bluuebunny/crossref_metadata_2025 huggingface dataset:
Nondiffracting Bessel optics: Theory and practice
Authors: Eberly J. H.
Published: October 2005

The theoretical foundation for the nondiffracting Bessel beams identified in 1985 by Durnin will be explained briefly, and several practical realizations of these beams in optical contexts will be described.
Development of an acoustic array for generating ADEPT pulses in air
Authors: Poindexter Alan P. , Baker Bruce D. , Cook Bill D.
Published: October 2005

A six-ring circular array with each ring driven by separate electronics has been developed for the purposes of exploring acoustic directed energy pulse trains in air. Each array clement is a small loudspeaker with 1-in.-diam cone. A different signal is to be applied to each ring simultaneously. These signals come from an MACII computer with an off-the-shelf interface board. The signal must be compensated for the loudspeaker's system reponse and interaction between array elements. [This work is supported by Lawrence Livermore National Laboratory.]
Dynamic modulus measurements made with the dynamic mechanical thermal analyzer
Authors: Deigan R. J. , Dlubac J. J.
Published: October 2005

Polymer Laboratories' dynamic mechanical thermal analyzer (DTMA) measures the dynamic Young's or shear modulus and loss factor of viscoelastic materials. The DMTA infers the modulus and loss factor by measuring the response of a small sample to forced vibrations in a small material sample. An optimal sample geometry is selected for a given Young's modulus range by considering effects such as thermal stress, shear deflection in short beams, and system compliance. The goal is to minimize these combined errors in the dynamic Young's modulus and loss factor measurement over the full madulus range. As a result of this analysis recommended DMTA test procedures are presented.
Dynamic finite element analysis of a compressionally loaded viscoelastic cylinder
Authors: Harrison R. W.
Published: October 2005

A nonlinear finite element program utilizing the Mooney-Rivlin strain-energy tensor was used to obtain the static deflection of a compressionally loaded viscoelastic cylinder. The deformed coordinates and the mass and stress tensors were then passed to a complementary dynamic finite element program where the vibration isolation as a function of the applied vibrational frequency and the impedance of the top surface were calculated. The predictions of the finite element programs were verified with both Instron and accelerometer data. The calculations show the effect of compressional loading on the vibration absorption response and indicate possible geometrical modifications to enhance this absorption for application in the theory of motor mount isolation.
Prediction of the loss factor of a double pipe damping system
Authors: Li Meng , White Robert G.
Published: October 2005

It is well known that increasing the damping of machinery structures will reduce noise radiation due to ringing vibration. Damping in industrial pipes can be increased by various damping techniques. This study was motivated by the need to decrease the noise radiation and vibration of pipework in power plants, particularly at elevated temperatures. Based upon the physical phenomenon of the air film damping of two parallel plates, a double pipe damping system with a very small air gap was investigated. In analysis, the Flugge shell equations of motion and the Navier-Stokes equation for viscous fluid were employed. First, a traveling wave-type solution was taken for shell and fluid. Then, from the interface conditions between the shell and fluid, the solution for the fluid medium was expressed in terms of shell displacements. Finally, solving the fluid equation of motion gave the fluid velocity profiles and stresses in the clearance between two cylindrical shells. From the definition of energy dissipated in the fluid, the equation of loss factor of the whole system was derived. The analytical results showed a very good qualitative agreement with the experiment. The study showed that different shell modes had a different contribution to the squeezing film damping, and the optimal design of the double pipe damping system can be deduced. The theory has been extended to studying the squeezing film damping system with various kinds of viscous fluid, such as lubrication oil, etc.
An adaptive two-dimensional algorithm for solving finite-difference implementations of parabolic equations
Authors: Robertson J. S. , Arney D. C. , Jacobson M. J. , Siegmann W. L.
Published: October 2005

In previous work algorithms have been developed for enhancing the efficient solution of finite-difference implementations of the parabolic approximation that adaptively adjust range-step size [J. Acoust. Soc. Am. 86, 252–260 (1989)] and depth-step size [Proc. Second IMACS Symp. Comput. Acoust. (to appear)]. Here, an algorithm is presented that integrates previous work, i.e., both range- and depth-step sizes are adaptively selected by a procedure within the implicit finite-difference (IFD) implementation of the parabolic approximation. An error indicator that includes components resulting from both range and depth discretization is computed at each range step, and its value is compared to a user-specified error tolerance window. If the error indicator falls outside this window, the range- and depth-step sizes are adjusted so that the error is equidistributed between both components. Further adjustments are not made until the indicator leaves the tolerance window. In this sense, the algorithm generates a two-dimensional grid that is optimal and that often leads to substantial decreases in run time. Examples are presented that illustrate the accuracy and efficiency of the algorithm. [Work supported by ONR.]
Full-wave bottom reverberation modeling
Authors: Tappert F. , Ryan F.
Published: October 2005

