bluuebunny/crossref_metadata_2025 huggingface dataset:
A statistical method to determine signal processing and stimulus parameter settings for optimal evoked potential detection
Authors: Morledge David E.
Published: October 2005

A frequency-domain detection method, based on statistical estimates derived from coherence measures, was used to determine optimal signal processing and stimulus parameter settings. The method was evaluated using simulations of electrophysiologic recording noise and signals in noise, as well as, real physiologic noise and click evoked auditory brain-stem responses. Parameters varied in the study were artifact rejection level, spectral filter shape, and stimulus presentation rate. Signal detection was improved following objective selection of parameter settings by this method. Some systematic estimate biases were found under the noise conditions evaluated. This method should be useful in selecting stimulus and recording parameter settings for improved signal detectability. [Work supported by U.C. San Francisco Graduate Division.]
Masking of tones and formant frequencies
Authors: Alwan Abeer
Published: October 2005

The purpose of this study was to find out whether results of auditory masking theory, established primarily for pure tones, can be used to predict masking of vowel formant frequencies by broadband noise maskers. For this purpose, three sets of 2AFC experiments were conducted. In each experiment, four frozen white-noise waveforms (with similar statistical properties) were used. The task in the first set of experiments, was tone-in-noise detection in which the tones were located at 1100 and 1800 Hz. In the second set, one-formant synthetic vowels were used in vowel-in-noise detection tasks. The formant was located at 1100 Hz in one experiment, and at 1800 Hz, in another. Finally, listeners were presented with pairs of four-formant synthetic vowels in noise: one pair had identical vowels and the second vowel in the second pair had a missing second formant frequency (F2). Subjects were instructed to choose the interval thought to contain the different pair of vowels. The four-formant synthetic vowel was either /ɑ /-like (formants at 625, 1100, 2700, and 3250 Hz) or /schwa/-like (formants at 500, 1800, 2700, and 3500 Hz). It was found that, on the average, the masked thresholds of tones and of formant frequencies were predicted well from the signal-to-noise ratio in a critical band centered around the tone or the formant frequency. [Work supported by NIH and NSF.]
Within-signal versus between-signal frequency discrimination for combinations of complex tones
Authors: Pollack Irwin
Published: October 2005

(1) Irrelevant between-signal variation in the fundamental frequency of complex tones was introduced into discrimination tests of relevant within-signal frequency differences. (2) Correspondingly, irrelevant within-signal variation in the fundamental frequency of complex tones was introduced into discrimination tests of relevant between-signal frequency differences. The effect of (2) was substantially larger than that of (1), suggesting the relative primary of within-signal processing over between-signal processing for the fundamental frequency of complex tones.
Acoustic test facilities at the Applied Research Lab/Penn State
Authors: Hughes W. Jack
Published: October 2005

The acoustic test facilities at ARL/PSU consists of an anechoic tank and a high-pressure tank. The anechoic tank is isolated from the building and is used for acoustic measurements such as calibrated frequency responses, directivity patterns, impedance parameters, target strength, echo reduction, etc., over a frequency range of 300 Hz to 500 kHz. The 5-ft-diam high-pressure tank is used to perform frequency response calibrations up to a pressure of 15 000 psi. A computer-controlled measurement system (based on an HP3570/HP3330 combination and an HP series 9000 model 320 computer) typically generates short-pulsed gated signals and a time-delayed gate for testing. Long pulse or continuous wave operation is also used to calibrate transducers at close distances and low frequencies. A description of the printed data output and thoughts on a replacement data collection system will be presented.
Proportional-bandwidth high-resolution time-frequency representations for musical signals
Authors: Pielemeier William J. , Wakefield Gregory H.
Published: October 2005

Time resolution, frequency resolution, and superposition requirements present a difficult trade-off in time-frequency representations for musical signals, both for individual instrument and ensemble analysis. Existing methods have significant limitations, particularly at low frequencies. A representation is developed in Cohen’s class [L. Cohen, Proc. IEEE 77(7), 941–981 (1989)], which improves on existing methods for a large class of musical signals. The pseudo-Wigner distribution is extended to a proportional-bandwidth form well suited to the frequency resolution required for musical signals and musical frequency analysis, paralleling a constant-Q transform [J. Brown, J. Acoust. Soc. Am 89, 425–434 (1991)]. This has excellent resolution, but fails to provide even limited superposition. A kernel is designed for Cohen’s class, based on a musical signal model, providing a limited superposition property while maintaining improved resolution when applied to this distribution. Favorable performance is demonstrated relative to constant-Q methods and spectrograms for both single instrument and ensemble cases, which are beyond the capabilities of phase vocoder and Fourier series-based methods.
Developments in phase-matching filter techniques
Authors: Orris G. , McDonald B. E. , Kuperman W.
Published: October 2005

