bluuebunny/crossref_metadata_2025 huggingface dataset:
Effect of perceptually separating spectral components on profile discrimination
Authors: Dai Huanping
Published: October 2005

In spectral-shape discrimination [D. M. Green, Profile Analysis (Oxford U.P., Oxford, UK, 1988)], listeners discriminate a standard spectrum in which all components are equal in level, from a signal-plus-standard spectrum in which the level of one component (the signal component) is higher than the level of other nonsignal components. Because the overall level is randomly selected from a wide range of values on each stimulus presentation, the absolute level of the signal component is not a reliable cue for discrimination. Instead, listeners must make a relative level comparison between the signal and nonsignal components. Under normal conditions of profile experiments, listeners can detect signal increments as small as 1 dB. To be discussed here are special cases where the signal component is perceived as a separate sound from the nonsignal components. Perceptual separation between the signal and nonsignal components was achieved by introducing asynchrony in their onsets and offsets, by modulating them differently, or by presenting them in separate ears. Experimental results showed that such separation made spectral-shape discrimination very difficult. Thus optimum performance requires that the signal and nonsignal components be perceived as belonging to a single auditory object. [Work supported by NIH.]
Evaluation of a representational model for auditory scene analysis
Authors: Meyer Georg F. , Ainsworth William A.
Published: October 2005

Physiological data suggest that a two-dimensional signal representation of amplitude-modulation frequency against center frequency is extracted in the central nucleus of the inferior colliculus [C. E. Schreiner and G. Langner, J. Neurophysiol. 60, 1823–1840 (1988)]. The representation groups signals with common harmonics, and can easily be adapted to also enhance common onsets. This makes it an attractive basis for a representational model of auditory scene analyis. The map is used as a front end to a neural network pattern matching stage for vowel recognition. The model is tested against human performance in a concurrent vowel recognition task [P. F. Assmann and Q. Summerfield, J. Acoust. Soc. Am. 88, 680–697 (1990)]. It predicts human performance for a F0 grouping task, even for vowels with up to a 12-dB intensity difference, well. Another grouping cue, demonstrated in abstract sounds, is common onset. The double-vowel task was modified to include onset asynchrony. A model based on the change in the AM-map representation between successive frames predicts human performance well. It is possible to directly drive a pattern matching stage with the AM map representation removing the need for an explicit grouping processes. The model is inherently robust when driven by stimuli that violate grouping cues.
Perceptual tolerance of formant frequency in vowel–vowel transitions
Authors: Ainsworth William A. , Meyer Georg F.
Published: October 2005

The frequency difference limen for steady-state formants was estimated many years ago [J. L. Flanagan, J. Acoust. Soc. Am. 27, 613–617 (1955)] and more recently [D. Kewley-Port and C. S. Watson, J. Acoust. Soc. Am. 95, 485–496 (1994)]. It was not generally realized, however, that the perceptual tolerance of transitions is somewhat greater [W. A. Ainsworth, Proc. 13th Int. Congr. Phon. Sci., Stockholm, pp. 837–840 (1995)] at least for vowel–vowel pairs. There appears to be a perceptual continuum from simple sounds (tone glides) to more complex, speechlike sounds [A. van Wieringen and L. Pols, J. Acoust. Soc. Am. 98, 1304–1312 (1995)]. In the present series of experiments the perceptual tolerance of the center of F1 and F2 vowel–vowel transitions was estimated. It was found that this tolerance is much reduced if the transition is not completed. This tolerance is also reduced if the center transition frequencies of bothF 1 and F2 were varied simultaneously.
Acoustic and auditory representations of the voicing contrast
Authors: Damper Robert I. , Gore Mat O. , Harnad Stevan
Published: October 2005

Key aspects of the categorical perception of initial stop consonants by human and animal listeners are replicated by a ‘‘synthetic listener’’ in the form of a computational auditory model. This model is a composite of a physiologically faithful ‘‘front end’’ [M. J. Pont and R. I. Damper, J. Acoust. Soc. Am. 89, 1213–1228 (1991)] and an artificial neural network (ANN) ‘‘back end.’’ ANNs are trained on neurograms (i.e., the front end’s auditory-nerve representation) for the 0- and 80-ms end points of three voice-onset time continua (bilabial, alveolar, and velar), and tested on the intermediate (10, 20, ..., 70 ms) patterns. Replication of the behavioral data—in terms of warping of similarity space and shift of phoneme boundary with place of articulation—is robust in that it is maintained across a variety of ANN architectures and learning algorithms. Unlike real listeners, the synthetic listener can be analyzed easily and manipulated purposefully to reveal the acoustic and auditory features underlying the voicing contrast. A contribution analysis of the weights and activations of the ANNs highlights the auditory features. Replacement of the Pont–Damper front end by a simple Fourier-based spectral analysis abolishes the boundary-shift effect, although similarity warping is maintained. This suggests that restructuring of the acoustic features by the peripheral auditory system is crucial to phonetic classification.
Acoustic characteristics of the human paranasal sinuses derived from transmission characteristic measurement and morphological observation
Authors: Dang Jianwu , Honda Kiyoshi
Published: October 2005

