bluuebunny/crossref_metadata_2025 huggingface dataset:
Evaluation of a reiterant force impulse task: A constructive replication
Authors: Bunton Kate , Weismer Gary
Published: July 2002

In the current study, characteristics of labial force–impulse tasks were investigated under three speaking conditions using a minipressure transducer. In the task, neurologically normal adults were first asked to read aloud two sentences of different lengths (ten repetitions each) containing bilabial voiced and unvoiced consonants, and were then required to repeat the series of utterances using mimed speech. Mimed speech was produced with subjects holding their breath; subsequently, the experimenter cued subjects to use a normal airflow and the mimed utterances were repeated. Subjects were then trained to produced sequences of labial force impulses modeled on the real-speech utterances by substituting /ba/ or /pa/ for each syllable. The reiterant condition was also recorded as mimed. The goal of the analyses were to (a) compare the timing of the real-speech sequences to the timing of mimed and reiterant versions, and (b) compare the force magnitudes associated with labial stress produced during real speech to mimed and reiterant repetitions. Results will be discussed in terms of the relationship between orofacial, nonspeech motor performance and speech production performance. [Work supported by NIH Award No. T32 DC00042 and R01 DC00319.]
Magnetometer observation of articulation in sitting and supine conditions
Authors: Tiede Mark K. , Masaki Shinobu , Wakumoto Masahiko , Vatikiotis-Bateson Eric
Published: July 2002

Magnetic resonance imaging (MRI) has recently emerged as a useful tool for the investigation of the three-dimensional vocal tract shape in sustainable articulatory targets such as vowels and fricatives. However, existing scanning devices require that subjects phonate in a supine position. Previous work [Whalen, J. Acoust. Soc. Am. Suppl. 1 88 (1990)] compared MRI to cineradiographic data collected in a sitting posture, and reported limited effects of gravity on velar height differences in vowel production. This study used electromagnetometry [Perkell et al., J. Acoust. Soc. Am. 92, 3078–3096 (1992)] to investigate whether detectable systematic differences in midsagittal tongue posture exist between the two phonation conditions. Subjects fitted with nine transducers (four on the tongue, mandible, upper and lower lips, and two for head correction reference) produced both running speech and sustained vowels mimicking MRI protocol, repeated in sitting and supine postures. Analysis of separately recorded acoustics showed systematic effects of production posture on lower formant bandwidth. Results of comparing tongue height differences between the two conditions will be presented.
Three-dimensional visualization of electropalatography
Authors: Wakumoto Masahiko , Masaki Shinobu
Published: July 2002

Electropalatography (EPG) is a widely used noninvasive technique for monitoring dynamic tongue-palate contact patterns in speech research and speech clinics. However, the conventional two-dimensional (2D) visualization system is incapable of informing individual characteristics of three-dimensional (3-D) palatal shape or the relationship between electrode positions and anatomical landmarks (e.g., teeth). This paper reports a method for visualizing tongue-palate contact patterns on the 3-D palatal shape display. A convex palatal shape model was formed by silicone impression material from an individual dental cast, and electrode positions on the individual-specific artificial palate (RION SP-01, 63 electrodes) were marked on the surface of the convex model. The 3-D geometry and the surface texture information of the convex model were extracted by a laser beam scanner (Cyberware 4020). Then, palatographic data were collected through the EPG system (RION DP-20) and the tongue-palate contact patterns were mapped on the 3-D graphical display of the palate. Preliminary observation of Japanese fricatives revealed vertical differences in constriction between [s] and [■]. Its application to speech production research and speech training will be discussed.
A physiological model of the tongue and jaw for simulating deformation in the midsagittal and parasagittal planes
Authors: Dang Jianwu , Honda Kiyoshi
Published: July 2002

A physiological articulatory model was developed for simulation of 3-D deformation and movement of speech organs. The model is composed of a midsagittal portion of the tongue and the jaw and hyoid bone that yield rotation and translation. The tongue model consists of a midsagittal layer and two parasagittal layers with 1-cm intervals. The form and muscle geometry of the whole model are extracted from MR images of a speaker. All the soft tissue and rigid organs are modeled by node points with mass and two types of branches to adjacent points: soft branches for muscles and connective tissue between node points, and hard branches for bony organs. The muscles are defined based on an improved Hill’s model. Muscle activation signals, in part derived from EMG recordings, serve as the model’s control parameters. Preliminary experiments revealed that the model can produce realistic 3-D deformation and movement for vowels and a sustained consonant /r/. Since all the displacements of soft tissue and rigid organs are computed with the same algorithm, the model produces plausible dynamic patterns of the tongue–jaw complex with relatively short computational time in comparison with the finite-element method.
An application of laser-based ultrasonic nondestructive evaluation using a fiber-optics-based Fabry–Perot interferometer
Authors: Huber Robert D. , Candy James V. , Chinn Diane J. , Thomas Graham H.
Published: July 2002

