bluuebunny/crossref_metadata_2025 huggingface dataset:
An approach to noise cancellation for speech recognition systems
Authors: Kurokawa Shuji , Mito Michiharu , Horio Yoshihiko , Nakamura Shogo
Published: October 2005

This paper describes an approach to simple noise cancellation for speech recognition, such as template matching on time-spectrum patterns. This system has two inputs. The primary input Ip receives a speech signal S(t) corrupted by an environmental noise N(t). The second input Ir, used as a reference signal, receives kN(t + td), that is, a correlated version of the noise in Ip, where k represents the level ratio of kN(t + td) to N(t) included in the primary signal, and td is the time difference between Ip and Ir. The two input signals are analyzed using a bandpass filter bank (BPFB). The noise cancellation is performed in the following manner: (1) The level ratio k is estimated; (2) the time difference td is also estimated using k; (3) the cancellation is performed on every output of the BPFB using k and td. At the same time, a speech section is detected by using the variations in each spectrum difference between the two input signals. The values of the parameters k and td are fixed at the beginning of the speech section while a speech signal exists.
Precise detection of isolated words in noise by the linear prediction method
Authors: Kobatake Hidefumi , Tawa Katsuhisa
Published: October 2005

This paper presents a new method of detection for speech in stationary noise. First, a fundamental principle of segmentation is derived. It is shown that the final prediction error (FPE) decreases monotonically as the length of the processed signal increases, if the signal is stationary or homogeneous, and the FPE tends to increase when the homogeneity of the signal is lost. This means that the instant when the FPE is at its minimum corresponds to the position of the initial word. Based on this principle, the final prediction error is proposed as a criterion to find the speech segments in noise precisely. Second, experiments to test the performance of the proposed method are presented. Experimental results show that the proposed method is sensitive enough to detect changes in signal characteristics. A detailed comparison of the accuracy of the proposed method and that of conventional methods has been made and the effectiveness of the proposed method is shown.
Nonlinear time-scale modification of speech signal with varied segmental duration characteristics
Authors: Tohkura Yoh'ichi , Kitahara Yoshinori
Published: October 2005

Segmental duration of each phoneme changes depending upon the speaking rate. Generally, vowel parts are easier to be compressed or expanded than consonant parts are in fast or slow speech, respectively. Questions raised in this paper include how the speaking rate can be extracted from the speech signal without knowing the content (i.e., phonetic information) and what kind of time-scale modification can be chosen in order to control speaking rate. First, the segmental duration compressibility of the speech signal was defined by path slopes in DTW spectral matching when utterances with various kinds of speaking rates were matched to a reference utterance of a normal speaking rate. On the assumption that the compressibility is inversely proportional to segmental spectrum changes, the relationship between the compressibility and the average cepstral time difference Δcep [S. Furui, IEEE Trans. Acoust. Speech Signal Process. ASSP-34, 52–59 (1986)] was studied. The results showed that the Δcep is an efficient parameter to represent the compressibility. By using the Δcep as a control parameter of the speaking rate, nonlinear time-scale modification can be achieved without speech quality degradation.
Advances in microparticle characterization using high-frequency (30 MHz) acoustic scattering
Authors: Chen X. , Roy R. A. , Apfel R. E.
Published: October 2005

There are a variety of characterization techniques for individual and ensembles of particles. Of the former category, however, there are none that characterize microparticles (in the micron range) by the mechanical descriptors: density and compressibility. An apparatus developed in the Yale Acoustics Laboratory accomplishes this objective, thereby allowing one to determine the distribution of a given property for a population of particles individually convected past the focal zone of confocally positioned acoustic transmitter and receivers. In earlier work, two- and three-transducer systems operating at 30 MHz have produced data sets that have allowed for the prediction of density and compressibility of particles (e.g., red blood cells) given a priori information about the size of these particles. More recently, the two-transducer acoustic apparatus has been married to a modified electrozone sensing cell (such as found in Coulter counters) that provides size information for each cell. Particles pass first through the electrozone unit and then through the focal zone of the acoustic unit. The result is that now the volume, compressibility, and density can be determined for each particle. Initial results and limitations of the apparatus will be presented. [Work supported by Grant R01-GM30419 of the U.S. National Institutes of Health.]
Shock wave production in medical ultrasound—A new computational approach
Authors: Christopher P.T. , Parker K.J.
Published: October 2005

