bluuebunny/crossref_metadata_2025 huggingface dataset:
Application of cardioid beamforming in both impedance tube and free-field measurements of ground impedance
Authors: Swanson David C. , Maniet Edward
Published: October 2005

Measurements of the specific acoustic impedance of ground covered by institutional grass were performed using both a cardioid beamforming method and the established transfer function method [J. Y. Chung and D. A. Blaser, 907–921 (1980)]. The cardioid method steers a beampattern to separate the incident and reflected waves. Both methods were applied to impedance tube and free-field measurements. With proper calibration, the cardioid and transfer function methods are mathematically identical and produce similar experimental results. Comparison of the impedance tube and free-field measurements suggest that the latter approach produces more reliable results. However, due to practical limitations of the source/receiver geometry free-field measurements at low frequencies are more complicated in that the plane wave assumption is not valid.
Multichannel block adaptive filters for real-time, nonparametric identification of time-varying plants
Authors: Nowlin William C. , Guthart Gary S. , Toth Gregory K.
Published: October 2005

Many real-world applications of active noise control are characterized by transfer functions that vary significantly and unpredictably. The controller’s transfer function models must adapt to these variations. Presented here is a class of adaptive filters that accomplish quasiperiodic system identification updates for feedforward control by using blocks of input–output histories. The algorithms form a one-dimensional family linking normalized LMS adaptive filters and optimal Wiener filters, and are termed ‘‘block projection’’ algorithms. The system identification proceeds noninvasively, producing nonparametric (FIR) impulse responses. The multichannel generalization and application of these algorithms to system identification, as presented here, is novel. Considerations are described that arise from the algorithms’ implementation in the context of system identification; in particular, the proper weighting of input and output data pairs is discussed. The resulting multichannel control algorithms have been implemented successfully for quieting of a compact distributed source in an anechoic environment, and for local quieting of a diffuse field in a reverberant room. In both cases, error microphones could be moved about, providing a ‘‘mobile quiet zone,’’ and performance was obtained for bandwidths exceeding a decade.
Active control of sound transmission through a thin metal plate
Authors: Bao Xiaoqi , Varadan Vasundara V. , Varadan Vijay K.
Published: October 2005

The performance of active sound isolation by using a piezoelectric sensor and an actuator on a thin plate is investigated experimentally. In the experiment, the aluminum plate covers an opening of an acoustic enclosure, where a sound source is located. The outside acoustic field is measured by an 11-element microphone array, which scans on a surface of a hemisphere. The isolation performance of the original plate is poor at its resonance frequencies. The sound transmission from inside to outside of the enclosure is actively controlled by a one-sensor one-actuator control system which minimizes the output of the on-plate sensor. Global reductions of 15–20 dB are achieved at the first three resonance frequencies.
Principal component analysis applied to the active control of turbofan inlet noise
Authors: Cabell Ran , Fuller Chris
Published: October 2005

An analytical study of the active control of fan noise in a turbofan engine with wall-mounted acoustic sources is described. The transfer functions between the acoustic sources and microphones outside the duct are decomposed into principal components (PCs). The active control problem is then solved using a subset of the principal components. The subset consists of PCs that are statistically significant for accurately modeling the primary radiated noise. The resulting elimination of certain PCs is a mathematically efficient way to optimally reduce the number of channels of control. In certain cases it can be shown that the PCs describe the radiating modes of the control system. This provides substantial insight into the mechanisms of control, and can be used in the controller to derive specific modes to the exclusion of all other modes. [Work supported by NASA Langley Research Center.]
Optimal design of enclosures for active noise control and isolation
Authors: Varadan Vasundara V. , Kim Jaehwan , Varadan Vijay K.
Published: October 2005

An optimal design technique is used to find the optimal conditions for several parameters of an enclosure to reduce the noise radiated from an internal source. Design variables in the optimization procedure are the size and thickness of the enclosure, size and thickness of actuators, the location of actuators, and the applied voltage on the actuators. The objective function in optimal design is the total radiated sound power at far field. The practical limits of the design variables are taken as constraints. The acoustic field acts as a distributed load on the walls of the enclosure. A 3-D finite element model is used to model the walls and the PZT actuators. With an applied voltage on the actuators and with acoustic pressure loading, the velocity distribution on the outside of the walls can be found by solving the finite element equations. These are used in the Helmholtz integral representation to compute the radiated sound field to the exterior of the enclosure. The design is performed at a single frequency which may represent the dominant frequency in the noise spectrum. By comparing the initial design and the optimal design results, a noise reduction of 12–18 dB has been achieved.
Experimental investigation of low-frequency noise from jet engine test cells
Authors: Gibson Robert G.
Published: October 2005