Fiber optics lend increased flexibility to laser-based ultrasonic nondestructive evaluation (NDE). In this work, fiber-optic cables are used to transmit light from a laser to the detection site, and then from the detection site to a Fabry–Perot interferometer. The use of fibers allows both the detection laser and interferometer to be placed at a considerable distance from the object under test. A direct line-of-sight of the object from the main equipment is not required, since the fibers may be fed through walls and around obstacles. In addition, by containing the laser light in the fibers, the chance of accidental exposure to powerful laser beams that may otherwise be transmitted through air is decreased. Laser-based ultrasonics is generally less sensitive to traditional contact ultrasonics, and in addition, some light is lost in the coupling of laser light energy into optical fibers, further decreasing the sensitivity; thus the need for signal processing of the received signals is of great importance. In this work, the waveforms obtained using the Fabry–Perot interferometer and the corresponding signal processing performed on the data to enhance the resulting image for NDE are discussed.
Comparison of conventional acoustic and MIR radar/acoustic processing of speech signals
Authors: Gable T. G. , Burnett G. C. , Holzrichter J. F. , Ng L. C. , Lea W. A.
Published: July 2002

Applications of the micropower impulse radar (MIR) to speech research at the Lawrence Livermore National Laboratory has produced potentially new methods of speech processing. They include the accurate calculation of vocal tract transfer functions, formant, and pitch analysis, and basic phoneme synthesis. These speech parameters have traditionally been in the realm of all-pole LPC calculations. Related research using the MIR radar has supplied an increasingly accurate voiced excitation function, which makes possible transfer function calculations using both poles and zeros, yielding more accurate formant information and more natural sounding synthesis. This paper compares the newly obtained results with traditional LPC and cepstral approaches and demonstrates the improvements based on experimental data from several male and female subjects. The radar data also allow extremely accurate pitch tracking, which is simpler and more robust than that calculated by traditional means. This information can significantly enhance acoustic-based speech processing by allowing pitch adaptive processing.
Broadband matched-field processing
Authors: Baggeroer Arthur B.
Published: July 2002

Several algorithms have been proposed for broadband matched-field processing; however, the issue of signal coherence across frequency and its implications for the signal coherence has not been treated consistently. If one uses a wide sense stationary random process model for a passive source, then spectral bands are uncorrelated, and one is led to an incoherent sum of energy of the optimal beamformer across frequency bins. Active sources imply a model where there is signal coherence across frequency which can be characterized using the two-frequency coherence function. Here, it is important to preserve phase in the output of the optimum beamformer since it contains the group speed information of the several rays and/or modes of the propagation. Models for transient signals fall in between the passive and active ones. The limited duration of the transient contains multipath/multimode information; however, the signal waveform is usually not known so matched filtering is not possible. The presentation will relate the issues of signal coherence broadband MFP algorithms and provide examples of the performance under different model assumptions.
Absolute units for acoustic pressure levels: Shades of Carl Eckart and Bob Urick [ASA medalists as Pioneers in Underwater Acoustics, 1973 and 1988, respectively]
Authors: Fisher F. H.
Published: July 2002

In Principles of Underwater Sound (1946), edited and largely written by Carl Eckart, he showed what acoustic pressure levels in dynes per square centimeter (rms) meant in decibels (air and water) compared with one another and also in watts per square centimeter. In Principles of Underwater Sound by Bob Urick, 3rd ed. (1983), Sec. 1.6 deals very clearly with absolute units of intensity with respect to the new standard pressure reference level of 1 (μPa)=10−5 dyn/cm2 (rms) for both air and underwater acoustics; see ‘‘Preferred Reference Quantities for Acoustic Levels,’’ ANSI S1.8-1969(R1974). In physical units this intensity amounts to 0.67×10−22 W/cm2. A few examples will be given of difficulties encountered by not including absolute units, most recently with respect to whales and sound sources radiating only 250 W. The inclusion of absolute units would facilitate understanding of acoustics to the scientific, technical, and public communities.
Near-surface source depth resolution in a shallow-water environment
Authors: Hodgkiss W. S. , Murray J. J. , Booth N. O. , Schey P. W.
Published: July 2002

The depth discrimination capability of matched-field processing provides the possibility of separating surface from submerged sources of acoustic energy. Data collected in 200-m water during SWellEx-96 will be used to illustrate the achieved depth resolution in a downward refracting, shallow-water environment. During the event analyzed, two sources were towed simultaneously. The nominal depths of the deep and shallow sources were 60 and 10 m, respectively. Both projected unique, multitone transmissions with the deep source covering the 50–400 Hz band and the shallow source covering the 100–400 Hz band. In addition, the source ship herself (R/V SPROUL) had a detectable radiated acoustic signature across this same frequency region. The data were received by a 120-m aperture, 64-element vertical array deployed from the R/P FLIP. Broadband adaptive matched-field processing has been carried out on the data. The depth resolution characteristics of each of the three sources of acoustic energy will be presented. [Work supported by ONR, Code 321US.]
Paul S. Veneklasen, 1945–1996, an overview
Authors: Christoff Jerry P.
Published: July 2002

