bluuebunny/crossref_metadata_2025 huggingface dataset:
Vocal closure patterns and source-tract acoustic interaction
Authors: Rothenberg Martin
Published: October 2005

For a strong or well-carrying voice, glottal volume velocity waveforms obtained by inverse filtering are generally consistent with the hypothesis that an interaction between the time-varying glottal impedance and an inertive component in the vocal tract impedance is causing an increase in the relative strength of higher harmonics and a decrease in average air flow. However, the required ratio of vocal tract inertance to glottal impedance is about a factor of 2 less than the greatest estimate that can be made from the glottal and vocal tract geometries using standard acoustic principles. The hypothesis that the unexplained difference is related to the shape of the vocal folds during the closing phase of their vibratory cycle was investigated by comparing glottal air flow and vocal fold contact area waveforms, with the latter obtained using an electroglottograph. To obtain an adequately accurate glottal airflow pulse and adequate synchronization of the recorded waveforms, the glottal flow waveshape was obtained by inverse filtering pharyngeal air pressure instead of the more commonly used radiated pressure or oral air flow. The transfer function of the required inverse filter is described, as well as problems and advantages of this procedure.
Estimating subglottal pressure from esophageal pressure
Authors: Rothenberg Martin , Mahshie James
Published: October 2005

A continuing problem in speech research has been the measurement of air pressure in the trachea during the speech act without invading the trachea itself. A common but problematic approach has been to use a pressure measured in the esophagus, at the level of the trachea, via a small balloon at the end of a catheter which leads to an external transducer. However, many of the problems inherent when using an external transducer can be avoided if esophageal pressure is measured using a miniature, catheter-mounted pressure transducer positioned directly in the esophagus in such a way as to have the diaphragm protected from direct contact with the esophageal walls. Recorded in this manner, esophageal pressure yields fairly accurate records of tracheal pressure, except during esophageal contraction. The frequency range available extends low enough to include the variations in average subglottal pressure that drive the various acoustic sources within the vocal tract, and high enough to include the energy at the lowest subglottal resonance. Data will be presented to support these claims and to illustrate the use of the technique in measuring the coordination between subglottal pressure changes and articulatory movements.
Accuracy of vocal tract area function estimates from acoustic reflection measurements
Authors: Milenkovic Paul
Published: October 2005

A broadband dynamic transducer is used to apply a continuous acoustic excitation at the lips for purposes of measuring the vocal tract area function. A pair of closely spaced microphones inside the tube that conveys this excitation signal is used to resolve forward and backward going acoustic wave components. The reflection coefficients of the vocal tract are obtained by a migration operator which transforms wave components measured at the lips into the equivalent causal response to a finite duration wavelet incident at successive depths into the vocal tract. The accuracy of this measurement technique is influenced by (1) the bandwidth limit imposed by the separation of the microphone pair, (2) the stability of microphone gain, and (3) the sensitivity of the microphones to structure borne vibrations. The combined influence of these limits on accuracy is determined experimentally by making acoustic measurements of the area functions of epoxy molds made to the dimensions of vowels derived from Fant's x-ray data. [Work supported by NIH grants NS 2156 and NS 16377.]
Mandibular rotation and translation during speech
Authors: Edwards Jan , Harris Katherine S.
Published: October 2005

Jaw movement during speech has typically been represented as the movement of a single point even though this representation does not provide adequate information to predict the position of other points on the jaw. However, since the tongue rests on the jaw, such information is needed in order to calculate the jaw component of tongue position for different points on the tongue. In this experiment, a two-dimensional rigid-body model of jaw movement was developed to describe speech-related jaw opening and closing gestures. Jaw movements were decomposed into three components: rotation about the terminal hinge axis and the horizontal and vertical translation of that axis. Data were collected for three subjects during two separate recording sessions. The two-dimensional, rigid-body model of jaw movement appeared to provide useful information on the control of jaw movement. Furthermore, it proved robust enough to preserve inter-speaker differences across the two separate recording sessions. [Work supported by NINCDS grant NS-13617 to Haskins Laboratories.]
Low-frequency noise fields
Authors: Carey W. M. , Wagstaff R. A.
Published: October 2005

Since the classic paper of Wenz [J. Acoust. Soc. Am. 34, 1936 (1962)], ambient noise has been an extensively studied phenomena. Morris [SIO Ref. 75-34, MPL/Scripps (1975)] emphasized the importance of ships as their signals are enhanced when they cross over seamounts or proceed over the continential slopes. Wagstaff [J. Acoust. Soc. Am. 69, 1009 (1981)] showed by the comparison of measurements and calculations that coastal shipping (ships over the continental slope and on the shelf near the slope) must be considered in order that the horizontal directionality be correctly described. He showed that these coastal sources would also affect the vertical directionality. This paper reinforces these findings with new results from down-slope transmission loss (TL), coherence, and noise directionality experiments. High-resolution noise measurements show long-term persistent directional characteristics associated with distant shipping lanes and density patterns. Short-term averages show a temporally dynamic field composed of resolved distant shipping and uncorrelated background noise. These results emphasize the importance of the coherent contribution from shipping to the mid-ocean noise field. Vertical directionality measurements by Anderson et al. (1972) show a broad angular distribution of noise intensity near the horizontal at low frequencies and a peaked distribution about the horizontal at high frequencies. This broad angular distribution near the horizontal was found smooth and indicates that in addition to surface ships, environmental noise influences the vertical directionality.
Inverse scattering from an elastic sphere
Authors: Ayres V. M. , Gaunaurd G.
Published: October 2005