Scattering by roughness of the penetrable water-sediment interface is regarded as volume scattering and two coupled parabolic equations for outgoing and incoming waves are derived. These equations are solved using a multiple-forward-scatter/single-back-scatter approximation plus reciprocity. In this way, a novel full-wave, fully range-dependent, bottom reverberation model has been developed for long-range sub-kilohertz frequency applications. In contrast to all known existing reverberation models, scattering strength is not input as a function of vertical angle. Instead, a statistical model of seafloor roughness is used in which the local power spectrum is input together with range-dependent geophysical properties of the sediments and basement. The output is reverberation pressure level (RPL) as a function of range, bearing, and frequency. Examples of predicted RPL in the Canary Basin operational area will be shown. [Work supported by ONT.]
Manifestly reciprocal scattering amplitudes for rough interface scattering
Authors: Berman David H. , Dacol Dalcio K.
Published: October 2005

A new exact expression for scattering amplitudes for rough interface scattering is presented. This expression is explicitly reciprocal and it is shown to hold for a variety of boundary conditions: Dirichlet (pressure release), Neumann (rigid surface), impedance, fluid-fluid, and fluid-solid interfaces. This expression is shown to be a convenient starting point for deriving approximations that also respect reciprocity. Examples include a small-slope approximation, a reciprocal phase perturbation approximation, and a reciprocal smoothing approximation. [Work supported by ONR.]
A numerical treatment of the fluid/elastic interface for the potential equations
Authors: Knightly G. H. , Li G.-Q. , St Mary D. F. , Lee D.
Published: October 2005

Recently Shang and Lee [J. Acoust. Soc. Am. 85, 654–660 (1989)] carried out an investigation of the fluid/elastic interface boundary. The development was done in such a manner as to provide consistency between the standard parabolic approximation in the fluid medium and the potential equations for the elastic wave equations in the elastic medium. The development is designed in such a manner as to allow the use of existing implicit finite-difference marching schemes in the liquid medium. Here, an enhancement of the interface treatment is presented, stability of the schemes is discussed, and numerical comparisons are made with a known solution having an interface. [Work supported in part by ONR.]
Underwater acoustic propagation simulation using a 3-D sea-surface scattering model and a 2-D propagation model
Authors: Keiffer Richard S. , Norton Guy V. , King David B. , Chin-Bing Stanley A. , Novarini Jorge C.
Published: October 2005

A three-dimensional time-domain sea-surface scattering model based on the wedge assemblage method has been incorporated into a two-dimensional frequency-domain propagation model based on the parabolic equation (PE) method. The sea-surface scattering model computes the time-domain scattering acoustic field using multiple realizations of sea-surface spectra to give a realistic stochastic 3-D sea-surface scattering prediction. The interfacing of the two models is achieved by obtaining the output from the sea-surface scattering model as a function of mode propagation angle and redistributing the scattered field into the proper modes in the propagation model. This new field is then marched forward by the PE model until the next significant sea-surface scattering event occurs where the interfacing method is again employed. Examples will be presented using various interfacing schemes.
A connectionist model for classifying speech into silence, glottal source, burst friction, or mixed categories
Authors: Sadoff Steven J.
Published: October 2005

An algorithm for classifying speech into four classes (silence, only glottal source, only burst friction, or mixed) is being developed. This scheme primarily differs from the standard silence/aperiodic/periodic classification in that the defining characteristics of glottal source and burst friction sounds do not depend on periodicity distinctions, but on the locus of the energy concentrations. One male speaker reciting the Rainbow passage has been recorded and analyzed. Utilizing a strictly layered backpropagation network, the automated learning procedure is trained using the first 30 s of the passage; the final 75 s are used for testing. Quantitative results, along with several examples illustrating this model, will be presented. Additionally, the performance and classification strategy of the network will be compared to that of human observers. [Work supported by AFOSR.]
Context-modulated discrimination of similar vowels using second-order connectionist networks
Authors: Watrous Raymond
Published: October 2005

Discrimination of medial adjacent vowels in the context of voiced and unvoiced stop consonants using connectionist networks is investigated. Separate discrimination networks were generated for one speaker from samples of the vowel centers of [e,æ] for the six contexts [b,d,g,p,t,k]. A single context-independent network was similarly generated. The context-specific error rate was 1%, whereas the context-independent error was 9%. A method for merging isomorphic networks into a single network is described. The method uses singular value decomposition to find a minimal basis for the set of context-specific weight vectors. Context-dependent linear combinations of the basis vectors may then be computed using second-order network units. Compact networks can thus be obtained in which the vowel discrimination surfaces are modulated by the phonetic context. In a preliminary experiment, as the number of basis vectors was reduced from 6 to 3, the error rate increased from 1% to 3%. Experiments with nonlinear optimization of context-modulated, second-order networks for this task are underway and will be reported.
Automated language acquisition
Authors: Gorin A. L. , Levinson S. E.
Published: October 2005

