bluuebunny/crossref_metadata_2025 huggingface dataset:
Music analysis/synthesis using the analysis-by-synthesis/overlap-add sinusoidal model
Authors: George E. Bryan , Smith Mark J. T.
Published: October 2005

Sinusoidal models have been successfully applied to a variety of speech processing problems, including speech analysis/synthesis. Due to their generality they are also becoming important for the analysis, synthesis, and modification of music signals. The analysis-by-synthesis/overlap-add (ABS/OLA) system is a sinusoidal modeling system which is well adapted to represent and synthesize musical instrument tones. The ABS/OLA system uses an analysis-by-synthesis strategy to determine the parameters of well-resolved sinusoids much more accurately than is possible by direct Fourier analysis, and uses a computationally efficient overlap-add synthesis model that has been generalized to produce artifact-free synthetic tones in the presence of time- and frequency-scale modification. This paper summarizes the ABS/OLA system, details its application to the analysis, synthesis, and modification of musical tones, and presents recent results in its development, including generalization to quasiharmonic tones and improvements in the computational efficiency of analysis.
Perceptual evaluation of a programmable radiation pattern loudspeaker simulating a real instrument
Authors: Caussé René , Warusfel Olivier , Bloch Georges
Published: October 2005

Conventional loudspeaker radiation patterns have no relationship to those of the sound sources which they purport to simulate [Caussé etal., Proceedings of I.S.M.A. Tokyo, 67–70 (1992)]. In this particular study, the perceptual difference between a real instrument and a programmable radiation pattern loudspeaker system (12 independently programmable loudspeakers mounted on a polyhedral structure) was investigated. In a first experiment, the trombone was chosen for its radiation pattern which presents a cylindrical symmetry and a monotonic variation according to frequency. For purposes of reproducibility during perceptive tests, the programmable radiation pattern loudspeaker was compared with an artificially driven trombone: The musical material was recorded just after the U-bend of the bell joint. This signal was reproduced by a driver, corrected by inverse filtering techniques, and mounted at the same location. To supply the programmable radiation pattern loudspeaker system, close-microphone recordings on axis and off axis were used. The programmable loudspeaker was controlled with the help of filtering processes that were calculated from previous measurements on the radiation pattern of the trombone. Results of the listening tests will be presented and discussed.
A new lattice gas model for 1-D sound propagation with dissipation
Authors: Sudo Yasushi , Sparrow Victor W.
Published: October 2005

The authors have proposed new lattice gas models for sound propagation [‘‘A New Lattice Gas Model for 1-D Sound Propagation,’’ J. Comp. Acs. (to be published); ‘‘Sound Propagation Simulations Using Lattice Gas Methods,’’ 15th AIAA Aeroacoustics Conf., AIAA 93-4404 (1993)]. These models are dissipationless. In this paper, it will be shown how dissipation effects can be combined with these lattice gas models. It is known that a random walk of lattice gas particles can simulate dissipation effects. The random walk and the motion of lattice gas particles for wave propagation can be combined using a similar method to that used to obtain a lattice gas sound propagation model for a two-dimensional square lattice from a one-dimensional lattice gas sound propagation model. One can also apply the concept of our lattice gas models to the simulation of fluid dynamic nonlinearities and the simulation of sound propagation in moving media.
Computation of beam diffraction using a non-asymptotic ray theory
Authors: Roberts Ronald A.
Published: October 2005

A technique for numerical computation of acoustic beam propagation is presented based on a nonasymptotic ray theory. Similar to conventional ray theory, an ansatz of the form φ(x)=A(x)eif(x) is used to transform the Helmholtz equation for time harmonic waves. Imposing A(x) and f(x) are both real leads to two coupled nonlinear PDEs for A(x) and f(x). Conventional ray methods employ an expansion in frequency to decouple one of these equations (eikonal equation) which is solved by ray tracing (method of characteristics). Solution of the remaining transport equation then follows. In the present method, the two coupled nonlinear PDEs are solved self-consistently without an expansion in frequency, resulting in a ray tracing procedure which automatically diffracts at caustics, etc., producing exact field results. Discussion will focus on procedures to insure numerical stability. Application to beam diffraction problems (focusing, beam spreading) will be demonstrated.
Space-time domain acoustic wave fields in media with attenuation: An analysis based on the symbolic manipulation approach
Authors: Verweij Martin D.
Published: October 2005

