bluuebunny/crossref_metadata_2025 huggingface dataset:
From touch to string vibrations—The initial course of the piano tone
Authors: Askenfelt Anders , Jansson Erik
Published: October 2005

In a pilot study [J. Acoust. Soc. Am. Suppl. 1 71, S92 (1982)], measurements of key velocity, hammer acceleration, and string vibrations were reported. The study also covered the timing of the parts in the action and contact durations between hammer and string. The project was now continued by verifying measurements of key motion and string vibrations by means of a novel optical detector. In addition, the motion of the hammer has been recorded simultaneously with key motion under different playing conditions (pianist, nonpianist, pendulum). The effects of adjustment of the action on timing, as well as the effects of weight, shape, and softness (intonation) of the hammer on the string waveform and spectrum, have also been investigated. The methods and the results of the measurements will be presented and discussed.
Source location and identification by backpropagation in the time domain
Authors: Porter Robert P. , Chen Jei Shuan
Published: October 2005

Generalized backpropagation in the time domain has been applied to seismic inversion and is now being extended to the source location and identification problem in underwater acoustics. The method is based on a discrete implementation of the Huygens-Kirchhoff integral and is applicable to a wide range of wave-front migration problems including underwater acoustics. The computation is simplified by using a few nearest neighbors to calculate the field along a nearby coplanar surface. A brief discussion of generalized backpropagation in both the frequency and the time domains will be presented. The method will be illustrated by showing reconstructions of some of the numerous spread-source geometries that have been studied. Some implications of the use of this method for locating and identifying sources in the ocean will be discussed.
Signal field matching in range, depth, and bearing
Authors: Ozard John M.
Published: October 2005

Advances in propagation modeling have enabled increasingly accurate calculations of the acoustic field for situations where fairly complete knowledge of the acoustic environment is available. To take advantage of the improved propagation models, various matched field signal processing schemes have been developed. Because of the large number of possible combinations of range, depth, and bearing for a moving acoustic source, it is desirable to speed up the matching process and reduce the search region. This can be achieved by using an initial coarse global grid search followed by a fine optimization of source position to produce field matches. Further gains can be obtained by predicting the next source position so that subsequent searches can be restricted to small regions around the predicted positions. In this presentation, the essence of the normal mode propagation model and suitable processing and prediction schemes are described and applied to a shallow water environment. This is followed by illustrations of performance for a generalized beamformer and for an orthogonal technique under a variety of environmental conditions, source parameters, and array configurations.
A simple method of range and depth estimation
Authors: Yang T. C.
Published: October 2005

The range and depth of an acoustic source in the ocean can be determined by decomposing array data and beamforming on the mode amplitudes. In particular, because the mode amplitudes are proportional to exp(− ikirs) as a function of source range, the product of the mode amplitudes with the steering vector V [where Vi = exp(ikir)] is maximum for the true source range (r = rs). Accurate range and depth estimation with this approach, however, requires reliable estimates of the mode amplitudes—often difficult to obtain. In this paper, an eigenvector decomposition technique is used to extract the mode amplitudes for data received on a long (1-km) vertical array. Range and depth are successfully estimated both for simulated data and for data from the 1982 Fram IV Experiment in the Arctic Ocean. The effect of the number of modes on the estimates is also illustrated.
Range-depth ambiguity surfaces for vertical arrays in shallow-water environments
Authors: Heitmeyer Richard , Hamson Rachel
Published: October 2005

This paper describes an analytical study of range-depth ambiguity surfaces obtained on a vertical array in shallow water through the spatial correlation of the incident acoustic field with a replica of that field. The results indicate that a high-quality ambiguity surface (narrow main lobe-low sidelobe levels) can be obtained provided that the incident acoustic field is formed from a significant number of “effective” modes and that the array spans a large fraction of the water column. Changes in the acoustic environment that reduce the effective number of modes diminish the quality of the ambiguity surface. Thus the quality is reduced by either a decrease in the water depth or the source frequency or by an increase in either the source range or the bottom loss. Furthermore, although the array length is a key parameter, hydrophone spacings over twice that required to satisfy a Nyquist criteria (water depth/number modes) can be used without substantial degradation. Finally, errors in the bottom parameters, which produce errors in the replica field, shift the main-lobe peak, whereas errors in the sound-speed profile can degrade the ambiguity surface to the point where the main-lobe peak is indistinguishable from the sidelobe peaks.
The phonetic relevance of temporal decomposition
Authors: Marcus Stephen M. , Syrdal Ann K.
Published: October 2005

