bluuebunny/crossref_metadata_2025 huggingface dataset:
On indirect acoustic impedance computations involving fixed free-space attenuation or fixed boundary loss factors.
Authors: Oni Adeboyejo A.
Published: October 2005

A class of methods for speedily estimating the acoustic impedance of surfaces is presented. The methods utilize a lattice search algorithm to estimate the impedance that will validate analytical expressions for ground attenuation of sound, using experimental measurements of continuous-wave sound attenuation changes between two above-ground sound source locations with respect to a single microphone close to the ground. In the two cases considered, opportunities appear to exist for a considerable reduction in computation complexity: in the first case, the arrangement allows spherical spreading components of the excess attenuation terms to essentially cancel out. In the second case, the arrangement allows the boundary loss factor to remain constant while the source sound location changes. Variations in meteorological parameters between the two sound source locations are assumed negligible. These proposed analytical and experimental arrangements display the potential to provide a means of obtaining speedy and efficient estimates of ground impedance for various surfaces. [Work supported by the U.S. Army Research Office.]
The propagation of plane N waves through a statistically isotropic turbulent field as a singular perturbation problem.
Authors: Lipkens Bart
Published: October 2005

Reported here is a theoretical investigation to explain experimental observations of the effect of turbulence on N-wave propagation [B. Lipkens and D. T. Blackstock, J. Acoust. Soc. Am. 90, 2307(A) (1991)]. An adaptation of von Kármán’s spectral model for incompressible, isotropic turbulence is used to generate a statistical realization of a turbulent field. The 3-D, random, isotropic velocity field consists of a collection of discrete Fourier velocity modes [M. Karweit etal., J. Acoust. Soc. Am. 89, 52–62 (1991)]. A lossless wave equation is derived that incorporates the effect of the turbulence field [see, for example, K. Plotkin, Ph.D. thesis, Cornell University (January 1971)]. The turbulence is assumed to be frozen during passage of an initially plane N wave. A regular perturbation expansion in the turbulence Mach number is used to solve the problem. At second order the problem becomes singular, and coordinate straining is applied to remove the singularity. The first-order perturbation makes the N wave ragged. The second-order perturbation appears to introduce changes in the phase speed. [Work supported by NASA.]
Molecular relaxation effects on the thickness of weak shock waves in seawater.
Authors: Kang Jongmin , Akay Adnan
Published: October 2005

Many works have been done for predicting peak pressure and the decay time constant of underwater explosion waves; however, little efforts had been made for determining a detailed structure of the leading edges. Marsh etal. [J. Acoust. Soc. Am. 38, 326–338 (1966)] investigated the effects of the finite amplitude and MgSO4 relaxation on the explosion waveforms in seawater, and showed the anomalous increase of the rise time from their measurements (from 5 to 12 μs). Recently, a molecular relaxation model for determining a rise phase of weak shocks propagating through the relaxing atmosphere was developed by Kang and Pierce, where O2 and N2 are dominant relaxation processes. The theory is applied to the rise phase of weak shocks in seawater, in which, equivalently, two relaxation processes are important: MgSO4 and B(OH)3 [Fisher and Simmons, J. Acoust. Soc. Am. 62, 558–564 (1977)]. The molecular relaxation in seawater depends on temperature, ambient pressure, salinity, and pH. Numerical simulations show the dependency of shock thickness on peak pressure and local properties of seawater such as temperature, ambient pressure, and salinity, and also clarify the individual role of each relaxation process and the classical absorption.
Acoustic surface waves propagating horizontally above a seasonal snow cover.
Authors: Albert Donald G.
Published: October 2005

Experimental measurements of acoustic impulses propagating horizontally above a seasonal snow cover reveal the presence of acoustic surface waves that were not observed under grass or frozen ground conditions. The measurements were obtained using blank pistol shots fired 1 m above the snow as the source of acoustic impulses; the resultant acoustic waveforms were measured by microphones at heights of from 0.1 to 4.75 m after 60 m of horizontal propagation along the surface. the snow cover was 0.20 m thick with an average density of 200 kg m−3 and crystal sizes of 0.5 to 1 mm. The peak pressures measured at all of the microphones were markedly reduced compared to pressures measured when a snow cover was not present because of the well-known absorptive effect of the snow. In addition, waveforms near the surface displayed a strong, low-frequency ‘‘tail’’ following the impulsive arrival from the shot. This tail was found to decay exponentially with height z above the surface [∼e−αz], a diagnostic feature of surface waves, with a measured attenuation coefficient α≊0.5 m−1. Waveforms calculated using Attenborough’s four-parameter, layered model of ground impedance are shown to agree with the observed waveforms when an assumed surface effective flow resistivity of 20 kN s m−4 is used. [Work supported by the Directorate of Research and Development, U.S. Army Corps of Engineers, Project 4A161102AT24.]
Contemporary issues in masking hailing from Zwicker’s work.
Authors: Buus So/ren
Published: October 2005

