bluuebunny/crossref_metadata_2025 huggingface dataset:
A perturbation method applied in a study of Lamb wave sensors
Authors: Zhu Zhemin , Li Jian , Wu Junru
Published: October 2005

There are basically two approaches in studies of sensing using Lamb wave sensors [J. Acoust. Soc. Am. 91, 861–867 (1992)]: the dispersion equation method and the acoustic impedance method. The first approach is exact but complicated; the latter is simple but approximate. In many Lamb wave sensing applications, changes of the boundary conditions to be detected can be considered to be small perturbations to the propagation of free Lamb wave in a thin plate. General dispersion equations for Lamb waves in a plate with proper perturbed boundary conditions are derived. The equations are then applied to various sensing applications including viscous mass loading. [Work of JWU supported by NSF and Vermont EPSCoR.]
Design of an acoustical bench
Authors: Sinibaldi Jose O. , Raichel Daniel R. , Arveson Paul T.
Published: October 2005

An acoustical bench was designed to facilitate ultrasonic target strength experiments requiring monostatic geometry for frequencies up to 100 kHz. The design of an acoustical bench is analogous to that of an optical bench. Its purpose is to facilitate accurate positioning in setting-up and running acoustical experiments. As an example of its use, the acoustical bench will enable the researcher to move precisely the source and receiver and obtain backscattered target strength as a function of azimuthal angle. The acoustical bench is designed to accommodate a variety of target sizes and shapes. The distance of the source and receiver from the target can be easily adjusted to provide both geometric near and far field conditions for most targets. To reduce errors due to diffraction effects, the source and receiver are mounted flush on a planar baffle.
Neural network-based analysis of cues for vowel and consonant identification
Authors: Talley Jim
Published: October 2005

Many static and dynamic features of the acoustic speech signal have been proposed in the literature as cues for identification of phonetic categories. Ultimately, such features’ cue validity is most appropriately studied via well-designed perceptual experiments involving human subjects, but such studies are hindered by their expense (especially in terms of time) and inherent confounds (stimuli must be sufficiently speech-like to garner treatment as such). This paper examines ways in which neural networks (NNs) can be utilized as auxiliary tools in phonetics/speech perception research. Discussion includes the application of NNs to the task of learning relevant speech category discriminations from various restricted characterizations of a corpus of naturally spoken CVC syllables. That task examines learnability as a function of the available featural information. In addition, various ‘‘post-mortem’’ techniques (e.g., transfer function mapping, weight analysis,...) are discussed which, when applied to trained NNs, yield estimates of the cue validity of (ensembles of) features with respect to phonetic category discriminations. These methods cannot be blindly interpreted as producing valid characterizations of human speech perception, however, they represent useful tools that are inexpensive and highly targetable (confounds can be controlled) and can serve as guides to fruitful experiments with human subjects. [Work supported by NSF.]
The interaction of F0 and vowel quality in tonal and nontonal languages
Authors: Tsay Jane S. , Sawusch James R.
Published: October 2005

The issue investigated in the present study was the interaction between F0 and vowel formants in natural speech. F0 and vowel formants are normally assumed to be independent of each other. However, there are systematic correlations between tone (F0) and vowel height in some tonal languages, including Fuzhou and Mandarin. This study examined changes in the first three vowel formants that accompany tonal changes in the speech of native speakers of Fuzhou and Mandarin. Data from English speakers with manipulated pitch contours were used for comparison. Preliminary results indicate that in Fuzhou the tonal changes seem to be accompanied by changes in the formants, with one vowel phoneme being replaced by another. By contrast, in Mandarin the tonal changes seem to be accompanied only by allophonic changes in the vowels. In both tonal languages the change in vowel quality accompanying tonal changes is greater than that accompanying pitch changes in English. Implications of the complete results for theoretical phonology will also be discussed. [Work supported by NIDCD Grant Nos. R01 DC00219 and T32 DC00036.]
Inferring degree of coarticulation and relational invariance of stop place in CV’s using locus equations
Authors: Crowther Court S.
Published: October 2005

Locus equations relating target and onset frequency of F2 (F2t and F2o, respectively) have been successful in distinguishing stop consonant place of articulation independent of vowel context [e.g., Sussman etal., J. Acoust. Soc. Am. 90, 1309–1325 (1991)]. These linear equations are of the form F2o=a(F2t)+c. The slope parameter a is generally interpreted as indicating not only place of articulation, but extent of coarticulation [e.g., D. Krull, PERILUS V, 43–61 (1987)] as well. The intercept parameter c often is not interpreted explicitly, except in cases where the slope is found to be near zero. The locus equation technique is considered from several different viewpoints. Locus equations are analyzed empirically by examining changes that occur in estimated slope and intercept when a trained phonetician simulates different degrees of C-V coarticulation. Differences in extent of coarticulation are also measured using electropalatography, and these measurements are compared with slope and intercept parameters derived from the acoustic data. Finally, the Locus equation technique is compared with other methods for inferring coarticulation, including absolute comparisons of individual F2o and F2t frequency differences. [Work supported by NIH.]
Durational characteristics of proper names common words
Authors: Whalen D. H. , Wenk Heidi E.
Published: October 2005

