bluuebunny/crossref_metadata_2025 huggingface dataset:
Filter shapes for brief signals as a function of preceding stimulation
Authors: Strickland Elizabeth A.
Published: July 2002

The threshold for a brief probe tone in a broadband noise was measured as a function of probe frequency (1 or 4 kHz) and masker level (0–40 dB spectrum level). The probe was always presented 2 ms after the onset of the masker, and the masker was preceded either by silence, or by a precursor with the same spectrum and level as the masker. For each subject there were some conditions in which probe threshold was higher with no precursor than with a precursor (overshoot), and some conditions in which probe threshold did not differ with the presence of a precursor. Filter shapes for the same brief probes were then measured using the notched noise method. The probe was always presented with a 2-ms delay from the onset of the notched-noise masker. The masker was preceded by silence or by a precursor. The probe level was set 10 dB above the threshold measured for a given condition in the overshoot experiment. A comparison will be made of filter shape and the presence or absence of overshoot. [Work supported by NINCDS.]
Physiological correlates of the ‘‘mean-term’’ auditory adaptation
Authors: Bibikov N. G.
Published: July 2002

In psychophysics, adaptation is defined usually as a decline of loudness of steady stimulus as a function of time. It occurs over periods ranging from seconds to minutes. Recent psychophysical data show that during a ‘‘mean-term’’ adaptation period (the first 30 seconds of the sound) the detection of amplitude changes considerably improves. Physiological correlates of ‘‘mean-term’’ adaptation in single units located in different regions (cochlear nucleus, superior olive, torus semicircularis) of the auditory system of the immobilized frog Rana temporaria are explored. Driven responses to long-duration pure tones and AM tones decreased over time periods ranging from milliseconds to minutes. The adaptation increased at the upper levels of the auditory system. The response to pure tone stimuli declined more rapidly than the response to AM tones. For AM stimuli the adaptation increased as the modulation depth decreased. The considerable enhancement of the phase locking to the sound envelope could be observed during the first 10–15 s of the AM tones with low modulation indexes. The effect of phase-locking enhancement during the mean-term adaptation matched some psychoacoustical findings. [Work supported by the RFBR.]
Waveguide absorbers for structural damping
Authors: Ungar Eric
Published: July 2002

For a series of experimental measurements, Manfred Heckl attached strips of damping tape extending outward from the plate edges to increase the plate’s damping. He understood that vibrations induced in the tapes would extract vibrational energy from the plates, but did not explore their effects in detail. The present paper presents an analysis of the damping effects that attached wave-bearing systems can produce and illustrates realizations of waveguide absorbers in the shapes of tapered rods and beams. It also presents related experimental data and describes the general potential of waveguide absorbers.
Concentrated lateral excitation of structures
Authors: Petersson Bjorn A. T.
Published: July 2002

Supplementing a recent study, aimed at clarifying theoretical requirements regarding models for prediction and design of transmission characteristics for built-up structures [Petersson and Heckl (1996)], vibration transmission relating to lateral force and moment excitation has been investigated for plates and beams of arbitrary depths. Through the associated mobilities, the power transmission and the cross-coupling between lateral translational and lateral rotational components of excitation and motion are studied and the influence of the volumetric nearfield is examined. Complete descriptions are developed for the two associated point-cross mobility elements. From the theoretical results it can be concluded that reciprocity with respect to the point-cross mobility elements is strictly only valid in the hypothetical case of point excitation. The numerical results demonstrate, however, that reciprocity can be invoked in an approximate sense also for realistic excitation conditions. It is shown that a reducttion of a lateral force excitation, remote from the neutral layer, to a combined force and moment excitation at the neutral layer is not applicable except for very small Helmholtz numbers. It is also concluded that the volumetric nearfield affects all mobility elements. Finally, some manageable estimation procedures for the mobilities will be proposed.
Stop-consonant production by dysarthric speakers: Use of models to interpret acoustic data
Authors: Poort Kelly L.
Published: July 2002

Acoustic measurements have been made on stop consonants produced by several normal and dysarthric speakers. The acoustic data were previously recorded by H. P. Chang and H. Chen at MIT. In the present study, various aspects of production following release of the oral closure were quantified through the use of acoustic measures such as spectra and durations of noise bursts and aspiration noise, as well as shifts in frequencies of spectral prominences. Through comparison of these measurements from the normal and dysarthric speech, and based upon models of stop-consonant production, inferences were drawn regarding articulator placement, rate of articulator release, tongue-body movements, and vocal-fold state. The dysarthric speakers deviated from normals particularly with respect to alveolar constriction location, rate of release, and tongue-body movement into the following vowel. For example, the lowest front-cavity resonance in the burst spectrum of an alveolar stop is normally in the range 3500–5500 Hz. For three of eight dysarthric speakers, the range was lowered to 1500–2800 Hz, indicating either placement of the tongue tip further back on the palate or formation of the constriction with the tongue body in a location similar to that of a velar stop. [Work supported in part by NIH Grant DC00075.]
The effects of utterance length on temporal control in aphasia
Authors: Baum Shari R.
Published: July 2002