The echoes (or cross sections) returned by an elastic sphere of unknown material composition can be used to determine its composition. Three parameters uniquely define the composition: the density ρ and the speeds of dilatational and shear waves, Cd,Ct. We propose an approach which capitalizes on the findings of the resonance scattering theory, which has shown how to isolate the modal resonances contained within a cross section by a process of background subtraction. These modal resonances, or equivalently the modal mechanical impedances (n = 0,1,2,…) contain all the information required to completely and uniquely determine the composition in an asymptotic way. The problem splits naturally into two types of compositions (rubbery, or nonmetallic, and metallic), and the available cross sections can be of the monostatic or bistatic types. In all combinations of cases, the modal resonances or the modal impedances, can be used to determine the composition in an accurate way that could become the standard method for composition determination by remote sensing. Material composition can be determined as easily a the modal resonances can be extracted from the return, and we illustrate in detail how this is done in several instances.
Wave vector frequency spectrum of a transitioning boundary layer
Authors: Josserand Michel A. , Lauchle Gerald C.
Published: October 2005

The laminar-to-turbulent transition zone continues to be of interest from the flow noise point of view. Lauchle [G. C. Lauchle, J. Acoust. Soc. Am. 67, 158–168 (1980)] has developed a theoretical model to predict the radiated component of the pressure fluctuations that occur in this zone (wavenumber less than or equal to the sonic wavenumber). In the present work, the space-time correlations of a nonhomogeneous indicator function, which monitors the existence of turbulent bursts in the transition zone, have been measured in a subsonic wind tunnel using a zero pressure gradient flat plate. An empirical model based on the measured space-time correlations has been deduced. Using existing models for the statistics of wall pressure fluctuations inside a turbulent boundary layer, the wave vector frequency spectrum of the transition zone is derived for wavenumbers greater than the sonic wavenumber. This result is of fundamental importance and may be useful in the prediction of the coupling of pressure fluctuations inside a transition zone with structural vibrations.
Low-frequency vibration and possible noise influence on Performance
Authors: Irwin Andy W.
Published: October 2005

Task performance in a range of situations is considered. These situations include the effects of just-perceptible vibration in quiet conditions on intricate processes to the influences of higher magnitude vibration, with or without noise intrusion, on the performance of manual dexterity tasks, reading, and in decision processes. The motion forms encompass single axis horizontal and vertical vibration, yaw vibration, and combinations of these components. Data from several laboratory studies, a wide variety of field situations, and diverse population types are described and used in the formulation of assessment methods and criteria.
Low-frequency vibration effects in the space station
Authors: Willshire Kelli F.
Published: October 2005

In response to a Presidential directive, the National Aeronautics and Space Administration (NASA) will launch a space station in the early 1990's. An objective of the Station program is to make customer use of the station cost-effective. To help meet this objective, the Space Station will be capable of permanent habitation with crew rotations of 90 days. Consequently, means of enhancing and sustaining crew productivity are receiving increased attention. Environmental factors can influence productivity either by direct effects on performance or by indirect effects which ultimately impact performance. Noise and vibration, including low-frequency vibration, are two factors being studied. This paper will discuss potential low-frequency vibration sources, impacts on Space Station crew performance, and possible solutions.
Acoustics in an undergraduate engineering curriculum
Authors: Holger David K.
Published: October 2005

The current status of acoustics in typical undergraduate engineering curricula, difficulties associated with increasing the exposure of undergraduate engineering students to acoustics, and strategies for doing so are discussed. Engineering undergraduates are often exposed to selected acoustical topics in an introductory physics course. Most undergraduate engineering curricula do not require formal coursework in acoustics. As a result, typical exposure to acoustical topics occurs either in elective courses that are devoted primarily to acoustics or in required courses such as vibrations or fluid mechanics where acoustics is of secondary importance. Typically, only a small percentage of engineering undergraduates select acoustics courses as electives. A practical strategy for increasing students' exposure to acoustical topics is to make appropriate elective courses available and attractive. In addition to conventional methods for making electives attractive to students and advisers, acoustics courses can be attractive from an engineering accreditation point of view. The potential for design project work, computational work, laboratory experience, and increased basic science background is present in all acoustics courses.
Strategies and struggles for incorporating engineering acoustics in undergraduate curricula
Authors: Lord Harold W.
Published: October 2005

