bluuebunny/crossref_metadata_2025 huggingface dataset:
Low-frequency diffraction from a free surface coupled to a semi-infinite elastic surface as modeled by sea ice properties
Authors: Dahl Peter H. , Frisk George V.
Published: October 2005

This paper discusses the solution of a low-frequency plane wave incident upon a semi-infinite elastic plate, such as an Arctic ice lend or free edge, using the Wiener-Hopf method. By low-frequency it is meant that the elastic properties of the plate are adequately described by the thin plate equation. For example, in a floating ice sheet, this translates into frequency-ice thickness products that are ≲ 150. A key issue here is the fluid loading pertaining to sea ice and low-frequency acoustics, which cannot be characterized by simplifying heavy or light fluid loading limits. An approximation to the exact kernel of the Wiener-Hopf functional equation is used here, which is valid in this midrange fluid loading regime. The farfield diffracted pressure is found, which includes a fluid-loaded, sub-sonic (relative to the water) flexural wave in the ice plate. Comparisons are also made with the locally reacting approximation to the input impedance of an ice plate. The combined effects of the ice lead diffraction process represent loss mechanisms that contribute to the transmission loss in long-range Arctic acoustic propagation.
Low-frequency SAW sensors for the detection of shear stresses in the turbulent boundary layer
Authors: Roh Yongrae , Varadan Vasundara V. , Varadan Vijay K.
Published: October 2005

To simulate a low-frequency SAW device, surface wave propagation at the boundary between water and a thin film of PZT-SH on steel was investigated extensively through numerical analysis. Two- and three-dimensional dispersion curves, attenuation mechanism, and displacement variation were obtained for each of the propagation modes of all types of the surface waves (Rayleigh, Scholte, and Love) in the medium. The energy distributions of the Rayleigh and Scholte waves have also been obtained. With these results, the optimum geometry (crystal cut, propagation direction, and nondimensional wavenumber) for maximum launching efficiency was determined. The acoustic fluid was replaced by a turbulent flow, and the variation of the propagation velocity of the SAW due to turbulence was investigated. These results show a new method to distinguish the effect of shear stress fluctuations from that of normal pressure in a turbulent flow.
Linear transformations for vowel normalization
Authors: Zahorian Stephen A. , Jagharghi Amir J.
Published: October 2005

The results of an evaluation of multivariable linear regression techniques for speaker normalization of vowel data for 11 vowel classes will be presented. The database for the study consisted of the central vowel portions of 2922 CVC syllables obtained from ten males, ten females, and ten children. Each stimulus was represented both by three formants and in terms of overall spectral shape, via the discrete cosine transform coefficients (DCTCs) of the magnitude spectra. In all classification experiments, half the database was used to train the classifier and the other half was used for evaluation. For the case of formants, the classification accuracy on the evaluation data was 63.3% if different speakers were used for training and testing (and thus no speaker normalization), 63.9% if the same speakers were used in the training and test sets but without explicit normalization, and 75.2% with speaker-normalized data. The rates for the corresponding conditions, but with DCTCs as parameters, were 58.0%, 66.1%, and 77.9%. These results indicate a reduction in classification error rate due to speaker normalization of 32.4% for the case of formants and 47.3% for the case of DCTCs. The speaker-normalization techniques are an extension of the techniques reported in the literature [S. F. Disner, J. Acoust. Soc. Am. 67, 253–261 (1980)]. These results also extend our results reported at the Fall 1987 ASA Meeting [S. A. Zahorian and A. J. Jagharghi, J. Acoust. Soc. Am. Suppl. 1 82, S37 (1987)] to a larger, more varied database and indicate that automatic classification of speaker-normalized vowels is comparable to either spectral shape parameters or formants as initial features. [Work supported by NSF.]
F0 normalization and talker variability
Authors: Johnson Keith
Published: October 2005

Fundamental frequency (F0) normalization is an effect observed in vowel perception in which the F0 of the vowel plays a role in perceived vowel quality. It was hypothesized that this perceptual effect involves a process in which hearers adjust a perceptual vowel space when they detect a change of speaker. A seven-step vowel continuum from “hood” to “bud” was synthesized at two levels of F0 (120 and 240 Hz). These two continua were presented to subjects blocked by F0 level and with items of different F0 randomly intermixed with each other. In the mixed F0 condition, there was a large shift in the identification functions of the vowel continuum as a result of changing F0, while there was no effect of F0 in the blocked F0 condition. Analysis of the reaction time data showed that subjects identified the items more slowly in the mixed condition than in the blocked condition and that this increase occurred for items that were preceded by an item with different F0 and not when successive items had the same F0. These data suggest that hearers adjust a perceptual vowel space when they detect a change of speaker. [Work supported by NIH Training Center Grant No. 50-315-26 to Indiana University.]
Target zones for synthetic vowels
Authors: Miller James D. , Hawks John W.
Published: October 2005

