bluuebunny/crossref_metadata_2025 huggingface dataset:
Tests for categorizing the auditory speech discrimination in the severely/profoundly deaf young adult
Authors: Webster J. C.
Published: October 2005

The phonetic constant, word familiarity and grade level comprehension of the 1922 monosyllabic words used in ten free responses and eight rhyme tests were analyzed. Printed versions of the predictable SPIN sentences were given for 35 NTID students to fill the missing last (key) word. Based on analysis of these words three new tests were conceived. (1) A six-choice vowel initial and final consonant and numerial test was found to categorize 35 NTID students more efficiently than the present spondaic word/CHABA (CID) sentences profiling tests. (2) The SPINIKER test, a competing message test using the concepts (and many sentences) of SPIN to be used for speech (lip) reading and/or auditory discrimination testing. (3) A common-word more or less unlearnable question or sentence test using multiple choices among interrogatives (where, when,…), verbal auxiliaries (can, will,…), personal pronouns and free responses among verbs or nouns to be used in hearing aid evaluations. Typical sentences are “What could she drink?” or “Where is my bag?”
Use of optical information in phonetic perception
Authors: Summerfield Quentin
Published: October 2005

The accuracy with which normal listeners can transcribe sentences presented 12 dB below a background of continuous prose was compared with accuracy in four audio-visually supplemented conditions. With monochrome displays of the talker showing (i) the face, (ii) the lips, and (iii) four points at the centres of the lips and the corners of the mouth, accuracy improved by 43%, 31%, and 8%, respectively. In contrast, no improvement was obtained with an optical specification of syllabification by a Lissajous circle whose diameter varied according to the amplitude envelope of the test sentences. The differences suggest that optical concomitants of articulation specify linguistic information and do not merely focus attention when there is a competing background. This conclusion was reinforced in a second experiment, in which identification functions were obtained for continua of synthetic utterances. These ranged between /aba/, /ada/, and /aga/, and were presented both in isolation and in combination with videorecordings of a speaker uttering each of the disyllables corresponding to the endpoints. Percepts in audiovisual conditions differed from those in audio-alone conditions: audio-visually, /b/ was only perceived when lip-closure was specified optically, and, if lip closure was specified optically, /b/ was generally perceived. Interpretation of the detailed results suggests that perceivers make use of articulatory constraints on the combined audiovisual specification of phonetic events.
Identification and discrimination of VOT by listeners with moderate, severe and profound sensorineurai hearing loss
Authors: Parady Susan , Dorman M. F. , Whaley P. , Raphael L. J.
Published: October 2005

To determine whether sensorineural hearing loss affects listeners' ability to detect the voice onset time (VOT) of stop consonants, listeners with moderate, severe, and profound hearing loss were presented stimuli from along a VOT continuum in both identification and discrimination tasks. The location of the phoneme boundary did not differ among the normal listeners, the listeners with moderate loss, and most listeners with severe loss. Some listeners with severe loss, however, evidenced longer boundaries while others could not identify the signals at all. A similar outcome was obtained for the listeners with profound loss. For the most part discrimination data mirrored identification data. However, in several instances listeners who were unable to identify voiced and voiceless stops were able to discriminate between them in a normal manner. Since identification and discrimination tasks assess different aspects of information processing, an evaluation of sensory capabilities in the hearing impaired should include both tasks.
Identification and discrimination of synthesized syllables by normally hearing six- and ten-year-old children and by adults
Authors: Elliott Lois , Longinotti Cheryl , Meyer Dianne , Zucker Karen , Raz Israel
Published: October 2005

Adaptive test procedures were used to estimate: (1) boundaries between perceptual “categories” of a labeling, or identification, task and (2) the just-discriminable difference to each side of the midpoint of a 13-item /ba/-/da/-/ga/ continuum of 4-formant, synthesized syllables (AX task). At the locus of the /ba/-/da/ boundary: (1) both 10-year olds and adults had significantly better discrimination than 6-year olds; (2) 10-year-olds' categorical boundaries were significantly different from those of 6-year olds; and (3) only adults showed a significant difference between the discrimination limen and the boundary between “categories.” At the locus of the /da/-/ga/ boundary, performance of 6-year-olds differed significantly from adults'. These results were obtained even though all Ss scored approximately 100% correct on a speech discrimination test—WIPI for children and W-22 for adults—and parallel results obtained on a monosyllabic word perception task [Elliott, L. L. et al., J. Acoust. Soc. Am. 66, 12–21 (1979)]. [Supported in part by BEH, NINCDS, and NSF.]
Perception thresholds during transient displacements of basilar membrane
Authors: Escudier P. , Schwartz J. L. , Dolmazon J. M. , Carré R.
Published: October 2005

