bluuebunny/crossref_metadata_2025 huggingface dataset:
Human hearing without efferent input to the cochlea
Authors: Scharf Bertram
Published: October 2005

The measurement of hearing without efferent input is possible in human patients whose vestibular nerve has been sectioned, to relieve severe vertigo usually caused by Ménière’s disease. The olivocochlear bundle, which runs in the inferior vestibular nerve, is also sectioned. Psychoacoustic measures have been made in over a dozen such patients, whose hearing levels range from normal to 40 or 50 dB. Most measurements were made only after the operation, but many were also made before. Thresholds measured before and after the operation revealed no change in the ability to detect tones either in the quiet or in noise, except when selective frequency focusing was involved. Comparisons between operated and normal ears showed a similar pattern. Other psychoacoustic tests, including measurements of intensity, frequency, and gap discrimination, of loudness functions and loudness adaptation, of lateralization, of the auditory filter (by notched noise), of overshoot, of TTS, showed either normal auditory functions or only the changes usually associated with sensorineural impairment. Besides the laboratory tests, most patients’ reports indicate no change in hearing subsequent to the vestibular neurectomy. Apparently, the one change—in selective listening—has little effect on such common auditory functions as speech perception. [Research supported by NIH.]
Manufacturing disks and disk drives: Open problems
Authors: Essa Kevin
Published: October 2005

The talk will concentrate on the significant open problems, still on hand, in the daily manufacturing reality of the disks and the disk drives. New, fresh approaches in terms of intelligent controls and software mimicking will be emphasized.
Plate volume probing in manufacturing (or, extension modes in thin plates excited by broad normal sound beams)
Authors: Hazony Dov
Published: October 2005

A thin plate is excited by a normal acoustic beam of a finite area. The beam forms a propagation channel across the plate. This propagation mode is analyzed when the driving signals are sharppulses. But, the main emphasis is on leaks that radiate off the main channel. These contain extension modes that travel along the off-channel axis in a coherent manner. Further, it will be seen that these may be singled out to monitor volumes of plates, walls of tubes, and inspection of welds; especially when access is restricted to a single surface. Experimental results will be presented and discussed. [Work supported by ONR and TecSonics, Inc.]
Acoustic emissions in micromanufacturing
Authors: Ünal A. , Wu C. H. , Mackertich S.
Published: October 2005

Micro manufacturing is posing fundamental problems in terms of realizing miniaturized parts and systems in real world of manufacturing [A. Ünal, J. Acoust. Soc. Am. Suppl. 1 88, S20 (1990)]. Acoustic emissions combined with modern artificial neural network based intelligent signals analysis may present new possibilities. Some real life industrial examples will be considered in this presentation.
Syllt for building deltas: Simple speech synthesis for teaching and research
Authors: Hertz Susan R. , Zsiga Elizabeth C. , Huffman Marie K.
Published: October 2005

Syllt is a new computer software tool for speech synthesis on PCs and Sun workstations. Using the Delta System [Hertz, PapersinLaboratoryPhonologyI (1990)], Syllt implements a phone-to-speech rule set that synthesizes high-quality CVC syllables from a string of phones entered by the user. The program allows rapid creation of synthetic stimuli for perception experiments, and provides a tool for teaching acoustic phonetics. In the process of creating speech output, Syllt produces a multi-tiered utterance representation (a ‘‘delta’’) that coordinates phonological units (such as phonemes), phonetic units (such as bursts), and quantitative parameter values (such as formant frequencies) for a Klatt synthesizer. Users can interactively manipulate the value and relative timing of any of these elements—for example, timing of voicing relative to stop release, amplitude and frequency of burst noise, or formant trajectories. The result of any change can be immediately heard and evaluated. Delta manipulation can be easily automated to generate continua of stimuli that differ along one or more dimensions. The user has access to the rule set itself, which is extensively documented and can be customized for particular research needs. The structure and performance of Syllt will be demonstrated. [Work supported by NIH.]
The use of sampled consonants for improved intelligibility in formant synthesizers
Authors: Moran Heather , Hata Kazue , Pearson Steve
Published: October 2005

This paper investigates the improvement of intelligibility in a formant synthesizer by using a library of sampled consonants extracted from natural waveforms. Four hundred and seventy-one monosyllable words containing consonants in all possible vowel environments from a male American English speaker were recorded. The current focus is on the voiceless consonants. A comparison test was conducted between our current synthesizer [K. Matsui etal., Proc. ICASSP 2, 769–772 (1991)] and the sampled consonant system. Eight naive listeners participated in a simple intelligibility test. The stimuli list consisted of 110 tokens, 60% of which were nonsense words. The results showed that the sampled consonant system was significantly higher (by 20%) in overall intelligibility score. In terms of consonant classes, initial stops showed the most improvement with a 26% increase. Weak fricatives did not show a dramatic difference between the two systems. In further experiments these were improved utilizing additional factors such as modification of the spectral shape, manipulation of transitional cues into vowels and durational adjustments. The relative importance of each factor will be discussed.
Time-frequency analysis of modulated speech noises
Authors: Ravera Bertrand , d’Alessandro Christopher
Published: October 2005