A new approach to developing large-vocabulary speech understanding systems, in which the system automatically acquires a language model for its task based on semantic information, is introduced. This is in contrast to previous work, in which language was preprogrammed. An important consequence of this approach is that it leads to habitable language models. To accomplish this task, use is made of a medium-grain neural network, together with a novel adaptive training procedure for estimating the set of connection weights. The resulting connection weights have an information-theoretic interpretation, and do not require gradient search techniques for their estimation. A conversational-mode system that serves as a test bed for the network is described. The application scenario is inward-call management, where a customer telephones a large organization and encounters an operator whose function is to forward the call to one of several departments. The network's performance is first experimentally evaluated on three text databases: sentences collected from the inward-call management system, transcriptions of customer responses to an automated call transaction system (ACTR), and a collection of idiomatic phrases connoting yes and no. Finally, preliminary results evaluating the inward-call management system using speech data comprising unconstrained sequences of isolated words, which are phonetically transcribed via an ergodic HMM and provided as noisy input to the network, are reported.
Minimum mean-square error speech enhancement using hidden Markov models
Authors: Ephraim Yariv
Published: October 2005

A minimum mean-square error (MMSE) estimation approach for enhancing speech signals degraded by statistically independent additive noise is developed. This approach is motivated by the fact that optimal waveform coding in the MSE sense, and optimal autoregressive model vector quantization in the Itakura-Saito sense, given noisy speech, can be performed using MMSE estimators of the speech waveform and its sample spectrum, respectively. Estimators for the clean speech signal, and for the spectral magnitude and sample spectrum of that signal, are developed, based upon hidden Markov modeling of the clean signal and the noise process. The parameters of the models for the two processes are estimated from training sequences of clean speech and noise samples. It is shown that the MMSE estimator of the clean signal comprises a weighted sum of Weiner filters, one for each state of the hidden Markov model for the clean speech, where the weighting factors at each time instance are the probabilities of the filters to be the correct filters for the given noisy speech. The other MMSE estimators considered here have similar structure. At 10-dB input signal-to-noise ratio (SNR), the proposed approach provided an improvement of 4.5–5.5 dB in SNR.
Speech recognition using HMMs with an LVQ-trained codebook
Authors: Iwamida Hitoshi , Katagiri Shigeru , McDermott Erik
Published: October 2005

A new speech recognition algorithm using HMMs with a codebook trained by learning vector quantization (LVQ) is here described. Both HMMs and LVQ are stochastic algorithms holding considerable promise for speech recognition. In particular, HMMs have the significant advantage that phone models can easily be concatenated to produce a long utterance model, such as a word or sentence model. LVQ, on the other hand, is a powerful classifier, as shown in the high phoneme recognition rates obtained in McDermott and Katagiri (1989). The new algorithm described here combines the advantages inherent in each of these two algorithms. To evaluate this LVQ-HMM hybrid, phoneme recognition experiments were performed using the same data as used in TDNN (Waibel, 1988) and shift-tolerant LVQ (McDermott and Katagiri, 1989) experiments. Applied to various phoneme recognition tasks, the LVQ-HMM hybrid achieved recognition rates much higher than those of a conventional HMM with a codebook designed by K-means clustering. For small codebook sizes, the hybrid model realized a more than tenfold decrease in recognition error rates, yielding extremely high phoneme recognition performance, comparable to that of TDNN or shift-tolerant LVQ.
A new discriminative HMM training procedure
Authors: Huang Eng-Fong , Soong Frank K.
Published: October 2005

A training procedure is proposed for improving the discriminative power of a maximum likelihood (ML) hidden Markov model (HMM) without sacrificing its classification capability. The proposed discriminative HMM consists of a conventionally trained ML model and a discriminative model. The training data are utilized in two different modes. In the first mode, conventional ML models, denoted as master models, are trained. In the second mode, discriminative models, denoted as slave models, are trained by aligning training tokens of a certain word with all but the correct word master models and the model parameters are estimated by maximizing the conditional likelihood of the training tokens given the fact that they are aligned with incorrect-word master models. In recognition, the scores of the master models are reinforced by the scores of the input tokens, which are compared with corresponding slave models. A speaker-independent, 39-word, alpha-digit database was used to evaluate the new training procedure. Experimental results indicate that the new training can improve the recognition performance by 1%–2%. However, the discriminative power of the slave models decreases gradually when more sophisticated models and features are used.
Reflectivity of monolayers of bubblelike scatterers at water/solid interfaces
Authors: Tolstoy Ivan
Published: October 2005

