bluuebunny/crossref_metadata_2025 huggingface dataset:
Active control of higher-order acoustic modes in ducts
Authors: Zander Anthony C. , Hansen Colin H.
Published: October 2005

The active control of higher-order acoustic modes within a semi-infinite hard-walled rectangular duct is examined analytically for harmonic constant volume velocity sound sources. The effect of source location, size, strength, and relative phase upon the total mean acoustic power flow is investigated for both single and dual control source configurations, at a number of excitation frequencies. For the analysis, the primary source is mounted in the plane of the duct cross section, which may be a uniform surface of arbitrary finite impedance. It was found that the reduction in total acoustic power is dependent upon the relative location of the primary and control sources, both laterally across the wall of the duct and axially along the duct. The reduction in total acoustic power was found to be a maximum at axial source separation distances corresponding to multiples of half-wavelengths of the propagating modes, and at relative lateral locations such that the primary and control source mode shape function values were equivalent.
Dynamic behavior of structural-acoustic systems in feedforward control of sound radiation
Authors: Burdisso Ricardo A. , Fuller Chris R.
Published: October 2005

The dynamic characteristics of a beam under active control to minimize the coupled sound radiation is investigated. The beam is subjected to a harmonic point force and the resulting radiated field minimized by utilizing a piezoceramic actuator as control input. The error information from the acoustic medium is obtained by placing a single microphone in the acoustic far field. A feedforward control approach is used to derive the optimum control input applied to the actuator. The analysis shows that when the error microphone is placed along a line perpendicular to the radiating surface, the controlled structure has new eigenvalues and eigenfunctions. The controlled system eigenproperties are only a function of the control force location and independent of the disturbance input. A numerical example demonstrates the applicability of the formulation. The numerical results are also corroborated experimentally using a feedforward controller implemented on a TMS320C20 DSP processor.
Single-frequency ultrasonic measurement of kinetic constants and volume and compressibility effects of the proton-transfer reaction in aqueous solutions
Authors: Kharakoz Dmitry P.
Published: October 2005

A method for determining the acoustic relaxation characteristics (relaxation time, volume effects, and compressibility effects of ionization) in proton-transfer reactions is proposed. The basis is the simultaneous measurements of sound velocity, absorption, and degree of ionization at a single frequency. The single-frequency ultrasonic method applied in a systematic study is easier than the normally used wide-frequency ultrasonic method, however it is restricted to solutions with a known number of relaxation processes. Included is an illustration of the single-frequency method applied to an aqueous solution of monocarboxylic acid.
A derivation of the differential equations of coupled-mode propagation
Authors: Fawcett John A.
Published: October 2005

In this paper the differential equations of the coupled-mode theory of acoustic propagation are derived. The derivation properly accounts for the boundary conditions imposed at an interface and gives rise to terms not normally found in other coupled-mode formulations. The solutions of this couple-mode system agree very well in numerical experiments with those obtained from a boundary integral equation method.
Acoustic wave propagation in one-dimensional stratified gas–liquid media: The different regimes
Authors: Sornette Didier , Legrand Olivier
Published: October 2005

The problem of the propagation of acoustic waves in a one-dimensional bubbly liquid, either periodic or random is considered. This example of wave–scatterers interaction is complex enough to illustrate many of its general characteristics and yet simple enough so that an exact solution of the theoretical model can be found. Indeed, it is possible to compute exactly the response of the bubbly liquid to an incident wave in any range of concentrations, bubble size polydispersity and frequency range. In the near-field or high concentration regime, these solutions are particularly interesting since the 3-D problem cannot be solved in general. This analysis exhibits three main regimes, depending on the average value of the bubble radius R, average distance d between bubbles, and wavelength λ in the liquid. (1) For R≤λ and d≤λ, this is the ‘‘phonon’’ or ‘‘mass-spring’’ regime: the compressible bubbles act as springs connecting rigid incompressible fluid slabs that play the role of the masses in between the ‘‘springs’’. (2) For R≤λ≤d, bubbles present a kind of resonance, which is the 1-D analog of the standard 3-D bubble internal resonance. Furthermore, bubbles are coupled to each other by their radiations that propagate away in the compressible fluid: This is the ‘‘bubble’’ regime, which is the 1-D counterpart of the standard 3-D problem of an acoustic wave propagating in a diluted solution of small bubbles. (3) For λ≤R and λ≤d, the compressibility of the bubbles does not lead anymore to a resonance. This is the ‘‘stratified’’ regime, where the liquid and gas layers play symmetric roles. In the periodic systems, the dispersion relations and band structures are found. In the random systems, we compute the Anderson localization length, both numerically and analytically in the limit of uncorrelated disorder, for various system realizations.
Renormalization of propagation in a waveguide with rough boundaries
Authors: Berman David H.
Published: October 2005