By comparing the same syllables produced in a variety of utterance-length conditions, the present investigation examined the hypothesis that fluent aphasics exhibit a shorter-than-normal speech planning domain, whereas nonfluent aphasic patients exhibit impaired speech timing at all levels. Subjects included nonfluent aphasics, fluent aphasics, RHD controls, and normal controls. The base stimuli were 10 monosyllabic nouns from which 10 bisyllabic words were derived. Each of the 20 words also appeared medially in the context of a short (4–6 syllable) and a long (8–10 syllable) sentence. Productions were elicited two times each and recorded for acoustic analysis. Durations of the target monosyllable as a proportion of whole utterance durations were computed. Preliminary analyses revealed that normal speakers produced target syllables with shorter durations in 2-syllable relative to 1-syllable words in both long and short utterances. Target syllables were shorter in the long sentences, as expected. A similar pattern was found for the other speaker gruops, but the magnitude of the differences was larger in the fluent and nonfluent aphasic patients productions. Moreover, the nonfluent speakers’ proportional durations were aberrant due to abnormally long sentence durations. Results are discussed in rlation to theories of temporal control in brain-damaged patients. [Work supported by MRC.]
An interactive construction system of three-dimensional vocal tract shapes from tomogram
Authors: Yokoyama Tohru , Miki Nobuhiro , Ogawa Yoshihiko
Published: July 2002

A new interactive system is developed to construct the 3-D vocal tract shape from tomogram, for the purpose of understanding the relationship between the speech production process and the geometrical figure of the human vocal tract. The shaded 3-D vocal tract shape can be seen with this system from any direction, by construction from a set of 2-D images measured by, for example, MRI or x-ray CT, and basic geometrical information is obtained for estimating the area function and transfer function, or creating the mesh model for 3-D FEM. For constructing a 3-D vocal tract shape, the following three operations are conducted for the system. First, the 3-D locations of the tomogram images are interactively determined. Second, the contours of the air tract in the images are extracted after several image processings. Third, the wire frame model of the vocal tract, including a branching case, is constructed computationally from the extracted contours. The system is developed using the C++ language as an object-oriented programming language, Motif toolkit as GUI, and PEX as 3-D graphic libraries. The proposed system provides a convenient tool for speech research on the X-Window environment.
Perceptually motivated modeling of noise in pathological voices
Authors: Gabelman Brian , Kreiman Jody , Gerratt Bruce R. , Antonanzas-Barroso Norma , Alwan Abeer
Published: July 2002

Research investigating the correlation of acoustic measures of noise and the perception of pathological voice quality has consistently demonstrated a moderate association. However, this correlational approach cannot address basic questions concerning their cause and effect relationship, such as how carefully noise must be modeled to preserve the perceived pathological vocal quality, and what particular aspects of the noise are necessary to preserve natural vocal quality. To address these questions, an analysis by synthesis approach was applied to a large sample of natural pathological vowels varying in severity of pathology. Several synthetic versions of each voice were created using an LF source model and a formant synthesizer. Amplitude and spectral characteristics of the noise were varied as necessary. Stimuli were compared perceptually to the original voice and to each other. A discussion will focus on the procedures necessary to adequately model noise in voices with severe vocal pathology, and on the perceptual importance of differences in how vocal noise is synthesized. [Research supported by NIH.]
Mechanisms underlying recovery from forward masking in electrical stimulation
Authors: Chatterjee Monita
Published: July 2002

In normal hearing, adaptation is hypothesized to be closely related to recovery from masking. In electrical stimulation, adaptation at the inner hair cell synapse is absent. However, refractory mechanisms can result in an adaptationlike decrement at high stimulation rates. Shannon [ J. Acoust. Soc. Am. 88, 741–744 (1990)] found that normalized psychophysical forward masking functions were similiar in normal and electric stimulation, and suggested that the mechanisms underlying forward masking are primarily retro-cochlear. Recovery from forward masking was studied in five Nucleus-22 cochlear implant users. Results were found to be strongly subject-dependent. In general, a rapid decay in the first 15 ms after masker offset, and a slow decay persisting up to 200 ms after masker offset, were observed. Under some conditions, plateaus and nonmonotonicities were observed in the region of transition between the two processes. Recovery was independent of masker level. Masker duration, pulse rate, and masker-probe spatial separation strongly influenced the relative weight of the rapid component. The amount of masking near masker offset did not determine the time course of recovery. It is hypothesized that the rapid mechanism is due to auditory nerve refractoriness, while the slow mechanism is likely to be more central in origin. [Work supported by NIDCD.]
Neurophysiological and psychophysical measures of duration discrimination in normal-hearing adults and adults with cochlear implants
Authors: Trautwein Patricia G. , Ponton Curtis W. , Kwong Betty , Waring Michael D.
Published: July 2002