The production of some speech sounds involves the periodic modulation of a noise component (e.g., the voiced fricatives in French, or aspiration noises in breathy vowels). According to the linear acoustic model of speech production, the speech signal is produced by filtering an excitation signal with a linear time-varying filter (which represent the vocal tract transfer function and sound radiation). For instance, in voiced fricatives, the excitation signal is a mixed signal resulting from the modulation of a frication noise source by the glottal flow. Two representations, taking into account the nonstationary nature of mixed-excitation speech sounds, were studied: cyclostationary analysis (using a spectral correlation (SC) estimator of the cyclic frequency-frequency spectrum) and nonstationary analysis [using a smoothed pseudo Wigner–Ville (WV) estimator of the Wigner–Ville spectrum]. The theoretical and experimental results obtained on test signals and actual speech show that some acoustic parameters can be estimated using these analysis methods (frequency of modulation using SC; time structure of the modulation spectrum and vocal tract filter transfer function using WV). Nevertheless, estimation of other acoustic parameters (for instance the spectral density of the excitation noise) appeared difficult, if not impossible.
An acoustic signal representation for speech perception in noise
Authors: Drake Laura A. , Rutledge Janet C.
Published: October 2005

Spectral representations of speech have been used extensively in speech communication applications such as hearing loss compensation algorithms and speech recognizers. Here, a representation incorporation additional perceptual characteristics is created. It is hoped that such a representation would improve the performance of application systems with noisy input by emphasizing more perceptual cues. For example, the shape of the speech signal is modified in hearing loss compensation to fit the residual spectrum of the listener. Information from perceptual cues relating to the interaction of key components can be used to adjust the compensation algorithm to enhance intelligibility. These same cues that are important to human perception may be incorporated into the feature vector to enhance the performance of a speech recognition system such as an HMM or neural network. Results will be presented at the conference.
Precise determination of fundamental frequency by utilizing harmonic structure of speech
Authors: Suzuki Jouji , Seto Miyuki , Shimamura Tetsuya
Published: October 2005

Precise and correct measurement of fundamental frequency F0 is requested in the detailed analysis of speech sound and analysis synthesis telephony employing F0. This paper proposes a new method to estimate F0 employing the harmonic number and its frequency. At first, 256 pts. of speech samples are windowed and ‘‘0’’s are added to 1024 pts. The magnitude spectrum is calculated by FFT and smoothing is carried out by moving average of 5 pts. Such procedures make it easy to identify the spectrum peaks and improve the accuracy of their frequency. Every harmonic is identified from the peak of the spectrum and an harmonic number is assigned. Now fundamental frequency is calculated by (nth) harmonic frequency)/n. Finally the fundamental frequency is computed as the weighted mean, which is proportional to n, on harmonics located under 2 kHz. This idea is quite similar to getting F0 as the coefficient of a regression line passing through the origin. The accuracy of F0 obtained by this method is expected to be less than 3 Hz in female voice when the sampling frequency is 10 kHz. Experimental results indicate that the proposed method gives more precise and reasonable pitch information than the cepstrum or correlation method does.
Acoustic scattering from doubly reinforced elastic plates
Authors: Karali Angela K. , Hayek Sabih I.
Published: October 2005

Almost all known structures are designed for low weight and high strength. For metal structures and especially aircrafts and underwater vehicles this means reinforcements in the form of ribs. This paper presents the development of analytic models that can be used in predicting the acoustic scattered field from such structures, assuming that the geometry is that of an elastic plate of infinite dimensions reinforced with a periodic double array of ribs. The plate is in contact with a homogeneous and isotropic acoustic fluid on one side only. Mindlin’s plate theory is used in modeling the plate and this allows the evaluation of the nonspecular scattered field, generated at the reinforcements, at frequencies below and above the coincidence frequency of the plate. The reinforcements are assumed to exert both forces and moments on the plate and they are included in the mathematical model through their transverse and rotational impedances. A Fourier transform technique is used to evaluate the nonspecularly reflected field due to the periodic double array of ribs. Analytic solutions are obtained and implemented numerically for frequencies ranging from 1 to 15 times the classical coincidence frequency of the plate.
Wavepacket synthesis of far-field echoes from a ribbed cylindrical shell with hemispherical endcaps
Authors: Hughes David H. , Gaumond Charles F. , Houston Brian H.
Published: October 2005