A scheme is presented for computing the mean field in a waveguide with rough boundaries using nonperturbative half-space scattering amplitudes. It is shown that the presence of a second scattering surface renormalizes the mean half-space amplitudes even when surface roughness on the two bounding surfaces is uncorrelated. The method of ‘‘smoothing’’ is used to show how recent work on half-space scattering amplitudes might be incorporated into solutions of the wave equation in more realistic settings.
Measurements of acoustic backscatter from the Sohm Abyssal Plain
Authors: Hines Paul C. , Barry Patrick J.
Published: October 2005

Acoustic backscatter data were collected from the ocean bottom at four essentially smooth sites on the Sohm Abyssal Plain. The bottom sediment at the four sites varied from mud/clay to silt/gravel. Using a simple array consisting of a free flooding ring projector which was omnidirectional in azimuth, and an omnidirectional hydrophone, data were collected over the frequency range of 800–2400 Hz in 200-Hz bandwidths. By deploying the array to a depth approaching 5000 m, bottom backscatter data were obtained at grazing angles down to 4° before the onset of the first surface return. To obtain sufficient signal-to-noise ratio without impairing spatial resolution, linear FM pulses were employed. In this paper, the experiment and the bottom topography of the four sites are described, and the backscatter data are presented.
Reflection loss at a bottom with a fluid sediment layer over a hard solid half-space
Authors: Hovem Jens M. , Kristensen Åge
Published: October 2005

Numerical modeling and experiments have shown that for acoustic propagation in shallow water where the bottom has a layer of relatively soft sediment over a hard rock subbottom, high losses may occur for certain frequencies and angles. These high losses have been attributed to shear waves in the sediment, but this paper demonstrates that the high losses also can occur without shear waves in the sedimentary layer and can be attributed to the excitation of interface waves of the Stoneley type at the fluid/solid interface between the sediment and the substrate. These waves are generated by the evanescent waves in the sedimentary layer for incoming plane waves in the water at grazing angles smaller than critical. The role of the sediment layer is to produce the inhomogeneous wave that is required to excite the interface wave. From an analysis of the dispersion equations, one can predict the necessary conditions for the high losses to occur.
Investigations of underwater acoustic transducers using TV holography, acoustic phase shifting, and digital image processing
Authors: Ellingsrud S. , Nilssen Å. Bergh
Published: October 2005

A measuring system, based on TV holography and image processing, that is used for full field mapping of surface displacements of underwater acoustic transducers is described. In the technique’s normal operation mode, the time-averaged recording, a realistic amplitude detection range in water is from about λ/5 to 2.5λ, where λ is the wavelength of the applied laser light. Combined with sinusoidal phase modulation and discrete vibration phase shifts, the system provides numerical phase and amplitude values of sinusoidally vibrating transducers. This combined technique has an amplitude detection range in water from about λ/1000 to λ/10. Results are presented of single-element and array transducers in active and passive mode, both as ordinary time-averaged fringe patterns and as numerical phase and amplitude maps. The transducers vibrated in a frequency range of 20–250 kHz.
Standing wave analysis of a thermoacoustic prime mover below onset of self-oscillation
Authors: Atchley Anthony A.
Published: October 2005

A standing wave analysis of a thermoacoustic prime mover is presented. The analysis, motivated by Swift [G. W. Swift, ‘‘Thermoacoustic engines,’’ J. Acoust. Soc. Am. 84, 1145–1180 (1988)], assumes that the acoustic pressure and velocity can be described in terms of a standing wave throughout the prime mover. Because the analysis is not limited by either the small boundary layer or short stack approximations, it is more general than Swift’s short stack, boundary layer approach. However, it is not as general as a fully numerical approach. The results of the analysis compare well to measurements of the quality factor of the first three longitudinal modes of a helium-filled prime mover below onset of self-oscillation. An analysis is presented of the sensitivity of the calculations to experimental errors and to various assumptions regarding the temperature distribution in the prime mover stack. It is concluded that, in most cases, using reasonable, simplifying assumptions regarding the temperature distribution produces acceptable results.
Directional flextensional transducer
Authors: Butler Stephen C. , Butler Alexander L. , Butler John L.
Published: October 2005

A 3.25-kHz directional class IV flextensional transducer, which produces enhanced motion on one surface and canceled motion on the second surface, has been developed. The transducer radiating surface consists of a thick-walled oval aluminum shell. A Navy type III piezoelectric ceramic stack with two active sections and one inactive section is used to drive the shell along its major axis into its directional mode of operation. The directionality is achieved by exciting the drive stack into both extensional and bending modes. Both measurements and a finite element analysis on this device showed the center portion of one side to be moving at a relative displacement level more than 20 dB greater than the corresponding center portion of the other side.
Evidence for two discrete sources of 2f1−f2 distortion-product otoacoustic emission in rabbit. II: Differential physiological vulnerability
Authors: Whitehead M. L. , Lonsbury-Martin B. L. , Martin G. K.
Published: October 2005

