a basis for segregating concurrent sounds that differ in fundamental frequency. Common interaural time delay may play an analogous role in segregating complex sounds whose sources differ in spatial position. To explore this analogy, masked identification thresholds were measured for synthesized vowels in pink noise in several binaural conditions. The across-frequency coherence of each sound’s interaural phase relationships was manipulated. The results were interpreted in terms of an interaural cross-correlation mechanism that groups components that produce cross-correlation maxima at the same delay. Masking release (compared to the condition where both noise and vowel were in phase at each ear) was found to depend mainly upon the coincidence across frequency, at the same correlation delay, of cross-correlation maxima due to the noise. Masking release was largely independent of the pattern of interaural correlations generated by the target vowel, providing the pattern differed from that of the masking noise. The result is analogous to the demonstration that the periodicity of maskers, not target vowels determines ease of segreagation based on differences in fundamental frequency, and shows that the binaural masking release of complex sounds includes an across-frequency grouping mechanism.
The proportion-of-the-total-duration (PTD) rule holds for duration discrimination
Authors: Kidd Gary R. , Watson Charles S.
Published: October 2005

In previous work, it was demonstrated that frequency resolving power for each individual component of an unfamiliar sequence of tones increases with the component’s proportion of the total sequence duration [G. R. Kidd and C. S. Watson, J. Acoust. Soc. Am. (to be published)]. This work has now been extended to the case of duration discrimination. In this case, the dimension affected by changes in PTD (i.e., time) is also the primary dimension of variation within the patterns, as well as the dimension to which listeners must attend to perform the task. Listeners were asked to detect a change in the duration of a single tone in a five-tone pattern using a modified two-alternative forced choice procedure. Target-tone durations were determined by the PTD value (0.1, 0.2, or 0.4) and the total pattern duration (250 or 750 ms). Context-tone durations were determined randomly on each trial. A single frequency pattern, consisting of a sequence of ascending frequencies, was used throughout the experiment. The pattern of results obtained was essentially the same as that found in the frequency-discrimination experiments. Increases in the proportion of the total pattern duration occupied by the target tone consistently resulted in lower duration-discrimination thresholds. [Work supported by NIH with partial support from AFOSR to the Institute for the Study of Human Capabilities.]
Identification of novel multidimensional speech-like sounds and the effects of training
Authors: Christopherson Laurel A. , Humes Larry E.
Published: October 2005

Listeners classified Klatt synthesized nonspeech stimuli that differed on three independent and discriminable dimensions. The three dimensions of the stimuli were fricative spectrum, frequency transition, and temporal gap. Each of the three dimensions could take on one of three possible values. Twenty-seven stimuli were synthesized using all possible combinations of the dimension values. Multidimensional scaling of paired-comparison similarity judgments confirmed the existence of three perceptual dimensions. Subjects were then trained to classify three exemplar stimuli as ‘‘circle,’’ ‘‘triangle,’’ and ‘‘square,’’ respectively. Following this training, subjects classified all 27 stimuli as ‘‘circle,’’ ‘‘triangle,’’ or ‘‘square.’’ From this it was determined how attention was distributed among the dimensions. Finally, the subjects’ ability to shift their attention from dimension to dimension was examined. [Work supported by NIA and AFOSR.]
Developments toward a computer-based course in acoustic phonetics
Authors: Berkovitz Robert A. , Pickett J. M.
Published: October 2005

A course consisting of computer-based lessons is under development to teach principles of speech sound production and acoustic patterns of phonetic features. The course employs an IBM-compatible personal computer and a highly flexible graphic format, and makes extensive use of the speech synthesis and analysis capabilities of the SpeechStation system. The teaching approach will be demonstrated and made available for hands-on, interactive use with lessons under development. [Work supported by the National Institute on Deafness and Other Communication Disorders.]
A telemetry scheme for ocean acoustic tomography: Real time monitoring
Authors: Spiesberger John L. , Bowlin James B.
Published: October 2005

A new telemetry scheme is described that will make it possible to monitor ocean temperatures in real time using acoustic tomography. The system incorporates autonomous sources mounted on subsurface moorings and receivers that are either cabled to shore or suspended from surface units that drift. The telemetry scheme predicts that the corruption of acoustic travel times, due to wander of the source’s mooring, can be largely eliminated in real time by shifting the start times of the tomographic transmissions in special ways. Corrections to source wander can be obtained without expending battery energy over and above that used in conventional tomography experiments. Standard techniques can be used to correct clock errors at the source in real time.
Acoustic transfer impedance of a complex piping system
Authors: Song Limin
Published: October 2005