Using algorithms developed by the second author (JWH), he synthesized 7703 vowel tokens such that the values of F0, F1, F2, and F3 uniformly sampled locations in a plane in the middle of the vowel slab of the auditory-perceptual space [J. D. Miller, “Auditory-perceptual interpretation of the vowel,” J. Acoust. Soc. Am. (in press)]. Each token was 400 ms and had a rise-fall pitch contour. The tokens were arranged in random order and presented in groups of 100 to one listener (JDM), who identified each token as [IY, IH, EH, EY, AE, AA, AH, AO, OW, UH, or UW] and rated the clarity of each token on a scale from 1 (poor) to 5 (excellent). The sizes and shapes of the target zones derived from these experiments will be compared to those based on measurements of natural speech. The data are, in general, consistent with the concept of vowel target zones as being large with irregular and abutting boundaries. [Work supported by NINCDS.]
Detection thresholds for isolated vowels
Authors: Kewley-Port Diane
Published: October 2005

A series of experiments on the detectability of vowels in isolation has been completed. Stimuli consisted of three sets often vowels, one synthetic, one from a male talker, and one from a female talker. Vowel durations ranged from 20–160 ms. Thresholds for detecting the vowels in isolation were obtained from well-trained, normal-hearing listeners using an adaptive-tracking paradigm. Detection thresholds for vowels calibrated for equal sound pressure at the earphones differed by as much as 20 dB across vowels. However, the same patterns of thresholds obtained for the vowel set were obtained for all vowel durations. An orderly decrease in vowel thresholds was obtained for increased duration, as predicted from temporal integration for pure tones. Several different analyses have been conducted in an attempt to explain the differential detectability for the various vowels within each set. The most detailed of these analysis involved a model developed by Moore and Glasberg (1987) for calculating excitation patterns and the corresponding loudness level in phons. While that model provided improved explanatory power over other models tried, further refinements of perceptual models of loudness will be needed to explain these data. [Research supported by NIH and AFOSR.]
Perceived modulation magnitudes of second-formant, center-frequency variations
Authors: Sullivan L. S. , Porter R. J. , Cullen J. K. , Collins M. J.
Published: October 2005

Speech signals convey information via rapid changes in formant frequencies. Listeners can reliably judge the relative “modulation magnitude” of frequency modulations [e.g., H. Fastl, Hear. Res., 59–69 (1982)]. Listeners were asked if they could also judge modulation magnitude for center-frequency-modulated formants by presenting pairs of 500-ms stimuli with centered, “notchlike” modulations drawn from synthetic continua with varying notch duration (30–300 ms) or depth (50–500 Hz). The first, reference, stimulus (either a low- or high-modulation extreme of a continuum) was assigned a reference magnitude (1 for low-modulation-reference trial blocks; 10 for high-). Listeners assigned a relative “modulation strength” (1 to 10) to the second stimulus. Judged magnitude appears as an uneven perceptual surface in the magnitude × depth × duration space: Highest modulation strengths are heard for deep-long stimuli; deep-short stimuli have slightly lower judged magnitudes; shallow-long modulations are much weaker; shallow-short modulations are the weakest. The possible relation between perceived modulation magnitude and judgments of articulatory-phonetic differences will be discussed. [Work supported in part by NIH-NINCDS and the Louisiana Lions Eye Foundation.]
The use of rate information during phonetic perception depends on pitch continuity
Authors: Green Kerry P. , Stevens Erica B. , Kuhl Patricia K.
Published: October 2005

Studies have demonstrated that speaking rate provides an important context for the perception of certain acoustic properties of speech. For example, syllable duration, which varies as a function of speaking rate, has been shown to influence the perception of voice-onset-time (VOT) for syllable-initial stop consonants. The purpose of the present experiments was to examine the influence of syllable duration when the initial portion of the syllable was produced by one talker, and the remainder of the syllable was produced by a different talker. A short duration and a long duration /bi-pi/ continuum was synthesized with pitch and formant values appropriate to a female talker. When presented to listeners for identification, these stimuli demonstrated the typical effect of syllable duration on the voicing boundary: a shorter VOT boundary for the short stimuli relative to the long. An /i/ vowel, synthesized with pitch and formant values appropriate to a male talker, was added to the end of each of the short tokens producing a new “hybrid” continuum. Although the overall syllable durations of the hybrid stimuli equaled the original long stimuli, they produced a VOT boundary similar to the short stimuli. In a follow-up experiment, two new /i/ vowels were synthesized. One had a pitch appropriate to a female talker with formant values appropriate to a male talker, while the second had a pitch appropriate to a male talker and formants appropriate to a female talker. These vowels were used to create two new hybrid continua. The results of this second experiment indicated that continuity of pitch but not of formant structure appears to be the important factor in the integration of speaking rate within a syllable. [Work supported by NIH.]
Phonetic and nonphonetic fusion in duplex perception
Authors: Nygaard Lynne C. , Eimas Peter D.
Published: October 2005