By redistributing the energy flux of an incident plane wave, a monolayer of identical bubblelike scatterers at an interface may, at frequencies close to the monopole resonance ω0, drastically alter the reflectivity of the surface. Reflectivity calculations are given for a wave incident normally upon infinite square lattices (of basis l) for three models: (1) air bubbles in a liquid fullspace, (2) air bubbles at a water/hard surface or water/elastic plate interface, and (3) air-filled cavities in a rubber layer at such an interface. These illustrate the essential role of multiple scatter and, most interestingly, the possibility of reflectivity nulls for incident frequencies ω≃ω0. For case (3), use of rubber constants given by earlier writers [Gaunaurd et al., J. Acoust. Soc. Am. 65, 573–594 (1979)] shows that it is theoretically possible to select values of kl such that a hard wall (or an elastic plate) becomes fully anechoic for given ω≃ω0. For the rubber types considered, the effective bandwidth of this effect (99% anechoicity) Δω/ω0 varies between 0.01 and 0.05. [Work supported by ONR.]
Sound scattering from a submerged finite cylinder with ribs
Authors: Moser Philip J. , Yuan Jian-Ren , Überall Herbert
Published: October 2005

The effects of an almost-periodically spaced set of ribs on the scattering of sound from a submerged cylinder of finite length are investigated. The cylinder is either assumed soft (corresponding to a very thin air-filled shell) or rigid (corresponding to a heavy thick shell), and its scattering amplitude is obtained using the geometrical theory of diffraction. The rib scattering amplitude is found from curvature corrected results of the theory for a flat plate carrying a rib. Numerical results are shown for the case of soft cylinders; they exhibit the onset and emergence of rib effects over the background of scattering from the body of the cylinder.
An old cochlear frequency-position function compared to more recent data
Authors: Greenwood D. D.
Published: October 2005

Accurate cochlear frequency-position functions based on physiological data would assist interpretation of physiological and psychoacoustic data within and across species. In 1961, a family of almost-exponential functions was developed [D. D. Greenwood, J. Acoust. Sac. Am. 33, 1344–1356 (1961)] by integrating an exponential function fitted to a subset of critical bandwidth estimates. The integral fitted well cochlear data on frequency-place in human cadavers, and, with changes in constants, similar data on several other species—as well as behavioral-anatomical data on cats. Since 1961, mechanical data from dead human ears and various data from living animals (cat, guinea pig, chinchilla, monkey, gerbil) have appeared. It is shown that they are fitted by the same basic function. As in 1961, in both cadaver and living ears, the same slope parameter can be used if partition length is normalized to 1; i.e., slope is scaled if distance is expressed in physical units. Slope uniformity suggests that the cadaver data may well describe at least the function's slope in living human cochleas. To set the upper frequency limit requires only the empirical choice of a single other parameter. It is suggested that these cochlear frequency-position functions be used in plotting not only physiological data but also psychoacoustic data (and in modeling work with speech and other bioacoustic signals), since, if the cochlear data are correct, the functions are consequently independent of, and an appropriate means to examine, psychoacoustic data and assumptions. [Work supported by NSERC, Canada.]
Kurtosis measurements: Implications for noise exposure criteria
Authors: Hamernik R. P. , Hsueh K. D. , Ahroon W. A. , Davis R. I. , Turrentine G. A.
Published: October 2005

Two conventional measures of an industrial noise environment are the overall sound-pressure level (SPL) and the power spectrum. While these two measures are necessary, they may not be sufficient to describe a noise environment for the purposes of evaluating the hazard to hearing. This presentation will describe the results of an experiment designed to test the hypothesis that for equal SPL and power spectrum a high-kurtosis noise exposure is more hazardous to hearing than is a low-kurtosis noise. Two groups of chinchillas, with two animals per group, were exposed continuously for 5 days, to one of two classes of noise at a 90 dB SPL. The power spectrum of both noises was identical but the noises had different values of kurtosis. The results clearly showed that there was up to a 20 dB greater permanent hearing loss for the animals exposed to the high-kurtosis noise. Detailed results of asymptotic and permanent threshold shifts and sensory cell losses will be presented. These results would appear to be a clear indication that, even for moderate levels of exposure, energy is not a sufficient variable upon which to base noise exposure standards. [Work supported by NIOSH.]
Ultrasonic absorption in the magnetic superconducting system Er1−x\HoxRh4B4
Authors: Sun Keun J.
Published: October 2005

Ultrasonic attenuation measurements as a function of temperature at constant magnetic fields and as a function of magnetic field at constant temperatures in Er1−xHoxRh4B4 show interesting behavior at low temperatures. The enhanced attenuation observed in the superconducting state and the sharp change in attenuation at magnetic phase transition for the superconductors in this system are attributed to spin-phonon interaction. The peculiar dependence of attenuation on orientation of magnetic field with respect to sound wave propagation direction evidences indirectly the coexistence of magnetic order and superconductivity. A two-level model of phonon-phonon interaction is proposed to elucidate the occurrence of relaxation type attenuation maxima observed in the temperature-dependent attenuation curves of all the samples. Furthermore, the attenuation behavior at low temperatures in high magnetic fields may also manifest the occurrence of a new magnetoelastic coupling effect.
Ultrasonic attenuation measurements on the heavy fermion superconductor UPt3
Authors: Sarma B. K. , Levy M. , Schenstrom A. , Qian Y.-J. , Xu M.-F. , Adenwalla S. , Ketterson J. B. , Zhao Z. , Hinks D.
Published: October 2005