A transfer impedance method has been developed to analyze the acoustic response of a complex piping system to low-frequency pulsation sources such as a reciprocating compressor. The acoustic transfer impedance of a piping system is defined as the complex amplitude of acoustic pressure in the piping system generated by a constant-frequency source at a given location that has a unity complex amplitude of acoustic volume velocity. In this method, the transfer impedance in each pipe component is expressed in terms of transmission and reflection coefficients that are determined through boundary conditions. Recurrence relations are derived to relate the transmission and reflection coefficients of one pipe component to those of an adjacent pipe component. This method is particularly useful in analyzing piping acoustic resonance and pulsation mode shape in which the location of maximum acoustic pressure can be easily identified. If the pulsation source is known, the pulsation levels in the piping system can be calculated. Compared with most of existing methods, this method provides a better understanding on how acoustic energy propagates in the piping system in addition to giving a quicker and simpler solution. Further development of this method may lead to design optimization of pulsation control devices.
The beginnings of resonant ultrasound spectroscopy and its applications to geophysics
Authors: Anderson Orson L.
Published: October 2005

It is no accident that resonant ultrasound spectroscopy (RUS) work is deeply connected to geophysics problems. A great triumph of geophysics in the past two decades has been the inversion of the measured resonant vibration modes of the oscillating Earth as a whole, resulting from a major earthquake, to find the density distribution with depth. Over the last 20 years the fundamental theory has been extended by a series of steps to include nonspherical shapes (parallelepipeds, cones, ellipsoids, and right circular cylinders) and also materials of lower symmetry (including cubic down to monoclinic). The advent of supercomputers with their large computation capacity has made all of this possible. The benefits of applying resonant ultrasound spectroscopy to geophysical problems are also benefits to materials science, since the geophysical approach is almost identical to a good materials science approach. In this paper, some results are presented of using RUS to find solutions to some outstanding geophysical problems mainly related to high temperature applications. The materials science applications of the work are found when one emphasizes measurements of insulators of low crystal symmetry at high temperatures (up to 1800 K). This is, of course, the special domain of high-temperature ceramics. A large portion of this paper is devoted to a historical review of RUS.
Experimental study of sound scattering by elastic spheroids, and the excitation of their resonances
Authors: Bao X. L. , Überall H. , Niemiec Jan
Published: October 2005

Experimental studies of ultrasonic scattering by submerged steel spheroids, both oblate and prolate, and of steel spheres have been carried out using incident short, sinusoidally modulated pulses. The time waveform and frequency responses of the scattering objects are obtained directly from experimental data via numerical processing. The sound pulses are incident in the axial direction and also in discrete oblique directions at every 5 deg from the axis. The sound paths of surface waves generated on the objects, causing a chain of echo pulses, are identified, echo arrival times are shown to agree with the surface wave interpretation, and resonanc spectra are extracted using the Numrich–de Billy method. For explaining the resonance spectrum, the phase-matching principle is employed. The target strength of the objects is extracted from experiment using various methods of interpretation (specular echoes, elastic-wave echoes, total echo energy, or entrance windows), and compared to the geometrical target strength. It is shown that specular echoes alone are not sufficient to explain the entire target strength, and the presence of elastic surface waves makes itself felt here.
A study of materials at high temperature using miniaturized resonant tuning forks
Authors: Schindel D. W. , Hutchins D. A. , Smith S.
Published: October 2005

Miniaturized tuning forks, whose resonant frequencies are temperature dependent, have been manufactured from various materials using a precision milling machine under computer control. Two noncontact measurement systems that rely upon capacitive transducers as well as a pulsed ruby laser have allowed the generation and detection of fork vibrations at temperatures up to 900 °C. The usefulness of such a system for materials characterization will be demonstrated through the successful detection of the precipitation and dissociation of hydride platelets in a fork consisting of a Zr–Nb alloy. Also to be discussed are studies of single-crystal silicon tuning forks and the resultant program currently underway for the development of micromachined thermometers for corrosive and high-temperature environments.
Acoustical modeling of flexible perforated screens attached directly to elastic porous materials
Authors: Bolton J. S. , Green E. R.
Published: October 2005