A method is presented for the determination of the space-time domain acoustic wave field, and in particular the space-time domain Green’s function, in homogeneous, isotropic equivalent fluid media that represent materials with a complicated viscoelastic behavior. The viscoelastic properties are modeled with the aid of arbitrarily intricate compliance memory functions, and the inertia behavior of the media is allowed to show memory effects as well. The presented integral transformation-type method consists of the transform domain Neumann series solution or WKBJ iterative solution, and the Cagniard–De Hoop method of inverse transformation. Analysis of the terms of the Newmann series solution yields a recurrence scheme that can ideally be evaluated using symbolic manipulation. Application of the Cagniard–De Hoop method results in an analytical transformation back to the space-time domain, which makes the method computationally fast. After explaining the theory, numerical results for several lossy media with an almost constant-Q behavior, like many types of rock, are presented. It is shown that it is sufficient to take into account only a few terms of the Newmann series solution to obtain accurate results, even for media with high losses.
New constitutive equation for the volume viscosity of fluids
Authors: Zuckerwar Allan J. , Ash Robert L.
Published: October 2005

The traditional constitutive equation for the volume viscosity of fluids, written for one-dimensional flow as σ′x=(λ+2μ)∂u/∂x, has no theoretical foundation and no established relationship to known relaxation processes in fluids. [for notation see H. Schlichting, BoundaryLayerTheory (McGraw-Hill Classic Textbook Reissue Series, New York, 1987)]. When applied to periodic flow, i.e., sound propagation, this equation yields expressions for the dispersion and absorption of sound, which indeed have the expected forms for a single relaxation process; but it inherently leads to the absurd conclusion that the relaxation strength must equal unity (among other deficiencies). It is shown here that a constitutive equation of the form σ′x=ηv(∂u/∂x)+ηp(Dp/Dt) yields expressions which conform to the acoustical single relaxation process, whereby ηv=−ρ0c20τps, ηp=−τvs, and τps and τvs are the isentropic relaxation times at constant pressure and constant volume, respectively. An application to a simple problem in steady compressible flow is presented.
Axisymmetric propagation modes in a liquid-filled elastic waveguide
Authors: Lafleur L. Dwynn , Shields F. Douglas , Lawrenson Christopher C.
Published: October 2005

Using Del Grosso’s theoretical treatment [V. A. Del Grosso, ‘‘Errors in Ultrasonic Propagation Parameter Measurements. Part 4—Effect of Finite Thickness Solid Tubes Enclosing the Liquid Cylinder of Interest,’’ NRL Report 6852, Naval Research Laboratory, Washington, DC (1968)] as a foundation, axisymmetric modes of progressive wave propagation in a liquid-filled elastic waveguide are studied. The waveguide system consists of a liquid cylinder surrounded by a hollow cylinder of elastic solid of arbitrary thickness. Dispersion curves and particle displacement profiles for propagating modes are calculated for several liquid/solid combinations. Modal characteristics are compared to those in the extreme cases of the rigidly bound cylinder and the empty hollow elastic cylinder. It is also shown that certain liquid/solid combinations can produce a near-plane-wave mode propagating at the intrinsic sound speed of the liquid over a range of frequencies. Experimental measurements demonstrating modal properties are presented. [Work supported by ONR.]
Axial propagation in piezoelectric, continuously twisted, structurally chiral mediums
Authors: Lakhtakia Akhlesh
Published: October 2005

Axial propagation in continuously twisted, structurally chiral mediums (CTSCMs) has been analyzed in a recent paper [A. Lakhtakia, J. Acoust. Soc. Am. 95, 597–600 (1994)], and that analysis is extended here to piezoelectric CTSCMs (PCTSCMs). The stress tensor in a PCTSCM is symmetric, while the stiffness, the permittivity and the piezoelectric stress matrices vary helicoidally about the axis of spirality. Exact analytic solutions are reported for wave propagation along the axis of spirality in a PCTSCM. Because the eigenmodes are elliptically polarized with the vibration ellipse varying along the propagation direction, application of PCTSCMs for transducers with special polarizing properties is envisaged. The general analytical work reported is exemplified by application to a PCTSCM with a base Hex. 6 mm symmetry.
Ultrasonic NDE of 3-D textile composites
Authors: Chen Julie , Krauss Gordon
Published: October 2005

Three-dimensional (3-D) textile composites provide a unique challenge to current ultrasonic NDE techniques. In 3-D textile composites, through-the-thickness stitching or tows provide reinforcement against delamination, which is common cause of failure in traditional aligned fiber laminates. However, standard nondestructive techniques for quality control and inspection are made less effective by the complex scattering created by the multidirectional fiber architecture. Although no single method will be suitable for detecting all potential defects and damage, the extent to which any given method is applicable is still relatively unexplored. The present research investigates the benefits of using plane waves to detect defects near the through-thickness reinforcements. Preliminary results will be presented on the effect of various structural parameters (e.g., interlaminar spacing, interlacing density, tow size, etc...) on the ultrasonic signal response.
A computational model of prosodic perception
Authors: Todd Neil P. McAngus , Brown Guy
Published: October 2005