Recent investigations concerning acoustical signal decomposition of Wigner distributions [N. C. Yen, ‘‘Wavepacket decomposition,’’ accepted for publication in J. Acoust. Soc. Am.] have motivated further investigations of echoes from more complex submerged targets. The signal decomposition method of Boudreaux-Bartels [G. F. Boudreaux-Bartel and T. W. Parks, ‘‘Wigner distribution synthesis techniques,’’ IEEE Trans. Acoust. Speech Signal Process. ASSP-34, 442–451 (1986)] is applied to monostatic echoes from a finite, ribbed cylinder with hemispherical endcaps at frequencies below ka=42. The cylinder has a length of 1 m long, a radius of 7.0 cm, and an internal rib spacing of 1 cm. Distinct wavepackets associated with Bloch waves and Bragg waves on the target are synthesized over the suitable range of monostatic target angle.
Speech coding: A review of applications, requirements and standardization
Authors: Dimolitsas Spiros
Published: October 2005

The rapid growth of new telecommunications services such as long-distance, cellular, mobile satellite, and personal communications has been driving the development of speech coding technology. However, for speech coding to be economically viable while offering optimum quality, uniformity is required. This uniformity has traditionally being served by standards, which have typically been developed to address network and/or end-user terminal applications. These two different applications often impose different types of requirements that speech coding systems must meet. For example, for network speech coders, emphasis is often placed on transparency to signaling in addition to speech signals. On the other hand, for end-user applications, low complexity is often a crucial requirement while transparency to nonvoice signals is not. Internationally, speech coding standardization activities are addressed by different fora that can be associated with the two types of applications mentioned above. For network applications, the Telecommunications Standardization sector of the International Telecommunications Union and the ANSI Committee T1 on Telecommunications are active in this area. For end-user applications, the US Government, the International Standards Organization, the ANSI Telecommunications Industry Association, and the European Telecommunications Standards’ Institute are key players.
Robust waveform interpolation for the entire speech signal
Authors: Kleijn W. Bastiaan , Haagen Jesper
Published: October 2005

In waveform interpolation (WI), a speech signal is reconstructed by the concatenation of infinitesimal segments of an evolving characteristic waveform, which is obtained by interpolation over time [W. B. Kleijn, IEEE Trans. Speech Audio Process. 1, 386–399 (1993)]. WI leads to efficient coding of voiced speech, but current implementations switch to CELP for nonperiodic signals. The WI paradigm is extended to provide an effective basis for the coding of voiced and unvoiced speech and background noise. Prototype waveforms are extracted every 2.5 ms. At this high rate the WI analysis–synthesis system results in transparent speech quality. Each prototype waveform is decomposed into a slowly evolving waveform (SEW), obtained by convolution in time with a 40-ms smoothing window, and a remainder, the rapidly evolving waveform (REW). Because of its low bandwidth, a low bit rate suffices for the SEW (additional processing lowers the bit rate further), while the REW requires only a rough statistical description (e.g., its phase spectrum can be randomized). The new paradigm facilitates efficient, robust speech coding in the range 2–8 kb/s.
Porosity, permeability, and shear strength images of the seabed measured by acoustic cross-well tomography experiments
Authors: Yamamoto Tokuo
Published: October 2005

Cross-well tomography of the sedimentary seabeds below man-made islands in Tokyo Bay and Florida Bay was affected by very high attenuation and very high background noise; nevertheless, high resolution (0.2 to 1 m) velocity images were obtained between wells separated by long distances (50 to 250 m). A piezoelectric source in a 60-m-deep well-used long sequences (4095 cycles) of phase-modulated pseudo-random binary codes at high carrier frequencies (up to 10 kHz). A 24-channel hydrophone array in another well received the signal. Inversion of first arrival times by damped least-squares imaged the compressional wave velocities. Assuming the normal consolidation condition, the porosity and shear strength images were predicted from the compressional wave velocity image. The differences in the compressional wave velocity images obtained using different carrier frequencies were used to determine the permeability image of sediments based on the Biot theory. The direct measurements of porosity and shear strength conducted on the cores and the standard penetration test boreholes were used to verify the tomographic predictions, resulting in good agreements. The direct measurements of permeability from pumping tests confirmed the tomographically determined permeability images. [Work supported by KSC and ONR.]
Real-time telemetry of acoustic array data for tomography and other applications
Authors: von der Heydt Keith , Kemp John , Lynch James F. , Chiu Ching-Sang , Miller James
Published: October 2005

Acoustic phased arrays are of great utility in various applications for acoustical oceanography and ocean acoustics, such as tomography and matched-field processing. Internal storage of data from such arrays both limits the total amount of data taken (usually to a few gigabytes) and deprives the user of real-time advantages, such as remote control of the array’s sampling, much greater data capacity, and real-time evaluation and processing. To ensure high bandwidth, radio telemetry to a ship or shore station is required. In August 1992, a collaboration of WHOI and NPS researchers deployed a prototype telemetered vertical line array (VLA) system using a radio link at 902–928 MHz with a one nautical mile effective range. A local area network (LAN) ethernet link was established between the surface buoy computer controlling the array and the shipboard computer ensuring rapid, error-free communications. This and other details of how this technology is implemented, other advantages and disadvantages, and the future of such technology will be discussed. [Work supported by ONR.]
Real-time data acquisition from mid-ocean observatories
Authors: Catipovic Josko
Published: October 2005