A simplified view of speech production considers it to be the result of a sequence of gestures toward articulatory targets, these gestures overlapping in time and thus resulting in spectral transitions in the acoustic speech signal. Temporal decomposition is an acoustic analysis technique which, without using any detailed phonetic knowledge, decomposes a speech signal into a sequence of spectral target vectors, each associated with a time varying target function. [B. S. Atal. Proc. ICASSP 83, 81–84 (1983); S. M. Marcus and R. A. J. M. van Lieshout, IPO Annual Progress Report, 19, 25–31 (1984)]. This report will further evaluate relationships between the acoustic characteristics of phonetic units and the target vectors and functions resulting from temporal decomposition. Comparisons will be made between the temporal alignment of phonetic segments and target functions. The effect on target function duration of systematic contextural effects known to affect segment durations. such as final consonant voicing on preceding vowel duration, will be explored. Spectral properties characteristic of representative vowels and consonants will be compared with those of the corresponding target vectors, and some estimates of the stability of these values will be given.
The “effective second formant” F′2: A parameter for the classification of French front vowels
Authors: Mantakas Marios , Escudier Pierre , Schwartz Jean-Luc
Published: October 2005

The “effective second formant” F′2 was analytically tested as a cue for rounding opposition in French front vowels (/i/-/y/ and /e/-/ø/ opposition). For this purpose, a corpus of these four vowels introduced into contexts maximizing and minimizing the rounding opposition [C. Abry, L-J. Boë, P. Corsi, R. Descout, M. Gentil, and P. Graillot, Labialité et Phonétique (Université des Langues et des Lettres, Grenoble, France, 1980)] was acoustically studied. The first four formant frequencies and levels were measured. The F′2 of the analyzed vowels was next estimated with a model based on psychophysical and vowel perception hypotheses and using the frequencies and levels of F2, F3, and F4 [M. Mantakas, J-L. Schwartz, and P. Escudier, Actes des 15e Journées d'études sur la Parole (GALF, Aix-en-Provence, France, 1986), pp. 157–161]. The effectiveness of F′2 as a classification parameter was furthermore compared with that of other acoustic cues proposed for French. [Work supported by CNET, France.]
Discrimination of duration ratios in Estonian and English
Authors: Fox Robert Allen , Lehiste Ilse
Published: October 2005

Estonian has been described as having a three-way quantity distinction among disyllabic words based upon whether the ratio of the durations of syllable 1 to syllable 2 is approximately 2:3, 3:2, or 2:1. The present study was designed to examine the ability of listeners to discriminate among such duration ratios independent of other phonetic factors, such as fundamental frequency or segmental variations. Pairs of noise bursts (with 10 ms rise/fall times) whose durations were in the ratios of 1:2, 2:3, 3:2, and 2:1, were constructed. In half of these noise sequences the combined duration of noise 1 + noise 2 was 350 ms; in the other half the combined duration was 450 ms. Subjects were presented with two such sequences and were required to respond whether the duration ratios in the two sequences were the same or different. Equal numbers of sames and differents were randomly presented to 56 native listeners (and five trained phoneticians). Twenty-eight listeners were native Estonians (whose data were collected in Tallinn, Estonia) and 28 listeners were native American English speakers (whose data collected in Columbus, OH). Obtained results show that both groups of listeners clearly recognized only two contrastive patterns: 1:2 and 2:3 vs 3:2 and 2:1. In addition, overall sequence duration had a significant effect upon duration ratio discriminations. These results will be discussed in light of possible language group difference, phonetic training differences, and in terms of psychoacoustic limits on the contrastive use of duration in languages.
A cross-language study of vowel spaces and interference
Authors: Jongman Allard , Fourakis Marios
Published: October 2005

This is a study of the spectral characteristics of the vowels of German and modern Greek in relation to those of American English. Two native speakers of German produced 18 words containing the short and long vowels of German, and two native speakers of Greek produced five words containing the five pure vowels of Greek. All speakers also produced nine English words containing the nine simple vowels of English. Vowel target zones for German, Greek, and foreign-accented English are established within the auditory perceptual theory [J. D. Miller, J. Acoust. Soc. Am. Suppl. 1 76, S79 (1984)] and compared to the target zones of native speakers of English. A metric of the distance between the foreign, foreign-accented English, and native English target zones will be developed. Perception experiments are planned to obtain both identification scores and paired-comparison ratings of foreign accentedness. The positions of vowels in both the native and acquired language will be correlated with these ratings to determine if an objective measure of foreign accent and phonetic interference can be derived.
Vowel duration in English word and sentence patterns as spoken by non-native speakers
Authors: Fokes Joann , Bond Zinny
Published: October 2005

Non-native speakers from five different language backgrounds (Chinese, Hausa, Japanese, Persian, and Spanish) and three native American English speakers produced tokens of two-syllable words (complete), three-syllable words (completion), and four-syllable words (competition) in isolation and in sentence context. Vowel duration, a correlate of stress, was measured in the first two syllables in words from spectrograms. The Americans used very short vowels in the reduced syllables and shortened stressed vowels in suffixed words and in sentence context. The primary difficulties of the non-native speakers were vowel reduction in unstressed syllables and the stress patterns of four-syllable words.
Accents, focus distribution, and the perceived distribution of given and new information: An experiment
Authors: Nooteboom S. G. , Kruyt J. G.
Published: October 2005