Single fiber responses, produced by trapezoidal time pattern of basilar membrane displacements [Sokolich et al., J. Acoust. Soc. Am. 59, 963–974 (1976)] reveal, on each fiber, an increased firing rate during displacements toward scala tympani or toward scala vestibuli, according to the characteristic frequency of the fiber. Such signals have been tested, using our peripheral auditory system modeling [J. M. Dolmazon and L. Bastet, J. Acoust. Soc. Am. Suppl. 1 65, S14 (1979)]. Simulation results show good agreement with experimental data. In the experiments described here, such test signals have been studied by psychoacoustic tests. The perception thresholds of medium or high frequency tone bursts superimposed on a low frequency (40 Hz) trapezoidal signal have been specified for different position of this modulation inside the fundamental period. In order to excite fibers with different characteristic frequencies, experiments have been done with several modulation frequencies. Results are described and discussed in light of single fiber responses and simulation results from our model.
The frequency response of vibrators with coupled modes, such as cylindrical and spherical shell
Authors: Skudrzyk E. J.
Published: October 2005

In the standard theory, the mode masses are multiplied by a mode shape factor to account for the energy of coupled-in-modes. This procedure represents an excellent approximation in the frequency range near the resonance peaks, but it is not exact. If, for instance, a spherical shell is driven by a point force, the stretching modes are not excited directly since the point of attack is a node for these modes; nevertheless, they all become excited because of their coupling with the transverse vibration of the shell. Because of the distributed nature of their excitation, they can be excited with a negative sign. For instance, in the case of a spherical shell at low frequencies, the stretching modes counteract the transverse motion. It is proved in network theory, that coupled systems can be represented by parallel connections of simple tuned circuits with frequency-independent elements, and that, because of the coupling, some of their signs can be negative. Similar results apply to shells. As a consequence, the driving point admittance does exhibit not only sharp antiresonance, but exhibits also some shallow minima. The deviations between the exact and the standard mode solution will be discussed and the frequency response of coupled vibrators will be illustrated by the mean-value method for cylindrical and spherical shells. [This work was sponsored by the Office of Naval Research (Code 474).]
Effect of a fluid-viscoelastic composite baffle on the signal pressure
Authors: Ko Sung-Hwan
Published: October 2005

A theoretical model was developed to evaluate the effect of a fluid-viscoelastic composite baffle on the signal received by an array of hydrophones. The composite baffle placed between the vibrating plate and the semi-infinite fluid medium (water) is designed primarily for reducing the pressure fluctuations in the water originated from the vibratory motion of the plate. Due to their acoustic softness, noise reduction baffles for underwater use degrade the signal being received by producing a partially out-of-phase reflected wave. Therefore, it is desirable to improve the signal-to-noise ratio for a given type of the baffle. The viscoelastic portion of the composite baffle (which may be called the signal conditioning plate) is a rubberlike material that has a loss factor associated with the shear modulus. The present model is compared to the case where the viscoelastic signal conditioning plate is replaced by an elastic thin plate. Effects of various parameters on the signal pressure are discussed. [Work supported by Naval Material Command and Naval Sea Systems Command.]
Performance of transform and subband coding systems combined with harmonic scaling of speech
Authors: Malah David , Crochiere Ronald E. , Cox Richard V.
Published: October 2005

In this presentation we report on a nearly 2:1 reduction in the bit rate of two high quality waveform encoding schemes—adaptive transform coding (ATC) and subband coding, without impairing the quality of the reconstructed speech. The bit rate reduction is achieved by operating the waveform coders on a frequency scaled speech signal, occupying half of its original band. The frequency scaling is performed by a recently developed efficient time-domain harmonic scaling (TDHS) algorithm. This algorithm is pitch adaptive and is based on weighting adjacent pitch periods by an appropriately chosen window function. Once the pitch period is known, the TDHS algorithm can be implemented with only one multiplication and two additions per output sample. The aim of this study was to achieve high-quality, natural sounding speech at or below 9.6 kbit/s. By combining TDHS with ATC, high-quality encoded speech is demonstrated at 4.8 kbit/s. For high-quality speech at 9.6 kbit/s the combination of TDHS and subband coding offers a system with a lower complexity than ATC and which is potentially amenable for hardware implementation with current technology. Examples of computer processed speech by the above schemes will be played.
A real-time “homomorphic-vocoder driven” transform coder for speech
Authors: Cox R. V. , Crochiere R. E.
Published: October 2005