In a previous report, it was shown that, in normal rabbit ears, the amplitude and phase of 2f1−f2 distortion-product otoacoustic emissions (DPOAEs) elicited by low-level (<60–70 dB SPL) stimuli display a differential dependence on stimulus parameters to those evoked by high-level (>60–70 dB SPL) stimuli, indicating differences in the underlying generation mechanisms. In the present study, the physiological vulnerability of DPOAEs in each of the two 2f1−f2 DPOAE-response regions identified on the basis of differential parametric properties, was characterized. Thus emissions evoked using stimulus levels from 45–75 dB SPL were measured over time upon: (1) induction of lethal anoxia, (2) acute injection of ethacrynic acid, and (3) acute injection of ethacrynic acid 2 h after a single administration of gentamicin. The DPOAEs evoked by low-level stimuli (45 dB SPL) were abolished within 3–4 min of induction of anoxia, whereas DPOAEs evoked by high-level stimuli (75 dB SPL) were unchanged in this period. The high-level emissions decreased with a complex time course post-mortem, and demonstrated behaviors, including evidence of susceptibility to fatigue, suggesting a dependence upon a cochlear energy supply. Low-level DPOAEs could be temporarily abolished, with complete recovery, by an acute administration of ethacrynic acid that had little effect on high-level DPOAEs. Treatment with the gentamicin and ethacrynic-acid combination, which would be expected to produce widespread hair-cell damage, eliminated low-level DPOAEs, and greatly reduced high-level emissions. In combination with previously published data, these findings strongly suggest that low- and high-level 2f1−f2 DPOAEs arise from discrete sources. The data are consistent with the proposal that the low-level DPOAE source is an active, micromechanical process, but suggest that the proposed origin of high-level DPOAEs exclusively in the passive macromechanics of the cochlear partition may be incorrect. The elimination of both low- and high-level DPOAEs revealed the presence of a third, residual 2f1−f2 DPOAE component, approximately 75–80 dB below the stimulus-tone levels, that may reflect the true passive-distortion response of the cochlea.
A combined finite element method/boundary element method technique for V(z) curves of anisotropic-layer/substrate configurations
Authors: Liu G. R. , Achenbach J. D. , Kim J. O. , Li Z. L.
Published: October 2005

A numerical technique is presented to calculate V(z) curves for a line-focus acoustic microscope, and for the specimen configuration of an anisotropic layer deposited on an isotropic elastic substrate. In this technique, the finite element method (FEM) and the boundary element method (BEM) are combined to calculate the scattered wave field on the specimen surface. The FEM is used in the layer and the BEM is used in the coupling fluid and the substrate. By combining the FEM with the BEM, the shortcomings of the two methods are avoided and their advantages are maintained. As a result, the present technique can be used for very thin and anisotropic layers. The V(z) curves are obtained by substituting the calculated wave field in an electromechanical reciprocity relation. The attenuation of the water used as the coupling fluid is taken into account by adjusting the calculated V(z) curves. The adjusted curves are compared with measured data, and very good agreement, both in the peak spacing and the amplitudes of the oscillatory curves, is observed.
Heat transfer effect on sound propagation in foam
Authors: Goldfarb I. I. , Schreiber I. R. , Vafina F. I.
Published: October 2005

The influence of heat transfer on the propagation of acoustic disturbances in a gas–liquid foam of polyhedral structure is considered in this paper. The presence of quasiordered structure of microcapillaries (Plateau–Gibbs channels) results in the appearance of a hydrodynamic effect caused by liquid motion along the channels and due to the competition with the contribution of heat transfer in general wave dissipation. The interphase heat transfer process in a gas-filled foam is considered within the framework of a cell model. Liquid motion through the proposed channels depends on the radial dynamics of the foam cell and on the hydroconductivity of the foam defining a free liquid motion through the channels. The Rayleigh equation analog, which takes into account liquid motion through the channels of the foam and small moisture content, is obtained. The dispersion relation shows that the influence of surface tension can be ignored. The effects of added liquid mass and the viscous interphase mechanism in the frame of applicability of the suggested model are analyzed. It is concluded that the general influence on the evolution of the initial perturbation has been exerted by the heat transfer process between the carrying and dispersion phases and liquid motion through the channels. The attenuation decrement consists of two parts: thermal and viscous. The intensity is defined by the foam dispersion a0 (size of the foam cell). Where a0<a*0 the signal dissipation, to a marked degree, is caused by liquid motion, and it is possible to neglect heat transfer.