Recent advances in underwater acoustic communication have enabled real-time contact with ocean bottom instrumentation. This overview talk summarizes developments in acoustic, radio, and satellite communication links and in undersea networks constructed for telemetry from mid-ocean instrumentation. Acoustic communication has made major advances in terms of energy efficiency, data rate, and robustness. Data from several experiments illustrating capabilities of the acoustic modems, algorithm structure of the communication link, and its relationship to the characteristics of the acoustic channel are presented. Technology resulting from this work is incorporated in a line of underwater modems that are being used extensively by the oceanographic community. The modems are of course useful as direct point-to-point communication devices; however, incorporating modems into communication networks greatly increases their utility. Several deployed networks are discussed, in particular an under-ice system recently develoyed on Lake Winnipesaukee and in the Arctic off Pt. Barrow. The under-ice network is a part of the Autonomous Oceanographic Sampling Network (AOSN), a long-term development program aimed at improving the timeliness and cost-effectiveness of oceanographic data acquisition.
High-resolution sonar volume scattering and attenuation measurements in marine sediment
Authors: LeBlanc Lester R. , Schock Steven G. , DeBruin Darryl L. , Munro Lachlan I.
Published: October 2005

This paper examines the use of normal incidence high-resolution sonar for seafloor sediment classification. Volume scattering, a measure of the scattering strength per cubic meter, and attenuation measurements taken at several sites are compared to the acoustic characteristics of cored sediments. A relaxation time model utilizes the shift in the center frequency of acoustic reflections from within the sediment to measure attenuation. Using a statistical model to detect sediment transition layers, volume scattering and attenuation values are calculated between detected layers. The scattered energy measurements reveal two separate probability density functions. The probability density of scattering from layer boundaries is characterized by a Rayleigh density function whereas volume back scattering is characterized by an exponential density function. When the measured volume scattering coefficient is compared to sediment properties, generally, sediments with large grain components produce larger volume scattering coefficients. In comparing attenuation measurements with sediment types, sands attenuate acoustic energy more rapidly than clays and silts. By combining sediment volume scattering measurements, attenuation measurements, and detected layer impedance profiles, it is expected that insitu acoustic classification of marine sediments will become more viable. [Work supported by ONR.]
The measurement of sound speeds in the surface wave zone
Authors: Melville W. Kendall , Terrill Eric , Horn Ronald C.
Published: October 2005

Sound-speed profiles in the surface wave zone are of direct interest for assessing the role of surface reverberation in many acoustic models. The sound-speed profiles may also be used in acoustical oceanography applications related to wave breaking, bubble distributions, and air–sea fluxes. Earlier instruments for the direct measurement of the sound-speed profile in the wave zone [e.g., Lamarre and Melville, J. Acoust. Soc. Am. 95, 1317–1328 (1994)] could be deployed for only limited periods. In this paper an autonomous surface buoy is described which is designed to measure sound speed, temperature, and conductivity, on a deep-water mooring for a period of months. The design of the instrument is based on the tethered version of Lamarre and Melville but is extended to include on-board data processing and storage of sound speeds over the first 7 m of the water column. Samples of processed data, buoy location, and other ‘‘housekeeping’’ data are recovered via an Argos satellite link. Preliminary results from the initial deployment in the North Atlantic during the 1993–94 winter will be presented. [Work supported by ONR.]
Artificial bubble cloud targets for acoustic remote sensing of the ocean
Authors: Hwang Paul A. , Roy Ron A. , Crum Lawrence A.
Published: October 2005

Theoretical calculations indicate that bubble clouds with void fraction of O(1%) and size of O(20 cm) produce large scatter due to collective oscillations. It is suggested that artificial bubble cloud targets can be used for acoustical remote sensing of ocean environment. The flow following property of such targets is excellent because the target density is quasineutral buoyant. The void fraction, bubble size distribution, and cloud geometry can be controlled. A technique using polymer solutions to stabilize bubble clouds was developed. Resonant tube measurements indicated that the acoustic velocity and attenuation of artificial clouds are similar to bubble clouds in water. This technique can be applied to the designs of underwater acoustic targets for sonar calibration, simulation of wave breaking, and neutral buoyant tracers for tracking ocean currents. The use of artificial bubble clouds targets is specially valuable for low wind condition and applications in deeper regions where bubbles generated at the surface rarely survive. Larger bubbles can also be transported into deeper regions by artificial clouds. [Work supported by ARPA.]
A broadband signal processor for acoustic imaging using ambient noise
Authors: Glegg Stewart A. L. , Oliveri Marc
Published: October 2005