Adaptive transform coding (ATC) has recently been proposed as a technique for good quality, robust coding of speech at bit rates from 9.6 to 16 kb/s. In this paper we report on two new developments: (1) the use of a homomorphic vocoder model for the “side-information” channel in ATC and (2) the application of these techniques to a real-time implementation of ATC on an array processing computer. In the ATC coder speech is buffered into blocks and transformed into a frequency domain representation. The transform coefficients are then quantized for transmission. The discrete cosine transform (DCT) is generally used because it minimizes “end effects” in the blocks. The step sizes and number of bits in the quantizers are adaptively controlled to track the dynamic spectral variations of speech. The control of the step sizes and bit assignment is based on a local estimate of the spectrum which is parameterized, encoded, and transmitted as “side-information” to the receiver. In previous work an LPC vocoder model was used to efficiently model the spectrum in order to take advantage of known properties of speech. In this paper we investigate the use of a homomorphic vocoder model which accomplishes essentially the same thing. The properties of the “pseudo-cepstrum” in the homomorphic model (which is based on a DCT analysis) are explored in order to define the side information needed to characterize the DCT spectrum. It is also shown that the choice of the homomorphic model leads to a particularly convenient form of the ATC algorithm for a real-time block processing implementation on an array processing computer. Also it is shown that the log spectrum output of the homomorphic model is in a convenient form for both the bit assignment algorithm (which now becomes a straightforward quantization operation) and the quantization of the transform coefficients (which is done in the log domain). Details of the algorithm will be discussed and examples of real-time coded speech will be played. Good quality is achieved at 9.6 kb/s and nearly transparent quality is achieved at 16 kb/s.
An investigation of endpoint detection techniques for isolated word recognition
Authors: Lamel Lori F. , Rosenberg Aaron E. , Rabiner Lawrence R.
Published: October 2005

Accurate location of the endpoints of an isolated word is important for reliable and robust word recognition. The endpoint detection problem for isolated words where the background noise is stationary and low level is relatively easy to handle. However, often the beginning and end of an isolated word is obscured by speaker generated artifacts (such as mouth noises, e.g., clicks, pops, lip smackings, and breathiness) as well as those introduced by the recording environment and transmission system. Several techniques for endpoint detection of isolated words recorded over a dialed-up telephone line were studied. The techniques investigated are classified as either explicit, implicit, or hybrid. The class of explicit techniques are those in which the endpoints are located prior to and independent of the recognition and decision stages of the system. For implicit endpoint estimation, the endpoints of the isolated word are determined by the recognition and decision stages of the system; i.e., there is no separate stage for endpoint detection. The hybrid techniques incorporate aspects from both the explicit and implicit methods. Investigations showed that the hybrid techniques provided the best estimate of both endpoints and correspondingly the highest recognition accuracy of all three classes that were studied. A new hybrid technique for endpoint detection is proposed which reduces the recording rejection rate by 30% while maintaining the same recognition accuracy as obtained using earlier techniques with clean speech (i.e., no artifacts).
Segmentation of speech signals by application of methods of hierarchical classification
Authors: Breitkopf P.
Published: October 2005

A multi-level segmentation of a one-dimensional signal may be induced by hierarchical ordering of subsets in a corresponding parameter space. This concept has been introduced to design a segmentation algorithm that creates a special three-level segmentation for the speech signal. For preclassification it uses the parameters short time prediction gain and short time variance to form the second level of a segment hierarchy containing the classes “pause,” “fricative,” “vocal,” and “nasal oriented.” By merging the segment classes “pause” and “fricative” as well as “vocal” and “nasal oriented” the first level is formed. Since the vocal parts comprise more than 50% of speech, a clustering procedure has been added to create a third level containing four classes that roughly correspond to four different vowel classes. The parameter vector for the clustering algorithm is the sampled LPC-generated log power spectrum together with the L2 distance. Five samples of speech, with a duration of 1 min each, have been processed. The resulting segmentation served as a basis for a number of segment length statistics which suggest applications in speaker verification and speech coding. [Work supported by VW Foundation, at the University of Hannover, Germany.]
Maximum likelihood parsing of speech in the presence of segmentation errors
Authors: Levinson Stephen E.
Published: October 2005

In this paper, a new method of syntactic analysis for speech recognition is presented. It is based on a novel algorithm which uses a formal model of grammatical structure to correct errors resulting from ambiguities not resolved by acoustic analysis of the continuous speech signal. The algorithm is unique in that it can correct not only misclassifications of symbols, substitution errors, but also those errors due to the presence of extra symbols and the omission of others, segmentation errors. It is proven that the algorithm is optimal in the sense of maximum likelihood and it is further shown that the algorithm is efficient, having a worst case execution time proportional to the length of the input and the square of an appropriate measure of the grammar size. An upper bound on the memory requirements is proven to the linear in both quantities. The algorithm has been tested on 496 Japanese phrases averaging 20 phonemes in length and spoken fluently by four male speakers. Based on input from an acoustic processor with an overall phoneme error rate of 40%, roughly equally divided between misclassifications and segmentation errors, a correct phrase recognition rate of 62% was obtained.
Combination of linear and nonlinear time normalization for isolated word recognition
Authors: Welch J. R.
Published: October 2005