Recent work on masking has focused on cases that appear inconsistent with the power-spectrum model of masking. This model holds that simultaneously masked thresholds are determined by the signal-to-masker ratios within frequency-selective channels. More than a decade ago, Zwicker noted that an amplitude-modulated broadband masker is much less effective than a steady-state masker with the same energy. This effect, known as comodulation masking release (CMR), might be mediated by detection of envelope correlation or by listening during brief periods of low masker intensity (in the valleys). Recent experiments show that CMR is obtained for brief tones presented in the valleys of the masker, but not at the peaks. This finding favors the ‘‘listening in the valleys’’ explanation. Zwicker also was among the pioneers in the exploration of ‘‘overshoot’’ of masking. Recent experiments show that the overshoot depends on the onset of energy outside the critical band centered on the signal and not on energy within it. Finally, Zwicker had a long-standing interest in how detection thresholds depend on signal bandwidth. Recent experiments show that the energy at detection threshold is nearly independent of bandwidth for brief signals, but increases with bandwidth for long-duration signals. This finding indicates that the decision rule used for detection depends on duration. All three phenomena can be modeled if the weights applied to the frequency-selective channels of the power-spectrum model are assumed to vary over time because they are driven by the stimulus. [Work supported by NIDCD.]
Temporal effects with single- and multiple-band maskers: Effects of level.
Authors: Bacon Sid P. , Takahashi Gail A. , Hicks Michelle L.
Published: October 2005

The temporal effect with broadband maskers was dubbed ‘‘overshoot’’ by Zwicker in 1965. Recent experiments suggest that frequency regions quite remote from the signal frequency are important for this effect. The purpose of the present study was to determine which frequency region(s) might be responsible for the observation [S. P. Bacon, J. Acoust. Soc. Am. 88, 698–702 (1990)] that overshoot is nonmonotonic with masker level. The signal was a 10-ms, 4.0-kHz sinusoid presented 1 or 195 ms after the onset of the 400-ms masker. The spectrum level of the masker ranged from −5 to 55 dB. Four masker configurations were employed: lower band (LB, 2.5 to 3.5 kHz), middle band (MB, 3.5 to 4.5 kHz), upper band (UB, 4.5 to 5.5 kHz), and all band (AB, 2.5 to 5.5 kHz). There was a significant temporal effect for all but the MB; it was monotonic with level for the UB, but nonmonotonic for the LB and AB, suggesting that the LB contributes most to the nonmonotonic effect in the AB condition. Preliminary analyses suggest that at low levels (where the temporal effect exists in the AB condition) the masking by the individual bands is combining in a much more nonlinear fashion at the 1-ms delay. [Work supported by NIDCD.]
Intensity discrimination in backward masking.
Authors: Plack Christopher J. , Viemeister Neal F.
Published: October 2005

It has been demonstrated that the Weber fraction for a brief tone burst is elevated when the tone is preceded by an intense narrow-band noise [F-G. Zeng etal., Hear. Res. 55, 223–230 (1991)]. This elevation is greatest when the tone burst is at medium sound levels (around 40–70 dB). The experiment of Zeng etal. was repeated except that the presentation order of the tone burst and the noise was reversed. Intensity discrimination was measured for a 1-kHz tone burst with a steady-state duration of 25 ms and 2-ms onset and offset ramps. The tone burst was presented in quiet or 100 ms before a 200-Hz-wide narrow-band noise centered on 1 kHz with a spectrum level of 70 dB and a duration of 100 ms. Not only was the ‘‘midlevel hump’’ evident in the backward masking conditions, but the effect was larger than in the equivalent forward masking case. This result will be discussed with regard to possible explanations for the existence of the midlevel hump. [Work supported by NIDCD Contract No. DC00683.]
Pulse compression using passive phase-conjugate processing.
Authors: Dowling David R. , Jackson Darrell R.
Published: October 2005

Passive phase-conjugate (PPC) processing consists of using the first arrival, from a stream of pulses that have traversed a refractive medium, as the matched filter for later pulse arrivals. Its connection with true ‘‘active’’ phase conjugation [see Jackson and Dowling, J. Acoust. Soc. Am. 89, 171 (1991)] lies in invoking reciprocity for the back propagation. When the intervening acoustic medium changes slowly, the processor creates unambiguous maxima regardless of the number of propagation paths between the source and the receiver. The technique can be easily extended to receiving arrays by coherently summing the PPC output from each hydrophone. Differences in the height and phase of the peaks in the PPC processor output from different pulses are produced by changes in the acoustic medium and the source-array geometry. The results from recent acoustic propagation experiments, including IWAC ’90, will be used to illustrate some of the benefits of this technique. Applications will also be discussed. [Work supported by NRL.]
Optimization of acoustic array design using the simulated annealing algorithm.
Authors: Hayward Thomas J.
Published: October 2005