For low signal-to-noise (S/N) data, an algorithm has been reported that is very effective when the shape of the noise spectrum is known and smooth as a function of frequency. The method uses the phases of the spectral components as free parameters in a variational problem. Improvements on the basic method to help relax the idealistic assumptions made have been investigated. Methods employing empirical orthogonal functions (EOFs) to exploit further degrees of freedom in the solution are presented. For cases of moderate S/N, an algorithm is presented that attempts to utilize signal-free intervals in improving S/N.
Acoustic classification using fuzzy sets
Authors: Leonetti Marc C. , Hand Edward A.
Published: October 2005

A prototype acoustic classification system is designed using fuzzy sets. Parameters relevant to classification are extracted from acoustic signals. Given n parameters, an n-dimensional hypercube is constructed. Each contact is represented by a vector within this hypercube. Known contact classifications map along the edges of the hypercube. Unknown contacts are represented by vectors whose elements are multivalent. The distances between the unknown contact and the known contact classifications are calculated. The contact classification is determined by a radius of similarity which is computed as one-half of the average distance between all known contacts.
Ontogeny and phylogeny of the cyclopean mantis ear
Authors: Yager David D.
Published: October 2005

Many praying mantises have a well-developed auditory sense mediated by a single, midline ear that hears only ultrasound. Hearing appears gradually in praying mantis nymphs. The neural elements of the peripheral auditory system are in place at hatching. The tympanum is represented only by a small slit and the ear components face posteriorly. The adult medial orientation is achieved by progressive rotation of the walls and broadening of the slit into a membrane. Increasing auditory sensitivity corresponds to the progressive development of impedance-matching tympanal air sacs. There are four major patterns of auditory structure and function among the 2000 mantis species. (1) Most mantises have a single ear and good ultrasonic hearing. (2) In a third of the species males have a functional ear, but females do not. (3) Mantises of one subfamily have not one, but two ears. The ears are serial homologs that hear in entirely different frequency ranges. (4) A few groups of mantises are deaf and have ‘‘ears’’ almost identical to those of newly hatched nymphs. Based on these results, it is possible to suggest an ‘‘auditory phylogeny’’ for the suborder Mantodea.
Cetacean perception of the marine environment−A theoretical approach
Authors: Raichel D. R.
Published: October 2005

The possibility exists that cetaceans can assess undersea environment through their sensing the modification of their probe signals in terms of harmonic content as well as the intensity of the reflected signals. The fluidic constitutivity or the elasticity (or a combination thereof) of the impinged surrounding objects affect the cetaceans’ probe signals so that additional harmonics appear in specific ways in the reflections. Absorptivity, spatial distribution, and the distance and acoustic impedance between the emitter and the reflector constitute the other factors that affect the probe signals. The harmonics aspect of this theory derives from previous work in the study of sound propagation through rheological fluids [D. R. Raichel and W. H. Kapfer, J. Appl. Mech. (Ser. E) 40, 1–6 (1973); M. Sokolov, ibid. 41, 823–825 (1974)], whereby perturbation techniques are applied to acoustic equations to yield the higher harmonics that result spontaneously from the inherent nonlinearity of sound wave travel through a constitutively real medium.
Active control of spinning mode noise from a turbofan engine
Authors: Burdisso R. A. , Thomas R. H. , Fuller C. R. , O’Brien W. F.
Published: October 2005

A JT15D turbofan engine is mounted in a test cell and operated to produce a fan blade passage frequency (BPF) of 2300 Hz. Twenty-seven flow disturbing rods are placed upstream of the 28 fan blades exciting to dominance the first circumferential spinning mode. A feedforward filtered-x adaptive multichannel LMS algorithm is implemented into a TMS320C30 DSP board. This three-channel controller is used to attenuate both the BPF tone and the first harmonic of the BPF (2BPF) tone components of the fan noise. The BPF and 2BPF frequency are sensed with a magnetic probe in the casing of the fan. This reference signal is an input to three adaptive finite-impulse response filters and whose outputs are fed forward to an array of 12 acoustic drivers mounted around the circumference of the engine inlet. The 12 drivers are divided into three channels of four drivers each. The far-field acoustic error information is obtained from three polyvinylidene fluoride piezoelectric polymer (PVDF) microphones placed in front of the engine. Experiments have demonstrated reductions of up to 10 dB in significant sectors. [Research supported by NASA Langley Research Center.]
A developmental study of native and non-native vowel perception
Authors: Flege J. E. , Walley A. C. , Randazza L.
Published: October 2005