Low-frequency noise emission from facilities designed for the ground runup testing of jet engines can be a source of annoyance to communities near military air bases. Noise measurements, using arrays of microphones and accelerometers at a number of U.S. Air Force hush houses, provide information regarding the source characteristics. Comparisons are made between noise from a single aircraft both inside and outside a hush house. Source location calculations indicate that low-frequency noise originates from a region near the trailing edge of the jet exhaust deflector. Coherence between selected near-field and far-field sensors is found to be sufficiently high to warrant experimental investigation of the active reduction of noise from a full scale hush house, using feedforward control techniques, following on from the small scale hush house experiments described by J. P. Smith etal. [ 3332(A) (1994)]. [Work sponsored by the U.S. Air Force.]
Studies of source–load interaction in a duct system using acoustic noise synthesis
Authors: Bhate Anilkumar , Prasad M. G.
Published: October 2005

Acoustic noise synthesis (ANS) for duct systems was first developed by the authors (M. G. Prasad and T. V. Ananthapadmanabha, INTER-NOISE 89 Proceedings) based on the principles of speech synthesis using a source-filter model. The ANS allows the designer of a duct system to simulate the perceptual quality of the acoustical output of the system. The ANS is based on the transfer function which is evaluated using the known geometry of the duct system. The present work describes the application of ANS to study the interaction between source and load on the perceptual quality of a duct system acoustic output. An electroacoustic driver as a source and ducts of various lengths are used as loads. Good results are obtained which validate the application of acoustic noise synthesis in design of duct systems with emphasis on the perceptual quality and subjective evaluation.
USAF flight test investigation of focused sonic booms
Authors: Downing Micah , Zamot Noel
Published: October 2005

In April 1994, the USAF Armstrong Laboratory in cooperation with USAF Test Pilot School conducted a measurement study of controlled focus booms generated by supersonic maneuvers. The objective of this study was to collect focus and post-focus booms and to assess the ability of aircrews to control the placement of the focal region during basic maneuvers. Forty-nine flights were performed and included level linear acceleration, level turn, accelerating dives, and climbout/pushover maneuvers. Sonic booms were collected by up to 25 boom event analyzer recorders (BEARs) placed in a 13 000-ft linear array. The BEAR units were spaced 500 to 2000 ft apart with the denser spacing at the expected focal region. This spacing was chosen to evaluate the thickness of both the focal and post-focal regions. Of the 49 flights, a focus boom was successfully placed within the array 35 times which demonstrated the ability to place controlled focus booms. This ability of the aircrew can be employed to avoid collateral damage to noise sensitive receptors. Along with capturing focus U waves, complex N–U signatures were recorded at distances away from the foci. Turbulent conditions had a defocusing effect, resulting in smaller maximum overpressures.
Sonic-boom noise penetration into the ocean: 1995 update
Authors: Sparrow Victor W.
Published: October 2005

This talk is an update to a previously given presentation [ 1850–1851 (A) (1993)] in which a first-pass model for the penetration of sonic boom noise into the ocean was described. Since that presentation was given, a number of refinements to the theory have been identified, the primary of which is the effect of aircraft Mach number on the duration of the sonic boom. The general trend for aircraft flying at a fixed altitude is that the evanescent acoustic wave underwater due to a sonic boom penetrates more deeply for increased aircraft speeds [ 159–162 (1995)]. Other refinements currently under development include appropriate initial models for sound level decay as a function of depth and for focusing and defocusing of the penetrating sonic boom noise by ‘‘frozen’’ sinusoidal and trochoidal ocean surface waves. [Work supported by NASA Research Grant NAG-1-1638.]
Onset-sensitive time-frequency masking and its application to speech recognition
Authors: Aikawa Kiyoaki
Published: October 2005

This paper proposes an onset-sensitive time-frequency masking mechanism in order to improve dynamic feature extraction. Application of the proposed mechanism to Japanese 23-phoneme recognition using hidden Markov models demonstrated that onset-sensitive MASP outperforms time-invariant MASP. MaskedSpectrum (MASP) [Aikawa etal., Proc. ICASSP93 II, 668–671 (1993)] is a new spectral representation incorporating time-frequency forward masking and has been reported to provide excellent performance when used for speaker-dependent and speaker-independent speech recognition. The masking pattern production mechanism was previously modeled by a time-invariant time-frequency filter, but the masking level rises at the onsets and offsets in a speech sound [T. Hirahara, J. Acoust. Soc. Jpn. E12 (2), 57–68 (1991); E. Miyasaka, J. Acoust. Soc. Jpn. 39 (9), 614–623 (1983)]. This phenomenon suggests that an adaptive masking mechanism is effective for balancing instantaneous and transitional spectral features depending on vowels or consonants. The masking pattern is calculated as the weighted sum of the smoothed preceding spectra obtained by time–distance-dependent spectral smoothing lifters. The masking level is controlled by the slope of the temporal contour of the instantaneous sound energy. The masked spectrum is obtained by subtracting the masking pattern from the current spectrum. Onset–offset-sensitive masking models are also examined.
Difference limens for vowel–vowel formant transitions
Authors: Ainsworth William A.
Published: October 2005