The author joined Paul S. Veneklasen & Associates and the Western Electro-Acoustic Laboratory in 1956, the day after his graduation from UCLA with a B.S. in Applied Physics and a specialization in acoustics. During the next forty years, he was privileged to work side by side with Paul Veneklasen on a variety of projects ranging from speech communication in noise, acoustical model testing, and a wide variety of architectual acoustical commissions. Most people think of Paul in only one field of endeavor, but those who knew him best knew that as a physicist, he would be interested and try anything to make an advance in technology. The paper provides a brief chronology of his major interests, projects, patents, and unpublished ideas.
The acoustical modeling of Paul Veneklasen
Authors: Siebein Gary W. , Gold Martin A.
Published: July 2002

The pioneering acoustical modeling work of Paul Veneklasen has served as an inspiration to much current work in the area, including that conducted at the University of Florida. Paul constructed small models of spaces he was designing and conducted a variety of acoustical tests in them to assist in making design decisions on many innovative rooms. He determined criteria by which to evaluate the acoustic response of the rooms obtained from the models from listening to simulated sounds in his laboratory. These criteria included: providing a strong direct sound throughout the room; providing reflections from overhead within 40 ms or less of the direct sound for clarity and loudness; providing lateral reflections within 80 ms of the direct sound for envelopment; and providing a rich reverberant sound across a full range of frequencies. These criteria are still used today as the basis for interpreting the impulse responses obtained in the design process of rooms. The experiments he engaged in as a part of the process of many rooms involved research and design components including listening, modeling, designing, and building can be traced in concept to the work of Wallace Sabine. This process has become an established approach to acoustical design today.
Sound radiation and postglottal filtering in frogs
Authors: Purgue Alejandro P.
Published: July 2002

This work presents a comparison across selected species of several aspects of the mechanism of sound radiation in frogs. Measurements of the magnitude of the transfer function of the radiating structures show that the structures radiating the bulk of the energy present in the call vary depending on the species. Bullfrogs (Rana catesbeiana) radiate most of the energy (89% sound level) in their calls through their eardrums. Additionally, the transfer function of the eardrums displays peaks coincident with those observed in the mating and release call of this species. The vocal sac and gular area contribute energy only in the lower band (150–400 Hz). The ears are responsible for radiating additional frequency bands to the ones radiated through the vocal sacs. In Rana pipiens, the ears also broadcast a significant portion of the energy present in the call (63% sound level), but the frequencies of the aural emissions are a subset of those frequencies radiated through the vocal sacs. Finally, the barking treefrog (Hyla gratiosa) appears to use two different structures to radiate his call. The low-frequency band is preferentially radiated through the lungs while the high-frequency components are radiated through the vocal sac.
Response of elephant seals to acoustic thermometry of ocean climate sound transmissions
Authors: Costa Daniel , Crocker Daniel , Gedamke James , Webb Paul , Le Boeuf Burney , Waples Danielle , Hayes Sean , Ganong James
Published: July 2002

The hypothesis that northern elephant seals would respond to acoustic transmissions from the ATOC sound source was tested. Elephant seals were chosen because they have the best low-frequency hearing of any pinniped, are abundant, naturally migrate past the Pioneer Seamount, and are deep divers. ARGOS satellite tags provided information on animal location while at sea, while archival tags provided information on swim speed, time, depth, ambient acoustic environment, and ambient sound pressure levels. Instruments were deployed on 14 naturally migrating adult male elephant seals that were expected to swim near the source site, and upon 29 juvenile animals that were translocated and released 1 h prior to transmission. Measured mean intensity of ATOC exposure ranged from 120 to 135 dB for 60–90 Hz (n=6) compared to ambient levels of 100–107 dB (60–90 Hz). Animals did not alter return track, diving pattern, or swim speed, did not go to the surface, and often continued to dive closer to the sound source if on the descending segment of a dive. [Work funded by ONR and ARPA.]
Quantitative and qualitative analysis of the ascendent frequency whistles of tucuxi, Sotalia fluviatilis, in the Sepetiba Bay, Rio de Janeiro, Brazil
Authors: Simao Sheila M. , Figueiredo Luciana D. , Siciliano Salvatore
Published: July 2002

Sotalia fluviatilis was the only cetacean species observed in all 16 boat surveys (82 h) conducted in the Sepetiba Bay (22° 58′, 44° 02′W), between December 1993 and August 1995. During the boat survey of 5 January, 1995—a very calm day (Beaufort = 1)—it was possible to make good subaquatic sound recordings (30 min) of a group of ca. 20 young and subadult dolphins. As there is little bioacoustic information on tucuxi, sonogram analysis was used to characterize the tucuxi’s whistles of ascendent frequency. The qualitative analysis resulted in 866 whistles divided in 48 types. Some of these sonograms shapes match the ones presented by other cetacean species sonograms (Tursiops truncatus, Delphinus delphis, and Globicephala macrorhynchus). The quantitative analysis was based on: number of harmonics; number of inflexions; beginning, final, maximum, and minimum frequecies; duration; and frequency modulation. Minimum, maximum, and average values, standard deviation, and coefficient of variation were calculated for each type of whistle. The S. fluviatilis whistles are extremely diverse (high coefficient of variation values for each type) and frequent (38.5 whistles/min), seeming to have an important role in social relations among these animals. Such whistle diversity and quantities could be explained by the composition of the dolphin group, i.e, it was exclusively formed by young and subadult animals.
A sonobouy array for two-dimensional location of dolphin vocalizations
Authors: Howarth Eric S. , Defran R. H.
Published: July 2002