This paper reports on an experiment examining some perceptual consequences of correspondences between accent patterns, plus and minus focus, and new and given information in Dutch-spoken sentences. The main question examined was to what extent plus focus (accented) constituents are generally perceived as conveying recently introduced (“new”) information, and minus focus (unaccented) constituents as conveying earlier introduced (“given”) information. The linguistic material for the experiment was formed by brief radio news items, each two sentences long. First sentences determined the distribution of new and given information in second sentences. The accent patterns in the second sentences were varied systematically by manipulating their synthetic pitch contours according to the rules for Dutch intonation. Subjects were asked both to find a fitting first sentence to each second sentence, and to rate the acceptability of each possible combination of a first with a second sentence on a scale from 1–10. Results showed that the most preferred or acceptable distribution of new and given information closely matches the distributions of plus and minus focus, as marked in the accent patterns. New information can hardly ever be acceptably associated with minus focus, but given information can, rather often, but not always, be acceptably associated with plus focus.
Interacting effects of preceding stress and phonemic vowel quantity on consonant duration in Finnish
Authors: Välimaa-Blum Riitta
Published: October 2005

Finnish has a quantity opposition in both vowels and consonants, and demarcative stress on the initial syllable. Stressed vowels are longer than corresponding unstressed ones. A peculiar fact about short and long vowels is that they are significantly longer after a short initial syllable [CV.CV(V)] than after any kind of long initial syllable [CVV.CV(V), CVC.CV(V), CVVC.CV(V)]. A proposed explanation is that duration as a stress cue spills over to the second vowel in CV.CV(V) words in order to avoid neutralizing the phonemic vowel quantity on the first syllable. If this indeed is the source of the second-vowel lengthening, then we would expect a similar effect in syllable final consonants: The consonant after a short stressed vowel (CVC.) should be longer than the consonant after a long stressed vowel (CVVC.). The duration of geminates was studied in four word structures, CVC.CV(V) and CVVC.CV(V), and they were found to be longer in the former group than in the latter.
On the resyllabification of /l/ in English
Authors: Sproat Richard , Borowsky Toni
Published: October 2005

English /l/ is known to be light ([l]) or dark ([ɫ]) when syllable initial or syllable final, respectively. The acoustic difference is that [l]s have lower F1s and higher F2s than [ɫ]s. The effects of varying strengths of boundaries on the resyllabification of /l/ in English have been examined. Speakers were asked to produce two sets of sentences, both of which included /l/s in the vowel contexts /i_ɪ/u_ə/, and /o_ə/: (1) The control set had /l/s that were either necessarily syllable initial or syllable final; and (2) the test set had morpheme final /l/s with a vowel initial morpheme following and boundaries of varying strengths [+, ♯, ♯♯ (compound boundary), major phrase boundary, major intonational boundary] intervening. The data were analyzed by two techniques, namely traditional spectrograms, and by measuring the frequencies of the LPC-poles for the /l/s. It was found, for at least some speakers, that light /l/s are possible before all boundaries except a major intonational boundary. These results support the position that resyllabification is generally allowed across any boundary except a major intonational one. We have no data confirming the claim of Halle and Mohanan [“Segmental phonology of Modern English,” Linguistic Inquiry 16, 57–116 (1985)] that speakers resyllabify across compound boundaries but fail to do so across word or phrase boundaries.
An acoustic study of the semivowels /w,y,r,l/ in American English
Authors: Espy-Wilson Carol
Published: October 2005

The semivowels have acoustic properties that are similar to both vowels and other consonants. Ways to quantify acoustic cues that distinguish between the semivowels and separate the semivowels as a class from vowels and other consonants have been studied. To make these distinctions, the formant frequencies within the semivowels, the formant transitions between the semivowels and adjacent vowels, and relative energy measures felt to be correlates of such features as voiced, sonorant, nonsyllabic, and stoplike have been looked at. Some results of this study agree with the findings of previous studies. Namely, given minimal pair words, F1 separates the glides /w/ and /y/ from the liquids /l/ and /r/, F2 separates /w/ from /r/ and /l/ from /y/, and F3 separates /r/ and /l/. In addition, it was noted that the cues usually separate the semivowels from vowels and other consonants. In most cases where semivowels cannot be clearly distinguished on the basis of one or more features, considerable acoustic evidence of feature assimilation between them and the adjacent sound(s) exists. For example, often in words like “cartwheel” and “Norwegian,” where a vowel is followed by a postvocalic, but not word-final /r/, either the lowest point of F3 occurred at the beginning of the vowel or F3 remained at the same low frequency throughout the vowel and /r/ segments. That is, the features of the vowel and following /r/ overlapped. Results are based on measurements of 233 polysyllabic words that contain the semivowels in a variety of phonetic environments. Each word was spoken by two males and two females. [Work supported by a Xerox Fellowship.]
PHONASCII: An ASCII-based system for detailed phonetic transcription
Authors: Allen George D.
Published: October 2005