The simulated annealing algorithm [S. Kirkpatrick etal., Science (May 1983)] has been applied recently to a number of underwater acoustics problems. In this paper methods developed in recent years for the optimization of sonar receiver array configurations, including both horizontal and vertical arrays, are reviewed. Optimization of horizontal planar-array configuration to minimize noise gain in a 2-D-isotropic noise field results in highly ordered configurations resembling crystalline or quasicrystalline structures. Vertical-array configuration optimization for conventional beamforming is based on maximization of signal-to-noise ratio at the array output. The optimized vertical-array configurations exhibit far less regularity of spacing, due to the complexity of the acoustic field in the vertical. Nevertheless, the results are readily understood as a trade-off between placement of sensors at depths with high signal-to-noise ratio and separation of sensors to reduce the array noise gain. Optimization for matched-field processing is also considered.
Robust adaptive processing for underwater acoustic arrays.
Authors: Burkhardt Russell C. , Sibul Leon H.
Published: October 2005

Adaptive processors for underwater acoustic arrays are known to be sensitive to uncertainty in the spatial structure of the signal. The performance of an adaptive algorithm is also degraded in cases where the second-order noise statistics cannot be accurately estimated such as when an array with many elements is involved and/or when the data are highly nonstationary. The conventional approach to uncertainty in the signal structure is to use constraints. These constraints, in general, do not directly use apriori knowledge of the uncertainty in the signal and also ignore the existence of any error in the noise statistics, the presence of which tends to increase sensitivity to other errors. In this work, an approach to this problem is developed in which realistic models of the uncertainties in both the signal and noise structures are used to derive robust adaptive algorithms based upon a minimax criterion. The realistic signal uncertainty models are also used to formulate robust matched-field processors. In addition, a transformation to mode space based upon a minimax criterion for uncertainty in the characteristics of the acoustic channel is used to derive a robust matched mode processor.
On the gain available from passive acoustic synthetic aperture processing.
Authors: Sullivan Edmund J. , Carey William M.
Published: October 2005

Since passive synthetic aperture processing (PSAP) has long been understood to involve a trade-off between spatial and temporal gain, it has often been stated that there is no ‘‘new gain’’ to be realized. The argument, based on the model of a plane wave in isotropic noise, is as follows. For a fixed time available for processing, PSAP reduces the amount of integration time available for each sensor while increasing the number of sensors available. Thus, temporal gain is reduced while spatial gain is increased by an equal amount. However, when the noise field is nonisotropic, a different result obtains. It is shown that in the case of fields containing localized noise sources, PSAP can provide an increase in overall gain due to the ability to beamform on a desired source while rejecting undesired sources. A generalized formulation of the spatial-temporal processing gain provided by PSAP in terms of the ratio of coherent noise to incoherent noise power is presented that demonstrates the gains achievable in such a nonisotropic noise field. It is further shown that, even in the case of an isotropic noise field, if sufficient signal to noise is available, an improvement in bearing resolution can be achieved.
Woodwind models incorporating nonlinear air-column losses.
Authors: Keefe Douglas H.
Published: October 2005

The sound level within woodwinds under playing conditions is sufficiently high so that nonlinear losses are significant, yet time-domain models of sound production have assumed that the air column responds linearly. Using a linear air-column model, time-domain simulations of saxophone tones give results qualitatively similar to performed tones for low-register tones, but dissimilar results for second-register tones and multiphonics, both of which employ an open register hole. A time-domain analysis of a cylindrical tube terminated by an open orifice characterized by a nonlinear resistance shows that the reflection function amplitude is reduced as the Strouhal number tends toward unity, but the time delay is not affected. This nonlinear reduction is most important for small-diameter orifices, such as the register hole. Calculations of the saxophone input impedance for second register notes demonstrate that the nonlinear resistance of the register hole reduces the level of the fundamental resonance by 8 dB and lowers its frequency, thereby reducing the degree of harmonicity with higher frequency resonances and enabling the simulated performer to produce a tone in the upper register. Other resonances are not affected. Similar behavior is predicted for multiphonics.
Steel drums: ‘‘Steel Appeal.’’
Authors: Duerden Darren , Duerden Jennifer , Jackson Jim
Published: October 2005

Some of the unique acoustical properties of steel drums will be demonstrated in the performance of short musical selections.
Noise protection measures on ranges for small arms using sound attenuating ceilings.
Authors: Buchta Edmund
Published: October 2005

In the vicinity of shooting ranges for small arms, people are highly annoyed by impulsive shooting noise. Only the combination of shielding frames and attenuating ceilings leads to noise protection with sufficient efficiency. Such a ceiling should not prevent the natural illumination and may not stop the circulation of air with respect to the removal of exhaust gas of the ammunition. A coffered ceiling, consisting of crossing baffle plates, meets these requirements. Experimental data for ceilings with different size parameters show that excess attenuation in residential areas at distances from 1.0 to 2.0 km of more than 15 dB(A) could be achieved with acceptable efforts.
Automatic measurement of intonation.
Authors: Spaai Gerard W. G. , Storm Arent , Hermes Dik J.
Published: October 2005