Second-formant transitions in vowel–vowel utterances are not always of the same duration as those of the first formant and they often begin and end at different instants. In other cases the formant frequencies sometimes first move in a different direction from their final targets. In order to investigate whether these formant movements are perceptually significant, a number of difference limens for formant transitions have been measured for synthesized versions of the vowel pair /a/–/i/. It was found that differences in duration between the first- and second-formant transitions of up to 70 ms were not perceived. It was also found that delays between the starts and ends of the first and second transitions of up to 50 ms were not perceived. These results suggest that the differences in durations and delays between the first and second formants found in natural vowel–vowel utterances are unlikely to be of perceptual significance. [Work supported by EC Science Contract SC1-CT92-0786.]
Vocal tract simulations based on fluid dynamic analysis
Authors: Richard G. , Liu M. , Sinder D. , Duncan H. , Lin Q. , Flanagan J. , Levinson S. , Davis D. , Simon S.
Published: October 2005

An alternate approach to speech synthesis based on direct numerical solution of Navier–Stokes (NS) and Reynolds-averaged-Navier–Stokes (RANS) equations is described. Unlike the traditional methods based on linear acoustic theory, the NS and RANS formulations are not limited by the assumptions of linearity, negligible viscous effects, and plane-wave propagation. The expected results are high-quality synthesis and a new parametrization of speech for applications in automatic recognition and low bit-rate coding resulting from a parsimonious modeling of articulatory shapes and dynamics. In the present formulation, the Navier–Stokes equations are discretized and solved using a finite-difference method. An initial application involves a 2-D simulation of flow through ideal channels (straight, dilating, and constricting tubes) with rigid walls and constant boundary conditions (constant flow velocity at inlet, zero pressure at outlet). As expected for these simple geometries, the resonance frequencies correspond to those predicted by linear acoustics. In a second application, the formulation is applied to the geometry of the three cardinal vowels. For these cases, periodic inflow boundary conditions are also used (a train of short pulses to represent vocal cord excitation). Synthetic speech sounds of encouraging quality are obtained for the three vowels. [Research supported by NSF/ARPA IRI-9314946 and ARPA DAST 63-93-C-0064.]
A computationally efficient articulatory synthesizer
Authors: Rael Jacob , Chang James , Alwan Abeer
Published: October 2005

In this study, a computationally efficient articulatory synthesizer that utilizes the popular analog circuit simulator SPICE is developed. The synthesizer uses a transmission-line analog model of the vocal tract. An analog model has many advantages over digital representations: (1) Side branches (needed for modeling nasals and /l/) can be simulated easily by additional transmission lines in parallel; (2) drive-dependent sources, at any location, could be added; and (3) the number of sections can be varied without changing the sampling rate, as is the case with a digital synthesizer. A computer interface, using MATLAB, is developed such that the input to the synthesizer can be specified in terms of the area function of the vocal tract and the type and location of dependent or independent sources (voltage or current.) By simulating the transfer function of the vocal tract, transient and steady-state responses are generated. Using Fant’s vowel area functions (1960), vowels were synthesized with their first four formant frequencies almost identical to those given by Fant. The feasibility of implementing the analog synthesizer using modern ICs, such as the gyrator-based inductance simulator and switched capacitor filter circuits, is assessed.
Speech synthesis based on an overlap-add sinusoidal model
Authors: Macon Michael W. , Clements Mark A.
Published: October 2005

A general framework for waveform synthesis in a concatenation-based text-to-speech system is presented. Natural speech is segmented into subword units and analyzed using an iterative analysis-by-synthesis procedure originally presented in [E. B. George and M. J. T. Smith, J. Audio Eng. Soc. 40, 497–516 (1992)]. Synthesis is then performed by an efficient overlap-add resynthesis and modification method. This method eliminates the need for precise, hand-corrected pitch pulse marking in analysis (as required in some other popular concatenation methods), by incorporating a pitch pulse onset time estimation function based on [R. J. McAulay and T. F. Quatieri, Proc. ICASSP, 1713–1715 (1986)]. The sinusoidal model is capable of natural-sounding prosodic modification of both continuous speech and concatenated segments, making it an ideal candidate for application in a TTS system. Furthermore, the model provides for a computationally tractible and conceptually simple decoupling of various properties of the speech signal, making it an excellent platform for other transformations of the synthesized speech.
Toward automatic segmentation and syllabification of dysarthric speech
Authors: Zhang Guangsheng , Bunnell H. Timothy
Published: October 2005

A program has been developed which uses heuristic processing of the output of a neural network to locate syllable boundaries in connected speech. The neural network attempts a broad segmental (8 category) classification. This is followed by a program which locates syllable boundaries using phonological constraints and the pattern of network classifications over time. On test data consisting of 150 sentences (about 1800 syllables) from the TIMIT database, this program correctly identifies between 83% and 88% of the syllable boundaries (depending on the error criteria), including syllables bounded by sonorant consonants and syllables containing syllabic consonants. This program is now being applied to the identification of syllable boundaries in speech from talkers with cerebral palsy. In preliminary analyses of 20 sentences from ten dysarthric talkers (140 syllables) only 12 boundary omissions occurred. However, since dysarthric speech often contains hesitations, false starts, and extraneous noises, assessing boundary insertion errors is more difficult; about 50 boundary insertions occurred. Roughly 2/3 of these represented instances in which the talker inserted additional syllabic features into the speech. Further studies will examine methods for reducing the number of these ‘‘unintended’’ boundaries identified by the algorithm. [Work supported by the Nemours Foundation.]
What acoustic tomography can tell us about the state of the ocean: Theoretical considerations
Authors: Miller Robert N.
Published: October 2005

