bluuebunny/crossref_metadata_2025 huggingface dataset:
Spatial perception, the acoustics of the external ear, and interactions with earphones
Authors: Shaw Edgar A. G.
Published: July 2002

In free-field listening, the perception of source elevation and discrimination between forward and rearward source positions appear to be largely dependent on the availability of high-frequency spectral cues associated with the mode structure of the concha and diffraction by the pinna, respectively. The role of the concha is particularly clear between 5 and 10 kHz, while pinna-diffraction effects are strongly developed between 2.5 and 6 kHz. When sounds are presented through earphones, the free-field characteristics of the ear are replaced by very different characteristics that are dependent on complex and unreliable interactions between the individual earphone and the individual ear. Since these interactions are highly dependent on earphone design, it may be appropriate to focus attention on the special characteristics that may be desirable in earphones intended for use in the creation of virtual acoustical environments. These may include coupling that (i) allows the response at the eardrum to follow the primary resonance curve of the individual ear and (ii) minimizes the inter- and intrasubject variations in response to other frequencies, including frequencies above 5 kHz where the wave characteristics of the concha come into play.
Transforming the head-related transfer functions
Authors: Farias Julio G. , Perrott David R.
Published: July 2002

Auditory and visual localization of a real sound source under both natural and artificial listening conditions were examined. Five subjects were employed in a visual search task paradigm which contained some conditions that were aurally aided. The aurally aided conditions utilized white noise, emitted at the subjects horizon, from a single speaker at 72 dB(A). The paradigm also contained conditions in which a fully enclosed helmet was used to occlude the pinna in order to create the artificial listening conditions in which normal head-related transfer functions (HRTFs) are transformed. Reaction time and accuracy were simultaneously used within each condition to measure the extent to which normal HRTFs were transformed by the artificial listening conditions. The data were analyzed and significance was found between the natural and artificial conditions [F(8,32)=3.59; p=0.005]. There was an increase in both accuracy and reaction time between the natural and transformed HRTFs. (The experiment began with six subjects, but due to scheduling difficulties, their data were not available at the time this abstract was written. Their data will be included in the analysis for the presentation.)
Probe signals for obtaining impulse response of doubly-spread channels
Authors: McDonald Vincent K. , Shockley Richard C. , Rice Joseph A. , Green Dale
Published: July 2002

The U.S. Navy is developing undersea acoustic telemetry and ranging (telesonar) technology using shallow oceans as the physical transmission channel. The telesonar signaling philosophy is to achieve high reliability by designing for anticipated time and frequency spreading in the bounded, nonstationary, forward-scattered transmission channel. Obtaining the time-variant acoustic impulse response of representative channels supports appropriate signal design. These data allow measurement of time and bandwidth coherence, and aid understanding of boundary effects. Also, convolving signal waveforms with the impulse response simulates the influence of the transmission channel. This paper examines candidate probe signals for acquiring bandlimited acoustic impulse responses of shallow ocean channels. These probe signals are intended for systematic sampling of various littoral channels during telesonar sea tests. [Work supported by ONR-321.]
The MPL sound velocimeter: An instrument for in situ sound velocity measurements in the deep ocean
Authors: Sweeney Aaron D. , Spiess Fred N. , Boegeman Dwight E. , Jabson David M. , Zimmerman Richard
Published: July 2002

A sound velocimeter designed to measure sound speed to a precision of 1 part in 105 has recently been tested at sea. The device was used in conjunction with a CTD to check agreement with sound speeds computed from popular empirical equations [C.-T. Chen and F. J. Millero, J. Acoust. Soc. Am. 62, 1129–1135 (1977); V. A. DelGrosso, J. Acoust. Soc. Am. 56, 1084–1091 (1974)]. Disagreements will be discussed. High-precision travel-time measurement is achieved through use of a phase-comparison technique. For this purpose, the device determines the number of cycles of the acoustic signal between the transmitted pulse and received echo, adjusting the frequency until the echo and transmitted pulse are in phase. The travel time is the product of the signal period and the number of cycles between the outgoing and incoming pulses. The velocimeter’s capability of measuring travel time at several different frequencies near 4 MHz serves as a check on the timing consistency, since the travel time should be independent of frequency over this narrow range. The total distance traveled by the pulse is about 8.6 cm. With proper calibration, this pathlength can be known to a precision of 1 part in 105.
Fiber-optic acoustic and vibration sensors and systems
Authors: Brown David A.
Published: July 2002

Fiber-optic interferometric acoustic sensors and accelerometers have been developed for a wide variety of marine and geophysical applications. They offer advantages including compatibility with optical fiber telemetry, light weight, and removal of electronic in the wet-end. This presentation will review sensor designs used in sonar applications. Interrogation and multiplexing schemes using 3×3 couplers and fiber-optic Bragg gratings for acoustic and vibration applications will also be presented. [Work supported by ONR and NUWC.]
Active control of structural intensity in a T-beam
Authors: Hayek Sabih I. , Mayfield Patrick J.
Published: July 2002