The superconducting pairing mechanism in the heavy fermion superconductor UPt3 is unconventional, as evidenced from any of its low-temperature properties. Recent ultrasonic absorption, heat capacity, and magnetization measurements show a multiplicity of phases in the H-T plane, suggesting an exotic order parameter. The experimental results will be reviewed. [Work supported by ONR and NSF.]
Relating macroscopic deformation to site symmetry changes around the Cr+++ in shocked ruby crystals
Authors: Gupta Y. M.
Published: October 2005

An important but very difficult problem in shock wave research is to understand and interpret time-resolved, continuum measurements in condensed materials in terms of changes at the atomic/molecular level. As a start toward this challenging problem, experimental and theoretical studies on understanding the luminescence R-line spectra in shocked ruby crystals have been carried out. Time-resolved experiments have been performed to obtain wavelength shifts under macroscopically, well-defined compression and tension loading, and for shock wave propagation along the crystal c and a axes. Below the Hugoniot elastic limit (≈13 GPa), the observed R-line spectra are strongly anisotropic and show nonlinear shifts with density for both compression and tension. For the theoretical analysis, the deformation potential is written in terms of symmetry adapted basis according to the irreducible representation of local octahedral group. Using the symmetry of the macroscopic strain, the perturbative effects of the deformation potential are evaluated for the 2E state that gives rise to the R lines. Results not only from the shock experiments but also from hydrostatic and uniaxial stress experiments can be analyzed consistently using these analytic developments. A summary of the experimental and analytical work will be presented. [This work was carried out in collaboration with S. M. Sharma, X. A. Shen, P. D. Horn, and J. A. Butt and supported by ONR Contract N00014-86-0307.]
Language modeling for very-large-vocabulary speech recognition
Authors: O'Shaughnessy D. , Gupta V. , Lennig M. , Seitz F. , Mermelstein P.
Published: October 2005

Virtually all recognition systems, both research and commercial, place severe limits on the vocabulary and syntactic structures a speaker may use. This recognition system is the most advanced in the world in terms of freedom from this constraint—it allows an essentially unrestricted English vocabulary and sentence structure—although it is currently required that each word be followed by a short pause. Many speech recognition systems make little use of syntactic and semantic information. A trigram language model is employed here that examines the statistical likelihood of each vocabulary word's occurrence as a function of the two preceding words. A global context-sensitive parser is also used which accepts, as input, a long sequence of word candidates. (The speaker is not required to pronounce punctuation marks.) Using rules of English syntax, the parser identifies many erroneous candidates and estimates locations to insert punctuation in the output text. The combined parser and trigram model have raised the accuracy of this system to 92% for ten speakers on an 86 000-word vocabulary. [Work supported by NSERC.]
A database demonstrating the sounds of the world's languages
Authors: Ladefoged Peter , Choi John
Published: October 2005

A database illustrating the full range of the sounds of the world's languages would be an enormous undertaking. Consequently, this paper is restricted to trying to provide illustrations of a large number of the less familiar sounds. Currently, samples of about 50 languages were obtained illustrating different airstream mechanisms (!Xóō clicks, Navaho ejectives, Igbo voiceless implosives, etc.), phonation types (Sindhi voiced aspirates, Bruu tense phonations, etc.), and places and manners of articulation (Agul epiglottal stops, Yeletnye bilabial trills, etc.). Each recording has been digitized at 22 000 Hz, with 8-bit samples, care being taken to maintain the full 48-dB signal-to-noise (S/N) ratio. (However, some of the original field recordings do not have this S/N ratio.) The database is organized as a 15-meg set of Macintosh Hypercard stacks. Clicking on a word, or a row, or column of words, reproduces the corresponding items, and makes it simple to repeat contrasting sounds. The entire stack is in the public domain, and is available (for Macintosh computers with Hypercard) at cost. [Work supported by NSF.]
LPC vector quantization based on critical-band analysis
Authors: Lin Daniel , McCarthy Brian M.
Published: October 2005

Critical-band spectrum is known to represent the redundancies of the human auditory system. Vector quantization constitutes a minimal redundant code set for a given signal source. In this study, these techniques are combined in a low-rate speech analysis/resynthesis system. An important design criterion for a perceptually based source coder is to define an appropriate mapping between the analysis codebook in critical-band domain and the synthesis codebook in LPC domain. Several methods for generating and searching the synthesis codebooks based on the perceptual distance criterion are proposed. Their performance with other LPC-based distance measures (e.g., log area, LSP, cepstral) is compared. These results indicate a substantial improvement in the quality of synthetic speech for the perception-based system due to smoother spectral trajectories in the low-frequency regions.
Evaluation of a glottal ARMA modeling scheme
Authors: Lobo Arthur P. , Ainsworth William A.
Published: October 2005