Previous research has shown that infrequent words are longer in pure frequency lists than frequent words. Lists with proper/common homonyms (e.g., Grant/grant) were read as quickly as possible, with a surprising result: Although (printed) names are less frequent than words, name lists were faster than word lists. This could be due to the difficulty of focusing on one aspect of a homonym, commonness of name lists, greater frequency for names in speech, or special treatment for names. When only one word or name was read at a time, however, the names were longer in duration. This suggests that frequency works for names as it does for words, but that lists in which all items have similar or limited semantics are easier to produce. [Work supported by NIH Grant No. DC-00825.]
Syllable duration and word boundaries in French: A preliminary report
Authors: Bell-Berti Fredericka , Andreotta Karen , Gelfer Carole E.
Published: October 2005

Requisite to understanding a spoken message is the ability to recognize the grouping of segments into syllables and syllables into words. The durations of vowels and consonants are known to vary with their positions in words and phrases, with segments in final syllables being longer than those in nonfinal syllables. Furthermore, relative duration, or rhythm, has been shown to be important in identifying word-boundary location in English noun phrases [L. H. Nakatani and K. Shaffer, J. Acoust. Soc. Am. 63, 234–245 (1978)]. However, the situation in French, a ‘‘syllable-timed’’ language, is less clear. Durational data will be reported for sets of three-syllable utterances of one, two, and three words, produced in isolation and within sentences by three native speakers of French, as part of this study to identify the cues to word-boundary identification in a nonstress-timed language. [Work supported by St. John’s Univsersity and by NIDCD Grant No. DC-00121 to the Haskins Labs.]
Speaking rate, VOT, and quantity: The search for higher-order invariants for two Icelandic speech cues
Authors: Pind Jörgen
Published: October 2005

The temporal structure of speech is highly variable. Speaking rate, stress among other factors influence the duration of speech sounds. A number of speech cues are by their nature temporal. This holds, e.g., for voice onset time as well as for duration as a cue to phonemic quantity. The highly ‘‘elastic’’ nature of speech would seem to pose a problem for the listener: How does he or she achieve an invariant percept in the face of this great temporal variability? Data for two Icelandic speech cues under different speaking rates are reported. Results show that manipulations of rate have the effect of slightly blurring the distinction between unaspirated and aspirated stops. In spite of great changes in the absolute durations of long and short vowels and consonants the two categories of syllables, V:C and VC:, are in fact kept totally distinct. In perceptual experiments it is shown that the ratio of vowel to consonant duration is the primary cue to quantity and remains invariant at different rates. Perceptual experiments show a small effect of rate on the boundaries for VOT. The two temporal speech cues of VOT and quantity are thus shown to behave quite differently under transformations of rate.
Skill acquisition, coarticulation, and rate effects in a neural network model of speech production
Authors: Guenther Frank H.
Published: October 2005

This work describes a neural network model of speech motor skill acquisition and speech production that explains a wide range of data on contextual variability, motor equivalence, coarticulation, and speaking rate effects. Model parameters are learned during a babbling phase. To explain how infants learn phoneme-specific and language-specific limits on acceptable articulatory variability, the learned speech sound targets take the form of regions, or convexhulls, in orosensory coordinates. This leads to an explanation of coarticulation wherein the target for a speech sound is reduced in size based on context to provide a more efficient sequence of articulator movements. Furthermore, reduction of target size for better accuracy during slower speech (in accordance with Fitt’s law) leads to differential effects for vowels and consonants, as seen in speaking rate experiments that were previously explained by positing separate control processes for the two sound classes. The babbling process also naturally accounts for the formation of coordinativestructures, or groups of articulator movements marshalled together to perform orosensory tasks. Coordinative structures provide motor equivalence, including automatic compensation to perturbations or constraints on the articulators. Computer simulations verify the model’s motor equivalence, coarticulation, and speaking rate properties. [Work partially supported by AFOSR F49620-92-J-0499.]
Cross-language analysis of VCV coarticulation
Authors: Molis Michelle R. , Lindblom Björn , Castleman Wendy , Carré René
Published: October 2005

Öhman [J. Acoust. Soc. Am. 39, 151–168 (1966)] reported superposition of a consonant closure gesture on a vowel-to-vowel transition was sufficient to describe VCV coarticulation; however, other researchers [R. McAllister and O. Engstrand, Fonetik, 115–119 (1992)] have found a language-dependent articulatory trough in the movement of the tongue during some VCV sequences. Such a trough would limit the possible extent of coarticulation. In this study, one male speaker each of American English, French, and Swedish produced VCV sequences. Vowels included /i/, /a/, and /u/. For each of three stop consonants (/b/, /p/, and /d/), an index of coarticulation was obtained through calculation of a locus equation. In addition, coarticulation indices were obtained from the output of an acoustic tube model that uses superposition to generate VCV sequences. Preliminary results indicated that superposition alone predicted coarticulation of unaspirated stops in all languages, but was not sufficient to explain the reduced coarticulation observed for aspirated stops. Reduction in degree of coarticulation corresponded qualitatively with the language dependent extent of the articulatory trough. With modifications intended to simulate an articulatory trough, the model successfully simulated the qualitative change in cross-linguistic coarticulation.
Evaluation of individual differences in vowel perception through mimicry
Authors: Hawks John W. , Quinn Bonnie E. , Anthony Kimberly A.
Published: October 2005

