bluuebunny/crossref_metadata_2025 huggingface dataset:
Consonant type and consonantal effects on tone in Musey
Authors: Shryock Aaron
Published: October 2005

In Musey, a Chadic language of northern Cameroon, the tonal pattern of a word is partly predictable from the initial consonant of the word. There are two series of obstruents and two h's that contrast word initially. One series is associated with low initial tone, the other with high or midtone; there are, however, minimal pairs which indicate that the initial consonantal contrast in Musey is independent of the tonal contrast. Consonantally conditioned alternations in tone in other languages have been associated with voiced and voiceless consonants or breathy voiced consonants. However, preliminary studies of the acoustic nature of the Musey contrast have shown no systematic contrast in voice onset time or breathiness. In order to understand this question better, additional parameters of contrast that might underlie this effect will be investigated. The short-term phonetic effect on fundamental frequency at the release of these consonants will also be examined. The issues raised by this problematic consonant distinction for understanding the relationship of consonant type and tone will be discussed.
Shaped hydrophone measurement near the surface of a structure
Authors: Ng Kin W. , Covey Joel F.
Published: October 2005

Acoustical measurements in the vicinity of structures are generally contaminated by scattering from the structures. One approach toward reducing the contamination in measurements of free-field pressure is to use the large area polymer (LAP) hydrophone. Results will be presented to demonstrate the effectiveness of a shaped, area-averaging hydrophone in obtaining a more accurate measurement of the pressure field near the surface of an idealized structure.
Articulatory and acoustic differences between palatal and velar stops
Authors: Keating Patricia A. , Lahiri Aditi
Published: October 2005

Many experimental studies have shown that velar consonants have a more forward place of articulation in front vowel contexts than in back vowel contexts. On some accounts, such velar fronting is basically the same process as the secondary articulation of palatalization, and furthermore, both “fronted” and “palatalized” velars are equated with the “palatal” cononants of other languages. In this study, velar stops (in English, Russian, Czech, Hungarian) are compared with palatal stops (in Czech, Hungarian) and with palatalized velar stops (in Russian) in different vowel contexts. X-ray and palatographic data from the literature show that palatalized velars are very much like velars in high front vowel context, but that both of these are very different from palatals. Acoustic analysis comparing spectra of consonant release and the following vowel agrees with the articulatory data. It also makes clear that languages with palatals also have fronted velars, differing in the distribution of energy across frequencies. These results indicate that palatalization of velars is phonetically equivalent to fronting, but that neither fronted nor palatalized velars are like the palatal consonants.
Production of the English word-final /t/-/d/ contrast by native speakers of Mandarin and Spanish
Authors: Flege J. , Munro M. , Skelton L.
Published: October 2005

Previous second-language (L2) research has shown that if the voicing of word-final obstruents exerts little or no effect on preceding vowel duration in a language, then native speakers of that language will produce a far smaller vowel duration difference in isolated English words like beat and bead than do native speakers of English. This might be due to phonetic interference, resulting from the perceptual identification of word-final consonants in English and the L1. If so, one would expect native speakers of languages without word-final obstruents to succeed better in acquiring the vowel duration cue to the word-final English /t/ vs /d/ contrast. To test this, native speakers of English, Spanish, and Chinese read lists of minimally paired /bVt/ and /bVd/ words. Native English-speaking listeners later identified far more final stops produced by native speakers of English than Spanish or Mandarin (95% vs 72%, 63%). When closure voicing and release burst cues were removed, the correct identification rates remained well above chance for the native but not the non-native speakers (88% vs 46%, 58%). This was probably because the vowel duration difference in words ending in /t/ and /d/ was far greater in words spoken by the native speakers of English than by the native speakers of Spanish and Chinese (87 vs 33, 34 ms). Also, F1 offset frequency was significantly higher in /t/- than /d/-final words spoken by the native but not by the non-native speakers. The results suggest that the inability of adult L2 learners to master the vowel duration cue to the English word-final /t/-/d/ contrast is not the result of cross-language phonetic interference. Some nonperceptual explanation must be found to account for this seeming limitation.
Sound production in fricative consonants
Authors: Shadle Christine H. , Oeters Donald I.
Published: October 2005

A series of experiments with mechanical models of fricative consonant articulatory configurations has been conducted to determine where in the tract the turbulence noise is generated and the spectral characteristics of that noise. The latest models, based on a combination of x-ray, EPG, and photographic data, have the correct midsagittal profile and area function, and thus have the most realistic shape of model work to date. Data obtained from /s, ∫/ substantiate earlier results based on a different subject [C. H. Shadle, J. Acoust. Soc. Am. Suppl. 1 84, S115 (1988): also in Speech Production and Speech Modelling, edited by W. Hardcastle and A. Marchal, Proc. of NATO-ASI, Kluwer Acad., pp. 127–219 (1990)] and results from extremely idealized models [C. H. Shadle, Proc. 12th ICA, paper A3–4, Toronto (1986)]. Comparisons across a range of flow rates, with and without sublingual cavity, between measured source and farfield spectra, and between speech and model data for /s, ∫, ç, x/, lead to source parameters, a distinction between two source types, and to the conclusion that the three-dimensional shape of the tract is crucial in determining source parameters: These parameters can be used in a model based on one-dimensional sound propagation. Three-way comparisons between far-field sound measured (1) for the models, (2) for actual utterances, and (3) for far-field sound predicted from measured source parameters used in a model based on one-dimensional sound propagation will be shown. [Work supported by SERC.]
Sentence final fricative spectrum variation as a function of prosodic context
Authors: McRoberts Gerald W. , Faber Alice
Published: October 2005