At high intensities, the presence of nonlinear propagation effects can influence the performance of medical ultrasound devices. In general, models that can fully account for the combined effects of tissue nonlinearities, tissue attenuation, and beam diffraction effects are quite complex, require extensive computational time, and provide little physical insight as to the interaction of these processes. A novel, computationally efficient model has been developed that accounts for the effects of nonlinearities, attenuation, and diffraction. The model propagates a circularly symmetric beampattern forward in small axial increments. Over the increment, the fundamental and harmonics are diffracted and attenuated linearly using angular spectrum decomposition techniques. Then, the complex amplitudes of the fundamental and harmonics are altered according to theory of plane-wave propagation in a medium with nonlinear behavior. This hybrid, incremental algorithm is shown to reproduce theoretical Gaussian and focused piston beampatterns described by others, and also matches experimental data from focused ultrasound beampatterns measured in the present study. The new hybrid algorithm is computationally efficient and provides some insight into the relative importance of nonlinearities, attenuation, and focusing in medical imaging and lithotripsy conditions. [Work supported by NIH.]
Penn State's graduate program in acoustics
Authors: Stuart Alan D.
Published: October 2005

The posters will present a comprehensive overview of the graduate program in acoustics at Penn State. In part, the posters will contain the following: (1) descriptions of the acoustics and related courses offered; (2) examples of the facilities available and the research being conducted; (3) listings of graduate theses, completed and in progress; (4) information on special extended education programs—telecommunication and summer—leading to a master's degree in acoustics.
The graduate program in neuroscience at Syracuse University
Authors: Smith Robert L. , Verrillo Ronald T.
Published: October 2005

The Ph.D. program in neuroscience at the Institute for Sensory Research (ISR) of the L.C. Smith College of Engineering provides an excellent opportunity for graduate study and research in many areas of interest to members of ASA. The program is housed at ISR, a multidisciplinary research center devoted to the advanced study of the senses, particularly hearing, touch, and vision. Faculty include several Fellows and Members of ASA: J. J. Zwislocki (auditory psychophysics and biophysics), R. T. Verrillo (vibrotactile psychophysics), R. L. Smith (auditory electrophysiology), E. M. Relkin (auditory electrophysiology), N. B. Slepecky (auditory anatomy), S. J. Bolanowski (cutaneous electrophysiology), and G. A. Gescheider (vibrotactile psychophysics), along with R. B. Barlow (vision), S.C. Chamberlain (anatomy), G. A. Engbretson (vision), D. G. Pelli (vision), and J. Baumgold (neurochemistry). The program of study provides an intensive background in pertinent physical, life, and engineering sciences. Students are directly involved in research work at ISR. They have undergraduate backgrounds in a variety of scientific and engineering areas and generally receive full support through assistant-ships and fellowships. ISR consists of 27 000 square feet of mostly new, fully computerized, laboratory, office, and auxiliary space and is located on the scenic Skytop campus. A visit is encouraged during the May 1989 ASA meeting.
Target recognition in acoustics
Authors: Gaunaurd G. C.
Published: October 2005

The classification/identification of underwater stationary objects can be acoustically achieved by means of passive or active sonar. Moving targets are the “easy case,” since they can be identified by their Doppler factor, and will not be discussed here. By means of passive sonar, one merely listens with the best possible receiver to the noises/sounds emitted by the noisy target. The targets are passively “identified” when the spectra of the radiated sound fields they emit are seen to match certain spectral lines known a priori for certain desired objects. Identification via active sonar is independent of how noisy the target may be. Here, the echoes returned by the targets when they are insonified by sonar pings are analyzed. These are classical inverse scattering situations in which a returned “signature” in the time or the frequency domain is used to extract physical characteristics of the targets in question. This paper will concentrate on this type of (active) classification technique [i.e., G. Gaunaurd, IEEE J. Ocean. Engr. OE-12, 419–422 (1987)], and discuss means to accurately determine size, shape, and composition information about submerged elastic targets from the RESONANCE features present in their monostatic and/or bistatic scattering cross sections. Emphasis will be placed on the resonance scattering techniques (RST) developed for poles in the complex-frequency domain, and their Fourier-related, time-dependent waveforms. The approach is ideally amenable to implementation by expert (AI) systems.
SPREX: Speech recognition expert
Authors: Mizoguchi Riichiro , Tsujino Katsuhiko , Kakusho Osamu
Published: October 2005

Conventional speech recognition systems employ procedural languages such as FORTRAN to encode a large amount of heuristic knowledge, which makes the systems inflexible. SPREX (a SPeech Recognition EXpert) is implemented as a knowledge-based system that is superior to procedural ones in the use of heuristics. This approach is expected to contribute to the establishment of a new methodology of speech recognition research. SPREX has a knowledge base for feature parameter trajectory reading and the knowledge is encoded in the form of production rules. It has the following three advantages. (1) It can cope with the coarticulation effects. (2) It can be constructed by stepwise development. (3) It is easy to debug. In addition to the recognition system, SPREX also has a powerful environment for knowledge base construction, which consists of five subsystems; a speech database, rule database, rule translator, rule learning module, and interface. The environment helps developers in many respects by employing knowledge engineering techniques. The current version of SPREX is implemented on Symbolics 3620 in Lisp and OPS5e and attains a 95% segmentation rate and an 87% recognition rate for continuous speech 70 s long uttered by six male adults at a speech rate of 6 morae/s. [Work partly supported by a Grant-in-Aid for Scientific Research on Priority Areas.]
Comparison of spatial processing techniques for sonar array data
Authors: Wyllie David V. , Ferguson Brian G. , Speechley Garry C.
Published: October 2005