When the thermal processes define the attenuation, the liquid motion can be ignored. The formula for the critical dispersion a*0 is obtained. For the water–air foam with volumetric water content α10=10−2, the critical dispersion is a*0=5×10−4 m. It is shown that the frequency range of validity of the suggested model can be divided into the two subranges. A brief review of the published experimental data on research of acoustic disturbances propagating in gas–liquid foam is presented. The experimental data analysis shows that the calculation of only heat exchange processes is insufficient for their interpretation. The comparison of experimental curves (obtained by Z. M. Orenbakh and G. A. Shushkov at the Institute of Northern Development) and calculated profiles for different parameters of gas–liquid foam is presented. It is shown that the calculation of liquid motion in the frames of the suggested model permits a more accurate description of the experimental results.
Nonlinear inversion for ocean-bottom properties
Authors: Collins Michael D. , Kuperman W. A. , Schmidt Henrik
Published: October 2005

High-resolution methods based on simulated annealing and full-wave sound propagation models are developed for nonlinear inversion for ocean-bottom properties. Simulated annealing is used to search the high-dimensional parameter space of ocean bottoms for the parameter set corresponding to the best replica field. The parabolic equation method is used to solve range-dependent inversion problems. For data taken by Lynch et al. from a range-dependent region of the Gulf of Mexico [J. Acoust. Soc. Am. 89, 648–665 (1991)], this approach achieves excellent agreement between the theoretical and measured acoustic pressures. The recovered sediment parameters suggest that a sound-speed boundary layer exists in the upper part of the sediment and that the depth of an interface in the sediment is range dependent. For locally range-independent problems, inversion is performed in wave-number space. Large efficiency gains are possible with this approach because the number of wave-number samples required for inversion is much smaller than the number of wave-number samples required for computing replica fields.
Low-frequency single interaction acoustic scatter from the sea surface using short pulses
Authors: Chester John B.
Published: October 2005

During February/March 1992 a comprehensive low-frequency (<1000 Hz) and low-grazing angle (<30 deg) acoustic sea surface scatter and air/sea interaction experiment was conducted in the Gulf of Alaska. For comparison with acoustic models, measurements were made of air/sea boundary descriptions such as near-surface bubble characteristics, wind speed, wave height, whitecap coverage, volume scatterers (fish), wind stress, etc. A variety of waveforms of short duration (<500 ms) were scattered from a surface path simultaneously isonified by three collinear vertical towed source arrays at frequencies around 250, 400, and 1000 Hz. Surface-scattered energy was received on a horizontal line array towed by the source ship. During the measurement wind speeds from 4 to 19 m/s and rms wave heights from 0.9 to 2.3 m occurred associated with sea states 2 to 6. Surface grazing angles of 4 to 32 deg and horizontal bistatic angles of 15 to 82 deg were observed. Scattering strength analysis includes frequency, grazing angle, waveform, and wind speed dependence as well as correlation with air/sea boundary descriptors. Current scattering strength model predictions utilizing the appropriate measured environmental parameters will be compared to measured data.
Low-frequency distant surface/near-surface reverberation measurements in the Gulf of Alaska
Authors: Gauss Roger C. , Soukup Raymond J. , Hayek C. Scott
Published: October 2005

Distant reverberation measurements in the surface/near-surface and air–sea interaction experiment in the Gulf of Alaska (February/March 1992) were designed to assess the Doppler and spatial characters of primarily low grazing angle reverberation as functions of frequency environmental conditions, and source steering angle. Twenty-two tests using cw and HFM pulses of 8- to 12-s duration were conducted during a variety of sea conditions, at wind speeds ranging from 4 to 19 m/s. These data were collected simultaneously at two frequencies (usually 250 and 1000 Hz). The data illustrate the contrast in the spectral and temporal characteristics of long-range reverberation (20 to 30 km) between low sea states where fish scattering is important and higher sea states where subsurface bubbles are important.
The analysis of noises in a fiber optic microphone
Authors: He Gang , Cuomo Frank W.
Published: October 2005

The general formulations to evaluate the pressure detection limit determined by various noise sources within the fiber optic microphone are developed including the shot and thermal noises of the opto-electronic detector. The formulations to calculate the optical power level of the thermal radiation from the sensing membrane and optical fibers, which will degrade the microphone performance by increasing the detector shot noise, particularly at high temperatures, are developed as a function of the temperature along the optical fibers as well as the microphone parameters. These factors determine the limit that the optical power output from the optical fiber microphone must exceed in order to perform pressure detection properly within the intended temperature range. The eventual pressure detection limit is evaluated based on the thermally generated membrane random vibration while the conditions to measure this noise effect are also established. The analysis provides a general approach to design the fiber optic microphones for a particular application and for a wide temperature range. Calculations illustrating our approach based on a typical fiber optic microphone design are also included.
Laser generation of narrow-band surface waves
Authors: Huang Jin , Krishnaswamy Sridhar , Achenbach Jan D.
Published: October 2005

Tone-burst-like narrow-band surface waves are generated in the thermoelastic regime by illuminating the surface of a solid with an array of laser-generated line sources. The laser line array is formed by a system of lenses and an optical diffraction grating that provide for flexible and easy control of the line-array parameters. It is shown experimentally, consistent with theory, that the generation of narrow-band surface waves can be controlled by adjusting the line-array parameters such as the number of line sources in the array, the width of each line source, and the separation distance between them. Certain optimum generation conditions are experimentally determined whereby the amplitude of the narrow-band surface waves can be increased by a factor of N (the effective number of lines in the array) over the corresponding broadband signals that would be generated using only a single line source. A laser-interferometric system is used to detect the generated surface waves. By suitably bandpass filtering the detected signal, the greater part of the white noise in the system can be eliminated, leading to a significant improvement in the signal-to-noise ratio and hence the sensitivity of the whole laser ultrasonic system.
Time normalization in voice analysis
Authors: Qi Yingyong
Published: October 2005