The spectra of word final [s] in /CVCs/ words (where V is one of {a i o u} and C2 one of {k t}) in sentence final position for questions and statements produced by female talkers were compared. The results indicate that the frequency of peak spectral energy in the [s] was higher for questions than for statements by several hundred hertz. The difference in spectra was generally greater following /k/ than following /t/. The expected lower-frequency spectral energy was observed following the back vowels /o u/, despite the intervening consonant. These results are consistent with Honda's (1983) hypothesis of a biomechanical interaction between supralaryngeal articulation and laryngeal F0 control. Honda suggests a “supplementary tensor mechanism” by which genioglossus and geniohyoid activity displaces the hyoid bone forward and hence rotates laryngeal structures. By the same token, increased tension of the vocal folds may require modification of the shape of the upper vocal tract. It is proposed that the supplementary tensor mechanism facilitates F0 raising during a question final rise. Increased tongue fronting, especially during a /k-s/ sequence, results in a smaller resonating cavity for /s/ and therefore higher-frequency energy distribution during the /s/ for questions than for statements. [Work supported by NIDCD grant DC-00016 and MH41511.]
Preferred syllables
Authors: Maddieson Ian , Precoda Kristin
Published: October 2005

It is commonplace that universally preferred syllables reflect speaker or listener preferences in producing or decoding speech. Two strategies are usually discussed: Speakers prefer to adjoin segments that involve the least articulatory effort in the transition; listeners prefer adjoining acoustically well-distinguished segments since such sequences are easier to decode. In a study of CV sequences in five languages, Janson (1986) reported a preference for combining front consonants with front vowels and back consonants with back vowels, and concluded that “mak[ing] things easier for the speaker clearly overrides the…needs of the listener.” A reanalysis of Sanson's materials suggests that this conclusion is not supported by his data, but neither is there evidence for acoustic distinctiveness playing a decisive role. This inconclusive result might arise from using text frequency counts, where rare syllable types in frequent words can skew totals. Hence, studies of preferred syllables in a number of additional languages, particularly ones with small inventories of permitted syllables, have been conducted using lexical frequency counts. These studies continue to show that the general preference strategies outlined above account for little of the pattern of syllable frequencies. Instead, answers may more likely be found in detailed analyses of individual consonant-vowel pairings.
The problem of scales in scattering from rough surfaces
Authors: Dashen Roger
Published: October 2005

Theoretical work on scattering from rough surfaces is often plagued by an extreme range of scales of surface roughness. In particular, a wide range of scales can make numerical calculations impractical. Roughness on scales longer than a wavelength can be handled by an improved composite model. Scales smaller than a wavelength can, in certain circumstances, cause serious trouble. The cross section can depend explicitly on the cutoff at high wave numbers. [Work supported by ONR.]
The accuracy of the single backscattering, multiple forward scattering approximation for low grazing angle sea surface reverberation
Authors: Thorsos Eric I.
Published: October 2005

Accurate modeling of reverberation from the rough air-sea interface in the presence of general sound-speed profiles is difficult at low grazing angles because multiple forward scattering may occur. This paper examines the accuracy of an approximation in which multiple forward scattering, but only single backscattering, is taken into account. This approximation has been used by Tappert and Ryan [J. Acoust. Soc. Am. Suppl. 1 86, S65 (1989)]. The 2-D scattering problem (1-D rough surface) ignoring bubbles is considered. Isovelocity conditions are assumed so exact that integral equation solutions can be used as ground truth. To implement the approximate method, the surface is partitioned into large- and small-scale surfaces. The scattering problem is treated as a sequence of three steps: (1) propagation to the surface and multiple forward scattering along the large-scale surface; (2) local backscattering using first-order perturbation theory applied to the small-scale surface; (3) multiple “forward” scattering along the large-scale surface in the backward direction and propagation away from the surface. The accuracy of this method is shown to be good (for all scattering directions) at low frequencies when integral equation solutions are used for steps (1) and (3). The integral equation solution for the total surface yields the exact solution. Next it is shown that the accuracy remains good for low grazing angle backscattering when the PE approximation is used. This is done using PE integral equation solutions in steps (1) and (3). Finally, in collaboration with Ballard and Ewart (APL-UW), this method has been applied to marching algorithms in the PE approximation; the result is a marching backscattered field. The marching approach can be used with general sound-speed profiles. The techniques described are useful for testing other (more computationally efficient) approaches to the sea surface reverberation problem. [Work supported by ONR.]
Full-wave three-dimensional modeling of long-range oceanic boundary reverberation
Authors: Tappert Frederick
Published: October 2005