The total vibration structural intensity in a T-shaped connected beam is modeled for a point mechanical noise source located at one end of the horizontal branch of the T-beam. For a point force lying in the plane of the T-beam, only flexural and extensional intensities are included. In order to minimize the total structural intensity propagating into the vertical leg of the T-beam, three control strategies were investigated. These include a point force, a point couple, and a combined point force and couple, where the location of the control actuators were the same for all three strategies. The effectiveness of each strategy is evaluated for control on or off resonances. The total injected power required for control for each control strategy was also evaluated to rate the efficiency of each method.
A comparison of an analytical model and experimental results for a curved piezostructure
Authors: Henry James K. , Clark Robert L.
Published: July 2002

Active structural acoustic control (ASAC) research has previously concentrated on flat plates with attached piezoelectric sensoriactuators. Application of this ASAC research to aircraft interior noise requires the consideration of the curvature of the fuselage panels in the dynamic modeling. This paper investigates the effects of curvature on the dynamics of fuselage panels. A state variable model of a simply supported curved panel with attached piezoelectric sensoriactuators is developed for ASAC applications. The model is verified by comparison to previous work on curved panels and flat plates. Further, the model is validated by experimental data obtained from a simply supported curved panel test structure. [Research supported by NASA Langley Research Center under Grant No. NCC 1 250.]
End correction in vibrational power flow measurement through joints
Authors: Wu Qunli
Published: July 2002

In the determination of coupling loss factors for complex joints, such as adhesive or bearing joints, analytical methods may not provide satisfactory results. The experimental approaches are normally preferred. In the measurements of the coupling loss factors for the joints, it is assumed that the end reflection can be neglected. However, it is difficult to create such nonreflection termination in the experiments. In this paper, the effects of end reflection in the determination of the coupling loss factors were studied. A new method using the measured reflection coefficients of the termination to correct the measured coupling loss factors was proposed. The experimental study was carried out in the stepped beams and adhesively bonded beams. The experimentally corrected results were compared with the theory.
Detection and classification of the acoustic emission signals by the vector receiver in an elastic environment
Authors: Tran Dat H. , Krasilnikov Alexsandr J. , Oboznenko Igor. L
Published: July 2002

The novel method of detection and classification of the developing microfissures in elastic environments is proposed. A probability model of the microfissures development is assigned that depends on time and space. The problem is reduced to the solution of the equation of elasticity with the limit and primary condition assigned by it. The rule of the development of the fissures’ radius as discrete in time and space or as a random process is assigned. Two variants of the given problem are analyzed: Voltaire’s equation or the ordinary differential equation with variable coefficients. The numerical results are given. A six-component vector-phase receiver settled on the unrestricted boundary is used as a receiver. The projection of the vectors of tension and shift are measured and calculated: their mutual correlation is possible. As a result, information about the intensity vector is provided that determines the coordinates and the intensity of the acoustic emission.
Aircraft pipeline quality control in a commercial medium
Authors: Shkvarnytska Tannia Yu.
Published: July 2002

The method of controlling the presence of microcracks in metal pipeline materials for aircraft systems in a commercial medium (on the stage of quality selection of the systems for the specific aircraft model) is proposed. The structure of the echosignal from the cylindrical pipeline in water is investigated in comparison with the pipeline standard section without microcracks. In the presence of microcracks in shell material, changes in the effective Young’s modulus and Poisson’s ratio due to echosignal structure and the meaning of statistical longitudinal and shear material modules are defined. That effective modulus contains information about microcracks in the pipeline material measuring 1×3 mm (the cylindrical crack). Approximately 50 cm of the length of the main part is allowed to register changes in material. In particular, it is noticed in the frequency range f(3–10) MHz, where the resonance microcrack’s peculiarities become apparent. The analytical formulas of effective elasticity modules that contain information about the presence of the many microcracks (including the sole) are obtained. The proposed method can be applied in the case of aircraft pipeline quality control.
Windowed displays of broadband impulse response measurements for finite cylindrical shells
Authors: Morse Scot F. , Marston Philip L. , Kaduchak Gregory
Published: July 2002

Previously interpreted angle-frequency domain displays of impulse response measurements for thick nd thin finite cylindrical shells [S. F. Morse et al., J. Acoust. Soc. Am. 101, 3043(A) (1997)] reveal backscattering enhancements due to elastic effects on the cylinder over a range of target tilt angles. Presently, these signatures are examined in the angle-time and time-frequency domains. For selected angles, time-frequency analyses are presented as well as synthetic aperture images for selected apertures. From these it is possible to identify the spatial and angular locations of many of the important elastic contributions. Features associated with a meridional ray, as well as helical rays, have the appearance of originating from the back end of the cylinder. One major elastic reponse observed at large tilt angles is found below the coincidence frequency. [Work sponsored by the Office of Naval Research.]
Development of aids to speechreading based on automatic production of cued speech
Authors: Braida Louis D. , Duchnowski Paul , Lum David S. , Sexton Matthew G. , Bratakos Maroula S.
Published: July 2002