The ability to discriminate duration contrasts (a first approximation of voice onset time differences) was evaluated in adult cochlear implant users and normal-hearing adults using a neurophysiological response (the mismatch negativity—MMN). The MMN is an evoked response generated by a deviant stimulus when embedded in a series of standard homogeneous stimuli. Magnitude of the MMN correlates with psychophysical discrimination thresholds indicating that the MMN provides a measure of perceived acoustic change [Kraus et al. (1996)]. The MMN thus provides a neurophysiological correlate of acoustic discrimination. The present study compared an objective statistical measure of MMN magnitude to psychophysical measures in response to duration differences of brief auditory stimuli. A ten-click train was paired with shorter trains to create an oddball sequence. The selected duration differences were similar to voice onset time differences that distinguish voiced from voiceless speech phonemes. The MMN magnitude was measured at four scalp-electrode locations. The relationship between neurophysiological and psychophysical thresholds was determined and comparisons were made between the normal-hearing adults and adults with cochlear implants. While some similarities were found, the results suggest that the sensitivity to duration differences differs between normal-hearing and cochlear implant subjects.
Intensity discrimination of ramped and damped tones
Authors: Schlauch Robert S. , Ries Dennis T. , DiGiovanni Jeffrey J. , Elliot Sara , Campbell Shari L.
Published: July 2002

Intensity discrimination of 1.0-kHz tones that had exponentially ramped or damped envelopes was measured as a function of duration and level. Standard levels ranged from near threshold to 90 dB SPL and durations ranged from 10 to 200 ms. Just-noticeable differences for some subjects were smaller for damped tones than for ramped tones for the same duration and level. For these conditions, damped tones showed more temporal integration than ramped tones. This result is somewhat surprising given that the subjective duration of ramped tones increased by a factor of 100 for stimuli ranging from 10 to 200 ms, whereas the subjective duration of damped tones increased by a much smaller factor of 28 over the same range of durations. Possible physiological and cognitive explanations for these differences will be discussed. [Work supported by NIH-NIDCD R29 DC01542.]
Preferential detection of rising versus falling intensity
Authors: Neuhoff John G. , Wess Jennifer
Published: July 2002

An increase in intensity caused by the approach of a sound source can signal an ecological threat or opportunity, and therefore a potentially important event. A receding source produces a pattern of decreasing intensity change and a potentially less important event. Therefore preferential responding by an organism to rising intensity stimuli may provide a selective advantage. To investigate the hypothesis that listeners respond preferentially to rising intensity stimuli, listeners were presented with white-noise samples that either rose or fell dynamically 20 dB (SPL) in intensity. Stimulus duration ranged from 0.5 s to 1.5 s, and onset of intensity change ranged from 0 s to 1 s. In a speeded 2AFC task listeners had to determine whether intensity rose or fell. Results show that listeners detected increases in intensity more quickly and more accurately than equivalent decreases. The findings are consistent with the interpretation that the auditory system processes some acoustic events according to a set of ecologically defined priorities.
The perception of three tone patterns by Indian and American listeners
Authors: Iyer Nandini , Hoglund Evelyn M. , Jones Mari Reiss
Published: July 2002

Serial identification of three tone patterns by Indian and American listeners was investigated in three experiments. Six pitch patterns combined a high-frequency (H) and a low-frequency tone (L) (e.g., H, H, L or L, L, H). The first experiment presented all patterns at each of three different tempi (fast, medium, and slow trial blocks) with blocks presented in either ascending or descending order of tempi. Each pattern occurred six times (randomly) within a block. The second experiment investigated the differences in pattern perception between musicians and nonmusicians among the previously tested Indian and American listeners. In the third experiment, frequency separation between the high and low tones was investigated at two different tempi. A repeating cycle of three tone patterns was presented for identification and streaming thresholds were determined for Indian and American listeners. Preliminary results indicate that the two groups of listeners differ in their ability to identify three tone patterns. These differences were affected by the listeners’ musical background. Frequency separation and tempo also differentially affected pattern perception among the two groups of listeners.
Temporal correlates of foot-level timing in English and Japanese
Authors: Tajima Keiichi , Port Robert F.
Published: July 2002

Languages of the world are felt to be spoken with different kinds of rhythm. But what is the basis for this impression? Notions like ‘‘stress-timed’’ and ‘‘syllable-timed’’ have been difficult to verify experimentally because rhythm in spoken language rarely yields consistent temporal regularities. Experiments were conducted in which native English and Japanese speakers repeated phrases using a specified rhythm in time with an auditory metronome. Results demonstrate that temporal correlates exist for prosodic units approximately the size of the ‘‘foot,’’ consisting of one relatively strong syllable followed by zero or more weak syllables. Strong (foot-initial) syllables show greater temporal stability than do weak (foot-internal) syllables when the timing of syllables was perturbed by manipulating inherent segmental durations. This is found for both English and Japanese. However, in English, increasing the foot size by the addition of weak syllables is accompanied by compensatory shortening only within the foot, while in Japanese, compensation occurs both within and between adjacent feet, suggesting a weaker role for this unit. By eliciting speech in a rhythmic context, such as ‘‘speech cycling,’’ both language-general and language-specific properties of speech rhythm can be directly measured.
Categorization of temporal intervals
Authors: Large Edward W.
Published: July 2002