If speech intonation is only represented by an unprocessed series of pitch measurements, the interpretation can be hampered by three factors. First, because of the presence of unvoiced parts in an utterance, the continuously perceived pitch contour is disturbed by interruptions. Second, in many cases, speech is characterized by involuntary pitch perturbations that either cannot be heard at all, or do not contribute to the perception of intonation. Third, the perceptual meaning of a pitch movement depends upon its position within the syllable, in many cases with respect to the vowel onset. A correct interpretation of a pitch contour, therefore, requires the position of the vowel onsets to be known, too. These problems can be solved by interpolating the pitch at unvoiced parts from the adjacent pitch measurements, by removing all perceptually irrelevant details from the contour, and by indicating the vowel onsets. Besides presenting the procedures, a system will be presented which performs these tasks in real time, and which is currently used in an evaluation of its usefulness in teaching intonation to deaf persons. The applicability for the use of this intonation meter in training intonation of foreign languages will be indicated. [Work supported by Instituut voor Doven, St.–Michielsgestel, Netherlands.]
A new technique for formant extraction by adaptive smoothing of Fourier spectra.
Authors: Chari Venkatesh R. , Espy-Wilson Carol Y.
Published: October 2005

A new technique to automatically track formant frequencies in voiced speech was developed. Continuity constraints were imposed by adaptively smoothing the short-time Fourier representation. At each point in the time-frequency field, the average energy distribution within a linear window was computed as a function of angular displacement. The angle at which the energy density was closest in magnitude to the point under consideration was chosen as the most likely direction of energy correlation. Weighted smoothing was then performed in that direction. This allowed potential candidate peaks to be enhanced if they lay on a formant track but were not robust enough to be picked on their own merit. Continuous and distinct peak tracks were thus generated even in cases where formants were close. Higher level speech knowledge sources were then called upon to assign peak tracks to formant slots. A measure of reliability was also estimated for each assignment. Experiments conducted on sentences from the TIMIT database yielded very encouraging results. Results will be presented. [Work supported by NSF.]
Temporal arrival structure from matched-filter processing of an LFM signal across two convergence zones using Difar sonobuoys.
Authors: Lawrence Thomas N. , Holub Kirk L. , Bedford Nancy R.
Published: October 2005

An experiment conducted near the Nares Abyssal Plain broadcast LFM and cw signals to investigate propagation across two convergence zones (CZ). Air-launched Difar sonobuoys were placed in the first and second CZs simultaneously, and cw bursts of about 1000 Hz, and LFMs covering 900 to 1100 Hz were recorded. A time-delayed echo repeater, which rebroadcast received signals omnidirectionally, was located in the first CZ. These signals were recorded via an expanded videocassette recording system [S. G. Payne, N. R. Bedford, and L. A. Thompson, ‘‘The AEAS Poco Amigo Airborne Acoustic Measurement System,’’ ARL-TR-88-28], which demultiplexes signals from each Difar sonobuoy into its omnidirectional, north–south, and east–west components. Matched-filter processing was applied to these data to obtain a temporal arrival structure. In the case of the second CZ, all three components of the Difar buoy were separately recorded thus allowing horizontal directionality via beamforming. Using calibration information from directional hydrophones, vertical arrival angles were determined [M. B. Bennett, L. A. Thompson, and P. T. Eisman, ‘‘Moored Data Buoy for Shallow Water Acoustic Studies’’ (in preparation)]. The arrival structure of the primary source and the echo repeater will be contrasted. [This work is supported by the ASW Environmental Acoustic Support Program.]
Speech feature perception in prelingual deaf children with cochlear implants.
Authors: Tyler Richard S. , Fryauf-Bertschy Holly , Kelsay Danielle , Gantz Bruce
Published: October 2005

The speech-perception abilities of 15 prelingual deaf children, using the Nucleus 21-channel cochlear implant, were examined with a new audiovisual speech-feature test for young children. The test stimuli include the letters ‘‘b, d, c, p, t, v, z’’ and pictures of ‘‘me,’’ ‘‘knee,’’ and ‘‘key.’’ Preliminary test retest data suggest good reliability for a 60-item test. Performance is significantly correlated with a closed-set picture test for young children. A feature analysis indicated that the voicing and envelope features were understood moderately well when presented by audition, and the place feature was understood well when presented by vision. [Work supported by NIH.]
Speech and hearing applications of LabVIEWR
Authors: Espinoza-Varas B. , Shim Chongjoon , Venema Theodore H.
Published: October 2005

A system for research, teaching, and clinical applications in speech and hearing is described. The system utilizes LabVIEWR software, National InstrumentsR expansion boards (D/A converter, DIO and DMA boards), and a MacintoshR IIFX computer. LabVIEW is an integrated package for data acquisition, digital signal processing, instrument and process control, data analysis and graphics. The package: (a) employs an icon-based language that does not require writing of code; (b) contains a comprehensive library of modular, pre-programed functions (e.g., A/D conversion, digital filtering, FFT analysis, etc.) callable by displaying icons to the screen; (c) links modules with a graphical ‘‘wiring’’ utility; and (d) is intuitively simple to use. The system is being used for signal generation, control of psychoacoustic and speech-perception experiments, data analysis, and signal processing. A package of hearing-science laboratory demonstrations has been implemented also. [Work supported by OCAST Grant No. HSO-005 and Presbyterian Health Foundation.]
Acoustic scattering from fluid-loaded plates with arbitrary boundary conditions.
Authors: Kaina R. , Cuschieri J. M.
Published: October 2005