In many speech analysis/synthesis schemes, the source of excitation for voiced speech is a train of impulses. Although this is a mathematically sound scheme, there is no physiological basis for it. The quality of speech that has been attained due to the introduction of a dynamically varying source, e.g., a parametric source mode, multipulse excitation, etc., has been found to be better than that produced using impulse excitation. In this paper, a pitch synchronous glottal ARMA analysis/synthesis scheme is proposed. A parametric voice source model is discussed. The voice source and vocal tract parameters are simultaneously estimated. The AR and MA orders of the vocal tract model are variable and their values are decided every period depending on whether a resynthesis efficiency threshold is crossed during the analysis/synthesis procedure. This scheme is compared with two other schemes, viz., (1) closed phase LPC analysis/synthesis and (2) robust LPC analysis/synthesis. The superiority of the proposed scheme over the latter two methods is shown in terms of better formant/bandwidth tracking ability and efficiency of resynthesis.
Transducer effect on automatic speech recognizer
Authors: Chang Harry M.
Published: October 2005

It is generally recognized that the types of transducers used to input speech to automatic speech recognizers (ASR) will affect their performance. However, any ASR application over the telephone network implies the use of a large number of different telephones as input transducers to ASR. It is, therefore, very important to quantify the performance degradation due to transducer-to-transducer variation introduced in many potential ASR applications over the telephone network and identify the source of the degradation. To conduct this experiment, a database based on an 11-digit vocabulary was created from 20 native English speakers in 11 sessions in a 2-month period. The utterances were digitally recorded on two tracks simultaneously: one from a head-mounted Shure SM-10A dynamic microphone, and the other from one of 11 selected telephones that will be different in each session. A commerical speaker-dependent recognition system was tested with both the subdatabase MIC created from the Shure SM-10A microphone and PHONE from the 11 telephones. The quantitative analysis of the evaluation tests will be presented.
Modeling lexical stress in read and spontaneous speech
Authors: Polifroni Joseph H. , Rudnicky Alexander I.
Published: October 2005

Although prosodic information has long been thought important for speech recognition, few demonstrations exist of its effective use in recognition systems. Lexical stress information has been shown to improve recognition performance by allowing the differentiation of confusable words (e.g., Rudnicky and Li, DARPA Workshop on Speech Recogn., June 1988). In this study, lexical stress modeling for a spreadsheet system with significant number of confusable words (e.g., EIGHTY and EIGHTEEN) is examined. The models used here have been evaluated on both read and spontaneous speech. A database of over 400 spreadsheet and numeric utterances was available for training a (HMM-based) speaker-independent continuous-speech system with a 273-word vocabulary and language perplexity of about 51. Testing data used in this study were based on read utterances and data generated in a separate study examining the use of a spoken-language spreadsheet. This latter set includes: (a) a “spontaneous” set, composed of parsable utterances from a spreadsheet task; (b) a “read” set, consisting of the spontaneous sentences read by their original speakers. The use of lexical stress models was found to reduce the error rate for read speech by approximately 10%. A comparison with the spontaneous data will provide an insight into the nature of the improvement.
Time and frequency spectral derivative features for robust recognition of Lombard and noisy speech
Authors: Hanson Brian A. , Applebaum Ted H. , De Haan Gregory R.
Published: October 2005

Lombard speech approximates speech variations encountered in noisy environments. This paper addresses automatic, speaker-independent recognition of Lombard and noisy speech. Spectral information was derived from production-based (LP) or perceptually based LP analysis, and represented by cepstral or index-weighted cepstral (RPS, spectral frequency derivative) coefficients. Instantaneous, dynamic (first temporal derivative), and acceleration (second temporal derivative) features were then computed from one of the representations, and integrated during the likelihood calculation of a hidden Markov model recognizer. The recognizer was trained with normal speech to evaluate its robustness to speech variations. Strong interaction was found between the temporal derivative features, the spectral derivative, and the degree of smoothing in the analysis. Although the acceleration feature performed poorly by itself, when combined with other features it generally raised recognition rates for Lombard speech. This trend was more pronounced with the cepstral representation. Combining all three features gave the best Lombard speech recognition results: The perceptually based analysis with RPS coefficients and three features yielded 94% correct recognition, compared with 63% and 82% correct for standard cepstrum and RPS, respectively, when the instantaneous feature was used alone. Experiments were also done with additive noise. The results and implications for robust speech recognition are discussed.
On the use of speech recognition in articulation training
Authors: Ostendorf Mari , Gibbons Kerin , Gavett Elizabeth , Bankson Nicholas
Published: October 2005

Computer processing of speech has been used in many ways as an aid to clinicians in various types of speech therapy. In this work, use of speech recognition for detecting articulation errors, such as a lisp or a phoneme deletion, is investigated. The main features of this approach are (1) the use of a time-dependent score in template matching and (2) the adjustment of dynamic time-warping slope constraints according to the type of articulation error (deletions versus substitutions). The use of a time-dependent score is particularly important because it allows the clinician to target a particular portion of a word for correct pronunciation. The algorithm was implemented in near real time on a PC with a DSP processor. Speaker-dependent and -independent performance was investigated for simulated and actual articulation errors. [Work supported by Design Automation, Inc., through an SBIR grant for the National Institute for Deafness and Other Communications Disorders.]
Perception of synthetic vowels: A comparison of several classification schemes
Authors: Hawks John W.
Published: October 2005