There is now considerable evidence that auditory processing involves two parallel pathways. The first, a phasic pathway, which specializes in temporal processing, involves a hierarchy of temporal periodicities ranging from pitch periods to rhythmic periods. The second is a tonal pathway, which specializes in the spectral processing. Here a computational model of the phasic process based on a multi-timescale decomposition of the output from a cochlear model is presented. This model determines (1) the locatation of the syllables of a speech signal, (2) the hierarchical grouping of the syllables, and (3) the stress of the syllables. In order to test this model its output was compared with that of conventional metrical phonology for a variety of utterances.
Effect of speaking style on parameters of fundamental frequency contour
Authors: Higuchi Norio , Hirai Toshio , Sagisaka Yoshinori
Published: October 2005

The authors have analyzed the fundamental frequency (F0) contours of Japanese sentences spoken in four styles, e.g., normal, hurried, angry, and kind, for the synthesis of natural sounding speech. Thirty-five sentences in each speaking style spoken by a professional narrator were analyzed. The parameters of the F0 generation model proposed by Fujisaki, e.g., the minimum value of F0 (Fmin), the amplitude of the phrase commands (Ap) and the amplitude of the accent commands (Aa), are used here as key factors in the analysis. In the case of the sentences spoken angrily, Fmin is kept high, and the change due to both the phrase component and the accent component is minimal. Consequently, the F0 contours of sentences spoken angrily are flat. On the other hand, in the sentences spoken softly (kind) the dynamic range due to the accent component is greater than for the others, and in order to keep it high the amplitude of the phrase component is accordingly supressed. The F0 contours of the sentences spoken hurriedly are similar except that the amplitude of the accent commands is slightly smaller than those spoken normally. It was found that these parameters are useful to express the difference due to the speaking styles.
Active synthetic aperture sonar: The past 25 Years
Authors: Christoff James T.
Published: October 2005

This paper presents a brief description of the active synthetic aperture concept, its advantages and disadvantages, followed by a historical overview from Coastal Systems Station viewpoint. The overview starts in the late 1960s with the initial concept studies, progresses to the simple single-channel systems developed in the mid to late 1980s, and concludes with the present high area search rate, multi-channel, multi-aspect, realtime, motion compensated systems. Images produced by each system will be presented. The paper concludes with a brief description of present SAS efforts at CSS and the most pressing research areas for future SAS development.
Synthetic aperture sonar processing using wideband coded waveforms
Authors: Halsema John , Irza John
Published: October 2005

Traditional narrow-band synthetic aperture sonar (SAS) suffers from reduced along-track resolution and reduced pulse repetition as the speed of the sonar platform increases. Wideband SAS systems offer a method for reducing these undesirable effects at the expense of increased system complexity. Most recently, Gough and Hayes have demonstrated the performance of a wideband SAS that uses continuous-tone frequency modulation (CTFM) as the transmitted waveform [J. Acoust. Soc. Am. 86 (1989)]. This paper presents the system design for a wideband SAS that uses binary phase-shift keyed (BPSK) modulation for the transmitted waveform. In addition to being wideband (like CTFM), the BPSK signal offers unique advantages over CTFM in active SAS systems. In the system presented here, maximal-length pseudo-random (or pseudo-noise, PN) sequences are used to encode the BPSK waveform, offering the ability to transmit uniquely different coded waveforms in rapid succession, hence providing a higher spatial sampling rate than is achievable with uncoded waveforms. In addition to the system design, computer simulations are presented that compare images generated by narrow-band, CTFM, and PN-BPSK processing. Tradeoffs between image quality versus processing complexity are discussed.
Simulation of synthetic aperture sonars
Authors: Sammelmann G. S. , Christoff J. T. , Lathrop J. D.
Published: October 2005

Coastal Systems Station is developing a high-fidelity simulation of low- and high-frequency imaging synthetic aperture sonars in coastal regions. This simulation includes a coherent bottom and surface reverberation model that treats reverberation as a Gaussian random process; 3-D coherent beam patterns for the projector and receiver arrays; a 3-D coherent target strength model of an arbitrarily shaped target that treats the target strength as the coherent sum of the inelastic scattering from a large number of facets, edges, and point scatterers; and the inclusion of multipaths in the calculation of the reverberation and target returns. This simulation generates synthetic images of proud and buried targets. A number of examples of images will be presented showing the effects of system parameters and operating conditions on synthetic aperture sonar imaging. [Work supported by ONR.]
Oceanographic remote sensing system
Authors: Ekhaus Ira
Published: October 2005

A new sonar system design is proposed to detect, localize, and classify seafloor bottom laying targets, discriminating man-made items from natural ones in the presence of seafloor reverberation. Target-physics-based characteristics of multilook, multi-aperture return signals from seafloor and minelike objects (MLO) targets are exploited to improve system performance. The proposed approach is a synthesis and extension of techniques found in areas including standard and interferometric synthetic aperture radar (SAR/IFSAR), swath bathymetry (SB), monopulse radar tracking applications, and 3-D radar target identification and imaging. Additionally, the proposed approach generates on output a reusable oceanographic data format that increases the utility of acoustic databases derived from the format and provides a convenient paradigm for visualization of the data.
A performance comparison of motion estimation and correction techniques for synthetic aperture sonar imaging
Authors: Silkaitis John M. , Douglas Bretton L. , Lee Hua
Published: October 2005