A four-sonobouy array was deployed 0.25 km offshore in southern California waters. The array was developed and deployed to passively locate coastal bottlenosed dolphins (Tursiops truncatus). Sonobouy mooring sites were selected to overlap areas shown by the Cetacean Behavior Laboratory at San Diego State University to be highly utilized by these coastal dolphins. Arrival-time-difference measurements were used to provide two-dimensional locations of underwater biological sounds such as dolphin vocalizations. Sonobouys were separated by a range of between 367 and 1585 m and the accuracy of source location depended on the relative proximity to the nearest sonobouy. Global Positioning Satellite and theodolite data were used to verify sonobouy locations and, in turn, the acoustic arrival-time differences of on-board calibration noises. Bottlenosed dolphin whistles, including mimics of the calibration sound, were among the biologically identifiable sounds tracked by the array. Other acoustic signals recorded but not located were snapping shrimp (Californiensis dentripes) and sources judged to be fish. Recorded sound levels varied over a considerable range, even on identical hydrophones, and were probably due to the signal transmission distance, shallow-water propagation effects, and diurnal fluctuations in the sources.
Experimental study of cavity resonance suppression methods
Authors: Johnson J. Scott , Mongeau Luc G.
Published: July 2002

Flow-excited cavity resonance is a common, generally undesirable, sound-generating phenomenon that occurs in many engineering applications. In this study, low-speed cavity resonance suppression techniques were investigated. The focus was on two methods: leading edge spoilers and leading edge air mass injection. Experiments were conducted in a closed section wind tunnel, which included a secondary blowing apparatus for controlling boundary layer thickness. The cavity, attached to the wind tunnel test section, acted as a side-branch resonator. Smoke flow visualization was performed for various operating conditions. For example, a nondimensional spoiler length of 0.156 (based on cavity length) inclined at a 50-deg angle of attack typically provided a 12 dB (re: 20 μPa) reduction of the 140-dB, 124-Hz tonal cavity pulsation. Air mass injection through a perforated array at a rate of 2.8 liter/s yielded a 27-dB reduction. In general, the results suggested that such leading edge suppression techniques yield greater orifice shear layer thicknesses, similar to that of increasing the boundary layer thickness. The thickened shear layer may cause greater vortex diffusion, a key parameter establishing the level of the cavity pressure fluctuations. Diffuse vortices lead to weaker cavity pressure oscillations and lower vortex convection velocities.
Turbulent flow noise estimate by use of a velocity sensor embedded in a three-layered composite structure
Authors: Ko Sung H.
Published: July 2002

A theoretical model was developed to evaluate flow noise levels received by a velocity sensor that is embedded within an outer decoupler mounted on a low acoustic impedance surface. The model, a three-layer structure, consists of a layer of nonvoided elastomer (outer decoupler), which covers a layer of microvoided elastomer (inner decoupler) backed by an elastic plate. The rubberlike material of the outer decoupler is designed to reduce the flow noise generated by turbulent boundary layer pressure fluctuations, and the low acoustic impedance (soft) material of the inner decoupler is configured to reduce the flexural wave noise generated by the vibration of the elastic plate. The upper surface of the outer decoupler is in contact with turbulent flow (water) and the lower surface of the backing plate is in contact with semi-infinite space (air). This work uses the model of the three-layer composite structure for the analyses of both the flow noise and signal levels received by the embedded velocity sensor. The final noise levels are the equivalent plane-wave levels, which were obtained by subtracting the signal levels from the flow noise levels. [Work supported by the Office of Naval Research, Code 321SS.]
Understanding and following a jazz improvisation in real time: A statistical approach
Authors: Toiviainen Petri H.
Published: July 2002

A system for following an improvised jazz solo and inferring the current location in the chord progression is presented. The system will ultimately work in real time in a MIDI environment: It listens to an improvised jazz melody played on a MIDI instrument and joins in the performance with a synthesized rhythm section. It is comprised of two stages: a beat tracker and a performance analyzer. The beat tracker is based on a continuously adapting nonlinear oscillator, which synchronizes to the beats of the improvised solo by continuously adapting its phase and period. It can follow local variations of tempo and produce expectations of next beats. The performance analyzer attempts to infer the current location in the harmonic progression by comparing the notes played in the improvised melody with statistical distributions of notes obtained from improvisations of well-known jazz musicians [T. Jarvinen, Tonal Dynamics and Metrical Structures in Jazz Improvisation (Dept. of Musicology, Univ. of Jyvaskyla, Finland) and trying to find the best match from several possible alternatives. [Work supported by the Academy of Finland.]
A superior Kirchhoff method for aeroacoustic noise prediction: The Ffowcs Williams–Hawkings equation
Authors: Brentner Kenneth S.
Published: July 2002

