bluuebunny/crossref_metadata_2025 huggingface dataset:
Automatic generation of word models using piecewise linear segment lattices
Authors: Kojima Hiroaki , Tanaka Kazuyo
Published: October 2005

A framework for ‘‘phonological concept formation’’ has been proposed, aiming to generate robust speech recognition models [Kojima et al., Proc. ICSLP 92, Vol. 1, pp. 269–272 (1992)]. For this purpose, a ‘‘piecewise linear segment lattice’’ model is proposed. The structure is represented as a lattice of segments, each of which is represented as regression coefficients of feature vectors within the segment. Compared with typical stochastic models like HMM, the advantages are: (1) It needs fewer samples to learn; (2) it represents objects in voluntary precision; and (3) its structure can be dynamically changed by less calculation. An outline of the generation algorithm is as follows: (1) Dividing each sample into segments using DP, where the number of segments is decided based on an MDL-like criterion; (2) matching between the sequences of segments within the same word by DP; (3) modifying the division according to their matching scores; (4) picking up similar (i.e., near) subsequences and gathering them into a phonelike cluster. Speaker-independent isolated word recognition is carried out using the proposed models which are generated in several conditions. The results show that the recognition rate is improved by forming phonelike clusters.
Preliminary report on syllable level organization observed in Parkinsonian speech obtained in individuals before and after posteroventral pallidotomy
Authors: Wang Q. Emily , Shannon Kathleen
Published: October 2005

Syllable level organization has been evidenced in the articulatory movement patterns in different languages [C. P. Browman and L. Goldstein, Producing Speech: Contemporary Issues, 19–34 (1995); R. A. Krakow, ‘‘The articulatory organization of syllables: A kinematic analysis of labial and velar gestures,’’ Ph.D. dissertation, Yale University (1989); Q. E. Wang, ‘‘Are syllables units of speech motor organization?—A kinematic analysis of labial and velar gestures in Cantonese,’’ Ph.D. dissertation, University of Connecticut (1995)]. This study analyzed speech samples produced by nondemented individuals with idiopathic Parkinson’s disease (Hoehn and Yahr stage 2–3) who underwent posteroventral pallidotomy. The data were collected with the patients on and off their medications as well as pre- and post-operatively. The preliminary results indicated that the patients were able to produce stimuli with syllable-initial nasals with less difficulty than those containing syllable-final nasals, and as the patients’ motor performance improved, their ability to produce the stimuli with syllable-final nasals also improved. This may suggest that the speech motor programming and execution are different for the phonemically identical phonemes in syllable-initial and syllable-final positions. [Work supported by Rush University and NIH Grant DC-00121 to the Haskins Laboratories.]
Are selective adaptation effects independent of cognitive load?
Authors: Kat Donna , Samuel Arthur G.
Published: October 2005

Selective adaptation occurs through the repeated presentation of a sound (the ‘‘adaptor’’), and leads to a reduction in the perception of similar sounds. Adaptation has been used to investigate the nature of early speech representations. Work in this laboratory has recently demonstrated that perceptually restored phonemes can produce reliable adaptation effects, and that these effects are occurring at relatively early levels (e.g., phonemic rather than lexical) of processing [A. G. Samuel, CognitivePsychology (in press)]. The current study is designed to determine whether the adaptation effects are so low-level and automatic that they do not require cognitive resources. There are three conditions in the current study: (1) adaptation alone (control), (2) adaptation during continuous arithmetic problems, and (3) adaptation during continuous rhyming judgments (presented visually). Preliminary results indicate that continuously solving arithmetic problems does not reduce the adaptation effect, indicating no general cognitive involvement in adaptation. The rhyming task tests for any more specific involvement of language processors. Immunity to this secondary task would support a very low-level, automatic basis for adaptation. [Work supported by NIMH.]
The unit responses to various complex sounds in the auditory cortex of the Japanese monkey
Authors: Kamada Tsutomu
Published: October 2005

It is well investigated in the mammalian cortex that physiological parameters like tonotopicity and binaural response properties are represented in the primary auditory cortex, while the coding mechanism of complex sounds is not always understood in the primary as well as secondary cortex. In these circumstances it is still necessary to examine the response properties of neurons in a wide area of auditory cortices. The author has examined the neuronal responses to monkey vocalizations as well as pure tones, white noise, and synthetic sounds in the auditory cortex of the Japanese monkey in the chronic condition. More than 200 hundreds of units were recorded, most of which responded to either of the sound stimuli while the specific response to a sound was rare. Sharp tuning units with clear characteristic frequencies were located mainly in the primary cortex, while broad tuning ones were in periphery of the koniocortex. Some neurons responded, like the frequency analyzer when examined, to ‘‘coo’’ vocalizations and to synthetic harmonic components of the vocalization. Other responses were compared to acoustic structures of the stimuli. The correlations between stimuli and neural responses were discussed in association with the location of the neurons in the cortex.
Recent advances in exterior and interior NAH at NRL
Authors: Williams Earl G.
Published: October 2005