It is demonstrated that adaptive beamforming techniques are superior to conventional methods for the spatial processing of sonar array data. For purposes of comparison, both conventional and adaptive frequency-wavenumber spectra are presented for various sonar arrays. Adaptive processing is found to improve the spatial resolution performance of an array of acoustic sensors and to suppress the spatial leakage inherent in conventional processing.
The effect of target motion on bearing estimates from a beam interpolation algorithm
Authors: LaTourette Robert A. , Ng Lawrence C.
Published: October 2005

An initial estimate of a target's bearing from an acoustic array of preformed beams is determined by choosing the beam pointing direction with the largest detected energy. An attempt to improve on this initial bearing estimate is accomplished by interpolating the response of three beams centered about the peak response. This interpolation process is designed to yield an improved estimate of the target bearing between beam pointing directions. This paper is a study of the merits of this beam interpolation process in the presence of target motion.
Detection performance of normalizer for multiple signals subject to partially correlated fading with chi-squared statistics
Authors: Nuttall Albert H.
Published: October 2005

The false alarm and detection probabilities for a multipulse signal subject to partially correlated fading, in the presence of Gaussian noise of unknown level, are derived in closed form. The number K of signal pulses, as well as the number L of noise-only pulses used to estimate the noise background power level, are arbitrary. The power fading is characterized by a chi-squared distribution with 2m degrees of freedom and a normalized set of covariance coefficients {ρkt}, all of which can be selected arbitrarily, in order to match an experimental realization or an actual measured situation. The performance capability of this processor depends additionally on the received signal-to-noise ratio. This study covers the case of a nonconstant threshold; comparisons of this normalizer with earlier results (for L = ∞) enable a quantitative evaluation of the losses incurred by lack of knowledge of the noise level. The important capability of constant false alarm rate is achieved by this normalizer.
Prognosis and diagnosis using acoustics
Authors: Antonelli David , Fu Caroline
Published: October 2005

Present methods of prognostics and diagnostics on engines, especially aircraft engines, involve performance monitoring and wrap testing using built-in test (BIT) and built-in test equipment (BITE). Performance monitoring is used primarily for measuring liquid flow, temperatures, and current flow. Wrap testing is used primarily for testing electrical circuits. If the hypothesis holds that vibration and/or acoustical signals can be interpreted to detect changes in a mechanical operation, this will add another dimension to engine monitoring, diagnostics, and prognostics. Abnormalities in acoustic signals emitted by worn mechanical parts could be detected and warn the operator of imminent failure. This paper describes how the SPIRE research tool could be modified and integrated with AI techniques to interpret acoustic patterns to predict engine failure.
Inductive generation of phoneme recognition knowledge for a continuous speech recognition system SPREX
Authors: Tsujino Katsuhiko , Mizoguchi Riichiro , Kakusho Osamu
Published: October 2005

A continuous speech recognition expert system named SPREX has been constructed. It simulates the behavior of human experts who can recognize speech by inspecting the trajectories of feature parameters such as formant and power [R. Mizoguchi et al., Proc. ICASSP '86 2, 1221–1224 (1986)]. In order to construct a speaker-independent knowledge base, it is necessary to learn by concrete speech samples. ARIS (adaptive rule induction system) is developed as an inductive learning subsystem for SPREX. It has the following three characteristics that are also interesting from a knowledge of engineering point of view: (1) It can generate new attributes for separating the training samples efficiently; (2) it can cope with the noisy data problem by using heuristic generalization operators; and (3) it can deal with numerical attributes as well as symbolic ones. The latest version of a knowledge base generated by ARIS recognizes 82% of consonants and 95% of vowels. Some of these performances exceed that of the rules written by human experts. [Work partly supported by a Grant-in-Aid for Scientific Research on Priority Areas.]
Acoustic holography for wideband, arbitrarily shaped noise sources
Authors: Maynard J. D.
Published: October 2005

Acoustic holography has been shown to be a powerful technique for the study of sound radiation from complex structures. In its most rudimentary form, acoustic holography involves the measurement of a single frequency of sound throughout a spatial surface that corresponds to the level surface of a separable coordinate system. Most holography systems are limited to this basic form because of data acquisition and processing limitations. However, a high-speed microphone array data acquisition system with an on-line array processor and new data processing algorithms that permit acoustic holographic reconstructions of wideband noise sources with arbitrarily shaped surfaces has been developed. The data acquisition system is now described in detail in the literature. [Maynard et al., J. Acoust. Soc. Am. 78, 1395–1413 (1985); W. A. Veronesi and J. D. Maynard, J. Acoust. Soc. Am. 81, 1307–1322 (1987)]. In this paper, holography for odd-shaped surfaces, i.e., surfaces that do not coincide with level surfaces of separable coordinate systems, will be emphasized. Current algorithms involve finite element techniques and singular value decomposition. [Work supported by the Office of Naval Research.]
Current advances in cylindrical nearfield acoustical holography
Authors: Williams Earl G.
Published: October 2005