The Lighthill acoustic analogy, as embodied in the Ffowcs Williams–Hawkings (FW–H) equation, is compared with the Kirchhoff formulation for moving surfaces. A comparison of the two governing equations reveals that the main Kirchhoff advantage (namely, nonlinear flow effects are included in the surface integration) is also available to the FW–H method if the integration surface used in the FW–H equation is not assumed impenetrable. The FW–H equation is analytically superior for aeroacoustics because it is based upon the conservation laws of fluid mechanics rather than the wave equation. Hence, the FW–H equation is valid even if the integration surface is in the nonlinear region. This is demonstrated numerically for helicopter rotor applications in the paper. The Kirchhoff approach can lead to substantial errors if the integration surface is not positioned in the linear region (i.e., if the input data are not a solution to the wave equation). These errors may be hard to identify in some cases.
Numerical simulation of superoscillations of a BSA-bearing drop in microgravity
Authors: Chen Xiaohui , Apfel Robert E. , Shi Tao
Published: July 2002

Large-amplitude nonlinear oscillations of a water drop, radius 2.53 cm and aspect ratio 2.10, with surfactant BSA (bovine serum albumin) 1.0×10−5 g/ml(1CMC), is numerically simulated based on the boundary integral method. BSA is a globular, large molecular-weight protein. Two surface viscosities, surface dilatational viscosity and surface shear viscosity, as well as the Marangoni effect, are considered. The high-surface viscosities make the rotational velocity the same order as the irrotational velocity in the boundary layer. A boundary-layer method is used to calculate the rotational velocity. When the surface dilatational viscosity is 0.5 sp and the surface shear viscosity is 0.15 sp, the numerical simulation results are in good agreement with the experiment results observed in the space shuttle during the second United States Microgravity Laboratory, USML-2, in October, 1995. The evolution of the drop oscillation for both experiment and simulation is given. The change of the BSA distribution along the surface, and the relative importance of the rotational and irrotational velocities, as the drop oscillates is also shown. [Work supported by NASA through JPL, Contract No. 958722.]
Measurements of the transient response of single-bubble sonoluminescence subject to an abrupt change in the drive pressure amplitude
Authors: Matula Thomas J. , Crum Lawrence A.
Published: July 2002

A novel method utilizing abrupt drive pressure amplitude changes is described for exploring the dynamics of single-bubble sonoluminescence. Air bubbles that are initially below the luminescence threshold are subject to an abrupt increase in the drive pressure amplitude. The resulting turn-on time for light emission is qualitatively similar to nitrogen bubbles, except that over time scales of seconds, the light intensity of an air bubble increases [B. P. Barber et al., ‘‘Sensitivity of sonoluminescence to experimental parameters,’’ Phys. Rev. Lett. 72, 1380 (1994)], while the intensity from a nitrogen bubble remains low. However, when we begin with a stable sonoluminescing air bubble above the luminescence threshold, and then generate an abrupt decrease in the pressure amplitude below the threshold (no light regime), followed again by an abrupt increase back to its original (sonoluminescing) state, the resulting turn-on time for light emission appears qualitatively similar to noble gas bubbles: sudden, and relatively intense, unlike what occurs for air bubbles that were initially below the luminescence threshold. These measurements imply that sonoluminescence from an air bubble depends on the time the bubble spends in the sonoluminescing state. [Research supported by NSF.]
First and second pitch shift effects as alternative manifestations of a single phenomenon
Authors: Vassilakis Pantelis N.
Published: July 2002

The second pitch shift effect describes a drop in the perceived pitch of a complex stimulus when the frequency spacing among its components is increased. Based on existing experimental data describing the first pitch shift effect, the present paper demonstrates that the mathematical model introduced by de Boer [Doctoral dissertation, University of Amsterdam (1956)] and modified later by Smoorenburg [J. Acoust. Soc. Am. 48, 1055–1060 (1970)], predicts without any further modification the second pitch shift effect as well. The Smoorenburg model also predicts that the perceived pitch of complex stimuli will not always drop when increasing the frequency spacing among components, but may rise depending on the structure of the stimuli. A perceptual experiment was conducted using nine complex stimuli. For eight of the stimuli, the model predicted a rise in pitch with increasing frequency spacing while for the ninth stimulus the opposite was predicted. The results of the experiment support the conclusion that the relationship between the direction of pitch motion (rise/drop) and the direction of changes in frequency spacing among components of complex stimuli (increasing/decreasing) can be predicted by the same model that explains the first pitch shift effect, making the second pitch shift effect an unnecessary concept.
A psychophysically based model of consonant perception by multichannel cochlear implant users
Authors: Meyer Ted A. , Svirsky Mario A. , Castor Michael B. , Falsafi Sassan , Simmons Peter M.
Published: July 2002