The rhythms of music and speech comprise sequences of temporal intervals separating the onsets of component events (e.g., notes and syllables). The perception of rhythmic patterning in such sequences involves the identification of distinct interval categories. Notating a musical rhythm, for example, relies upon the perception of nominal relationships between time intervals (e.g., 2:1, 3:1). Such categorization appears robust to perturbations that alter the absolute duration of temporal intervals. Thus it makes sense to ask whether the perception of categorical temporal relationships is ‘‘categorical perception.’’ The answer depends not only on the ability to make category judgments, but also on the differential discriminability of temporal intervals. Entrainment models propose that perceived onset time depends upon onset phase relative to an internal oscillation [Large and Kolen, Conn. Sci. 6, 177–280 (1995); McAuley, Proc. Cog. Sci. Soc., 615–620 (1996)]. Such models predict discrimination performance, however, they do not address categorization. This paper presents an extension of entrainment models that addresses categorization: In addition to relative phase dynamics, it explicitly considers the amplitude dynamics of internal oscillations. It captures existing data on interval categorization and makes new predictions regarding categorization and discrimination performance. Tests of the predictions are discussed.
Time series analysis of propagation in range-dependent shallow-water environment
Authors: Jensen Finn B. , Bini-Verona Francesco , Nielsen Peter L. , Gerstoft Peter
Published: July 2002

One track of the Formiche—97 shallow-water experiment in the Mediterranean involved broadband propagation over a sloping bottom with water depths varying between 120 m at the 16-element receive array and 600 m at the end of the 35-km track. The 0.82-kg SUS charges were detonated at two depths, 18 and 90 m, and at range intervals of around 1.2 km. Signals were recorded over a bandwidth of 50–5000 Hz on hydrophones covering the full water column. Detailed environmental measurements were also performed. Using state-of-the-art numerical modeling tools it was demonstrated the degree of accuracy with which broadband signals can be modeled in complex ocean environments, after having first fine-tuned the sediment parameters via a broadband acoustic inversion. The forward model used is a modified version of the ORCA code [Westwood et al., J. Acoust. Soc. Am. 100, 3631–3645 (1996)], which uses adiabatic mode theory combined with frequency-interpolation techniques to efficiently compute the broadband frequency response of the propagation channel. The broadband inversion was done using the SAGA code [Gerstoft, SACLANTCEN Rep. SM-333 (1997)], which does parameter optimization via a directed Monte Carlo search based on genetic algorithms. [Work was supported by the EC under the MAST-III project PROSIM, Contract No. MAS3-CT95-0008.]
A search algorithm to predict the resonance frequency of a shallow-water soliton packet
Authors: Chin-Bing Stanley A. , King David B. , Warn Varnas Alex , Zingarelli Robert A.
Published: July 2002

Recently a number of researchers have performed acoustic simulations of shallow-water regions that contain large amplitude internal waves (solitons). Their results have confirmed earlier findings: (a) mode conversions, due to acoustic interactions with soliton packets, can produce a large loss in acoustic transmission; (b) this loss occurs within a narrow band of acoustic frequencies (about the resonance frequency); and, (c) the transmission loss is most prominent when there is strong coupling between the lower-order (water-borne) propagation modes and the higher-order, very lossy (bottom interacting) modes. Using this knowledge, development of a search algorithm has begun to predict the resonance frequency of a shallow-water soliton packet. In this proof-of-concept work, a shallow-water soliton model was used to generate replicas of the soliton fields in the Strait of Messina. The allowable acoustic mode conversions were calculated and correlated with the dominate spatial wave numbers of the soliton packet. This was used to predict the resonance frequency. The resonance frequency was later confirmed by making hundreds of acoustic runs. The search algorithm will be discussed and examples shown. [Work supported by ONR/NRL and by a Federal High Performance Computing DoD grant.]
The existence and coherence of ray bundles in a refractive ocean
Authors: Smirnov Ivan P. , Khil’ko Alexander I. , Caruthers Jerald W.
Published: July 2002

Due to the existence of relatively smooth local extrema in the dependencies of ray-cycle lengths on initial path angles, different kinds of weakly diverging bundles (WDB) of rays can arise in refractive oceanic waveguides. As Brekhovskikh et al. have shown [Acoust. J. (in Russian) 36, 5, 824–831 (1990)], such bundles transporting high-level acoustic intensity can be interesting with respect to these applications in ocean acoustic tomography (OAT). Conditions for the existence of WDB for piecewise linear approximations of sound velocity profiles are formulated in this work. Based on these results, the possibilities for the existence of WDB for different regions of the world’s oceans are discussed. Important for tomographical application, the coherence of WDB is investigated under assumptions that the acoustic signals are emitted by noise sources with limited lengths of spatial coherence and the oceanic environment is randomly inhomogeneous in spatial coherence, as described by a Gaussian distribution. Numerically, it is shown that, because within a WDB ray parameters differ only slightly, coherence in WDB is maintained for propagation to long distances. In contrast are the rays outside the WDB. The optimal space/time filtering for OAT on the basis of the developed multiscale coherence approach is discussed.
Vertical noise coherence measurements in shallow water using Lagrangian drifters
Authors: Desharnais Fancine , MacDonald Blair R. , Mah Kenneth J.
Published: July 2002