The capability of coupled two-way parabolic equations to model oceanic volume reverberation [F. Tappert, The Parabolic Approximation Method (Springer-Verlag, Berlin, 1977)] has been extended to three dimensions and to boundary reverberation [F. Tappert and F. Ryan, J. Acoust. Soc. Am. Suppl. 1 86, S65 (1989)]. A single backscattering, multiple forward scattering approximation [E. Thorsos, J. Acoust. Soc. Am. Suppl. 1 88, S83–S84 (1990), which is the paper previous to this one], together with reciprocity, provides computational feasibility for real oceanic conditions. Results from this model that illustrate the complicated nonlinear interactions between surface and bottom scattering will be discussed. For example, at higher sea states the effects of multiple forward scattering by surface wind waves cause bottom reverberation levels to increase at short ranges and to decrease at long ranges due to altered ensonification of the seafloor. [Work supported by ONR and ONT.]
Ocean surface reverberation: Evidence for backscatter from near-surface vertical filamentary structures
Authors: McDonald B. Edward
Published: October 2005

A surface reverberation model is proposed based on volume scatterers whose geometry is descriptive of recent ocean observation [e.g., Farmer and Vagle, J. Acoust. Soc. Am. 88, 1897 (1989)]. Scatterers in this model are vertical cylinders of elliptical cross section representing either filamentary- or sheet-like sound-speed inhomogeneities whose strength decreases exponentially with depth. This geometry is intended to describe intermediate-aged microbubble “fossils” of breaking waves and/or convective processes. Born approximation backscatter results for this geometry show substantial agreement with high-frequency surface reverberation data. Backscatter cross sections are examined as a function of wind speed, acoustic frequency, and grazing angle. It is demonstrated that almost all the high-frequency backscatter in the model is specular reflection from surfaces of volume scatterers. Some preliminary speculations involving Langmuir circulation are offered for the application of this model to the low-frequency surface reverberation problem, where data are scarce. [Work supported by ONR.]
Simulation of acoustic backscatter from bubble plumes using a finite-element model and a cw full-field backscatter method
Authors: Chin-Bing Stanley A. , Murphy Joseph E.
Published: October 2005

Backscatter from large bubble plumes has been suggested as a possible cause of anomalous reverberation. This possibility has been examined using a finite-element full-wave acoustic model, FOAM [Murphy and Chin-Bing, J. Acoust. Soc. Am. 86, 1478–1483 (1989)] and PE-FFRAME, a hybrid marching version of FOAM, applied to a large sea-surface plume having sound speed and density representative of a 30% volume fraction. A cw differencing technique was applied to isolate the backscatter. The technique involves calculating the total full-wave acoustic field with and without the object (plume) present; taking the difference of the two complex pressure fields in the backscattered direction gives a measure of the backscattered acoustic field caused by the object. The cw full-field backscatter method (cw FFBM) has been benchmarked against an analytic calculation of the backscattered field from a rigid plate. This and other benchmark examples will be presented that validate the cw FFBM. Plume backscatter using the cw FFBM was examined for environments in the Norwegian Sea and the Pacific. Internal wave fields were also introduced into both the Norwegian Sea and Pacific environments via an internal wave model. Numerical simulations using PE-FFRAME and the cw FFBM will be presented that show the combined effects of long-range (one convergence zone) monostatic and bistatic backscatter from a large bubble plume in the presence of an internal wave field and the sea-surface boundary. [Work supported by ONR/NOARL.]
Acoustic Doppler evidence for a near-surface bubble layer
Authors: Alexandrou D.
Published: October 2005

Acoustic reverberation data were collected with a horizontally directed pulse-to-pulse incoherent Doppler sonar operating at 75 kHz and deployed near the surface. The spectral moment estimation method was used to estimate velocity profiles. Clear evidence of interaction with the surface is presented by both the reverberation intensity and velocity signatures. However, an interesting anomaly is revealed in the joint behavior of surface velocity and surface reverberation intensity. Specifically, the surface velocity event does not coincide in range with the corresponding intensity feature. An interpretation of this behavior is offered based on the sonar beam geometry and the hypothesized presence of a thin bubble layer entrained just below the surface. Realistic computer simulations are used to provide support for this hypothesis and to estimate the backscattering coefficient of the volume reverberation produced by the bubble layer. This backscattering coefficient, combined with the theoretical target strength of a resonant bubble, leads to an estimate of bubble concentration. [Work supported by ONR.]
Scattering from very rough surfaces based on the modified second-order Kirchhoff approximation with angular and propagation shadowing
Authors: Ishimaru Akira , Chen Jei S.
Published: October 2005