When listeners hear a synthetic consonant-vowel syllable split with its third-formant transition presented to one ear and the rest of the syllable (the base) presented to the other ear, they report two distinct percepts: A nonspeech chirp sound in the ear presented the transition and a complete syllable in the ear presented the base. The present studies investigated the nature of fusion of information yielding the phonetic percept. By introducing a third-formant transition into the base, fusion of information in the nonspeech percept was also studied. In this case, listeners reported hearing both a nonspeech chirp centrally lateralized and a complete, but now clearer, syllable in the ear presented the base. Experiment 1 replicated the original phenomenon of duplex perception and demonstrated reliable phonetic and nonphonetic fusion of the isolated and embedded transitions. In experiments 2–5, fundamental frequency, amplitude, spectral composition, and onset times of the chirp were varied relative to the base that did or did not contain a third-formant transition. It was found that listeners' ability to detect the additional third-formant transition differed for the phonetic and chirp percepts for each of the four variables. These results suggest that, as expected, the mechanisms for the perception of speech and location are different. Moreover, neither system captures all the available information and thus the fusion of information can occur in both directions.
The role of spectral, temporal, and dynamic cues in the perception of English vowels by native and nonnative speakers
Authors: Miranda Salvatore , Strange Winifred
Published: October 2005

Previous cross-language studies have demonstrated that different languages may utilize different phonetic features to signal phonemic distinctions. This study investigated which phonetic features are utilized by language learners perceiving speech in the target language and if the acoustic parameters that specify the features are processed in the same way by native and nonnative speakers. Native speakers of English and students of English as a second language (ESL) identified American English vowels that were electronically modified to vary systematically in the type of acoustic information available. The role of three types of acoustic information was investigated: vowel targets available in quasisteady-state syllable nuclei; dynamic information contained in rapidly changing formant transitions; and intrinsic duration information. Results indicated a remarkably similar error pattern across conditions for both native and nonnative subjects, except for one less-advanced subgroup of nonnatives who identified vowels in certain modified conditions more poorly than vowels in an unmodified control condition, suggesting that less-advanced nonnatives may need all possible sources of information to identify English vowels correctly. [Work supported by NINCDS.]
Perception of speaker age using a paired stimuli technique
Authors: Morris Richard J. , Brown W. S.
Published: October 2005

Listeners have consistently demonstrated the ability to estimate the age of taped voice within a few years of the chronological age of the speaker. In this study, a paired comparison technique was used to determine the consistency of raters' judgments. The listeners rated the age of voices from two age groups of speakers. Both intrajudge and interjudge consistency were determined. The speakers for the study consisted of ten women who were 20–30 years of age and ten women who were 75–90 years. of age. A 3-s sustained /a/ at comfortable frequency and intensity levels was used as the stimulus for the listeners. The listeners were 22 undergraduate students who rated the second voice in each pair relative to the first one. The voices were rated as sounding “much older,” “older,” “the same age,” “younger,” or “much younger” than the first voice. A multiple correlation statistical paradigm was used to analyze the data. The results will be discussed in relation to theories of aging speech production and perception.
Respiratory response to oral flow and pressure perturbations during speech
Authors: Rochet Anne Putnam , Morr Kathleen E. , Warren Donald W.
Published: October 2005

Evidence for the regulation/control hypothesis of Warren for speech aeromechanics [Cleft Palate J. 23, 251–260 (1986)] was observed in nine normal men who produced /pʌ/ and /si/ under conditions in which a translabial device released intraoral pressure during /p/, and bite blocks opened the anterior bite artificially during /s/. Intraoral pressures, oral flows, and respiratory volumes associated with utterance were the dependent variables. For /p/, intraoral pressure decreased and translabial air leakage increased as bleed orifice area increased, but mean peak intraoral pressure never fell below 4 cm H2O. For /s/, mean oral flow increased slightly as the anterior bite was opened, but mean peak oral pressure did not vary significantly from control values. Mean flow on the postconsonantal vowels did not vary significantly across the experimental conditions. There were no significant differences among mean inspiratory volumes within or across translabial bleed conditions, nor within or across bite-Mock conditions. Expiratory volume increased significantly as bleed orifice area increased, but not as bite-block size increased. These data may reflect the regulation control options inherent in each condition. The respiratory system alone could be relied upon to maintain intraoral pressure within requisite limits for speech during the bleed conditions; during the bite-block trials, speakers could employ increased respiratory drive, structural adjustments of the sibilant orifice, or both, to compensate for aeromechanical perturbation introduced by the blocks. [Work supported by AHFMR.]
The application of frequency-domain adaptive filters to speech enhancement
Authors: Haweel Tarek I. , Clarkson Peter M.
Published: October 2005