Ideally, a synthetic aperture sonar platform traverses a perfectly straight path as it forms the synthetic aperture. However, medium turbulence and currents affect platform motion. Errors between ideal trajectory and true trajectory of a fraction of an acoustic wavelength will cause a loss in phase coherence during data acquisition resulting in unfocused images. Autofocus algorithms, commonly used in synthetic aperture radar (SAR) imaging, have been developed to overcome phase-error problems. However, phase errors in synthetic aperture sonar data are different in nature due to receiver configurations and pulse repetition frequencies. Synthetic aperture radars typically use a single receiver and a high pulse repetition frequency producing a smoothly varying phase error over the synthetic aperture. To achieve satisfactory mapping rates and maximum range, a synthetic aperture sonar receiver needs to use a multiple-element receiver and lower pulse repetition frequency. In synthetic aperture sonar, the resulting phase errors between adjacent receiver elements are typically small and linear, however, phase errors between echo data obtained on successive pulses are significant. An autofocus algorithm has been developed specifically for correcting phase errors in synthetic aperture sonar data. In this paper, its performance is compared with conventional SAR autofocus algorithms on both simulated and experimental data. [This research is supported by UC MICRO program and Sonatech, Inc.]
The extraction of acoustic propagation parameters in shallow water using synthetic aperture processing
Authors: Lawrence Thomas N. , Bedford Nancy R.
Published: October 2005

Synthetic aperture processing (SAP) is a method of extracting horizontal wave numbers from recordings on one, or on a very few hydrophones. The authors have reported results from a range-dependent experiment using an array in deep water [J. Acoust. Soc. Am. 93, 2374(A) (1993)]. In the experiment reported in this presentation SAP have been combined with the robustness of a 48-element vertical array covering most of the water column in 400 m of water. A determination was made of detailed shallow water propagation parameters using a moving cw sound source projecting 25 and 45 Hz. Estimations can be made for mode depth functions, mode attenuation, and mode eigenvalues as a function of source range. At 25 Hz almost all propagating modes are bottom interacting. So a determination of the effects of bottom properties on acoustic propagation can be made without direct measurement of bottom parameters. The environment is slightly range dependent, and experimental results will be compared with parameters obtained from an adiabatic normal mode model.
Data-adaptive wave-vector filtering for acoustic arrays
Authors: Ferguson Brian G.
Published: October 2005

Both conventional and adaptive wave-vector filtering techniques are applied to microphone data from an experimental land-based acoustic surveillance array and to hydrophone data from a horizontal line array towed below the sea surface. After transformation from the time domain to the frequency domain, the outputs of each array’s acoustic sensors are processed in the spatial domain by each of the wave-vector filtering techniques. A comparison of the results shows that the data-adaptive technique, which requires inversion of the cross-power spectral matrix, offers superior performance by suppressing spatial leakage in the wave-number domain, enhancing the spatial resolution through narrower beamwidths and providing superdirective array gain at frequencies well below the spatial Nyquist frequency of the array. Adaptive wave-vector filtering is shown to be a powerful diagnostic tool for analyzing the self-noise in the towed array.
Eigenstructure-based algorithms for time varying array
Authors: Friedlander Benjamin , Zeira Ariela
Published: October 2005

This study considers the problem of finding the directions of narrow-band signals using a time-varying array whose elements move during the observation interval in an arbitrary but known way. The emphasis is on the case where the shape of the array changes significantly during the observation interval. The motivation for this study are problems involving direction finding systems employing more than one platform such as a spatially referenced synthetic aperture sonar. Assuming that the signal is uncorrelated from snapshot to snap-shot, two eignestructure based algorithms for this problem were derived. These algorithms are modifications of techniques developed originally for time invariant arrays. The first uses array interpolation, and the second uses the ideal of focusing matrices. Like other eigenstructure methods, these algorithms require a modest amount of computations in comparison with the maximum likelihood estimator. The performance of the algorithms is evaluated by Monte-Carlo simulations and compared to the performance of the time-varying beamformer and to the Cramer–Rao bound. Although both techniques were successful for wideband array processing with time invariant arrays, it was found that only the interpolated array algorithm is useful for direction finding with time varying arrays.
Implementation of adaptive beamforming processing schemes in a towed array sonar system
Authors: Riley J. , Trider R. C. , Stergiopoulos S. , Ashley A. T. , Ferguson B. G.
Published: October 2005