This paper reports on the acoustic characteristics of the paranasal sinuses as determined from transmission characteristic measurements and morphological examinations. A new experimental approach was developed to explore the correspondence between antiresonance frequencies and the causal resonators [J. Dang and K. Honda, J. Acoust. Soc. Jpn. (E) 17, 93–99 (1996)], and it was adopted to determine the antiresonance frequency of each sinus cavity. In this study, the antiresonance frequencies and the locations of the sinus openings were estimated from transmission characteristics of the nasal tract for three subjects, and then MRI-based morphological data for the subjects were used to relate each antiresonance frequency to its causal sinus cavity. The results indicate that each of the three major sinuses, i.e., the sphenoidal, maxillary, and frontal sinuses, contributes its own antiresonances to the transmission characteristics of the nasal tract. The estimated antiresonance frequencies were compared with computed natural frequencies of Helmholtz resonators, and the differences were within 10% for the sinuses. On the basis of the frequency distribution of the sinus antiresonance, the acoustic characteristics of the paranasal sinuses were modeled by four Helmholtz resonators. The simulation with the four-zero model showed that the paranasal sinuses not only introduce antiresonances in the transfer function, but also change the spectral shape of the nasal formants.
Prosodic rules for the implementation of phrase boundaries in synthetic speech
Authors: Sanderman Angelien A. , Collier René
Published: October 2005

From previous research it is known that speakers use the prosodic cues pause and pitch to audibly structure their spoken messages. Listeners, on the other hand, use these phonetic cues to determine the degree of disjuncture in the flow of speech, which supposedly helps them to process the meaning of the utterances. In the research reported here, a professional speaker’s phrasing behavior was modeled in various sets of rules, corresponding to different levels of prosodic boundary strength. These phrasing rules were evaluated as to their acceptability and it appeared that several of them improve the quality of the synthetic speech. The rule set implementing five levels of boundary strength improved this quality more than rule sets with fewer levels. In fact, it appeared that this rule set produces synthetic speech which is prosodically almost as good as a copy-synthesis version with natural prosody.
Effect of relative amplitude, presentation level, and vowel duration on perception of voiceless stop consonants by normal and hearing-impaired listeners
Authors: Hedrick Mark S. , Jesteadt Walt
Published: October 2005

Previous work on the influence of relative amplitude and presentation level in listeners with normal hearing and those with sensorineural hearing loss was extended to include (1) a comparison of relative amplitude manipulation in the presence of neutral formant transitions versus manipulation of both formant transition and relative amplitude and (2) the additional variable of vowel duration. Synthetic consonant–vowel (CV) stimuli were used, and the amplitude of the burst relative to the vowel in the F4–F5 frequency range was varied across a 20-dB range using a /p–t/ contrast. In experiment I, some stimuli had neutral formant transition values and relative amplitude manipulations; other stimuli had both formant transition and relative amplitude manipulations. For stimuli in experiment II, neutral formant transition values were used, relative amplitude was manipulated, and vowel duration ranged from 14 to 200 ms. Results from experiment I showed no significant difference between listener groups when only relative amplitude information was manipulated, but significant differences when both relative amplitude and formant transition information was present. These results suggest that the listeners with normal hearing weighted the two acoustic cues differently from listeners with sensorineural hearing loss. Results from experiment II indicated that increasing vowel duration generally increased the number of labial responses from listeners with normal hearing, but did not always increase the number of labial responses from listeners with sensorineural hearing loss.
Landmark detection for distinctive feature-based speech recognition
Authors: Liu Sharlene A.
Published: October 2005

This work is a component of a proposed knowledge-based speech recognition system which uses landmarks to guide the search for distinctive features. In the speech signal, landmarks identify times when the acoustic manifestations of the linguistically motivated distinctive features are most salient. This paper describes an algorithm for automatically detecting acoustically abrupt landmarks. Some examples of acoustically abrupt landmarks are stop closures and releases, nasal closures and releases, and the point of cessation of free vocal fold vibration due to a velopharyngeal port closure at a nasal-to-obstruent juncture. As a consequence of landmark detection, the algorithm provides estimates of the broad phonetic class (articulator-free features) of the underlying segment. The algorithm is hierarchically structured, and is rooted in linguistic and speech production theory. It uses several factors to detect landmarks: energy abruptness in five frequency bands and at two levels of temporal resolution, segmental duration, broad phonetic class constraints, and articulatory constraints. Tested on a database of continuous, clean speech of women and men, the landmark detector has detection rates over 90%. A large majority of the detections were within 20 ms of the landmark transcription, and almost all were within 30 ms. The results are analyzed by landmark type and phonetic class.
Signal processing techniques for acoustic measurement of sperm whale body lengths
Authors: Goold John C.
Published: October 2005