In a mapping study using synthetic versions of isolated vowels, certain stimuli seemingly evoked very consistent perceptions of a given vowel quality to individuals that was different from the plurality of listeners. An imitation paradigm was employed to further investigate these differences. Subjects mimicked exemplar, ambiguous, and their individually ‘‘different’’ tokens in both speeded and unspeeded conditions. Subsequent identifications of the imitated vowels by phonetically trained judges suggested that subjects’ productions in both conditions generally agree with their perceptions. Disagreements were more frequent in the unspeeded task or related to neutralization in the speeded task. Relationships between formant patterns of the stimuli, imitations, and individual vowel spaces will be presented. Implications of the results related to the link between perception and production will be discussed. [Work supported by NIH.]
High-resolution geoacoustic modeling for SWellEX-1 using a geographic information system (GIS)
Authors: Bachman Richard T.
Published: October 2005

The SWellEX-1 (shallow-water evaluation cell) area is a complex shallow water environment of approximately 400 km2 immediately offshore of San Diego, California. In it, thinly sedimented terraces separate steep scarps. The terraces and scarps are old erosional surfaces with moderate relief. Sediment thickness ranges from zero to tens of meters over a basement of sandstone and mudstone. The IDRISI geographic information system was used to assemble and analyze geologic and geoacoustic data, and to provide that data to a database server that will support high-fidelity acoustic simulation codes. Data for the initial effort consisted of SeaBeam and National Ocean Service depth soundings, published geological maps, published sediment thickness charts combined with the results of a shallow seismic reflection survey, and published sediment charts. Data provided to the acoustic codes at a grid spacing of 2 arcseconds are depth; sediment thickness; sediment compressional and shear properties and density (varying with depth in the sediment); and basement properties.
Experimental spatiotemporal signal correlation functions in a shallow water environment (SWellEX-1)
Authors: Shook L. A. , Ryan F. J.
Published: October 2005

During 2 weeks in August 1993, a high-fidelity acoustic experiment, SWellEX-1, was conducted in shallow waters off San Diego, CA. The SWellEX-1 site is a 50-to 200-m bottom-limited shallow water environment characterized by thin sediments and strong range dependence. During the course of the experiment extensive synoptic oceanographic measurements were collected. The site’s bathymetry and geoacoustic bottom properties are characterized on a 2-arcsecond lat/lon grid. Broadband and tonal signals (50–750 Hz) from calibrated projectors were received on a 48-channel, 90-m aperture vertical line array (VLA) spanning the lower half of the water column. Source tows were conducted along the isobath and up-slope relative to the VLA. Experimental results of signal propagation and spatiotemporal vertical correlation functions will be presented along with high-fidelity full-wave simulations. The impact of a strongly bottom-limited shallow water environment on signal correlation and the resulting effect on plane-wave signal processing schemes will be investigated. Relative signal phase stability results will be presented and its impact on inverse synthetic aperture processing discussed. [Work supported by ONR under NRaD block CS3A.]
Structures and intensities of interference zones in a ducting shallow water environment
Authors: Del Balzo Donald R. , Ferla Carlo
Published: October 2005

Observations of regularly spaced zones with high propagation loss in a shallow water ducting environment were reported at the Denver ASA meeting [D. Del Balzo, J. Acoust. Soc. Am. 93, 1886(A) (1993)]. These zones resulted from interference between a small number of acoustic modes from a low-bandwidth (5-Hz) signal centered at 342.5 Hz. The current work is a modeling effort to describe the zonal structures and to determine their sensitivity to various factors; e.g., duct size and strength, source depth, center frequency, and bandwidth. Narrow-band and broadband normal mode calculations are employed to ascertain the physical size, location, and intensity of the zonal features.
Binaural advantage for sound pattern identification
Authors: Kidd Gerald , Mason Christine R. , Rohtla Tanya L.
Published: October 2005

Listeners learned to identify six tone patterns in continuous broadband noise. The patterns consisted of eight sequential 48-ms tones having frequencies that were: (1) constant, (2) rising, (3) falling, (4) alternating, (5) one-step up, or (6) one-step down. The frequency extent was 14% of the nominal center frequency. Masking level differences (MLDs) were measured in a 2I2AFC adaptive detection task by comparing thresholds for NoSo and NoSπ conditions. Identification level differences (ILDs) were measured in a 1I6AFC fixed-level identification task by comparing the psychometric functions for the two interaural conditions. Two center frequencies, 500 and 3000 Hz, were used. In addition, for ILDs, the center frequency was randomly roved over the 2500-Hz range. Average MLDs and ILDs were 12 dB at 500 Hz and 3 dB at 3000 Hz. However, in the roved condition, the psychometric function for the NoSπ condition was much shallower than for the NoSo condition, so ILDs decreased with increasing S/N level. This result is in good agreement with similar findings for speech [H. Levitt and L. R. Rabiner, J. Acoust. Soc. Am. 42, 601–608 (1967)]. The orderly relationship between detection and identification breaks down in certain conditions when an ‘‘informational masker’’ is used. [Work supported by NIH/NINDCD.]
Effects of interaural configuration of wideband forward and backward fringes on detection of an interaural phase shift of a narrow portion of wideband noise
Authors: Grange A. N. , Dye R. H.
Published: October 2005