In manual cued speech (MCS) a speaker produces hand gestures to resolve ambiguities among speech elements that are often confused by speechreaders. The shape of the hand distinguishes among consonants; the position of the hand relative to the face among vowels. Experienced receivers of MCS achieve nearly perfect reception of everyday connected speech. MCS has been taught to very young deaf children and greatly facilitates language learning, communication, and general education. A system that can produce cued speech automatically in real time is currently being developed at MIT. Cues are derived by a speaker-dependent HMM speech recognizer that uses context-dependent phone models and presented visually by superimposing animated handshapes on the face of the talker. The benefits provided by these cues strongly depends on articulation of hand movements and on precise synchronization of the actions of the hands and the face. Cue receivers experienced in the reception of MCS can recognize roughly two-thirds of the keywords in cued low-context sentences correctly, compared to roughly one-third by speechreading alone. Ongoing development aims at increasing the accuracy and robustness of the cue recognizer, refining the display, and simplifying the hardware required for use in classroom settings. [Work supported by NIH.]
Lexical influences on the McGurk effect
Authors: Brancazio Lawrence
Published: July 2002

Much research has been devoted to the study of whether lexical knowledge affects lower-level phonetic processing. For example, in the Ganong effect (Ganong, JEP:HPP 6, 110–125), the lexical status of the end points of a /b/–/p/ voicing continuum affects the point at which a shift from /b/ to /p/ responses is made (e.g., more /b/ responses in a /bif/–/pif/ continuum than in a /bis/–/pis/ one). This study addresses whether lexical effects influence the perception of discrepant audiovisual stimuli (the McGurk effect). Pairs of words and nonwords were constructed that differed only in the place of articulation of the initial phoneme; in each pair either both members were words, only one was a word, or neither was a word. The stimuli were presented audiovisually, with the audio from one pair member and the video from the other; subjects identified the intial phoneme. Analyses of a proportion of video responses (McGurk responses) indicated significant effects of the lexical status of the auditorily-presented token and of the visually-presented token: the McGurk effect was stronger when the video token was a word, and weaker when the audio token was a word. Implications of these findings will be discussed.
Effects of identification training on Japanese adults’ perception of American English vowels
Authors: Strange Winifred , Akahane-Yamada Reiko
Published: July 2002

In a pre-test, 20 Japanese listeners’ identification of American English vowels /ε, æ, ■, ■, ■/ was examined, using syllables /bVb, bVp, dVd, dVt, gVg, gVk/ embedded in sentences produced by four men (CVCs) and real words produced by two men and one woman (Words). Experimental subjects (N=10) completed nine training sessions on /æ, ■, ■/, using CVCs (three speakers). Pre-test-Post-test comparisons showed that identification of trained vowels in CVCs improved for the experimental group (42% to 85% correct), but not for the control group (44% to 44%). The training effect generalized to the speaker not used in training. Experimental subjects also improved significantly on trained vowels in words (43% to 63%); the control group did not (44% to 47%). Again, the training effect transferred to novel speakers. Accuracy on untrained vowels /ε, ■/ decreased somewhat for the experimental group, while it improved for the control group, suggesting the presence of response biases. Nevertheless, overall greater performance gains for the experimental group (23% for CVCs, 10% for words) versus the control group (6% for CVCs, 6% for words) suggest that robust changes in perception of non-native vowels are possible with identification training. [Work supported by NIDCD and ATR.]
Communicative factors in speech variation: Hyperarticulatory compensation for lexical competition
Authors: Wright Richard
Published: July 2002

Based on factors in lexical competition such as usage frequency and similarity-neighborhood density, previous research has characterized words as ‘‘easy’’ or ‘‘hard’’ to identify. Lindblom [in W. Hardcastle and A. Marchal (Eds.), 403–439 (1990)] has proposed that talkers accommodate listeners’ communicative needs by controlling the degree of reduction (hyper- and hypoarticulation) in different contextual conditions. This study examines the degree of centralization of vowels in 38 ‘‘easy’’ and 38 ‘‘hard’’ monosyllabic (CVC) words of equal familiarity spoken in isolation by ten talkers. Measurements of the first two vowel formants (F1, F2) are made at the point of maximal displacement in the vowel (excluding the initial and final 50 ms of the vowel). Centralization is measured by calculating the Euclidean distance from the center of a talker’s F1–F2 vowel space. Three results emerge: (1) overall ‘‘easy’’ words were significantly more centralized than ‘‘hard’’ words; (2) peripheral vowels, such as /i/, /a/, /u/, showed the greatest effect; and (3) there was considerable between-talker variability both in overall vowel centralization and in the magnitude of the difference between vowels in ‘‘easy’’ and ‘‘hard’’ words. [Work supported by NIH Grant No. DC00012.]
Three experiments in aurally guided visual search
Authors: McKinley Richard L. , D’Angelo William R. , Nelson W. Todd , Perrott David R. , Bolia Robert S.
Published: July 2002