Two vowel continua were created using the Klatt synthesizer. F1 frequency varied in the same eight equal mel steps in both. F2 frequency also varied in eight equal mel steps but, whereas it increased relative to /i/ in one continuum, it decreased relative to /i/ in the other. As a result, vowels in the ‘‘native–native’’ continuum ranged from /i/ to /i/, those in the native-foreign continuum from /i/ to a vowel symbolized as /Y/. Adults and children ranging in age from 4 to 12 years (ten per group) were trained to identify the endpoints of both continua as either /i/ or ‘‘the other vowel.’’ There was no effect of subjects’ age on the number of stimuli identified as /i/. This failed to support the hypothesis that vowels not bounded by another native language category−such as /i/ in the /i/–/Y/ but not the /i/–/i/ continuum−will expand outward with age. Slope analyses revealed that the 4- to 6-year-old children had shallower identification functions than the adults. Much the same effect of age was evident for the /i/–/i/ and /i/–/Y/ continua. This argues against attributing the observed age effect on slopes to a poorer definition of vowel categories by the 4- to 6-year-olds than by the adults. In metalinguistic tests, the 4-year-olds performed a rhyming task more poorly than 5- to 8-year-olds. The 4- and 5-year-olds were less able than 6- to 8-year-olds to segment initial and final consonants. The slope differences may therefore have arisen from age-related differences in ability to perform perceptual tasks involving localized sound segments. [Work supported by NIH.]
Detection of a submerged object insonified by surface noise in an ocean waveguide
Authors: Makris N. C. , Kuperman W. A. , Ingenito F.
Published: October 2005

Recent results indicate the possibility of imaging an object using ambient noise in the frequency region of 5 to 45 kHz [Buckingham etal., Nature 356, 327–329 (1992)]. For lower frequencies, numerical simulations have been performed to determine whether an object submerged in an ocean waveguide and insonified by surface noise can be detected. An object is placed in a surface-generated noise field of an ocean waveguide [Kuperman and Ingenito, J. Acoust. Soc. Am. 67, 1988–1996 (1980)]. The noise field is then scattered from the object using a modification of a previously developed theory for scattering from an object in a waveguide [Ingenito, J. Acoust. Soc. Am. 82, 2051–2059 (1987)]. An assortment of array processing algorithms are then used to detect the object.
Prefiltering for improved correlation detection of narrow- and medium-band energy signals
Authors: Ioup George E. , Pflug Lisa A. , Ioup Juliette W. , Field Robert L.
Published: October 2005

Narrow-band processing is commonly applied to achieve signal-to-noise ratio (SNR) advantage when the signal is concentrated in frequency and the noise bandwidth is broader. The matched filter (ordinary cross correlator) rejects noise outside the signal passband when the source is known. If only the passband of an energy transient is known, but other signal details are unknown, prefiltering of the data can sometimes improve ordinary correlation detectors and in some cases, produce even greater improvement in higher-order correlation detectors. For example, in the cross power spectrum, if the passband of the signal is half the passband of the noise and the noise is not preferentially concentrated in frequency, for a given probability of detection and fixed probability of false alarm, there will be a potential two-to-one maximum improvement in SNR with prefiltering. With one-dimensional prefiltering, for the cross bispectrum the potential maximum improvement rises to four-to-one and for the trispectrum, to eight-to-one. Larger maximum improvements are possible with higher-order prefiltering. In practice, one expects improvements less than the maximum. The prefiltering makes possible substantial improvements in detector performance which are shown using Monte Carlo simulations. Improvements of as much as a factor of 4 in SNR over the unfiltered case have been observed. [Work supported by ONR/NRL-SSC.]
Detection of cubic difference tones using the trispectrum
Authors: Rao Preeti , Bilger Robert
Published: October 2005

Auditory nonlinearities can give rise to an intermodulation distortion product at frequency=2f1−f2 from existing tones at f1 and f2. Such a phenomenon has been noted in ears with multiple spontaneous otoacoustic emissions (SOEs), where emissions are often observed at harmonically related frequencies [Burns etal., Hear. Res. 16, 271–278 (1984)]. SOEs, narrow-band signals with random amplitude, and phase fluctuations, are typically detected and estimated using the power spectrum. The power spectrum, however, suppresses all phase relations and cannot distinguish phase-coupled emissions from spontaneously excited ones. Cubically phase-coupled harmonics may be detected by the trispectrum [Swami and Mendel, IEEE Trans. ASSP 39, 1099–1109 (1991)]. Here, cubic differences tone SOEs are detected and the degree of phase coupling is quantified by a tricoherence index defined in terms of the power spectrum and an appropriately defined trispectrum of the input process. Data from the ears of subjects with multiple SOEs are analyzed for cubic coupling and the results are presented in the form of tricoherence spectra. [Work supported by DC00174.]
Active structural acoustic control of radiation due to broadband disturbances
Authors: Smith Jerome P. , Fuller Chris R. , Burdisso Ricardo
Published: October 2005

Active structural acoustic control (ASAC) using the adaptive feedforward LMS algorithm has been demonstrated for narrow-band (harmonic) disturbances. This paper presents an extension of this work to the attenuation of acoustic radiation from structures subject to broadband (random) disturbances. The control compensators are adaptive finite impulse response (FIR) filters. The LMS control algorithm is implemented on a floating point digital signal processor. The control inputs are implemented with piezoelectric ceramic actuators (PZT), and both polyvinylidene flouride (PVDF) structural sensors designed to observe the efficient acoustic radiating modes and far-field microphones are used as error sensors. In the control of harmonically excited systems, satisfactory attenuation is achieved with a single-input, single-output (SISO) controller. In contrast, for systems excited with broadband disturbances, a multi-input, multi-output (MIMO) controller is necessary for significant acoustic attenuation. A MIMO broadband acoustic control system is developed and results for the control of a simply supported plate is presented. [Work supported by Office of Naval Research.]
A hybrid formulation for blended adaptive feedforward and feedback closed-loop control
Authors: Saunders William R. , Robertshaw Harry H. , Burdisso Ricardo A.
Published: October 2005