Waveform cross correlation and cepstrum analysis were used to demonstrate possible techniques to measure pulse intervals within sperm whale sonar clicks. The structure of sperm whale clicks takes the form of a series of decaying broadband pulses separated by a time interval that is a function of sound velocity in spermaceti oil and the length of the spermaceti sac within the whales’ head. Click signals were bandpass filtered and waveform cross correlation used on the filtered signals to obtain maxima in the correlation function. Such maxima occur when successive pulses within the filtered click waveforms align after time shifting of the replica waveform by integer multiples of the interpulse interval. As an alternative approach, cepstrum analysis was used on the spectra of individual clicks, which were found to contain ripples with periods corresponding to the reciprocal of the interpulse interval. Variable signal quality lead to the conclusion that neither method was reliable for spot measurements of IPIs from individual clicks. However, calculating IPIs by either method for several hundred clicks in 6-min sequences, and smoothing the results with moving averages, allowed realistic mean values to be obtained and interpulse interval trends to be observed with dive time. Interpulse intervals were generally found to decrease with dive time, in accordance with known sound velocity characteristics of spermaceti oil under increasing pressure. Mean values of interpulse intervals obtained by cepstrum analysis for each click sequence were used to estimate body lengths of the respective animals.
The coupling of the A0 and interface Scholte modes in fluid-loaded plates
Authors: Dabirikhah H. , Turner Charles W.
Published: October 2005

In a recent Letter by Dickey et al., the velocity dispersion curve for the A0 mode for a fluid-loaded plate shows a splitting. However, this dispersion curve is incomplete and cannot reveal the reason for this anomaly. The nature of this anomaly has been investigated and the complete dispersion curve is presented. It is believed that this anomaly occurs because of the coupling of the A0, antisymmetrical zeroth mode, and the AS, antisymmetrical interface Scholte wave. The strength of this coupling depends on the ratio of the density of the loading fluid to the density of the plate material. The results of this numerical investigation for a solid plate loaded on the top and bottom surfaces with the same fluid and two different fluids are also presented.
Continuous structures as ‘‘fuzzy’’ substructures
Authors: Strasberg M.
Published: October 2005

This Letter discusses when and how the representation of Soize’s ‘‘fuzzy’’ substructures by a multitude of sprung masses can be applied to a single continuous or complicated substructure which itself has many closely spaced frequencies of resonance and antiresonance.
An evaluation of STAMINA 2.0 using the ORNAMENT ground attenuation algorithm
Authors: Herman Lloyd A.
Published: October 2005

The United States Federal Highway Administration (FHWA) computer program STAMINA 2.0 was evaluated with the ground attenuation algorithm used in the Ontario Road Noise Analysis Method for Environment and Transportation (ORNAMENT). The goal: To determine this method’s potential to improve noise level predictions at receivers located near traffic roadways with barriers in the propagation path. The study of 41 sites indicated that this method reduced the mean overprediction of noise levels from 2.6 dB using STAMINA 2.0 alone, to 0.5 dB using STAMINA 2.0 with the ORNAMENT ground attenuation algorithm.
An illusory transfer of temporal gaps between crossing tones
Authors: Kanafuka Kyoko , Tanaka Shunsuke , Nakajima Yoshitaka , Sasaki Takayuki
Published: October 2005

When a long ascending tone with a temporal gap in the middle and a short continuous descending tone cross each other, a long continuous ascending tone and a short descending tone are often perceived with a temporal gap in the middle. This is an illusory phenomenon because the physical temporal gap of the ascending tone seems to be transferred to the descending tone perceptually. This phenomenon was named ‘‘the gap transfer illusion.’’ Many related stimulus patterns were compared which confirmed the robustness of this illusion. An onset and a proximate termination tended to be connected to each other perceptually, forming an auditory event. The proximity of the temporal dimension seemed more important than the proximity of the frequency dimension. When onsets had no proximate termination, the illusion disappeared. Several new phenomena to which the same idea can be applied were found.
Modeling issues for fuzzy internals connected to the master structure at multiple points
Authors: Pierce Allan D.
Published: October 2005

The present paper considers a sequence of modeling problems involving idealized mechanical attachments that are connected to an external elastic structure at either a single point or a number of points. Previous analyses for cases when the attachments were rods and beams are reviewed along with the fuzzy-structure idealization that considers the mass per unit natural frequency to be continuous, and in which the asymptotic high-frequency limit is taken. Although the first approximation to this limit leads to a divergent integral for the frequency-dependent effective mass contributed by the internal structure, it is possible to obtain realistic high-frequency limits with incorporation of the knowledge of the total mass of the appendage. The remainder of the paper concentrates on two-dimensional elastic bodies, especially plates, that are connected to the master structure at one or more places, and explores the asymptotic tendencies of the elements of the modal mass matrix and the circumstances in which the number of attachments may be effectively blurred as a continuous distribution of nonconnected one-dimensional fuzzy attachments. [Work supported by the Office of Naval Research.]
When is a ‘‘fuzzy’’ substructure not a fuzzy substructure?
Authors: Strasberg M.
Published: October 2005