Some time normalization is necessary for reliable word recognition. Recently, attention has been given to nonlinear time warping using dynamic programming. Implementation of full dynamic programming time warping in real time, however, requires substantial computational power. A real-time speech recognizer is described which employes linear time normalization followed by a nonlinear time warp between the linearly normalized input and reference samples. A symmetrical time warp function with maximum magnitude deviation of 12.5% from the main diagonal is used. This combination of linear and nonlinear time warping was compared to linear time normalization with three shift values. In a test with one experienced user speaking natural alphabet data, recognition performance was improved 2.1% to a level of 96.3% correct. In a second test with an experienced user speaking one- and two-digit pairs, the improvement was 0.7% to a level of 99.4%. In tests with 9 mostly inexperienced users speaking digits and phonetic alphabet, the improvement was 0.6% to a level of 98.0%.
Evaluation of an automatic directory listing retrieval system based on isolated word recognition
Authors: Aldefeld Bernd , Rabiner Lawrence R. , Rosenberg Aaron E. , Wilpon Jay G.
Published: October 2005

Automated directory listing retrieval has been a goal of the Bell System for a long time. An early form of such a system, based on the use of spoken spelled names, was a speaker trained system and it used a simplified directory search algorithm. Subsequent improvements and modifications to both the recognition algorithm and the directory search procedure have led to the current implementation in which the overall system is speaker independent and can automatically find the name in the directory which provides the best acoustic match to the spoken name. The new system can automatically detect and correct simple (i.e., single letter) anomalies in the spelling of the name, including letter substitutions, inversions, deletions, and insertions. If a conflict in the detected name occurs (e.g., two or more names with the same or close acoustic distance scores), the system automatically requests additional information to help resolve the ambiguity. In evaluational tests on an 18 000 name Bell Laboratories directory, the directory listing retrieval system found the unique correct name in 98.3% of the trials, on average, even though the acoustic recognizer provided the correct letters only about 70% of the time.
The role of professional societies in noise-control education
Authors: Johnson John C.
Published: October 2005

Although societies of technical professions are not chartered to provide studies for certified credit (in the sense of educational institutions), each society has addressed the matter of education in its respective field. Such concerns are manifested in various forms: society-sponsored workshops, seminars, special sessions, topical conferences, and even selected certification of individual technical qualifications. Additional society contributions include directories of accredited course offerings by educational institutions within selected fields of special concern and information on job opportunities and advanced degree graduates produced by educational institutions within selected subfields of specialization. Thus, even though professional societies do not offer accredited courses of study, they tend to be an active partner in education-related functions. During the past decade, noise-related studies have received increased attention by both educational and professional institutions, resulting in a marked upturn of events relating to educational activities by professional societies. Specific education-related activities by professional societies have been selected for abbreviated description of the important role they play in the total educational process in the field of noise control.
Educational needs from the point of view of consultants
Authors: Bruce Robert
Published: October 2005

This paper summarizes the results of a survey of consultants practicing noise control engineering. In general, consultants in noise control engineering recommend that students obtain a broad educational background with undergraduate and graduate degrees in Engineering, obtain practical work experience in a variety of engineering environments, and develop expert skills in written and spoken communication. Management skills such as planning projects and controlling costs are also important.
Graduate level education in noise control—Needs and availability
Authors: Cohen Raymond , Crocker Malcolm J.
Published: October 2005

This paper will explain the reasons for graduate level education in noise control, typical programs of study at the M.S. and Ph.D. levels and survey several of the graduate programs available.
How we match industrial needs to the educational process
Authors: Lang William W.
Published: October 2005

Earlier papers in this special session have considered the needs of the user of the end product of the process. This paper describes how users can influence the educational process by means of two written examinations administered by the Institute of Noise Control Engineering (INCE/USA). The INCE Fundamentals Examination is a two-hour, closed-book, multiple-choice examination covering the fundamentals of acoustics and noise control engineering. The INCE Professional Examination is divided into two parts, each of four hours duration. The first part, Principles of Noise Control Engineering, covers fundamental concepts including the application of basic acoustics, mechanical dynamics and properties of the ear to noise control problems. The second part, Practice of Noise Control Engineering, requires a selection of problems from a number of individual specialities, including instrumentation and measurements, hearing conservation, noise problems in buildings, in the community, and in several specific industries. A concerted effort has been made to develop questions that are typical of those encountered by the practicing engineer. Hence, the exam questions are of a practical, rather than academic nature. By adjusting the educational process to meet the requirements of the INCE examination program, industries' needs are matched.
A technique for the study of drop dynamics in liquid-liquid systems
Authors: Trinh E. , Wang T. , Lee M.
Published: October 2005