Development and implementation of a novel control algorithm, suitable for active vibration control (AVC) or active structural acoustic control (ASAC), is introduced in this paper. The hybrid algorithm combines the closed-loop properties of the stochastically optimal linear quadratic Gaussian (LQG) design with the robust filtered-X LMS (FXLMS) adaptive algorithm. The resulting compensator provides active damping of plant poles in combination with optimal suppression of persistent disturbances. It is shown that the inclusion of a FXLMS compensator in the forward path leads to more robust harmonic disturbance compensation than is practical for completely feedback-based (e.g., LQG disturbance modeling) techniques. Additionally, the pole-placement properties retained in the hybrid controller provide simultaneous suppression of transient disturbances. The hybrid paradigm uses the Kalman filter estimate of the plant disturbance for the reference signal to the FXLMS block. This nice feature makes the hybrid controller practical for structures where an adequate reference signal is not readily available. Theoretical development of the hybrid controller equations are presented along with a demonstration of a hybrid AVC controller on a simply supported plate. [Work supported by ONR.]
Advanced time domain sensing techniques for active structural acoustic control
Authors: Maillard J. P. , Fuller C. R.
Published: October 2005

This paper discusses new work concerned with developing structural sensors and associated signal processing that provide time domain estimates of far-field pressure or structural wave-number information. The sensor arrangement consists of multiple accelerometers whose outputs are passed through an array of linear filters. The impulse response of the filters is constructed from the appropriate Green’s function for the elemental source area associated with each sensor. The outputs of the filter array are then summed in order to provide far-field pressure or wave-number information somewhat analogous to the well-known boundary element technique. A major significance of the approach is that it provides time domain information. The use of the sensing procedure is illustrated through simulations on radiation control from a baffled simply supported beam excited by both narrow-band and broadband disturbances. The control approach used is the time domain adaptive LMS algorithm. [Work supported by Office of Naval Research.]
An analog cochlear model for multiresolution speech analysis
Authors: Liu Weimin , Andreou Andreas G. , Goldstein Moise H.
Published: October 2005

Uniform bandwidth spectral analysis methods, such as spectrogram, are not suitable for rapidly changing broadband speech signals since the tradeoff between temporal and spectral resolutions is fixed for all frequencies. Described here is the signal representation by a model of the auditory periphery, based on neurophysiological data and realized in analog integrated circuitry. The main modules in the model are the basilar membrane filter bank and the reservoir model of hair cells/synapses [earlier version described in Liu etal., IEEE Trans. Neural Nets 3, 477–487 (1992)]. The instantaneous firing rates of the auditory-nerve fibers are modeled. Although nonconstant Q conditions in the cochlear filter are necessary in order to match the model output to the neural data, the model does exhibit properties of a wavelet analysis. Temporal analysis of the model output yields accurate determination of both time and frequency of each component in synthetic and natural speech signals. Finally, the low-power large-scale integrated circuits operate in real-continuous time (equivalent of 2.8×109 arithmetic operations/second), and all model parameters are electronically tunable.
Human cochlear traveling wave delays in normal and impaired ears
Authors: Kimberley Barry P. , Lee Anya , Brown David , Eggermont Jos
Published: October 2005

Previous work using electrocochleographic measures [J. J. Eggermont, J. Acoust. Soc. Am. 65, 463–470 (1979)] suggested that human cochlear traveling wave velocities were often increased in ears with cochlear loss. A new method to measure cochlear delays using DPE phase responses has been developed [Kimberley etal., J. Acoust. Soc. Am. (to be published)]. This method relies on more purely cochlear responses in comparison with previous techniques. The method is also rapid and noninvasive. Cochlear traveling wave delays were estimated using the phase responses of distortion product emissions associated with f2 frequencies ranging from 10 kHz to 700 Hz. Frequency-specific delay comparisons were made between two groups, namely 21 young adults (42 ears) with normal hearing and 20 adults (40 ears) with mild/moderate high-frequency hearing losses (30–50 dB HL) at two or more audiometric frequencies. The effect of frequency-specific cochlear hearing loss on traveling wave delays are analyzed and compared with the results previously obtained by Eggermont using electrocochleographic techniques. [Work supported by the Alberta Heritage Foundation for Medical Research.]
Conformal baffled line array for the AUTEC ocean haul down facility
Authors: Menoche Richard K. , Santos Gerald M. , Stein Peter J.
Published: October 2005

