bluuebunny/crossref_metadata_2025_split huggingface dataset:
Use of Lanczos vectors in fluid/structure interaction problems
Authors: Jeans R. , Mathews I. C. 
Published: October 2005

The goals of any numerical computational technique used for the solution of structural acoustics problems in the exterior infinite domain should be ones of accuracy with rapid convergence, robustness, and computational efficiency. A computer program has recently been developed to achieve each of these three goals. Accuracy and robustness in the numerical representation of the integral equations used to represent the infinite fluid was attained through the use of boundary element implementations of the surface Helmholtz integral equations. The computational efficiency was resolved through the use of Lanczos vectors to model the deformation characteristics of the structure. The difficulty hindering the use of the boundary element formulation has been in finding a suitable numerical representation of the hypersingular integral operator contained in the differentiated form of the Helmholtz integral equations. This difficulty has hampered the convergence, accuracy, and robustness of many previous numerical integral equation implementations used for the modeling of the infinite fluid field. The authors have recently developed both collocation and variational techniques to overcome the difficulties previously encountered in the numerical implementation of the hypersingular integral operator.
The Cauchy singularity present in the integral formulation is made numerically amenable through the use of tangential derivatives in both the collocation and variational techniques. The variational approach has the advantage that the resulting added fluid mass term is symmetric and combines efficiently with a finite element approximation of the structural elastic response. Several different strategies making use of the Lanczos vectors have been investigated. The first involved the use of Lanczos vectors solely to characterize the structural response. This reduced form of the structural dynamical matrix was then substituted back into a Burton and Miller formulation of the acoustic problem, from which the normal surface pressure was obtained. The second strategy investigated involved forming the complex Lanzcos vectors of the dynamical matrix formed from the addition of a symmetrical added fluid matrix to the structural mass matrix. The size of resultant matrix equation set solved at each frequency for this strategy is determined by the number of Lanczos vectors used.
Timing of lip and jaw movements in the production of vowel types /i, a, u/ and voicing contrast /p, b/
Authors: Sock Rudolph , Löfqvist Anders 
Published: October 2005

The temporal organization of lip and jaw movements was investigated for one subject, producing six pseudowords. The words, produced at two speech rates, normal and fast, were: /pipi/, /papa/, /pupu/, /bibi/, /baba/, /bubu/. Movement data from the upper and lower lips, and the jaw were obtained via an optoelectronic tracking device, together with an audio recording. Based on articulatory and acoustic events in the velocity and audio signals, respectively, four cycles were identified in the movement of each articulator, and two cycles in the acoustic waveform. The movement cycles are: (1) maximum vocalic velocity; (2) vocalic cycle onset; (3) maximum consonantal velocity; and (4) consonantal cycle onset. Velocity cycles were determined, as the interval between successive positive or negative peaks associated with the raising or lowering gesture in the production of either a vowel or a consonant. Onset cycles were defined as the recurrence of zero-crossings on velocity traces, reflecting the onset of a lowering or a raising gesture to produce a vowel or a consonant. The acoustic cycles are: (1) release cycle; and (2) closure cycle. Vocalic and consonantal phases were defined, within vocalic and consonantal movement cycles, respectively, as specific timing relations of articulator lowering or raising gestures, and within the appropriate acoustic cycle, as the interval that presents a stable formant structure for the vocalic phase, and as the obstruent portion, for the consonantal phase.
Gestural and acoustic phasing patterns obtained within the cycles suggest that: (1) the lower lip is an efficient articulator in distinguishing the front vowels /i, a/ from the back vowel /u/, in both consonantal contexts and in all four cycles; (2) the upper lip distinguishes all vowel types in both consonantal contexts in the vocalic cycle onset, and voicing contrast in the maximum consonantal velocity; (3) distinct phasing patterns in normal speech tend to merge with increased speech rate; (4) the jaw, although a carrier articulator, does not contribute actively to separating the phonetic tasks required in this study; (5) the closure cycle is the better of the two acoustic cycles in separating vocalic and consonantal classes. Results are analyzed and discussed in terms of cognitive tasks and emergent properties of the speech production system. [Work supported by the Fyssen Foundation and NIH.]
Statistical grammar inference
Authors: Schabes Yves 
Published: October 2005