A source of confusion in discussions of ‘‘fuzzy’’ substructures is use of the same term for two types of structures with different characteristics. The fuzzy substructures discussed, e.g., by Strasberg and Feit [J. Acoust. Soc. Am. 99, 335–344 (1996)] have a multitude of closely spaced, deterministic frequencies of antiresonance and sufficient damping so that ‘‘modal overlap’’ exists over the frequency range of interest. On the other hand, Soize and other investigators consider substructures described statistically in terms of an ensemble of substructures, each member substructure having many antiresonance frequencies which do not necessarily have modal overlap, with their antiresonance frequencies and modal masses specified by probability distributions. Although the real, dissipative portions of the impedance determined for these two types of substructure are similar, the nondissipative, reactive portion may be different, resulting in significant differences in transient response. Moreover, the response determined for the ensemble of substructures may differ significantly from that of individual member substructures. Examples of the differences will be displayed.
Wavelet and Melnikov function analysis of oscillations of a submarine
Authors: Zeitlin Michael G. , Fedorova Antonina N.
Published: October 2005

The explicit time description, computer modeling, and Melnikov function approach for the problem of detecting signals from oscillations of submarines are given. This problem in the Galerkin approximation is reduced to the problem of solving the system of differential equations with polynomial nonlinearities and variable coefficients. The first part of the construction is a variational approach to this problem, which reduces the initial problem to the problems of the solution of functional equations and algebraical problems. In the general case the solution is represented in a compactly supported wavelet basis. Multiresolution expansion is the second part of the construction. The solution is parametrized by solutions of two reduced algebraical problems: One is a polynomial algebraical system of equations, and the second is some linear problem, which is obtained from one of the following wavelet constructions: the fast wavelet transform, stationary subdivision schemes, and the method of connection coefficients. This solution, computer calculations, and the Melnikov function analysis offer the possibility of analyzing, in the parameter space, the time evolution of the system and the corresponding transition between different regimes: deterministic, quasiperiodic, and chaotic dynamics.
Experimental study on sound and vibration generated by a valve in a water piping system
Authors: Murayama Chie , Ohkawa Heiichiro , Kaite Masahiro , Hiramatsu Tomotaka
Published: October 2005

Various acoustic troubles are caused by the structure-borne sound generated by water piping systems connected with pumps at hotels or condominiums. Sound and vibration generated by water piping sytems are uncomfortable factors in living environments which require silent conditions. In order to predict and control sound and vibration generated by water piping systems, it is necessary to obtain some data about acoustic characteristics of sound and vibration sources such as pumps and valves. Vibration of pipe walls and sound in water generated by valves according to flow quantity obtained by pump rotation control and valve control, and also water temperature, were measured, and the basic tendencies of sound and vibration characteristics were obtained. In this paper, these investigative results are described.
Diphthong dynamics in Swabian
Authors: Geumann Anja , Hiller Markus
Published: October 2005

The vowel quality in some diphthongs of Swabian (a dialect of German) was determined by measurement of first and second formant values. A minimal contrast could be shown between two different diphthong qualities /æe/ and /■i/, where for standard German only one is assumed, viz., /ai/. The two diphthong qualities differ only slightly in onset and offset vowel quality, so a better understanding of their relation was expected from an examination of the diphthong dynamics. Formant values (first and second formant) were taken at eight equidistant points during the diphthongs. The results suggest that there is a significant difference in the temporal patterns of the two diphthongs. Where /æe/ could be broadly described by a steady state onset to glide pattern, the other one is more adequately characterized as a glide to steady-state offset pattern. This dynamic aspect seems to be a very important field in need of further investigation into other diphthong pairs in Swabian, as in other languages.
Pitch dependency of formants and its individuality in Japanese stationary vowels
Authors: Kuroda Masahiro , Miki Nobuhiro , Ogawa Yoshihiko
Published: October 2005

It has been known that pitch frequency variation influences the formant frequency. In conventional studies, the data during a few pitches are regarded as stationary signals. If the estimated spectrum is the average spectrum in a few pitches, this cannot take into account the transition of formant frequencies and the influence of variation of pitch frequencies upon the vocal transfer function in each pitch period. The electroglottographic (EGG) is employed with speech data in order to cut the corresponding waveform of the glottal closure in each pitch cycle. The vocal tract transfer function of the Japanese back-low vowel /a/, front-high /i/ and back-high /u/ was estimated for Japanese adults using the Fejér kernel mapping for short time analysis estimates fundamental frequencies and glottal closure quotients also. The facts that the variation of formant frequency is related to that of pitch, and that variation has an individual difference are discussed.
Evaluation of the trajectory of vowel glide based on vector analysis
Authors: Toyama Jun , Shimbo Masaru
Published: October 2005