Synthetic vowel tokens (1725) were randomly presented twice to eight naive speakers of Midwestern American English for classification as one of 12 vowel categories, /IY,IH,EH,EY,AE,AA,AH,AO,OW,UH, UW,ER/. Subjects rated the certainty of their responses on a scale from one (very unsure) to five (very sure). The vowels were synthesized in null context and utilized a male F0 contour. The frequency values of F1, F2, and F3 were selected such that the entire area where vowels may be represented in the auditory-perceputal space [J. D. Miller, J. Acoust. Soc. Am. 85, 2114–2134 (1989)] was equidistantly sampled. The results of this experiment support the view that F1 and F2 are the primary determinants in the perception of nonretroflex vowels. In addition, monophthongal versions of the diphthongs /EY/ and /OW/ may be distinctly classified. The results will be presented graphically as target zones constructed on the basis of the plurality identifications for each token. These target zones are abutting and nonoverlapping, and correctly classify 99.9% of the plurality judgments (less 48 tokens where ties occurred) and 75% of the 27 600 total judgments. Other vowel classification schemes will also be compared for their accuracy in classifying the results of this experiment, as well as results from other studies. [Work supported by NINCD.]
Including frequency-dependent losses in a time-domain, articulatory synthesizer
Authors: McGowan R. S.
Published: October 2005

The Webster horn equation describes acoustic wave propagation in tubes with cylindrical symmetry. This differential equation can be derived from the linearized equations of motion for air neglecting viscosity and heat conduction. A modified horn equation including viscous loss in the boundary-layer approximation can be derived. This modified equation includes a convolution integral over time; thus the computational problem is to find an approximation to the integral. First, the integral is discretized, and then a rational Padé approximation is used to approximate the resulting infinite sum. The frequency dependence of this approximation is examined and compared to the ideal frequency dependence. [Work supported by NIH grants HD-01994 and NS-13617 to Haskins Laboratories.]
Phonological rules for a semantics-to-speech system of Japanese: A rule application experiment
Authors: Sato Shigeru
Published: October 2005

In order to confirm validity of the phonological rules of Japanese and their application algorithms, a phonetic form generation experiment is performed in the phonological component of a semantics-to-speech system implemented in the computer. The rules are tiered according, first, to the nature of data they handle: segmental versus accentual, and, second, to their rule-applicational relevance to syntax: cyclic versus noncyclic. Successful integration of segmental and accentual phases of phonological processing is shown to be possible using the concept of recursive adjunction of a suffix to a stem. The rules were installed in the computer in a human-editable format, translated by a rule compiler into executable Lisp functions, tested, and found to be valid in actual realization of synthesized speech.
Robust ARMA analysis for the determination of voice source and vocal tract control parameters in speech synthesis
Authors: de Veth Johan , Boves Louis , van Golstein Brouwers Wim
Published: October 2005

A cascade six pole-pair and five zero-pair synthesizer has been developed as part of a text-to-speech conversion system for Dutch. Control information for this synthesizer is derived from a.o. measurements on natural speech. A pitch synchronous robust ARMA analysis technique was developed and applied to utterances produced by a number of adult male talkers. The resulting pole-zero parameters were separated into sets pertaining to the source of the vocal tract. The vocal tract parameters were corrected in those frames where the analysis method made occasional mistakes. Analysis-resynthesis of sentence material using the corrected vocal tract parameters to control the synthesizer driven by impulse and noise excitation yielded high-quality synthetic speech. The tract parameters were then used to inverse filter the speech, to obtain the source function, that was subsequently parametrized using the Liljencrants-Fant model. It is hoped that the speech quality will be improved by replacing the impulse excitation by the controllable source model.
Restoration of tracheoesophageal voice with LPC resynthesis
Authors: Qi Yingyong
Published: October 2005

Four vowels, [il, [æ], [a], and [u], and one diphthong [ou], produced by two male and two female tracheoesophageal speakers, were analyzed with the LPC autocorrelation method. The vowels were synthesized by replacing the original source with an impulse train. The fundamental frequency of the impulse train was 100 Hz for the male and 200 Hz for the female speakers. The results of an identification experiment indicated that both the vowel and the gender of the speaker can be better identified from the synthesized vowels than from the original ones. The possibility of improving the quality of tracheoesophageal speech and building prosthetic device with the LPC technique will be discussed.
Low-frequency acoustic tomography using matched field processing
Authors: Tolstoy I. A.
Published: October 2005