Language can be talked, written, printed, or encoded in numerous different ways. As any form of communication, each of these codings can be analyzed statistically for the purpose of comparative judgment and predicting power. Early proposals of language models such as Markov models, N-gram models [C. E. Shannon, Bell Syst. Tech. J. 27(3), 379–423 (1948)] although efficient in practice, have been quickly refuted in theory since they are unable to capture long distance dependencies or to describe hierarchically the syntax of natural languages. Stochastic context-free grammar [T. Booth, in 10th Annual IEEE Symp. on Switching and Automata Theory (1969)] is a hierarchical model that assigns a probability to each context-free rewriting rule. However, none of such proposals perform as well as the simpler Markov models because of the difficulty of capturing lexical information [Jelinek etal., Tech. Rep. RC 16374 (72684), IBM, Yorktown Heights, NY 10598 (1990)] [K. Lari and S. J. Young, Comput. Speech Lang. 4, 35–56 (1990)] This is the case even if supervised training is used [F. Pereira and Y. Schabes, in ACL ’1992]. Stochastic lexicalized tree-adjoining grammar (SLTAG) has been recently suggested as the basis for training algorithm [Y. Schabes, in COLING ’1992]. The parameters of a SLTAG correspond to the probability of combining two structures each one associated with a word. This system reconciles abstract structure and concrete data in natural language processing since it is is lexically sensitive and yet hierarchical.
The need for a lexical and still hierarchical statistical language model is partially corroborated by preliminary experiments which show that SLTAG enable better statistical modeling than its hierarchical counterpart, stochastic context-free grammars. The experiments also show that SLTAG is capable of capturing lexical distributions as well as bigram models while maintaining the hierarchical nature of language.
Scaling the perceptual clarity of the presence or absence of target tones embedded in modulated noise maskers
Authors: Singh Punita G. , Bregman Albert S. 
Published: October 2005

Perceptual judgments of the presence or absence of tones embedded in modulated noise maskers were obtained using a six-point rating scale (1=clearly absent, 6=clearly present). The maskers comprised four simultaneous noise bands centered at 500, 1000, 1500, and 2000 Hz, with bandwidths of all bands either 12.5, 25, 50, or 100 Hz. The bands were derived either from the same noise source by filtering and transposition, or from different sources, and were subsequently multiplied by same or different (phase shifted) modulators. A sine tone at the center frequency of one of the noise bands was used as the signal. A single noise interval was preceded by a cue tone that was either present or absent in the noise at a fixed S/N≊−13 dB SPL. The cue and noise intervals were both 400 ms long, and separated by 1 s of silence. Rating scores from ten normally hearing listeners were generally higher for tones embedded in maskers with common envelopes, indicating less masking and improved perceptual clarity of the tones. Common origin of noise source did not lead to improved audibility of the tones. There was no significant effect of tone frequency. However, bandwidth affected judgments significantly. For the 12.5- and 25-Hz bandwidths, the difference between modulation conditions was reduced, while false alarm rate increased substantially. These narrow-band noises apparently sounded tonal even in the absence of the signal, thereby receiving higher ratings. For the 50- and 100-Hz-wide maskers, the effect of modulation pattern was enhanced, and there were fewer false alarms. Similarity of amplitude envelopes within a masker has been invoked as a simultaneous grouping cue facilitating segregation of a concurrent signal.
It seems that sequential grouping factors such as similarity of perceptual attributes (e.g., pitch or timbre) of the cue and masker also play a role in signal detection. [Work supported by NSERC.]
Acoustical analysis of false starts in spontaneous speech
Authors: O’Shaughnessy Douglas 
Published: October 2005