Acoustic tomographic data consist of line integrals of sound speed along ray paths. The community has little experience with line integral data, and most of the methods for mapping oceanic properties and for data assimilation (i.e., combining observed data with results from dynamical models) have been designed and tested with point measurements in mind. In this work, the tools of estimation theory were used to investigate the potential for reconstructing time series of oceanic pressure and density fields by assimilation of line integral data into a simple ocean model. Under ideal conditions, with noise-free data and ocean dynamics given exactly by a simple combination of linear waves, time series of integral quantities along the edges and diagonals of a square array are sufficient to determine time series of amplitudes of a large collection of waves, most of whose wavelengths are much shorter than the length of the edge of the array. Simulation experiments were performed to investigate the consequences of imperfect, possibly nonlinear dynamical ocean models and noisy data.
Evolution of the large-scale temperature field in the Greenland Sea during 1988–89 from tomographic measurements
Authors: Morawitz Werner M. L. , Worcester Peter F. , Cornuelle Bruce D. , Sutton Philip J. , Lynch James F. , Pawlowicz Richard
Published: October 2005

Moored thermistor, hydrographic, and tomographic measurements have been combined using least-squares inverse methods to study the evolution of the 40 km and larger three-dimensional temperature field in the Greenland Sea during winter 1988–89. In February, the sub-surface temperature maximum at around 200-m depth disappears over a large area. Upper waters warm around this time, while intermediate waters cool, consistent with vertical mixing. A chimney structure reaching depths in excess of 1000 m is observed to the southwest of the gyre center during March. The chimney has a spatial scale of about 50 km, a time scale of about 10 days, and breaks up in about 3–6 days. A one-dimensional vertical heat balance adequately describes changes in total heat content in the chimney region from autumn 1988 until the time of chimney break-up. A simple one-dimensional mixed layer model is successful in reproducing fall to winter bulk temperature and salinity changes, as well as the observed evolution of the mixed layer. The estimated annual deep water production rate in the Greenland Sea for 1988–89 is about 0.1 Sverdrups, comparable to the decreased production rates since 1983 derived from tracer measurements. [Work supported by NSF and ONR.]
Characteristics of signals recorded near shore in the Mediterranean during the THETIS 2 experiment
Authors: Desaubies Y. , Terre T. , Faure B. , Gaillard F.
Published: October 2005

Most of the western Mediterranean coast is characterized by a narrow shelf and steep rise. Thus it should be fairly easy to install sources and/or receivers cabled to shore for permanent acoustic monitoring of the basin. The study of the feasibility of this option was one of the goals of the THETIS 2 project. Results of a ship-borne acoustic survey over the continental rise will be presented. Signals were recorded at 26 stations, opening up in range, with an 8-channel vertical array. Several of the receptions implemented a synthetic aperture array concept. Where available, the results will be compared to those obtained at French Navy, fixed hydrophones locations, in particular, a long time series at a single shallow water hydrophone. [Work supported by EC/MAST and IFREMER.]
Ocean tomography via matched-field processing
Authors: Tolstoy A. , Sotirin B.
Published: October 2005

The tomographic determination of 4-D ocean sound speeds depends upon the nature of the system design, the analysis technique, and upon the resources used for generating and measuring the acoustic fields sampling the ocean region of interest. Matched-field tomography has recently been proposed as a high-resolution method potentially capable of synoptically imaging ocean regions as large as 1000 by 1000 km with rms accuracies less than 1 m/s [A. Tolstoy, J. Comput. Acoust. 2(1), 1–10 (1994)]. However, all results to date for the determination of deep ocean sound-speed profiles have been simulated. A major drawback to the actual implementation of the technique has been the requirement for several long (1000 m) vertical arrays, and these arrays have been neither affordable nor readily available. New technology is on the verge of producing relatively inexpensive, air-deployable, lightweight versions of these arrays suggesting that the MFP method may soon be testable. [Work supported by ONR.]
Investigation of the layered bottom structure influence on propagation of high-frequency acoustic signals
Authors: Fokina Margarita S. , Kerzhakov Boris V.
Published: October 2005

Peculiarities of frequency-angle dependences of Fresnel’s reflection coefficients V(fd,θ) (f=frequency, d=layer thickness, θ=grazing angle) from a system of elastic layers on the elastic substrate are investigated. The method of calculating the phase and the module of the reflection coefficient from the layered elastic absorbing substrate is numerically realized. The resonant behavior of the reflection coefficient at different grazing angles for a given bottom model is investigated. The influence of the set of parameters of bottom layers and the substrate on the frequency-angle dependence of the reflection coefficient and the behavior resonance of the reflection coefficient is considered. The effects of the layer structure of the elastic absorbing bottom on the space-frequency dependences of acoustic fields in oceanic waveguides in the frequency range 20–1000 Hz is studied. The relationship between the type of sediments that forms bottom layers and frequency ranges, where the least losses due to propagation are observed, i.e., optimum propagation frequencies, is ascertained. [Work supported by RFFI project N 94-05-1725-a.]
Numerical investigation of ray path stability in the Mediterranean under the condition of nonstationary signal receiving
Authors: Fokin Vladimir N. , Fokina Margarita S.
Published: October 2005