The concept of acoustic imaging using ambient noise was introduced by Buckingham etal. [Nature 356, 327 (1992)] as a method for passively detecting objects in the ocean. It was suggested that two-dimensional pictorial images could be obtained using this approach, and it has been shown by Potter [J. Acoust. Soc. Am. 95, 21 (1994)] that at least 900 pixels are required to show the details of a spherical object in an underwater sound channel. This paper will describe an imaging technique that makes use of the cross correlation between an array of sensors to generate an image with over 1000 pixels. The difference between this and other approaches is that broadband signal processing is utilized, which optimizes the use of the available sound energy scattered by the object, and eliminates many of the problems associated with spatial aliasing inherent to narrow-band analysis. Consequently images can be obtained using a relatively small number of sensors. This method has been evaluated using a simple experiment in air, scaled to be equivalent to an ocean application, using loudspeakers to generate the ambient noise field. Results will be presented that show the images of various objects. [Work supported by ONR.]
Adaptive processing of communication signals in shallow water acoustic telemetry networks
Authors: Zvonar Zoran , Stojanovic Milica , Bowlin James B.
Published: October 2005

Underwater acoustic communications have received much attention in recent years leading to the development of powerful and reliable receiver algorithms for signal processing in a variety of ocean environments. With the feasibility of high-speed coherent underwater communications established, the focus of current research is shifting to more demanding communication scenarios, as encountered in underwater acoustic networks. The problem of signal detection in highly variant shallow water environments, in the presence of strong cochannel interference from other acoustic modems in the network is addressed. The signals of different users occupy the same bandwidth, making the respective channel responses and the underlying data symbols sequences the only distinction among distorted replicas observed at the receiver. The performance of a single-sensor decision-feedback equalizer (DFE) is compared to the performance of a centralized multiuser receiver that jointly performs adaptive equalization and interference cancellation, and a decentralized multi-sensor DFE that performs spatial signal combining and multichannel equalization. Performance at different receiver structures was tested on real data collected at a one-mile range in the Woods Hole harbor, from two closely separated sources. Excellent results were obtained at signal-to-interference ratios as low as −10 dB, showing the performance improvement of proposed techniques over DFE in a network scenario.
Performance of a high-rate communication link on a shallow water acoustic channel
Authors: Stojanovic Milica , Bowlin James B. , Catipovic Josko A. , Proakis John G.
Published: October 2005

Latest achievements in underwater acoustic communications are leading the way to modern instrumentation for acoustical oceanography by eliminating the need for tethering or diver’s contact with submerged sensors. To meet the various telemetry needs, an acoustic link must provide both reliability and a high rate of data transfer, requirements that are often conflicting. The first approximation of an acoustic channel is that of a time-invariant multipath communication channel. It is well known that by increasing the transmission rate on such a channel, multipath propagation causes longer intersymbol interference, ultimately limiting system performance. Recent experimental results obtained with a newly developed receiver algorithm, however, showed a seemingly surprising result: an increase in transmission rate resulted in improved system performance. An explanation for this phenomenon is found in the time variation of the ocean multipath. Notably emphasized in the fully saturated shallow water channels, higher transmission rates allow finer sampling of the rapidly varying channel, and therefore better tracking. Experimental results obtained in the Woods Hole harbor are presented showing a dramatic improvement in performance of QPSK coherent detection over a one-mile range as the data rate is increased from 5 to 30 kilobits-per-second. A theoretical analysis based on stochastic channel modeling supports these observations.
High Q, low-temperature oscillators: Some examples
Authors: Parpia Jeevak M.
Published: October 2005

High Q audio frequency mechanical oscillators have been used in a number of experimental investigations of superfluidity [V. Kotsubo etal., Phys. Rev. Lett. 58, 804–807 (1987)] as well as flux lattice melting in superconductors at low temperatures [P. L. Gammel etal., Phys. Rev. B 40, 7354–7356 (1989)]. The design, construction, and use of mechanical devices fabricated from metal and silicon will be presented. In particular, single-crystal silicon oscillators [R. E. Mihailovich etal., Phys. Rev. Lett. 68, 3052–3055 (1992)] offer the potential for the observation of very low dissipation phenomena, with energy sensitivities approaching 10−24 J/cycle. [Work supported by the NSF through Grant No. DMR-9123857.]
High Q and cryogenic resonant gravitational wave detectors
Authors: Johnson Warren W.
Published: October 2005

Gravitational wave detectors, aimed at detecting the momentary gravitational perturbation that might come from the gravitational collapse of a star, share almost every attribute of a very sensitive acoustic detector, except the requirement that they be completely decoupled from any true acoustic source. There are now four multiton cryogenic resonant detectors around the world that have operated with burst amplitude sensitivity in the attometer (10−18-m) range, including the ALLEGRO detector at LSU. Even at liquid helium temperature, one possible fundamental noise source is the Langevin force, whose spectral density is determined by the mechanical Q of the relevant antenna resonances. Two practical materials, aluminum–magnesium alloy and high-purity niobium, have proven to have Q’s greater than 10 million under some conditions. Now that many other noise sources have been eliminated, further improvements in sensitivity will come from further increases in Q.
High Q piezoelectric resonators
Authors: Ballato Arthur , Gualtieri John G. , Kosinski John A.
Published: October 2005