A device based on the effects of acoustic radiation pressure, conceptually similar to that used by P. Marston and R. Apfel [J. Coll. Interface Sci. 68, 280 (1979)], has been developed for the investigation of driven large amplitude shape oscillations of liquid drops immersed in immiscible hosts. The system primarily consists of a liquid resonant cavity in which acoustic standing waves are used to trap drops and to drive them into resonant shape oscillations through amplitude modulation. The sizes of the drops currently used range from about 0.5 to 1.5 cm in diameter, and the present liquid combination is silicone oil in a water-methanol mixture. Measurements of the frequencies of resonant modes as functions of oscillation amplitude and viscosity, and of damping constant for free oscillations can be performed. A triaxial system allows a nonaxisymmetric drive through selective excitation in various directions. The analysis of complex oscillations of compound drops (or liquid shells) is also possible. [This work is supported by NASA.]
Computer assisted optical measurements of acoustic fields
Authors: Clark J. A. , Kia T.
Published: October 2005

Optical fringe patterns characterizing acoustic fields can be photographed by a variety of interferometric, schlieren, and shadow-graphic methods. It is possible to simultaneously determine several independent measures of the acoustic field (such as acoustic pressure and particle velocity) by spatially or temporally differentiating or integrating the optical data since the data are available for an extended region of space and period of time. However, digital processing techniques are required to achieve practical results with the large quantity of data usually involved. In this talk procedures for digitizing, enhancing and compacting data from optical fringe patterns with a solid-state video camera and microprocessor will be demonstrated. Digital image processing approaches which have been implemented on two large computers will be described. Fringe patterns generated from the processed data which correspond to several new measures of acoustic fields will be presented. These techniques will be used to analyze the scattering of sound and shock waves at boundaries. [Research supported by ONR and assisted by Dr. H. Dardy, NRL.]
Sparse planar microphone array for estimating bearing to near-ground sources: System considerations
Authors: Miller G. Kirby , Babian Fred E.
Published: October 2005

The maximum likelihood bearing estimator has been simulated on a computer and has been used to examine the effects on performance of both environmentally imposed conditions and system parameters. The imposed conditions include source spectra, ambient noise spatial character, wind, temperature, and array tilt due to terrain. The system parameters include microphone position error, channel amplitude match, channel phase match, operating frequency band, number of microphones, size of array, number of azimuth bins, and number of delay line taps. The results of this study indicate that a passive acoustic bearing estimation system for ground sources should be able to achieve an accuracy of a few degrees under most conditions.
Sound radiation from convex and concave domes in an infinite baffle
Authors: Suzuki Hideo , Tichy Jiri
Published: October 2005

The sound radiation from convex and concave rigid diaphragms represented by portions of a spherical surface was investigated for the purpose of modeling a dome loudspeaker. Pressure responses, radiation impedances, directivity patterns, and some other characteristics were obtained and compared with those of a circular flat diaphragm of the same radius. While the on-axis pressure response of the rigid flat piston remains constant, the response of the convex dome decreases for about ka ≥ 1 as the height of the dome increases. The concave dome has a wide smooth peak around ka = 1.5 and many peaks and dips for larger values of ka. The radiation resistance of the convex dome increases by the rate of 12 dB/oct or less up to about ka = 2.0 and remains almost constant in higher frequency regions. On the other hand, the radiation resistance of the concave dome has its largest peak at the frequency corresponding to the peak of sound pressure response and shows many peaks and dips in higher frequency region. The radiation reactance of the convex dome is smooth and always positive, while the concave dome also exhibits negative reactance due to the resonance in its cavity. The directivity patterns of either the convex or concave dome do not have minima with zero sound pressure and their radiation patterns are generally wider than those of the circular plate.
Response of a fluid-loaded panel supported by several ribs
Authors: Crighton D. G.
Published: October 2005

An infinite fluid-loaded surface, idealized here as a membrane, is supported by two ribs of infinite impedance, situated at equal distances on either side of a third central rib which is driven by a prescribed line force. Formal exact solutions are readily obtained for the structural response and the radiated sound field, and these are simplified and interpreted on the basis that a fluid-loading parameter at the coincidence condition is small. The arguments rely heavily on the asymptotic structure of the Green's function for the infinite fluid-loaded membrane, as described at the 98th Meeting [D. G. Crighton, J. Acoust. Soc. Am. Suppl. 1, 66, S1 (1979)]. Parameter ranges in which resonances and antiresonances are to be found are determined, and explicit expressions are given for the amplitude of resonant and antiresonant responses in regard to the sound field, the drive-point response, and the surface wave fields transmitted across the ribs. Arbitrary configurations of multiple ribs can be dealt with in a similar manner.
Homophonic adaptation compared to critical bandwidth
Authors: Weiler Ernest , Jhaveri Jyoti , Ogden Christine , Agnello Joseph
Published: October 2005

Glass and Weiler (1979) found a steady decrease in loudness adaptation as bandwidth of the adapting noise increased, when it was assessed with a 1 kHz comparison tone. This differed from the findings of Carterette using a homophonic assessment of adaptation at increasing bandwidths. The present study used a homophonic technique to compare adaptation within the critical bandwidth for 1 kHz (100 Hz bw), and outside the critical bandwidth (300 Hz bw). The values for ten listeners agreed closely with the findings of Glass and Weiler for these same two stimulus values. The subjective increase in unadapted loudness for the 300 Hz bw, was not extended to the adaptation, which decreased. The difference again is not in the same direction reported for the larger noise bandwidths of Carterette.
Dynamic testing of elastomeric materials
Authors: Hobaica E. C.
Published: October 2005