Probably no other technique available to the experimenter provides such an exhaustive exposition of the structural acoustics of fluid loaded vibrating cylinders in such a short time, as does broadband nearfield acoustical holography. In a single experiment consisting of the measurement of the pulsed pressure on a 2-D cylindrical contour, the complete 3-D acoustic fields outside the cylinder are obtained over a frequency range of 2 octaves. Recent results on high-mode density cylinders will be shown, including the validation of the broadband results by comparison of the holographic, radial velocity reconstruction with direct measurement from a surface mounted accelerometer. Because of the volume of data provided by this technique, computer graphics are used to help expose physical phenomena. A videotape displaying simultaneously the oscillation of the surface velocity and nearfield pressure fields resulting from a shaker driven cylinder, as functions of frequency, will be shown. Current work will be discussed in boundary element implementation of holography using the Helmholtz integral, for application to axisymmetric, capped cylinder structures.
Acoustical imaging and farfield radiation patterns in air and water using STSF
Authors: Clark Joseph A.
Published: October 2005

The spatial transformation of sound fields (STSF) technique involves a planar scanning across the source under investigation. From each scan position, the cross spectrum is measured to every one of a set of references and, further, the cross spectrum is measured between each pair of the references. From the measured cross spectra, any descriptor of the nearfield (pressure, particle velocity, intensity, etc.) can be investigated by means of nearfield acoustic holography, while the more distant field can be determined by application of Helmholtz's integral equation. The cross spectrum formulation implemented in the STSF technique has the advantage of putting no restrictions on the coherence of the sound field. Only a certain stationarity of the source during the measurement procedure has to be assumed. This paper describes the use of the STSF measurement method in an investigation of radiation from dipoles both in air and in water. The discussion will briefly review the principle of the STSF method and then will present pressure, velocity, and active, and reactive intensity plots calculated in the plane of the dipole sources from data measured in the acoustic nearfield. Farfield radiation patterns calculated from the same nearfield measurement data will also be shown and compared with sound-pressure levels measured directly in the farfield. Some factors influencing the accuracy of STSF measurements of farfield radiated noise will be identified.
Visualization of the aerial ultrasound field in the nearfield of a transducer
Authors: Otsuka Tetsuro , Shiode Takashi , Seya Koichiro
Published: October 2005

High-power aerial ultrasound sources having a stepped circular vibrating plate were designed for frequencies of 20 and 28 kHz. The diameter of the vibrating plate was about 5 λ (λ is the wavelength in air). The purpose of this sound source was to make a piston sound source having a diameter much larger than the wavelength so that the single beam directional pattern could be obtained on the center axis of the sound field. To investigate the characteristics of the sound source, the imaging of the ultrasound field radiated from the sound source was experimentally performed. The signal from a scanning microphone in the sound field was introduced to a lock-in amplifier and was compared with the reference signal. The output signals of the phase and the amplitude from the lock-in amplifier were linked to a microcomputer and a work station. The ultrasound field was visualized on a CRT display with 256 different colors. The characteristics of the propagating ultrasound from the stepped circular vibrating plate are discussed.
Scattering in a shallow water waveguide
Authors: Sammelmann Gary S. , Hackman Roger H.
Published: October 2005

The bistatic scattering from an elastic target in a shallow-water waveguide with a rigid bottom was discussed at a previous meeting of the society [G. S. Sammelmann and R. H. Hackman, J. Acoust. Soc. Am. Suppl. 1 82, S49 (1987)]. One emphasis of this work was the effect of modal dispersion and the strong elastic signature of the flexural resonances on the scattered waveform. Here, an extension of the model to realistic bottoms is given. In particular, the effects of bottom structure on the time domain structure of the scattered signal are discussed and the generation and propagation of interface waves and the scattering of such waves by a target in the waveguide are considered.
A modal/WKB inversion method for determining sound-speed profiles in the ocean and ocean bottom
Authors: Casey Kevin D. , Frisk George V.
Published: October 2005