In spontaneous speech many false starts occur, where a speaker interrupts the flow of speech to restart the utterance. The acoustic aspects of such restarts in a common database were examined for duration and fundamental frequency. Automatically identifying the type of restart could improve speech recognition performance, by eliminating one version of any repeated words (or parts), and in the case of changed words, suppressing the unwanted words, so that the recognizer operates on only desired words. In virtually all current recognizers, words in a restart either simply pass to the textual component of the recognizer or cause difficulties in having a proper interpretation in the language-model component (since the language model is invariably trained only on fluent text). The spoken data consists of 42 speakers, each speaking about 30 different utterances (median length of about 12 words). There were 60 occasions with simple repeated words (or portions), 30 cases of inserted words, and 25 occurrences of new words substituted. When a word was simply repeated in a restart, it had virtually the same prosodics usually, but occasionally the repeated word had less stress. With a substitution or insertion in the restart, the modified word was virtually always more stressed. For restarts where the speaker stopped in the middle of a word and simply ‘‘backed up,’’ the pause lasted 100–400 ms in 85% of the examples. About 75% of the interrrupted words did not have a completion of the vowel in the intended word’s first syllable. In virtually all examples, the speaker completed at least 100 ms of the word, however, before pausing for at least 100 ms.
When the pause occurred at a word boundary, the words repeated after the pause were either a straight repetition with little prosodic change or a repetition where the repeated words shortened up to 50%. As for the repeated words (after the pause) prior to the inserted word, function words showed little or no shortening, but usually had lower F0; content words here exhibited significant shortening and lower F0. Such prosodic change only applied to nonprepausal words, because words immediately prior to a pause were often subject to significant prepausal lengthening. [Work supported by NSERC.]
Comodulation masking release in subjects with unilateral and bilateral hearing impairment
Authors: Moore Brian C. J. , Shailer Michael J. , Hall Joseph W. , Schooneveldt Gregory P. 
Published: October 2005

Three subjects with unilateral cochlear hearing loss and three subjects with bilateral cochlear hearing loss were tested in three experiments. In the first, their auditory filter shapes were measured for center frequencies of 700 and 2000 Hz, using the notched-noise method. The auditory filters were generally broader for the impaired than for the normal ears. In experiment 2, the threshold for detecting a 2000-Hz signal centered in a band of noise was measured as a function of the noise bandwidth for a Gaussian noise, and for that same noise multiplied (modulated) by a second noise low-pass filtered at 12.5 Hz. For the Gaussian noise, thresholds increased up to a certain noise bandwidth and then flattened off. This bandwidth was usually greater for the impaired than for the normal ears, consistent with the broader auditory filters of the impaired ears. For the modulated noise, thresholds tended to decrease when the noise bandwidth was increased beyond a certain value, indicating comodulation masking release (CMR).
The decrease occurred at wider bandwidths for the impaired than for the normal ears. For the unilaterally impaired subjects, the amount of decrease was smaller for the impaired than for the normal ears when tested at equal SPL, but not when tested at equal SL. In experiment 3, the threshold for detecting a 700-Hz signal centered in a 20-Hz-wide band of noise (the on-frequency band, ONB) was measured in the presence of eight flanking bands (FBs) whose envelopes were either identical with that of the ONB (correlated condition) or were uncorrelated. CMR was defined as the difference in threshold between the correlated and uncorrelated conditions. The ONB and the FBs were presented either to the same ear (monaural condition) or to opposite ears (dichotic condition). CMRs tended to be greatest at high levels of the ONB and the FBs. CMRs in the monaural condition were smaller for hearing-impaired than for normal ears. However, at high levels, CMRs in the dichotic condition were similar for normal, bilaterally impaired, and unilaterally impaired subjects.
In the latter case, CMRs were similar when the ONB was presented to the normal ear and to the impaired ear of each subject. It is concluded that: (1) the small CMRs often found for subjects with cochlear hearing loss can be partly attributed to the low sensation level of the stimuli; (2) reduced frequency selectivity may also contribute to small CMRs; and (3) independence of the output of the auditory filters centered on the ONB and the FBs, as occurs under dichotic conditions, can lead to large CMRs in impaired subjects.
Effects of forward masking on intensity discrimination, frequency discrimination, and the detection of tones in noise
Authors: Carlyon Robert P. , Beveridge Hazel A. 
Published: October 2005