Two significant recent advances in near-field acoustical holography are discussed. First, the formulation and results are presented for interior NAH used for the reconstruction of fuselage vibration, interior pressure, and acoustic intensity in an aircraft fuselage. The formulation includes derivation of a new Neumann Green function which closely matches the physics of the fluid/structure interaction and is extremely efficient for solving the inverse problem solved by NAH. Second, pertaining to exterior NAH, a method is presented which processes NAH data in order to localize and quantify radiating ‘‘hot spots’’ on cylindrical and plate structures, identifying and source ranking (using supersonic intensity) regions of the surface of the structure which radiate to the far field. The derivation of the method is rigorous, and ranks the hot spots according to their fractional (or percentage) contribution to the total radiated power. Corroborating results are presented for experiments on an internally excited, complex shell structure underwater.
The use of near-field acoustical holography (NAH) and partial-field decomposition to identify and quantify the sources of exterior noise radiated from a vehicle
Authors: Takata Hiroshi , Nishi Takuo , Jiang WeiKang , Bolton J. Stuart
Published: October 2005

Since power-train noise sources are ‘‘hidden’’ within the engine compartment, it is difficult to use NAH to identify the sources and the associated partial radiation fields that create the exterior noise field radiated by a motor vehicle. Integrated near-field acoustical holography (INAH) has been developed to address this concern. INAH represents a combination of NAH, reference microphone selection procedures, and coherence techniques. The procedure entails sensing the sources inside the engine compartment by using an array of reference microphones, and then calculating the associated partial sound fields by using NAH. A key factor in the success of this procedure is the selection of a good reference microphone subset. A selection procedure has been developed by combining condition number and coherence analyses. The partial field determination problem has been approached by using both partial coherence and singular value decompositon (SVD) procedures. It has been found that the partial coherence technique maintains a closer relationship with the physical partial fields than does the SVD procedure, and thus conveys more information than the latter technique. In this presentation, INAH will be demonstrated through identification of the hidden sources and the partial fields that contribute to the exterior noise field of a vehicle.
Application of near-field holograms that contain noise to predict the far-field scattered pressure of a finite cylinder with spherical endcaps
Authors: Hughes Ronald G. , Carroll Gerard P.
Published: October 2005

Near-field acoustic holography has been used in several studies to understand the radiation characteristics of various submerged objects. Further, the process has also been used to a lesser extent to examine acoustic scattering. Near-field holograms in either radiation or scattering studies are degraded by errors associated with background noise and sensor placements. In addition to these sources of error, the application of near-field holography to characterize the scattered field are further degraded by the requirement to remove the incident field from the scattered field. The significance of placement errors, background noise, and the removal of the incident field from the scattered field are presented for the case of a cylinder with spherical endcaps. The velocity and pressure distributions on the cylinder surface were calculated using SARA 2D and the pressure field was calculated using a Helmholtz integral method. Quantitative predictions of the degradation of the far-field pressure for a cylinder with spherical endcaps based upon reconstructions from near-field hologram planes that include various types of errors will be presented. [Work supported by ONR.]
The phonatory pressure-flow-adduction space
Authors: Scherer Ronald C. , Alipour Fariborz , Guo Chwen G.
Published: October 2005

Relationships among average subglottal pressure, average glottal airflow, and glottal adduction (vocal process gap), variables related to glottal competence, were obtained for one human subject. The data define the pressure-flow-adduction space. Results are similar to those of a study using excised canine larynges [Alipour et al., ‘‘Pressure-flow relationships during phonation as a function of adduction,’’ J. Voice (in press)] especially for the differential flow resistance as a function of adduction. For set values of adduction, glottal flow resistance increased by about 2.5 times between subglottal pressure values of 5 and 15 cm H2O. At set values of subglottal pressure, flow resistance increased by a factor of approximately 9.5 between pressed and breathy phonation. Any particular flow resistance value does not predict a particular adduction value due to the dependence on subglottal pressure. However, a particular subglottal pressure, glottal flow pair appears to predict a particular adduction level. Subglottal aerodynamic power increases with subglottal pressure at set levels of adduction, increasing by a factor of about 3.7 between 5 and 15 cm H2O. For a specific value of subglottal pressure, aerodynamic power increases as adduction decreases. [Work supported by NIH Grant No. P60 DC00976.]
Measurements of subglottal resonances and their influence on vowel spectra
Authors: Hanson Helen M.
Published: October 2005