Perforated screens are frequently used to face sound absorbing porous materials since they serve both to protect the porous material and to enhance its effectiveness at low frequencies. Recently, exact theories describing the combination of a perforated screen and fibrous materials that can be modeled as equivalent fluids have been presented for both normal and arbitrary incidence angles. In this paper a similar analysis is presented that is appropriate when the sound absorbing material is an elastic porous material. The perforated screen itself is modeled as an Euler–Bernoulli panel whose stiffness and mass per unit area are modified by its degree of perforation. The boundary conditions describing the interaction of the elastic porous material and the screen require, among other things, that the velocities of both the solid and fluid phases of the porous material are simultaneously equal to the normal velocity of the nonperforated area of the panel. In addition, the volume velocity of the air within the perforations is required to equal the porosity weighted sum of the fluid and solid velocities of the porous material in the regions in which the porous material is adjacent to a perforation. By applying the boundary conditions appropriately, closed-form expressions may be found for the amplitude of each of the infinite number of plane waves scattered and transmitted by periodically arranged perforations. Example calculations that illustrate both the nature of sound absorption by the combination of polyurethane foam and a perforated screen and sound transmission through the same combination will be presented.
Theoretical and numerical aspects of absorbing ducts with asymmetrical lining
Authors: Ochmann Martin
Published: October 2005

The propagation loss of absorbing ducts will be calculated with the aim to find the optimal design of the absorber for a specific problem. An infinite, rectangular duct lined with locally reacting absorbers will be considered. The impedances of the lining surfaces could be different, so that the asymmetrical case is included. To calculate the propagation loss due to absorption the boundary value problem of the wave equation in the absorbing channel has to be solved and the eigenfunctions (or modes) and eigenvalues have to be determined numerically. It can be shown that these modes are biorthogonal and that it is possible to expand a prescribed incident wave field into a sum of absorber modes. Since the underlying mathematical boundary problem is non-self-adjoint it is not possible to derive the expansion theorem by using the usual Sturm–Liouville theory. Instead of this Cauchy’s integral formula has to be used for the series expansion in connection with Green’s function. The eigenvalue problem will be solved numerically by using Muller’s method. Numerical results for different absorber configurations will be presented and discussed.
Reciprocal technique for determining the sound radiation due to loudspeaker enclosure vibrations
Authors: Starobin Bradley M. , Arzoumanian Sevag H.
Published: October 2005

The contribution of enclosure-borne sound to a loudspeaker’s total radiated sound was determined indirectly by a combination of analytical and experimental techniques. The transfer functions that relate a loudspeaker enclosure’s average vibratory response both to point force excitation and to external acoustic excitation were determined experimentally for several different enclosure prototypes. A reciprocity relation that equates the radiated pressure to force transfer function, which was not measured directly, to the acoustically induced vibrational velocity to point source volume velocity transfer function, determined experimentally, was used in evaluating the loudspeaker enclosure radiation resistance. Finally, enclosure-borne radiated sound power, as derived from measured enclosure vibrational velocity, was related to radiated sound power due to rigid body loudspeaker cone vibration. Some of the enclosures exhibited certain structural modes which, through these studies, were identified as efficient sound radiators, and as such, candidates for further study in order to limit their radiated output.
Three-point transfer function method for impedance-tube measurements
Authors: Chu W. T.
Published: October 2005

Using computer simulations, Bolton has shown that measurements with three equally spaced microphones may be used simultaneously to determine the complex propagation constant and the forward- and backward-going wave components in an impedance tube [J. S. Bolton and M. Kompella, J. Acoust. Soc. Am. Suppl. 1 84, S96 (1988)]. The impedance of the test material may then be determined. The practicality of this proposal has been studied in more detail, using both computer simulations (with realistic inputs containing random errors) and actual experimental implementation with the single microphone technique [W. T. Chu, J. Acoust. Soc. Am. 83, 2255–2260 (1988)]. Results indicate that the method is feasible provided that the microphone positions are properly chosen. This requirement will limit its application.
Performance of shaded hydrophones in the reduction of turbulent flow noise
Authors: Ko Sung H.
Published: October 2005