Threshold interaural phase shifts were measured for a 20-Hz-wide section (CF=500 Hz) of wideband noise. Each interval of a 2-afc task consisted of an observation interval (either 100 or 250 ms), and forward and backward temporal fringes, all having identical amplitude spectra. A spectral band (250, 500, or 750 Hz wide) surrounding the target band was diotic throughout the entire stimulus presentation. Interaural phases of frequencies above and below the diotic band (flanking bands) differed for the temporal fringes and the observation interval. For one set of conditions, the flanking bands were diotic during the temporal fringes and interaurally phase-shifted ±90° during the observation interval. In other conditions, the flanking bands contained the 90° interaural phase shift in the temporal fringes, and were diotic in the observation interval. In general, thresholds were higher for conditions in which the temporal fringes were completely diotic, compared to conditions in which the interaural parameters of the flanking bands initially differed from the diotic band surrounding the target. This difference in thresholds decreased as the width of the diotic band and the duration of the observation interval increased. [Work supported by NIH.]
Effects of center frequency and bandwidth on echo threshold and buildup of echo suppression
Authors: Yang Xuefeng , Grantham D. Wesley
Published: October 2005

Echo threshold is the delay of a reflected sound relative to the direct sound at which a subject can no longer hear the reflection as a separate event. Echo threshold was measured for a single 4-ms noise burst as a joint function of center frequency (500–4000 Hz) and bandwidth (0.08–1.00 octave) of the burst. The direct burst came from a loudspeaker at −45° azimuth, and the simulated echo from a loudspeaker at 0° azimuth. A single-interval, forced-choice procedure was employed, in which the subject had to say on each trial whether one (direct sound alone) or two (direct plus reflected sound) events occurred. Thresholds ranged from 3–61 ms, with marked differences among individual subject’s functions; however, there was some tendency for threshold to increase with decreasing center frequency and decreasing bandwidth. Buildupofechosuppression [R. L. Freyman etal., J. Acoust. Soc. Am. 90, 874–884 (1991)] was measured in the same conditions by determining echo threshold for the noise burst when it followed a train of 9 like-filtered bursts. Elevation in threshold ranged from 0–49 ms, with staggering inter-subject differences across conditions. These results support the conclusions from earlier studies that the suppression of echos and the buildup of this suppression is mediated at a high level in the auditory system. [Work supported by NIDCD.]
Higher-order mode acoustic radiation from fluid-loaded, multi-layer composite plate
Authors: Kim M. , Hwang Y. F.
Published: October 2005

One of the outstanding features of the composite plate is that one can reduce the longitudinal wave (the first symmetrical mode) speed significantly at higher frequencies by intelligently arranging the various layers of elastic and viscoelastic materials. In such cases, however, the higher-order modes shift to a lower frequency region as compared to those of a single-layer elastic plate. Consequently, those higher-order modes may now become susceptible to an excitation band. An exact mathematical formalism was developed to calculate the acoustic radiation of a fluid-loaded, multi-layer composite plate subjected to a harmonic excitation. The formal solution is developed using the classical integral transform technique incorporated with the multilayer elasticity theory. The formalism was validated with the solution of a fluid-loaded, infinite, single-layer elastic plate. Numerical results are then presented for the far-field directivity pressure pattern of various plate configurations subjected to normal- and tangential-line forces. The pressure pattern and in-vacuo dispersion curve of the plate are used to explain radiating mechanisms and evaluate the acoustic significance of the higher-order modes. In addition, the effects of fluid-loading and damping on higher-order supersonic waves are also discussed. [Work supported by ONR.]
Vibration and sound radiation analysis of laminated plates in a fluid
Authors: Sorokin Sergey V.
Published: October 2005

Vibrations and sound radiation from laminated plates of finite length submerged in a fluid are formulated as a problem of optimal design width and of advanced materials. Design parameters are the coefficients of the constitutive matrix for the plate’s composite material. The sequence of constitutive laws is examined for laminated plates with sharply different plies. Interactions between an acoustic medium and a plate and between a rigid (soft) acoustic baffle and a plate are described by boundary integral equations assembled in a two-level system. The first level boundary equations govern the dynamics of a plate and the second level boundary equation describes the structural–acoustic interaction. The classical boundary equation related to contact acoustic pressure is modified by a substitution of a Somigliana’s type formula for normal displacements of points of the plate’s surface in contact with a fluid. Special attention is paid to the construction of Green’s functions for vibrations of unbounded plates in the framework of various plate’s theories. The asymptotic analysis of Green’s functions for laminated plates is performed along with the asymptotic analysis of boundary equations. Based on these analyses some simplified models of fluid–laminated plate interaction are developed. A semi-analytical sensitivity analysis of an energy flux and a directivity of radiation to the parameters of design is performed. Active control by the regulation of material parameters of plies is discussed.
A new algorithm for determining the elastic constants of transversely isotropic solids
Authors: Šribar Rok , Sachse Wolfgang
Published: October 2005