The Occupational Safety and Health Act of 1970, established the principle that the employer is responsible for in-plant health and safety. The subsequent Noise Control Act of 1972 focused on the adverse effects of noise in our work and living environment. The promulgation of these laws extended the responsibility of engineers to include the welfare of those who operate the devices and processes they design or manage, who may be exposed to unwanted emission of health related by-products. Hence, public health and safety concerns should be fundamental to engineering education and engineering acoustics should, therefore, be a logical part of the undergraduate engineering curriculum. This paper addresses the difficulties and strategies for incorporating engineering acoustics in undergraduate curricula at a time when there exists an explosive rate of new technology, and ever increasing pressures to incorporate new course material in an already over-burdened engineering curriculm. These issues are discussed in the light of recent studies and reports which deal with engineering education, namely: (1) a report by a technical panel for the National Institute of Occupational Safety and Health (1984) which gives recommendation for improving engineering practice, education, and research as it relates to occupational safety and health; (2) a report sponsored by the National Research Council (1985) which looks at the state and future of engineering education and practice in the United States; and (3) the criteria for evaluating engineering curricula set forth by the Accreditation Board for Engineering and Technology (1985–86).
Acoustics in a school of music
Authors: Keefe Douglas H.
Published: October 2005

While music students are commonly exposed to musical acoustics in a formal course, there is a trend for music schools to become increasingly active in research involving acoustics, with applications to relevant sub-disciplines of music. Upper-level undergraduate courses in a music school have been designed for use also by graduate students in the following areas: a full-year sequence in music science, computer applications to music, and musical applications of digital signal processing. All have instructional components involving acoustics, and all make use of a microcomputer center comprised of six music workstations. Specialized peripherals and software demonstrate acoustical phenomena and serve as tools for work in musical acoustics, music perception and cognition, music engineering, computer music, and automated music transcription. One important benefit is that effective use of computers helps music students without extensive background in calculus or the physical sciences grasp technical concepts more readily.
Waves á la Fourier
Authors: Ayers R. Dean
Published: October 2005

A senior/graduate elective course on Fourier transforms and the physics of vibrations and waves has now been taught for 6 years. The textbook by Ronald Bracewell is used in the first half of the semester to lay the mathematical foundations: piecewise functions, convolution, generalized functions, the transform and the series, theorems on transform pairs, and techniques for evaluating transforms. The second half of the semester is devoted to physical applications, with relatively little time spent on vibrations or waveforms; the major focus is on the use of spatial transforms to describe basic processes of radiation and imaging. Examples are drawn from acoustics, optics, solid-state physics, and medical imaging. The emphasis throughout this course is on the unifying structure of this approach to linear physics. Computational techniques and applications involving statistics or noise are deliberately avoided. An optional laboratory allows the students to see the ideas of the lecture course illustrated in concrete examples. Several students have gone on to do master's theses in acoustics after taking this course, and others have commented on its usefulness to them in a variety of fields.
Acoustics laboratory experiments in the undergraduate curriculum
Authors: Rossing Thomas D.
Published: October 2005

It is impossible to overemphasize the importance of laboratory experiments to the student learning acoustics. We have developed an acoustics laboratory with a library of over 50 acoustics experiments, ranging from the introductory to the advanced undergraduate/graduate level. This laboratory serves our undergraduate physics curriculum in a variety of ways, by providing: (1) experiments for a one-credit acoustics laboratory course; (2) optional experiments and “experimental problems” for an intermediate course in waves and vibration, an advanced acoustics course, and other courses; (3) open-ended experiments for undergraduate research projects; (4) demonstration experiments for lectures in mechanics, acoustics, and other intermediate courses; (5) acoustics experiments for our intermediate laboratory course; and (6) (fortunately or unfortunately) a band of well-maintained loanable test equipment for other teachers. We have prepared an acoustics laboratory manual, which is now distributed commerically by Cenco. Several experiments will be described and demonstrated as time permits.
Perceptual categorization of a compressed vowel space
Authors: Varley Douglas , Landahl Karen , Gould Herbert Jay
Published: October 2005

Acoustic properties of vowel tokens produced by a speaker with radical malformation of the vocal tract have been investigated. A study of vowel formant structures demonstrated that the speaker does not maintain expected intravowel formant relationships which results in a diminished vowel space. However, the speakerapos;s vowel categories do display formant structures which are patterned after the norm [Peterson-Falzone and Landahl, Speech and Language (Academic, New York, 1981), Vol. 6]. The question remains in what way the anomalous formant structures will affect vowel categorization by listeners. A previous report of modeled speech [Landahl and Gould, J. Acoust. Soc. Am. Suppl. 1 77, S100 (1985)] suggests that this speaker's perceived vowel space will not include vowel categories at the periphery. The current study reports perceptual testing of vowel tokens produced by this speaker. The results of these tests concerning the existence of peripheral vowels are in accord with earlier predictions that the speaker would be unable to maintain category distinctions among more central vowels, all vowel categories being compressed in the diminished vowel space. We predicted [I]→[ɛ], [æ]→[ɛ], and [ᴜ]→[ɔ] mismatches. The results are essentially in line with these predictions. [Work supported by NINCDS.]
Design of high-intensity, high-efficiency sirens for acoustic agglomeration
Authors: Pla F. G. , Reethof G.
Published: October 2005