Using the concrete c(z) data (c=sound speed, z=depth) obtained in the Mediterranean during the THETIS-2 experiment on board the ACADEMICIANSERGEYVAVILOV research vessel, the possibility of ray path identification using nonstationary receiving are numerically investigated. Sufficient c(z) data were obtained for numerical simulation of the sound propagation along different acoustic routes. Sensitivity of different rays paths to errors caused by the erroneous determination of depth and distance is analyzed. The possibility of using the data obtained from the research vessel for ocean acoustic tomography is discussed.
Definition of geoacoustic characteristics of an elastic layered bottom using the spatial dependence of an acoustic field
Authors: Fokina Margarita S. , Fokin Vladimir N. , Kerzhakov Boris V.
Published: October 2005

The effect of the layer structure of the elastic absorbing bottom on the space-frequency dependences of acoustic fields in shallow and deep seas in the frequency range 20–500 Hz is studied. The comparison of the experimental losses due to sound propagation in the deep-water part of the Atlantic Ocean at three frequencies and the results of numerical calculations by ray programs has shown that the experimental and calculated losses may agree satisfactory only if the layer bottom structure is taken into account. The numerical study in the frames of the approximation of adiabatic modes has shown that an absorbing layer of the geological bottom structure of the Barents Sea influences significantly the characteristics of sound fields and enriches the field mode spectrum. It is shown that the degree of this effect depends on the seasonal fluctuations of the sound velocity profile structure. The relation between the type of sediments that form bottom layers and frequency ranges, where the least losses due to propagation are observed, i.e., optimum propagation frequencies, is ascertained. [Work supported by RFFI project N 94-05-1725-a.]
Nonauditory effects of waterborne sound: Subjective responses during under water hearing conservation studies
Authors: Steevens C. C. , Smith P. F.
Published: October 2005

Divers have reported that intense waterborne sound can be felt as well as heard. Subjective reports of nonauditory sensations were obtained from 19 divers during recent underwater hearing conservation studies. The divers were interviewed following each noise exposure trial. The waterborne exposure stimuli were warble tones (±5% center frequency) at center frequencies varying from 125 to 6000 Hz at sound-pressure levels as high as 145 dB re: 20 μPa. Exposure durations ranged from 43 s–15 min per exposure. Reports of direct effects on divers’ bodies, and effects on diver equipment were solicited. Nonauditory sensations were reported for frequencies of 1000 Hz and below, but none was reported above 1000 Hz. The lower the frequency, the lower the threshold for nonauditory sensations. At 125 Hz, sensations were first reported at a level of 104 dB re: 20 μPa. Nonauditory sensations reported were primarily vibrations felt at various locations on the divers’ bodies. Anatomical localization of sensations varied with frequency, but were not exclusively related to gas filled viscera. There were also several reports of minor effects on diving equipment. No evidence for nonauditory, physiological hazard was found during these limited exposures. [Work supported by U.S. Navy.]
Acoustic response of interacting gas bubbles
Authors: Goldman Daniel , Barbone Paul E. , Nadim Ali
Published: October 2005

Following a method similar to that used to obtain the standard Rayleigh–Plesset equation, equations are derived describing the volume oscillations of two interacting spherical bubbles, whose centers are assumed to be a large distance (relative to the equilibrium radii of the bubbles) apart in an infinite fluid. Although the bubbles are initially assumed to be fixed, these approximate equations also describe the movement of the bubble centers. First, the natural frequencies of the two-bubble system are obtained via linearization. Then, for bubbles with the same equilibrium radii, the nonlinear stability of the symmetric and antisymmetric modes is studied both analytically and numerically. The response of the two-bubble system to a time-periodic pressure field is also studied. Finally, the equations for the two-bubble system are generalized to those describing an n-bubble system, and an estimate is obtained for the lowest resonant frequency of the n-bubble cloud.
The effect of turbulence drift on acoustical scattering into a shadow region
Authors: Stinson Michael R.
Published: October 2005

The turbulence structures responsible for acoustical scattering in the atmosphere are not static and evolve in time. As a first approximation, a uniform drift of the turbulence with the prevailing winds, consistent with Taylor’s hypothesis, can be anticipated. This drift can be accommodated through an extension of the Green’s function parabolic equation (GF-PE) approach. Normal implementations of the GF-PE compute the sound field assuming a static realization of turbulence. Such calculations can be repeated, though, for successive realizations of a turbulent atmosphere, with each realization corresponding to an additional shift in space of the initial turbulent structure. The resulting series of sound field ‘‘snapshots’’ shows the evolution of the sound field as the turbulence drifts. Recent simulations using the GF-PE have examined scattering by drifting turbulence into an acoustical shadow, formed during upward refracting conditions, and the time evolution of the scattered field will be presented. The nature of these time variations is relevant to the use of beamforming arrays in an acoustic shadow. Some observations about predicted array responses will be discussed.
Resonant behavior of two eclipsed shells: Quasiharmonic study
Authors: Pareige P. , Kheddioui Elm. , Conoir J. M. , Izbicki J. L.
Published: October 2005