This paper reports a new algorithm for determining the elastic constants of transversely isotropic solids. The elastic constants are calculated from experimental group velocity data measured in general directions belonging to the zonal plane. The algorithm incorporates a theoretical calculation of the group velocities, and a feed-forward neural network (FFNN). In the procedure, the experimental, and the theoretical group velocity data are fitted to polynomials in a linear least-square sense to obtain appropriate training sets for the FFNN. When convergence of the network is achieved, it is used to estimate the elastic constants. With this procedure, we can determine C11, C12, and C13 elastic constants of zinc with precision better than 5%, even when the initial estimates of this Cij’s are off by as much as their value. [Work partially supported by ONR.]
Acoustical holography in spherical coordinates for noise source identification
Authors: Devries L. A. , Bolton J. S.
Published: October 2005

Acoustical holography has been applied most often to visualize the sound field of planar sources driven by known forces. In this paper the application of acoustical holography to machinery noise source identification, when it is possible to enclose the source entirely by a spherical measurement surface, is considered. That procedure amounts to representing the sound field on the measurement surface as a summation of spherical harmonics, each of which is associated with a particular spherical Hankel function. Once the spherical harmonics have been identified, each may be propagated inward towards the source for the purpose of source identification, or outward to visualize the sound field. The combination of those procedures with the use of multiple reference microphones to visualize the sound field radiated by multiple incoherent sub-sources will also be described. The procedures will first be illustrated by visualizing the sound field radiated by several simple sources (i.e., monopoles and dipoles) when operated either individually or in combination. Next, the vibration of a refrigeration compressor shell is visualized, both when mechanically driven at a single interior point and by internal pressure pulsations. Finally, results for a compressor in normal operation will be shown.
The state of the art of holographic vibration analysis
Authors: Stetson Karl A.
Published: October 2005

Holographic vibration analysis has changed greatly since its discovery in 1964. Thirty years ago, holograms were recorded exclusively with photographic plates using lasers that emitted barely 1 mW of power. Many years ago, high-power lasers and high-speed photographic plates freed holography from the need to hold objects immobile for long exposure times. Now the development of electro-optic holography is freeing it from the use of photographic film. The use of electronic phase stepping, modern pipeline image processors, and solid-state TV cameras have resulted in compact holographic systems that not only display vibratory fringe patterns in real time, but also allow direct conversion of fringes into numerical data. This paper will outline the current state of this technology and how it can impact on the study of musical instruments.
Analysis of contrast in marimba stroke articulation
Authors: Roberts Ronald A. , Larkin Barry
Published: October 2005

A current issue in marimba performance is the role of legato and staccato stroke techniques in controlling note articulation characteristics. Pedagogical tradition teaches that an arrested stroke produces a staccato (choppy) sound as compared to a nonarrested stroke that is used to produce a legato (smoother) sound. It has recently been suggested, however, that perceived differences in the mallet stroke attacks are purely contextual, that is, a note is perceived as staccato if it is significantly louder than its neighbors in time. It has also been suggested that if pedagogically correct legato and staccato strokes are played with equal loudness (equal mallet velocity) outside of a musical context, no differences are perceived. This paper presents a time frequency analysis of equal loudness legato and staccato strokes that indicates that there is indeed a physical difference in the sound pressure fields generated by the two strokes. In the staccato stroke, the broadband third overtone generated by the marimba bar is noticeably greater in amplitude relative to other spectral components, particularly when played with a softer mallet. However, it has yet to be determined if these differences are perceived. Procedures and results of ongoing experiments will be presented and discussed.
Holographic studies of normal modes in bass handbells
Authors: Gangadharan Deepak , Rossing Thomas D.
Published: October 2005

Holographic interferometry has been used to compare the modes of vibration in two G1 handbells, one of bronze and one of aluminum [Rossing and Mansell, J. Acoust. Soc. Am. 93, 2382 (A) (1993)]. Except for the (2,0) and the (3,0) modes, tuned to have a 3:1 frequency ratio, all the other modes in the bronze and aluminum bells occur at different frequencies. In the bronze bell, we observed an unusual family of (m,3*) modes, not found in other bells. a)Present address: Phys. Dept., South Florida Univ., Tampa, FL 33620.
Acoustical characteristics of a conch shell trumpet
Authors: Taylor Lisa R. , Prasad M. G. , Bhat R. B.
Published: October 2005