Aurally guided visual search is a logical concept if one considers the evolutionary utility of the ability to localize sound sources, such as food or predators by primal man. The localized auditory stimulus would direct the gaze to the approximate location of the target or threat. The auditory system has a full 4π sr field while the visual system is limited to about 150 deg horizontally by about 60 deg vertically. The spatial resolution of the visual system is high, while the auditory system’s spatial resolution is low. A series of experiments is being conducted investigating the effects of spatial auditory cueing on visual search. This paper describes the design, methodology, and results of the first three aurally guided visual search experiments in this series. The experiments were conducted using both real and virtual spatial auditory stimuli and included visual target location acquisition and discrimination in a black visual field, visual target detection in a simple gray visual field, and visual target location acquisition and discrimination in visual fields of varying complexity. The results indicate that spatial auditory cues significantly improve performance in a visual search task in all visual environments investigated. [Work supported by AFOSR.]
Aurally aided visual search: The effects of individualized head-related transfer functions on response times
Authors: Bolia Robert S. , McKinley Richard L.
Published: July 2002

Previous experiments by the authors have investigated the effects of spatially correlated auditory cues on visual search times. In a recent study [D’Angelo et al., J. Acoust. Soc. Am. 101, 3107 (1997)], it was shown that the addition of a spatially correlated auditory cue, presented over headphones, effects a significant reduction in response times for a search for a single visual target among a varying number of similar distractors. The aim of the present study is to investigate the effects of individualized head-related transfer functions (HRTFs) on response times in an aurally aided visual search task. Subjects will search for a visual target among 1, 5, 10, 25, or 50 distractors with the aid of a spatially correlated auditory cue, using either their own or generic (90th percentile KEMAR) HRTFs. Results will be presented and the implications for the design of virtual auditory systems discussed. [Work supported by AFOSR.]
Three-dimensional auditory localization of nearby sources
Authors: Brungart Douglas S. , Durlach Nathaniel I. , Rabinowitz William M.
Published: July 2002

Auditory localization performance has been measured for a point- source stimulus within 1 m of a listener’s head. Broadband noise bursts, roved in intensity to eliminate loudness as a distance cue, were presented at random locations ranging from 0°–180° in azimuth, 60°–60° in elevation, and 15 cm–1 m in distance. The listeners responded to these stimuli by moving a response wand to the perceived location of the sound. Response errors were then calculated in azimuth, elevation, and distance. Directional localization error for nearby sources was comparable to that previously reported in localization experiments at distances greater than 1 m. Distance accuracy was found to depend on source azimuth, with best performance to the listener’s side and worst performance directly in front of the listener. Distance judgments were slightly more accurate when the amplitude of the stimulus was not randomized to eliminate distance effects. The improvement in accuracy for the fixed-amplitude stimulus was greater in front than to the side of the listener. Distance accuracy was substantially degraded when one of the listener’s ears was occluded by an earplug and muff. These results suggest the importance of binaural cues to near-field distance perception. [Work supported by AFOSR.]
The contribution of interaural intensity differences to the horizontal auditory localization of narrow bands of noise
Authors: Bakke Matthew H. , Levitt Harry
Published: July 2002

Brief noise bursts (340-ms, 20-ms rise/fall times) consisting of third-octave bands of noise (center frequencies 0.5, 1, 2, and 4 kHz) and bandpass noises with different degrees of low-frequency content (0.5–4, 1–4, and 2–4 kHz) were recorded binaurally from 17 different locations (90 deg on the left to 90 deg on the right in 11.25-deg steps) at the ears of KEMAR in an anechoic room and a reverberant room. The recorded sounds were processed by attenuating or removing interaural intensity differences (IIDs) and presented to five normally hearing subjects through insert transducers (ER-3A) in a sound-source identification task. The effect on localization accuracy of reducing IIDs by half was not significant. However, reducing IIDs to zero resulted in degraded localization accuracy for noise bands that contained energy above 1000 Hz. The results were similar for signals recorded in a reverberant room as for those recorded in an anechoic room. Results will be discussed with respect to the relative contributions of interaural time differences (ITDs) and IIDs for horizontal auditory localization. [Work supported in part by NIH, NIDRR.]
Measurement and analysis of the propagation of sound from the continental slope to the continental shelf
Authors: Chiu Ching-Sang , Smith Kevin B. , Lynch James F. , Gawarkiewicz Glen G. , Pickart Robert S. , Sperry Brian , Miller James H. , Robinson Allan R.
Published: July 2002