Data collected from several speakers are being studied to estimate subglottal resonant frequencies. The utterances were ten vowels of American English embedded in the nonsense syllable /■hVd/. Sound-pressure signals were recorded by a microphone, and signals transduced by an accelerometer placed on the neck below the larynx were recorded simultaneously. In the accelerometer signal the subglottal resonances appear as well-defined spectral peaks, whereas in the microphone signal these resonances are manifested as small perturbations in the spectra. These perturbations are more pronounced for a speaker who, based on other spectral data, phonates with a significant glottal chink [H. M. Hanson and K. N. Stevens, Proc. ICPhS 95, Stockholm]. Comparison of spectra of the accelerometer and sound-pressure signals for two female speakers has resulted in estimated subglottal resonant frequencies of 625, 1540, and 2110 Hz. The bandwidths of the first subglottal resonances have also been estimated for the two speakers at 110 and 210 Hz, respectively. These results are in keeping with previous reports. Discontinuities in apparent formant movements have been observed in the vicinity of the estimated subglottal resonances, suggesting that resonances of the subglottal system can influence the acoustic characteristics of vowels. [Work supported by NIH Grant Nos. DC00205-02, DC00075.]
The normal modes of vocal fold tissues
Authors: Berry David A. , Titze Ingo R.
Published: October 2005

The Ritz method is used to calculate the eigenmodes and eigenfrequencies of a 3-D vocal foldlike continuum. The investigation represents a rectification and extension of previous studies [I. R. Titze and W. J. Strong, J. Acoust. Soc. Am. 57, 736–744 (1975); I. R. Titze, J. Acoust. Soc. Am. 60, 1366–1380 (1976)], emphasizing the indispensability of utilizing natural boundary conditions when computing the characteristic modes of a system. Concurring with previous assertions, two of the eigenmodes are theorized to play a major role in facilitating self-oscillation of the vocal folds. For the case of incompressible tissues, the eigenfrequencies of the two eigenmodes are nearly identical throughout a large parameter space spanned by the mechanical constants. If truly indicative of mechanisms governing vocal fold dynamics, it may help explain why the two eigenmodes entrain so naturally across a wide range of phonatory adjustments.
Optimal glottal width and convergence angle for ease of phonation
Authors: Lucero Jorge C.
Published: October 2005

Recent experimental studies [I. R. Titze et al., J. Acoust. Soc. Am. 97, 3080–3083 (1995); R. W. Chan et al., J. Acoust. Soc. Am. 99, 2471(A) (1996)] have shown the existence of optimal values of the glottal width and convergence angle for ease of phonation, at which the phonation threshold pressure is minimum, not predicted by the previous theory [I. R. Titze, J. Acoust. Soc. Am. 83, 1536–1552 (1988); J. C. Lucero, J. Acoust. Soc. Am. 98, 779–784 (1995)]. This phenomena is analytically investigated using low-dimensional dynamical models of the vocal folds. The threshold pressure is determined through numerical techniques, linearizing first the equations of motion about the equilibrium position and calculating next the value of lung pressure for instability. The results agree with the experimental evidence, showing that the threshold pressure is minimum when the vocal folds are slightly separated and the glottis is rectangular. The minimum occurs as a consequence of large viscous losses at small glottal widths and airflow separation from a divergent glottis, which were not considered in the previous theory. [Work supported by the CNPq, Brazil.]
Numerical simulation of the flow through in vitro
models of the vocal folds
Authors: Hofmans Geert C. J. , Ranucci Massimo , Bosch R. , Hirschberg A.
Published: October 2005

In order to obtain a reasonable prediction of the flow through the vocal folds, for applications such as a two-mass model for voiced sound production, the use of a quasistationary flow model is considered. This model assumes a quasisteady jet and the flow separation is predicted by means of a boundary layer model. Limitations on the validity of such a model are investigated by means of in vitro experiments which provide global information in the form of flow visualizations and quantitative information in the form of wall pressure and hot wire measurements. A two-dimensional numerical simulation based on a viscous blob method provides further quantitative information on the details of the flow. However, such a simulation cannot predict the transition from laminar to turbulent jet flow which is an essentially three-dimensional effect. Furthermore, both the simplified model and the numerical simulation do not predict the occurrence of asymmetry in the flow which is sometimes observed in the experiments. Imbedding the numerical calculation of the locally incompressible flow in an acoustical model will be discussed. [Work supported by Technology Foundation Grant No. ETN33.2952.]
Formulation of methods for real-time simulation of biomechanical vocal tract
Authors: Wilhelms-Tricarico Reiner F.
Published: October 2005

Based on a method that is related to the mortar finite element method, a domain subdivision method was developed which can be used towards the goal of a real-time simulation of the vocal tract. Previous work [R. Wilhelms-Tricarico, J. Acoust. Soc. Am. 97, 3085–3098 (1995)] on simulation of soft tissue for speech production research has resulted in rather slow algorithms that are not very suitable for implementation in parallel multiprocessor machines. A new formulation uses finite elements with quadratic interpolation of the displacement and velocity fields as well as four pressure variables per element, to approximate tissue incompressibility. This formulation is combined with a domain subdivision method that is designed to implement the simulation algorithm on a multi-processor computer such as CM5. The interaction between the various connected domains is achieved by an augmented Lagrangian method. [Work supported by NIH.]
Finite-element method analysis of a three-dimensional vocal tract model with branches
Authors: Aoyama Kazuki , Matsuzaki Hiroki , Miki Nobuhiro , Ogawa Yoshihiko
Published: October 2005