Turbulent boundary layer pressure fluctuations can be reduced by filtering either a wave number through a finite hydrophone or a hydrophone array or by filtering the wave number through an elastomer layer. In general practice, various configurations of hydrophone arrays are embedded within a layer of elastomer, thus reducing the turbulent boundary layer pressure fluctuations. The theoretical model considered in this paper is a plane elastomer layer backed by an infinitely rigid surface; the other side of the layer is exposed to turbulent flow. This paper examines the performance of shaded, rectangular and circular hydrophones for the flow noise reduction. The rectangular hydrophone is shaded with a triangular or semicircular loading. Similarly, the circular hydrophone is shaded with a conical or hemispherical loading. Numerical calculations were made for both flush-mounted and embedded hydrophones. The results presented are comparisons of numerically calculated noise reductions between shaded and unshaded hydrophones.
Automatic sound source detector
Authors: Narasimhan Ravi
Published: October 2005

This event-driven system determines sound source location using CMOS-EPROM technology and a microprocessor interface. The time delay between a pair of microphones is measured. This time delay gives one direction. Another pair of microphones yields a necessary second direction. The source location is determined precisely by considering hyperbolic equations and the linear approximation. The microphone outputs are ∼200 mV. These signals, amplified to logic high level, are inputs to the Altera EPM5032 erasable programmable logic device (EPLD). The EPLD is programmed using MAX+PLUS software and a hardware programmer. A circuit designed to determine which microphone belonging to a pair received the sound first together with a 2-MHz crystal oscillator and a 12-bit binary counter used to measure the time delay. The counts are fetched by an assembly code in the NSC888 microprocessor board and sent to a program in the Macintosh, to calculate the sound location, through the RS232 modem port. The program calculates and immediately displays the sound location. The test results of this system illustrate its reliability and accuracy.
Observation of an aircraft transit using a land-based acoustic array
Authors: Ferguson Brian G. , Trider Robert C. , Hanlon James L.
Published: October 2005

A turbo-prop aircraft flew at a constant altitude and speed over and along the axis of a line array consisting of 15 microphones spaced 0.9 m apart. The acoustic data from the microphones were processed by a frequency-domain beamformer, which led to the acoustical Doppler effect being observed on the various output displays of the processor. Color displays showing the variation of the beamformer’s output power as a function of frequency and angle of arrival are presented here for a conventional beamformer, an adaptive beamformer, and an adaptive beamformer with a weight norm constraint. Adaptive beamforming maximizes the array gain resulting in the spatial leakage being lower and the widths of the beams being narrower when compared with conventional beamforming. Imposing an upper bound on the norm of the adaptive weight vector reduces the signal suppression that occurs in adaptive beamforming as a result of system errors. The observed variation of the Doppler shift in the propeller blade rate with the angle of arrival is compared with the variation predicted using an isospeed sound propagation model.
A multichannel digital system for use in calibrating towed arrays
Authors: Forsythe S. E. , Brown C. K.
Published: October 2005

A long-line hydrophone calibrator (LLHC) has been developed at NRL–USRD to serve as a platform for calibrating a variety of towed arrays. The LLHC consists of multiple projectors and hydrophones in a long water-filled pipe. Digital control of the projectors allows a precisely controlled acoustic field to be generated in the pipe. The LLHC makes extensive use of digital signal processing to control and monitor up to 128 acoustic channels. The DSP algorithm for field generation is distributed among 9 signal processors, each controlling 15 acoustic channels. The transfer function of the pipe is characterized by driving each projector separately and recording the responses of all hydrophones to it. To build a specified sound field in the pipe, the inverse of the response matrix is multiplied by the vector of desired hydrophone responses to form the vector of drives required to achieve the desired field. This presentation describes the details of the electronics and software necessary to build the desired field. [Work supported by NAVSEA.]
Analytical approach to vibration of a slotted beam
Authors: Cheng Xiuting
Published: October 2005

Vibration of a slotted beam was approached analytically by considering it equivalent to a uniform beam with local additions to bending moment, shear force, and distribution force. Natural frequencies and vibration modes of the slotted beam were studied using this analytical model. Changes in these vibration parameters were studied with respect to the slot location, slot size, etc. Comparison with other theoretical approaches and experimental results will be made. [Work supported by NSF Grant No. MSS-9024224.]
The three-dimensional laser vibrometer
Authors: Vignola Joseph , Houston Brian H.
Published: October 2005

A noncontact tool for measuring simultaneously the three components of vibration-induced displacement on the surface of underwater structures has been designed and constructed. From a distance of approximately 2 ft from the surface of a target structure the three components of motion can be measured. Data for acoustic studies can be obtained by collecting data at a series of points on the surface. Optical fiber is used so that a compact optical probe with no electronics is the only device in the water. The majority of the hardware, both optical and electrical, are located on an optical bench that does not need to be near the probe or target structure. An analysis of the sensitivity, frequency response, and other performance characteristics of the system will be included.
Analysis and characterization of ambient noise in shallow water
Authors: George Jacob
Published: October 2005