The purpose of this paper is the presentation of temporal, frequential, and angular experimental studies of the acoustic scattering by two identical air-filled elastic shells immersed in water. They are insonified by a quasiharmonic wave at normal incidence. The distance between the shells is about 1 to 3 diameters of one shell. The frequency range investigated corresponds to the whispering gallery wave resonances (l=2). The configuration considered is the eclipsed one: Emitter and shells are on the same axis. The complex structure of the backscattered temporal signal is analyzed and theoretically justified. According to the part of the temporal signal considered, different backscattering spectra are obtained and analyzed. The results are discussed in term of acoustic rays involving internal and external acoustic waves. In addition, a phenomenon of acoustic interference is clearly shown. At the resonance frequencies of one shell, the angular diagram shapes are justified and compared to those provided by an analytic model. In all cases, the results are compared to the well-known results provided by the study of a single shell.
An experimental investigation of the propagation of sound over a flat rough surface of finite impedance
Authors: Chambers James P. , Sabatier James M.
Published: October 2005

Recent work on rough surfaces by Medwin etal. [ 657–665 (1986)] has indicated the presence of acoustic boundary waves in the fluid directly above rough surfaces with high acoustic impedances. The effect of this boundary wave is that the sound field above the rough surface is louder than it would have been over a similarly smooth surface at close ranges due to multiple coherent scattering. The opposite is true at longer ranges since the scattering becomes incoherent. The ranges of coherent and incoherent scattering depend on the roughness scale and frequency of the sound source. The experimental work described above was performed over surfaces with high acoustic impedances such as wax and steel. The purpose of this present work is to find out if similar trends exist in the propagation of sound over rough surfaces that possess a lower acoustic impedance. Sound signals will be measured above a rough surface and compared to the signal above a smooth surface of the same material. Preliminary experiments using Styrofoam indicate that similar trends exist above the low impedance rough surface as did over the high impedance rough surfaces. [Work supported by USDA.]
The effects of talker-specific information on immediate memory span
Authors: Saldaña Helena M.
Published: October 2005

Recent evidence suggests that talker-specific information is retained along with codes for words and phonemes in long-term memory [Palmeri etal., JEP:LMC 19, 309–328 (1993); J. W. Mullenix and D. B. Pisoni, 365–378 (1990)]. If this indexical information is retained, one would expect talker-specific information to also effect a listener’s performance on short-term memory tasks. However, the predominant trace decay theories of short-term memory do not predict this. For example, in the articulatory loop model, items are stored as memory traces that fade after approximately 2 s unless revived by an articulatory control process [A. D. Baddeley and G. J. Hitch, Psych. Learn. Motiv. (1974)]. The number of items that can be reactivated within the decay time can be retained indefinitely. Therefore, immediate memory span is defined in terms of time or duration of items. In the present investigation, memory span experiments were conducted in which voice information was held constant or was changed for each item in the list. Initial results indicate that a change in voice information results in a decrease in memory span. Follow-up experiments manipulate speaking rate and presentation level. A reconceptualization of the articulatory loop model is discussed which takes into account the encoding of stimulus variability. [Work supported by NIH.]
The role of experience in the development of category structures
Authors: Acker Barbara E. , Pastore Richard E.
Published: October 2005

Recent speech research and music work have found empirical evidence for prototypes or exemplars. Whereas the function of a speech prototype is debatable, experiments using musically trained subjects found better discrimination around major chord prototypes than near less representative members of the category. While the specific function of prototypes is of interest, the development of prototypes is also important in elucidating perceptual processes. Developmental processes in speech and music have been studied by comparing infant and adult data. Utilizing musically naive adult subjects and musical stimuli, the current work investigates perceptual differences attributable to training and experience, but avoids problems associated with infant subjects. The ability of nonmusically trained subjects to discriminate in-tune (prototype [P] set) and mistuned (nonprototype [NP] set) major chords was evaluated in an AX task. Unlike musicians who showed significantly better performance with the P set, the naive subjects showed no differences between the P and NP sets of stimuli. These results demonstrate the need for experience for the development or activation of prototypes. Whereas the music results may not provide a direct parallel to speech, they do provide a basis for comparison. [Work supported by AFOSR.]
Age differences in the uptake of acoustic cues in spoken word recognition
Authors: Fox Robert Allen , McGory Julie
Published: October 2005