Sound wave propagation in a vocal tract model is investigated using a three-dimensional finite-element method (FEM). In the present study, a new FEM model is proposed with branches of the vocal tract measured by MRI. The branches correspond to the pyriform fossa and the vallecula of epiglottis. Using this vocal tract model, sound pressure and particle velocity are computed. Evaluations are as follows: first, the difference of vocal tract transfer function (VTTF) between the model with the branches and without; and, second, the difference of VTTF between the model taking into account the wall impedance of the tongue and not taking it into account. From the simulation results, it is shown that formant peak levels are influenced by the branches in the high-frequency region over 4 kHz, and the distribution of the sound-pressure pattern around the branches is seen by a complicated spherical surface. Furthermore, in the case of the model with the wall impedance, it is shown that the formant peak levels have a tendency to decline, due to the loss on the wall.
Applying morphological constraints to estimate three-dimensional vocal-tract shapes from partial profile and acoustic information
Authors: Yehia Hani , Tiede Mark K. , Vatikiotis-Bateson Eric , Itakura Fumitada
Published: October 2005

Dynamic three-dimensional (3-D) vocal-tract (VT) shapes are fundamental to successfully quantifying the relation between articulatory and acoustic characteristics of the vocal tract. Unfortunately, direct measurements of the vocal tract are currently restricted to the acquisition of either static 3-D or dynamic 2-D shapes. Here, an indirect method to estimate dynamic 3-D VT shapes is described and evaluated. The procedure is as follows: first a set of static 3-D VT shapes (derived from MRI) is used to construct a parametric model of the vocal tract. This model constrains the 3-D shape space to be morphologically plausible. Next, 2-D point trajectories (obtained midsagitally with a magnetometer) are used in conjunction with the simultaneously measured first three formant frequencies to estimate the corresponding vocal-tract parameters. Indirect evaluation is performed in two ways: (i) Midsagittal 2-D points are extracted from MIR 3-D VT shapes, and the corresponding formant frequencies are calculated. This information is used to obtain estimated 3-D VT shapes which are then compared with the original MRI shapes. (ii) Area function trajectories are derived from estimated 3-D VT shape trajectories, and used as input to an area-driven speech synthesizer whose output is compared analytically with the original speech.
Acoustic measurement of vocal tract area function: An experimental study
Authors: Mochida Takemi , Honda Masaaki
Published: October 2005

The aim of this study was to develop a method for estimating the cross-sectional area function of the vocal tract by acoustic measurement. In this method, the area function is estimated using the transient response determined from the incident and reflected signals measured at the lips. The measurements are made with the vocal tract at one end of uniform tube that is excited at the other end by a speaker. With this method, it is extremely difficult to estimate the area function behind the narrow constriction, because the input and output signals are measured on the same side of the vocal tract. To improve estimation accuracy, a wide range and high energy level of frequency characteristics is needed for the input signal. The method for obtaining the desired input signal introduced here is based on the LMS error of the speaker output. To verify the estimation accuracy, an experiment was carried out using some cavities of known geometry. This measuring system shows good performance for the cavity whose ratio of minimum to maximum area is about 1:20. The estimated area functions of the human vocal tract were compared with midsagittal shapes simultaneously measured by a magnetic position sensing device.
Tongue articulators as muscle synergies
Authors: Sanguineti Vittorio , Laboissière Rafael , Payan Yohan
Published: October 2005

Tongue movements during speech production were investigated by means of a simple yet realistic biomechanical model, based on a finite-element modeling of soft tissues, in the framework of the equilibrium-point hypothesis (λ model) of motor control. The model has been exploited to estimate the ‘‘central’’ control commands issued to the muscles, for a data set of mid-sagittal digitized tracings of vocal tract shape, recorded by means of low-intensity x-ray cineradiographies during speech. An analysis of commands needed to achieve the empirically observed tongue shapes has shown that, in spite of the great mobility of the tongue and the highly complex arrangement of tongue muscles, central commands can be described in terms of a simple linear control model, with a small number of factors that can be associated with elementary tongue gestures. This result provides a biomechanical substrate to the concept of ‘‘tongue articulator.’’ Moreover, it is suggested that, because of the peculiar geometrical arrangement of tongue muscles, the central nervous system (CNS) may not need a detailed representation of tongue mechanics but, indeed, may make use of a relatively small number of muscle synergies that are invariant over the whole space of tongue configurations.
Measurement of palatolingual contact stresses and patterns during consonant production using a force-sensor-mounted palatal plate
Authors: Matsumura Masafumi , Niikawa Takuya , Tachimura Takashi , Wada Takeshi
Published: October 2005