Acoustic agglomeration requires sound pressure levels in the order of 155 to 165 dB, and powers ranging from a few hundred up to several hundred kilowatts, depending on the application. The cost of producing such power in the 1- to 4-kHz range is very significant and dictates the need for very efficient sound sources such as sirens. It is shown how nonlinear distortion of the acoustic wave and its related losses affect the performance of a siren. The presence of the mean flow through the siren passages is very important in the way it changes the local speed of sound and smoothes out the resonances of the various parts of the system. The magnitude of these effects is related to the siren operating point, i.e., the pressure ratio at which it is run and to the size of the stator ports chosen to reach the desired sound pressure level in the agglomerator. The importance of the acoustic properties of the agglomeration chamber in which the siren radiates is also critical and the chamber can be designed to minimized the losses due to nonlinear acoustic wave propagation.
The modeling of rigid-walled acoustic horns with sharp flare discontinuities
Authors: McLean James , Hixson Elmer L.
Published: October 2005

A composite model capable of predicting the throat impedance of so-called “constant directivity” horns is presented. These horns cannot be modeled using methods based solely on Webster's equation because they possess a sharp discontinuity which is necessary to achieve the desired radiative characteristics. The composite model combines one- and two-dimensional analyses. A two-dimensional mode-matching technique is used to modal the portion of the horn in which the flare is discontinuous. The parts of the horn which flare smoothly are modeled using a generalized transmission line approach which can account for viscous losses and is shown to be equivalent to Webster's equation when losses are absent. It is necessary to use this generalized transmission line approach for the smoothly flared portions of the horn because the complex geometry of the horn prohibits analytical solution of Webster's equation. The composite model is used to predict the performance of a horn typical of the “constant directivity” horns and the predictions are compared to experimental data showing good correlation. [Work supported by JBL Inc.]
Finite difference errors in a two-microphone technique
Authors: Mielnicka-Pate A. L.
Published: October 2005

A numerical investigation of finite difference errors will be presented for a monopole and a baffled piston. The research is an extension of work reported earlier in a paper entitled “Finite Difference Nearfield Errors for a Baffled Vibrating Piston” [A. L. Mielnicka-Pate, J. Acoust. Soc. Am. Suppl. 1 76, S45 (1984)]. The finite difference errors were calculated for the acoustic pressure, particle velocity, sound intensity, and radiated power. The analysis included three different integration surfaces such as a plane located in a nearfield, a sphere, and a cube centered around a source. Directions and locations for significant errors in pressure, velocity, and intensity were identified. The finite difference errors will be discussed in terms of source characteristics and microphone locations. Numerical results of the radiated power calculations will be presented to illustrate the effects of a choice of integration surfaces.
A potential link between structural parameters and the radiated sound field of a vibrator
Authors: Ebbitt Gordon , Strong William Y.
Published: October 2005

Two powerful techniques available for studying structural vibration and radiation are nearfield acoustical holography (NAH) and modal analysis. The NAH technique measures the pressure field over a surface sound field in the half-space above the radiator. The modal analysis technique determines the structural parameters of the vibrator. Since both techniques can determine the velocity of the vibrating surface, it seems likely that linking these two techniques via the measured velocity could provide useful insight into how the structural parameters affect the radiated sound field. A modal analysis and a NAH analysis of a simple plate has been performed and the results of attempting to link these two techniques will be discussed.
An algorithm for selecting transducer element array positions
Authors: Wiemann King W. , Hughes W. Jack
Published: October 2005

A lumped-parameter equivalent circuit of a tonpilz transducer is used to predict element amplitude and phase tolerance for different radiation loadings, based on in-air measurements of the transducers. Relationships among the measurable parameters of a transducer and its performance characteristics have been explored. Preliminary estimates of acceptable tolerances for each circuit parameter have been established. A two-part technique to determine the placement of transducer elements in an array which minimizes the impact of element tolerances on the directional beam pattern has been developed. This technique has been used to place, in an array, elements with tolerances of ± 9% in amplitude and ± 11° in phase. The resulting theoretical array response achieved sidelobe levels that were within 1 dB of the −40-dB design. The first half of the technique selects four elements at a time from a larger selection pool, and places them in a manner that partially cancels their respective amplitude and phase variations. The second half of the technique uses a permutation search algorithm which rearranges the initial placement of elements in and out of the array looking for improvements in the array response.
A doubly resonant longitudinal vibrator transducer
Authors: Thompson Stephen C.
Published: October 2005