Carried out jointly by the Woods Hole Oceanographic Institution (WHOI), Naval Postgraduate School (NPS), Harvard University, and the University of Rhode Island (URI), the measurement program of an integrated acoustic-oceanographic field study called Shelfbreak PRIMER took place in the Middle Atlantic Bight. One of the goals of Shelfbreak PRIMER is to characterize and understand the propagation of sound from the continental slope to the continental shelf, including the effects of shelfbreak frontal processes, seasonal stratification, and topographic variations. The field work included two intensive 3-week experiments, one in July 1996 (summer) and the other one in February 1997 (winter). In particular, each of the two experiments employed a suite of acoustic and oceanographic sensors including several transceivers/sources and two vertical hydrophone arrays (VLAs) straddling the shelfbreak front and a SeaSoar that provided several high-resolution, three-dimensional surveys of the frontal region. The results from a data and modeling analysis of the pulse transmissions from a fixed sound source on the slope to a fixed VLA on the shelf are discussed. The objective of this analysis is to gain fundamental insights into the tidal-to-seasonal variability of the amplitudes and travel times of the strongly coupled normal modes. [Work supported by ONR.]
Sound propagation in regions overlaying gaseous range-dependent sublayers
Authors: Sidorovskaia Natalia A. , Werby Michael F.
Published: July 2002

Some regions of the ocean consist of bottom layers, which have imbedded bubbles. The bubbles may consist of methane gas. The consequence is that the particular layer has a sound speed, which is lower than in the overlying layer. Depending on the wavelength of the sound source relative to the layer thickness, sound may either reflect off the layer, or it may partially tunnel through the layer to the next set of layers. This effect is frequency dependent, and at some point, when the wavelength in the layer is small relative to the layer thickness, no sound will traverse the layer. The effect of this is that the modal eigenspectrum will be altered and will be largely characterized by the gaseous layer structure in horizontal and vertical directions. The influence of range-dependent gaseous regions on modal spectrum and measured signal structure in different underwater waveguides, which can be considered as strongly range dependent for spatially localized gaseous regions, is modeled by the normal mode code SWAMP for different frequency ranges. [Work sponsored by ONR, the Naval Research Laboratory, and the University of New Orleans.]
Room noise criteria standards: What features are important?
Authors: Pei Hsien-sheng (Jason) , Hellweg Robert D. , Peppin Richard
Published: July 2002

At the joint meeting of ASA and NOISE-CON 97 in June 1997, the authors organized a special session on room noise criteria, in which a ‘‘questionnaire’’ on the features of several existing noise criteria methods was answered by panel experts and distributed to the noise control community for their inputs. The four methods were: the A-weighted sound-pressure level; the noise criteria (NC) tangency method contained in older versions of the ASHRAE handbook; the balanced noise criterion method (NCB) contained in ANSI S12.2; and the room criteria method (RC) contained in ANSI S12.2 and the current ASHRAE handbook. The features considered were: speech interference, high-frequency annoyance, mid/low frequency annoyance, very low-frequency annoyance, pure tones, temporal fluctuations, user friendliness, wider frequency range, speech privacy, costs, and ability to handle noises with abnormal characteristics and complexity. The questionnaire rating was twofold: (1) how well does each method adequately account for each of the ‘‘features’’ (or characteristics) relative to the other methods, and (2) what is the relative importance of each of the ‘‘features’’ in a noise criteria method. This paper presents the results of the questionnaires and reports progress since the joint meeting in ANSI S12 WG18 and ASHRAE TC2.6.
Biologically inspired SCAT sonar receiver for 2-D imaging
Authors: Simmons James A. , Saillant Prestor A. , Boatright-Horowitz Seth
Published: July 2002

The SCAT (spectrogram correlation and transformation) model of biosonar in bats represents FM broadcasts and echoes as spectrographs with an integration time of 300–400 μs. The model then mimics signal processing by bats with two parallel computational paths: (1) determination of the delay of echo spectrograms using delay lines and coincidence detectors, and (2) deconvolution of the echo spectrum to determine delay separations smaller than the integration-time window [Saillant et al., J. Acoust. Soc. Am. (1993)] to produce high-resolution echo-delay images. The new, binaural version of the SCAT model receives broadcasts and echoes at two ‘‘ears’’ and uses binaural algorithms to produce range/azimuth sonar images. The model incorporates several stages of binaural interactions that facilitate different functions including determination of azimuth itself, enhancement of target detection, and separation of adaptive imaging from these other functions. The model produces images that locate the principal glints in complex targets from only a few sonar broadcasts without having to sample many different target aspect angles. [Work supported by ONR and NSF.]
Damage of living tissue cells in water by shock waves and mathematical model
Authors: Tamagawa Masaaki , Akamatsu Teruaki
Published: July 2002

In this study, the damage experiments of the living tissue cells (red blood cells and cancer cells) [M. Tamagawa and T. Akamatsu, Proc. 20th Int. Shock Wave Symp., Pasadena, USA, 517–518 (1995)], and the experimental model for cells (microcapsules including dye stuff) by plane shock waves are executed. It is shown from the results that using a microcapsule is an effective method for sensing injury to living tissue cells. To explain these phenomena, the living tissue cells are modeled mathematically as one and two spherical elastic shells filled with liquid toward the shock wave. Using stationary and transient analysis of a spherical shell, the dynamic characteristics of the one living tissue cell and two with a mutual interaction are evaluated. The results show that damage to living tissue cells depends on: (1) the elastic modulus of the cell, (2) the bulk modulus of intracellular material, (3) the thickness of the cell membrane, (4) the distance between the cells on shock-induced damage, and (5) the rise time of the shock wave.
Nine-year study of NHRA drag racing noise
Authors: Bricken Gordon , Jean Christopher
Published: July 2002