A theory of scattering from very rough surfaces is presented. Both the surface rms height and the correlation distance are close to a wavelength and the rms slope is close to unity. The theory is based on the first- and modified second-order Kirchhoff approximations. The second-order scattering includes the incident and scattering shadowing and the angular and propagation tapering shadowing. The angular shadowing limits the angular spectrum of the second-order scattering within those intercepted by the surface, while the propagation shadowing limits the propagation distance within those intercepted by the surface. The calculations are made for the rms height equal to 1.5, 1.0, and 0.5 wavelengths and the correlation distance equal to 1.8 wavelengths and show close agreement with the Monte Carlo simulation of the exact integral equation. The results show that the first-order Kirchhoff scattering is dominant for rms height equal to 0.5 wavelengths, but for rms heights equal to 1.0 and 1.5 wavelengths, the second-order scattering becomes comparable to the first-order and produces the enhanced backscattering. Therefore, this theory provides a possible physical explanation for the enhancement. The range of validity of the theory is also discussed. [Work supported by ARO and NSF.]
Characterizing pulse signals reflected from rough surfaces
Authors: Bozma Ömur , Kuc Roman
Published: October 2005

Most of the work on scattering from rough surfaces is developed for monochromatic wave sources. In this work, the wideband pulse signal reflected from a three-dimensional rough surface is analyzed. The frequency correlation function of the acoustic channel is derived using the Kirchhoff approximation method for a rough surface having the Gaussian roughness spectrum. The expected value of the received signal and its intensity are characterized as time-evolving functions. The intensity of the reflected pulse is related to the two-dimensional inverse Fourier transform of the product of the pulse power spectrum and the frequency correlation function. The coherent and incoherent components of the reflected intensity are described. The width of the coherently reflected pulse depends on the transmitted pulse width, whereas the width of the incoherent component is a function of the illuminated area on the surface. The theoretical predictions are verified with results obtained using the Polaroid transducer operating at a center frequency of 60 kHz from two different rough surfaces. The theoretical predictions are shown to agree reasonably over the range of observable angles of incidence from 0° to 60°. [Work supported by NSF: ECS-8802627.]
Phase resetting and rhythmic performance
Authors: Lame Gerald D.
Published: October 2005

Both music and speech are transmitted primarily by sound, and both can be characterized as rhythmic. When people hear a regular series of sounds, they are able to “follow” it much better than they can a similar series of flashing lights, whether this involves synchronizing overt movements with the series, or merely counting. A possible explanation for this morality difference is that sounds, but not lights, have a strong automatic phase resetting influence on an internal timing process, or clock, that underlies rhythmic perception and production. Experimental evidence supporting this hypothesis will be presented. Phase-dependent phase resetting, as found in circadian systems, neural pattern generators, and electronic phase-locked loops, ensures entrainment of an endogenous rhythmic process to an exogenous one, and resetting of a central timer by feedback from movements it controls reduces their variability. The operation of such automatic resetting processes in human rhythmic performance suggests explanations for a great many facts about time perception, temporal numerosity judgments, auditory feedback, and other phenomena. The possible relevance of some of these to speech and music will be discussed.
Effect of loudness and timing variation on rhythm perception
Authors: Norris Mary Ann , Rosenthal David
Published: October 2005

A rhythmically ambiguous musical passage, interpretable as being in either duple or triple time, was constructed. The loudness and timing of notes in the passage were varied in order to systematically explore the space of expressive variation of these parameters that would be used by a performer to induce a particular rhythmic interpretation. The resulting stimuli were played to musically trained experimental subjects, who were then asked to make a forced choice between duple and triple rhythmic interpretations. The subjects' responses were statistically analyzed and compared with a hypothetical model of their interpretive processes. It was found that, while loudness variation affected subjects' responses in a manner consistent with previous results and with the hypothetical model, timing variation, surprisingly, has no obvious effect on the subjects' responses. These results have implications for machine interpretation of rhythm, as well as for an understanding of human rhythm processing.
Signal duration required to distinguish the tone chroma, tone height, and timbre of a sound
Authors: Robinson Ken , Patterson Roy
Published: October 2005

Current auditory models suggest that pitch is a primitive feature extracted early on and then used to assist processing of timbre information in music and speech. To test this hypothesis, the number of cycles of a sound required to identify the tone chroma (note name), tone height (octave), or timbre (instrument) of a musical note was determined. There were four instruments (brass, flute, harpsichord, and strings), four octaves centered on C1, C2, C3, and C4 and four notes, C, D, E, and F. The 64 stimuli were presented randomly at a variety of durations, and, in a given session, the listener identified the stimulus within one perceptual category: tone chroma, tone height, or timbre. The results show that performance rises with duration to an asymptotic level. Performance rises fastest to the highest asymptote in the case of instrument identification, and slowest to the lowest asymptote for tone chroma. This suggests that timbre processing can proceed before detailed information about the pitch is available. [Research supported by UK RAE 2239.]
Melodic character versus voice position in the perception of a two-voice musical structure
Authors: Baczewski Philip , Killam Rosemary N.
Published: October 2005