Elastomeric materials display a very complex time-dependent response to stress. When attempting to characterize the behavior of such viscoelastic materials, the number of meaningful test methods are found to be limited and test results must be carefully applied over limited conditions. This presentation reviews the methods of obtaining dynamic properties of elastomers over the full range of frequencies that may be encountered. It starts with a brief discussion of viscoelastic behavior, covers the temperature-frequency equivalence postulated by Williams-Landel-Ferry, before reviewing test methods. It groups the methods by frequency. Methods discussed range from the torsional pendulum to wave propagation methods and include the Rheovibron, the Fitzgerald apparatus, and the transmitted force method among others. Particular attention is given the flexure wave apparatus used at Electric Boat.
Determination of dynamic circumferential properties of a composite viscoelastie hose
Authors: Hauptmann Robert
Published: October 2005

The hose wall of many towed line arrays is made up of a composite of several materials which are chosen to obtain particular longitudinal and circumferential properties. These properties are normally predicted by measurements made on the individual materials which made up the composite hose. A new method has been developed to measure the circumferential hose properties directly. The method utilized impulse measurement techniques and piezoelectric polymer film to measure a pulse as it propagates down an oil filled section of array hose. The impulse technique allows measurements over a broad frequency range to be made before reflections from the end terminations complicate the picture. The 1-mil thick piezoelectric polymer film glued circumferentially around the hose allows measurements of the pulse propagation in the compliant hose wall without affecting its characteristics. A dual channel FFT analyzer is used to capture the pulses, compute a transfer function and hose wall properties. The method has been used to determine the circumferential properties of several hose walls over a 1- to 700-Hz frequency range.
Measurement of viscoelastic properties by strain response of a piezoelectric plate
Authors: Schapery R. A. , Harbert B. C.
Published: October 2005

A piezoelectric plate device called the “duomorph” has been developed for use in measuring dynamic and transient viscoelastic properties of both soft and hard materials. This device consists of two crystalline piezoelectric plates which are bonded together to form a driver-sensor pair or a pair which produces pure bending and on which are mounted electrical resistance strain gauges. Electrical excitation of the duomorph causes bending and/or in-plane straining which is affected by any material with which it makes contact. Through analysis and comparison of the response to that measured when the gauge is tested by itself. the dynamic and transient viscoelastic properties of the contacting material are found. Both phase angle and amplitude are used in determining properties. The duomorph was originally developed by Aerojet and Texas A&M several years ago for the purpose of measuring in situ complex modulus during aging of solid rocket fuel. Since then we have employed several different versions of the duomorph when fully embedded or when on the surface of various materials, including clay, asphalt, and rubber. Its most recent application consists of measuring viscoelastic properties of resin during and after curing. It has potential application to in-process monitoring of polymeric laminates, and this application is presently under study. In this lecture we describe the device, its fabrication, calibration, and various applications. [Sponsored by the Air Force Office of Scientific Research and the Office of Naval Research.]
Vowel formant frequency characteristics of preadolescent males and females
Authors: Bennett Suzanne
Published: October 2005

This report describes the vowel formant frequency characteristics (F1-F4 of five vowels produced in a fixed phonetic context) of 42 seven and eight year old boys and girls and the relationship of vocal tract resonances to several indices of body size. Results showed that the vowel resonances of male children were consistently lower than those of females, and that the extent of the sexual differences varied as a function of formant number and vowel category. Averaged across all measured formants of all five vowels, the overall sexual distinction was approximately 10%. The range of differences extended from 3% for F1 of /i/ to 26% for F1 of /æ/. Measures of body size were always significantly related to these children's formant frequencies (range in multiple R's −0.506 to −0.866). The origin of the sexual dimorphism in vocal tract resonance characteristics is discussed with reference to male/female differences in vocal tract size as well as certain articulatory behaviors.
Perception of rising and falling F2 transitions in speech stimuli
Authors: Gordon-Salant Sandra , Wightman Frederic L.
Published: October 2005

The effects of monotic CV competition on perception of place were determined in identification tasks using synthetic speech stimuli. Three continua of stimuli were used: ba-da-ga, bi-di-gi, and bε-dε-gε, with stimuli in a series varying only in F2 starting frequency. Standard three-choice identification functions were obtained for each continuum in quiet, and in the presence of a monotic “masker” competing at a signal-to-competition ratio of 0 dB. The competition consisted of a clearly identified bV, dV, or gV exemplar from the test continuum. Masked identification functions demonstrated that maskers with rising F2 transitions had no effect, while muskets with falling F2 transitions markedly affected stimulus identification. The current results are similar to those obtained previously [C. Berlin et al., J. Acoust. Soc. Am. 53, 699–709 (1973)] with dichotically-presented, natural CV stimuli. The present data are consistent with the hypothesis that complex, time-varying stimuli with falling F2 transitions are more “salient” than those with rising F2 transitions.
Perceptual competition of CV syllables and of isolated F2
Authors: Syrdal-Lasky Ann K.
Published: October 2005