This study compares the uptake of acoustic-phonetic information during lexical access in younger (20–30 years) and older (55–70 years) listeners to determine if these groups are equally efficient in utilizing the coarticulatory information in VC transitions. Examined were place contrasts between stops (p/t and p/k) and fricatives (s/sh and z/yog), and manner contrasts between stops and nasals (b/m and d/n) using the gating paradigm [Warren and Marslen-Wilson, Percept. Psychophys. 41, 262–275 (1987)]. Stimuli consisted of 21 pairs of CVC words differing only in terms of the final consonant (e.g., pot versus pop). These words were heard in their natural or cross-spliced forms in which the initial CV of one word was appended to the consonant offset of its match. Gated versions of each word were constructed on the basis of this ‘‘splicing’’ point such that the first gate included the initial CV up to 100 msec before end of the vowel and then proceeded in 20-ms increments (gates that included portions of voiceless stop closure providing no additional acoustic information were eliminated). Results will be discussed in terms of possible age-related decrements in phonetic processing. [Work supported by NIA.]
On the ‘‘locus equation’’ and its relation with the consonant place of articulation
Authors: Chennoukh Samir , Carré René , Lindblom Björn
Published: October 2005

The specification of the consonant place of articulation from the locus equation coefficients [Sussman etal., 1256–1268 (1991)] is subject to discussion. For instance, Fowler [Percept. Psychophys. 55, 597–610 (1994)] claims that these coefficients can vary significantly within homorganic consonants. In order to study the relation between these coefficients and place of articulation, V1CV2 utterances were stimulated using the distinctive region model and three different instants of initiating the second vowel: before the acoustic segment of the consonant, at the beginning of that segment, and, in the third case, after it. The results show that linear locus equations can be acceptably fitted to those conditions, but that their coefficients vary as a function of the second vowel onset. It was noted that the locus equations are sensitive to degree of consonant–vowel coarticulation. Also, when y intercepts were plotted against the associated slopes for a given place and across the simulated timing conditions, highly linear relationships were observed. Thus the coefficients of such ‘‘second-order’’ locus equations stay constant for a given place despite variations in temporal and coarticulatory conditions. Hence, for natural speech, such ‘‘second-order’’ parameters might be a more stable place correlate than first-order coefficients.
Perceptual interaction of F1 and F0
Authors: Benki Jose R.
Published: October 2005

While F1 is a primary perceptual correlate of vowel height, other dimensions of vowels, such as F0, covary with F1 in natural speech. This study investigates the perceptual interaction of F0 and F1 in back vowels using the Garner paradigm. Two factors are hypothesized to determine the interaction: the magnitude of F1−F0 and the locations of the harmonics. F1 and F0 are predicted to integrate negatively for vowels in a threshold region (3.0 Bark <F1−F0<3.5 Bark), whereas subthreshold (F1−F0<3.0 Bark) and suprathreshold vowels (F1−F0≫3.0 Bark) are predicted to show no interaction. Previous work [Benki etal., 2977(A) (1994)] suggests that F1 and F0 do not interact in the threshold region but integrate positively in the suprathreshold region. The predicted dependence on the locations of the harmonics follows from work by Hughes and Diehl [ 2978(A) (1994)] showing that F1 discriminability is enhanced when a harmonic is near nominal F1. To assess these effects, CVC stimuli ranging in F1 from 300–600 Hz and F0 from 90–120 Hz were presented to listeners in the Garner baseline and correlated classification tasks. A perceptual F0/F1 space is inferred using a detection-theory analysis of the accuracy data. [Work supported by NSF and NIH.]
Word frequency effects on the acoustic duration of morphemes
Authors: Losiewicz Beth L.
Published: October 2005

The acoustic duration of the English past tense (ED) morpheme was measured for matched high- and low-frequency verbs (e.g., KNEADED/NEEDED). The ED on low-frequency verbs was of longer acoustic duration than the ED on matched high-frequency words. The rhyming portion of the matched verbs also showed a lengthening effect for the low-frequency words; in contrast to previous reports that word frequency does not affect word acoustic duration [Geffen and Luszcz, Mem. Cogn. 11, 13–15; Wright Mem. Cogn. 7, 411–419]. However, this effect was statistically independent of the ED length effect, and the final phonetic segment of a low-frequency monomorphemic verb stem was not longer in acoustic duration than a homophonous segment on a matched high-frequency verb (e.g., the /d/ in KNEAD/NEED). Further, the ED morpheme is of longer acoustic duration than a homophonous segment in a nonverb homophone (e.g., RAPPED/RAPT), as earlier reported for the morpheme /s/ [Walsh and Parker, J. Phon. 11, 201–206]. This set of evidence corroborates a frequency-dependent dual-access processing theory of linguistic morphology: that high-frequency complex words are processed holistically and low-frequency complex words are processed componentially [cf. Bybee, Morphology (1985)].
Adaptation in speech production to transformed auditory feedback
Authors: Houde John F. , Jordan Michael I.
Published: October 2005