This experimental paper is concerned with the implementation of the minimum variance distortionless response (MVDR) method [IEEE Trans. 57(8), 1408–1418 (1969)] in a towed array sonar system. The paper discusses the issues associated with the implementation of the MVDR in a real time processing system and examines the robustness of the algorithm using a variety of sea trial data. The performance of an algorithm that pre-processes the data before the MVDR algorithm is applied is also discussed and compared with the standard approach. The experimental results obtained suggest that, as expected, the robustness of the MVDR algorithm is the key to its successful implementation in a real time system. When this condition is met, the MVDR has demonstrated the capability to enhance the conventional beamformer by providing improvements in both angular resolution and detection performance. The implementation of the MVDR algorithm in a real time towed array processing system has necessitated the development of
Revolving-projection stereo imaging of hard-skinned sonar targets
Authors: Henderson Terry L.
Published: October 2005

It is well known that the three-dimensional shell geometry of the visible portion of a distant hard-skinned sonar target can be scanned in xy coordinates by beamforming a planar array, with the associated z coordinates being provided by time gating the acoustic return. However, a simple line array running along the x axis can provide a projection view of the target, with its upper and lower surfaces being projected semi-transparently onto the xz plane, giving a picture analogous to that of a paramecium viewed with a slide microscope. If the line array is now tumbled in the xy plane, it will appear that the target is revolving around the z axis. When this tumbling view is presented 14° out of phase to the left and right eyes, a stereoscopic view of the tumbling target is obtained, thus revealing its three-dimensional form (with the aid of the brain’s innate tomographic signal processing powers). Further simplifications of the array structure and processing are possible, with some limitations and weaknesses. A videotape of results for several targets has been prepared for presentation.
Absolute calibration of high-intensity focused transducers used in pulsed mode
Authors: Kapodistrias George , Madanshetty Sameer I.
Published: October 2005

Absolute calibration of high-intensity focused transducers is attempted. The difficulty with using the high-intensity focused fields in underwater applications is that at the high pressure amplitudes involved, cavitation can occur, which can destroy the reference transducer used for calibration. Reference transducers of the PVDF membrane type are expensive and cannot be sacrificially used for calibration purposes. A robust alternative to using the sacrificial reference transducer is to use an economical needle type DapcoTM transducer to pick the acoustic pulses to be calibrated. It is used to measure the relative positive and negative peak amplitude variations with increasing input voltage. [At high intensities the nonlinearities on the acoustic pulse manifest as heightened positive pressure peak accompanied by shallow negative pressure (tensile) peaks.] Additionally, a radiation force balance is used to measure the net acoustic output. Knowing the beam profile and applying a nonlinearity correction factor developed absolute magnitudes of the positive and the negative pressure peaks can be inferred for pulsed applications. It is to be noted that for cavitation studies one needs to know the peak negative pressure (the tensile component).
Test results and design analysis for a thermoacoustic underwater projector
Authors: Ward William C. , Gabrielson Thomas B.
Published: October 2005

An experimental thermoacoustic projector (a heat-driven sound source without moving parts) produced source levels near 190 dB at 120 Hz during recent tests at the Navy’s Seneca Lake facility. These data were taken near 60 m depth; in thermoacoustic projectors, the source level increases linearly with depth. The device is composed of two coupled vertical tubes. The upper driver tube is filled with helium and contains a thermoacoustic stack with hot and cold heat exchangers. The lower tube is an impedance matching device filled with water up to a variable level. The tube opening is necessarily small compared to wavelength. For a radiation impedance with such a small resistive component, the test device demonstrated a resonant mode ambiguity before reaching the optimum tuning point. This effect reduced the maximum source level by 3–5 dB below the design level, and had a similar effect on the overall efficiency. This presentation will give an overview of the experimental results and introduce design modifications to overcome the observed limitations and reduce the size of the projector as well.
Acceleration-insensitive hydrophone
Authors: Beason M. K. , Trivett D. H. , Luker L. D.
Published: October 2005

Generally, the measurement of sound pressure in water is not significantly affected by the small acceleration sensitivity of the hydrophone. In a standing-wave tube, however, regions exist with very large particle velocities and small pressures. The acceleration sensitivity of the hydrophone can, therefore, cause a large error in the measurement of the pressure. The long-line-hydrophone calibrator (LLHC) uses the electroacoustic transfer matrix of the system to determine the projector drives required to simulate a plane-wave pressure field within the calibrator. The algorithm employed operates the calibrator as a standing wave tube to measure the transfer matrix. Thus measurement inaccuracies caused by acceleration-sensitive hydrophones will affect the quality of the field. A hydrophone design is presented that reduces the acceleration sensitivity by using spherical piezoelectric elements and ‘‘free-floating’’ mounts. Results are also presented that compare relative measurements of pressure errors caused by hydrophone acceleration sensitivity in a standing-wave tube. [Work supported by NAVSEA.]
A computational theory of auditory event detection
Authors: Irino Toshio , Patterson Roy D.
Published: October 2005