The conventionally designed longitudinal vibrator transducer consisting of a double-mass-loaded active material stack is widely used in both single element and array configurations. This device, commonly called a “tonpilz” transducer, is well suited for operation at frequencies near its longitudinal resonance; however practical considerations normally limit the bandwidth attainable. It is difficult to design a tonpilz transducer with an operating mode Q of less than three which provides a limiting fractional bandwidth of approximately 0.30. This paper describes a modification to the simple tonpilz design which creates a new class of longitudinal vibrator transducer. This new device has two longitudinal vibration modes each of which can be used for operation. It is possible to design transducers of this type to be operated throughout the entire frequency band between the two resonance frequencies. Several devices of this type have been built and tested to verify that the predicted performance can be attained. Electrical and acoustical test data from these devices, which provide a flat transmitting bandwidth greater than an octave, will be presented. [Work supported by Gould Independent Research and Development Program. A patent is pending on this design type.]
Application of diffusion bonding to thin ceramic low-voltage sonar transducers
Authors: Johnson Michael P.
Published: October 2005

Piezoelectric ceramic stacks used in sonar transducers have previously been bonded together using organic adhesives. These adhesives in general have moduli an order of magnitude lower and loss factors an order of magnitude higher than the ceramics themselves. As a consequence, degradation of the electromechanical coupling factor and decreased efficiency occur. This leads to increased operating temperatures and possibly thermal runaway. To avoid these difficulties, it has been necessary to use relatively thick ceramic to minimize the number of adhesive bonds. This results in the need for step up transformers to provide the required voltages for high power sonar. A diffusion bonding process, reported on previously [J. Acoust. Soc. Am. Suppl. 1 72, S23 (1982)] can be used to circumvent these difficulties and eliminate the need for step up transformers. A previously developed and tested transducer design has been modified using this process. Electrical as well as acoustical data will be presented for a 3 × 3 subarray of diffusion bonded elements. A comparison of effective ceramic parameters between the conventional, diffusion bonded, and modified adhesive bonded stacks will also be presented.
Frequencies and modeshape calculations for uniform membranes
Authors: Worman W. E.
Published: October 2005

Most methods for calculating frequencies of a membrane of arbitrary shape require a reasonable guess at an approximate wavefunction. In the method presented in this paper both approximate wavefunctions and frequencies are determined. A wavefunction can always be expressed in the form w(r,θ) =  ∑ n = 0∞ AnJn(kr) cos nθ +  ∑ n = 1∞ BnJn (kr) sin nθ. Approximate by terminating the summations at n = N. Require then that the boundary condition be satisfied at 2N + 1 discrete points on the boundary to generate 2N + 1 simultaneous homogeneous equations linear in the A's and B's. These have nontrivial solutions for the discrete k values for which the determinant of the matrix of coefficients vanishes. For one of these k values (wavenumbers) one can determine the A's and B's and hence the wavefunction. Examples of this procedure will be presented for membranes of various shapes. The method is accurate and effective requiring only a reasonable amount of time. [Work completed at Case Western Reserve University, Cleveland, OH 44106.]
Effects of 5 years of filler and varnish seasoning on the eigenmodes in four pairs of viola plates
Authors: Hutchins Carleen M.
Published: October 2005

In 1979–1980, four pairs of top and back viola plates, finished and tuned according to Hutchins' plate tuning method, were measured for the frequency and damping of modes ♯1, ♯2, and ♯5 and weighted carefully. Two of the above pairs were tested with filler on, and two “in the white” with subsequent application of filler. Each pair of top and back plates was then glued to its own set of ribs and six coats of violin oil varnish applied. The four varnished “corpuses” were hung in an unheated attic for 5 years. The plates were then removed from their ribs and a similar series of tests applied. The measured changes in mode frequencies, damping, and overall weight of each plate will be presented and implications for violin making discussed.
Acoustic comparison of soprano solo and choir singing
Authors: Rossing Thomas D. , Sundberg Johan , Ternström Sten
Published: October 2005

Soprano singers were recorded while singing similar texts in both choir and solo modes of performance. A comparison of long-term-average spectra of similar passages in both modes indicates that subjects used different tactics to achieve somewhat higher concentrations of energy in the 2–4 kHz range when singing in the solo mode. It is likely that this effect resulted both from a slight change of the voice source from choir to solo singing, and from a denser clustering of the higher formants, similar to that which leads to a “singer's formant” in male singers. The subjects used slightly more vibrato when singing in the solo mode. The differences observed between solo and choir singing, though not as great as previously reported in male singers [J. Acoust. Soc. Am. 76, 541 (1984); paper submitted to J. Acoust. Soc. Am.], they are nevertheless considered to be significant. [Work supported by NSF and by the Swedish Research Council for Humanities and Social Sciences.]
Generic spectrum envelope functions for orchestral wind instruments
Authors: Benade A. H.
Published: October 2005