In this paper the measurement of palatolingual contact stresses and patterns during consonant production using a force-sensor-mounted palatal plate is reported. The force sensor is composed of a strain gauge and a cantilever mounted on a palatal plate which is attached to a hard palate. The palatolingual contact stress is concentrated at one spot on the cantilever by a protuberance bonded on to the cantilever. When the tongue touches the force sensor, the palatolingual contact stress is detected as the strain in the cantilever. A prototype system was developed consisting of a 16-force-sensor-mounted palatal plate (thickness=1 mm) and a multichannel strainmeter. The output of the force sensor was proportional to the stress applied to the cantilever and had no hysteresis. Error due to mechanical interference among the sensors was less than 5.4%. The palatolingual contact stresses and patterns for an adult male were measured during production of the consonants /t/, /d/, and /n/ using the present system.
Control of the oral release in bilabial stop consonant production
Authors: Lofqvist Anders
Published: October 2005

This study examines kinematic patterns of lip and jaw movements in bilabial stop production at the release of the oral closure. Lip and jaw movements were recorded using a magnetometer system. Five subjects participated and produced 50 tokens of the sequences /aCV/, where C was either /p/ or /b/ and V one of /i, a, u/. The organization of jaw movements differed between subjects. In two of the subjects, the jaw did not reach any peak position during the stop but moved continuously during the stop to a peak position during the following vowel; this was the case when the second vowel was one of /i, u/. The open vowel /a/ was reliably associated with a jaw lowering movement of greater amplitude and velocity than the vowels /i, u/. The onset of the lower lip release movement generally occurred before the onset of the jaw lowering movement. The peak velocity of both the lower lip and jaw release movement sometimes occurred before the release of the oral closure. Stop consonant voicing had no consistent influence on the release movements of the lips and the jaw, nor on a derived measure of lip aperture. [Work supported by NIH.]
Towards a three-dimensional articulographic system
Authors: Tillmann Hans-G. , Zierdt Andreas , Hoole Philip
Published: October 2005

Existing EMMA systems have provided invaluable information on speech kinematics. Two main drawbacks remain, however. First, phonetically relevant tongue movements are inherently three dimensional. Second, the necessity for midsagittal alignment severely restricts subjects’ freedom of head movement, potentially compromising the naturalness of their speech. In previous work an arrangement of six transmitter coils allowing three-dimensional reconstruction of sensor coordinates (plus alignment about two axes of rotation) was designed. Now, a two-stage design strategy is reported to arrive at a full-fledged measurement apparatus, involving the development of a virtual instrument followed by its hardware implementation. The geometrical complexity of the transmitter arrangement and of the resulting magnetic fields necessitates the use of complex numeric algorithms to determine sensor location and orientation. A promising algorithm has been developed and will be briefly discussed. The crucial task is then to compare the performance of the algorithm under ideal and expected real-life conditions. For this reason a virtual instrument will simulate the entire measurement chain, to test the sensitivity of the algorithm to errors from quantization, carrier-frequency cross-talk, etc. The results of these simulation experiments will be presented and their implications for realization of the system in hardware will be discussed.
Traditional temporal measures are unstable and thus are inappropriate parameters to compare speech kinematics of control and stuttering populations
Authors: Alfonso Peter J.
Published: October 2005

The aims of the study are to: (1) explore further the short- and long-term flexibility in normal speech kinematics, and (2) identify relatively stable temporal characteristics that could function as appropriate experimental and control group comparative parameters. Movements of the tongue, lips, and jaw were transduced by electromagnetic midsagittal articulography. A single session included 20 perceptually fluent repetitions of /pap/, /tat/, and /sas/ imbedded in a carrier phrase at normal, slow, and fast speech rates. Seven stutterers and matched controls completed three sessions at 2-week intervals. Reported here are across session comparisons of temporal ordering and relative timing for bilabial closure at normal rates. The prevalence and stability of the sequence pattern is dependent upon interarticulator relative time. Four control subjects and two stutterers showed idiosyncratic but stable temporal organizational characteristics within and across sessions. However, three of the controls and five of the stutterers were inconsistent in at least one of the two temporal parameters across sessions. Thus neither temporal ordering nor interarticulator relative time is an appropriate group comparative parameter because of natural long-term flexibility. Natural flexibility is compatible with the notion of coordinative structures within a task-dynamic point of view.
Lip and tongue force control in patients with Parkinson’s disease
Authors: Gentil Michèle , Perrin Sabine , Pollak Pierre , Benabid Alim L.
Published: October 2005

The objective of the present study was to characterize the motor control of idiopathic Parkinson’s disease patients and normal subjects during orofacial nonverbal tasks. The fine force control associated with speechlike force magnitudes in three articulatory organs: Upper lip, lower lip, and tongue were examined by means of force transducers. Using visual feedback, subjects produced ramp-and-hold compression forces as rapidly and accurately as possible to end-point target levels ranging from 0.25 to 2 N. The relationship among several parameters of the ramp-and-hold force contraction and target force level was quantified in 14 Parkinsonian patients (7 males and 7 females) and 14 age- and sex-matched control subjects. All patients were successively evaluated in two conditions, that is, without and with levodopa therapy. The results demonstrated an abnormal isometric control force of the articulatory organs in patients with Parkinson’s disease. Moreover, differences were observed between lip and tongue motor performance. Most of the results were not improved by L-dopa. This suggests that Parkinsonian dysarthria probably results from nondopaminergic lesions.
Voice changes associated with sinus disease: Pre- and post-operative data
Authors: Gracco L. Carol , Vining Eugenia
Published: October 2005