The dynamic response and acoustic scattering from a finite, fluid-loaded, rectangular plate, simply supported on two edges and with arbitrary boundary conditions on the remaining two edges are investigated. An analytic solution of the equation of motion of the plate is obtained by using a modal decomposition in the direction perpendicular to the simply supported edges and by using a spatial transform into the wave-number domain for the remaining direction. Since the plate is finite, when taking the spatial transform, four response-related parameters are obtained in the result of the transform. From knowledge of the boundary conditions a set of simultaneous equations can be set up and solved to evaluate these parameters. By obtaining the solution in the wave-number domain, the need for assuming a known mode shape is eliminated. Using this approach for solving the fluid–structure coupling problem, arbitrary boundary conditions can be considered. In this paper, the above approach is used to solve for the input and scattered acoustical power from two edge-coupled plates with a discontinuity at the common junction. [Work sponsored by ONR.]
Transient analytic and numerical results for the fluid–solid interaction of prolate spheroidal shells.
Authors: Jones-Oliveira Janet B.
Published: October 2005

A transient solution is presented that models the fluid–solid interaction of a thin elastic prolate spheroidal shell loaded end-on by a nonconservative acoustic shock wave. Solutions to the Lagrangian equations of motion are provided for the normal and tangential shell displacement fields, as well as for the incident, scattered, and radiated fluid pressure fields. The temporally and spatially dependent shell displacement and fluid pressure fields are expressed modally in terms of an orthonormal basis that is selected specifically to satisfy the kinematic boundary conditions at the fluid–solid interface identically. Further, the modal representations for the shell displacements resolve the boundary conditions imposed by the problem configuration; and the modal representations for the fluid field satisfy causality. The solutions are developed in terms of prolate spheroidal angular and radial wave functions. For both analytic and computational reasons, the shell displacement and fluid pressure fields are ultimately expressed in terms of Legendre polynomials and modified spherical Bessel functions of the first and third kinds. The resulting solutions are exact within the limit of series solutions. A novel analytic/numerical hybrid solution technique is introduced for solving the inhomogeneous coupled PDE’s; the spectral method is a Galerkin variation. The explicit analytic solutions converge to the exact solution of the actual coupled differential equations over the entire temporal and spatial domains both in the structure and in the fluid.

Series truncation is postponed until the final expressions and their mathematical structure are known, thereby ensuring that all relevant physics has been correctly modeled having taken full advantage of inherent symmetries. Numerical results for the free vibrations and for the transient fluid–solid interactions of a fluid-loaded prolate spheroidal shell are presented. Specifically, the axisymmetric in-vacuo nondimensionalized frequencies are tabulated for three characteristic cases corresponding to low, intermediate, and high aspect ratio geometries. The fluid loading is shown to shift the frequencies, as well as introduce additional structural frequencies. Plots of the shell displacements, velocities, accelerations, strains, and strain rates are presented for the submerged shock-loaded low aspect ratio case. Insights into the qualitative and quantitative effects of the fluid on the structural response are revealed. DOE-macsyma was used extensively to develop and verify the solutions to this problem. [This work was performed under the auspices of DOE by LLNL under Contract No. W-7405-Eng-48.]
Lattice hidden Markov models for speech recognition.
Authors: Chang Lu , Bayoumi M. M.
Published: October 2005

The hidden Markov model (HMM) has been successfully applied to the area of automatic speech recognition. However, HMM has two assumptions, namely, Markovian transition of its underlying states and time-independent emissions of observations. As is widely known, speech signals are highly correlated. And for speech recognition, temporal information is very important. A lattice HMM (LHMM) is proposed to address the difficulty of conventional HMM. The new model has two sequences, a hidden state sequence and an observable group sequence. Both sequences are assumed to have a Markovian property. However, the transitions of these two sequences are dependent on each other. The observations of the model are functions of the state and the group sequence. It should be noted that although the group sequence is observable, it is represented as a transformation of the observation sequence. The re-estimation formula for the model will be presented. The training of LHMM needs some care. Segmented data are clustered using DTW to find the typical group transition paths. The group transition matrix is estimated from the transition paths. This procedure is devised to increase the discrimination power of the model.
Feature extraction from a cochlear model representation: A novel supervised/unsupervised neural network hybrid for speech recognition.
Authors: Tajchman Gary N. , Intrator Nathan
Published: October 2005