Two approaches to determining sound-speed profiles in the ocean and ocean bottom using measured acoustic modal eigenvalues are examined. Both methods use measured eigenvalues and mode-dependent assumed values of the WKB phase integral as input data and use the WKB phase integral as a starting point for relating the index of refraction to depth. Inversion method 1 is restricted to monotonic or symmetric sound-speed profiles and requires a measurement of the sound speed at one depth to convert the index of refraction profile to a sound-speed profile. Inversion method 2 assumes that the sound speed at the ocean surface and the minimum sound speed in the profile are known and is applicable to monotonic profiles and to general single duct sound-speed profiles. For asymmetric profiles, inversion method 2 gives the depth difference between two points of equal sound speed in the portion of the profile having two turning points, and in the remainder of the profile it gives sound speed versus depth directly. A numerical implementation of the methods is demonstrated using idealized ocean sound-speed profiles. The two methods are used also to determine the sediment sound-speed profiles in two shallow water waveguide models, and inversion method 1 is used to find the sediment sound-speed profile using data from an experiment perfomed in the Gulf of Mexico. [Work supported by ONR.]
Effects of surface bubble layers on coastal acoustic tomography transmissions
Authors: Nakano Iwao , Lynch James F.
Published: October 2005

Recent work by Lynch, Miller, and Chu has shown that one can obtain surface wave frequency-directional spectra with low-frequency (∼225 Hz) acoustic tomography in coastal regions Lynch et al., J. Geophys. Res. 92, 6869–6885 (1987)] using the travel-time fluctuations of the acoustic multipaths as data. However, no account was taken in this work to estimate the effects on the acoustics of the near surface bubble layer produced by the surface waves. This bubble layer reduces the sound speed near the surface and produces two effects: First, a travel-time delay is produced, i.e., the bubble layer “biases” the travel time estimate; and second, the interaction angle of the ray with the surface is steepened. The dependence of both of these effects on wind speed and ray angle are studied for a simple waveguide example. A comparison of the magnitude of the bubble layer effects with the surface scattering effects is also made.
A neural network model of musical chord classification
Authors: Laden Bernice , Keefe Douglas H.
Published: October 2005

A neural network, which models a musical task, specifically the chord classification task, was built. The model “listens” to a chord and classifies it as a major, minor, or diminished chord. Two basic approaches were taken in the development of the model: (1) pitch class, and (2) harmonic template. The pitch class approach represents chords as they are customarily represented in music notation, namely by specifying the pitch class of each tone in the chord. The harmonic template approach defines each pitch in terms of an equivalent harmonic complex, specified by the most appropriate pitch class label for each of the first five harmonics. This approach is motivated by pitch perception theories based upon pattern matching (Goldstein, 1973; Terhardt, 1974). Connection strengths between nodes in the network were derived using an error propagation learning algorithm (Rumelhart, Hinton, and Williams, 1986). The pitch class network performed poorly by classifying only 33%–72% of the musical chords correctly. The harmonic template network classified 100% of the musical chords correctly. [Work partially supported by the University of Washington Graduate School Research Fund and ONR Grant N00014-86-C-0065.]
The self-excited oscillation of and the acoustic radiation from reed instruments
Authors: Vaidya P. G.
Published: October 2005

A harmonium reed undergoes a self-excited oscillation, when the static pressure on its two sides differs by a value, greater than a critical amount. The reed vibrates at its natural frequency, yet the sound is radiated at multiple frequencies. In this paper, a fluid mechanical model by St. Hilaire has been extended to show how the self-excitation begins, how the oscillation reaches a limit cycle, and how, due to the monopole coupling, radiation at multiple frequencies becomes possible.
In situ acoustic calibration for a large aperture array
Authors: Sotirin Barbara J.
Published: October 2005

During September 1987, a large aperture acoustic array was deployed vertically in the Northeast Pacific to study low-frequency noise in the ocean. Coherent combination of the 120-channel outputs requires knowledge of individual element amplitude and phase response for accurate results. Two in situ methods of array calibration are described and results from the September experiment are presented. The first method used transmissions from a low-frequency source of known location and power level. Simulating the conditions encountered during the transmission, the power arriving at the array was predicted by several acoustic propagation models. By comparing the array response at specific frequencies to the response predicted by the models, an absolute calibration was obtained. An error curve for the phase data was generated by unwrapping the phase, accounting for a sampling offset in the array, and subtracting a multiple linear regression curve. The second method determines relative amplitude levels by examining the average ambient noise power output of a specified frequency band across the array. Using spectral, coherence, and directionality plots, the level of self-noise in the array was shown to be below that of the ambient noise being measured. These two independent methods provide a consistent set of element calibration values used for array beamforming. [Work supported by ONT.]
Influences of the geometry of the French horn mouthpiece upon intonation
Authors: Plitnik George R. , Lawson Bruce A.
Published: October 2005

A mouthpiece assembly consisting of adjustable components having different dimensions, shapes, and tapers for the rim, cup, throat, and backbone was used in conjunction with a Lawson B-flat French horn for this experiment. By accurately measuring the input impedance for the mouthpiece/open French horn assembly by means of a dual-sensor apparatus [see A. M. Bruneau, J. Acoust. Soc. Am. 81, 1168–1178 (1987)], the effect upon the amplitude and frequency of the impedance maxima could be individually gauged as each mouthpiece parameter was varied. By correlating performers' comments and subjective evaluations of the different mouthpiece combinations, an attempt has been made to relate changes in resonant frequencies and spectral envelopes to the intonation differences perceived by the player.
Control criteria for active noise reduction systems
Authors: Tichy Jiri
Published: October 2005