This study examines whether melodic character or voice position is a more influential factor in professional musicians' perception of a two-voice musical structure. A previous study [P. Baczewski and R. N. Killam, J. Acoust. Soc. Am. Suppl. 1 86, S17 (1989)] examined 15 highly trained listeners' response to a stimulus consisting of a 16-measure piece of music. They were presented with a taped performance (prepared by professional musicians) of three repetitions of a Mozart duet, together with the experiment's instructions. Experimental design allowed measurement of response to each stimulus repetition. A difference was noted in the perceptual strategies for notation of top and bottom lines of the duet. This current study examines a repetition of the above experiment in which the positions of the lines are reversed (transposition of the bottom line into the range of the top and vice versa). This procedure yields insight into whether melodic character was responsible for the lesser degree of total correct response for the lower line of the previous experiment, or whether lower lines are inherently less well perceived and notated by musically trained subjects. [Work supported in part by an Apple Education Foundation Grant.]
Monaural ear differences for melody recall
Authors: Perry David W.
Published: October 2005

Recall of monaurally presented diatonic melodies by experienced pianists was recorded with a computer-interfaced keyboard. Responses were analyzed by computer in terms of pitch accuracy and in terms of the timing of keystrokes (response latency, keyboard rehearsal time, final recall speed, and temporal segregation). For right-handed pianists (N = 10), left-ear presentation resulted in higher pitch recall accuracy (p < 0.001) and longer response latency (p < 0.005) for the first presentation. Individual ear asymmetries (EAs) for pitch accuracy and response latency were also positively correlated with each other (r = 0.70). EAs for the second presentation were not significant. None of the other performance measures was significantly asymmetric. These results agree with those of Zatorre [Neuropsychologia 17, 607–617 (1979)] for melody recognition, and indicate that perception of novel pitch sequences by musically experienced subjects, under conditions demanding attention to the entire melody, consistently enlists specialized processors in the nonlanguage dominant hemisphere. The positive correlation between EAs for response latency and pitch accuracy, together with subject reports, suggests that covert singing of the melody may be characteristic of the strategy that preferentially enlists right hemisphere processors.
Effects of tone sets and serial orders on melodic tracking: Dependence or independence?
Authors: Cohen Annabel J. , Frankland Bradley , Lamoureux Michael , Dunphy Debora
Published: October 2005

Forty listeners, differing in level of musical experience, indicated the serial order of eight tones in 32 sequences representing all combinations of four tone sets or scales (major, minor, chromatic, and three-semitone spacing) and eight serial orders. Responses were produced with a mouse, which controlled the vertical height (representing pitch height) of a series of eight sliders (representing the eight temporal positions in the sequence) on a computer screen. Response accuracy was higher for tone sets with wider frequency ranges, for serial orders with simpler contours, for sequences in which the tone set did not vary from trial to trial and for listeners with musical training. An interaction between tone set and serial order indicated their interdependent contributions to melodic tracking performance. This interaction was previously observed using a less refined technique [Cohen and Frankland, Canadian Acoustics (submitted)] and the results are here extended to more rapid sequences, to complementary response-time information and to specific serial position analysis of relative and absolute errors. [Work supported by NSERC.]
Expectancies generated by tonal contour
Authors: Frankland Bradley , Cohen Annabel J.
Published: October 2005

Expectancies generated by tonal contour were examined in a two-part study. Listeners (N = 61) were presented with eight-note sequences followed by a probe tone. Subjects rated the fit of the probe to the sequence in a two-level (good or poor), forced-choice decision. For each probe, the proportion of good-fit replies provided a rating of its expectancy. In part 1, ascending and descending scales provided a key-defining context [Krumhansl and Kessler, Psychol. Rev. 89, 334–368 (1982)]. Cluster analyses on individual probe-tone profiles for all 13×2 contexts classified subjects into 12 groups: 5 were clearly identified as triadic (–n = 11), diatonic (n = 17, n = 3), or proximity (n = 3, n = 3). In part 2, 12 sequences representing degrees of higher-order sequential expectations provided eight-note contexts for the probe-tone task. For the triadic and the large diatonic groups, profiles from the 12 melodies when compared against the baseline profiles indicated effects of hierarchical contour. Thus sensitivity to higher-order contour structure influences expectations for particular tones. [Work supported by NSERC.]
Organization and discrimination of repeating sound sequences by newborn infants
Authors: McAdams Stephen , Bertoncini Josiane , Bobrow Jacqueline
Published: October 2005