The harmonics-to-noise ratio (HNR) has been widely accepted for quantifying the irregular or noise component of voice. HNR, however, is usually inflated by cycle-to-cycle variations of fundamental frequency period because zero padding is used for time normalization of the wavelet. In this study, a new method was developed for analyzing waveform perturbations of voice. In this method, noise components of voice were calculated from the discrepancies between wavelets after they had been optimally aligned in time. The optimal time normalization of wavelets was accomplished using procedures of dynamic time warping (DTW). This method was evaluated using both synthetic and natural voices, and significant reductions in noise were obtained. The harmonics-to-noise ratio obtained using DTW for time normalization was also shown to be independent of fundamental frequency perturbations.
Large eddy simulation of aerosound using homogeneous turbulence as example
Authors: Lee Seungbae , Meecham William C.
Published: October 2005

Acoustic radiation from turbulent fluids, aeroacoustics, has been a major research area especially since the pioneering work of Lighthill. The fundamental mechanism of acoustic radiation from homogeneous turbulence has been discussed, using a similarity hypothesis [W. C. Meecham and G. W. Ford, J. Acoust. Soc. Am. 30, 318 (1958)]. This paper presents a new application of large eddy simulation (LES) techniques to the calculation of the acoustic far field produced by turbulent fluids. The acoustic radiation problem consists of two steps; in the first step one simulates homogeneous turbulent fields using LES; and in the second step one generates the far-field sound using Lighthill’s analogy. It is found that the radiated power spectrum is proportional to ω4M3, where M is the Mach number, at high frequencies as predicted in the early work. Other features of the sound field support the early theory. Similar calculations for jet noise will be discussed. Color computer graphics will be shown.
Designing synchronous chaotic systems
Authors: Vaidya P. G. , He Rong
Published: October 2005

Consider two identical chaotic system that are started from virtually identical initial conditions. In a short time, they would be observed to diverge from one another. While it is true that, given ample passage of time, the systems will come within an arbitrarily close distance of one another, they would drift apart once again. In their fascinating paper, Pecora and Carroll have recently described a system of two Lorenz oscillators. They could be metaphorically described as a master and a slave system. The master system, undergoing chaos, drives a part of the slave system. Incredibly enough, the two systems remain in perfect synchronization. Further analysis of this system and its generalization will be presented. Based on the generalization, a method to design chaotically synchronous systems will be presented. Practical applications of this phenomenon will be discussed.
Speaker-independent connected digit recognition algorithm for a single chip signal processor
Authors: Glinski Stephen C.
Published: October 2005

An algorithm was designed for a fixed point arithmetic signal processor chip to perform real-time speaker-independent English digit recognition. Each word was represented by a single 10-state Markov model, the states of which were 9-way mixtures of 24-dimensional Gaussian densities of cepstral features. Algorithms for feature extraction include autocorrelation, linear predictive coding, and the computation of both cepstra and differential cepstra. Algorithms for pattern matching include a Laplacian distance measure, viterbi decoder, best choice, and partial traceback. To achieve real-time operation, single precision arithmetic was employed for the Laplacian distance metric, which is the bottleneck in the recognizer. Memory storage was minimized by quantizing model parameters to 10 bits and dynamically pruning a tree of word candidates. Recognition accuracy of about 98% per word was obtained; this is approximately the same as that obtained with a floating point simulator as tested on a connected digits NIST database.
Speaker-independent sound recognition
Authors: Hanna S. A. , Stuart Laubstein Ann
Published: October 2005

The speech recognition machine must be able to (i) recognize intended types by filtering out variability in tokens and (ii) segment a continuous speech stream. Given the infinite (at least in principle) set of possible phrases and sentences, the segments must be smaller units than these such as words (practical only when dealing with a limited vocabulary), syllables, or phonemes. Phoneme-based recognition systems such as the one proposed here divide the speech signal into a string of phoneme-like units, which are subsequently used to recognize large word vocabularies composed of these smaller units. The broad category classification of sound elements developed here is intended as an initial stage in a two-phase phoneme-based speech recognition system. The role of the first phase is to automatically break down the continuous signal into a string of speaker-independent broad sound classes. This is done in two steps. First, the continuous speech signal is segmented and then these segments are assigned to one of the following acoustic categories: vowel-like, voiced-fricative-like, unvoiced-fricative-like, voiced-stop-like, unvoiced-stop-like, and silence. This classification algorithm has been applied to speech sentences uttered by five different speakers. Informal subjective tests indicate that classification accuracy ranges from 90% to 94%.
Dynamic cepstral parameter incorporating time-frequency masking and its application to speech recognition
Authors: Aikawa Kiyoaki , Kawahara Hideki , Tohkura Yoh’ichi
Published: October 2005