The basic behavior of the room-average spectrum envelopes E(f) = M(f)T(f) of orchestral winds is describable via a limited number of functional forms, each with a breakpoint frequency parameter characteristic of the instrument itself. At the mezzoforte playing level the internal (mouthpiece) spectrum envelope comprises two factors, M and T1 for the nonclarinet woodwinds and brasses. For woodwinds, Mw (x) = sin (πx)/(πx)2, where x = f/fbreak, and for the brasses the coresponding function is M>b = (1/x) (1 + x2)1/2. The clarinet odd harmonics fit Mco = (1/x)/(1 + x4)1/2, while Mce = x/(1 + x8)1/2 for the evens. The function T1 = [(1 − x)2 + Bx2]1/2 depends on the mouthpiece cavity-and-constriction plus main bore entrance taper. T1 = 1 for clarinets. The transfer out to room-average spectra involves a tonehole based transfer function Th = x/(1 + x2)1/2 for nonclarinet woodwinds, and a bell-controlled function Tb = x3/2/(1 + x3)1/2 for the brasses. For the clarinet odd harmonics Tco = Th, while Tce = 1 for the even components. Playing-level modifications to the envelopes are similarly describable for the various instrumental types. Numerous illustrative examples will be presented. [Work supported by NSF.]
Distortion analysis using Hermite polynomials
Authors: Williamson Malcolm J.
Published: October 2005

This paper describes a mathematical method for analyzing the distortion caused by a memoryless nonlinear amplifier. Two common examples of such nonlinear systems are a mu-law compander and a peak clipper. The motivation for this work is to quantify the distortion which would be present in some algorithms proposed for speech processing. The most common method for measuring distortion employs one or two sinusoids at the input and measures energy in harmonics and cross-product distortions. However, those techniques do not fully carry over to more complex signals such as speech. Our technique is to analyze the nonlinear function in terms of Hermite polynomials. This gives a measure of the total distortion as well as the distortion in each harmonic. This approach is equivalent to using Gaussian (not necessarily white) noise as a probe signal and calculating the linear component using the zero-lag cross correlation. The mathematical predictions compare closely with experimental results using short sections of speech.
Application of smoothed Wigner distribution (WD) to speech signals
Authors: Garudadri H. , Benguerel A.-P. , Gilbert J. H. V. , Beddoes M. P.
Published: October 2005

The WD is a powerful tool for the simultaneous time-frequency analysis of time-varying signals [Martin and Flandrin, IEEE Trans. Acoust. Speech Signal Process. ASSP-33 (Dec. 1985)]. For signals with simple time-frequency structures (e.g., frequency-modulated signals), the WD provides excellent resolution in the time-frequency (t − ω) plane, as compared with conventional short-time spectral analysis techniques. The interpretation of the WD becomes more difficult as the time-frequency relationship of the signal gets more complex (e.g., in speech), because of the presence of cross terms and of regions below (i.e., negative) the t − ω plane. This disadvantage can be overcome by smoothing the WD in the t and ω directions over a region TΩ > 1 (T and Ω are the “smear” values in time and frequency) [Janssen and Claasen, IEEE Trans. Acoust. Speech Signal Process. ASSP-33 (Aug. 1985)]. Such a smoothing process, however, makes the WD and the spectrogram equivalent in terms of the time and frequency resolutions. This presentation is directed towards comparing the performances of the WD (with partial smoothing, i.e., T < 1) and the spectrogram. Nonspeech signals as well as speech signals will be used to demonstrate the application of WD to the analysis of speech signals. [Work supported by NSERC, Canada.]
Multidimensional scaling of confusions produced by speech recognition systems
Authors: Yuchtman Moshe , Nusbaum Howard C. , Davis Chris K.
Published: October 2005

The performance of an automatic speech recognition device is typically characterized by recognition accuracy, either in terms percentage of words recognized correctly or some measure of the types of recognition errors made by the system. As long as performance is significantly above chance or below perfect recognition, these measures cannot completely describe the operational characteristics of a recognition system. In order to provide a more complete account of recognition performance, we used multidimensional scaling of confusion matrices to compare the performance of three speech recognizers. The test vocabulary consisted of the “E” subset of the alphabet: B, C, D, E, G, P, T, V, Z. Scaling solutions were obtained for each recognizer using direct distance scores produced by the recognition algorithm, as well as distance measures (d′) between each pair of words derived from confusion matrices constructed from recognition responses. The results of these analyses indicate that the patterns of errors have an underlying structure that is distinctively different for each of the three recognizers. In addition, excellent agreement exists between solutions obtained for recognizer-generated distance scores and actual recognition data. The implications of these findings for performance assessment, improved training procedures, and decision-making algorithms will be discussed. [Work supported in part by IBM.]
Automatic recognition of intervocalic voiced stops
Authors: Franco Horacio
Published: October 2005

The recognition accuracy for intervocalic voiced plosives using a simple consonant detector combined with a context-dependent Bayesian classifier was studied. The consonantal segments were located by running a dip detector over the smoothed log-energy contour. The vocalic nuclei were located at the segments of least spectral change. LPC spectra along the vocalic segments were the features for their classification which was performed by a compound Bayes decision procedure. The feature set for the Bayesian context-dependent classification of voiced stop sounds was a sequence of three selected LPC spectra located along the amplitude dip. One spectrum was obtained at the point of minimal energy and the other two were obtained at the points of maximum slope of the smoothed log-energy contour at the beginning and end of the consonantal segment. A recognition performance of 92% was obtained in a speaker-dependent manner using a database consisting of nearly 1300 consonants embedded in VCVCVCV nonsense sequences where the V's were the vowels /a,i,u/ and the C's were the voiced stops /b,d,g/ uttered by two male Argentine Spanish speakers.
Formant transitions as partial invariants in the identification of voiced stops
Authors: Nearey T. M. , Shammass S. E. , Dow M. L.
Published: October 2005