A novel supervised/unsupervised hybrid neural network training algorithm [Intrator, ‘‘Combining exploratory projection pursuit with projection pursuit regression,’’ preprint (1992)] is applied to a speech recognition task. The input representation for the neural network is produced by Richard Lyon’s cochlear model as implemented by Slaney [M. Slaney, ‘‘Lyon’s Cochlear Model,’’ Apple Tech. Report ♯13, Apple Comput., Inc., Cupertino, CA 95014]. This detailed, high-dimensional representation is projected through the units of the hidden layer of a back-propagation-like architecture into a much lower dimensional space (spanned by the number of hidden units). This space is constructed by optimizing a combination of the familiar MSE minimization and an unsupervised measure of the goodness of the projections. The goodness measure is based on the multimodality of the projected distributions. Both constraints are powerful and useful when used alone. However, when they are combined, the resulting network has the potential to learn and generalize much more robustly than either alone. This technique was applied to the classification of 16 classes of stressed vowels extracted from the TIMIT Acoustic-Phonetic Continuous Speech Corpus [NIST Speech Disc 1-1.1 (October 1990)]. Comparisons between the hybrid method and plain back propagation are discussed.
Efficient representation of LPC excitation using nonuniform frequency-domain sampling.
Authors: Gupta Sunil K. , Atal Bishnu S.
Published: October 2005

An efficient representation of the LPC excitation is essential in predictive coding systems for synthesizing high-quality speech at low bit rates. In this paper, a method is presented that takes advantage of the nonuniform spacing of auditory critical bands to achieve an efficient frequency-domain representation of LPC excitation. A segment of LPC excitation with a duration of N samples, represented as a Fourier series, requires N/2 sinusoidal components uniformly spaced along the frequency axis for exact reproduction of the excitation. Thus, for speech bandlimited to 4 kHz, a 10-ms segment requires 40 frequency components for exact reproduction. It was found that, by using uniform frequency spacing below 1 kHz and logarithmic spacing above 1 kHz, the number of sinusoidal components can be reduced to 15 without introducing any audible distortion in the synthetic speech signal. Subjective tests were conducted to determine the effective signal-to-noise ratio of synthetic speech for different numbers of sinusoidal components. These results and a tape demonstration of synthetic speech will be presented at the meeting.
Control of Klatt speech synthesizer with high-level parameters.
Authors: Bickley Corine A. , Stevens Kenneth N. , Williams David R.
Published: October 2005

This paper describes a system for synthesis of speech with a terminal analog synthesizer like the Klatt synthesizer. The control of the synthesizer is accomplished with a small number (about 10) of high-level (HL) parameters that capture directly certain attributes of the state of the vocal tract during speech production. The multiplicity of low-level parameters in the synthesizer are derived from the HL parameters through a set of mapping relations. The HL parameters consist of (1) the fundamental frequency; (2) a specification of the vocal tract in terms of the first three or four natural frequencies of the tract assuming no opening to the glottis or the nose; (3) the areas of three major orifices in the vocal tract: the glottal opening, the velopharyngeal opening, and (for consonants) the principal constriction in the vocal tract; (4) a measure of active expansion or contraction of the vocal-tract volume for obstruents; (5) a measure of the efficiency of turbulence noise generation for obstruents. The mapping relations for obstruents involve the calculation of airflows, pressures, wall displacements, glottal waveform characteristics, vocal-tract losses, and noise due to turbulence. Some examples of the mapping relations for both obstruents and sonorants are given. [Research supported by the National Institutes of Health.]
Examples of speech synthesis with high-level parameters controlling a Klatt synthesizer.
Authors: Williams David R. , Bickley Corine A.
Published: October 2005

This paper describes the synthesis of a number of simple syllables with a Klatt synthesizer that is augmented to include the capability of controlling ten high-level (HL) parameters. These HL parameters are mapped into a much larger number of low-level (LL) parameters in a way that incorporates constraints on these parameters imposed by the acoustics and aerodynamics of the speech production system. Examples are presented to show that the synthesis of a pair of utterances that differ in one feature (such as /pa, ba/, or /ba, ma/, or /ba, da/) can be realized by keeping most HL parameters the same and changing only one or two parameters (such as glottal opening, velopharyngeal opening, or trajectories of one or two formants). Although a complex pattern of LL parameters with precise temporal synchrony is required to synthesize these utterances, the pattern of HL parameters is relatively simple, and the timing relations are often rather lax. Comparisons of the parameter sets needed to synthesize a variety of consonants reveal how HL synthesis captures generalizations about speech sound classes that traditional acoustic synthesis does not. [Research supported by the National Institutes of Health.]
Influence of an internal reference system and cross-modality matching the subjective rating of speech synthesizers.
Authors: Pavlovic Chaslav V. , Rossi Mario , Espesser Robert
Published: October 2005