A ‘‘dynamic cepstrum’’ is proposed as a new spectral parameter that outperforms conventional cepstrum in speech recognition. The dynamic cepstrum incorporates both the static and dynamic aspects of speech spectral sequences by implementing forward masking, which is one of the most important mechanisms for extracting the spectral dynamics that provide acoustic cues in speech perception. Recent research on auditory perception [E. Miyasaka, J. Acoust. Soc. Jpn. 39, 614–623 (1983) (in Japanese)] reports that forward masking becomes more widespread over the frequency axis as the masker-signal interval increases. This masking characteristic facilitates the novel filtering methodology of time-dependent spectral filtering. The new dynamic cepstrum spectral parameter can simulate this function. The parameter is obtained by subtracting the masking level from the current cepstrum. The masking level at the current time is calculated as the sum of the masking levels obtained by filtering the preceding spectral sequence, where the cut-off frequency of the low-pass filter shifts lower as a function of masker-signal interval. A /b,d,g,m,n,N/recognition experiment that applies the dynamic cepstrum to hidden Markov models demonstrates that the dynamic cepstrum outperforms the combination of cepstrum and delta cepstrum, where the combination requires twice as many parameters as the dynamic cepstrum.
Very large vocabulary lexical access in real time from continuous phonemes
Authors: Mattei Michael D. , Graham James H. , Desoky Ahmed H.
Published: October 2005

Very large dictionary (over 5000 words) real-time lexical access for continuous speech recognition continues to be a difficult problem for speech researchers. One search technique that seems to overcome many of the limitations of hidden Markov models and beam search is based on inverted file database management. This inverted file search technique is able to operate in real time on a microcomputer with less than 8 MIPS processing speed using dictionaries well over 5000 words. This paper presents an overview of the technique along with specific performance statistics. The results are obtained using a 33 000-word dictionary on a 386 microcomputer. The inputs to the process are 100-word informal speech passages of continuous phonemes with no syllable or word boundary information. Although a simple search termination heuristic is employed, reasonably accurate word identification results are obtained with no post processing grammatical analysis.
Developing jazz improvisational skills
Authors: Thurmond Dennis
Published: October 2005

The use of music sequencing software to develop idiomatic keyboard improvisational skills based on the scalar motives of Keith Jarrett will be discussed. The presentation will include different models of real-time performance practices using a keyboard controller with sequenced bass and rhythm tracks. With the advent of sophisticated music sequencing software it is now possible to develop applications to aid the developing jazz musician acquire specific improvisational skills within the sonic framework of a specific improvisor (in this case, Keith Jarrett). Notational examples will include Mr. Jarrett’s work from the recordings: KolnConcert and ForestFlower.
Description of the sound scattering and radiation from compact but complex bubble distributions using multipole expansions
Authors: Carey W. M.
Published: October 2005

Dynamic bubble distributions in liquids and their acoustic properties are important in a large variety of applications ranging from the measurement of nitrogen bubbles in blood, fish schools, to clouds from breaking waves. Unlike individual bubbles, these distributions maintain their compactness by viscous and hydrodynamic vorticity. The specific case of compact distributions that have complex shapes may be treated by the use of multipole expansions. This paper discusses the application of multipole expansions to complex shapes in general and compares the first-order radiation and scattering fields from simple geometrical shapes, spheres, cylinders, and ellipsoids. The paper shows that when the regions are acoustically compact the lowest-order radiation from these features can be described by a modified Minneart formula describing the net volume fluctuation. In the case of noncompact distributions scattering regimes are shown to be delineated by several nondimensional numbers.
Coupling finite element and boundary element methods for the analysis of the acoustic scattering by elastic structures
Authors: Lavie Antoine , Dubus Bertrand
Published: October 2005

Considerable interest has been expressed recently in the scattering of a plane acoustic wave by an elastic structure. In this paper, a method that couples the finite element method and a boundary element method is proposed. A finite element code (ATILA) is used to model the elastic structure and a boundary element code (EQI) to describe the propagating waves in the infinite fluid medium surrounding the body. The analysis is performed in three steps for each frequency: (i) the effect of the fluid is characterized by a nodal impedance matrix using BEM, (ii) the analysis of the structure using FEM including fluid effects via the impedance matrix provides the displacement field, (iii) from these displacements, the near-field and far-field pressures are obtained using BEM. The method is successfully applied to the scattering by spherical shells of various thicknesses for ka varying from 0 to 15 (k is the acoustic wave number in the fluid and a is the external radius of the sphere). The advantages of the method (analysis of complex geometries with various materials, direct knowledge of the displacement field, exact description of Sommerfeld’s radiation condition) and its drawbacks (irregular frequencies associated with BEM, limitation to medium values of ka) are discussed. [Work supported by D.R.E.T. Paris.]
Statistical analysis on low-frequency sea bottom reverberation measurement data
Authors: Zhang Li , Goodman Ralph R.
Published: October 2005