The sound spectrum and duct geometry of a conch shell trumpet is examined in this study. The fundamental frequency of a TurbinellaPyrum was sounded and its spectrum recorded in an anechoic chamber. Spectral analysis shows a strong fundamental frequency and five overtones at integer multiples that form a smoothly decreasing envelope. The placement of the harmonics and their relative amplitudes suggest the conch shell duct has an exponential flare similar to a french horn. It was also found that the fundamental frequency of the conch shell could be lowered by insertion of fingers into the shell opening. X-ray tomography pictures reveal the shape and size of the duct cross section at every half-turn, as well as its corresponding location with respect to the longitudinal axis. This information was used to approximate the duct profile as it would appear if the conch were unwrapped. As expected, the duct profile has an exponential flare that gives the conch a characteristic timbre and allows its sound to project over long distances.
The modern baroque trumpet
Authors: Pyle Robert W. , Pyle Robinson
Published: October 2005

Recent years have seen a great increase in the frequency of historically informed performances of early music. While the phrase ‘‘modern Baroque trumpet’’ appears to be an oxymoron, most of the trumpets used professionally in such performances have not been copied from historic specimens. To aid the player in achieving the accuracy and intonation expected by the audience (and conductor!), the design of the modern Baroque trumpet departs from that of its forebears, typically by adding from one to four side holes to be opened and closed by the player’s fingers. Another nontraditional feature present on some modern instruments is a tapered mouthpipe just downstream of the mouthpiece. The changes in the input impedance of the instrument produced by the sideholes and tapered mouthpipe will be shown and the relationship between these changes and the playing qualities of the instrument will be discussed. The talk will include demonstrations on at least one instrument.
Synthesis of the sound pressure radiated by a guitar using a mixed time- and frequency-domain modeling technique
Authors: Carter Thomas , Guyader Jean-Louis , Chaigne Antoine
Published: October 2005

A general method has been previously developed by O. Beslin etal. [J. Acoust. Soc. Am. 89, 1868 (A) (1991)], using a variational method to describe the motion of a perforated plate, and a integral formulation for the modeling of the consecutive radiated sound pressure. This model is now adapted to the particular case of a guitar, whose top plate is represented by a rectangular orthotropic plate with hole and back cavity. The sound pressure computation is carried out in the frequency domain, for a point force acting at the bridge. This yields the pressure versus force transfer function H(ω), from which the corresponding impulse response h(t) can be derived by means of inverse Fourier transforms. Finally, the guitar tones are obtained by convolving h(t) with a time-domain formulation of the force F(t) imparted to the bridge by a guitar string, using a finite difference method previously developed by A. Chaigne [J. Acoust. 5(2), 181–211 (1992)]. This model is used for investigating the influence of various plate parameters on timbre and sound quality of the instrument.
Highlights of hearing protection
Authors: Veneklasen Paul S.
Published: October 2005

Before WWII Dr. Vern Knudsen began the development of ear defenders. In December 1941 this project suddenly took on great urgency; models were developed through V29. Performance was measured by threshold shift. Extreme industrial and military exposures were measured and tentative limit suggested. Development continued at Harvard Psycho-Acoustic Laboratory, resulting in V51. This device was produced in huge quantities for military and industry by MSA. At Harvard two other devices were developed: a comfortable dual-seal circum-aural headphone cushion and an insert device, called the Harveltip, derived from the V51 for coupling a miniature receiver. Both of these were in response to interference of increasing noise with military communication. At Harvard Bob Wallace conceived and tested the possibility of ‘‘active’’ noise suppression. Substantial improvement in direct person-to-person speech communication occurred with a good ear defender. After the war, research progressed with improvement over the Harvintip. The limit of achievable protection with exposed head was due to bone conduction. A method was developed for measurement of attenuation at much higher exposure. All this experience yielded clarification for necessary design parameters.
Circumaural hearing protector development at the WPI Acoustics Laboratory
Authors: Campbell R. H.
Published: October 2005

Professor William B. Wadsworth (1915–1993), affectionately known as Waddy, advanced the state of the art in circumaural hearing protectors when he devised the electrical equivalent circuit for these devices in 1953 and, subsequently, was awarded a patent. This paper briefly describes the life and times at the WPI Acoustics Laboratory surrounding this invention, the struggle to establish measurement standards to perform real-ear threshold shift attenuation tests, and the development of other devices and test methods. The author was a test subject in 1958 and a staff member from 1959 to 1961 at the WPI Acoustics Laboratory, and thereafter as a contributing member of several standards committees in this area. He continues working professionally in the field of hearing protectors and communications in high noise, as well as teaching acoustics and audio engineering at WPI.
The invention of the negative hearing aid
Authors: Hawley Mones E.
Published: October 2005