Piezoelectric crystal resonators for frequency control and selection are characterized primarily by the values of their acoustic quality factor, Q. Depending on the application, other measures, such as the electromechanical coupling factor are also used. None of the usual performance figures is entirely adequate for characterizing resonators for applications at high frequencies. This paper describes various metrics traditionally applied to resonators, such as acoustic Q, coupling factor, and equivalent circuit parameters. It tabulates observed values for older materials like quartz and lithium niobate and newer substances such as langasite and thin-film aluminum nitride. It then introduces a new metric for judging material suitability for high-frequency resonator use, and tabulates these values. The new measure combines all physical quantities associated with acoustic wave propagation in piezocrystals: mass density, dielectric permittivity, piezoelectric modulus, elastic stiffness, and viscosity. Its use leads to a number of unobvious constraints on the future technology.
Acoustic viscometer for measurements in gases
Authors: Mehl James B. , Gillis Keith A. , Boyes Stephen J. , Moldover Michael R.
Published: October 2005

Martin Greenspan and Francis N. Wimenitz [‘‘An Acoustic Viscometer for Gases,’’ NBS Rep. 2658 (1953)] proposed the use of double Helmholtz resonators for measurement of the viscous diffusivity of gases. Such resonators, consisting of two cavities coupled by a duct, have a low-frequency mode where the gas oscillates between the cavities. A theoretical model that includes the viscous and thermal losses in the duct, thermal losses in the cavities, duct-end corrections, and fill ducts, have been developed. The resonator dimensions can be chosen so that the damping of the lowest mode is primarily due to viscous losses at the duct walls, and thermal losses at the cavity walls are very small. Several resonators have been constructed, including all-metal and mainly glass designs. Measurements on argon, helium, and propane samples are in very good agreement with the theoretical model.
Acoustical observations of anthropogenic discharge plumes
Authors: Proni John R.
Published: October 2005

Acoustical methods such as backscatter and Doppler current measurements are gaining substantial acceptance in both point source and nonpoint source oceanic pollution studies. Using acoustical backscattered energy, chemical and biological samplers are guided to oceanic water column regions of higher contaminant concentration. In typical discharge operations, e.g., dredged material, sewage effluents, oil-well blowouts, after-discharge backscatter levels are initially at least three to four orders of magnitude above normal background backscatter levels and are available for tracking of plumes and guidance of sampling operations. Additionally, acoustical backscatter may also be used to map naturally occurring surfaces within the oceanic water column along which both discharged material and naturally recurring particulates may accumulate. Thus nonacoustical sampling may continue to be guided indirectly far beyond the time or distance at which plume concentration levels may be considered to have diminished to average water column background levels. Furthermore, such surfaces may be loci of increased probability of exposure to contaminants for plankton, larvae, and juvenile fish and as such should be closely studied. Acoustical detection of incursions of discharged material along sub-oceanic surfaces for both negatively and positively buoyant plumes will be illustrated using sewage sludge discharges and secondarily treated wastewater effluent discharges. Preferential accumulation surfaces present: (1) at the boundary region of Hudson/Raritan River water and continental shelf water and (2) within Chesapeake Bay, detected through backscatter and transverse Doppler, measurements will be shown.
Scattering and attenuation of elastic wave by a viscoelastic cylinder. I. P-wave incident
Authors: Liu Yin-Bin , Wu Ru-Shan
Published: October 2005

The analytical expressions for scattered and transmitted fields from a circular elastic cylinder with and without intrinsic attenuation caused by a normally incident plane P wave or a normally incident cylindrical P wave are derived. The corresponding acoustic cases are also given by letting the shear modulus either inside or outside the cylinder be equal to zero. In the Rayleigh wave scattering regime, the solution agrees with standard approximate solutions. Resonance scattering, radiation pattern, scattering cross section, and synthetic seismogram are calculated. For a given mode number m in some discrete frequencies, either scattered compressional or scattered shear waves vanish for a nonabsorbing cylinder, and the amplitudes of scattered compressional and shear waves are always nonzero for an absorbing cylinder. The sawtooth shapes of a scattering cross section that appear in a low-velocity nonabsorbing cylinder will disappear in the absorbing cylinder. The scattered fields are the superposition of the geometrically reflected waves and a series of surface waves that circumnavigate the cylinder azimuthally, the amplitudes of scattered compressional waves are much greater than those of scattered shear waves (ka≫1). [Work supported by NSF.]
Comparison of membrane wave coupling coefficients
Authors: Rebinsky Douglas A.
Published: October 2005

Comparisons are made between various estimates of the coupling coefficients that characterize the longitudinal wave contribution to acoustic scattering from thin elastic circular cylindrical and spherical shells. A common method of obtaining the coupling coefficient is through the application of the Sommerfeld–Watson transformation to the exact modal series solution. From this result, a high-frequency approximation can be derived that accurately describes the magnitude, but it does not adequately represent the phase. These are compared to coupling coefficients determined using a ray-theoretic approach. This representation describes both the magnitude and the phase well when compared to that obtained from the exact modal series result using the Sommerfeld–Watson transformation. [Work supported by ONR.]
Sound scattering by bulkheads in cylindrical shells
Authors: Guo Y. P.
Published: October 2005