In previous studies it was concluded that contextual invariance and subject invariance of categorical and magnitude estimates of speech quality could be improved by introducing a reference system and by normalizing the results with respect to it. The reference signal used in the previous studies was natural speech. The use of such a reference system may present problems for applications where cross-language comparisons of synthesizers are made. In particular, this refers to the difficulty of ensuring equal subjective quality of different talkers in different languages. In this study the possibility of substituting an actual reference signal with an ‘‘internal’’ reference defined to the subject as the system of optimal quality is investigated. Another objective of this study is to explore whether a sometimes difficult task of free number production required in magnitude estimations could be replaced by cross-modality matches using lines of various lengths produced by subjects on a computer screen. The main concern here was related to the unknown effects of the limited width of the computer screen on the magnitude estimation task. [This research was made possible by grant No. 2589 from the EEC Espirit SAM project.] a) Also at Univ. of Iowa City, IA.
An unsupervised method for learning to track tongue position from an acoustic signal.
Authors: Hogden John , Rubin Philip , Saltzman Elliot
Published: October 2005

A procedure for learning to recover the relative positions of the articulators from speech signals is demonstrated. The algorithm learns without supervision, that is, it does not require information about which articulator configurations created the acoustic signals in the training set. The procedure consists of vector quantizing short time windows of a speech signal, then using multidimensional scaling to represent quantization codes that were temporally close in the encoded speech signal by nearby points in a continuitymap. Since temporally close sounds must have been produced by similar articulator configurations, sounds which were produced by similar articulator positions should be represented close to each other in the continuity map. Using an articulatory speech synthesizer to produce acoustic signals from known articulator positions, relative articulator positions were estimated from synthesized acoustic signals and compared to the synthesizer’s actual articulator positions. High rank-order correlations, ranging from 0.92 to 0.99, were found between the estimated and actual articulator positions. Reasonable estimates of relative articulator positions were made using 32 categories of sound, and the accuracy improved when more sound categories were used.
Description of contextual factors affecting vowel duration.
Authors: van Santen Jan P. H.
Published: October 2005

As an initial phase of a project on duration models for predicting segmental durations from contextual factors, two natural speech databases produced by a male and a female speaker with more than 50 000 manually measured segmental durations were analyzed. This large quantity of data made it possible to perform a detailed analysis of the effects of several contextual factors, including lexical stress, word accent, the identities of adjacent segments, the syllabic structure of a word, and proximity to a syntactic boundary. Among the key results were the following. (1) The contextual factors accounted for up to 90% of the variance, and reduced the within-vowel standard deviation by a factor of 3. (2) There were complex interactions between factors, in particular between boundary proximity and postvocalic consonant identity and between lexical stress and syllabic word structure. (3) The effects of adjacent segments were reducible to the effects of voicing and manner of production; effects of place of articulation were negligible. (4) Proximity to a boundary should be measured in terms of syllabic and segmental position, not in terms of the sum of the intrinsic durations of segments between the target and the boundary. The results were compared with data reported by Crystal and House [J. Acoust. Soc. Am. 83, 1551–1573 and 1574–1585 (1988)].
Level discrimination of stop-consonant noise bursts by normal-hearing and hearing-impaired listeners.
Authors: Brown Cynthia D. , Espinoza-Varas Blas
Published: October 2005

Level discrimination for the consonant noise bursts of the word ‘‘pack’’ was measured in normal-hearing and hearing-impaired listeners. The utterance, produced by a female talker, was digitized at 20 kHz, edited and output via a D/A converter. Using a constant-stimuli 2IFC procedure, listeners had to discriminate level increments ranging from 0.5–7.0 dB in noise bursts either ‘‘in isolation’’ (i.e., excerpted from the word) or in word context. The sensorineural impaired ears had fairly uniform losses ranging from 30–70 dB. In all conditions, burst presentation level was 59–60 dB SPL for normal ears, and 25–30 dB above the PTA for impaired ears (i.e., most of the burst’s spectra were clearly above the audibility thresholds). Discrimination thresholds [P(c)=75] were interpolated from probits fitted to nine-point psychometric functions. In normal ears, thresholds were 1.5–2.0 dB for both the /p/ and /k/ bursts presented ‘‘in isolation.’’ Thresholds ‘‘in context’’ were ∼2.5 dB for /k/ and 3.0–4.0 dB for /p/. Some impaired ears showed discrimination thresholds similar to those of normal ears. [Work supported by OCAST Grant No. HSO-005 and Presbyterian Health Foundation.]
Measurement of cavitation thresholds for human blood.
Authors: Xu Qihong , Holland Christy K. , Apfel Robert E.
Published: October 2005

To gauge the likelihood of producing transient microcavitation invivo with medical ultrasonic equipment, several cavitation experiments were conducted with bank blood using acoustic signals that are similar to those deployed by typical medical diagnostic ultrasound. Using a specially designed chamber, filled with bank blood samples having different volume concentrations of red blood cells (RBC), the cavitation threshold of the blood and the backscattering signal from the blood were measured with an active cavitation detector [Roy etal., J. Acoust. Soc. Am. 87, 2451–2458 (1990)] with the frequency centered at 30 MHz. The results show that the acoustic pressure causing microcavitation in blood is in the range of the pressure generated by some medical diagnostic devices, the cavitation threshold of the blood varies linearly with the volume concentration of the RBC, and the noise level due to backscatters is inversely proportional to the volume concentration of the RBC. [Work supported by NIH Grant No. 5RO1CA39374.]
Respiratory control during singing.
Authors: Hixon Thomas J. , Watson Peter J.
Published: October 2005