The radiated noise goals for various underwater vehicles are becoming more stringent. As these goals are approached, novel acoustic test and evaluation schemes will be required to overcome ambient noise limitations and confirm performance. Customized directive arrays will be required in order to make the measurements. Large multielement volumetric arrays will be cost prohibitive. Here, a unique baffled vertical line array deployed at the AUTEC ocean haul down facility is described. This device was designed to overcome shrimp noise from a nearby reef and make acoustic measurement of buoyant test vehicles and weapons under shallow-water conditions. This array meets the following specialized criteria: (1) very high backside rejection to shield the array from shrimp noise, (2) very wide (120°) frontside horizontal beam pattern to ensure capturing the vehicle as it passes the array, (3) beam patterns stable to within 1 dB over the beamwidth for accurate measurements, and (4) the ability to discriminate against surface noise. Diffraction problems from the edge of the baffle were solved by randomizing the edges thereby creating what is called ‘‘diffraction wings.’’ Plate vibration problems were solved by constructing a baffle with blocks welded to the back of a relatively thin plate (thereby having mass with no stiffness) and the addition of damping treatment. Receiver standoff problems were solved by the use of conformal PVDF sensors.
Pulse-tube measurements with conditioned signals
Authors: Moore R. E. , Dlubac J. J. , Schreppler S. C. , Deigan R. J.
Published: October 2005

Operation of pulse tubes typically involve a gated signal, either a tone or a bandlimited impulse. The incident and reflected signals are separated in time and Fourier transformed to determine the response of the test sample in the pulse tube. Both gated tones and impulses, however, suffer from the inevitable turn-on transients of the transducer and the receiver which corrupt the desired spectral output. Since the transfer function between the signal generator and the receiver can be determined using convenient digital-processing techniques it is possible to condition the output at the receiver. This method is applied to modify broadband impulses so as to maintain the desired spectral output at the receiver. The technique requires one iteration to converge to an optimal waveform input to the transducer from the signal generator. The technique is demonstrated using data collected in a 2-in. diam×72-in. pulse tube.
Design and construction of Cooper Union’s acoustics facility for education and research
Authors: Raichel D. R. , Winter P. A.
Published: October 2005

The newly constructed Acoustics Research Center at the Cooper Union was carefully planned and designed to provide a full range of laboratory experience, augmenting acoustics courses being taken by undergraduate and graduate students. The laboratory includes the usual array of signal generators, sound level meters, FFT spectrum analyzers, microphones, ultrasonic transducers, oscilloscopes, and white/pink noise generators. Additionally the laboratory incorporates a walk-in anechoic chamber, a reverberation chamber, and two open laboratory areas. Incandescent lighting is installed in the anechoic and reverberation chambers; both fluorescent (for economy) and incandescent (for minimal interference) lighting are provided elsewhere. A library that contains back and current issues of professional journals as well as a number of texts is situated on these premises for ready access; the materials are listed in the school’s main library catalog. Students are encouraged to conduct advanced projects reflecting their specific interests. [Work supported by the Kresge Foundation and the New York State Science and Technology Foundation.]
An introduction to chaos and its control
Authors: Schroeder Manfred R.
Published: October 2005

Nonlinear dynamic systems often show sensitive dependence on initial conditions leading to unpredictable behavior with exponentially growing uncertainty. Despite the perfect determinism of the underlying physical laws, such systems are characterized by chaotic behavior, called deterministicchaos. Chaos is encountered in many nonlinear systems from the double pendulum to the three-body problem. The malfunction of the vocal chords and turbulence in wind instruments are examples of acoustic chaos. This paper is intended as an introduction to chaos, its control, and the related concepts of fractals and self-similarity [M. Schroeder, Fractals, Chaos, PowerLaws: MinutesfromanInfiniteParadise (Freeman, New York, 1991)].
Connectionist models for musical control of nonlinear dynamical systems
Authors: Wessel David
Published: October 2005

Musical instruments, whether acoustic or electronic, are often nonlinear. So too are many compositional algorithms used to generate sequences of notes. Effective artistic use of such instruments and compositional procedures requires controllers that transform the musician’s intention into the parameters that operate on the nonlinear system. Although a number of thorny theoretical issues remain unresolved, neural networks show considerable promise for the identification and control of nonlinear dynamical systems. Neural network or connectionist controllers have the further advantage of adapting to the particular or even idiosyncratic features of a individual musician’s performance style. Connectionist controllers trained with back propagation learning that map performance gestures to the parameter space of sound synthesis and compositional algorithms are demonstrated. Many of the most interesting sounds and musical effects occur on the verge of chaos as opposed to the regions of full blown chaotic behavior. Operating on this border of chaos requires refined control and exposes a number of challenging research issues in nonlinear control theory.
Underwater acoustic test and calibration facilities in Canada
Authors: Hughes Steven J.
Published: October 2005