Patients who suffer from sinus disease often report voice changes following sinus surgery. This is typically characterized as a reduction in vocal quality described perceptually as denasal. Pre-operative voices are also perceptually mild to moderately hoarse or breathy indicating some degree of irregularity in vocal fold vibration. This investigation characterized the laryngeal and nasal components of the acoustic signal in patients with sinus disease treated surgically. Acoustic measures from digital tape-recorded speech samples were analyzed pre- and post-operatively. Post-operative voice samples demonstrated acoustic characteristics consistent with increased nasal resonance. This is contrasted with pre-operative samples in which the acoustic correlates of nasalization were diminished. Noise factors associated with a loss of harmonic structure and perturbation in vocal fold vibration often associated with hoarse or breathy vocal quality were found. Significant laryngeal pathology was apparent in all samples using laryngeal videostroboscopic endoscopy. This study demonstrates significant vocal quality changes associated with sinus disease which are improved following surgery. Further, there is a strong correlation between hoarse vocal quality, laryngeal pathology, and sinus disease not often considered in the evaluation or treatment of patients with sinus disease. [Work supported by NIH.]
Patterns of generalization in speech sensorimotor adaptation
Authors: Houde John F. , Jordan Michael I.
Published: October 2005

It has previously been shown that many speakers will exhibit sensorimotor adaptation in whispered vowel production; i.e., they will alter their articulation of a whispered vowel in response to alteration of perceived formant frequencies in their normal acoustic feedback [ J. F. Houde and M. I. Jordan, J. Acoust. Soc. Am. 97, 3243(A) (1995)]. This previous study showed the alteration of the vowel’s articulation always being in a direction which minimized the perceived shift of the vowel’s normal formant frequencies. This previous study also included a pilot investigation of context generalization, which showed this learned adaptation of a vowel in one CVC context affected articulation of the same vowel in other CVC contexts. The current study is a more complete investigation of context generalization, as well as another type of generalization called vowel target generalization, showing the learned adaptation of a vowel in one CVC context affects articulation of different vowels in the same CVC context. The study also found (via post-experiment interview) that no subject was aware of the acoustic feedback having been altered, nor did any report noticing change in their vowel articulations, indicating the automatic, unconscious nature of the adaptation effect.
Examination of the open-end correction coefficient for side branches within an acoustic tube
Authors: Dang J. , Kawanishi Y. , Shadle C. , Honda K. , Suzuki H.
Published: October 2005

The open-end correction of an acoustic tube has been widely investigated in a free-field condition (cf. Hall, 1987). However, no experimental data give accurate open-end corrections for side branches within an acoustic tube, although some proposals have been made [see Fant (1960)]. To incorporate side branches such as the paranasal sinuses and piriform fossa in an acoustic model of the vocal tract, an end correction coefficient is definitely needed. Thus this study estimated the end correction coefficient of side branches experimentally and computationally. A number of acoustic tubes were used to examine effects of (1) the angle from the branch to the main tube, (2) the distance between the open end and the opposite tube wall, and (3) the geometric shape around the open end of the branch. The results indicate that the end correction coefficient depends on the dimension in front of the open end, and is independent of the angle of the branch to the main tube. The open-end correction coefficient (Ce) of a branch can be described by the diameter (D) of its open end and the distance (L) from the end to the opposite tube wall as shown: Ce=0.414+0.517 ×log(L/D2), for 0.158<(L/D2)<6.24. Application of this empirical formula to actual vocal geometries will be discussed.
Acoustic time reversal in shallow water
Authors: Roux Philippe , Fink Mathias
Published: October 2005

The aim of this paper is to study a way to overcome the problem of multipath distorsion in acoustic underwater communications by using a time-reversal mirror (TRM). A set of experiments is presented using a TRM made of 64 reversible transducers and different results will be emphasized. One result is related to the high focusing property obtained with a TRM operating in a bounded channel. It will be shown that the time-reversed beam is much thinner than the one observed in an unbounded fluid. The second result is the time compression observed for the time-reversed acoustic multipath pulse. The influence of the number of elements of the TRM on the focusing and on the time compression is discussed.
Narrow-band source localization in the presence of internal waves
Authors: Book Paul J. , Nolte Loren W.
Published: October 2005