A theory of asymmetric intensity enhancement around acoustic transients is presented. Experiments with ramped and damped tones [R. D. Patterson, J. Acoust. Soc. Am. 93, 2293 (A) (1993)] have shown that reversing an asymmetric envelope changes the timbre of the sound. A ‘‘delta-gamma’’ filter is introduced to produce asymmetric intensity enhancement around transients to explain the change in perception. The filter is the derivative of a gamma function, ∂Gm(f,t)/∂t. Sound envelope changes are then detected by the sign of ∂Gm(f,t)/∂t*‖s(t)‖, for sound envelope ‖s(t)‖. If it is assumed that the onset of a sound is more important biologically than what comes shortly thereafter, then this onset information can be enhanced by the filter at the expense of suppressing the following part during the negative value of the delta-gamma filter. It is shown that this filter can be implemented as a difference between the original and delayed envelopes of the output of the gammatone filter. The compression, suppression, and envelope estimation processes in the auditory periphery do not affect the sign of the delta-gamma filter. The theory can explain the experimental results obtained with damped and ramped tones.
A model of the effective auditory signal processing. I: Masking experiments
Authors: Dau T. , Pueschel D. , Kollmeier B.
Published: October 2005

A model is described that simulates the detection characteristics of the human auditory system for different psychoacoustical tasks. An optimal detection process is attached to the model after the preprocessing stages. The statistical basis was provided by the assumption of an internal Gaussian noise following nonlinear processing in the peripheral auditory system. The nonlinear processing describes essential features of human temporal processing, such as forward masking, temporal integration, and dynamic range compression. Simulations and measurements of backward, simultaneous, and forward masking are presented employing deterministic (frozen) and stochastic (running) broadband noise maskers. The stimuli were presented to the model in an paradigm (adaptive forced-choice procedure) identical to the one used with human subjects. In most cases, a very good agreement between simulated and measured thresholds is achieved.
Temporal integration, multiple looks, and signal uncertainty
Authors: Wakefield Gregory H.
Published: October 2005

Viemeister and Wakefield [J. Acoust. Soc. Am. 90, 858–865 (1991)] presented experimental results that challenged the notion of temporal integration as some form of decision process characterized by a long time constant. When the signal consists of brief temporal pulses, detection thresholds vary with the square root of duration, which is consistent with a multiple-look observer. However, when the signal is a continuous sinusoid, detection thresholds vary linearly with duration. The present paper introduces a multiple-look observer in which signal uncertainty and sequential estimation are incorporated. The general form of this observer involves a sequential refinement of the alternative and null hypotheses (from highly uncertain to highly certain stimulus conditions) during the observation interval. Several simplifying assumptions yield an ideal observer whose performance, which can be evaluated analytically, predicts the form of temporal integration for a variety of signals including sinusoids, wideband noise, and pulse sequences. The generality of the theoretical results and experimental verification of the sequential nature of the observer are discussed. [Work supported by grants from NIH.]
Optimal decision rules for the oddity task
Authors: Versfeld Niek J. , Dai Huanping , Green David M.
Published: October 2005

In the most general oddity task, m-1 stimulus alternatives are the same and one is different. The decision maker must identify the odd stimulus. Two likelihood-ratio decision rules are derived by treating the stimuli as Gaussian random variables with means that depend on the stimulus conditions. The two rules depend on the correlation between the observations generated by the stimuli. For m=3, which is by far the most popular oddity task, two extreme cases are discussed. If the observations are highly correlated, the optimum decision rule is equivalent to the so-called triangular rule. The other decision rule is slightly different, and is optimum when the stimulus observations are independent. Different rules are applicable to experiment using fixed versus random standards. A COSS (conditional on a single stimulus) analysis provides a convenient way of estimating the rule actually used by a decision maker in a typical decision task. [Work supported by the Dutch Ministry of Defence and by the U. S. AFOSR.]
Computation of the focusing of sound by solid spheres in water
Authors: Hickling Robert , Petrovic Mirjana
Published: October 2005

Previously [R. Hickling, J. Acoust. Soc. Am. 94, 1815(A) (1993)] computed visualizations of sound-power flow were presented that demonstrate the focusing of sound by solid spheres, as a function of frequency, for a number of different materials. In the past, fluid-filled spheres have been used to obtain focusing and to increase target strength [e.g., E. O. Belcher, Proc. Oceans 93 Conf. (Oct. 1993)]. Solid spheres, however, avoid possible leakage and are probably less susceptible to temperature change. The use of liquid-filled spheres for focusing is usually based on simple ray-refraction theory. The evaluation of solid spheres for focusing requires a greater amount of computation. Computational methods and techniques are described. Potential applications include omnidirectional passive surveillance in a half-space and beam forming using a driven system.
Scattering from complex fluid-loaded structures using asymptotic methods and computers
Authors: Norris Andrew
Published: October 2005