A series of experiments investigated the deterioration in intensity discrimination at intermediate levels caused by preceding the brief sinusoidal stimuli by an intense narrow-band noise [Zeng et al., Hear. Res. 55, 223–230 (1991)]. Experiment 1 showed that this deterioration acted in a roughly additive manner with the previously reported midlevel deterioration for the intensity discrimination of brief 6500-Hz tones in quiet. The experiment also showed that a smaller, although still substantial, deterioration occurred for the detection of brief tones in bursts of synchronous noise. In experiment 2, the masker could occur either before the signal (louder tone) or standard (softer tone), the choice being selected at random on a trial-by-trial basis, and with the Weber fraction tracked for each condition separately using two interleaved, adaptive procedures. Weber fractions were lower when the masker preceded the signal than when it preceded the standard, arguing against a major role for adaptation in causing the midlevel deterioration. Instead, it is suggested that the deterioration shared a common mechanism with the phenomenon of ‘‘loudness enhancement’’ [R. J. Irwin and J. J. Zwislocki, Percept. Psychophys. 10, 189–192 (1971); Galambos et al., J. Acoust. Soc. Am. 52, 1127–1130 (1972)].
The third experiment showed that Weber fractions did not depend greatly on whether the forward masker occurred only in interval one or only in interval two of a two-interval trial. A fourth experiment investigated the effect of a forward masker on frequency difference limens (DLs). When the masker was presented in both intervals of each trial, there was only a moderate increase in frequency DLs which, in contrast to the elevation in Weber fractions for intensity discrimination, was independent of signal level. Presenting the masker before only the lower frequency tone of a pair to be discriminated usually yielded lower DLs than presenting it only before the higher frequency tone, but this effect was less consistent than for intensity discrimination. The results of all experiments are discussed with respect to the various ways in which an intense narrow-band masker can affect the perception of subsequent brief sounds.
Pitch-synchronous frame-by-frame and segment-based articulatory analysis by synthesis
Authors: Gupta Sunil K. , Schroeter Juergen 
Published: October 2005

This paper presents a pitch-synchronous analysis-by-synthesis procedure for estimating model parameters for voiced speech. These model parameters describe the vocal-tract shape and the time derivative of the glottal area function. The excitation waveform is derived from the glottal area function by incorporating source-tract interaction using the current vocal-tract input impedance. The corresponding analysis procedure for estimating the model parameters once every pitch period is outlined. A significant improvement in quality was obtained for the new pitch-synchronous analysis/synthesis procedure relative to the fixed-frame-length-based scheme used previously. It was also found that the new pitch-synchronous articulatory analysis/synthesis scheme achieves lower rms spectral distortion values than the 2.4 kb/s. Federal standard LPC-10E algorithm. A segment-based procedure for estimating the vocal-tract model parameters at a rate much lower than the current pitch is described. In this segment-based analysis-by-synthesis approach, the model parameters are estimated every 50–100 ms. The parameters for the intermediate pitch periods are derived by interpolation. The segments are selected using a maximum likelihood segmentation algorithm that segments an utterance into diphonelike units. A segment-based parameter optimization scheme could lead to a highly economical representation of the speech signal for potential applications in very low bit rate speech coding and speech storage.
The above schemes were optimized for a pilot test sentence and then evaluated using eight test sentences for a log area and the Coker articulatory model representation of the vocal tract. Nine listeners were asked to judge the quality of the synthesis in a paired-comparison test and the results were analyzed using a nonparametric one-tailed sign test. For the log-area representation of the vocal tract, we found a significant degradation in speech quality for the segment-based optimization procedure relative to the frame-based procedure. However, for the Coker model representation, the degradation was found to be insignificant. This shows that unlike cross-sectional areas, the movement of various articulators in the vocal tract during speech production can be described with sufficient accuracy by specifying the position of these articulators and by using an interpolation function at time intervals much longer than a pitch period.
A new stochastic model for auditory-nerve discharge
Authors: Miller Michael I. , Wang Jing 
Published: October 2005