Several authors have applied the time-domain least-mean-squares adaptive filter to the problem of (voiced) speech enhancement. Generally, these efforts have achieved only limited success due, in part at least, to the nonuniform convergence of the adaptive filter when faced with frequency components of highly disparate spectral power (the so-called “eigenvalue disparity” problem). This problem is addressed by employing the normalization capacity of the frequency-domain adaptive filter (FDAF). The first part of this paper deals with the analysis of the FDAF for strictly harmonic signals (this reflects the quasiperiodic nature of voiced speech). It is shown that the behavior of the filler for each weight of the FDAF can be described by a linear transfer function relating the desired input to the output signal. It is further shown that the product of the input power and the feedback component (in each frequency bin) determines the stability and convergence of the filter. Normalization can be achieved by adjusting this product for each weight. Simulations comparing time- and frequency-domain approaches have shown that the FDAF enhancer significantly improves performance.
The intelligibility of digitally high-pass filtering the discrete digital frequencies in amplitude- and time-quantized speech
Authors: O'Brien Edward M.
Published: October 2005

Speech was first amplitude quantized into two levels, depending on whether the speech signal was above or below a 0-V reference. Time quantization was next accomplished using clocking rates of 25, 16.7, 10, and 6.25 kHz. Thus only discrete digital frequencies remain in the processed waveform. Having the speech in a digital format makes it amiable for further processing. Determining the discrete digital frequencies is relatively straightforward and these can then be displayed onto the skin using electrical stimulation to implement a tactile hearing aid. Since only a finite number of electrodes can be used, the effect of high-pass filtering the quantized speech was investigated using seven normal-hearing subjects. Phonetically balanced monosyllabic words were used as the test tokens. With no high-pass filtering, the percent word intelligibilities were 91%, 86%, 77%, and 54% for quantizing rates of 25, 16.7, 10, and 6.25 kHz, respectively. Several filters were tested for each quantizing rate. However, 70% word intelligibility (considered adequate for conversational speech) was obtained with only 15, 11, and 9 discrete digital frequencies remaining in the filtered speech for quantizing rates of 25, 16.7, and 10 kHz, respectively.
Voice quality in males: A multidimensional scaling study
Authors: Gelfer Marylou Pausewang
Published: October 2005

The purpose of this study was to investigate the perceptual dimensions that listeners use in discriminating male voices, and how these dimensions relate to perceptual rating scale judgments. Results were compared to earlier research using female voices. Listeners were 18 certified speech pathologists who judged the dissimilarity of sentences produced by 20 males (20–26 years of age). The listeners also evaluated each voice on 17 perceptual rating scales (e.g., high pitch-low pitch, clear-hoarse). Results of the multidimensional scaling (MDS) analysis produced a four-dimensional solution. A stepwise linear regression related the perceptual attributes of speech rate and vocal pitch to the first two dimensions, while the second two dimensions were best explained by descriptors pertaining to quality (e.g., pleasant, smooth, melodious). Compared to the MDS analysis of female voices, the variance accounted for was reduced, suggesting more random variability among male voices, or that these voices were more difficult for listeners to rate in a consistent manner. Combined results from both investigations suggest that the perceptual dimensions that listeners use in discriminating among male voices may not be substantially different from the dimension used in discriminating among female voices. [Research supported by BRSG Grant S07 RR07031, DRR, NIH.]
Intelligibility of speech produced in noise and while wearing an oxygen mask
Authors: Bond Z. S. , Moore Thomas I.
Published: October 2005

A number of laboratories have described relatively systematic changes in the acoustic-phonetic structure of speech produced in the presence of noise relative to that produced under more benign speaking circumstances. At least some of these changes may be attempts to make speech clearer and more intelligible. In order to evaluate the intelligibility of speech produced in various circumstances, a male talker speaking isolated words in quiet and in the presence of 95 dB SPL of pink noise was recorded. Speech samples were collected both with and without an oxygen mask. The peak amplitudes of each word were normalized, and the words were mixed with pink noise and presented for identification to 45 native and to 10 nonnative English speakers. The order of relative intelligibility was the same for both groups. Speech produced while listening to noise alone was most intelligible. Speech in the laboratory and in mask conditions was considerably less intelligible. The noise + mask condition led to marginally the least intelligible speech. [Work supported by AFSOR.]
Speech intelligibility as a function of environment and speaking style
Authors: Payton Karen L. , Uchanski Rosalie M. , Braida Louis D.
Published: October 2005

Sentences spoken “clearly” have been shown to be more intelligible than those spoken “conversationally,” both for hearing-impaired listeners in a quiet background [Picheny et al., J. Speech Hear. Res. 28, 96–103 (1985)] and for normal-hearing listeners in a noise background [R. M. Uchanski, Ph.D thesis, MIT (1988)]. This study examined the effects of noise and reverberation on the intelligibility of these two speaking styles for listeners with normal hearing. Six environmental conditions were tested: three reverberation times (anechoic, T = 0.18 s, and T = 0.6 s) by two levels of noise (+ ∞ and 0 dB S/N). As the environmental degradation increased (calculated from the STI), intelligibility scores decreased for both speaking styles. However, clear speech was more intelligible than conversational speech for each condition. Further, the clear speech scores declined at a more gradual rate than the conversational scores as the STI decreased. Envelope spectra of clear and conversational speech are being examined to derive a predictor of intelligibility that includes the characteristics of both the speech source and the acoustic environment.
Improvements to a speaker-independent algorithm
Authors: Scott Brian , Lin Lisan , Newell Mark , Smith Lloyd
Published: October 2005