Can matched field processing be used to determine the sound-speed environment given a known source? Recent computations are very promising in spite of expected difficulties relating to the nonuniqueness of solutions. In the range-independent case, the essence of the approach is to find “the” sound-speed profile that will maximize the power seen at the array. This has resulted in a profile that is “close to” the true profile and found by means of an exhaustive search of the parameter space describing the family of potential profiles. The parameter space consists of the coefficients of the major empirical orthogonal functions (EOFs) computed from the covariance matrix of sound-speed profile samples. Subsurfaces of the parameter space are easily plotted and show well-defined, informative trends. Unfortunately, iterative techniques investigated so far have shown convergence difficulties in that they find local maxima which correspond to profiles which are not close to the true profile. Results will show the procedure sensitivity as a function of array length, source range, depth, and frequency. The key to success for the technique is the computation of “good” EOFs.
Environmental mismatching effects on source localization processing in mode space
Authors: Shang E. C. , Wang Y. Y.
Published: October 2005

The performance degradation of the matched field processing (MFP) caused by environmental mismatching has been an intriguing subject. A great deal of study has been done by numerical simulations. In this paper, this problem is studied analytically. Analytic investigation is important because it reveals some of the physical insights and provides guidelines for a “self-coherent” scheme [H. Bucker, J. Acoust. Soc. Am. Suppl. 1 84, S19 (1988)]. It has been illustrated that the source range information is associated with the modal phase difference and the source depth information is associated with the modal amplitude ratio when MFP is performed in mode space. Therefore, the modal wavenumber plays a key role in source localization. Mismatching effects can be investigated analytically by taking advantage of calculating the modal wavenumber error by modal perturbation theory. [Work supported by NOAA and NOSC.]
Effects of spectral smearing on speech perception
Authors: ter Keurs Mariken , Festen Joost M. , Plomp Reinier
Published: October 2005

Connected speech presented in quiet is highly redundant. For such a condition, the resolution of spectral contrasts by the ear seems to be much larger than would be required. This suggests that the ear's high selectivity in frequency is particularly important to understanding speech in the presence of interfering sounds. The effect of reduced frequency resolution on the speech-reception threshold (SRT) for sentences in noise was investigated for eight normal-hearing subjects by simulating an auditory system with variable frequency selectivity. Signal processing was performed by short-time fast Fourier transforms (FFT), reduction of contrast in the spectral envelope without affecting the harmonic structure, and overlapping additions to reconstruct a continuous signal. The Spectral envelope in the frequency region from 100 to 8000 Hz was smeared over fixed relative bandwidths of 1/8, 1/4, 1/2, 1, 2, 4, and 8 octaves. Results show that the SRT increases progressively as the spectral envelope is smeared over bandwidths exceeding the ear's critical bandwidth. In a second experiment phoneme, confusions as a result of three different degrees of spectral smearing are studied in nonsense CVC syllables.
Articulation index importance functions for contextual speech materials
Authors: Bell Theodore S. , Dirks Donald D. , Trine Timothy
Published: October 2005

The relative importance of one-third octave frequency bands toward the intelligibility of speech in various contexts was examined. Thirty-five young normal hearing adults heard sentences in which the final word (target stimulus) was either predictable (probability-high; PH) or unpredictable (probability-low; PL) from the context of the sentence. Sentences were presented at S/N's from − 8 to + 14 dB in a noise shaped to conform to the peak spectrum of the speech, and crossover frequencies as related to the articulation index (AI) were determined by successively high-pass and low-pass filtering the stimuli to bisect intelligibility. Results indicated only slight difference in 1/3 oct importance functions as the result of context (PH vs PL), although the crossover frequency was significantly different and showed no interaction attributable to the noise. The effect of context in these speech materials was related to differences in perceptual dynamic range more so than the frequency importance function. These results are contrasted to other recent studies, and methodological and theoretical aspects of parameter estimation in the AI model are discussed.
An algorithm for distinguishing between voiced stops and voiced fricatives
Authors: Weigelt LaDeana F. , Sadoff Steven J. , Miller James D.
Published: October 2005

An algorithm has previously been reported which distinguished voiceless stops from voiceless fricatives with a success rate of 96.8% [J. Acoust. Soc. Am. Suppl. 1 85, S56 (1989)]. With only slight modifications, this algorithm also makes the voiced stop/fricative distinction. Here, results are presented on the modified algorithm. The input signal is high-pass filtered (cutoff frequency of 125 Hz) and two measures of the resulting waveform are used: the rms energy and the derivative of rms energy over time (termed rate of rise, ROR). The ROR is used as the primary classifier while energy pulse duration and relative level are used to discard spurious, irrelevant peaks. Peaks in ROR are considered in order of magnitude for relevance to the stop/fricative distinction. The resulting algorithm was tested on 420 CVC tokens (three male speakers, three female speakers, three stops and an affricate [B,D,G,JH], four fricatives [Z,ZH,V,DH], and ten vowel contexts [IY,IH,EH,AE,AA,AH,AO, UH,UW,ER]) recorded in an anechoic chamber. Data from two male and two female speakers (280 tokens) were used as a training set, and the remaining data (140 tokens) were used as a test set. The overall success rate was 97.9%. [Work supported by AFOSR.]