Despite advances in implant technology, cochlear implant (CI) users demonstrate a wide range in the ability to perceive speech in the absence of visual or contextual cues. Some progress has been made recently in the ability to explain perceptual performance with a CI. A quantitative, psychophysically based model of vowel perception by CI users of the SPEAK processing strategy generated confusion matrices that were remarkably similar to actual data [M. A. Svirsky and T. A. Meyer, Assoc. Res. Otolaryngol. Abs. 20, 59 (1997)]. The three dimensions of the parameter space are the centers of gravity of stimulation in the cochlea in response to the first three vowel formants. The free parameter of the model is the subject’s ability to scale pitch percepts associated with different electrodes. Although this single-parameter model was not successful in predicting consonant confusions, when the model was expanded to incorporate psychophysical estimates of gap detection and high- versus low-frequency intensity difference discrimination, the model successfully predicted performance on a medial consonant test [Meyer et al., Conf. Implant. Aud. Prosthes. Abs. (1997)]. Relations between predicted and obtained performance on selected psychophysical tests will be discussed. [Work supported by NIH.]
Discrimination of single-formant stimuli by chinchillas (Chinchilla villidera)
Authors: Holt Lori L. , Kluender Keith R. , Lotto Andrew J.
Published: July 2002

The present study assessed the behavioral capacity of six chinchillas (Chinchilla villidera) to discriminate complex acoustic differences, like those observed for the first formant (F1) of human speech, using operant methods. In a minimum-uncertainty task, chinchillas discriminated single-formant stimuli with fundamental frequencies of 100 or 200 Hz, 60 Hz (Q6) bandwidths, and center frequencies ranging from 2.5–7.0 Bark (approximately 253–709 Hz) in 0.25-Bark steps. Each trial began with the repeated presentation of a ‘‘target’’ stimulus. The chinchillas’ task was to discriminate whether this repeating target changed in center frequency or remained constant throughout the trial. The chinchillas completed 20 experimental sessions for each of six conditions in which fundamental frequency and center frequency of the target stimulus were varied. Stimulus characteristics were designed to reveal perceptual consequences of interactions between individual harmonics and overall spectral shape of the single formant. Behavioral sensitivities were estimated by d′ and percent correct. Data will be discussed in the context of human psychophysical performance and observations of single-unit responses in the chinchilla cochlear nucleus from studies utilizing the same stimulus corpus. [Work supported by NSF.]
The beginnings of word recognition in infancy
Authors: Jusczyk Peter W.
Published: July 2002

At around 7.5 months, English-learning infants begin to show some ability to segment words from fluent speech. Moreover, there is evidence that infants at this age begin to engage in some long-term encoding and retention of information about the sound patterns of words that occur frequently in fluent speech. Infants’ memory for these sound patterns apparently does not depend on their pairing with any concrete referents. Thus, at least in some instances, word learning may occur by first storing information about sound patterns, and then subsequently attaching a meaning to these. Additional findings from our laboratory suggest that these early representations of the sound patterns of words are rather detailed with respect to phonetic properties and may include information about such indexical properties as talker identity. [Work supported by NICHD and NIMH.]
Predicting the perception of words from the perception of phonemes
Authors: Nearey Terrance M.
Published: July 2002

The present paper explores the degree to which the perception of CVC syllables can be factored into the perception of its constituent phonemes. Categorization experiments with synthetic speech typically manifest such factorability to a remarkable degree (regardless of lexical status of the syllables.) Such factorability is also compatible with intelligibility studies of English CVCs [ e.g., A. Boothroyd and S. Nittrouer, J. Acoust. Soc. Am. 84, 101–114 (1988)]. These show that the identification of nonsense syllables can be predicted as the product of the probabilities of identification of its phoneme parts, while word identification is systematically higher. Simulation studies are reported here involving a ‘‘factored’’ perceptual model that consists of a set of phoneme-likelihood estimators whose outputs are modulated by prior probabilities related to lexical status. This model can approximate the patterns observed in human perception. Simulations were also run with nonfactorable models, where syllables and words involve information about unique stimulus properties that cannot be predicted from their constituent phonemes. Consistent with a conjecture of Allen [J. Allen, IEEE Trans. Speech Audio Process. 2, 567–577 (1994)], such syllable template models do not produce behavior compatible with human performance. [Work supported by SSHRC.]
Sound radiation from motion-related ice processes in the central Arctic
Authors: Stamoulis Catherine , Dyer Ira
Published: July 2002

The radiation characteristics of individual acoustic events from the central Arctic SIMI experiment data were studied, in the frequency range 10–350 Hz, with the purpose to identify the event generating ice mechanisms. Four event subpopulations and consequently four ice mechanisms were distinguished. The first subpopulation consisted of events attributed to floe unloading, a postfracture process, which was consistently detected in the ambient noise data following intervals of high ice fracture activity. The event radiation patterns were independent of azimuth and were modeled by a stationary vertical dipole. The second subpopulation consisted of events attributed to tensile fracture, the radiation pattern of which was modeled by a weighted superposition of longitudinal octopoles, modified by a Doppler factor to account for source motion. The last two subpopulations consisted of events attributed to shear fracture. A lateral octopole and a weighted combination of lateral and longitudinal octopoles, both modified by a Doppler factor, were used to describe the radiation characteristics of events in these subpopulations, respectively. The adequacy of the latter model indicated that events in the fourth subpopulation had been induced by shear fractures, which propagated through the formation of arrays of tensile cracks in their tips and edges.
A microprocessor-based tag to record acoustics experienced and generated by migrating northern elephant seals
Authors: Burgess William C. , Tyack Peter L. , LeBoeuf Burney J. , Costa Daniel P. , Crocker Daniel E.
Published: July 2002