An ASCII-based system is required in order to store phonetic transcriptions in computer memories. Although several phoneme-level systems (e.g., ARPABET) already exist, none of these permits the level of detail required for close transcription. Under the partial sponsorship of CHILDES (Child Language Data Exchange System, Carnegie-Mellon University), an IPA-to-ASCII system has been developed to permit the archiving of highly detailed transcriptions. This system will be described, and its relationship to other, similar projects (e.g., Alvey) will be discussed.
Possible production differences in children's and adults' fricatives
Authors: McGowan Richard S. , Nittrouer Susan
Published: October 2005

Most developmental studies of articulatory ability and phonological organization use acoustic measurements to make inferences about production differences between children and adults. A common assumption of this approach is that the only physical differences between children and adults are the lengths of various vocal tract regions. Accordingly, any acoustic difference not accounted for by this scaling is attributed either to a lack of articulatory skill or to poorly defined phonological categories. This report considers whether other physical differences might account for some of the acoustic differences between children and adults. Speech samples of 12 speakers (8 children and 4 adults) producing the fricatives /s/ and /∫/ followed by the vowels /i/ and /u/ were analyzed to locate the major spectral prominences. Acoustic measurements showed that the fricative low-frequency prominences for children's samples differed from those of adults in three important ways. They: (1) were generally higher in frequency; (2) were relatively greater in amplitude; and (3) showed greater effects of vowel context. The first finding can be explained by a simple scaling of adult models of fricative production to accommodate children's smaller vocal tracts. The last two findings suggest, however, that there are other anatomical and articulatory differences between children and adults affecting fricative production. The data suggest the possibility that one important difference may be that the ratio of the glottal area to fricative constriction area tends to be less for children than for adults. [Work supported by NIH Grants NS-07237 and NS-13617 to Haskins Laboratories.]
Vowel context effects in children's and adults' speech
Authors: Powers Joyce
Published: October 2005

Recent studies comparing coarticulation in adults' and children's speech have yielded contradictory results; some have indicated that children coarticulate more, and others that they coarticulate less. The studies which found more coarticulation in children's speech involved fricatives and the vowels /i/ and /u/; those finding more coarticulation among adults involved velar stops and the vowels /i/ and /a/. It may be that coarticulation is not a unified phenomenon, and that the conflicting results of earlier studies were a consequence of studying different kinds of coarticulation. The present study, therefore, considers all of the things that were separately investigated before. Adults and 3- and 4-year olds were recorded producing the syllables /si/, /su/, /sa/, /∫i/, /∫u/, /∫a/, /ki/, /ku/, and /ka/. The spectral peak of the fricative or stop will be determined, as will the F2 values at the onset of voicing of the vowel. The results of the study will be considered in terms of what they imply about how children acquire the phonological units of their language.
Hearing protector attenuation for impulse noise
Authors: Forshaw S. E.
Published: October 2005

The study of the effectiveness of hearing protection for impulse noise pose several problems. Due to the transient nature of the noise, real-ear-at-threshold (REAT) data are not readily applicable since current impulse-noise criteria are based on either (1) peak pressure, some measure of impulse duration, and the number of impulses, or (2) total A-weighted energy. It is possible to measure the effect of an earmuff on these parameters by positioning a miniature transducer under the muff at the wearer's ear canal. An alternative arrangement which permits the study of both ear muffs and plugs involves the use of an acoustic test fixture. The test fixture must simulate those characteristics of the head and outer and middle ears that affect the attenuation of hearing protectors on real ears. These include the viscoelastic characteristics of the circumaural and intra-aural skin, the inertial properties of the head, the frequency response of the auditory canal, and the mechanisms of bone conduction and physiological masking. DCIEM has sponsored by contract the development of such a test fixture [C. Giguere et al., J. Acoust. Soc. Am. Suppl. 1 78, S5 (1985)]. Attenuation results with the test fixture compare reasonably well with REAT-derived data. Agreement between the two methods could be improved for earplugs if the thickness of the artificial intra-aural skin were reduced. On-going research is directed towards studying human intra-aural skin, and measuring the peak attenuation, waveform-duration increase, and linear and A-weighted energy reduction of protectors for short- and long-duration impulses.
The development of hearing protective systems for use in the high-noise environments of army combat vehicles
Authors: Mayer Mitchell S. , Fung Tat Y.
Published: October 2005