Six regression formulas are used for fitting the measured sea-bottom scattering strength SS, which are regarded as functions of grazing angle θ and frequency f. The statistical analysis results are based on the measurement data from the published papers and technical reports since 1960. The data measured in the abyssal plain areas are closer to the Lambert’s law (SS=A+20 log sin θ) than those measured in the abyssal hills or ridge regions. But the formula SS=A+B log tan θ [A. W. Burstein and J. J. Keane, J. Acoust. Soc. Am. 36, 1596–1597 (1964)] is better than the Lambert’s law in all three regions. High correlation coefficients of the regressions can only be found if all the data are from same physiographic province. Otherwise the correlation coefficients are usually below 0.7, which indicate the current regression formulas are inadequate to describe the seafloor reverberation. In the 0.1- to 5-kHz band, the frequency dependence of scattering strength (SS vs log f) is very poor.
Reverberation modeling with the split step PE using back propagation
Authors: Schneider Hans G.
Published: October 2005

A method has been reported earlier to compute reverberation from ocean boundaries with the split step PE [J. Acoust. Soc. Am. 90, 2257 (1991)]. The formulation is based on the marching solution technique to solve the parabolic wave equation and uses generic backscatter functions for reverberation from the sea surface and/or ocean bottom. Two different schemes have been outlined. The first one, discussed earlier, uses direct reciprocity of propagation. This has the disadvantage that interference phenomena occurring during the forward propagation are also applied to the backscattered field that is unrealistic for incoherently scattered energy. The second approach, discussed here, avoids this by establishing the incoherent scattered field near the boundaries and propagating it backwards in range. In addition different source and receiver characteristics may be used easily and reverberation from bistatic geometries in the plane of propagation can be evaluated with one back-propagation run.
A model interference method for shallow water reverberation suppression
Authors: Williams Robert Bruce
Published: October 2005

A method of boundary reverberation reduction for shallow water ducts has been devised that makes use of a vertical array of sound sources. By controlling the waveforms of the sources, control over the phase and amplitudes of a selected set of modes can be made. This in turn allows formation of nulls in the field at specified range/depth pairs. At a given range, the nulls are chosen to be on the bottom of the duct where, for low and mid frequencies, most of the scattering is generated. However, at that same range, maxima in the field at other depths are present, thus enhancing the echo-to-reverberation ratio for reflectors at certain depths in the water column. Presented is the formalism in terms of mode theory, and field calculations for several environments. [Work supported by ONT.]
Acoustical measurements in lecture halls, theaters, and multi-use rooms
Authors: Siebein Gary W. , Chiang Wei-hwa , Cervone Richard P. , Doddington Harold W. , Schwab Wilhelm K.
Published: October 2005

A series of monaural and binaural acoustical measurements were taken at several source locations and multiple receiver locations in lecture halls, theaters, and multi-use rooms using the acoustical research instrumentation for architectural spaces (ARIAS) system. The rooms studied included a fire stair, small class rooms, lecture halls, theaters, and large, multi-use auditoria among others. Measurements of reverberation time, early decay time, early to late temporal energy ratios at various time periods, center time, lateral energy fractions, loudness, and interaural cross correlations were made. Reflectograms for locations within each room were also made. Summary graphs showing ranges of each measurement within each of the rooms and discussion of similarities and differences among the groups of rooms will be presented. [Work supported by the National Science Foundation.]
An acoustic measurement program for evaluating auditoriums and related spaces based on the early/late sound energy ratio
Authors: Marshall L. Gerald
Published: October 2005

Sound energy ratios in one form or another have long been used to quantify and predict clarity. Recognizing the critical importance of early reflections in the sound-decay process to room acoustic character, curves showing early/late ratios in the 20- to 200-ms period may be used as a way of examining the early reflection period, besides providing the established clarity ratings of C50 and C80. An automated measurement program producing octave-spaced energy-time curves and displays of the associated early/late sound energy ratios in the period between 20 and 200 ms will be discussed.
SYSid−audio band measurement and analysis system
Authors: Puria Sunil , Allen Jont B.
Published: October 2005

SYSid stands for SYStem IDentification. The software package runs on a DOS platform with the Ariel DSP-16+ installed. The basic concept of SYSid operation is simple. SYSid generates a stimulus (i.e., chirp, tones, user-defined, etc.) and synchronously averages the measured response. It is important to synchronously average a system response in order to obtain accurate phase information. SYSid then uses FFT techniques to deconvolve the stimulus from the measured response and to further analyze the data. From this basic mode of operation SYSid can perform many types of analyses including phase responses, group delay, impulse response, Hilbert envelope, reverse energy time curve, RT60, waterfall displays, electrical impedance, etc. In addition to these linear measurements, SYSid also provides the capabilities to make distortion measurements due to nonlinearities in the system. These include harmonic distortion, intermodulation distortion, THD+N, and input–output functions. With the current incarnation of SYSid each channel can be up to 32k sample points. Each channel is 16 bits and can be sampled at a maximum rate of 50 kHz. Examples of measurements made in areas of room acoustics, transducer design, and auditory physiology will be presented.
An acoustical study of a large rotunda
Authors: Moritz C. , Bruce R. D.
Published: October 2005