A survey of existing naval and commercial underwater acoustic test and calibration facilities in Canada is presented, with particular emphasis being placed on the calibration facilities operated by the Defence Research Establishment Atlantic. Each facility is described in terms of its geographical location and capabilities, its physical characteristics (including water depth, length, width), its weight handling capability, its data acquisition system(s), and the types of tests and calibrations conducted. Problems and deficiencies of each facility are discussed.
Verifying the accuracy of audible simulation (auralization) systems
Authors: Jacob Kenneth D. , Jo/rgensen Morten , Ickler Christopher B.
Published: October 2005

Auralization systems are beginning to appear in the professional sound industry. Each claims the ultimate benefit of ‘‘hearing the sound system before it is installed.’’ But how much work has been done to substantiate these claims? Can the customer always judge the sound system performance based on what he or she hears on the auralization system? Before any auralization system can be used, it must be ‘‘authenticated.’’ Authentication is the process of verifying that what people hear on an auralization system is the same−or almost the same−as what they would hear in the real environment. A systematic approach to authentication of auralization systems will be described. In this approach, the purpose of the auralization system, along with the perceptual parameters to be tested must first be defined. Then subject-based listening tests−typically one for each perceptual parameter−must be designed, conducted, and evaluated. It will be shown that this authentication process, when followed, can be used to characterize any auralization system.
A real-time multichannel room simulator
Authors: Gardner William G.
Published: October 2005

A room simulator has been developed as part of a project involving virtual acoustic environments. The system is similar to auditorium simulators for home use. The simulated reverberant field is rendered using four to six loudspeakers evenly spaced around the perimeter of a listening area. Listeners are not constrained to any particular orientation, although best results are obtained near the center of the space. The simulation is driven from a simple description of the desired room and the location of the sound source. The system accepts monophonic input sound and renders the simulated reverberant field in real time. Early echo generation is based on the source image model, which determines a finite impulse response filter per output channel. Diffuse reverberant field generation is accomplished using infinite impulse response reverberators based on nested and cascaded allpass filters. The system is implemented using Motorola 56001 digital signal processors, one per output channel. The talk will discuss implementation, design tradeoffs, and results.
The development of an audible simulator of sound fields and verification of simulation accuracy
Authors: Miyajima Tohru , Nakagawa Kiyoshi , Tahara Yasuhiko , Takane Shoichi , Ozawa Kenji , Suzuki Yoichi , Sone Toshio
Published: October 2005

A multichannel-type audible simulator with two special features, such as scattering sound and a way to determine production condition of reverberation, is developed. Using geometrical sound analysis with sound-scattering phenomenon, a sound field model of the simulator is composed of direct sound, initial reflection, reverberation, and scattered sound. Also, a new way to determine the production condition of reverberation is proposed. To examine the similarity between a simulated sound field and a real sound field, physical and psychological experiments are performed. It is found that acoustic indices, such as C80 (clarity), modulation transfer index (MTI), and time gravity (schwerpunktzeit), of simulated sound fields approximately agree qualitatively with those of the real sound field. The results of psychological experiments using dummy head recordings show that definition has good similarity, and it is necessary to improve reverberation.
Tongue surface deformation during obstruent stop consonants
Authors: Svirsky Mario A. , Stevens Kenneth N. , Matthies Melanie L. , Perkell Joseph S.
Published: October 2005

When an obstruent stop consonant is produced, there is an increase in intraoral pressure, and this increased pressure can result in outward movement of the vocal tract walls. Following release of the consonant, the intraoral pressure decreases and the walls are expected to move inward with a time constant that depends on the physical properties of the surfaces. An electromagnetic midsagittal articulometer (EMMA) was used to measure the vertical position of the tongue dorsum during a number of repetitions of five utterances: /ama/, /apa/, /aba/, /ampa/, and /amba/. Results seem to confirm the physical description given above: the average decrease in tongue dorsum position from implosion to release of the consonant was 0.4 mm for /m/, 2.7 mm for /b/, and 1.3 mm for /p/. The smaller movement for /p/ than for /b/ may be due to active stiffening of the tongue during /p/, partly counteracting the effect due to intraoral pressure. These displacements and their rates of movement are consistent with published data on average subglottal pressure, and compliance and resistance of vocal tract surfaces. These data have been used to refine current estimates of physical parameters of articulator surfaces during speech production. [Work supported by NIH Grant No. DC00075.]
Asymptotically correct shell theories with fluid loading for the thin spherical shell: Further results
Authors: Dean Cleon E. , Werby Michael F.
Published: October 2005

In a previous paper [C. E. Dean and M. F. Werby, J. Acoust. Soc. Am. 91, 2440 (1992)] initial results in the derivation of a so-called ‘‘shell theory’’ incorporating proper high ka asymptotic behavior for the Lamb flexural and extensional modes for a thin spherical shell were given. So-called shell theories give reasonably good results for the motion of a bounded elastic shell by positing that various parts of the shell move together in some reasonable manner. They also can give physical insight into the motions of the shell while using less computational time and resources than exact elastodynamic calculations. The use of various assumptions about the motions and fluid loading of the thin spherical shell gives rise to several shell theories. Further results from the derivation of an asymptotically correct shell theory with fluid loading for the thin spherical shell are compared with the exact results from a modal analysis with particular emphasis on the large size parameter (large ka) limit for the flexural and extensional Lamb modes. Limitations of each of the methods are then outlined as well as those of shell methods in general. [Work supported by ONR/NRL and by ONT Postdoctoral Fellowship Program.]
A theoretical study of waves on a conic shell
Authors: Guo Yue-Ping
Published: October 2005