Speech communication in noisy aircraft was recognized as a problem as early as World War I. The problem persisted throughout World War II, particularly between aircrew members with both talker and listener in loud noise. In 1947 the U.S. Air Force awarded RCA a contract to develop a high-intelligibility speech communications system for use in aircraft. The systems were tested in 120-dB sound pressure levels of synthesized propeller and jet noise. Dr. Harry Olson at RCA had invented an active noise-canceling system using loudspeakers for use in industrial work places. Elvin Simshauser and Mones Hawley of RCA invented a noise-canceling headset—a negative hearing aid—for interphone use in aircraft and other noisy military environments. A microphone on the outside of an earcap monitored the ambient noise near the user’s ear. The signal from the microphone was fed into a filtering and phase-shifting device that matched the attenuation and phase shift of the earcap. The resulting signal had its phase reversed and introduced into the earphone in the ear cap. The sound pressure level from the noise environment under the ear cap was reduced by 10 dB in the demonstration system. When the speech communication signal was superimposed on the signal from the noise-cancellation device, intelligibility was much improved.
History and development of the E-A-RR foam earplug
Authors: Gardner Ross , Berger Elliott H.
Published: October 2005

What does one do with a new vinyl material with exceptionally high energy absorption characteristics? Noise, vibration, and shock control come to mind. If the material can also be foamed and is safe for dermal contact, even broader applications could be anticipated. But, making it into a hearing protector—that’s a step that no one took seriously until the first author of this paper, a chemist who drew on lessons learned through association with an acoustical consultant, decided to try making earplugs. His early work at National Research Corporation in the late 1960s, which he continued in the early 1970s after his research group was purchased by Cabot Corporation, led to the development of an exciting hearing protector, the E-A-RR foam earplug. This earplug, commercially introduced in 1972, which provides an exceptional combination of comfort and attenuation, rapidly became the world’s best-selling hearing protector, spawning numerous imitators and patent litigation in ten countries. The authors will share the story of the inception and development of this product as well as some of the concepts behind its success.
Binaural beats of mistuned consonances
Authors: Feeney M. Patrick , Burns Edward M.
Published: October 2005

In a previous paper [M. P. Feeney and E. M. Burns, J. Acoust. Soc. Am. 93, 2346(A) (1993)] data on the detection of dichotic beats of mistuned consonances (BMCs) were presented. The results showed that subjects could detect mistuning from small-integer frequency ratios for simultaneous-dichotic pure-tones in a 2I-2AFC task. However, there was no conclusive proof that they were using ‘‘beats’’ as the cue, nor conclusive evidence that this was a ‘‘binaural’’ phenomenon, i.e., that detection involved any fusion or lateralization percept. Here, the results of two ‘‘control experiments’’ suggested by alert audience members A. J. M. Houtsma and C. Trahiotis are presented. In the first experiment, subjects were asked to adjust simultaneous-dichotic octaves, using a ‘‘beat-minimization’’ as the criterion. The subject’s variability in this task was much smaller than their variability for adjustments of melodic octaves. In the second experiment, listeners were able to detect intracranial movement between ‘‘frozen’’ dichotic BMCs. Further data on the ability of subjects to detect mistuning from small-integer frequency ratios for dichotic modulation envelopes are also presented. Taken together, the results suggest that dichotic BMCs are beats, are binaural, and are an example of cross-channel processing. [Work supported by the Virginia Merrill Bloedel Hearing Research Center.]
Failure to hear binaural beats below threshold
Authors: Gu Xiang , Wright Beverly A. , Green David M.
Published: October 2005

The ability to hear binaural beats at 200 and 500 Hz was assessed using a tone presented at 20 dB SL to one ear and a tone presented at a variable level ranging from −20 to 20 dB SL in steps of 10 dB to the other ear. A two-alternative forced-choice test was used to measure the listener’s ability to detect the beats. In the ‘‘beating’’ interval, the frequencies of the tones at the two ears differed by 6 Hz. In the ‘‘nonbeating’’ interval, the tones at the two ears had the same frequency. Special care was taken to avoid a potential pitch cue by randomly presenting one of four paired frequencies to the two ears. The experiment was conducted in ‘‘burst’’ and ‘‘fringed’’ condition. There was no evidence to suggest that binaural beats were detectable when the intensity of the variable-level tone was below 0 dB SL. These data fail to replicate a result reported by Groen [Acta Oto-Laryngol. 57, 224–230 (1964)]. [Work supported by NIH.]
Which ear has the asynchronous signal? Interaural effects
Authors: Pollack Irwin
Published: October 2005

Pairs of repeated noise patterns were presented to each ear: one pair with the same fundamental repetition frequency, and one pair with a slightly different fundamental frequency. The task of the listener was to identify the ear with frequency asynchrony. Under conditions where the detection of frequency asynchrony of complex sounds is nearly perfect, the discrimination of which ear exhibits the asynchrony is near chance. Several lines of evidence suggest that the latter difficulty is related to binaural processing of frequency asynchrony across the two ears. Evidence: (1) the accuracy of detection of binaural frequency asynchrony rivals that of monaural frequency asynchrony; (2) the introduction of interaural intensity differences between the two ears improves which-ear discrimination, and reduces binaural asynchrony discrimination; (3) under conditions of equivalent monaural asynchrony, which-ear discrimination reflects differences in binaural asynchrony.
Visualization of acoustic fields using broadband acoustic holography from intensity measurements and the application visualization system
Authors: Burns Thomas H. , Thompson William , Burroughs Courtney B. , Eschenburg Kent
Published: October 2005