Military vehicles are designed for maximum capabilities in areas of performance, armament, and endurance. Acoustic noise suppression at the source is not considered in the system design because of its degradation on performance, as well as its cost impact. Therefore, it is left to the crew helmet to provide for the hearing protection of the crew member. In order to protect the hearing and improve the job performance of the crew, there are two major noise paths to be considered: first is the noise pickup by the communication microphone, along with the speech communication; second is the environmental noise that penetrates the earcup. It is believed that additional improvements to such devices as noise canceling microphones and earcup/transducer assemblies have reached their practical limits. Therefore, research efforts are presently being conducted to incorporate active speech processing for rejecting noise picked up by the communication microphone, and active noise reduction to reduce the environmental noise penetrating the earcup. These efforts will be discussed, along with new types of test equipment and evaluation procedures that have been developed to more accurately evaluate existing communication components and future active speech processing/noise reduction systems.
Anticipatory coarticulation in children with good and poor syntactic abilities
Authors: Lieberman Philip , Sereno Joan A.
Published: October 2005

The present study investigates the relation between speech coarticulation and syntactic processing in children. Specifically, acoustic and perceptual measurements of anticipatory labial and lingual coarticulation are compared to the syntactic abilities of children, as measured by the Rhode Island Test of Language Structure. Two children with good (greater than 95% correct) and two children with poor (less than 85% correct) syntactic comprehension scores produced five tokens each of the following CV syllables ([si], [su], [zi], [zu], [ti], [tu], [di], [du], and [ki], and [ka]). The effects of the vowel on the preceding consonant were acoustically assessed by measuring both second formant and characteristic spectral prominence regions in the consonantal stimuli. Perceptual measurements were then performed by presenting these excised consonantal segments for vowel identification. The correlation between the coarticulation data and the measurements of the children's syntactic abilities is evaluated. The results are discussed in terms of theories of motor programming, the evolution of the hominid brain, and the relationship of speech motor programming to the development of syntax.
Theoretical models of ultrasound absorption in tissues
Authors: Greenleaf James F. , Sehgal Chandra M.
Published: October 2005

On the basis of many measurements on attenuation, absorption, and speed of ultrasound in biological materials, one can make the following generalizations: (a) Biological tissues have high attenuation coefficients, about two to three orders greater than their major component, water; (b) the amplitude attenuation coefficient increases with frequency near-linearly in soft tissues and near-quadratically in fluids with frequency in the range 1 to 10 MHz, and possibly even up to 100 MHz; (c) speed dispersion is extremely small (∼0.1 to 1 m/s/ MHz); (d) absorption and attenuation coefficients either increase or decrease with an increase in temperature, depending on the nature of the tissue, and the frequency of irradiatiOn; (e) fluids containing proteins show a sharp increase in excess absorption in acidic and basic pH ranges, and finally; (f) at sufficiently high intensity of irradiation the absorption coefficient increases with intensity. Comprehensive models for the propagation of ultrasonic waves through lossy media should explain these experimental findings. We will review some of the current models and critically examine them in light of the definition of absorption in relation to attenuation and scattering. [Work supported by NIH CA 24085, CA 41324, and NSF 8310626.]
Biomolecular ultrasound absorption
Authors: Dunn F.
Published: October 2005

Studies of the bulk ultrasonic properties of intact tissues reveal the importance of biomacromolecules in determination of ultrasonic absorption. The absorption coefficient of blood has been found to be proportional to the weight percentage of the protein content, which has encouraged studies of the absorption by proteins and their analogs in solution. Studies of amino acids, polyamino acids, and soluble proteins in aqueous solution have suggested such mechanisms as relaxation of proton-transfer equilibria occurring at ionizable residues and relaxation of solvent-solute equilibria. Though other mechanisms have been proposed, no single mechanism appears to be predominant in being responsible for ultrasonic absorption by proteins.
Calculation of temperature elevation caused by ultrasound absorption
Authors: Nyborg Wesley L.
Published: October 2005

It has been known for decades that ultrasound is capable of producing physiological change in man and in laboratory animals. A dominant mechanism for the change is temperature elevation resulting from sound absorption. It has been shown for some situations that the processes for heat generation and transport are understood well enough to make it feasible to predict the temperature that results from specific acoustic exposures. Thus Pond, Robinson and Lele, Carstensen, Lizzi, and others have successfully calculated intensity-time combinations required for production of recognizable lesions in mammalian tissues and other media. There is now special interest in computational methods for application to ultrasonic hyperthermia, and also to the formulation of safety criteria for diagnostic ultrasound. For the latter purpose a simplified method has been developed, based on a Green's function solution to the bio-heat transfer equation. In this paper, previous work on temperature calculations is reviewed, and recent results discussed.
A design optimization for shipboard machinery foundations subjected to noise and shock requirements
Authors: Miller Russel , Nuss Ken , Kasper Peter , Amir George
Published: October 2005