This paper reports some theoretical study on waves on an elastic shell of conic shape. The method of multiple scales is used to derive analytical solutions for the shell response to external driving. This response is analyzed in terms of helical waves whose amplitude and phase both vary with traveling distance due to the conic geometry. Of particular interest are waves traveling toward the apex of the shell. For these waves, the wave amplitude increases because of the decreasing shell radius, but this increase in amplitude is limited by the effects of caustics; these waves do not reach the apex. They are turned back at caustics with some phase shift. The effects of the conic geometry on the dispersive properties of the helical waves are discussed. [Work supported by ONR.]
Modeling and experiments with low-frequency pressure wave propagation in liquid-filled, flexible tubes
Authors: Bjelland Cato , Bjo/rno/ Leif
Published: October 2005

A model for wave propagation in a liquid-filled viscoelastic tube with arrays of receivers inside, is being used to analyze the influence of noise generated by in-line vibrational noise sources. In this model, distensibility is of greater importance than compressibility of the liquid. The dispersion and attenuation is shown to be strongly dependent on the viscoelastic properties of the tube wall. The complex, frequency-dependent moduli of relevant tube materials have been measured in stress wave transfer function experiments. The moduli are used in the model to produce realistic dispersion relations and frequency-dependent attenuation. A 12-m-long, liquid-filled tube with interior stress members and connectors in each end is hanging vertically from an upper fixture. The lower end connector is excited by a power vibrator to generate the relevant wave modes. Measurements with reference accelerometers and arrays of hydrophones inside are compared to the theoretical model for wave propagation. A good agreement between experimental data and theoretical predictions is found.
Modal surface impedances for two spheres in a thermoviscous acoustic medium
Authors: Hasheminejad Mohamad , Geers Thomas L.
Published: October 2005

Two spherical bodies are submerged in an infinite thermoviscous fluid; one body is motionless and the other is vibrating at high frequency with a surface pattern that is axisymmetric with respect to the line joining the centers of the two spheres. Both bodies are sufficiently good conductors that their surface temperatures deviate insignificantly from the ambient temperature. The acoustic stress and velocity fields on the vibrating surface may be conveniently related by a modal surface impedance matrix based on field expansions in Legendre functions. This impedance matrix, which of course accounts for the presence of the motionless sphere, is here obtained from the field equations of Epstein and Carhart [J. Acoust. Soc. Am. 25, 553–565 (1953)] through the application of translational addition theorems for bispherical coordinates [Y. A. Ivanov, DiffractionofElectromagneticWavesonTwoBodies, NASA Tech. Trans. F-597 (1970)]. Impedance matrices for fluids that exhibit thermoviscous boundary layers of various thicknesses are compared with counterparts produced by the thin-boundary-layer model [A. D. Pierce, Acoustics (McGraw-Hill, New York, 1981)]. Special attention is devoted to the case when the motionless sphere is sufficiently large relative to the vibrating sphere that it approximates a rigid wall.
Timing of lip and jaw movements in the production of vowel types /i, a, u/ and voicing contrast /p, b/
Authors: Sock Rudolph , Löfqvist Anders
Published: October 2005

The temporal organization of lip and jaw movements was investigated for one subject, producing six pseudowords. The words, produced at two speech rates, normal and fast, were: /pipi/, /papa/, /pupu/, /bibi/, /baba/, /bubu/. Movement data from the upper and lower lips, and the jaw were obtained via an optoelectronic tracking device, together with an audio recording. Based on articulatory and acoustic events in the velocity and audio signals, respectively, four cycles were identified in the movement of each articulator, and two cycles in the acoustic waveform. The movement cycles are: (1) maximum vocalic velocity; (2) vocalic cycle onset; (3) maximum consonantal velocity; and (4) consonantal cycle onset. Velocity cycles were determined, as the interval between successive positive or negative peaks associated with the raising or lowering gesture in the production of either a vowel or a consonant. Onset cycles were defined as the recurrence of zero-crossings on velocity traces, reflecting the onset of a lowering or a raising gesture to produce a vowel or a consonant. The acoustic cycles are: (1) release cycle; and (2) closure cycle. Vocalic and consonantal phases were defined, within vocalic and consonantal movement cycles, respectively, as specific timing relations of articulator lowering or raising gestures, and within the appropriate acoustic cycle, as the interval that presents a stable formant structure for the vocalic phase, and as the obstruent portion, for the consonantal phase.