The vertical coherence of an acoustic noise field in shallow water (at frequencies dominated by sea surface noise) can be used to infer certain ocean bottom properties such as the compressional sound speed of the upper sediment layer. As coherence is a normalized measurement, it is independent of background noise level, or wind speed/sea state, and remains a stable measurement over a wide range of environment conditions. A set of three Lagrangian ambient noise drifters (LANDs) were developed for vertical coherence measurements. The drifters (drogued at 15-m depth) follow the near-surface currents, and take hourly measurements of coherence from their two hydrophones located at 50-m depth. The hydrophone separation is 1 m, and the measurements are made over a usable frequency range of 100 Hz to 1 kHz. The drifters were deployed at Western Bank (Scotian Shelf, Canada) and drifted for up to 2.5 days. The sea conditions varied from calm to 3–4-m waves. The stability of the coherence data through the various sea states, and the impact of shipping noise on the measurements will be discussed.
Acoustic source estimation based on physical modeling and optimization algorithm
Authors: Amditis Aggelos J. , Nikita K. S. , Uzunoglu Nikoloas K.
Published: July 2002

The propagation of acoustic waves in the sea is analyzed by employing a horizontally stratified model for the sea medium with an arbitrary number of layers. Through this physical model, the acoustic field distribution, originating from an elementary source at an arbitrary location point inside the sea medium, is calculated by implementing a Green’s function approach. The proposed optimization algorithm is based on the minimization of the quadratic error function between estimated source signal values and signal values corresponding to known acoustic sources for the frequency content of the measured signal. Initial source estimation is based on the measured signal and the transfer function obtained by the above described propagation model. The minimization of the error function with respect to the model parameters described is carried out independently for each signal, corresponding to a known source, by employing the downhill-simplex minimization method. The measured signal is positively recognized as one corresponding to a known acoustic source if the minimization procedure results in an error value less than a predefined limit.
A universal parallel sonar signal processing system
Authors: You Lifu , Sang Enfang
Published: July 2002

A modern signal processing algorithm has been widely used in the sonar system, which has improved the sonar system features, but has also given the sonar signal processing system more compute burden in the meantime. Parallel signal processing technology is a solution. A universal parallel signal processing module is designed, which is based on the VME bus, so it is convenient to get a more powerful processing system. This module is comprised of four TMS320C40 digital signal processors. As examples, some basic sonar signal processing algorithms such as FFT, beamforming, etc. have been implemented using this system. The efficiency and speedup are given in this paper. This parallel signal processing system is suitable for many kinds of sonar systems.
The effects of bandwidth, dispersion, and correlation length scale on shallow-water reverberation
Authors: LePage Kevin D.
Published: July 2002

The ensemble average of the short-time average of the monostatic reverberation power expected in a shallow-water waveguide is obtained explicitly in the parameters of modal phase speed, slowness, dispersion, and source bandwidth under the assumption of a Gaussian source envelope and a two-dimensional Gaussian spatial scatterer correlation function. Results show that in the limiting case of very narrow bandwidths, resonant scattering is expected, yielding lower backscattered levels, and significant propagation structure is superimposed on the backscatterer time series, consistent with the two way TL to the scattering patch at the center frequency. In the limit of patch size small compared to the correlation length scale, higher backscattered levels, consistent with the scattering strength of the individual scatterers themselves, are expected, and the superimposed propagation structure is reduced due to the spatial decorrelation of the patches interrogated by the individual modes. Expressions for the time varying vertical coherence are also obtained in this framework.
Spectral diffusion of seismo-acoustic waves in shallow water
Authors: Odom Robert I. , Peyton Valerie I.
Published: July 2002

The propagation of seismo-acoustic waves in a strongly forward scattering medium is modeled as an energy diffusion process satisfying a convection-diffusion equation in slowness space [Odom and Mercer, Geophys. J. Int. 127, 111–124 (1996)]. No energy propagates with a slowness greater than the Stoneley/Scholte wave slowness, and therefore must satisfy a zero flux boundary condition at the maximum slowness. At a slowness equal to 1/(minimum shear speed), energy is lost by radiation to shear waves, and for finite frequencies satisfies a mixed boundary condition in slowness space. Maupin [Geophys. J. Int. 126, 762–780 (1996)] has recently shown how the branch line integral for the elastic continuum can be written as a dyad, which does not suffer from common problems with other formulations employing leaky modes. Maupin’s dyadic representation for the elastic wave continuum is employed to derive the proper mixed boundary condition and illustrate its effect on the diffusion of seismo-acoustic wave energy in a shallow water waveguide. [Work supported by ONR.]
The use of semianalytical calculation models to verify SEA predictions of flanking transmission in building structures
Authors: Bosmans Ivan , Vermeir Gerrit
Published: July 2002