A series of three experiments investigated the ability of 3- to 4-day old babies to discriminate repeating, rising and falling four-tone sequences. in different experimental conditions the sequences varied in tempo (five or ten tones/s) and pitch interval (two semitones: C3, D3, E3, F3, or five semitones: E2, A2, D3, G3). Two configurations of instrumental timbre and spatial position of the four tones were also used, with two instruments (vibraphone and trumpet) being presented in separate speakers on opposite sides of the head: (a) The three lowest pitches were played by one instrument and the highest was played by a different instrument (3/1); (b) pitches 1 and 3 in the sequence were played by one instrument and pitches 2 and 4 were played by the other (2/2). When reversed, configuration 3/1 is discriminable and 2/2 is not discriminable from the original sequence by adults, due to organization on the basis of spatial position and timbre. Discrimination by babies was measured with an habituation-dishabituation paradigm based on infants' sucking behavior. At the criterion habituation point the sequence was reversed for experimental babies and remained the same for controls. Experiment 1 used the fast tempo and small pitch interval sequences in the 3/1 configuration. Infants were unable to discriminate the melody from its reversal. Experiment 2 used the slow tempo and large pitch interval sequences in the 3/1 configuration. Experimental subjects were able to discriminate the melodic contours. Experiment 3 used the same rate and interval as experiment 2, but used the 2/2 configuration. Here, the infants were unable to discriminate the melody change indicating that they organize the pitch sequences according to timbral and/or spatial position information. The results are discussed in terms of musical and speech sequence processing in newborns. [Work supported by a Fyssen Foundation grant to S. McAdams.]
Evaluation of the NeXT computer system for auditory research
Authors: Lamoureux Michael , Cohen Annabel J.
Published: October 2005

As a standard feature, the NeXT computer is equipped with 16-bit synthesis capabilities provided by a Motorola DSP56001, a 20-MHz digital signal-processing (DSP) chip. Coupled with other features of speed, storage capacity, Unix-based operating system, and graphics, the NeXT should be an ideal choice for supporting a variety of auditory experiments. To examine this possiblity, empirically obtained specifications (signal-to-noise ratio, total harmonic distortion of sine tones, frequency and power accuracy, spectral bandwidth, and stereo separation) of the digital audio hardware will be reported. An evaluation of the NeXT's digital signal processing hardware and its associated software toolkits will also be provided, including measurements of the quality of the algorithmically generated signals. The suitability of the NeXT computer for stimulus generation and response measurement in psychoacoustics, auditory and music research will then be discussed and examples of various experiments that use the NeXT will be briefly demonstrated. It will be concluded that with minor exceptions, the NeXT computer system does meet its expected potential. [Work supported by NSERC.]
Basin-scale tomography: A new tool for studying weather and climate
Authors: Spiesberger John L. , Metzger Kurt
Published: October 2005

Measurements of acoustic travel time across the northeast Pacific Ocean are used to detect large-scale (>500 km) thermal trends in the ocean interior. The trends are novel observations, and there is probably no method to verify them with other data sets. The results demonstrate the practicality of a new and powerful means to observe ocean temperature and to investigate its relation to weather, climate, and greenhouse warming.
Time-domain analysis of propagation codes over megameter ranges
Authors: Boden Linda , Bowlin Jim , Spiesberger John L.
Published: October 2005

From 1983–1989, tomographic experiments were conducted over megameter ranges in the northeast Pacific Ocean. Because the arrival time of acoustic pulses is dependent upon oceanic thermal structure, the data can be used to map changes in the heat content of the basin over time. These changes can be determined accurately only if the paths along which the energy travels can be identified; the errors in the propagation model must be known. In order to assess arrival time errors over megameter ranges, three codes are compared: the Kraken normal-mode (NM) program, the Navy standard parabolic equation (PE) model, and the MPP ray code. The analysis is first done in simple range-independent environments in which the normal-mode solution is theoretically exact in the far field. The arrival sequences of all three codes are compared for center frequencies of 50, 100, and 200 Hz and ranges of 1, 2, 3, and 4 Mm. The arrival patterns of a coupled NM-PE model and the MPP ray code are then compared to experimental data in a range-dependent environment.
Phase and impedance dependence of gains and losses in a thermoacoustic device
Authors: Ceperley Peter
Published: October 2005

Thermoacoustic devices can convert thermal energy into acoustic energy and can use acoustic energy to pump thermal energy. This paper will examine the energy conversion process of these devices, its gains and losses, as a function of the amplitude and phase of the pressure component relative to the velocity component of the acoustic field. Experimental data will also be presented. [Work supported by the Office of Naval Research.]
Interaural correlation discrimination and autocorrelation
Authors: Pollack Irwin
Published: October 2005

The discrimination of differences in interaural correlation was examined for auditory noise sources differing internally in terms of their autocorrelation. Interaural correlation discrimination is inversely related to the autocorrelation of the sound source: lowest interaural thresholds with independent random noise sources; highest thresholds with repeated periodic noise sources. Interaural thresholds increase with higher periodicities. Periodicity leads to higher thresholds, apparently, by reducing the number of independent samples for the calculation of interaural correlation.
Lateralization mechanisms and Deutsch's octave illusion
Authors: Divenyi Pierre L.
Published: October 2005