Although coarticulation frequently causes insertion and deletion errors in speech recognition systems, human beings can recognize it correctly, presumably because of correct estimation of glide between phonemes. The present study is concerned with the glide characteristics of vowels, and it is shown that the trajectory of the glide appears along geodesics calculated by the metric on a phonetical vowel space. First the spectrum envelope of the stable part of a vowel is digitized by representing it by a 40-dimensional vector. Then, this vector is projected to a three-dimensional parameter space by discriminant analysis. The trajectory of glide generally produced a curve, which suggests that the space is non-Euclidean, as the trajectory should be the shortest path in the space. Assuming that the distribution of each vowel is a normal distribution, it is shown that it is an ellipsoid with a covariance matrix. Second, by a transformation using metrics, based on the variance of vowels, the trajectory is transformed into Euclidean space. It is shown that the transformed trajectory of the vowel glide is a similar straight line.
Effects of formant contour on the perception of synthesized /hVd/ syllables
Authors: Hillenbrand James M. , Nearey Terrance M.
Published: October 2005

To study the role of formant frequency movements in vowel perception, 300 utterances were selected from a database consisting of 1668 /hVd/ syllables spoken by 45 men, 48 women, and 46 children [Hillenbrand et al., J. Acoust. Soc. Am. 97, 3099–3111 (1995)]. Measurements from these signals were used to synthesize two versions of each utterance: (1) an ‘‘original formant’’ version that followed the measured contours of F1−F3, and (2) a ‘‘flat formant’’ version with F1−F3 fixed at steady-state values. Results indicated: (1) the original, naturally produced signals were identified with greater accuracy than either of the synthetic versions and (2) the original-formant synthetic signals were identified with much greater accuracy than the flat-formant signals. Discriminant analysis methods of Nearey and Assmann [J. Acoust. Soc. Am. 80, 1297–1308 (1986)] show that differences between listeners’ responses in the flat formant and either the natural or the original formant stimuli are significantly correlated (p<0.01, by randomization tests) with predictions from a simple nucleus +offglide model of vowel trajectories. Logistic regression analysis of this data is reported in a companion paper, 3aSC13. [Work supported by NIH and SSHRC.]
Vowel environments in minimal pair sequences
Authors: Trout J. D.
Published: October 2005

An earlier study replicated the global fundamental frequency declination effect over the minimal pair sentences used here. In order to test for local influences near word boundaries, three classes of measurements (duration, mean fundamental frequency, and fundamental frequency standard deviation or movement) were made on vowels that immediately preceded critical segments in minimal pairs (He found the fur meager/He found the firm eager). Preliminary results suggest that important features of the preceding vowels in minimal pairs survive differences of word boundary position; those features of preceding vowels do not provide general, cross-contextual cues to presence of word boundary. Further analysis using a speech perception task will determine whether listeners can identify which sentence of the minimal pair has been uttered. [Work supported by NIH.]
Discrimination of geminate and single stops in Japanese by five different language speakers
Authors: Minagawa Yasuyo
Published: October 2005

Thirty Korean, Chinese, Thai, American, and Spanish, subjects heard two-syllable Japanese words including geminate or single stops in two pitch accent contexts. The subjects discriminated between geminate and single stops. Their discrimination ability was examined and compared among these different speakers. Five language groups were clearly separated into two groups according to the results. The first group (Korean, Chinese, and Thai) that had three-way stop contrasts different in tense or aspiration, showed a lower correct ratio. Also pitch accent (LH, HL) and consonant type (/p/,/t/,/k/) of the stimuli appeared to influence the discrimination similarly in this group. The second American and Spanish group could perform the discrimination task better than the first group and little influence of accent and consonant contexts were observed. The second group seemed to depend more on the absolute closure duration of stop consonants in judging geminate or single. These results suggest that different language speakers use different acoustic cues in discriminating non-native contrast. In particular, while Japanese speakers discriminate single stops from their geminate counterparts using only relative segmental duration, Korean, Chinese, and Thai speakers seem to use additional acoustic information such as tense and aspiration.
Articulatory characteristics of French consonants are maintained after the loss of schwa
Authors: Steriade Donca , Fougeron Cécile
Published: October 2005

The optional loss of schwa in French generates consonant clusters. Thus de rôle [d■rol] can be produced as d’role [drol] and is thought to become homophonous with drôle [drol]. This study compared articulatory and acoustic properties of [d] in these sequences in order to determine whether [d] maintains, after the loss of schwa, certain characteristics of the isolation form de. The sequences were inserted in a carrier sentence and repeated 20 times by three French speakers. Consonants were compared in terms of amount and location of linguopalatal contact (measured with EPG), their acoustic duration, and frequency of lenition (absence of closure). Preliminary results for one subject show that despite the complete disappearance of schwa, the [d] ind ’rôle has some of the same characteristics as the one in de rôle as opposed to drôle: longer duration, greater and more anterior linguopalatal contact, less lenition. These results suggest that the loss of schwa does not lead to resyllabification of [d] with either the preceding or following syllable, but rather that the consonant maintains the characteristic properties of the citation form. The significance of this result is that lexical distinctions are maintained in production, despite the apparent phonological neutralization induced by schwa loss.
An improvement of word recognition by using cues of place-of-articulation
Authors: Dutono Titon , Watanabe Akira , Ikeda Naomitsu
Published: October 2005