The coupling of acoustic energy to and from smooth fluid-loaded thin shells can be described using techniques similar to the theory of geometrical diffraction. The frequency range of interest is below flexural coincidence but still high enough that asymptotic methods are applicable, which is approximately 10<ka<60. Two distinct mechanisms dominate this mid-frequency range: (i) a ‘‘background’’ response determined by a local inertial impedance, and (ii) phase matching to longitudinal and shear waves, collectively called membrane waves. The background response can be approximated by a specular reflection, but membrane waves require global treatment over the whole structure. The methodology will be demonstrated by application to shells of nonseparable shape. Several distinct steps are involved: first the coupling curves are determined—these are the closed loci defined by phase matching with the incident wave. Fans of rays are sent out over the shell, each evolving according to a ray equation and a transport equation. The surface field is built up in this manner, becoming progressively more complex with increasing time. The interaction with shell discontinuities can be dealt with by including appropriate diffraction coefficients. Some of these are known, others remain to be determined either analytically or computationally. [Work supported by ONR.]
Prediction of vibrations using a wave sampling decomposition
Authors: Trentin D. , Guyader J. L.
Published: October 2005

At medium frequency range (MFR), numerous vibroacoustic problems deal with systems of large number of modes where classical modal methods must cope with resolution of huge linear systems. In the idea of reducing these models, a mode hybridization method [H. J-P. Morand, ‘‘A modal hybridization method for the reduction of dynamic models in the medium frequency range,’’ Proceedings of the European Conference on New Advances in Computational Structural Mechanics (1991)] gives one equivalent mode obtained with modes of the complex structure. Another method [D. Trentin and J. L. Guyader, ‘‘Vibrations of a master plate with attached masses using modal sampling method,’’ J. Acoust. Soc. Am. (to be published)] based on reducing size of dynamic model in the MFR, gives a prediction of the vibration response of a structure by using a limited number of modes of the homogeneous master structure in order to interpolate the response of modes of whole structure with attached heterogeneities. The method presented here gives a prediction of plate vibration using a wave decomposition instead of modal calculation. The corresponding wave function of the Fourier transform of the displacement is approximated the best by achieving the extremalization of the Hamiltonian of the plate. Quadratic velocity of the structure is derived and the convergence study of the method shows that global quantities involving frequency average can be obtained with reduced models of small size. [Work supported by the Institut National de Recherche et de Sécurité and by the Centre National d’Etudes Spatiales.]
Time-domain methods of analysis in structural acoustics
Authors: Stepanishen P. R.
Published: October 2005

An overview of the use of time-domain methods to analyze the mechanical response of fluid loaded structures invacuo and in fluid is presented. The response of a single degree of freedom mechanical oscillator will first be addressed and then used to build the solution to more complex plate and shell vibration problems using well established time-dependent eigenfunction or eigenvector expansions. In contrast to the invacuo problems in which the modal coefficients are uncoupled convolution integrals, the coefficients for the fluid-loaded problems are shown to be described by a set of coupled convolution integral equations. The underlying physics that naturally leads to the coupled convolution integral equations will be discussed. Direct time-domain methods of evaluating the convolution integrals and solving the convolution integral equations will be presented and discussed. Finally, numerical results will be presented for both in-vacuo and in-fluid plate and shell problems to illustrate typical response characteristics.
Exact analysis of the acoustic scattering by an arbitrarily insonified spherical body near a flat boundary
Authors: Huang H. , Gaunaurd G.
Published: October 2005

The acoustic scattering by spherical bodies (hard, soft, or elastic) near flat boundaries (hard or soft) insonified by plane waves at arbitrary incident angles is analyzed exactly in the low and intermediate frequency ranges. To satisfy the boundary conditions at the flat boundary as well as on the surface of the sphere, the mathematical problem is formulated using the image technique. The scattering wave fields are expanded in terms of the classical modal series of spherical wave functions utilizing the translational addition theorem. Quite similar to the problem of multiple scattering by spheres, numerical computation of the scattered wave pressure involves the solution of an ill-conditioned complex matrix system the size of which depends on how many terms of the modal series are required for convergence. This in turn depends on the value of the frequency, the proximity of the spherical body relative to the flat boundary and the incident angle. The ill-conditioned matrix equation can be solved using the Gauss–Seidel iteration method and back scattered echoes from the spherical bodies are extensively calculated. [Work supported by the NSWC Independent Research Program.]
Theory and application of harmonic wavelets
Authors: Newland David E.
Published: October 2005