In May 1988, a speaker-independent recognition algorithm was described at the 115th Meeting of the Acoustical Society of America [Scott et al., J. Acoust. Soc. Am. Suppl. 1 83, S55 (1988)]. The algorithm yielded 95.4% recognition accuracy on the 20-word TI database obtained from the National Bureau of Standards. The system has since been adapted for use over the telephone. In so doing, a new database was developed consisting of 16 words (zero-nine, oh, yes, no, cancel, terminate, and help) as spoken by 11 males and 11 females from various locations across the country. Although small (1001 utterances), the database represented a significant challenge as compared to the one obtained from the NBS. There were fewer training passes per word, more speakers, and there was considerably more noise in the database. Initial tests on this database yielded accuracies of approximately 60%. Four major enhancements to the algorithm improved the accuracy on this database to 95.3% Two of the enhancements compensate for time alignment problems inherent in both linear and nonlinear time normalization routines through the use of prosody and redundancy. The other two enhancements include the extraction of additional features and better use of variances. Results indicate that acceptable speaker-independent recognition can be obtained with minimal training and processor requirements, given effective normalization routines during the front-end signal processing.
An algorithm for distinguishing between voiceless stops and voiceless fricatives
Authors: Weigelt LaDeana F. , Sadoff Steven J. , Miller James D.
Published: October 2005

An algorithm has been developed for the purpose of voiceless stop/fricative distinction in word-initial position using three measures of the waveform: zero crossings, energy, and derivative of the energy over time (termed rate of rise, ROR). The ROR is used as the primary classifier while energy and zero crossings 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 treated on 720 CVC tokens (four male speakers, four female speakers, three stops and an affricate [P,T,K,CH], five fricatives [S,SH,F,TH,H], and ten vowel contexts [IY,IH,EH,AE,AA,AH,AO,UH,UW,ER]). Data from two male and two female speakers (360 tokens) were used as a training set, and the remaining data (360 tokens) were used as a test set. The overall success rate was 96.8%. [Work supported by AFOSR.]
Pattern processing of stop consonants: Part II
Authors: Garrison-Shaffer L.
Published: October 2005

The roles of F1, F2, and vowel environment in stop consonant pattern processing were investigated using interaural transfer and selective adaptation. Previous results indicate that different patterns may exist for processing of stops, depending upon whether the stop occurs in a front or back vowel environment [L. Garrison, J. Acoust. Soc. Am. Suppl. 1 83, S70 (1988)]. Such pattern processing may reflect differences in the placement of F2 relative to F1. This experiment systematically manipulated the relationships between formants as well as vowel environment for CV syllables. These manipulations resulted in six stimuli that were used as adaptors on a [ba]-[da] test series. One adaptor shared its vowel percept ([ba]) and three other adaptors shared vowel category (back) with the test series. Finally, two adaptors were from a front vowel category. The results will be discussed in relation to how formant relationships and vowel environment affect pattern processing of stop consonants. [Work supported by NINCDS.]
Acoustic features of nasal consonants in VC contexts
Authors: Qi Yingyong , Fox Robert Allen
Published: October 2005

The acoustic features of the nasals [m,n] in CV syllables and [m,n,ŋ] in VC syllables in English were first analyzed and compared using cepstrally smoothed running FFT spectra. Clear differences between nasals in CV and VC syllables were obtained. For example, the spectral energy transitions from vowel to nasal in VC syllables were found to be much less dramatic than in CV transitions. Next, given the recent interest in the efficacy of auditory representations in speech recognition schemes, the nasals in VC contexts were examined in terms of auditory transformed running spectra. Several features of interest were obtained: The spectra were generally dominated by the second formant and, when the preceding vowel was a low vowel, nasal place of articulation was distinguished by the second formant transition, which converged to 16–17 ERB for [m] and 19–20 ERB for [n]; [ŋ] was characterized by little formant movement. Consistent nasal place features were not found in the context of the vowels [i] and [u]. Finally, since the antiresonances of a nasal may provide place of articulation information, the system zeros of the nasals were analyzed using parametric spectral analysis based on the autoregressive moving average (ARMA) process. Results indicated that the antiresonances of nasals in a VC syllable could be consistently estimated by a two-step AR approximation method that could be used to distinguish reliably between [m] and [n].
ESPRIT: A signal processing environment with a visual programming interface
Authors: Gayvert Robert T. , Biles John A. , Rhody Harvey , Hillenbrand James
Published: October 2005