In general, the implementation of several parameter groups is seen to be effective in clearing up the nature of a phoneme, because each of them might have a typical feature which does not appear in the others. A speech visualization system has been developed as a speech training aid for the hearing impaired. Beside the formant tracker, the filter bank, and the neural network for manner-of-articulation, the system adopts neural networks for extracting cues of place-of-articulation. It was shown in a previous experiment that the information extracted by all of those devices are very useful for human judgment of the visual pattern. In this study of word recognition, in which the vocabulary is freely and easily constructed, an attempt was made to add cues of place-of-articulation to improve the performance. The results show that the addition of parameters was very effective, and the recognition rates using the proposed parameters were much higher than the usual rates. When 50 place names uttered by 30 males from the JEIDA database were tested, the recognition rates were 97.8% in the Bayes-decision-rule distance.
Reassessment of the syllable articulation score in consideration of place and manner of articulation
Authors: Takayasu Noriyuki , Suzuki Jouji , Yashima Hiroyuki
Published: October 2005

An important and subjective scale to measure the quality of speech is the syllable articulation score (AS). In the calculation of AS, all hearing errors are counted by the equal weight. Confusion usually occurs between phonemes having common characteristics, but confusion between quite different phonemes is observed when synthetic voice or an exotic voice is assessed. This paper proposes a three-dimensional articulatory model of consonants. The position of phonemes is assigned considering the place and manner of articulation. The weighted syllable articulation score (WAS) is estimated utilizing the distance between the phonemes uttered and the responded in this model. WAS of speech processed with HPF (high-pass filter) is 7% lower than that with LPF (low-pass filter), although their ASs are almost the same. Helium speech is quite unnatural and less intelligible than general speech. It is shown that the WAS of helium speech is low in comparison with the WAS of speech processed with LPF or HPF, although their ASs are almost common. Assessments by WAS of synthetic voice and various distorted speech will be presented. The final goal of this study is to make clear the relation between WAS and other subjective measures such as MOS (mean opinion score).
Speech recognition with altered spectral distribution of envelope cues
Authors: Shannon Robert V. , Zeng Fan-Gang , Wygonski John
Published: October 2005

Recognition of consonants, vowels, and sentences was measured in conditions of reduced and altered spectral information. Speech materials were passed through four bandpass filters (analysis bands). The output of each band was half-wave rectified, and low-pass filtered at 160 Hz to extract the temporal envelope. The envelope from each band modulated a bandlimited noise (carrier bands). Experiment I explored the importance of the cutoff frequencies of the bands, which proved not to be a critical parameter for speech recognition as long as the analysis and carrier bands were the same. Experiment II warped the spectral distribution of envelope cues by mismatching the analysis and carrier bands in frequency extent, resulting in a dramatic decrease in performance. Experiment III simulated the spectral smearing caused by broad auditory filters or electrode interaction in a cochlear implant by decreasing the slopes on the noise carrier bands. A decrease in performance was observed only for the most broadly overlapping carrier bands, indicating that speech pattern recognition is robust to spectral smearing. Overall, these results show that, for four bands, the most critical parameter is the frequency alignment of the analysis bands and carrier bands. [Supported by NIDCD.]
Broad-phonetic categorization of speech in cochlear-implant users
Authors: Carrell Thomas D. , Koch Dawn , Tremblay Kelly , Kraus Nina
Published: October 2005

Many studies have examined the ability of cochlear implant recipients to perceive words and sentences. The findings have frequently been analyzed within the structure of traditional linguistic units such as phonemes and features. However, it is possible that these units do not accurately represent the type of acoustic information encoded and transmitted by cochlear implants. In an alternate system of representation, broad phonetic categories (BPCs) parse the speech stream into acoustically based units [D. Shippman and V. Zue, IEEE ICASSP, Paris (1982)]. Although BPCs were developed to aid computer speech recognition systems in the face of ambiguous phonetic information, they may also be valuable in modeling the perception of words in cochlear implant recipients. The present experiment investigated whether BPCs provided a better representation of the speech signal in cochlear implant recipients than phonemes and features. Subjects were adults who had been implanted with either the Nucleus-22 or Clarion cochlear implant. A version of the modified rhyme test was used to assess phoneme and BPC recognition in word context. These results were then compared to the listeners’ scores on the CUNY (auditory-only) sentence test because of its closer relationship to everyday listening experiences. [Work supported by NIH.]
Perception of waveform envelope cues in speech and speech-correlated-noise by cochlear-implant users
Authors: Fox Robert Allen , Bicknell Ina , Welling D. Bradley
Published: October 2005