The overall methodology used to design foundations are sometimes a compromise between the various conflicting requirements of noise, shock, vibration, ship motion, thermal motion, and machinery imposed loads. The methodology developed is based on iterative structural design to achieve optimum stiffness and the predominant natural frequency that best mitigate shock and vibration motions prescribed by realistic shipboard environment. Experimental data of tested foundations in the prescribed shock environment is used to verify the validity of the structural optimization method developed. The paper describes the computer code developed for the structural optimization. Selection of various foundation categories are available for various locations in the ship and various installation configurations. Comparisons between structurally optimized and nonoptimized foundations are made demonstrating substantial cost and weight saving as well as improved hardness and vibration reduction. The iterative process of structural optimization for shock and vibration is linked to noise attenuation performance. The analysis of noise transmission through the foundation was performed using the direct dynamic stiffness method (DDSM). This approach provides velocity, force, and power transfer functions between input excitation positions and hull intersection positions.
Vibration damping performance—What we should know about it
Authors: Saha Pranab
Published: October 2005

The vibration damping performance of materials that are used in the automotive industry are usually tested by either: (1) the Geiger Plate Test Method, or (2) the Complex Modulus Test Method. The first test method has some advantages of testing materials that are unique to the automotive industry. Both of these methods have certain limitations, though the second method is a superior one in the sense that it can measure the damping performance with temperature and frequency. This paper discusses the limitations of some of the ways the data are typically presented using this method. The paper also discusses how design charts can be used to overcome these limitations and compute the damping performance of the composite material (composite loss factor) for a given application.
Dynamics of piezoelectric rods under initial stresses
Authors: Dokmeci M. Cengiz , nanI'D E. E.
Published: October 2005

The analysis presented is an extension of our earlier treatment of rods [Int. J. Solids Struct. 8, 1205–1222 (1972); and 10, 401–409 (1974)] in which the effect of initial stresses is neglected. In this study, this effect is considered on waves and vibrations of piezoelectric thin rods. First, a variational principle due to the author [Proc. 39th Freq. Control Symp., pp. 431–435 (1985)] is reformulated, and the field quantities are expanded in terms of Legendre polynomials. Then, the expansions together with the variational principle are used so as to consistently establish a hierarchical system of one-dimensional equations of crystal rods. The governing equations may incorporate the higher-order effects of piezoelectric rods by a proper truncation of expansions. They accommodate all the types of extensional, flexural, and torsional motions of rods. Further, the resulting equations are considered for special motions, and they are shown to agree with those of earlier ones. [Work supported in part by the U.S. Army through its European Research Office.]
Auditory system impedance measurement using the two-microphone wave-tube technique
Authors: Keefe Douglas H. , Burns Edward M. , Ling Robert
Published: October 2005

An impedance measurement technique suitable for middle ear investigations has been devised that has the advantages of fast data acquisition (less than 1 s) and large measurement bandwidth (200–6000 Hz). This technique does not require a transducer delivering a constant volume flow. The excitation used is a broadband chirp whose spectral shape may be varied to enhance signal to noise ratio. The method depends upon pressure measurements at two points within a smooth, cylindrical tube whose transmission characteristics are well known. In our implementation of this technique, the frequency-dependent viscous and thermal losses in the wave tube are accounted for. The wave tube is inserted into the ear canal, and the input impedance at the tube tip is computed based upon the tube transmission coefficients and the measured transfer function between the microphones. Measurements are presented of the input impedance in normal adult ears for comparison with the literature. Since the technique is fast and relatively noninvasive, it appears well-suited for our project goal of measuring the middle ear impedance of infants. [Work supported by NINCDS.]
Temporal integration in normal hearing, cochlear impairment, and impairment simulated by masking
Authors: Florentine Mary , Fastl Hugo , Buus Søren
Published: October 2005

To assess temporal integration in normal and impaired listeners, absolute thresholds for tones were measured as a function of duration. Durations ranged from 500 ms down to 15 ms at 0.25 kHz, 8 ms at 1 kHz, and 2 ms at 4 and 14 kHz. An adaptive 2I, 2AFC procedure with feedback was used. On each trial, two 500-ms observation intervals, marked by lights, were presented with an interstimulus interval of 250 ms, The monaural signal was presented in the temporal center of one observation interval. The results for four normal and six impaired listeners show (1) as expected, thresholds for normal listeners decrease about 8 dB per decade of duration, (2) for impaired listeners, the decrease usually is much less (2 to 6 dB) in the range of a hearing loss, but normal at frequencies where thresholds are normal, (3) for listeners with impairments simulated by masking, the slopes are nearly the same as those for normal listeners and steeper than those for the real impairments. These results indicate that the shallow slopes observed for impaired listeners probably are not due to splatter of energy to frequency regions where thresholds are low, but reflect diminished temporal integration, per se. [Work supported by NIH-NINCDS RO INS 18280.]
Comparison between loudness functions in noise and in noise-induced hearing loss
Authors: Hellman Rhona P.
Published: October 2005