Knowledge of the potential and kinetic energy densities, and the complex intensity vector is needed to characterize an acoustic field. Methods are described to obtain and visualize these quantities. Two independent techniques were combined to determine these acoustic field variables, namely, an acoustic intensity measurement technique and near-field acoustic holography (NAH). In the past, conventional acoustic intensity techniques have been used to estimate radiated power and, more recently, to localize and characterize sources. In this study, an acoustic intensity technique was used to obtain, indirectly, the complex pressure on a hologram plane in the near field of a source from a single broadband intensity measurement. From knowledge of the complex pressure on the hologram plane, the free-field was reconstructed using NAH [Loyau etal., J. Acoust. Soc. Am. 84, 1744–1750 (1988)]. After a brief review of these measurement techniques, various methods of visualizing both the steady-state and time-varying acoustic field variables will be introduced. Particular attention will be given to the complex acoustic intensity vector; the vector properties of this quantity will be exploited to characterize the acoustic field. [Work supported by International Jensen, Inc.]
Visualization techniques for acoustic scattered fields
Authors: Isler Sylvia K. , Mehta Vineet
Published: October 2005

A numerical simulation and visualization of time-varying pressure fields in inhomogeneous fluid media is considered. Medium inhomogeneities considered included attenuation, compressibility, and density contrasts. Of particular interest is the analysis of scattering from anisotropic media and the corresponding directional characteristics of the scattered field. The simulation model is based on a Padé approximant representation for the pressure. This model allows the computation of scattered fields at size parameters ka∼O(1). A finite set of wave-number responses is obtained by a solution of Green’s integral equation. The temporal response is obtained by inverse Fourier transformations of the wave-number response modulated by the incident field. Visualization techniques are developed to track strong scattering directions and to identify optimum measurement surfaces. [Work supported in part by and AT&T Bell Laboratories Cooperative Research Fellowship.]
Speech understanding in adult cochlear implant users
Authors: Tyler Richard S. , Lowder Mary , Woodworth George , Parkinson Aaron
Published: October 2005

This review discusses the speech perception results obtained from postlingually deafened adults with cochlear implants using a variety of speech coding strategies and electrode configurations. The speech feature information obtained from feature and whole-signal processing, and single and multichannel stimulation will be examined. Results will be discussed from implants developed in Innsbruck, London, Melbourne, Los Angeles, Paris, Duren, Utah, and San Francisco. Speech perception tests include simple pattern perception, audiovisual recognition, and the understanding of words without visual or contextual clues. Some patients perform poorly, whereas others are able to converse freely on the telephone. For example, word recognition from unknown lists of 50 words can range from 0% to 78% correct. Factors that contribute to successful implant use will be examined. Older patients and patients who have been deafened for long periods of time tend not to perform as well as recently deafened younger patients. The rate of learning over a five-year period shows large individual differences. Most patients show large improvements over the first 6 to 12 months, but some show continued real gains over 4 years.
Progress in the development of speech processors for cochlear prostheses
Authors: Wilson Blake S.
Published: October 2005

New strategies for representing acoustic information with multichannel cochlear implants have produced substantial improvements in speech recognition scores for implant users. One of those strategies, continuousinterleavedsampling (CIS), presents brief pulses in a nonoverlapping sequence across electrode channels, with the pulse amplitudes for each channel reflecting the envelope of a corresponding frequency band of the acoustic input. Recent studies with CIS and related processing strategies will be described, including (a) within-subject comparisons of CIS with the compressedanalog (CA) processor used in a standard clinical device, (b) parametric and control studies with CIS processors, and (c) a preliminary evaluation of a related strategy, virtualchannelinterleavedsampling (VCIS). VCIS processors add to the single-electrode channels of CIS processors virtual channels, produced by simultaneous stimulation of two or more electrodes and eliciting pitch percepts that are distinct from those of single-electrode channels. In general, the CIS/CA comparisons show higher levels of open-set speech recognition with CIS for each of 11 subjects. Results from the additional studies with CIS processors show how choices of pulse duration, pulse rate, and channel update order affect performance. Initial CIS/VCIS comparisons with three subjects do not show an immediate improvement in speech recognition scores with VCIS. [Work supported by NIH.]
A sloping-boundary condition for PE calculations in range-dependent acoustic media
Authors: Brooke Gary H. , Wort Philip M.
Published: October 2005

The traditional one-way parabolic equation (PE) formulation for range-dependent layered acoustic media is modified to include effects associated with the boundary conditions along a sloping interface. Essentially, the boundary condition for continuity of the normal derivative of pressure along a sloping interface can be cast in an approximate form which does not depend on range but does contain terms up to second order in the derivatives with respect to depth. The new sloping-boundary condition is then applied along an ‘‘equivalent’’ horizontal interface within each range-independent step of the PE. Numerical results obtained for the ASA benchmark-wedge problem are used to test the new approach against an accurate two-way PE algorithm. Good agreement in this case suggests that the sloping-boundary condition, incorporated into a one-way PE, can lead to an efficient and accurate alternative to forward calculations using two-way PE.