Previous studies of speech perception by cochlear-implant users have employed stimuli in which both waveform envelope and spectral (frequency-specific) information is present. It is unclear as to how much spectral (i.e., frequency-sensitive) information cochlear implant users may utilize in their perception of speech stimuli. In this study, the identification and discrimination of CV tokens are contrasted by normally hearing and cochlear-implant subjects in two conditions: CV tokens (digitized and normalized) and their speech-correlated-noise counterparts (SCN). In the SCN stimuli, frequency-sensitive information has been eliminated while waveform envelope information has been preserved. Preliminary analysis of the identification data indicates that normal-hearing subjects were able to identify unprocessed (UP) stimuli at a level of nearly 100% correct (96.1%), while the identification of cochlear-implant subjects was lower, but above chance (30.6%). However, absolute identification of consonants was better for cochlear-implant subjects than normal-hearing subjects in the speech-correlated-noise condition. Overall, the difference between the two groups in making manner, voicing, and place-of-articulation decisions indicated that these cochlear-implant users may have primarily utilized nonspectral information in their identifications and discriminations, although they are able to utilize some spectral information. [Work supported by the Deafness Research Foundation.]
Phonetic mechanisms of post-nasal voicing
Authors: Hayes Bruce P.
Published: October 2005

Many languages systematically replace voiceless stops by voiced stops when the immediately preceding sound is nasal. This is estimated to occur in 7% of all languages, and thus is likely to reflect preferences of the speech production mechanism. Post-nasal voicing can be explained by two factors. Coarticulation with an adjacent nasal induces nasal leak during a stop, which vents oral pressure and tends to maintain the transglottal pressure drop needed to sustain voicing. Coarticulation also causes the velum to rise or fall during the stop while the velar port is closed. This creates a pumping effect (rarefaction in post-nasal stops, compression in pre-nasal), which facilitates voicing for post-nasal stops and inhibits voicing for pre-nasal stops. The cross-linguistic pattern of post-nasal voicing reflects the combined effects of nasal leak and rarefactive velar pumping; the absence of pre-nasal voicing reflects the antagonistic effects of nasal leak and compressive velar pumping. To test this hypothesis, an aerodynamic vocal tract model [J. Westbury and P. Keating, J. Linguist.22 , 145–166 (1986)] was used. Over a range of estimated values for nasal leak and velar pumping, the model output a strong preference for extra voicing post-nasally, and no preference for pre-nasal voicing.
Perception of final consonant voicing in speech produced during simultaneous communication
Authors: Metz Dale Evan , Lessler Amy , Lawe Yvonne , Schiavetti Nicholas , Whitehead Robert L. , Whitehead Brenda H.
Published: October 2005

The temporal structure of speech is altered dramatically when a person uses sign language and speaks aloud simultaneously (i.e., simultaneous communication or SC). Notable among these temporal alterations are durational increases of vowels, pause times between words, and voice onset times. Temporal aspects of speech play a pivotal role in the perception of certain phonemic contrasts in spoken English. For example, cues for the perception of final consonant voicing are carried in the vowel that precedes the final consonant. The purpose of the present study was to investigate the effect of increased segment durations associated with speech produced during SC on final consonant voicing perception. Eight skilled SC users produced naturally spoken words that differed only in the voicing characteristic of the final consonant. The words were recorded under two conditions: (a) speech alone and (b) SC. Digital editing was used to remove the final consonant. The digitally altered words were played to 20 listeners who, in a forced-choice paradigm, circled the word they thought they heard. The listeners accurately identified final consonant voicing of 69.2% of the target words produced in the speech alone condition and 77.0% of the target words in the SC condition, a nonsignificant difference.
Mapping multidimensional perceptual consonant space for place contrasts
Authors: Pastore Richard E. , Liberto James W. , Crawley Edward J.
Published: October 2005

The nature and interaction of known cues for the perception of place distinctions were evaluated with a number of behavioral measures. The perceptual role of release burst and both F2 and F3 onset transitions were manipulated factorially in each of several vowel contexts. Within each vowel context, perception was evaluated using classification (labeling) tasks and category goodness ratings for each CV syllable, as well as ratings of similarity between pairs of CV syllables both within and across perceptual categories. For each of the vowel contexts, the results exhibit a generally high degree of consistency across the various behavioral measures. The findings provide complex, multidimensional perspectives of perceptual consonant space for each vowel. Within the perceptual space representations, some category boundaries are simple, whereas other boundaries are complex, being defined interactively across several variables. Perceptual quality or category goodness is seen to vary systematically within categories. Although the representation of the perceptual spaces are relatively complex, cross sections of the space are consistent with findings from earlier investigations, which utilized labeling tasks for unidimensional stimulus variables. [Research supported by a grant from AFOSR.]
Speech recognition in aphasia: An analysis of stop consonant confusions
Authors: Goldberg Lyn
Published: October 2005