The effects of internal waves on low-frequency (25-Hz) narrow-band source localization performance are investigated at very long ranges (1000 km). Sound-speed perturbations induced by an internal-wave obeying the Garrett–Munk spectrum are incorporated into the normal mode solution to the wave equation using the adiabatic approximation. Sensitivity of source localization on the internal wave modal content is examined. For the entire range of signal-to-noise ratios (SNR) and degrees of randomness in the internal wave field, the optimal uncertain field processor (OUFP) provides both the signal processing algorithm as well as the limitations on source localization performance. For high SNR, the localization errors are local and the performance level of the OUFP is predicted by the hybrid Cramer–Rao lower bound (CRLB). In fairly structured internal-wave environments, significant performance gains are made optimally localizing the source. Under very random internal-wave environments, the environmental variability limits the performance of all signal processing algorithms. [Research supported by ONR (Ocean Acoustics).]
Acoustic intensity fluctuations induced by internal waves in a shallow-water waveguide
Authors: Finette Steven , Turgut Altan , Apel John
Published: October 2005

The SWARM (Shallow Water Acoustics in a Random Medium) experiment performed during July–August 1995 made simultaneous acoustic and oceanographic measurements along a 45-km track on the continental shelf off the New Jersey coast. The area is a site of intense internal wave activity, primarily in the form of soliton packets that perturb the sound-speed profile in range, depth, and time. Incorporating environmental parameters estimated from SWARM measurements, computer simulation results are presented for acoustic propagation through this environment at frequencies of 224 and 400 Hz. A wide-angle parabolic equation method was used to compute multiple realizations of the acoustic field. The simulations employed bathymetry and sub-bottom acoustic parameters estimated from chirp sonar surveys. Oceanographic parameters were obtained from thermistor strings and yo-yo CTDs in conjunction with satellite images, which allowed for estimation of internal wave-field parameters that were used to compute sound-speed distributions. Numerical results include estimates of the full-field and modal-intensity fluctuations, as well as the spatial coherence at two vertical arrays located at 33 and 45 km from the moored sources. [Work supported by NRL and ONR. The authors wish to acknowledge the contribution of the SWARM group to this work.]
Extracting and classifying spatial variations in the angular response of seabed backscatter strength
Authors: Hughes Clarke John E.
Published: October 2005

Swath bathymetric sonar systems use seabed backscattered energy to estimate the slant range from the survey instrument to the seafloor. Because the slant range measurements are made over a range of angles, it is possible to estimate the bottom backscattering strength as a function of grazing angle. This approach is used for backscattered data collected at 95 kHz for grazing angles in the range 15 to 90 deg. The data are acquired from a Simrad EM1000 sonar in water depths ranging from 20–200 m at survey speeds of about 16 kn. These measurements have the advantage that they are made over wide regions. Operational problems that limit the quality of the available data include: degradation due to sea state; uncertainty in radiation and receive patterns of the sonar; and spatial variation in the seabed type within the footprint over which the measurements are made. Methods for minimizing the effect of these problems are presented. In order to use the estimates of local seabed angular response, a parametrization is used to relate the control that variations in seabed physical properties (density, velocity, roughness spectrum, volume inhomogeneites) have on the shape and magnitude of the seabed backscatter angular response.
Detection of narrowband harmonics using higher-order spectral methods
Authors: Barlett Martin L. , Meegan G. Douglas , Wilson Gary R.
Published: October 2005

An empirical investigation of detection performance for multiple harmonic components imbedded in controlled noise was performed using both spectral correlation and stationary bispectrum detection statistics. Conventional power spectral processing is used as a metric for detection gain obtained using the higher-order processing methods. Relative performance between power spectral and higher-order spectral methods will be presented as a function of both signal-to-noise ratio and harmonic ‘‘falloff’’ at fixed probability of false alarm. The empirical results will also be compared with analytic results obtained assuming asymptotic statistics [G. R. Wilson and K. R. Hardwicke, ‘‘Nonstationary Higher Order Spectral Analysis,’’ Appl. Res. Labs. Tech. Rep. ARL-TR-91-8, Applied Research Laboratories, University of Texas, Austin, TX (1991)]. [Work supported under the Independent Research and Development Program, Applied Research Laboratories, The University of Texas at Austin.]
Nonlinear spectrum estimation using a new neural network with a level estimator
Authors: Imai Hideaki , Miyanaga Yoshikazu , Tochinai Koji
Published: October 2005

This paper proposes a new nonlinear signal processing by using a three-layered network which is trained with self-organized clustering and supervised learning. The network consists of three layers, i.e., a self-organized layer, an evaluation layer, and an output layer. Since the evaluation layer is designed as a simple perceptron network and the output layer is designed as the fixed weight linear nodes, the training complexity is the same as the self-organized clustering and a simple perceptron network. In other words, quite high speed training can be realized. Generally speaking, since the data range usually used in signal processing is arbitrarily large, the network output should also cover this range. However, it may be difficult for only one node in the network to output these data. Instead of this technique, if this dynamic range is covered by using several nodes, the complexity of each node is reduced and the associated range is also quite limited. This results in a higher performance of this network than the conventional ones. As one of the objectives, this paper introduces the spectrum envelope estimation of speech waveforms. It is shown that accurate spectrum envelopes can be obtained.
Blind deconvolution based on common zeros of the z
transforms of two observed signals
Authors: Furuya Ken’ichi , Kaneda Yutaka
Published: October 2005