ESPRIT (Explorer speech processing system from the Rochester Institute of Technology) is an integrated speech research development environment that runs on the TI Explorer, optionally augmented by the TMS-320 based Odyssey DSP board. The goal of ESPRIT is to provide speech scientists, linguists, and engineers an intuitive environment in which to collect, process, and display speech signals. ESPRIT's module editor allows users who are not programmers to draw data-flow programs made up of built-in and user-defined speech processing algorithms, display functions, and standard utilities. ESPRIT's display editor allows users to manipulate the graphical displays that result from running these programs to zoom, scroll, rearrange, take precise measurements, and perform a variety of other operations. While ESPRIT provides standard signal processing algorithms (FFT,LPC) and displays (waveforms, spectrograms, waterfalls, spectral slices), users who develop their own Lisp or TMS 320 programs can easily install them to replace or augment the standard software. [Work supported by Rome Air Development Center under Contract No. F30602-87-D-0090.]
Microcomputer system for analysis of the verbal behavior of patients with neurological and laryngeal diseases
Authors: Yunik Maurice , Boyahov Boyan , Rahardjo Budi
Published: October 2005

The microcomputer system for verbal behavior analysis is built on the basis of the IBM-PC. It consists of two parts: hardware and software. The hardware is the condenser microphone and a two-channel ADC board, including: amplifiers with adjustable gain; two Bessel analog filters with cutoff frequencies of 800 Hz and 6 kHz; and two ADCs allowing sampling rate up to 40 kHz. The software is a program package which allows high precision pitch detection by means of the autocorrelaton function; finding the separate pitch periods by means of amplitude selection and parabolic interpolation; pitch period perturbation; number of subharmonic-to-fundamental frequency power; separation of segments where the signal is strictly periodic; realization of spectral analysis over these segments by means of fast Fourier and fast Walsh transforms; and evaluation of the degree of hoarseness from the spectra. Preliminary experimental research shows that the system may be very useful for analysis of the speech of patients with neurological and laryngeal diseases as well as the analysis for musical sounds. Results from analysis of pathologic voices agree with those of Kasuya et al. [“Preliminary Experiments on Voice Screening,” in Proceedings of the Symposium on Voice Acoustics and Dysphonia, Gotland, Sweden (August 1985)] and Laver et al. [“An Acoustic Screening System for the Detection of Laryngeal Pathology,” J. Phon. 14, 517–524 (1986)].
Acoustic-phonetic analysis of stress-induced speech
Authors: Gordon John W.
Published: October 2005

The Lincoln Laboratories stress speech database (LLStyle) has been used to test various speech recognition algorithms that might be used in conditions such as stressful environments, under which the input speech can be expected to vary from its trained norm. To facilitate the correlation of performance degradations with speech variabilities, the entire LLStyle database (consisting of over 10 000 utterances) was transcribed. The program SEARCH was then used to gather statistics of various acoustic-phonetic parameters. Grouping these statistics according to speaking style has provided some insight into which parameter changes cause problems in general for machine recognition. The analysis of individual words has helped to explain why certain assumptions made by some recognizer strategies (for constraining the search space) are invalid when applied to variable speech.
Recognition of prosodic contrasts in literal and idiomatic utterances by native and nonnative speakers of English
Authors: Van Lancker Diana
Published: October 2005

Abilities of three groups—native English speakers, fluent nonnative English speakers, and advanced students of English as a second language (ESL)—to identify literal (L) and idiomatic (I) meanings of ambiguous sentences (e.g., “He was at the end of his rope”) were tested. Tape-recorded utterances produced by two speakers with either intended L or I meanings were presented singly and in contrastive pairs; listeners judged whether each stimulus expressed an L or an I meaning. Native speakers performed significantly better than fluent nonnative speakers on both single and paired presentations, while the ESL speakers performed at chance on both tasks. Linguistic training was associated with greater accuracies in the native but not in the fluent nonnative speakers. Acoustic analyses had indicated that L and I utterances differed significantly on measures of fundamental frequency (F0), number of pitch contours per utterance, lexical and overall utterance durations, and pausing. I utterances were higher in F0, and had fewer contours, shorter durations, and fewer pauses than the L utterances. The results replicate the original finding that native speakers can discriminate L from I meanings from prosodic cues alone [Van Lancker et al., J. Speech Hear. Res. 24 (1981)], and further indicate that performance on the test was not attributable to a listening strategy and that native competency significantly affects abilities to process these prosodic cues.
Relationship of recorded EMG signals to within- and cross-utterance acoustic variation
Authors: Faber Alice , Raphael Lawrence J.
Published: October 2005

Analysis of simultaneous acoustic and EMG recordings of one speaker of New York English producing /əpVp/ utterances under several conditions shows differences between inter- and intrautterance interactions. For three anterior genioglossus insertions (♯1–♯3) and one posterior insertion (♯4), the peak activation amplitude relevant to articulation of the vowel was measured. Amplitude differences reliably differentiated target vowels. For vowel/muscle pairs with significant amplitudes, activity duration and phasing of the peak within the activity were also measured. Despite strong correlations among the peak amplitudes for the anterior insertions, these measures differed in their correlations with measures of F0, F1, and F2, both within and across utterances. Overall, F2 correlated best with ♯1/duration, F1 with ♯2/amplitude, and F0 with ♯2/amplitude. For /ɪ/, intertoken F2 variation correlated best with ♯3/phasing, F1 with ♯2/amplitude, and F0 with ♯3/phasing. The posterior ♯4/duration correlated best with /uw/ F1 and F0. These results suggest both the importance of applying similar intertoken analysis techniques to larger data sets and the advances in understanding of subgestural articulatory control likely to result. [Work supported by NIH.]
Using dynamic time warping to formulate duration rules for speech synthesis
Authors: Macchi Marion J. , Spiegel Murray F. , Wallace Karen L.
Published: October 2005