The pattern of phoneme recognition and phoneme confusions shown by fluent and nonfluent aphasic and nonaphasic subjects with regard to place of articulation and voicing was examined in a listening task using synthesized speech stimuli. Subjects were asked to recognize six stop consonant vowel syllables (/pa,ba,ta,da,ka,ga/) under three formant frequency conditions (synthesized presentation of F1, F2, and F3; attenuation of F2; and attenuation of F3). A series of confusion matrices was developed to document the voicing and place of articulation responses made by each group across all formant conditions. Analysis of phoneme recognition and phoneme confusions revealed that nonaphasic errors centered on substitutions along the place of articulation continuum. Both fluent and nonfluent aphasic groups demonstrated this place of articulation substitution pattern, but with more numerous errors, additional substitutions of voice, and combinations of voicing and place of articulation errors, particularly in the nonfluent aphasic group. Implications of the results are discussed.
Applications of microphone arrays to speech processing
Authors: Hoffman Michael W. , Li Zhao
Published: October 2005

Arrays of simple, unmatched microphones have been applied to two speech processing problems. The first application is noise reduction via robust constrained beamforming. Significant noise reduction has been observed (10 dB) in typical rooms at a source-to-array distance of 1 m. The noise reduction achievable is proportional to the direct-to-reverberant energy ratio of the interfering sound energy (from a single location in the room). The robust adaptive processing avoids the cancellation of the desired signal in all cases examined. The second application is speech coding. A standard code-excited-linear-prediction (CELP) codec is used in conjunction with the microphone array and noisy speech inputs. The quality of the desired speech coded in the presence of acoustic interference is enhanced by the array processing. Signal-to-noise ratios, intelligibility averaged gains, other spectral measures, and spectral envelope measures have been compared to assess the relative quality of the spatially filtered and coded noisy speech. [Work supported by NSF Grant No. IRI-9405286.]
Adaptive microphone arrays for speech enhancement in coherent and incoherent noise fields
Authors: Fischer Sven , Kammeyer Karl-Dirk , Simmer Klaus Uwe
Published: October 2005

To enhance the quality of speech in a hands–free telecommunication environment, microphone arrays consisting of multiple microphones can be used for sound pick–up. To circumvent the limitations associated with small array apertures adaptive signal processing techniques can be used. For extreme situations of a free (coherent) and a diffuse (incoherent) noise field, acceptable solutions for the noise reduction problem exist: In coherent noise fields, the adaptive beamforming techniques yield a high-noise reduction performance, but loose their efficiency in incoherent noise fields. On the other hand, for incoherent noise fields microphone arrays with adaptive post–filtering yield good performance; but in a coherent noise environment this method may cause unacceptable distortions in the output signal. In this contribution the advantages and disadvantages of these two techniques for noise reduction are discussed. A technique is presented which operates independently of the correlation properties of the noise fields, i.e., this method is able to reduce coherent and incoherent noise simultaneously. Some results are given using objective quality measures as well as subjective listening tests. The methods are also compared by using the recognition rate of a speaker-independent isolated word recognition system.
Near-field broadband array beamforming using wave equation transformation
Authors: Kennedy Rodney A. , Thushara P. , Abhayapala D. , Ward Darren B.
Published: October 2005

Given a broadband frequency versus angle beampattern specification in the near field, a methodology is developed to yield a filter-bank design for an array signal-processing structure. This methodology relies on two key tools: (i) a transformation using spherical harmonic solutions to the wave equation which takes a near-field beampattern specification and maps it to an equivalent far-field beampattern specification for a given frequency; and (ii) a solution of the broadband frequency versus angle far-field beampattern specification problem, which generalizes earlier results on broadband frequency invariant (far field) beamforming [Wardet al., J. Acoust. Soc. Am. 97, 1023–1034 (1995)]. The transformation exploits the uniqueness property of the solution to the wave equation to give a set of truly equivalent far-field patterns for a discrete set of frequencies. There is no restriction on the bandwidth of operation beyond that imposed by limitations on the number of sensors or filter complexity. A straightforward filter-bank design is then proposed to realize this equivalent far-field broadband frequency versus angle beampattern specification. The methodology is illustrated with a broadband frequency invariant near-field design which captures the full generality of the design methodology.
Optimum gains of a delay-and-sum microphone array for the near sound field
Authors: Nomura Hiroaki , Kaneda Yutaka , Kojima Junji
Published: October 2005

The optimum gains of a delay-and-sum microphone array for the near sound field, where the sound waves are spherical, have been investigated. The optimum gains give the highest signal-to-noise ratio in a sound field with background noise, and they give constant sensitivity to the sound source at every focal point. The optimum gains were theoretically derived as gi=C/ri, where ri represents the distance between the array focal point and the ith microphone element and C represents the normalizing parameter that keeps the sound source sensitivity constant, independent of the focal point. Parameter C is a function of ri and rc, where rc is the critical distance of the reverberant room. The results of computer simulation and indoor experiments (room volume: 83 m3, reverberation time: 0.2 s) showed a good array sensitivity distribution and power distribution estimation when each gain was optimized.