The octave illusion described by D. Deutsch [Hearing Research and Theory, edited by J. V. Tobias and E. D. Schubert. Vol. 1. pp. 100–142 (1981)] is a phenomenon observed when two continuous sinusoidal tones bearing an octave relation (e.g., 400 and 800 Hz) are presented dichotically such that the ear/frequency relationship is switched at regular intervals (every 250 ms). The illusion consists of the impression, reported by a majority of listeners, that the tone with the higher frequency in one ear (usually the right) alternates with the low frequency in the other ear. The present paper attempts to account for the mislocalization component of the illusion, based on two recognized properties of the binaural system: (1) In order for a stimulus to be lateralized, the dichotic information must be in the same band(s) [e.g., L. A. Jeffress, Foundations of Modern Auditory Theory, edited by J. V. Tobias (Academic, New York, 1972), Vol. 2, pp. 349–368]; and continuous pure tones may be mislocalized based on information not deriving from their onset [i.e., the Franssen effect, W. M. Hartmann and B. Rakerd, J. Acoust. Soc. Am. 86, 1366–1373 (1989)]. It will be shown that these two properties, together with some basic peripheral processing characteristics, can lead to lateral shifts in the dichotic image, consistent with the illusion. [Work supported by the Department of Veterans Affairs.]
Sound localization cues in the cat
Authors: Hind J. E. , Musicant A. D. , Chan J. C. K.
Published: October 2005

The changes in spectral properties of the cat free-field to eardrum pressure transfer function that occur with variation in the direction of a sound source have been described previously [J. A.c. oust. Soc. Am. 87, 757–781 (1990)]. In this paper, new measurements of direction-dependent interaural time differences defined in terms of the onset or group delay to an impulsive signal are reported. Considering the geometry of the cat head and ears, the results agree surprisingly well with the classic Woodworth model, which estimates the delay between two points on the surface of a sphere by calculating the time required for the incident wave to travel around the surface of the sphere. On the basis of human psychoacoustic findings and microelectrode studies of the cat auditory system, there is speculation as to how the observed spectral and temporal attributes of a free-field sound stimulus might be utilized in such discriminations as the determination of the minimum audible angle in both the horizontal and midsagittal planes as well as in the resolution of front-back confusion. [Work supported by NIH Grants NS12732 and NS07026.]
The integration of multiple cues in localization tasks
Authors: Zakarauskas Pierre , Cynader Max S.
Published: October 2005

Estimation theory is applied to the problem of combining multiple cues in acoustic source localization tasks. The process of localization is decomposed into the sub-tasks of cue estimation, likelihood function composition, combining the likelihood functions from different cues, and the estimation process itself. Calculation of the likelihood functions for a given cue is related to neural mechanisms. Several ways of combining information and making a localization decision are enumerated and compared. Localization performance is predicted for individual and combined cues based on the head-related transfer function of the subject. Finally, neurophysiological and psychophysical results are used to estimate some parameters of the theory, and experiments are proposed to estimate others.
Developments in using the boundary element method (BEM) to investigate the dynamic characteristics for foam earplug-earcanal and earmuff-earcanal systems
Authors: Royster Larry H. , Mourad Kassem , Xie Ke Jun , Ciskowski Robert
Published: October 2005

The boundary element method (BEM) has been utilized to formulate a three-dimensional math model for the foam earplug-earcanal and earmuff-earcanal systems. The BEM technique consists of transforming the partial differential equation describing the behavior of the unknowns, inside and on the boundary of the domain, into an integral equation relating only boundary values. In representing the viscoelastic properties of the foam earplug, both integer differential operator and fractional operator constitutive equations were utilized. The BEM model was then utilized to study the steady-state and transient responses (including the prediction of each system's insertion loss) for different earplug-earcanal and earmuff-earcanal configurations. The basic BEM model has demonstrated the ability to predict the internal resonances for a production earmuff shell-earcanal system and the measured insertion loss of a foam earplug-earcanal system subjected to an impulse. Together with these findings a general discussion of the need for improvements in the ability to model a variety of earplug- and earmuff-earcanal configurations will be presented.
Gas bubbles in ocean sediments and high-frequency acoustic backscattering strength
Authors: Chotiros Nicholas P. , Boyle Frank A.
Published: October 2005

The influence of trapped gas bubbles on the high-frequency acoustic bottom backscattering strength of ocean sediments will be examined. Statistical trends in the extant backscatter database at shallow grazing angles suggest that gas bubbles may be a dominant factor. The effect may be modeled in terms of an increase in the volume scattering strength of the sediment. Results from model and laboratory studies will be presented. [Work supported by ONR under NOARL management.]
Subjective and objective evaluations of rooms for music
Authors: Cervone Richard P. , Chiang Wei-hwa , Siebein Gary W. , Doddington Harold W. , Schwab Wilhelm K.
Published: October 2005