Two microprocessor-based acoustic recording tags, deployed on northern elephant seals migrating from California to Alaska, were recovered in August 1997 after over 4 months at sea. The tags’ hard disks contained measurements of pressure, temperature, and ambient noise, as well as acoustic signatures of swim speed, swim stroke rate, respiration, and cardiac function, obtained as one subject (‘‘Trane’’) swam across the northeastern Pacific and the other (‘‘Rocky’’) north along the West Coast. The subjects dove an average of 58 (Trane) and 81 (Rocky) dives per day during the 26 days that logger batteries supported data acquisition; maximum dive depths reached 780 m (Trane) and 770 m (Rocky). To function over prolonged periods under these demanding conditions, the tags were designed with robust enclosures and software- and hardware-based power conservation techniques. The hydrophones were potted in rear-facing conical resin housings, which were found empirically to reduce flow noise over 100–200 Hz by 10–15 dB compared with semicylindrical and tapered cylindrical hydrophone housings. The results suggest that hydrophone tagging methods offer a comprehensive and reliable means of sampling acoustic stimuli and associated behavior for free-ranging marine animals over long periods at sea. [Work supported by ONR and MBARI.]
Learning about sound with interactive exhibits
Authors: Feher Elsa
Published: July 2002

What and how can visitors learn from the interactive exhibits in a science museum? Using examples of actual exhibits that deal with sound, we will examine the process of developing them, and the ways in which visitors use them and interpret the phenomena that are displayed. a)Former director, Reuben Fleet Sci. Ctr.
A new graduate program in acoustics
Authors: Brown David A. , Payton Karen
Published: July 2002

The University of Massachusetts Dartmouth is offering a new graduate program in the area of Marine Acoustics leading to the degree of Masters of Science in Electrical Engineering with a specialization in Marine Acoustics. The Marine Acoustics specialization is multidisciplinary in nature bringing together the physical, mechanical, and electrical engineering aspects of the study of the production, propagation, and reception of underwater sound. The graduate program is currently running on campus as well as off-site at the Naval Undersea Warfare Center (NUWC) in Newport for as many as 60 students. Students in the NUWC program have completed the first 4 courses of an 11-course program consisting of a Mathematics Review (ECE403), the Fundamentals of Acoustics (ECE 557), Random Signals (ECE 521), Underwater Acoustics I (ECE 597), Underwater Acoustics II (ECE 597), Acoustic Transduction (ECE699), Digital Signal Processing (ECE574), Sonar Signal Processing (ECE575), Sonar Systems Engineering (ECE675), and a graduate thesis or two additional graduate courses (e.g., Computer Systems, Nonlinear Acoustics, etc.). The program is expected to provide the student with a good foundation in acoustics. Details will be elaborated on during the presentation. [Program sponsored by NUWC and UMass Dartmouth.]
Demonstrating the physics of brass musical instruments
Authors: Holmes Brian W.
Published: July 2002

Brass musical instruments, unlike woodwinds, have complicated shapes: a cup-shaped mouthpiece, a conical leadpipe, a cylindrical valve section, and a flared bell. A trumpet will be built, allowing the acoustical significance of these parts to be demonstrated. It will be shown why brasses rely on valves to get different notes, unlike woodwinds, which use side holes. Before the invention of the valve, hornplayers learned to augment their meager supply of open notes by changing the position of the hand in the bell. Although the modern horn uses valves, the right hand remains important. The acoustical and musical significance of the hand will be explained and demonstrated. The talk concludes with a performance of Beethoven’s Sonata for Horn and Piano, Op. 17, performed on a valveless horn similar to those in use in Beethoven’s time.
Analog and digital signal processing in a high-performance miniature sound-level meter
Authors: Goldberg Jack , Wong George S. K.
Published: July 2002

The design of a relatively low-cost Type 2 pocket-size sound-level meter with an embedded 8-bit microcontroller is described. The key design goal was to provide 110 dB of dynamic range without the need for user-controlled range switching. The miniature instrument meets all the current ANSI S1.4-1983 (R 1990) and IEC 651 (1979) sound-level meter standards. Relevant features in the future IEC 1672 sound level meter standard, such as level linearity, tone-burst response, over-range and under-range indication, measurement accuracy, and resolution are reviewed. Due to judicious separation of the functions performed by analog and digital hardware and software, the wide dynamic range is implemented with only 8 bits of A–D conversion. Extensive type testing confirmed that the instrument can satisfy the requirements of the standards over the required temperature range. [Work supported by Etymotic Research, Elk Grove Village, IL.]
Noise control under successive fulfillment of jobs specified by various degrees of labor tension
Authors: Levin Mark V. , Menshov Alexander A.
Published: July 2002