Singing presents special control tasks for the respiratory apparatus. These tasks are performed variously by singers and have been studied through the use of several technologies. This paper will discuss data gathered over the past decade in this laboratory on the nature of respiratory function during singing and interpret those data in terms of contemporary theory about neuromotor and biomechanical features of respiratory control. As well, attention will be devoted to respiratory control issues surrounding the development of singing skill and the potential to reorganize singing performance under selected biomechanical constraints. [Work supported by the National Institute on Deafness and Other Communication Disorders.]
Acoustics of the tenor high voice.
Authors: Titze Ingo R. , Mapes Sharon
Published: October 2005

Frequency spectra of five standard vowels sung at high pitches (F4, G4, A4, and B4) were obtained from six tenors who were in vocal training at Westminster Choir College. Because of the variation of the spectra over the vibrato cycle, measurements were made at the peak, the trough, and the middle of the cycle. Stylized power spectra were generated from averages over the vibrato cycle. A power spectrum model was then used to determine the extent to which tenors ‘‘tune’’ formants to specific harmonics of the source. Results indicate that, aside from the high level of acoustic energy in the region of the singer’s formant, tenors get most of their vocal intensity from the second, third, and fourth harmonics. In some cases, F1 or F2 is specifically tuned to one of these lower harmonics; in other cases, however, the desire to create smooth transitions between vowels seems to favor a distribution of energy across several harmonics. Unlike what has been observed in the soprano high voice, the fundamental almost never carries the greatest amount of energy.
Aperiodicities in the singer voice source.
Authors: Cook Perry R.
Published: October 2005

Aperiodicities in the glottal source waveforms of singers were investigated. Fast, interactive signal processing tools were developed for investigating aperiodicity. The expected deviations at the instant of glottal closure were found, along with two other commonly occurring types of disturbance. Pulses of noisy disturbance near the moment of glottal opening were often observable in low male tones. It was hypothesized that this noise production is caused by local turbulence near the glottal source which is modulated by the source flow wave itself. Analysis results for several bass singers producing multiple vowels across the vocal range were consistent with the dependence of the turbulence modulation on frequency and tube diameter downstream from the source. Subharmonic generation, often called diplophonia and considered a vocal pathology, is quite common and natural in normal singer voices. Subharmonic generation was investigated in 20 singers, across four octaves, three vowels, and three dynamic levels. Each subject exhibited the presence of measurable subharmonics on at least one produced tone, with over one third of all produced tones containing a second subharmonic component.
A parametric study on pitch of short vocal vibrato tones.
Authors: d’Alessandro Christophe , Castellengo Michèle
Published: October 2005

The pitch perceived for short vocal vibrato tones was measured using a method of adjustment. The stimuli were synthetic vocal tones, produced by a formant synthesizer. The main parameter under study was the tone duration, as a function of the fractional number of vibrato cycles. This parameter was examined in relation to: (1) the vibrato extent (0, 50, 100, and 200 cents); (2) the vibrato rate (4, 6, and 8 Hz); (3) the tone frequency (220, 440, 880, and 1500 Hz). Durations from 1/2 cycle to 5 cycles were studied. Part of the results were obtained for a relatively large group of musically educated subjects (20 subjects), and another part for a small group of selected subjects. Our results show that: (a) for short tones, the pitch does correspond to a weighted time average of the F0 pattern (a numerical model of which is in accordance with this data); (b) the pitch mean between the extreme frequencies as the duration increases; (c) the overall pattern of F0 has an influence on perception, and some simple patterns seem to behave better perceptually; (d) perception may be ambiguous above a threshold of duration, which is related to the absolute threshold of pitch change and to the trill threshold.
Detection of scattered ambient noise by fish.
Authors: Lewis Thomas N. , Rogers Peter H.
Published: October 2005

It has been hypothesized that one role of the fish’s auditory system may be to detect and localize nearby fish by ‘‘imaging’’ ambient noise scattered by their swimbladders. This is analogous to the role of the visual system of most animals, where the relevant signal is ambient light scattered by objects rather than light emitted by luminous objects. A classical conditioning experiment has been performed which indicates that the fish auditory system is capable of functioning in this manner. The ambient noise was provided by a J-9 transducer driven by Gaussian noise. Scattering of ambient noise by the resonant swimbladder was simulated by applying a filtered version of the same Gaussian noise to a small spherical projector. The target strength and bandwidth of the scattered noise were controlled by the attenuation and passband of the filter. The fish was conditioned to respond to the presence of the signal from the spherical projector. Thresholds of simulated scattered noise as a function of range to the subject were measured for the species Carassiusauratus (common goldfish). The fish’s ability to discriminate this signal is taken to be a measure of its ability to detect scattered ambient noise. [Work supported by ONR.]