Dynamic time warping (DTW), a speech recognition technique, was used to time align long-duration syllables with shorter-duration renditions of those syllables. A male speaker of English produced stressed syllables with long duration, like “libe,” and the same syllables with shorter durations in nonsense words like “libous,” “libegous,” "“libegass,” and “libotous.” Syllables were formant tracked and a frame-by-frame Euclidean bark-formant distance between each long syllable and each of its shorter counterparts was computed. DTW was applied to the resulting distance matrix. The DTW path, referenced to the long syllable, indicates which portions of the long syllable are shortened (and by how much) to match a shorter version of the syllable. The DTW paths showed regions of maximal and minimal shortening. These reactions were approximately aligned across syllable durations. The DTW paths for syllables of different durations were distinguished from one another primarily in regions of maximal shortening. The regions of maximal shortening were aligned with regions in the syllable with little formant movement. Consequently, a spectral-movement time function derived from the templates in an inventory could serve as the basis for duration rules for demisyllable speech synthesis. Further, this technique provides a method for making duration measurements automatically.
Speechreading sentences II: Application of a sequence comparator to data on CID sentences
Authors: Demorest Marilyn E. , Bernstein L. E.
Published: October 2005

When subjects speechread sentences, their performance is typically evaluated in terms of the number of words or keywords correct. A comparator that aligns stimulus and response sequences is being used as a heuristic for studying relationships between the lexical and the phonemic level of speechreading. Toward this end, a corpus of response sequences, previously analyzed in terms of words correct, was reanalyzed with the comparator. Stimuli were 50 CID sentences spoken by a male and a female talker and recorded on video laserdisc. Subjects were normal-hearing college students. One result showed that when sentences are short, computed visual stimulus-response similarity is well-correlated with number of words correct. But with few exceptions, when sentences are long, the correlation is reduced, suggesting that the two measures provide complementary information. The sequence comparator appears to be a useful tool for elucidating patterns of speechreading performance. [Work supported by NIH.]
Investigating randomness in foot timing patterns in English
Authors: Williams Briony , Hiller Steve
Published: October 2005

Isochrony has been considered only in terms of stressed syllables. However, it may also be a random property of unstressed syllables, and a control experiment was deemed necessary. A hand-transcribed database of 98 sentences, each produced by three speakers, formed the input to an algorithm calculating durations of feet, number of syllables per foot, and mean syllable duration within each foot. In each output dataset, feet were based on one of the following: stressed, tense, unreduced, random, or arbitrary syllables (the latter based on ordinal numbers of syllables within the utterance). Calculations were made of the correlations between foot duration and number of syllables per foot, and between foot duration and mean syllable duration. The first correlation was significant for all foot types; the second was significant (and negative) for all except the random and arbitrary types. The conclusion was that, although the mechanism of the tendency toward isochrony had by no means been discovered, it had been shown that the tendency was nonrandom and was due to linguistic rather than arbitrary factors.
Effects of absorption on the scattering of sound by sound
Authors: Darvennes Corinne M. , Hamilton Mark F. , Tjøtta Jacqueline Naze , Tjøtta Sigve
Published: October 2005

The scattering of sound by sound in a lossless fluid was discussed at an earlier meeting [Berntsen et al., J. Acoust. Soc. Am. Suppl. 1 83, S4 (1988), and Darvennes and Hamilton, J. Acoust. Soc. Am. Suppl. 1 83, S4 (1988)]. Here, the effects of absorption are included. The Khokhlov-Zabolotskaya-Kuznetsov equation is used to derive farfield asymptotic results for the sum and difference frequency sound due to the noncollinear interaction of real sound beams radiated from displaced sources. There are two main contributions to the nonlinearly generated sound in the farfield: the continuously pumped sound and the scattered sound. Weak absorption affects neither the locations nor the relative amplitudes of the pumped and scattered difference frequency sound. Strong absorption attenuates the pumped difference frequency sound faster than the scattered difference frequency sound. The scattered sum frequency sound is always attenuated faster than the pumped sum frequency sound, and there may be shifts in the locations of the maxima. Numerical results are presented for the case of Gaussian primary beams. [Work supported by ONR (CMD and MFH), IR&D Program of ARL:UT, and VISTA/STATOIL (JNT and ST).]
Effects of nuclei and host fluid parameters on the threshold for cavitation produced by pulsed ultrasound
Authors: Holland Christy K. , Apfel Robert E.
Published: October 2005