This paper investigates the mechanisms for history-dependent probability of eighth-nerve discharge, which is modeled as the probability that the excitatory postsynaptic potential (EPSP) process crosses afferent membrane threshold, with the discharge history dependence due to the dependence of postsynaptic threshold voltage on time since previous action potential. The model parameters are the Poisson intensity αt of vesicle release, the duration ε and probability density pV(v) of single-vesicle EPSP’s, and the threshold voltage curve θ(τ) for spiking. It is proven that the infinitesimal conditional probabilities of discharge exhibit two distinct behaviors. The first is associated with the time τ=TD, exactly the time the neuron releases from absolute refractory where there is no intensity [θ(τ)=∞, for τ&lt;TD]. At this time the neuron has a nonzero probability of discharge Λt(TD) = limΔ→0 Pr{Nt,t+Δ = 1‖t − wNt= TD}. The second regime corresponds to the time since previous spike being greater than dead time, τ≳TD, during which time the intensity exists λt(τ) = limΔ→0(1/Δ)Pr{Nt,t+Δ = 1‖t − wNt= τ≳ TD}. The fact that there is a nonzero probability of discharge following passage from the absolute refractory period predicts the nonmonotonic hazard intensity seen in high spontaneous neurons [R. P. Gaumond, Ph.D. thesis, Washington University, St. Louis (1980)] and high driven rate neurons. It is shown that for the lowest range of vesicle release intensities where the vesicle-release-rate/membrane-integration-time product αtε is small, the nonzero probability of discharge at a point is approximately equal to 0. The discharge intensity is dominated by a term linear in vesicle release intensity: λt(τ)≊αt exp(−∫t−εtασ dσ)∫θ (τ)∞pV(v)dv. This is precisely the Siebert–Gaumond intensity product form with monotonic recovered discharge probability.
At high vesicle release rates, such as for driven rate responses, the nonzero probability of discharging at a point becomes of nonsignificant size, and the intensity of discharge grows nonlinearly with αt, implying the product model does not hold. The model is demonstrated via the analysis of auditory nerve fibers from the cat.
ON ONE-MEMBRANE P SYSTEMS OPERATING IN SEQUENTIAL MODE
Authors: DANG ZHE , IBARRA OSCAR H. 
Published: October 2005

In the standard definition of a P system, a computation step consists of a parallel application of a "maximal" set of nondeterministically chosen rules. Referring to this system as a parallel P system, we consider in this paper a sequential P system, in which each step consists of an application of a single nondeterministically chosen rule. We show the following:1. For 1-membrane purely catalytic systems (pure CS's), the sequential version is strictly weaker than the parallel version in that the former defines (i.e., generates) exactly the semilinear sets, whereas the latter is known to define nonrecursive sets.2. For 1-membrane communicating P systems (CPS's), the sequential version can only define a proper subclass of the semilinear sets, whereas the parallel version is known to define nonrecursive sets.3. Adding a new type of rule of the form: ab → axbyccomedcometo the CPS (a natural generalization of the rule ab → axbyccomein the original model), where x, y ∈ {here, out}, to the sequential 1-membrane CPS makes it equivalent to a vector addition system.4. Sequential 1-membrane symport/antiport systems (SA's) are equivalent to vector addition systems, contrasting the known result that the parallel versions can define nonrecursive sets.5. Sequential 1-membrane SA's whose rules have radius 1, (1,1), (1,2) (i.e., of the form (a, out), (a, in), (a, out; b, in), (a, out; bc, in)) generate exactly the semilinear sets. However, if the rules have radius 1, (1,1), (2,1) (i.e., of the form (ab, out; c, in)), the SA's can only generate a proper subclass of the semilinear sets.