Effective control of auxiliary sources for active noise or vibration reduction can only be achieved by adaptive systems. The inputs into adaptive filters have to provide information on the signal to be canceled, as well as on the error signal from a sensor located in the area of sound field minimization. Although, in principle, some systems could achieve nearly perfect cancellation, their actual performance is limited by feedback problems from auxiliary sources which can cause system instabilities and limit the overall system performance. This problem becomes more severe with multiple feedback control which is needed, for instance, in multipath vibration transmission reduction of mechanical systems supported in several points, or multimicrophone control in enclosures. An overall analysis and approach to solutions will be presented.
Applications for active noise control: What has been achieved and what can be achieved
Authors: Warnaka Glenn E.
Published: October 2005

Active noise control is currently undergoing an explosive growth throughout the world. This rapid development is occurring because the concept allows improvement in existing noise control devices, often with potential improvements in size, weight, volume, and cost. It also permits new solutions to noise control problems that were previously difficult or unsolvable. The technique is particularly attractive because, for many applications, noise may be reduced at the listener's position without physical modification of existing noise sources or their arrangement. Active noise control seems particularly well suited to the reduction of low-frequency noise which, for technological reasons, cannot easily or conveniently be attenuated with existing hardware. Because this new concept applies to such a wide variety of industrial, commercial, and military needs, a large number of specific applications are now under development or proposed for development. This paper describes the current status of active noise control technology. It describes applications that have already been realized as well as the possibilities for future development.
Consideration of active noise control in space
Authors: Okda Josuke , Usagawa Tsuyoshi , Ebata Masanao
Published: October 2005

Active noise control in space is classified into four groups based on physical aspects, which can be further reduced into two groups, namely, the sound pressure or the sound power, and the ability or nonability of an additional sound source to act on the sound radiation of an original source. The latter is equivalent to the problem of whether the mutual impedance between the two sound sources must be considered or not. The noise reduction process in each group is analyzed using the impedance of the sources. The desirable condition of the additional source is discussed, and, when this condition is realized, the noise reduction effect is estimated. In general, when the additional source is placed near the original one, the more tightly they are coupled, the more effectively both the sound pressure and the sound power are reduced. It is difficult, however, to find the most desirable condition for reducing the sound power by means of an experimental procedure.
The use of electronic noise cancellation in the mining industry
Authors: Marraccini Leonard C. , Giardino Dennis A.
Published: October 2005

In order to reduce the noise exposure of people working in the mining industry, various types of noise control applications have been used. These have included standard retrofit noise controls, machinery redesign, and administrative controls such as modifying employee work cycles. Currently, the Mine Safety and Health Administration is investigating the use of electronic noise cancellation techniques for reducing worker noise exposure. This paper characterizes this work including background information, a description of the laboratory development work involving electronic noise cancellation, and the application of this technology in a field situation. By using the electronic cancellation system and loudspeakers, a zone or area of noise cancellation is produced.
Equivalent-network representation of an infinite elastic plate subject to plane-wave excitation
Authors: Rudgers Anthony J.
Published: October 2005

A plane wave incident upon an elastic plate at an oblique angle excites both dilatational and shear waves in the plate. On reflection at the plate surfaces, shear waves with displacement components nonparallel to the plate surfaces are partially converted to dilatational waves, and dilatational waves are partially converted to such shear waves. The mechanical behavior of a plate in such circumstances can be modeled, not only by means of equations derived using the theory of linear elasticity, but also with a four-port network that is equivalent to these equations. This network comprises two coupled transmission lines, one representing the dilatational and the other representing the shear-wave propagation. A pair of ports at each plate surface transmits the normal and the transverse forces and displacements appearing at that surface. Each port is connected to the two transmission lines through a network of ideal transformers. A layered elastic media can be represented by a number of such equivalent networks connected in cascade. [Work supported by ONR.]
The k-space formulation of acoustic scattering using the conjugate-gradient method
Authors: Koshigoe Shozo , Bojarski Norbert N. , Tubis Arnold
Published: October 2005

An extension of the iterative k-space method [N. Bojarski, J. Acoust. Soc. Am. 72, 570–584 (1982)] is used to calculate acoustic scattering. The conjugate gradient method, with guaranteed convergence, is used instead of the Neumann series for the iterative calculations. In order to accelerate the computation and reduce the memory-size requirement, the convolution type integral involved in the use of the conjugate gradient method is handled as in the k-space method. One-dimensional acoustic scattering by an arbitrary slab is calculated as a simple illustrative application of the formalism. In order to demonstrate the utility of the method, some of the calculated results are compared with those obtained by the Neumann method.
Product expansion of the S function for scattering from elastic spheres having multiple resonances
Authors: Marston Philip L.
Published: October 2005