The prediction of structure-borne sound transmission between coupled plates represents a key aspect when studying sound transmission in buildings, since vibrational energy flow between walls and floors leads to the existence of flanking paths. The resulting flanking transmission can be assessed theoretically using statistical energy analysis (SEA). In this method, the coupling between plates is usually estimated based on the wave approach for the corresponding semi-infinite plates. For a specific type of junction, one might consider verifying the predictive performance of SEA experimentally on an isolated plate junction. However, possible discrepancies between measured and predicted data can result either from the failure of the calculation model for the semi-infinite plates, or by the violation of the basic SEA assumptions. In this paper, a semianalytical calculation model based on a modal summation approach for finite-sized plates is proposed to determine the source of the prediction error. For this purpose, calculations are performed for two ensembles of similar corner junctions with randomly distributed dimensions. The analysis emphasizes the influence of the model overlap factor on the accuracy of SEA. [Research funded by the Flemish Institute for the promotion of Scientific Technological Research in Industry (IWT).]
Controlling subway noise in the LG Arts Center—Sangnam Hall
Authors: Korista D. Stanton , Abdelrazaq Ahmad
Published: July 2002

The LG Arts Center in Seoul, Korea is a multifunction facility including a Hall intended for a variety of performances including western drama and dance, chamber music, traditional Korean folklore plays, opera, oration, and digital recordings. The 1050 seat Hall is designed to guard against both airborne and groundborne noise. Airborne noise is controlled by enclosing the stage house and the audience chamber with 200-mm airspace double walls. Horizontal and vertical acoustical joints at the Hall structure interface with the main building and control groundborne noise generated by the subway trains in a tunnel adjacent to the site. The subway tunnel and the foundation structure of the building are in the same rock strata and the groundborne noise is transmitted as structure-borne noise to the Hall through the lower-level parking structures. Without isolation the noise from the subway would have intrusive effects on the multifunctional use of the performance hall and stage area. Vibration isolation rubber bearings are provided in the interface design to reduce the noise to acceptable levels in the noise critical space. The impact of the proposed isolation bearing pads on the building design and their acoustical benefits are discussed.
Real-time characterization of unstationary flows using a time-reversal mirror
Authors: Maurel Agnes , De Rosny Julien , Manneville Sebastien , Roux Philippe , Fink Mathias
Published: July 2002

Violation of time-reversal invariance for acoustic waves is a new method for characterizing a vorticity field with a double acoustic time-reversal mirror (TRM). Experiments show that the double TRM works as a vorticity amplifier and, even in the case of small vorticity field, provides the characteristics of the flow. Moreover, the TRM works in real-time and allows one to measure the vorticity field at a 20-Hz frequency rate. The case of a Von Karman vortex street is studied experimentally.
Results from cross-correlation measurement of estuarine current velocity
Authors: Bland Roger W. , Neuman Daniel G.
Published: July 2002

A technique for accurate measurement of the average water velocity in estuarine environments has been developed. The technique is based on measuring the travel time of 200-kHz ultrasonic signals. A high degree of precision has been obtained using a cross-correlation technique, and long, detailed time series have been recorded. However, in some geometries the precision is degraded by multipath interference. Measurements of current speed in California’s Sacramento-San Joaquin River Delta system will be discussed, with particular reference to the effects of multipath interference. Data were collected throughout a large flood in Dec. 1996–Jan. 1997. An analysis of the variation of the strength of the ultrasonic signal has been carried out to determine the attenuation of the signal due to the load of suspended sediment, and the results will be presented. The current-velocity time series recorded in the San Francisco Bay also show wave trains from boat wakes. These wakes will be discussed.
Selected topics in electroacoustical calibrations, measurements, and standards
Authors: Nedzelnitsky Victor
Published: July 2002

Modern instruments including multichannel dynamic signal analyzers capable of FFT and other signal processing, as well as improved concepts for applying these instruments to particular electroacoustical measurement and standardization issues, are essential tools in modern electroacoustical metrology. The overall uncertainty of a measurement, and its suitability for its intended purpose, may involve components of uncertainty that are individually and collectively difficult to evaluate. In such instances, the capacity to perform the same measurement in different ways can be especially valuable. Different measurements exploiting differences in source signals, signal processing implementations, and apparatus can provide valuable insight and can support the validity of measurement results and interpretations. Examples selected from recent and ongoing work at NIST include calibration or characterization of microphones and microphone arrays in a free field, examination of imperfections in anechoic chambers, and measurement of effects of instabilities and nonlinearities in electroacoustical systems. Recent ANSI (American National Standards Institute) and IEC (International Electrotechnical Commission) standardization work involving the coherence function method for examining nonlinearities in the electroacoustical performance of hearing aids has elicited comments proposing an alternative method involving the Hilbert transform. Some significant advantages and disadvantages of these methods are discussed.
Procedures for the precision of sound power determination by S. I. scanning method
Authors: Tachibana Hideki , Suzuki Hideo
Published: July 2002