Pomona International Raceway, Pomona, California is the premier venue for drag racing sanctioned by the National Hot Rod Association. Community noise was studied for 9 years from 1984–1993. The project provided a unique long-term study of professional drag racing noise at 14 locations in a built-up community adjacent to the starting line. The object was to establish a basis for documenting the effects of noise mitigation’s planned for a renovated facility. The study revealed startling diurnal variations in noise, year to year variations, and the wide variety of emission characteristics of racing types. The simultaneous community and trackside locations permitted accurate correlation of events and established the basis for predictive models. The large data base permitted creation of a benchmark location by which the construction mitigation measures could be verified in a statistically valid manner. The study also revealed that the data spread is so wide that only multiple event sampling will provide an accurate picture of the community noise levels.
Approximation to study the acoustic field diffracted in the shadow of a half-infinite barrier by the boundary element method
Authors: Sanchis Antonio , Marín Albert , Romero José , Giménez Alicia
Published: July 2002

The application of the BEM (direct form) for the external domain, with a lineal combination of Helmholtz’s equation integral and their derivative (Burton, Miller) makes possible the calculation of the unique solution for each number of waves. The fundamental characteristics of this model that works with lineal sound source are: (1) The calculation surface is the topographical profile adapted with adjacent plans, discertizated in elements defined by two extreme nodes and one central node. (2) The acoustic impedance of each element is determined starting from the Delany equation. (3) The unknowns are the sound field in each node φi. (4) A form function φ is determined starting from the φi of each node. (5) Helmholtz’s operators are obtained by integrating the φ function in each element. (6) By solving the equation system with these operators, the values of the φi, and the function φ, are obtained. (7) By integrating Helmholtz’s equation integral starting from φ, the value of the sound field in each point of the space is obtained. Finally, the model is applied to several simple barriers for which there are experimental values. The comparison of the obtained outputs shows the validity of the calculation.
The calculation of curved path sound propagation in temperature inversion conditions
Authors: Bronsdon Robert L.
Published: July 2002

Sound propagation models have typically been used to develop noise control solutions for a wide variety of troublesome sources from highways, power plants, and airports to stadiums and outdoor amphitheaters. Almost all of these models assume that there is no variability in the atmospheric properties of the propagating medium but this assumption is almost invariably false. Predictions from these models are usually validated during the day when propagation conditions are favorable, producing lower levels than the models calculate. The problems usually come at night, when temperature and wind variations in the atmosphere create propagation conditions which are not favorable, resulting in much higher noise levels in the community, often on the order of 15–20 dB. This paper describes work which has been undertaken by the Walt Disney Company in an effort to produce a more precise computer model for these unfavorable conditions. The model developed is based on Gaussian beams, a hybrid technique developed to retain some of the fundamental aspects of ray tracing while incorporating some of the more advanced GFPE concepts. The fundamental code, developed by Ken Gilbert and Xiao Di, is being integrated into SOUNDPLAN, a commercially available sound propagation computer program.
Sound propagation through a turbulent jet: Experiment and theory
Authors: Ostashev V. E. , Blanc-Benon Philippe , Juvé Daniel
Published: July 2002

The coherence functions of a spherical sound wave after passing through a turbulent jet were measured for different values of sound frequencies and mean speed of the jet [Ph. Blanc-Benon, Ext. Rev. Cethedec-Ondes Signal, No. 79-1984 (1984)]. In the presentation, the experimental data obtained are compared with the theoretical predictions based on the previous theories of sound propagation in turbulent media with medium velocity fluctuations [e.g., E. H. Brown and F. F. Hall, Rev. Geophys. Space Phys. 16 (1), 47–110 (1978)] and with the predictions of the theory recently developed [V. E. Ostashev, Waves Random Media 4, 403–428 (1994)]. The differences between these theories and their results for the coherence function of a spherical wave are explained. It is also shown that the predictions based on the previous theories do not agree with the experimental data. On the other hand, the predictions of the theory developed by V. Ostashev fit the measured coherence functions fairly well. [This material is based upon work supported in part by the U.S. Army Research Office under Contract No. DAAH04-95-1-0593.]
Reducing the effects of masker uncertainty with harmonicity and onset/offset synchrony
Authors: Oh Eunmi L. , Lutfi Robert A.
Published: July 2002