Using three experimental paradigms—single-response identification of dichotic and of electronically mixed monaural stimuli with simultaneous onsets and recognition of 50 ms leading stimulus of dichotic pair—perceptual competition was studied between pairings within three sets of stimuli—two-formant synthetic /bae/, /due/, /gae/syllables, identical /pae/, /tae/, /kae/ syllables with 50 ms F1 cutback, and identical isolated second formants. Tests were performed at two intensity levels such that the extended vowel measured 70 dB or 90 dB SPL. Stimuli containing frequency modulated F2 transitions generally dominated those with F2 of unchanging frequency. Stimuli having lower initial F2 frequency generally dominated those with higher initial F2 frequency, although this effect declined for 90 dB /bae/ pairings, in contrast to analogous pairings in other stimulus sets, presumably because of greater simultaneous masking of F1 on F2 at higher intensity. Psychoacoustic fusion [J. E. Cutting, Psychol. Rev. 83, 114–140 (1976); B. H. Repp, J. Acoust. Soc. Am. Suppl. 1 58, S76 (1975)] occurred equally often regardless of stimulus set and regardless of dichotic or mortaural presentation. [Work supported by NSF.]
Cross-series adaptation with vocalic stimuli
Authors: Godfrey John J.
Published: October 2005

The identification boundary shift after adaptation is now generally attributed to auditory contrast, rather than the operation of feature detectors. That this contrast must be in a prephonetic form was emphasized by the demonstration that mutual adaptation may take place even between vocalic and consonantal segments [J. J. Godfrey, J. Acoust. Soc. Am. Suppl. 1 66, S114(A) (1979)]. Further investigation shows that cross-series adaptation may be obtained with brief vocalic stimuli differing in the amount of formant frequency transitions from steady-state [ε] to [ει]. Cross-series adaptation was attempted with three adaptors: [ɔ], [ɔω̄], and [ɑι]. In the first two, F1 was identical to one of the test series stimuli, but F2 was different; in [ɑι], both F1 and F2 were different, but the directional pattern of formant changes resembled [ει]. Only the [ɑι] adaptor caused a boundary shift, suggesting a fairly abstract level of auditory contrast. [Work supported by NIH.]
A ray path analysis of sediment shear-wave effects on bottom reflection loss
Authors: Vidmar Paul J.
Published: October 2005

The effect of sediment shear-wave propagation on bottom reflection loss (RL) from a typical deep sea sediment layer is analyzed by means of a ray path decomposition of the acoustic field. The purpose of this analysis is the development of a simple physical model of the sediment shear-wave effects reported in a recent computational study of RL from a typical deep sea turbidite layer [P. J. Vidmar, J. Acoust. Soc. Am. Suppl. 1 66, S75 (1979)]. The ray path decomposition uses the interface reflection and transmission coefficients to obtain the amplitude of each shear (S) and compressional (P) wave. These coefficients are analyzed by means of an expansion in the parameter ε which is proportional to the relatively small S wave velocity of marine sediments. Ray paths with amplitudes up to order ε2 are identified and used to develop a qualitative ray path model of RL. P-wave conversion at the sediment-substrate interface is the dominant mechanism for sediment S-wave excitation. To order ε2 RL is the result of the interference of three waves in the water whose amplitudes are proportional to ε0, ε1, and ε2. Comparison with RL obtained from a computational model shows that the ray path model correctly predicts the dependence of RL on frequency, S-wave speed, S-wave attenuation, P-wave attenuation, and sediment thickness. [Work supported by Naval Ocean Research and Development Activity and Naval Electronic Systems Command.]
Frequency response of interferometric fiber-optic coil hydrophones
Authors: Jarzyuski J. , Hughes R. , Hickman T. R. , Bucaro J. A.
Published: October 2005

The frequency response of optical fiber hydrophones is examined both theoretically and experimentally. Previous models have considered only either static pressures or high-frequency (axially constrained) perturbations. In the present study, the acoustic sensitivities for fiber coils (5-cm-diam, 10-m total length) are given for the range 100 Hz to 50 kHz. The observed frequency and coil orientation dependence of the pressure sensitivity is explained in terms of an approximate theoretical model. According to this model the acoustic driving force on the coil is the sum of two terms, a surface pressure and a pressure gradient term, of opposite sign. It is assumed that only the lowest-order radial mode of vibration leads to a net optical phase shift in the coil. The observed peaks in the pressure sensitivity of the coils are found to be in the vicinity of the resonance frequency for the lowest-orderadial mode.
Measurement of complex sound speed in sonar transducer materials using impact testing techniques
Authors: Ricketts Donald , Erickson David J.
Published: October 2005