Gestural and acoustic phasing patterns obtained within the cycles suggest that: (1) the lower lip is an efficient articulator in distinguishing the front vowels /i, a/ from the back vowel /u/, in both consonantal contexts and in all four cycles; (2) the upper lip distinguishes all vowel types in both consonantal contexts in the vocalic cycle onset, and voicing contrast in the maximum consonantal velocity; (3) distinct phasing patterns in normal speech tend to merge with increased speech rate; (4) the jaw, although a carrier articulator, does not contribute actively to separating the phonetic tasks required in this study; (5) the closure cycle is the better of the two acoustic cycles in separating vocalic and consonantal classes. Results are analyzed and discussed in terms of cognitive tasks and emergent properties of the speech production system. [Work supported by the Fyssen Foundation and NIH.]
Effects of discard intervals and compression frequency on comprehension of connected discourse in time compressed speech
Authors: Poch Nancy E.
Published: October 2005

Time compressed speech studies have shown decreasing speech intelligibility for compression rates above 40%. The discard intervals and/or sampling periods, however, are rarely reported in these studies and have not been generally considered. In addition, most studies used primarily monosyllabic words or isolated sentences. This study examines the effects of discard intervals and sampling periods on comprehension of short paragraphs in normal hearing listeners at a constant 50% compression rate. Discard intervals from 15 to 135 ms were used. Subjects were given a set of 16 possible associations for each compressed paragraph and asked to select 3 that actually occurred in the paragraph. Comprehension scores were reduced for longer discard intervals. Detailed results and implications for audiological testing will be discussed.
Pure-tone thresholds in unmodulated and modulated noise for elderly listeners with normal or near-normal audiograms
Authors: Peters R. W. , Hall J. W.
Published: October 2005

Thresholds for the detection of a 1.0-kHz tone were measured in modulated and unmodulated noise for 11 elderly listeners, aged 62–83 with normal or near-normal audiograms, and for 5 younger normally hearing subjects. The noise was either narrow band, 1 ERB, or wide band, 1000 Hz, centered at 1000 Hz. The noise was either unmodulated or modulated at 10, 20, 30, 40, and 50 Hz. Auditory filter shapes were also estimated for the elderly subjects at 800, 1000, and 2000 Hz. Detection thresholds were significantly higher for the elderly as compared to the younger listeners for both unmodulated noise and narrow and wide-band-modulated noise. Improvement in detection in fluctuating as opposed to steady-state, unmodulated, noise was similar for the elderly listeners to that for younger listeners. Detection thresholds were not significantly correlated with auditory bandwidths or other auditory filters characteristics. There was an age effect beginning about age 75, in that pure-tone thresholds in both unmodulated and modulated noise, became progressively elevated. [Research supported NIDCD and the Andrus Foundation.]
Comparison and prediction of modal parameters of a strung and unstrung piano soundboard
Authors: Leggett Duane , Miller Michael W. , Celmer Robert D.
Published: October 2005

The first eight modes of an unstrung Steinway model-K upright piano soundboard were measured by testing both the front and back sides. Strong similarities in the first eight modal frequencies and mode shapes were found even though 40% fewer test points were used on the back side. Using the dynamic modification feature of the STARStruck modal analysis software system, a prediction of modal parameters was performed for the structural features of a strung piano. Comparison of these results to the actual modal frequencies and mode shapes found by testing the back side of a strung piano soundboard will be discussed. [Work supported by Steinway and Sons.]
Effects of air-jet angle and other jet characteristics on the spectral features of a jet-driven Helmholtz resonator
Authors: Cottingham James P. , Gresham Shenobie , Mackenzie Joshua W.
Published: October 2005

Previous studies of a Helmholtz resonator excited by an air jet explored the dependence of the frequency and amplitude of the Helmholtz mode on jet speed. It was observed that there are domains of jet speed, separated by narrow transition regions, for which a single frequency Helmholtz mode occurs [R. Khosropour and P. Millet, J. Acoust. Soc. Am. 88, 1211–1221 (1990)]. In the current investigation, a series of measurements were made exploring the dependence of amplitude, frequency, and domain width on the jet angle as well as jet speed and jet width. In addition, the amplitude variations of the harmonics of the Helmholtz mode and the appearances of the first standing wave mode were studied, especially near transition regions.
Ultrasonic measurements in superconducting UPt3
Authors: Adenwalla S. , Lin S.-W. , Zhao Z. , Ketterson J. B. , Levy M. , Sarma Bimal K.
Published: October 2005

Ultrasonic measurements on the heavy fermion superconductors have proved a useful tool in elucidating the nature of the superconducting state. Earlier measurements of sound attenuation in superconducting UPt3 showed power-law temperature dependences (rather than an exponential temperature dependence as expected for a BCS type superconductor), indicating the presence of nodes in the gap. Subsequent ultrasonic measurements have shown indications of phase transitions within the superconducting state. In particular, these ultrasonic velocity measurements show the presence of three superconducting phases in the mixed state. The size of the velocity jump can be related to the heat capacity jump. Further, using a thermodynamic analysis it is possible to deduce the order of the phase transition lines. [Work at NU supported by NSF, and the work at UWM supported by ONR.]