The F2 trajectories for /b,d,g/ in CV syllables are often summarized by initial F2 frequency (F2i) and steady-state vowel F2 (F2v). Trajectories were measured for 330 Canadian English CV syllables (3 stops × 11 vowels × 10 speakers.) Plots for each stop (vowels pooled) indicated a strong linear relationship between F2i and F2v. A regression line fitted to each plot represents an invariant relational property of the corresponding consonant, and F2 trajectories are not sufficient to uniquely specify the stops, since the lines for the three consonants intersect (indicating category overlap). However, the slopes and intercepts for the three consonants are distinct and thus represent “partly distinctive invariant properties” or “partial invariants.” Similar patterns obtain for F3. Use of partial invariants of F2/F3 trajectories in a classification algorithm (based on distance from category lines) results in an identification rate over 70%. Extensions of the algorithm to include spectral shape information will be discussed, as will relationships to perceptual data.
Some aspects of the auditory analysis of lateral consonants
Authors: Bladon Anthony , Burleigh Peter
Published: October 2005

From its consistent presence in sweep-frequency measurements and in natural speech data, the spectral antiformant in the 2700-Hz region might be thought to be a candidate perceptual feature involved in distinguishing laterals from vowels. However, in isolated synthetic steady-state stimuli, the presence versus absence of an antiformant is barely detectable. This finding can be interepreted (consistent with other experiments using fricatives) as indicating that the auditory filter “smoothes over” the spectral notch. The width of the notch was exaggerated experimentally, as a probe for the width of the auditory filter in speech. A value in excess of 1 bark, and consistent with the assumption of a 3- to 3.5-bark-wide filter, was found. This property of auditory analysis seems to underlie a rather large range of speech effects. A second question which arises is whether the antiformant might be detected by temporal auditory mechanisms, such as enhanced onset/offset of discharge in the appropriate channel. But, when stimuli with and without the antiformant were embedded in an intervocalic context, hardly any evidence of perceptual enhancement was found: Differences were again barely detectable. Instead, apparently, the contrast lateral versus vowel relies on grosser auditory characteristics such as transition duration, steady-state duration, and the amplitude envelope.
Effects of speaking rate on the perception of syllable-initial stop consonants
Authors: Jongman Allard
Published: October 2005

It has been shown that the effect of speaking rate on the perception of voice onset time (VOT) is strongest in the immediate vicinity of the target stop consonant: Listeners interpret a given VOT value in accordance with the duration of the CV syllable containing the stop consonant [A. Q. Summerfield, J. Exp. Psychol. Human Percept. Perform. 7, 1074–1095 (1981); J. L. Miller et al., J. Acoust. Soc. Am. Suppl. 1 76, S89 (1984)]. These studies used stimuli that are unlikely to occur in natural speech. The first study used simplified synthetic VOT continua; the second study used computer-edited natural-speech VOT continua, but in which syllable duration was kept constant, so that vowel duration decreased as VOT increased. The present study explored the effects of speaking rate on the perception of syllable-initial stop consonants in more natural circumstances. Natural-speech [bi-pi] continua were constructed, in which vowel rather than syllable duration was kept constant. Preliminary results indicate that the phoneme boundary is not, or only minimally, affected by changes in vowel duration.
The Biot-Stoll model: An experimental assessment
Authors: Brunson Burlie A. , Holland Charles W.
Published: October 2005

A promising approach in the prediction of acoustic wave speeds and attenuations in marine sediments is the use of the Biot theory as implemented by Stoll. The accuracy of the Biot-Stoll sediment model was examined for a variety of natural marine sediments taken from four shallow water sites in the Mediterranean Sea, including silty-clay, sand, and gravel. Comparisons were made between the Biot-Stoll model predictions of compressional velocity, compressional attenuation, and shear velocity with in situ and laboratory measurements. The model predictions on the whole showed excellent agreement with the measured data.
Ray-parameter method for acoustic velocity measurements of sediment layers
Authors: Zhang Shuying
Published: October 2005