This study investigated the ability of the speech production system to learn to compensate for changes in auditory feedback. The setup used for this was a DSP system that transformed the immediate feedback a subject received when speaking. This system can analyze a subject’s speech into a formantlike representation, possibly alter it, and then use it to resynthesize speech which is fed back to the subject with no noticeable delay (16 ms). The first of the experiments investigated whether subjects would learn to compensate for a change in vowel identity when producing CVC words. It was found that compensatory articulations were indeed learned, and that these persisted even when no auditory feedback was provided. The findings suggest similarities between speech and other sensorimotor tasks, such as reaching, which also show such adaptation. Other experiments characterizing the degree to which this effect generalizes across differing word and vowel environments will also be presented.
Contribution of different frequency regions to detection of additive noise in vowels
Authors: Kudumala Muralidhar R. , Espinoza-Varas Blas
Published: October 2005

An important issue in audio coding is the detection of quantization noise masked by speech. This investigation examined how different frequency regions contribute to the detection of broadband noise masked by a vowel. The vowel was synthesized by addition of the first 32 harmonics of a 200-Hz fundamental, with amplitudes appropriate to /i/. Broadband noise was synthesized by random-phase addition of the harmonics of a 10-Hz fundamental that fall within the vowel bandwidth. Noise detection thresholds were measured in a 2IFC task with an adaptive procedure (Levitt, 1971). Detection thresholds were obtained for noise with spectrum envelope parallel to that of the vowel, and for the same noise containing −6.0-dB spectral notches in either of the following bands: (a) 0.2–0.5 kHz; (b) 0.2–1.0 kHz; (c) 4.2–5.0 kHz; and (d) 4.7–5.0 kHz. Thresholds obtained in three highly trained listeners ranged from −27 to −32 dB (expressed in terms of noise-to-vowel power ratio). Threshold differences between spectral notch conditions were small. The results were compared to predictions of the excitation pattern model proposed by Moore and Glasberg [Hear. Res. 28, 209–225 (1987)]. [Work supported by OCAST-HR4-064.]
Toward understanding the role of formant transitions for distinctions of stops from glides
Authors: Revoile Sally G. , Nelson Peggy B. , Holden-Pitt Lisa
Published: October 2005

Our understanding is incomplete of the properties of vowel formant transitions that contribute to distinctions of voiced stop and glide consonants in speech. Research appears to have established some of the important transition cues for discernment of bilabial synthetic stops versus glides. However, the stop/glide transitions studied have typically been more stylized than those found in natural speech. This investigation examined the importance of transitions to listeners’ identification of initial stops and glides in spoken /CVk/ syllables. Performance was assessed for the stops and glides with progressive deletion of segments from the syllables’ onsets. Bilabial and velar stops and glides as well as alveolar stops were tested in /Ci/, /Copen ayek/, /Cash, ligaturek/ contexts to examine differences in transition use among phoneme environments. Twelve normal-hearing young adults participated as listeners. In general, when the initial stop bursts were deleted, the F2 transition frequency extent was significantly correlated with subjects’ consonant identification response patterns. That is, longer F2 frequency extents yielded a higher percentage of glide responses. In addition, shorter F2 frequency extents resulted in a higher proportion of ‘‘no initial consonant’’ responses. Neither F2 transition duration nor F1 transition duration/frequency extent significantly correlated with the subjects’ consonant identifications.
The role of formant synchrony in the coherence of vowels
Authors: Gordon Peter C. , Manning Erika
Published: October 2005

The coherence of vowels as auditory objects was studied by comparing identification thresholds in noise for synthetic vowel sounds (differing only in the center frequency of a single formant) to identification thresholds for the distinctive formant presented in isolation. The bandwidth of the noise masker was limited so that it only interfered with perception of the distinctive formant. Thresholds for accurately identifying the vowel sounds were lower than those for identifying the isolated formant. This demonstrates that vowel sounds cohere in the sense that unmasked formants reduce the masking of a formant embedded in noise. The advantage of a complete vowel over an isolated formant appears to depend on the temporal alignment of the formants. When the onset of the distinctive formant coincides with the offset of the other formants, then listeners can still identify the vowel sound in modest amounts of noise. However, in this case thresholds are not lower for vowel identifications than for identifications of isolated formants. This indicates that temporal synchrony plays a basic role in the psychoacoustic coherence of vowels.
Signal generation and processing for the R/P Flip ATOC transmissions
Authors: Metzger Kurt , Birdsall Ted
Published: October 2005

The waveforms transmitted during the November 1994 Acoustic Thermometry of Ocean Climate (ATOC) project R/P Flip measurement used an m-sequence phase modulated 75-Hz carrier, two carrier cycles per digit. Because the source had an inherent Q of 5, the drive waveform was modified in order to produce a two cycle per digit output. This paper describes the basic characteristics of the source, the procedure used to broaden the source bandwidth, the enforcement of a source peak power limit, and shows the resulting waveforms. The received waveforms possessed small amounts of Doppler ‘‘shift’’ caused by the motion of Flip on its mooring. Doppler limits the amount of time over which receptions can be integrated. Doppler normally must be treated as time compression/expansion. However, if the Doppler is sufficiently small it can be treated as simple time shift. This paper compares the results of using both approaches to account for the motion of Flip. [Work supported by the Strategic Environmental Research and Development Program through ARPA.]