A large rotunda with a dome ceiling was built as part of a large shopping mall. It was envisioned that this area of the mall would be used for concerts and public presentations. Musicians such as Ray Charles are scheduled to perform for the grand opening in August. However, as with many acoustically hard dome structures, problems arose with echoes. Reverberation times of 18 s at 500 Hz were measured in the center of the dome. An acoustical treatment was installed on part of the ceiling to reduce the echoes and reverberation time. Impulse response measurements and results of the treatments are discussed.
Influence of sound absorbing material type and its location on the sound transmission through a double panel wall specimen
Authors: Warnock A. C. C. , Quirt J. D. , Halliwell R. E. , Birta J. A.
Published: October 2005

Sound absorbing material placed in the cavity of a hollow wall reduces the transmission of sound through the wall. The influence of density of commonly used materials, such as glass fiber, has been the subject of some argument. Also, it is sometimes thought that the position of the sound absorbing material in the cavity is important. The work to be presented is part of a large series of measurements studying sound transmission through double panel walls. Several types of glass fiber and mineral wool were placed in the cavity formed between two sheets of 3-mm-thick Lexan. Densities ranged from 10 to 145 kg/m3 and airflow resistivity ranged from 5 to 50 krayl/m. For the materials studied, the influence of material type on sound transmission class (STC) was small: only one or two points. The influence on transmission loss varied with frequency. Largest effects on transmission loss (changes of about 8 dB) were seen at the frequencies around 1000 Hz. The thickness and position of the materials were varied. Highest transmission loss was obtained when the sound absorbing material covered the entire inner surface area of the specimen. The same volume of material filling the cavity width but only partially covering the inner surface area, filling the lower half for example, gave lower transmission loss values.
Effects of first formant onset frequency on voicing judgments result from auditory processes not specific to humans
Authors: Lotto Andrew J. , Kluender Keith R.
Published: October 2005

When Fl onset frequency is lower, a longer F1 cutback (VOT) is required for human listeners to perceive synthesized stop consonants as voiceless. K. R. Kluender [J. Acoust. Soc. Am. 90, 83–96 (1991)] found comparable effects of F1 onset frequency on the ‘‘labeling’’ of stop consonants by Japanese quail (Coturnixcoturnixjaponica) trained to distinguish stop consonants varying in F1 cutback. In that study, CV’s were synthesized with natural-like rising F1 transitions. Consequently, endpoint training stimuli (with 5- or 65-ms F1 cutback) differed in the onset frequency of F1 because a longer cutback resulted in a higher F1 onset. In order to assess whether earlier results were due to auditory predispositions or due to animals learning the covariance between F1 cutback and F1 onset frequency, the present experiment was conducted with synthetic continua having either a relatively low (375-Hz) or high (750-Hz) constant-frequency F1. Eight birds were trained to respond differentially to endpoint stimuli from three series of synthesized /Ca/’s varying in duration of F1 cutback. Second and third format transitions were appropriate for labial, alveolar, or velar stops. Despite the fact that there was no opportunity to use experienced covariation of F1 onset frequency and F1 cutback, quail exhibited reliably shorter ‘‘labeling’’ boundaries (more voiceless stops) for intermediate stimuli of the continua when F1 frequency was higher. Human performance with the same stimuli was like that for the birds. These results support the conclusion that the effect of F1 onset frequency may be adequately explained by general auditory processes. [Work supported by NIDCD/NIH Grant No. DC-00719.]
Acoustic comparisons of some Korean and English consonants
Authors: Schmidt Anna Marie
Published: October 2005

Presently available acoustic and perceptual data do not make it possible to evaluate the degree of phonetic similarity between vowels and consonants found in Korean and English. Recent models of second language acquisition [e.g., J. E. Flege, Q. J. Exptl. Psychol. 43A, 701–731 (1991)] require the availability of such phonetic measures for the formulation of specific testable hypotheses. One purpose of the present study was to expand the limited data set of Korean acoustic measures. Another purpose was to formulate hypotheses about which sounds native Korean speakers will use when learning English. Acoustic measurements were performed on six Korean consonants (/t/) lax, tense, aspirated, /s/ lax, tense, and /l/) in five vowel contexts. Measurements included vowel dimensions such as duration and format structure as well as consonant dimensions such as VOT and amplitude rise time. Similar measures were performed on comparable English consonants and vowels. Observations will be made of the relative degree of phonetic similarity of the Korean stop series to the English /t/ and /d/ and of the Korean fricative series to the English interdental, alveolar, and palatal fricatives. [Work supported by NIH.]