Associated with the scattering phase shift δn of the nth partial wave for a sphere of radius α is the function Sn (x) = exp[2iδn], where x = ka. A theorem from classical scattering theory [H. M. Nussenzveig, Causality and Dispersion Relations (Academic, New York, 1972), pp. 54–72; N. G. van Kampen, Phys. Rev. 89, 1072–1079 (1953)] leads to the following product expansion which appears to be important and novel for acoustics: Sn = ± e−2ix ∏ l(xnl* − x) (xnl + x)(xnl* + x) (xnl − x) ∏ j(iLnj − x)(iLnj + x), where the xnl ≡ Xnl − i(Γnl/2) lie in the fourth quadrant and Lnj > 0. Unlike a sum expansion (in some acoustic RST literature) Sn remains manifestly unitary even for multiple resonances l. This Sn may be used to split off elastic contributions to the partial-wave form function fn for backscattering in analogy with RST. In the case of only two resonances (labeled l = 1 and 2)fn≈fn(b) + fn1 + fn2 + fn(int), where fn(b) is a background term and it is assumed that x + Xnl ≫ Γnl. The fnl have a Breit-Wigner form fnl = exp(2iξn) [(2n + 1)/x]( −1)nΓnl/[Xnl − x − (i/2)Γnl] and ξn is the phase shift associated with fn(b). The interaction term fn(int) vanishes as Γnl/|x − Xnl| →0 for l = 1 or 2. This clarifies implicit assumptions of formal RST. [Work supported by ONR.]
The acoustic scattering by a submerged, elastic spherical shell—Pole trajectories in the complex k plane
Authors: Hackman Roger H. , Sammelmann Gary S.
Published: October 2005

At a previous meeting of the society [G. S. Sammelmann and R. H. Hackman, J. Acoust. Soc. Am. Suppl. 1 83, S94 (1988)], a fundamentally oriented analysis of the pole structure of the acoustic scattering amplitude for an elastic spherical shell was presented. In this presentation, it was demonstrated that fluid loading the first antisymmetric mode of a spherical shell in vacuum had a rather violent effect; the dispersion curve bifurcates near the frequency that the vacuum curve transitions from a subsonic to a supersonic phase velocity. Here, the precise relationship of the Franz wave poles of rigid scattering and the poles corresponding to the vacuum eigenvibrations to the acoustic scattering poles of the fluid-loaded spherical shell varying the elastic sound speeds of the shell are established. The fluid-loaded modes are found to “switch tails” under variation of the sound speeds and it is found that the unique identification of a given mode as either “diffractive” or “elastic” in character is not generally possible.
Solving the inverse scattering problem to obtain acoustic bubble spectra
Authors: Moritz Elan , Commander Kerry W.
Published: October 2005

Determination of acoustic bubble spectra near the surface of the ocean, using resonance scattering approximations, leads to considerable overestimation of the numbers of bubbles with radii less than 50 microns [J. Acoust. Soc. Am. Suppl. 1 82, S109 (1988)]. Several numerical methods for solving the exact inverse scattering problem were examined and results for each are presented. In these studies, attenuation and backscattering strengths, as a function of frequency, were used as input in each of the techniques, and the computer bubble distribution was compared with known (a priori) distributions. Sensitivity of the numerical techniques to dependence on the attenuation and backscattering functions was explored for each method as well. Improvements over resonance theory approximations are shown through a systematic study of each technique using known bubble distributions. [Work supported by ONT.]
Acoustic radiation force on a rigid sphere in the nearfield of a circular piston vibrator
Authors: Hasegawa Takahi , Inoue Naoki , Matsuzawa Kiichiro
Published: October 2005

The acoustic radiation force on a rigid sphere in the nearfield of a circular piston vibrator was investigated theoretically. The model considered is a rigid sphere suspended freely in an inviscid fluid, the center of the sphere being on the axis of a circular piston vibrator excited harmonically in an infinite baffle. The velocity potential in the nearfield was represented as an infinite series of the spherical harmonics by use of the diffraction integral for the vibrator. The result showed that the acoustic radiation force is given in the form of an infinite series as a function of the scattering coefficient and the diffraction integral. The nth diffraction integral fn is given in the form of a recursion formula as a function of the parameters of the vibrator and the distance between the sphere center and the vibrator. The formula obtained is valid for any elastic solid spheres. The present study will have useful applications in absolute measurements of ultrasound intensity in the nearfield at megahertz frequencies.
Overview of Japanese efforts toward an advanced man-machine interface through spoken language
Authors: Fujisaki Hiroya
Published: October 2005