An experimental apparatus has been developed to determine thresholds for cavitation produced in a fluid by short tone bursts of ultrasound at 0.76, 0.99, and 2.30 MHz [A. Atchley et al., Ultrasonics 26, 280–285 (1988)]. A fluid jet was used to convect potential cavitation nuclei, such as 1-μm polystyrene spheres, echo contrast spheres, and whole blood constituents, through the focal region of the insonifying transducer. Cavitation thresholds measured with this system in water and in a fluid with ten times the viscosity of water will be presented. Cavitation was detected by a passive acoustical technique that is sensitive to sound scattered from cavitation bubbles. Results from these experiments that permit the control of nuclei and host fluid properties will be compared to an approximate theory that predicts the onset of cavitation [C. K. Holland and R. E. Apfel, Trans. IEEE UFFC-36(2) (1989)]. [Work supported by NIH, grant number 1RO1-CA-39374.]
Ceramic-air composites for hydrostatic sensing
Authors: Kahn Manfred
Published: October 2005

The hydrostatic response (dh) of conventional PZT ceramic is only about 10% to 30% of its biaxial (d33) response. This is expressed in dh = d33 (1–2 γ), where γ is Poisson's ratio. Higher dh values can then be obtained by lowering the Poisson's ratio of the ceramic. This can be accomplished, without lowering the d33 value significantly, by introducing into the ceramic arrays of voids, ordered in such a manner that void-free columns of ceramic are maintained between the voids. The deposition of fugitive ink through photolithographically prepared screens onto unfired ceramic tapes, together with stacking, laminating, and subsequent firing, provide a method for the controllable preparation of PZT sensors. These not only exhibit dh values of ⩾200 pC/N, but the voids also control the capacitance, so that gh values of ⩾0.03 Vm/N are obtained. Capacitance values above 400 pf in a 0.5×0.5×0.1-in. sensor chip permit some cable capacitance and the use of bipolar rather than of FET preamplifiers. Hydrostatic tests of such devices have shown these to be also rugged, easy to connect and use, and capable of withstanding significant hydrostatic pressures.
Four channels mediate vibrotaction: Facts, models, and implications
Authors: Bolanowski S. J.
Published: October 2005

Four separate pathways or channels apparently mediate tactile (i.e., mechanical) stimuli. Each channel, as defined psychophysically by its frequency characteristics in response to vibratory stimuli and physiologically by additional considerations such as receptive field sizes and adaptation rates to ramp-and-hold stimuli, operates over a wide range of frequencies and intensities. The psychophysical channels are: Pacinian or P; non-Pacinian I or NPI; non-Pacinian II or NPII; and non-Pacinian III or NPIII. As defined physiologically and anatomically, the channels are hypothesized to be, respectively: Pacinian (Pacinian corpuscles) or P; rapidly adapting I (Meissner corpuscles) or RA I; slowly adapting II (Ruffini endings) or SA II; and slowly adapting I (Merkel-cell neurite complexes) or SA I. Because the channels are overlapping in their operating ranges, it is proposed that under certain conditions (e.g., low stimulus intensities) a single channel may mediate taction, while other conditions require that more than one channel contributes to tactile sensations and to overall perceptual impressions. From the standpoint of applied aspects such as the development of tactile aids for the deaf, it should be of interest to determine if the channels can be used independently and to determine the types of interactions that may occur for the purpose of enhancing information transfer via this sensory system.
Clinical aspects of vibrotactile devices with deaf children
Authors: Osberger Mary Joe , Robbins Amy M.
Published: October 2005

Longitudinal data will be presented on the speech perception and production abilities of 15 deaf children who use a vibrotactile aid (Tactaid II). Their performance is being compared to that of deaf children who use either a single- or multichannel implant. Initial findings indicate that the performance of the subjects with implants, even those with single-channel devices, is superior to that of the subjects who use the tactile aid. The subjects who achieved the highest scores with the tactile aid tended to have more sophisticated cognitive and linguistic skills that permitted them to develop strategies to decode the vibratory patterns. The importance of prior knowledge of the language in decoding information perceived via the skin is also supported by the data collected from postlingually deaf adults using the same tactile aid. The implications of generalizing data collected on adults to prelingually deaf children will be discussed. [Work supported by NIH.]
Vibrotactile forward masking in young and old subjects
Authors: Gescheider G. A. , Valetutti A.
Published: October 2005

Thresholds for the detection of a 50-ms test stimulus were measured as a function of the time interval between the offset of a 500-ms masking stimulus and the onset of the test stimulus (Δt). The frequency of the masker and the test stimulus was the same during a particular testing session and was either 20 or 250 Hz. The stimuli were presented through a 2.9-cm2 circular contactor applied to the thenar eminence. Thresholds were measured by a two interval-forced choice tracking procedure. A group of seven young subjects (average age 21 years) and a group of older subjects (average age of 67 years) were tested. The amount of forward masking expressed as threshold shift decreased as Δt increased. At all values of Δt, the older subjects exhibited significantly more masking than did the young subjects. The forward masking functions and the effects of age were essentially the same for stimuli that primarily affect the Pacinian system (250 Hz) and those that primarily affect non-Pacinian systems (20 Hz).