As the third international standard of ISO 9614 series (Acoustics—Determination of sound power levels of noise sources using sound intensity), the precision method by scanning is now being drafted by ISO/TC43/SC1/WG25. It will be Part 3 (Precision method for measurement by scanning), following Part 1 (Measurement at discrete points: precision, engineering and survey classes) and Part 2 (Measurement by scanning: engineering and survey classes). In this paper, the outline of the draft and discussions are introduced by focusing on the measurement accuracy, the methods for checking the adequacy of the measurement sound field (field indicators), and practical procedures.
Measuring system for the derivation of thermal loudspeaker parameters
Authors: Behler Gottfried K. , Bernhard Armin
Published: July 2002

Loudspeakers used in high-power applications are stressed by mechanical and thermal load. Both can be the reason for damage and should be controlled. Whereas the mechanical situation can be derived from the input voltage and the total impedance, which can be measured easily, it is rather difficult to calculate for the actual voice coil (VC) temperature under working conditions. It is possible to derive the VC temperature from the input voltage using linear mathematics if the thermal equivalent circuitry of the loudspeaker including the cabinet is known. Former papers have discussed the thermal equivalent circuitry of the loudspeaker without taking the cabinet into account. In the presented paper an equivalent circuit is presented, which can be used for loudspeakers in different cabinets and a measuring method is shown, that enables long-term measurements on loudspeaker systems with constant input power using a PC-based regulator. The main problem is to calculate the voice coil temperature from the measured impedance data. The solution of this problem is presented. At least the thermal parameters of the equivalent circuit are calculated from the measured data using curve fitting. Different loudspeakers and setups have been tested. The results will be presented.
Anomalous low frequencies from a bowed violin string
Authors: Hanson Roger J. , Halgedahl Frederick W.
Published: July 2002

With a bow force greater than the Schelleng maximum and careful bow control, it is possible on a violin to produce sounds of definite pitch ranging from approximately a musical third to as much as two octaves below the normal pitch. [Hanson et al., CAS J., 2, Series II, 1–7 (1994)]. The pitch is in agreement with the fundamental frequency of the observed harmonic series. The fundamental itself is very weak if the sounds are produced on the open violin G string. These anomalous low frequencies (ALF) occur when the bow force is great enough to prevent the Helmholtz kink from triggering the normal release of the string from the bow hair. The bow serves as a quasitermination point of the string so that there are pronounced bow-nut and bow-bridge reflections yielding at the bow a very complex waveform, some portion of which triggers the slipping of the string. The ALF can also be produced on a bowed monochord where the motion at several points is detected optically, and the sound is generated from the resulting signal. Some musical applications of the effect will be discussed.
Parameters involved in low-frequency noise annoyance
Authors: Landstrom Ulf
Published: July 2002

Noise annoyance and related adverse responses evoked during exposures in low-frequency environments are effected by a number of separate and interacting parameters. Laboratory as well as field studies have shown that in most cases the correlations between annoyance and noise levels or linear values are very weak. For descriptions of the fundamental correlations between low-frequency noise exposure, annoyance, and related adverse effects, deeper knowledge about the different interacting factors is needed. One consequence of this deficient knowledge of the critical factors for low-frequency noise annoyance is that well-founded methods for the evaluation of annoyance and related adverse effects are still missing. Noise control, building and rebuilding of new acceptable environments, or other types of measures against the low-frequency noise problems cannot be based only on evaluations of the dBA or other simple rating alternatives. For adequate evaluations and measures, deeper knowledge about interacting factors, e.g., tones, frequency character, bandwidth, exposure time, fluctuations, perception, masking and type of work, is needed. The present paper is a summary of the results and conclusions that could be drawn from studies directed upon some of the most important parameters involved in low-frequency noise annoyance.
Computational and subjective procedures for the assessment of sounds with weak tonal components
Authors: Ellermeier Wolfgang , Vormann Matthias
Published: July 2002

A discrete tone is defined as ‘‘prominent’’ if the level of the tone exceeds the level of the noise contained in its critical band by 6 dB. Tonal or narrow-band noise components of tire sounds often fall short of this level, but are nevertheless rated as prominent by test drivers. Tonality is closely related to the sensation of pitch strength, which depends upon the sensation level, bandwidth, duration, and frequency of the tonal components present. In order to compare a number of computational procedures for calculating the tonality of sounds, a broad range of tire noises, synthesized tire sounds, and artificial tone–noise mixtures in which the level and bandwidth of the tonal components was varied, were subjectively assessed using (a) a paired-comparison technique, and (b) categorical judgments collected from about 60 listeners. They were objectively assessed by tone-to-noise and prominence-ratio measurements, and by a new method based on a model of pitch sensation which accounts for the temporal and frequency resolution as well as for masking properties of the human ear. The advantages and limits of the different methods will be compared and discussed. Especially for weak narrow-band components, the pitch model seems to be superior to the other methods used to quantify the degree of tonality.
Dynamic and acoustic properties of beams of composite material
Authors: Tavallaey Shiva Sander
Published: July 2002