A mathematical model has been developed for evaluating the complex-valued sound speed in transducer materials. The associated measurement scheme employs a symmetrical longitudinal vibrator, wherein the preloaded test specimen constitutes the central piece of the test resonator. By impacting the test resonator, its resonance frequency and mechanical Q can be determined using a spectrum analyzer and vibration pickup. The mathematical model of the composite longitudinal vibrator-type test resonator facilitates the evaluation of the sound speed in the test sample from the measured resonance frequency and Qm. Experimental results are presented for several materials, with the measured sound speeds given as a function of prestress and temperature.
Technique for steering the virtual array of a parametric acoustic source
Authors: Woodsum Harvey C.
Published: October 2005

A particularly useful waveform for modulation of the carrier (primary) beam of a parametric acoustic source is one in which the “self demodulation” which occurs for θ = 0 is vastly different from the θ ≠ 0 case. The particular modulation considered here has the property of transmitting a cw signal with a directivity pattern which peaks off-axis while the on-axis transmission consists of impulsive noise. For active sonar applications, this technique allows one to form a conical rather than endfire beam by filtering on-axis energy from the echo returns. The MRA of the cw beam can be steered by varying the modulation or primary source level, thus allowing the possibility of higher volume scanning rates without sacrifice of angular resolution.
The reliability of social survey data on noise effects
Authors: Hall Fred L. , Taylor S. Martin
Published: October 2005

The results of household interviews provide essential data for the formulation of noise control policies. However, only a few studies have investigated the reliability of such data, in terms of a test-retest comparison. To help provide better evidence of reliability, 212 persons in the vicinity of Toronto International Airport were re-interviewed in the summer of 1979, using essentially the same questionnaire they had answered in the summer of 1978. To test whether the reliability is affected by noise level, the sample was stratified by both aircraft and road traffic noise levels. Results show that, on an individual level, reliability of attitude scores is not particularly high (correlation coefficients of 0.50 ± 0.05 for several scales for both aircraft and road traffic noise), and that the reliability is affected by noise level. On the aggregate level (based on the percentage of the population reporting certain responses), much better reliability can be obtained. For attitude data, percent highly annoyed is a much more reliable indicator than percent at all annoyed (r as high as 0.91 vs r no higher than 0.39).
Central representation of vowel duration
Authors: Westbury J. R. , Keating P. A.
Published: October 2005

For all languages for which there are data, the durations of vowels (of the same category) vary directly with the degree of tongue and mandible excursion specific to their articulation. This fact suggests that the durations of qualitatively different vowels can be explained in terms of peripheral constraints on articulatory movements. It is believed, therefore, that the underlying control specifications for vowels differing both temporally and in displacement require actual control only of the latter parameter. An explicit articulatory model of vowel production consistent with this view [B. E. F. Lindblom, STL-QPSR-4 (KTH), 1–29 (1967)] accounts for systematic variation in time and displacement by varying only the magnitude of force applied to articulators. New measures of acoustic vowel duration, mandible excursion, and EMG from the anterior belly of digastric are reported for two normal adult English speakers repeating isolated nonsense monosyllables. These results show, as expected, that vowel duration varies directly with mandible displacement. But, these results show also that both quantities vary directly with the magnitude and duration of activity in the jaw-lowering musculature. We interpret these data to indicate that the relative durations of vowels articulated with different mandible displacements are centrally represented. [Research supported by grants from NINCDS.]
Tongue contact duration for /t/ and /d/
Authors: Haynes Spencer , Hamlet Sandra
Published: October 2005

Tongue contact durations for /t/ and /d/ were investigated in word-initial position. Meaningful sentences containing the relevant phonetic contexts were spoken by four native speakers of American English. Tongue contact durations were measured by means of electropalatography. The microphone signal was also recorded. The signals were displayed together on two trace oscillograms run off at a paper speed such that each millisecond was represented by at least 1 mm of paper. Thus temporal measurements were highly accurate. Results indicated marked intersubject variability. For some subjects there was a strong relationship between consonant type and tongue contact duration, but the voiceless cognate was not always the longer. In other subjects the relationship was influenced by the following vowel. For one subject no significant difference was found between tongue contact duration for the two consonants regardless of the following vowel. A measurement of the period of voicelessness for /t/, from the cessation of voicing to the plosive burst, was also made. There were strong correlations between tongue contact durations and the period of voicelessness for only one subject. Although the burst was a good indicator of tongue release, as expected, there was no reliable spectral indicator of onset of tongue contact, particularly for /t/ if voicing ceased prior to contact. [Work supported by NIH Grant DE 03631.]