Pure-tone loudness functions were generated in a cochlear-impaired population with moderate to severe noise-induced losses. Three procedures were used: magnitude estimation, magnitude production, and cross-modality matching. The slope of the loudness function was determined over the stimulus range where cochlear impairment steepens the loudness function. Both the measured and predicted slopes show that the steepening of the loudness function depends on the severity of the hearing loss; that is, the higher the threshold the steeper the function. While this behavior is also observed in partially masked normal ears, the extent of the agreement between the partially masked and impaired loudness functions is determined by the bandwidth of the masking noise. The impaired loudness functions closely agree with those obtained in wide-band noise, but not with those obtained in narrow-band noise as wide as an octave. This finding implies that the effect of the cochlear impairment on the loudness function resembles that of an external noise with a broad frequency spectrum. [Work supported by the Rehabilitation Research and Development Service of the VA.]
Loudness judgments in recruiting ears as influenced by intensity, and interspersed changes in intensity
Authors: Kurman Maureen A. , Weiler Ernest M. , Dell'Aira Angel
Published: October 2005

It has been found that loudness judgments adapt to monaural stimulus increments [E. M. Weiler, D. E. Sandman, and L. M. Pederson, Brit. J. Audiol. 15, 201–204 (1981)]. The present study was undertaken to determine whether the changes in high-frequency recruiting ears measured at a nonrecruiting frequency (1000 Hz), were the same as in normal listeners. Both groups showed a similar decline in judgment to the baseline 660-dB tone. There was a significant interaction between groups, intensity, pulsed duration of an interspersed 10-dB increment, and successive judgments of loudness. Similar interactions were seen at 20 dB above the baseline intensity.
Modeling sensorineural hearing loss. I. Approach and application to frequency resolution
Authors: Humes Larry E. , Espinoza-Varas Blas , Watson Charles S.
Published: October 2005

In the present paper we evaluate the extent to which auditory-processing deficits observed in listeners with sensorineural hearing loss can be predicted entirely on the basis of loss of hearing sensitivity. In this model, processing deficits in the impaired ear are considered to be equivalent to those observed in normal listeners when masked by spectrally shaped noise. For normal listeners with such a simulated hearing loss, performance on tasks with actual masking noises thus involves the detection of signals in the presence of two maskers. One masker is used to simulate the hearing loss while the other is incorporated in the particular paradigm under study. The masked thresholds from each masker can be combined using a masking-additivity rule similar to that of Lutfi [J. Acoust. Soc. Am. 73, 262–267 (1983)]. This approach is used to predict published data from impaired ears on critical ratios, critical bandwidths, auditory filter shapes, masking patterns, and psychophysical tuning curves. For most of the paradigms studied, the psychoacoustic performance of the hearing impaired was predicted reasonably well with this approach. These results seem to suggest that the frequency-resolution deficits observed in the hearing impaired are primarily a consequence of altered hearing sensitivity. [Work supported by NINCDS.]
Modeling sensorineural hearing loss. II. Application to measures of temporal resolution
Authors: Espinoza-Varas Blas , Humes Larry E. , Watson Charles S.
Published: October 2005

Thresholds of hearing-impaired listeners in temporal-masking tasks were modeled by the performance of normal listeners with a noise-simulated hearing loss. This model predicts thresholds for such tasks by assuming the presence of two maskers; a spectrally shaped continuous masker used to simulate the hearing loss and the masker used in the temporal-masking task. The combined effect of the two maskers was predicted using a masking-additivity rule similar to that of Lutfi [J. Acoust. Soc. Am. 72, 262–267 (1983)]. Masked detection thresholds from impaired listeners and normal listeners with simulated hearing loss were examined for various temporal-masking paradigms. Data from the literature were examined for each of the following: (1) detection thresholds in continuous and modulated broadband noise; (2) growth of forward masking with masker level; and (3) decay of forward masking following masker offset. The masking-additivity approach predicted well the temporally masked detection thresholds for noise-masked normal listeners with simulated hearing loss, but underestimated the masked thresholds of the hearing impaired by as much as 15 dB. [Work supported by NINCDS.]
The design of a speech analysis workstation
Authors: Crump John M.
Published: October 2005

The recent availability of general purpose digital signal processing chips, inexpensive digital memories, and personal computers has opened the door to the development of a powerful workstation designed for the analysis of speech. A system can be developed with the advantages of a spectrograph (e.g., Sona-Graph), an oscillograph (e.g., Visicorder), a feature extractor (e.g., Visi-Pitch), and a general purpose computer (e.g., VAX with DSP software). This paper describes the development and capabilities of a standalone system based on a common microprocessor, powerful digital signal processing circuits, high-resolution graphic displays and high-speed DMA capabilities. The digital signal processing chip selected was the 32020 from Texas Instruments. Special high-speed DMA chips were used to facilitate high-speed data transfer between the different system modules (A/D to memory, memory to DSP circuits, DSP circuits to graphic circuits, and DSP circuits to printer). These DMA chips allow a 2-Mbyte/s transfer rate. The system management is performed by a Motorola 68000 and the system architecture has been defined to include up to 8 Mbytes of RAM. This system performs all the desired functions of a speech workstation and could provide a common platform for speech analysis research.
Vocal jitter and shimmer measurements with a personal computer
Authors: Drumright David G. , Seikel J. Anthony , Wilcox Kim A.
Published: October 2005