This paper studies sound scattering from cylindrical shells with internal bulkheads, modeled as circular elastic plates transversely loaded onto the shell. Fourier transformation and angular harmonics decomposition are used to derive solutions in terms of coupling forces and bending moments at the attachment, which themselves are determined by the dynamic balance between the shell and the internal plate, and hence, characterize the sound/shell/plate interactions. These forces and moments are discussed to identify dominant features in the scattering process, and the scattered acoustic field, both monostatic and bistatic, are examined to reveal the characteristics of the scattering. Also discussed are the effects of different attachment conditions on the far-field scattering and the total acoustic power scattered by the bulkhead.
Scattered field distributions from the near to the far field using an acoustic lens
Authors: McLeary Ross , Whitcher Russell E. , MacGillivray Ian R.
Published: October 2005

An experimental technique for measuring the acoustic scattering from bodies over a range of frequencies from approximately 100 kHz to 3 MHz is described. A compact arrangement comprising a large-area piezoelectric transmitter in conjunction with an acoustic lens allows scattering measurements to be made from the near to the far field in a laboratory water tank. An approximation to plane-wave or point-source insonification is achieved by the selection of an appropriate single or compound lens, respectively. Experimental results are compared with predictions for the scattering from a cylinder insonified normal to its axis. Results are also presented for monostatic scattering from a more complex body as a function of aspect angle. The technique has been found to be useful for the validation of computer or other models of scattering and for scale-model measurements.
Normal mode extraction technique in a viscous-damped linear dynamic system
Authors: Hashimoto Norihisa
Published: October 2005

A new method for normal mode identification in a viscous-damped linear dynamic system such as sound and vibration is presented in this paper. The present study reveals that the simple procedure for the transfer function of this system can automatically extract the mode shapes, and this method does not have the complex calculation and assumption features, which makes it different from the conventional experimental modal analysis. Through this experimental investigation, it is confirmed that this method can be utilized in practice when the transfer function is obtained accurately by using the adequate noise elimination technique.
Analysis of digitized waveforms using Shannon entropy. II. High-speed algorithms based on Green’s functions
Authors: Hughes Michael S.
Published: October 2005

An improved approach for calculating continuous waveform entropy Hc is described. This approach is based on the use of the Green’s function for the one-dimensional differential operator L[u]=u″ with boundary conditions u(1)=u(0)=0. The method is applied to data acquired from a Plexiglas block having artificial defects in order to produce an entropy image. The defect contrast obtained with the new Green’s function computation is then compared with that obtained from a previously described Fourier series approach. It is found that the new approach produces the same or higher image contrast in a time that is roughly three orders of magnitude smaller than that required by the Fourier series method. This speedup arises from two sources. The Green’s function approach has greater inherent immunity to noise, and the number of calculations required in the Green’s function approach is much smaller than the number required in the Fourier series approach.
New data on the short-term effects of tone exposure
Authors: Botte M. C. , Mönikheim S.
Published: October 2005

Three experiments studied the short-term effects of moderate- and high-level tone exposures on threshold and loudness. Experiment 1 measured temporary threshold and loudness shifts (TTS and TLS) due to a 1000-Hz exposure at a moderate level (65 dB SPL) as a function of exposure duration. Whereas there was no significant TTS, TLS was as high as 11 dB and recovered rapidly. TLS was significantly larger for the highest test levels. TTS and TLS were practically independent of exposure duration. Experiment 2 measured TTS and TLS as a function of test frequency following exposures to 500-, 1000-, or 3000-Hz tones at 65 dB SPL. The frequency patterns of TLS were established at test levels of 20–60 phons. TLS and TTS had similar frequency patterns with a maximum at the exposure frequency, but TLS patterns were higher and more extended than those of TTS. In experiment 3, subjects were exposed to a 1000-Hz tone at a high level (90 dB SPL). Frequency patterns for TTS or TLS at a 60-dB test level were measured repeatedly during 5 1/2 min after the end of exposure. Two short-term effects were observed immediately after the exposure: (1) a slight TTS (4.5 dB) at the exposure frequency which disappeared 2 to 3 min later; and (2) a large TLS (9 dB) which extended over the whole range of test frequencies, recovered rapidly, and had disappeared by 330 s after exposure. Concomitant with these short-term effects, an additional long-term TTS affected the test frequencies above the exposure frequency and reached a maximum of 17 dB at 150-s post-exposure, after which it began to recover relatively slowly. Thus, as well as causing ‘‘classical’’ long-term fatigue, high-level exposure also results in marked short-term effects that strongly reduce loudness over a wide frequency range.
Effects of frequency and amplitude modulation on the pitch of a complex tone with a mistuned harmonic
Authors: Darwin C. J. , Ciocca Valter , Sandell G. J.
Published: October 2005