An overview will be given of some recent efforts in Japan to promote and coordinate research activities toward the realization of an advanced man-machine interface through spoken language. In the field of speech processing, a large-scale national project was started in 1987 by a Grant-in-Aid for Scientific Research on Priority Areas from the Ministry of Education, Science and Culture (MESC) of Japan. It involves all the important academic groups as well as a number of experts from governmental organizations and private industries, and emphasizes the systematic coordination of research efforts through the use of common speech data, standardization of research tools, and shared research results. Another national project will start in 1989 to construct a database of the spoken dialects of Japanese. In the field of natural language processing, a separate project has also been carried out with the support of MESC. Efforts toward an integration of studies in the two areas, as well as toward closer cooperation between academic and industrial circles, have resulted in the establishment of a committee for academia—industry cooperation on the intelligent processing of linguistic information both in spoken and written forms under the auspices of the Japan Society for the Promotion of Science. Some highlights of these activities will be given in the lecture.
Speech processing in human-machine interaction: Research in the United Kingdom
Authors: Fallside Frank
Published: October 2005

Research into speech processing in human-machine interaction in the U.K. has been much stimulated by the Alvey initiative in advanced information processing. This will shortly be succeeded by the Information Engineering Directorate (IED) of the Department of Trade and Industry with support from other government bodies, including the Science and Engineering Research Council, together with support from the second phase of the European Economic Community's ESPRIT initiative. The paper describes the broad areas of research of these and other initiatives in the U.K. under three headings: (a) enabling research into speech recognition and synthesis, (b) systems, and (c) infrastructure. Work on speech recognition using the techniques of hidden Markov models, feature based methods, hybrid classifier, and connectionist models/neural networks is described, together with research into text to speech synthesis in several laboratories. The Alvey MAST system for speech transcription and VODIS system for database inquiry are described. A brief description is given of the work on speech assessment under the STAG and SAM projects. Finally, the IED strategy with its emphasis on multidisciplinary research is described.
Voice input output for the K.B.C.S. project of India
Authors: Rao P. V. S.
Published: October 2005

Financed by the Government of India in cooperation with the United Nations Development Program (UNDP), the Nodal Centre for Speech Input Output is concerned with the realization of a voice-oriented interactive computing environment (VOICE) under a knowledge-based computer system project. The VOICE system aims at speech input backed by provisions for (a) feedback by (i) oral repetition by the machine, (ii) visual display; and (b) correction by (i) repetition of the word, (ii) spelling by voice, (iii) typing in. Speech output of the machine, based on an earlier work [P. V. S. Rao and R. B. Thosar, IEEE Trans. ASSP ASSP-22, 217–225 (1974)] will be either (a) a well-formed sentence based on a restricted syntax, or (b) the speaker's utterance, recognized, resynthesized, and repeated for confirmation. For (a), predetermined intonational patterns are used. For (b), the speaker's intonational pattern is superimposed on the synthetic utterance. For recognition, conventional subword unit approaches, hidden Markov models, and rule-based direct phonetic feature analysis techniques are all being evaluated. It is hoped that the final system will in some sense integrate the best features of these.
Low-frequency reflectivity and transmissivity of gradual transition tank linings
Authors: Tanzosh John P. , Reader Wayne T.
Published: October 2005

Anechoic chambers and tanks that are intended to perform over a decade or greater frequency range typically employ a gradual transition lining consisting of pyramidal sections, wedgelike sections, or triangular sheets of porous or voided material. The purpose of these tapered pieces of absorptive material, of course, is to present a gradually changing impedance to the sound waves incident upon the wall, while simultaneously conserving absorptive material from that required for continuous layers. The reflectivity and transmissivity of a lining consisting of a periodic array of arbitrarily shaped geometric sections are computed assuming that the acoustic wavelength in the fluid surrounding the absorptive sections is much greater than the spacing of the sections. Then, the sections and fluid between them are characterized by an arbitrary number of layers with an effective density and compressibility. Examples will illustrate use of the analysis to determine the shape and height of pyramidal and triangular sections of a voided viscoelastic polymer required to achieve specified performance goals.
Chlorosulfinated polyethylene: A versatile polymer for acoustical applications
Authors: Reader Wayne T. , Deigan Richard J. , Megill Robert W.
Published: October 2005

Viscoelastic polymers are used extensively in the various fields of acoustics to reduce structureborne, airborne, and waterborne sound fields. Applications include their use as free or constrained layers adhered to vibrating plates to dissipate vibrational energy before it can generate noise in the surrounding field or can cause fatigue failure; as open cell foams applied to the walls of air chambers to absorb airborne sound before it can be reflected from the walls; and as closed cell foams that can be used to line the walls of water-filled tanks to render them nonreverberant and quiet. Each application requires that the elastic modulus and loss factor assume unique values within the operating frequency and temperature ranges. Chlorosulfinated polyethylenes, such as the HYPALON® family developed by E. I. Du Pont De Nemours and Co., possess the versatility needed to provide these unique properties and to be fabricated in the forms required for these acoustical applications. Discussed will be the range of elastic moduli, loss factors, and glass transition temperatures achievable by varying the HYPALON® polymer formulation, the cross-link density, and the amount and type of filler such as carbon black, oil, and type of blowing agent.