The ray-parameter method for acoustic velocity measurements of sediment layers presented by Bryan [Physics of Sound in Marine Sediments, edited by L. Hampton (Plenum, New York, 1974), pp. 119–130] has been extended to the case of layers having a common dip angle φ. The expressions for calculating the velocity V2 and thickness Δz of the second layer are V2 = ((x2 − x1)cos[sin−1(λV1)]λ(t2 − t1)cos[sin−1(λV1) − φ] − λ(x2 − x1)sin φ/V1)1/2, (1) ΔZ = V22(1 − λ2V22)1/2((t2 − t1) − (x2 − x1)sin φV1cos[sin(λV1) − φ]), (2) where the velocity of the first layer (V1) is premeasured, and the parameters λ, (x2 − x1), and (t2 − t1) are determined by a certain data processing procedure from seismic records. The errors of measuring velocities caused by uncertainties of λ and φ can be estimated as: (δV2V2)λ = 12δλλ(Z1ΔZtan θ1tan θ2cos2 θ2 − sin2 θ2), (3) where θ2 = sin−1[(V2/V1)sin θ1], θ1 is the angle of incidence to the layer, and z1 is the altitude of the receiving streamer above the layer; (δV2V2)φ = δφ2V1(t2 − t1)sin(θ1 − φ) − (x2 − x1)cos φV1(t2 − t1)cos(θ1 − φ) − (x2 − x1)sin φ. (4)
The determination of layer thickness in layered ocean bottoms using acoustic fields
Authors: Werby Michael F. , Ali Hassan B.
Published: October 2005

For a shallow water environment, it is theoretically possible to extract information on properties of the ocean bottom from the beamformed response of a cw source. In particular, using synthetic aperture offset measurements from a cw source (for discrete frequencies in the range 20–100 Hz), one can obtain information on bottom layering and the associated compressional and shear velocities. The technique rests on the correspondence between the maximum beamformer response in angular space and the normal mode response in wavenumber space. The present work extends the preceding method by using the properties so determined to deduce the thickness of the relevant layers. The technique is based on the comparison of results of calculations for the unknown layer with those obtained from a “reference” layer. Examples of the method are presented.
Reflection tomography imaging of submerged objects
Authors: Gaumond Charles F. , Abraham Phillip B.
Published: October 2005

In a continuing study of acoustic reflection tomography [C. F. Gaumond and P. B. Abraham, J. Acous. Soc. Am. Suppl. 1 78, S80 (1985)], results obtained from synthetic data for prolate spheroids will be presented. These will be compared with similar results for various spherical scatterers. This novel method of image reconstruction of strong scatterers will be briefly reviewed.
Bivariate swimbladder size allometry properties of mesopelagic fish and bioacoustic prediction
Authors: Saenger R. Alfred
Published: October 2005

One of the fundamental problems in modeling the low-frequency contributions to scattering strength of swimbladder fish is to transform a measured or theoretical size-depth distribution h(l|z) for an individual species to the corresponding distribution q(R|z) of equivalent swimbladder radius R. This is usually done using a y-on-x regression line relation for the species, where y = 1n R, x = 1n l, and l = fish standard length. Analysis of archival Ocean Acre swimbladder size allometry data for 38 species indicates that this procedure is physically inadequate. Due to biological variability, a wide range of R values is associated with a fish of length l. Since resonance scattering from gas-filled swimbladders is depth dependent, important scattering contributions may come from fish with bladder sizes R not lying on the regression line. To avoid systematic prediction errors, the x,y distribution p (x,y) rather than the regression line must be used to calculate q(R|z). This transformation problem and its modeling consequences are examined in light of the fact that p(x,y) for many swimbladder species has been found to be bivariate normal.
Acoustics of Oriental gongs and bells
Authors: Rossing Thomas D.
Published: October 2005

Many different types of bells and gongs are found throughout the Far East. Each has an interesting story to tell the acoustician as well as the musicologist. Among the instruments we will discuss are gamelan gongs from Indonesia, opera gongs from China, gongs from Thailand, temple bells from Japan, and chime bells from China.
Twelve Nasca panpipes
Authors: Haeberli Joerg
Published: October 2005

Tonometric meaurements for 12 Nasca panpipes, two of which can be dated, are presented. The acoustical properties of the two types of identified pipes differ significantly. Specific frequencies reoccur among different instruments. This allowed building of a sequence of steps which are based on equal frequency intervals in hertz, rather than on a logarithmic series. The accuracy of tuning was better than 10 Hz for over 60% of the pipes. This is adequate for panpipes that are activated by mouth blowing.
The ancient Chinese “calendrical” pitchpipes
Authors: McClain Ernest G.
Published: October 2005

The most important symbol of “order” or “natural law” in Han China (202 B.C. to 220 A.D.) was a set of 12 end-blown bamboo tubes known today as the “pitchpipes.” Each pipe sounded a single tone of the chromatic octave and served as “tonic” for a specific calendrical period. All weights and measures of the empire were derived from the dimensions and capacity of the fundamental pipe, from which the lengths of other pipes were proportioned according to string ratios. It is of interest to the history of acoustical physics and to the study of science and civilization in China to ask how much “order” such pipes display when blown. Within a narrow Han limitation to diameter/length ratios between 1/30 and 1/27, and to fundamental lengths somewhat shorter than the foot, the cyclic excess in “pure” wholetones of 9:8 offsets pipe “end correction” sufficiently to permit the wholetone scale on the tonic (i.e., the six yang, “male” tones, the only ones strictly designated lu, meaning “order”) to be sounded at or very near equal temperament values. Thus pitchpipes—within well-defined physical parameters—were arguably an appropriate cultural icon for cyclic orderliness. Sinologists overlook this lawfulness.