The Standards of occupational noise (A-weighted) in Russia, the Ukraine, and other countries in the former USSR are differentiated by degrees of labor tension (40–80 dB). Frequently there is a real necessity to solve the problem of noise control, which is summed up for a full-time day during which labor is exposed to different noise levels. The problem is addressed by the definition of equivalent sound shift level—the noise working loudness. This definition correctly sets forth one noise standard during a shift of one action, i.e., the fulfillment for a particular period by a worker of identical labor tension. The decision of this task is complicated by the presence during the shift of work distinguishing not only the noise levels but different labor tension. Under the circumstances, it is advisable to sum up noise control according to the quotient of noise loudness (QNL). It is defined by the overage in time exceeding the standards of sound level for every separate period of labor with different degrees of labor tension. The quotient of noise loudness (QNL) is calculated by the proposed formula. It allows one to make noise control under successive fulfillment of jobs specified by various degrees of labor tension.
Interaction between prosody and discourse structure in a simulated man–machine dialogue
Authors: Eklund Robert
Published: July 2002

Automatic speech understanding systems are beginning to attain a level of sophistication where commercial applications are within reach. However, if humans and machines are ever going to communicate in a natural way, it is of vital importance that language modeling go beyond the sentence level. A profound understanding of discourse structure is required, and to this end, knowledge concerning how prosody interacts with other linguistic phenomena is needed. Not only will better prosodic modeling of discourse lead to better speech recognition/understanding, it will also yield more natural-sounding speech synthesis. This paper reports on a dialogue/prosody project at Telia Research, Sweden. A Wizard-of-Oz simulation of a computerized reservation system was used to collect realistic speech data. Fifty subjects were given three tasks each that entailed the reservation of flights, trains, car hire, and hotel reservations. To avoid linguistic influence on the subjects’ utterances, the tasks were given as maps and icons. A ToBI-style analysis was applied, adapted to meet language-specific requirements. The dialogues were analyzed with regard to phrase boundaries, tones, disfluencies, syntax (functions/categories), new versus given information, and pitch range. This paper describes our observations concerning the interaction between prosodic, syntactic, and higher-level linguistic phenomena, such as discourse structure.
Pathologic discrimination through acoustic analysis of voice
Authors: de Oliveira Rosa Marcelo , Pereira Jose Carlos , Araujo Simone Adad , Grellet Marcos
Published: July 2002

The acoustic analysis of voice from dysphonic patients has been studied, mainly for classification of voice quality in terms of roughness, breathiness, and hoarseness. By a data reduction of speech signal, researchers have produced several acoustic measures in order to relate them with a perceptual diagnosis. In this work, a proposal was made to modify acoustic parameters in order to support a higher sample quantity, instead of using only a data window by digitalized speech. Also, robust statistics is used to obtain realiable measures that estimate the abnormal condition of a voice signal as well as to discriminate several pathologics of the larynx and vocal cords, without any perceptual consideration. Using only seven measures from adaptative inverse filtering (Kalman and Wiener filters) and a neural network for pattern classification on 228 voice signals of speakers (20 groups of distinct types of pathologies and one of normal speakers) evaluated through videolaringoscopy, results of the pathologic discrimination showed 22% of uncertainty. Redrawing the patient sample in seven distinct groups (five with defined pathologies, one with all others pathologies, and one control), then the error in the pathologic discrimination is 18%. [Work supported by Capes.]
Acoustic cues to syntatic structure
Authors: Tjaden Kris , Shapiro Lewis P. , Nicol Janet , Swinney David
Published: July 2002

Duration and fundamental frequency (f0) are used by listeners in on-line sentence processing as cues to syntactic structure [H. Nagel, L. P. Shapiro, and R. Nawy, J. Psycholing. Res. 23, 473–485 (1994)]. Research suggests, however, that speakers may not reliably use duration and f0 to disambiguate syntactic structures [D. W. Albritton, G. McKoon, and R. Ratcliff, J. Exp. Psych. 22, 714–735 (1996)]. The present study explored prosodic differences in a variety of syntactic structures for sentences read by five speakers. Speakers failed to utilize f0 or duration to disambiguate verbs followed by either a direct object or sentential clause; this finding does not support Nagel et al.’s (1994) results for one speaker. In contrast, intransitive verbs were characterized by longer durations and greater f0 excursion as compared to transitive verbs. Acoustic differences for noun phrases (NPs) following transitive and intransitive verbs also were present; NPs following transitive verbs were longer in duration and were associated with a steeper f0 change, as compared to NPs following intransitive verbs. The results suggest that speakers may distinguish some but not all syntactic structures using the prosodic cues of f0 and duration; implications for theories of on-line sentence processing will be discussed. [Work supported by NIH DC00494, DC03347.]