Modern engineering techniques require the use of more sophisticated and optimized structural design. One way towards this aim is the use of materials which in some way optimize their inherent properties. A concept which suits this purpose very well is composite/sandwich material. Sandwich/composite materials find numerous applications where high strength and low weight are important criteria. The dynamic behavior of a sandwich beam with different boundary conditions has been investigated. Measurements have been carried out. Free-free and clamped-clamped boundary conditions were included. Modal analysis has been used to estimate resonance frequencies and modal parameters. Further, acoustical parameters of a composite plate are measured. An analytical model for prediction of wave propagation constants in sandwich/composite structure is presented. Measured and predicted results are compared.
Modeling of porous material added damping on a vibrating plate
Authors: Dauchez Nicolas , Sahraoui Sohbi , Atalla Noureddine
Published: July 2002

Porous materials like plastic foam are well known for their ability to absorb sound. When bonded onto a vibrating structure they may add damping. Their damping performance is investigated here. The generic configuration is a free aluminum plate (28 cm×22 cm×1 mm) mechanically excited (20–500 Hz), damped by a 1- to 5-cm-thick foam layer. The analysis of porous material behavior, based on Biot–Allard theory [J.-F. Allard (Chapman and Hall, 1993)] shows that damping is mainly related to the frame viscoelasticity. A formulation of an equivalent plate, substituting the porous layer by a monophasic viscoelastic material, is presented. This formulation gives behavior indicators, like amount of added stiffness and damping. It also accelerates the computation of the structure vibration, compared with a 3-D finite element code, including poroviscoelastic elements [R. Panneton and N. Atalla, J. Acoust. Soc. Am. 100, 346–354 (1996)]. Good agreement is found with both the complete discretized formulation and experimental results. Maximum damping available from biphasic phenomena, like air flow induced by the layer flexure, is also evaluated and compared with structural damping. This shows that more sophisticated mounting conditions should be used to take advantage of the biphasic property.
Inversion of explosive shallow-water transmission loss data to obtain acoustic seabed models
Authors: Hall Marshall V.
Published: July 2002

Acoustic transmission losses (TL) measured with explosive charges along a track in shallow water have been analyzed at frequencies in octave steps starting from 16 Hz. The results have been inverted to obtain the acoustic properties of the seabed, which is modeled as a uniform half-space. There are five unknown parameters: sound and shear speeds (Cp and Cs), sound and shear absorption coefficients (Ap and As), and density. The number of unknowns is reduced to three by using a regression equation for density in terms of Cp, while As is set to its maximum physical value. By holding Cs fixed for subsets of the process, the number of unknowns is reduced to two (Cp and Ap). For a given seabed, the theoretical TL is computed as a function of range using TL models valid for stratified media. The rms residual between the measured and theoretical TL is computed over rectangles in the Cp–Ap plane. The position at which the rms residual is a minimum is noted, and expressed as a function of Cs. Criteria for selecting the optimum result are discussed. The half-space properties contain useful information, but since they vary with frequency, they do not describe the actual seabed.
Sonochemistry of sonodynamic therapy
Authors: Riesz Peter , Misik Vladimir
Published: July 2002

Sonodynamic therapy is a promising modality for cancer treatment based on the synergistic effect of cell killing by a combination of drugs (sonosensitizers) and ultrasound. The effectiveness of sonodynamic therapy was demonstrated in cell studies and in tumor-bearing animals. The mechanism of drug-dependent sonosensitization is unknown, but it seems likely that various mechanisms operate for different classes of sonosensitizers. Ultrasound-mediated sonodynamic activation of porphyrins is particularly poorly understood and hypotheses ranging from singlet oxygen production to peroxyl radical formation have been advanced. Evidence has been obtained against some of the arguments which were proposed in favor of the singlet oxygen hypothesis and the requirement of extracellular localization of gallium porphyrin ATX-70 molecules for sonosensitization in HL-525 cells has been established. Short-lived toxic intermediates produced from sonosensitizers by ultrasound are implicated in the mechanism of sonodynamic activation of some sensitizers. EPR spectroscopy was used to identify proposed intermediates in the activation of certain sonosensitizers (e.g., N,N-dimethylformamide, DMSO) by ultrasound. Certain water-soluble azo compounds, which are stable at hyperthermia temperatures and can be decomposed by ultrasound to give peroxyl radicals capable of initiating peroxidation of lipids and damaging other cellular sites, appear to be promising sonosensitizers.