It has previously been found that when a single low-numbered harmonic of a complex tone is progressively mistuned, for mistunings up to about 3%, the pitch of the complex changes in the direction of the mistuning but for larger mistunings (by about 8%) the pitch returns to its original value. This result is compatible with the operation of a mechanism such as a graded harmonic sieve, which can reject from the calculation of pitch those frequency components that are implausibly distant from a harmonic frequency. The first experiment shows that the tolerance of such a sieve is increased when all the components of the complex tone (including the mistuned component) share a common pattern of frequency modulation at a rate of 6 Hz. The second experiment shows that the tolerance of the sieve is not increased when the components share a common pattern of amplitude modulation at 17 Hz. The third experiment replicates these findings and further shows that the increase in sieve tolerance for FM, but not for AM, occurs at both 6 and at 17 Hz.
Detection and recognition of repeated tones and tonal patterns
Authors: McKinley Robert C. , Weber Daniel L.
Published: October 2005

This study examined the effect of multiple presentations on signal detection and on signal recognition (identifying one signal from a set of four possible signals) for three different sets of signals. One set was four sinusoids (100-ms duration, frequencies of 707, 1000, 1414, and 2000 Hz). Two sets contained tonal patterns each made of a sequence of seven, 100-ms, sinusoidal components. In the first set, the four patterns consisted of the same seven sinusoids in different orders. In the second set, the four patterns had the same order of relative frequencies, but had frequencies from different 1/4 oct bands centered at 707, 1000, 1414, and 2000 Hz. All stimuli were adjusted to be equally detectable in the presence of a continuous white noise (η0=20 dB SPL). Each trial contained 1, 2, 4, 8, or 16 presentations of a given signal plus noise (probability of a signal was 0.5) or noise alone. Detectability of the sinusoids generally increased as the square root of the number of presentations; detectability for the tonal patterns increased at a slower rate. Recognition was generally poorer than predicted by the recognition theorem [S. J. Starr, C. E. Metz, L. B. Lusted, and D. J. Goodenough, Radiology 116, 533–538 (1975)] and increased with multiple presentations only as much as expected from the increase in signal detectability.
The effect of conductive hearing loss on the masking-level difference: Insert versus standard earphones
Authors: Hall Joseph W. , Grose John H.
Published: October 2005

Previous studies have indicated that the masking-level difference is reduced in listeners having conductive hearing loss, even when sensation level is controlled. According to the crosstalk hypothesis, this deficit can be explained in terms of reduced binaural cues at the level of the cochleas, due to acoustic contributions from the bone conduction route. It is possible that such a deficit can be overcome through the use of insert earphones, because such transducers provide a high degree of interaural attenuation. In this study, NoSo and NoSπ thresholds were measured in normal-hearing subjects and in subjects having hearing loss due to otosclerosis, using both standard TDH earphones encased in NAF-48490-1 circumaural cushions and deeply seated Etymotic 3A insert earphones. Masking noise was presented both at a fixed sound-pressure level and at a fixed sensation level. The results did not indicate consistently larger MLDs with insert earphones for either group of subjects, for either type of presentation level. The MLDs of the hearing-impaired subjects often remained abnormally small even when the insert transducers were used and the noise was presented at a fixed SL. Whereas acoustic crosstalk appeared to have a role in the results obtained, other factors (perhaps related to long-term hearing threshold asymmetry and/or interaural phase mismatch) also were apparently involved.
Effect of reducing slow temporal modulations on speech reception
Authors: Drullman Rob , Festen Joost M. , Plomp Reinier
Published: October 2005

The effect of reducing low-frequency modulations in the temporal envelope on the speech-reception threshold (SRT) for sentences in noise and on phoneme identification was investigated. For this purpose, speech was split up into a series of frequency bands (1/4, 1/2, or 1 oct wide) and the amplitude envelope for each band was high-pass filtered at cutoff frequencies of 1, 2, 4, 8, 16, 32, 64, or 128 Hz, or ∞ (completely flattened). Results for 42 normal-hearing listeners show: (1) A clear reduction in sentence intelligibility with narrow-band processing for cutoff frequencies above 64 Hz; and (2) no reduction of sentence intelligibility when only amplitude variations below 4 Hz are reduced. Based on the modulation transfer function of some conditions, it is concluded that fast multichannel dynamic compression leads to an insignificant change in masked SRT. Combining these results with previous data on low-pass envelope filtering (temporal smearing) [Drullman et al., J. Acoust. Soc. Am. 95, 1053–1064 (1994)] shows that at 8–10 Hz the temporal modulation spectrum is divided into two equally important parts. Vowel and consonant identification with nonsense syllables were studied for cutoff frequencies of 2, 8, 32, 128 Hz, and ∞, processed in 1/4-oct bands. Results for 12 subjects indicate that, just as for low-pass envelope filtering, consonants are more affected than vowels. Errors in vowel identification mainly consist of reduced recognition of diphthongs and of durational confusions. For the consonants there are no clear confusion patterns, but stops appear to suffer least. In most cases, the responses tend to fall into the correct category (stop, fricative, or vowel-like).