Detection thresholds for a tone in an unfamiliar tonal pattern can be greatly elevated under conditions of masker uncertainty [E. Oh and R. A. Lutfi, J. Acoust. Soc. Am. 101, 3148 (1997)]. Two experiments were conducted to determine whether harmonicity and onset/offset synchrony of masker tones can reduce the detrimental effect of masker uncertainty. Masker uncertainty was introduced by randomly varying the frequencies and amplitudes of masker tones on each presentation. In experiment 1, maskers were composed of 2–49 tones that were multiples of 200 Hz, not including 1000 Hz. Anywhere from 2–10 dB less masking was observed for an inharmonically related signal at 1047 Hz than for a harmonic signal at 1000 Hz. In experiment 2, the onsets and offsets of masker tones covaried or varied independently of each other. As much as 30 dB less masking was obtained for the 1000-Hz tone signal when masker tones covaried in time. The results support the idea that harmonicity and onset/offset synchrony cause the signal to be perceptually segregated from the masker, thereby resulting in less masking.
A finite-element template for tongue modeling
Authors: Wilhelms-Tricarico Reiner , Wu Chao-Min
Published: July 2002

This report describes the design, data structure, and applicability of a refined finite-element template for the human tongue and connected oral structures. The finite-element model relies mainly on data sets from the Visible Human Project of the National Library of Medicine. The model is composed of macroblocks, which represent geometric subsections of the tongue. These blocks represent, in some cases, either individually or in combination, functional subsections, such as individual muscles. For each macroblock (or geometric region) a finite-element mesh can be generated such that the whole of the tongue can be modeled by a mesh of finite-elements, since the subdivision is compatible across block boundaries. The finite element template contains information about muscle tissue distribution of the tongue and velopharynx. Approximate muscle fiber directions are represented as direction fields. The model can be adapted to individual morphology if a set of morphological landmarks can be specified from measurements such as those from MRI data of an individual.
Mapping of muscle anatomy on three-dimensional magnetic resonance images of the human tongue based on morphological landmark selection
Authors: Wu Chao-Min , Wilhelms-Tricarico Reiner
Published: July 2002

The purpose of this study is to incorporate accurate muscle morphology in a computational 3-D tongue model based on anatomical literature and MRI data. Previous work [C.-M. Wu and R. Wilhelms-Tricarico, J. Acoust. Soc. Am. 100, 2659(A) (1996)] defined several landmarks within the 3-D space of the tongue tissue and developed a computational method to map Miyawaki’s drawing of muscle fiber directions [K. Miyawaki, Ann. Bul. RILP, Univ. Tokyo 8, 23–50 (1974)] onto the cross-sectional tongue images from the visible human data. In this study, the same technique is applied to identify a set of corresponding landmarks between Miyawaki’s data and MRI scans of a subject’s tongue forming a consistent articulatory posture. The geometrical mapping of these two data sets is achieved by the thin-plate spline mapping method. The result shows the sufficiency and robustness of the chosen landmark points to represent a general 3-D structure of the human tongue. This study is expected to provide a computational basis for quantitative comparison of MRI data across subjects and vowel types.
Articulatory based low bit-rate speech coding
Authors: Chennoukh Samir , Sinder Daniel , Richard Gael , Flanagan James
Published: July 2002

A voice mimic system has been designed to achieve an articulatory description of the speech signal based on an analytic description of the vocal tract (VT) parameters as a function of the first two formant frequencies. Articulatory based low bit-rate speech coding using the analytic mapping technique has been demonstrated. The natural speech input is analyzed to obtain the first two resonances of the VT, and the VT shape is estimated from the analytic acoustic-to-articulatory mapping. The shape is modeled using three parameters. Then, the articulatory parameters in addition to the frequency of the vocal cord vibration are transmitted. The receiver restores the VT shape from the articulatory parameters, computes the formant frequencies corresponding to the VT shape and synthesizes the output speech using a formant synthesizer. The bit-rate of the coding is determined from the parameter sampling rate and their digital representation. An intelligible output speech signal has been obtained for bit-rates as low as 624 bits/s using parameter sampling at 48 times/s, 3 bits for each of the VT parameters and 4 bits for the vocal cord vibration frequency. Coding at this bit-rate has been demonstrated on voiced sentences. [Research supported by ARPA DAST 63-93-C-0064.]
Name retrieval from pronunciation and spelling over the telephone, using HMM modeling and robust directory access
Authors: Gravier G. , Yvon F. , Etorre G. , Chollet G.
Published: July 2002

A system to retrieve names in a directory from their pronunciation and their spelling is presented, for telephone quality speech. Most of the previous approaches to letter recognition for spelled names are knowledge-based because of the high phonetic confusability between letters [C. S. Myers and L. R. Rabiner, J. Acoust. Soc. Am. 71(3), 716–727 (1982)] [Cole et al., Eurospeech’91, 479–482]. To avoid the knowledge-based approach, HMM modeling of speech units (SU) is used with word models for letters and phone models for name pronunciation. The directory is built automatically from a list of names using a grapheme-to-phoneme converter for the name and rules for the spelling, each entry in the directory consisting of several variants for both the pronounced and spelled names. From the acoustic recognition, the corresponding entry in the directory is found using DTW or a tree-search algorithm. Both methods allow insertion and deletion, and the cost for a substitution can either be fixed or determined from the train corpus confusion matrix. On a test database of 5000 names uttered by 3000 speakers (∼ 50 000 SU), 70% of the units are correctly recognized. Preliminary experiments based on spelled names with a DTW lexical search, showed that substitution costs defined using the confusion matrix significantly improved the results.
Crime automatic speaker verification and identification system
Authors: Gorban Igor I.
Published: July 2002