Wavelet analysis allows a signal f(x) to be decomposed into a family of orthogonal functions whose members have different scales and different positions along the x axis. There has been a great deal of research into the theory of dilation wavelets, which arise from the recursive solution of a special class of difference equation, and which cannot be expressed in the form of mathematical functions. In contrast, the author’s harmonic wavelengths have a simple structure that can be written in terms of harmonic functions. They are concentrated locally and are orthogonal to their own discrete translations and dilations. Their frequency bandwidth can be chosen arbitrarily. Various applications will be demonstrated, including the analysis of music and the extraction of wave velocity data from impulse response records. Results are shown in the form of wavelet maps which are similar in concept to sonograms and show the distribution of mean-square over position and frequency. There is a quick algorithm for computing the harmonic wavelet transform. This is implemented by the FFT and is faster than the usual algorithm for computing dilation wavelet transforms. It can be optimized in order to make the harmonic wavelet transform invariant to position (which the dilation wavelet transform is not).
Freedoms and constraints in computational prosody modeling from speech corpora
Authors: Sagisaka Yoshinori
Published: October 2005

With the growth of the availability of large speech corpora, statistical models of prosody control have been studied intensively. These computational models have improved the naturalness of synthetic speech and are expected to provide additional supra-segmental information for speech recognition. Though only conventional statistical methods such as linear regression, regression trees, or neural nets have been employed in these computational models, their success has been due to efforts to accommodate observed prosodic characteristics and qualitatively known control mechanisms. In this talk, a review will be presented on how observed prosodic characteristics have been modeled with these statistical tools and also how new statistical models can be designed to cope with the insufficiencies of conventional models. It is expected that the investigation of well-constrained models and their constraints will lead one to more efficient computational models and deeper understanding of prosody control mechanism through these modeling procedures.
Generating intonation with a dynamic lexicon of prosodic prototypes
Authors: Auberge Veronique
Published: October 2005

A model of intonation is described that uses hierarchically stored intonation contours and inferences about grammatical structure to generate complex intonation patterns for text-to-speech (TTS). The model makes two assumptions. First, intonational structure shares points of articulation with other linguistic structures—morphological, syntactic, and semantic—even though these structures may be quite different. The common nodes in intonational and other structures are called ‘‘rendez-vous.’’ Second, intonation contours, whose boundaries are marked by the rendez-vous, can be characterized by prosodic prototypes. Intonation is thus generated from the lexicon of prototypes, where parameters describing each prototype are defined by the rendez-vous. The lexicon itself is structured according to a rendez-vous hierarchy; it contains syllabic and supra-syllabic units associated with classes of F0 movements. For TTS, intonation generation is linked with a text parser that is able to deliver rendez-vous. A rule-governed multi-agent system allows interactive exchanges between the intonation generation module and the text analysis module. Interactions among the different language components (morphology, syntax, semantics) allow intonation parsing to facilitate the resolution of some structural ambiguities. In such a system, text analysis can be readily adapted for intonation generation in a variety of text and reading styles.
Text-to-speech synthesis and statistical analysis of natural speech corpora
Authors: van Santen Jan P. H.
Published: October 2005

A text-to-speech (TTS) system consists of modules that convert orthographic input via various intermediate representations into speech. The principles underlying these modules are quite diverse. The orthography-to-phoneme module (for some languages) can use pronunciation rules from the research literature. For most modules, however, the literature lacks the information needed for their implementation. Increasingly, these knowledge gaps are filled not by conducting new research but by applying statistical tools to speech corpora and absorbing the estimated parameter values directly in the TTS system. A challenge to statistical approaches, stemming from the input domain of TTS systems being {m open}, will be discussed. Specifically, the input domain has the property that the number of very rare contextual constellations is so large that the probability of some rare constellation occurring even in a small text sample is quite large. Consequently, corpora rarely cover all constellations that the TTS system nevertheless must be prepared for. This makes it important for statistical approaches to be based on corpus-independent regularities. Such regularities can be discovered with exploratory data analysis (a type of statistical analysis used prior to parameter estimation) together with an understanding of what could account for them. These points will be illustrated with speech timing and corpus-based concatenation.
Annotation and prosodic control in the Eloquence text-to-speech system
Authors: deJong Kenneth J. , Hertz Susan R.
Published: October 2005

A simple set of text annotations that enables users to produce sophisticated prosodic effects is being developed as part of the Eloquence text-to-speech system. The annotations relate to concepts such as ‘‘emphasis’’ and ‘‘level of excitement.’’ The rules interpret the typically sparse annotations in the process of building up a rich, ‘‘multi-stream’’ phonological and phonetic representation from which the final values for synthesis are derived. This structure includes prosodic phrases with associated tones, words with associated pitch accents, syllables and their nuclei, fundamental frequency values, and durations. For example, marking a work for emphasis triggers several actions: the rules place an accent on the word; they associate tones appropriate to the level of emphasis and phrase type; they attract nuclear stress to the word and deaccent following words in the phrase; they increase the pitch range at the emphasized word; and they lengthen the accented syllable in accordance with our nucleus-based timing model. The sophistication of the underlying rule set gives even untrained users access to a broad range of phonological and phonetic resources. The annotations along with their phonological and phonetic consequences will be described, and their effects on the final synthetic speech demonstrated. [Work supported by NIH.]