A new method is proposed for recovering an unknown source signal, which is observed through two unknown channels characterized by finite impulse response filters. Unlike conventional blind deconvolution methods, this method can recover not only the spectrum of the signal but also the phase characteristics. This method is based on a cost function that comes from a filter arrangement with a double-layer structure. The cost function is minimized when the common zeros of the z transforms of the two observed signals are extracted. If there are no common zeros between the system transfer functions of the two unknown channels, the common zeros of the observed signals represent the source signal and the noncommon zeros represent the characteristics of the two channels. Therefore, the source signal can be recovered by separating the common zeros from the other zeros, that is, by minimizing the cost function. Adaptive filters are used for this procedure. Computer simulations using room transfer functions as the two unknown channels demonstrate the effectiveness of this method.
Application of NORDEN transform to time-frequency characterization of time-varying signals
Authors: Yen Nai-chyuan , Wu Manli C.
Published: October 2005

The non-orthodox decomposition (NORDEN) transform developed by NASA [N. Huang et al., ‘‘The Empirical Mode Decomposition and the Hilbert Spectrum for Nonlinear and Nonstationary Time Series Analysis’’ (to be published)] is designed to adaptively break down a complex time-varying signal into a sum of several simple mode functions. Each single mode function can then be expressed in terms of an analytic signal whose amplitude and phase vary with time and can be displayed as a trace in a time frequency plot, the VEIN diagram, with magnitude expressed as line thickness or color coded. This methodology of signal analysis, unlike other traditional transform methods, i.e., Fourier, Wigner-Ville, and Wavelet, requires no special window, preselected kernel, or particularly selected mother wavelet and provides more detailed dynamic information about the signal under study in terms of its dominant components and their instantaneous frequency variation in the observation time. Several examples from various types of acoustic signals are examined with this algorithm to demonstrate its signal analysis capability.
A simple constant beamwidth transducer
Authors: Allwright D. J. , Power J.
Published: October 2005

Many designs for constant beamwidth transducers have appeared over the years, but these usually involve complicated array shading techniques or ‘‘lenses’’ applied to ultrasonic devices. With locally reacting materials, such as PVDF for example, it is possible to manufacture a composite material which automatically produces the appropriate frequency-dependent shading of the response in a single, continuous transducer. The material makes use of the capacitive (dielectric) nature of the active element in conjunction with a specially designed, electrically resistive layer, to which a single point connection is made. The resulting CR circuit acts as an integrator, and since the resistance to the remote parts of the transducer is largest, the response of those parts progressively reduces as frequency is increased. In other words, the effective size of the transducer reduces (i.e., ‘‘shrinks’’) as frequency is increased. Correct design of the composite provides a constant directivity characteristic. Several applications of this ‘‘shrinker technology’’ are discussed, including microphones, loudspeakers, and hydrophones.
Characterize reverberation spatial distribution using an L-shape array
Authors: Lee Yung P.
Published: October 2005

Matched-field processing (MFP) is a generalized beamforming. Instead of using plane-wave replica vectors in the plane-wave beamforming, in MFP, an acoustic model is used to calculate replica vectors in a search space. In addition to azimuthal and elevation information, which can be obtained by plane-wave beamforming, MFP also provides range and depth information. In active systems, scattered signals are usually modeled by two-way, source-to-scatterer and scatterer-to-receiver, propagation. Assuming that scatterers are illuminated and reradiate energy as point sources, the scattering field can be characterized by using just one-way, scatterer-to-receiver propagation. Passive MFP can then be used to map the spatial distribution of the scatterers. A shallow-water experiment took place off the Gulf of Mexico in November 1995; reverberation was measured on an L-shaped array from a bottomed cw source. Passive MFP was used to characterize the spatial distribution of the measured reverberation. [Work supported by U.S. Navy.]
A new towed array shape estimation scheme for real-time sonar systems
Authors: Lu Feng , Milios Evangelos , Stergiopoulos Stergios
Published: October 2005

In real-time towed array systems, performance degradation of array gain occurs when beamforming is carried out on the sensor outputs of a line array which is not straight. In this paper, a new method is proposed for array shape estimation. The procedure consists of two steps. First, the tow-point-induced motion is formulated in the time domain based on the constraints from the tow-point compass-sensor readings and from a discretized Paidoussis equation. At each time instance, the shape estimate is solved from a linear system of equations. It is shown that this solution is equivalent to a previous frequency-domain solution, while the new approach is much simpler. In the second step, the tail compass-sensor data are used to adjust the overall array shape. By noting that variations in the ship speed lead to a distortion in the normalized time axis, the predicted tail displacements are first registered with the tail sensor readings along the time axis. Then distortions in the estimated array shape over its length can be compensated accordingly. A slow-changing bias between sensor zeros is also modeled in order to remove systematic sensor errors. The effectiveness of the new algorithm is demonstrated with real sea-trial data.