Sources and characteristics of ambient noise in deep water have been the subject of extensive investigations [e.g., R. J. Urick, PrinciplesofUnderwaterSound (McGraw-Hill, New York, 1983), Chap. 7, and references therein]. More recently, fluctuations of ambient noise in shallow water has been studied by several authors [e.g., Clay etal., J. Acoust. Soc. Am. 77, 424 (1985), and references therein]. In the present work, ambient noise data from a shallow water experiment in the Pacific (water depth 600 ft; range of frequencies 50–500 Hz) are analyzed. Cumulative distribution functions and other features are examined and are contrasted with findings from the results of previous research. [Work supported by ONT.]
Acoustic measurement of sediment transport: An overview
Authors: Hay Alex E.
Published: October 2005

The field of sediment transport measurement using acoustic remote sensing technique is reviewed, summarizing recent advances and possible directions for the future. The focus is on elucidating sediment transport mechanisms in the bottom boundary layer, and particularly on estimating terms in the sediment mass transport equation. A key problem is the inversion of acoustic backscatter profiles to obtain suspended sediment size and concentration. Several fundamental issues are involved: the scattering cross sections of irregular particles; estimating scatterer concentration fluctuations from fluctuations in signal amplitude; the related problem of inversion stability in the presence of fluctuations; the spatial coherence scales of the concentration field; and sediment mass flux estimation. These issues are illustrated with results from measurements made both in the laboratory and in the field. While several of the above issues remain unsolved, the results are nevertheless quite promising, and indicate a bright future for the application of acoustics to the sediment transport measurement problem.
Field measurements of suspended sand using acoustic backscatter
Authors: Hanes Daniel M.
Published: October 2005

The suspension of sand by waves has been studied in the field under shoaled wave conditions through the application of acoustic backscatter techniques. The spatial and temporal resolution of the techniques allow for the examination of suspended sediment concentration on time scales corresponding to individual waves, and space scales of approximately 1 cm. This has led to the identification and description of near-bed intermittent suspension events, and provided insight into the mechanisms for suspended sediment transport in coastal regions. [Work supported by ONR.]
Impingement tones of axisymmetric choked jets on small plates
Authors: Henderson Brenda , Powell Alan
Published: October 2005

As reported previously, the impingement of axisymmetric choked jets on small normal flat plates produces two classes of intense tones. These occur with large amplitude oscillations of the stand-off shock wave that interacts with the shock also present in a free jet, while the sound waves emanate from at least two definite flow regions. In one class, the sound waves are quite impulsive and appear to be emitted from the stand-off shock region of the flow, the diameter of that shock and the jet varying periodically. The other class occurs when at least two shocks of the free jet are present. The influence of an upstream reflector implies that acoustic feedback to the nozzle occurs, while the preferred upstream radiation appears attributable to a simple source array at the multiple shocks in the supersonic conical deflected jet flow. [Work supported by Texas Advanced Research Program and NASA Graduate Fellowship Program.]
Estimation of the geoacoustic properties of the upper oceanic crust from bottom reflection loss measurements
Authors: Hannay David E. , Chapman N. Ross
Published: October 2005

Geoacoustic models have been estimated for the upper oceanic crust by inversion of bottom reflection loss versus angle data. The data were obtained in the PACIFIC ECHO experiments at thin-sediment sites in the North Pacific where the age of the crust increased from 40 to 64 million years. The measurements were carried out using a horizontal line array and small explosive charges, and the reflection loss was determined from beamformed array data in order to resolve the specular component. The geoacoustic models consist of profiles of the density, and the compressional (P) and shear (S) wave speeds and attenuations of both the sediment and basalt crust layers. The inversion assumes either a simple homogeneous half-space basement, or a layered model with constant sound-speed gradients in the upper crust overlying a homogeneous basement. The results indicate that the P-wave speed near the sea floor increases with increasing age of the oceanic crust from values of 2700 to about 3300 m/s. These values are significantly lower than those for the sound speed of basalt at thick sediment sites of similar crustal age. Also, there is evidence of a shear wave critical angle at the oldest site. The effects of seafloor scattering and sediment shear wave resonances are discussed.