A dedicated, computer-based analysis system was developed to perform a complete set of acoustical measurements of recent interest in full-size rooms and in scale models of rooms. The measurements included reverberation time, early reverberation time, loudness, early to late temporal energy ratios, lateral energy fractions, interaural cross correlation, and speech transmission index among others. Measurements were made at multiple locations in ten large concert halls. Groups of listeners evaluated live music performances at three locations in each of the rooms using a seven point semantic differential rating scale. Correlation analysis and statistical modeling identified significant relationships among the qualities of the music in the room rated by the listeners with the physical measurements made in the rooms. Variations of subjective qualities were identified among the different rooms and within each of the rooms as well. The subjective qualities that contributed to overall acoustical impression were also identified. [Work supported by NSF.]
Using piezoelectric film and ultrasound resonance to measure the elastic moduli of spherical ceramic particles
Authors: Spoor P. S. , McKenna M. J. , Maynard J. D. , Hellmann John R.
Published: October 2005

The search for alternate sources of energy has prompted interest in small ceramic beads, called “proppants,” which were developed as a means of “propping” open cracks during the hydraulic fracturing of bedrock in the vicinity of oil wells; recently, they have been considered as possible thermal transfer media for use in solar receivers [J. R. Hellmann et al., “Evaluation of Spherical Ceramic Particles for Solar Thermal Transfer Media,” SAND86-0981, Sandia National Laboratories, January 1987]. To monitor the effects of repeated thermal stresses on the proppants, one would like to have a reliable measure of their elastic constants; however, their spherical shape and small size (≃ 500 μm) make conventional techniques, such as pulse-echo, inapplicable. Using a special piezoelectric film transducer and a small-sample resonance technique [J. D. Maynard, J. Acoust. Soc. Am. Suppl. 1 85, S20 (1989)], the authors have been able to determine the elastic constants and their variation as a function of heat treatment. [Work supported by the Office of Naval Research and NSF Grant DMR 9000549.]
Characterization of anisotropy in wood composites
Authors: Bucur V.
Published: October 2005

The aim of this report is to study the anisotropic behavior of wood composites using the ultrasonic velocity method and acoustic emission method. Velocities of ultrasonic longitudinal and transversal waves were used for the estimation of five elastic constants. Complementary, stimulated acoustic emission, induced by breaking 0.5-ram pencil lead on the surface of the specimen, was employed to measure five parameters of the acoustic emission signal (duration, counts number, energy, amplitude, and rise time). The anisotropy was estimated as the ratio of velocities, of acoustic invariants, and acoustic emission parameters.
Flow noise induced in large arrays via the flexure of the support
Authors: Dubus B. , Montgomery R. E.
Published: October 2005

The noise induced by the turbulent boundary layer in sonar arrays is usually split into a direct path where the fluctuating pressures directly excite the hydrophones after traveling through an elastomer layer backed by a rigid surface, and the indirect path where the excitation of the support by the turbulent boundary layer induces the noise in the array. To describe this last phenomena, the force-modal transform method [M. C. Junger and D. Feit, Sound, Structures, and Their Interaction (MIT Press, Cambridge, MA)] has been extended to describe the flexure of a submerged multilayered plate and the wave-number spectrum of the noise sensed by the array. Several configurations are analyzed showing the effect of the stiffness and the damping of the support and of the distribution of the hydrophones. From these results, some solutions to reduce the noise level are proposed. [Work partially supported by Direction des Recherches Etudes et Techniques, Paris.]
Finite element model for a planar fiber optic hydrophone
Authors: Zoccola Paul J. , Shang Paul C.
Published: October 2005

A finite element model has been developed for estimating the acoustic sensitivity of a planar fiber optic hydrophone. The sensor consists of a single-mode fiber surrounded by a thin layer of silicon and a hytrel jacket. This strand is coiled and embedded in an elastomer layer. The model will allow use of nonisotropic materials also; these materials are less suitable for analytical modeling. The finite element model uses linear solid elements and uses symmetry about two or three planes, depending on the configuration. Because the sensor is designed for low-frequency measurements, hydrostatic pressure loading is used. Sensitivity is calculated using the strain on the fiber. The results are compared to closed form solutions for cylindrical configurations and to experimental results. They show that for most practical purposes, sensitivity is mostly controlled by the volume of the elastomer surrounding the fiber.
Theoretical study of a fiberacoustic waveguide with continuous variation of the radial acoustic velocity
Authors: Yin A. G. , Varadan V. K. , Varadan V. V.
Published: October 2005

In this paper, a geometrical acoustics method has been presented to describe acoustic wave propagation in a fiberacoustic waveguide in which the acoustic velocity varies continuously in the radial direction. Based on the acoustic wave equation, a WKBJ approximate method has been used to analyze the propagation and cutoff characteristics of guided acoustic modes in this waveguide. It is believed that this acoustic fiber can transmit acoustic and optical signals at the same time when the refractive index of the core material is greater than that of the cladding material and the acoustic velocity of the core material is less than that of the cladding material. Furthermore, this type of acoustic fiber is suitable because of low loss and hence long distance transmission paths.