Vocal fundamental frequency is one of the most widely analyzed components in speech acoustics. As such, there is a need for an efficient and accurate means of producing fundamental frequency data. This paper describes a series of machine language and basic routines that yield average fundamental frequency, vocal jitter, and vocal shimmer measures with a resolution sufficient for most research purposes. The algorithm permits the user to determine major period divisions through an efficient interactive graphic process. The microcomputer is then responsible for computing accurate period and intensity data for each cycle. In the present installation, the accuracy of these procedures is limited only by the 20 000-Hz maximum sampling rate of the hardware. In addition to the f0 analysis routines, programs for high-speed digital sampling, waveform editing, and waveform manipulation have been developed for use with PC-compatible computers. Documentation of the installation, use, and accuracy of the routines will be provided. [Work supported by NINCDS and U. of Kansas General Research Fund.]
Some results of speech analysis using a parallel image processor
Authors: Hemdal John F.
Published: October 2005

Special purpose parallel pipelined image processors are not only faster for images than general purpose computers but also allow one to investigate the applications of the latest image processing algorithms and techniques to continuous speech. This is particularly valuable when trying to automate the spectrogram reading capabilities of Victor Zue [Proc. IEEE 73, 1602–1615 (1985)]. Several sentences of continuous speech were sampled and converted to images using the FFT. These images were analyzed on a parallel cellular image processor called the Cytocomputer™. Suitable structuring elements were selected to smooth the inherently bumpy FFT spectra and to remove noise. Several of the distinctive features of speech are described in mathematical morphology terms and these features were tracked. Specifically, stop consonants, fricatives, vowels, and nasals were located and labeled. Examples of intermediate image transformations and resultant labeling were recorded by photographic process. [Work supported by Ohio Board of Regents.]
ARMA parameter estimation of speech
Authors: Wygonski John J. , Sohie Guy
Published: October 2005

Autoregressive moving average (ARMA) or pole-zero modeling techniques are known to provide better spectral estimation capabilities than autoregressive (AR) or all-pole methods such as linear prediction for nasalized speech and speech corrupted by noise. This paper describes the results of applying two ARMA modeling techniques to synthesized and natural segments of noisy and clean speech. The overdetermined normal equation (ODNE) and the extended-order singular value decomposition (SVD) methods of ARMA parameter estimation are compared to the covariance method of linear prediction and are evaluated using parametric distance measures in the cepstral domain. Use of the AR part of the ARMA model to estimate the envelope of noisy speech is demonstrated. Coefficient transformations among ARMA, AR, and cepstral parameters are also discussed.
A codebook of articulatory shapes
Authors: Sondhi M. M. , Schroeter J. , Larar J. N.
Published: October 2005

The acoustical properties of a vocal tract depend on its shape which, at frequencies below about 4 kHz, is adequately described by the area function (i.e., cross-sectional area as a function of position from glottis to lips). Obtaining even this simplified description of the tract is a difficult problem. The solution that is proposed is to construct a linked codebook of area functions and corresponding LPC vectors. Then, a given LPC vector is transformed to an area function by finding the closest LPC vector in the codebook and accessing the linked area function. The authors have constructed just such a codebook (to our knowledge, the first). At present, it has about 1500 shapes clustered from a training set of about 10 000 shapes generated by a vocal tract model [P. Mermelstein, J. Acoust. Soc. Am. 53, 1070–1082 (1973)]. Distance between area functions is defined as the Itakura distance between the corresponding LPC vectars. Handling the unique problems arising from this definition of distance and from the large size of the training set will be described. The plan for refining and enlarging the codebook will also be discussed.
Statistical representation of word-initial obstruents: General considerations
Authors: Weismer Gary , Forrest Karen , Milenkovic Paul
Published: October 2005

The description of obstruent spectra has typically been done in categorical terms [e.g., S. E. Blumstein and K. N. Stevens, J. Acoust. Soc. Am. 66, 1001–1017 (1979)]. Whereas this may be satisfactory for some purposes, the categorical approach is not helpful in cases where obstruent sounds are produced that are not easily described according to the normal set of segmental contrasts. In particular, many dysarthric and apraxic subjects produce stops and fricatives that give the perceptual impression of being “between” the normal segmental categories. The purpose of this paper is to describe some of the general considerations that led us to develop measures capable of indexing spectra associated with such indeterminate obstruent productions. A statistical representation of obstruent spectra will be described that is similar to certain approaches to describing a more general class of impulse noises [J. Erdreich, J. Acoust. Soc. Am. 79, 990–998 (1986)]. [Work supported by NIH Awards NS13274 and NS20976.]