A new automatic speaker verification process was presented at the Eighth Annual International Conference on Signal Processing Applications and Technology (ICSPAT ’97), San Francisco, CA (I. I. Gorban, Robust Speaker Verification Algorithms). On the basis of these algorithms the crime automatic speaker verification and identification (CASVI) system has been created. The main feature of the system is that it is possible to verify and identify a speaker when the speech records are very corrupted and/or contain extraneous noise. Testing conducted under different conditions has shown that the CASVI system works well when the level of frequency distortions is lower than 35 dB and the level of the signal-to-noise ratio is lower than 12 dB. These results have been obtained for a new modification of the algorithms in which the adaptive threshold verification process is included and the identification algorithms are added. A description of the CASVI system, it’s algorithms, the methodology, and the results of the system testing will also be presented at the meeting.
MRI observation of dynamic articulatory movements using a synchronized sampling method
Authors: Masaki Shinobu , Tiede Mark K. , Honda Kiyoshi , Shimada Yasuhiro , Fujimoto Ichiro , Nakamura Yuji , Ninomiya Noboru
Published: July 2002

The use of magnetic resonance imaging (MRI) in speech research has, typically, been limited to static, sustainable vocal tract configurations due to the long acquisition times required [e.g., Baer et al., J. Acoust. Soc. Am. 90, 799–828 (1991)]. This work describes the application of a synchronized MRI sampling technique developed for cardiac motion imaging to the visualization of dynamic articulatory movements. Subjects produced 128 repetitions of a target utterance (/tata/ and /kaka/) coincident with audible tone bursts synchronized with gating pulses controlling MRI scanning. During each repetition, 35 scans at 25-ms intervals were acquired, each scan providing one row of frequency-domain data for each of the 35 output frames. After repetitive scans were completed, the full sequence was processed to give a 40-frames-per-second midsagittally oriented movie. Analysis of the resulting movies revealed that tongue shapes for the interconsonantal /a/ varied between the two utterance types, indicating a place-of-articulation effect of consonants on coproduced vowel tongue shape. This suggests that the synchronized sampling method provides a practical technique for investigation of the dynamic characteristics of articulatory movements during short utterances.
Geometrical display of speech spectra as an aid to lipreading
Authors: Hunter Eric J. , Strong William J.
Published: July 2002

A geometrical display of speech spectra intended as an adjunct to lipreading was developed. Spectra were calculated at 5-ms intervals from speech sound pairs ambiguous to lipreaders. The spectra were displayed as sequences of irregular decagons. Human subjects were asked to discriminate between pairs of spectral decagon sequences derived from pairs of ambiguous speech sounds. Subjects were able to discriminate between most of the visual spectral patterns derived from ambiguous sounds. However, spectral patterns associated with the voiced/unvoiced contrast in some stop pairs were not discriminated consistently.
Acoustic time reversal in nonlinear wave propagation
Authors: Hallaj Ibrahim M. , Kargl Steven G. , Barbone Paul E.
Published: July 2002

The application of acoustic time reversal in nonlinear underwater propagation is investigated theoretically and numerically. An acoustic wave equation containing thermal and viscous absorption as well as finite-amplitude terms is presented. The behavior of such a system under time reversal is discussed. Theoretical results and numerical examples are presented, showing when and why the propagation is time reversible, and when the equation is no longer time invariant. [IMH and SGK’s work is supported by the Office of Naval Research.]
The auditorium measurement chain: A study of the use of auditorium synthesis in the design of concert halls
Authors: LoVerde John J.
Published: July 2002

In the early 1960s, Paul Veneklasen began to develop an idea to synthesize an indoor acoustical space coming to fruition with the development and construction of an electroacoustic system called Auditorium Synthesis. This system allows the listener to preview various types of spaces prior to construction [Paul S. Veneklasen, Int. Symposium on Arch. Acoustics, 21–43 (1974)]. Listener preferential settings including source strength (G), early decay time (EDT), early/late energy (C), and lateral energy fraction from the Otis A. Singletary Center for the Arts (Lexington, Kentucky). A comparison of preferred orchestral settings and modeled seat locations will be performed. With impulse response data from the completed concert hall at the Singletary Center for the Arts, a comparison will be shown of newer parameters between the modeled seat locations from Auditorium Synthesis and concert hall data.
Paul S. Veneklasen, mentor and associate
Authors: Christoff Jerry P. , Ortega Jose C.
Published: July 2002

The authors both arrived at Paul S. Veneklasen & Associates and Western Electro-Acoustic Laboratory in 1956 and 1957, fresh from UCLA’s acoustics program which was part of applied physics. In an era when most people change jobs numerous times during their careers, what was the glue that kept the authors there? Security? Technical challenge? Opportunity to work with a truly creative, renaissance person? Probably all of the above. But more than acoustics and business were learned from Paul. A few anecdotes illustrate the affection for Paul and the legacy he left.