bluuebunny/crossref_metadata_2025 huggingface dataset:
Optimized constant beam width arrays
Authors: Harrell Jefferson A. , Hixson Elmer L.
Published: October 2005

It has been previously shown by the authors [JAES 34, 221 (April 1990); J. Acoust. Soc. Am. 91, 2326 (A) (1992)] that the main beamwidth of a loudspeaker array can be held constant over an octave by the superposition of two arrays by using suitable filter functions. The beam patterns are identical at the end (design) frequencies, however, deviations occur at frequencies between the extremes. The deviations are representative of unwanted acoustic energy if the deviation is positive when referenced to the desired pattern. While the array has some error at these design frequencies (for example, undesired side and backlobes), the total error energy integrated over the octave may be greater than that of other arrays that use element weights that allow more error at the design frequencies. Presented is an algorithm to minimize the deviations between design frequencies while minimizing the error energies, which is based on a Wiener adaptation. The results are compared to hand-generated guesses. The impact of linear and nonlinear cost functions is considered.
Speed of sound in pure water as a function of temperature
Authors: Bilaniuk Nykolai , Wong George S. K.
Published: October 2005

In view of the adoption of the International Temperature Scale of 1990 (ITS-90), which defines the International Celsius Temperatures, t90, the dependence on temperature of the speed of sound in pure water is examined. Drawing on the experimental data published previously [V. A. Del Grosso and C. W. Mader, ‘‘Speed of Sound in Pure Water,’’ J. Acoust. Soc. Am. 52, 1442–1446 (1972)], it is found that the change from the previous t68 scale is significant. At 100 °C, the difference between the two scales (t68−t90) is 0.026 °C, resulting in a difference of 0.022 m/s for the speed of sound. The speed of sound is fitted to a new fifth-order polynomial applicable over the t90 range 0–100 °C.
Auditory scene analysis: Theory and phenomena
Authors: Bregman Albert S.
Published: October 2005

Many natural listening environments face the auditory system with a jumble of overlapping sound-producing events, from which it must recover separate descriptions of the individual sounds (auditory scene analysis). Its most basic method is ‘‘primitive’’ auditory grouping, a pre-attentive process that analyzes the incoming signal into components and then uses certain acoustic relations among them to link them into sets that later-acting recognition processes will prefer to treat as coming from separate events. Components (or features) will be linked when they exhibit relationships to one another that would be very improbable had they actually come from unrelated events. The evidence for grouping, derived from different relations, is allowed to compete and collaborate, in the linking of subsets of components or features. In a mixture, the listener tends to hear these subsets as distinct but overlapping sound sequences, each with its own properties. The perceived qualities that can be affected by this organizing of data include the melodic and rhythmic qualities of the perceived sounds, their pitches, timbres, numerosity, temporal order, and even their perceived spatial positions.
Perceptual organization of speech signals: Clues from studies of sinewave replicas of utterances
Authors: Remez Robert E. , Pardo Jennifer S. , Rubin Philip E.
Published: October 2005

The perceptual organization of speech is a challenge for accounts proposing a two-stage model in which primitive auditory grouping is followed by schematic secondary processes. This difficulty stems from the diversity of acoustic constituents of speech signals, which is attributable in turn to the mechanics of vocal sound production. Primitive mechanisms that rely on similarity principles to group like elements therefore fail to parse the sound stream isomorphically to the sound sources. The insufficiency of primitive grouping mechanisms for organizing speech signals is often admitted, and knowledge-based latterly occurring processes are often invoked to explain the self-evident perceptual coherence of utterances. Studies with sinewave replicas of words and sentences show that phonetic perception occurs in conditions that thwart the application of primitive grouping mechanisms as they are presently understood. Psychophysical tests with sinewave words suggest that the perceptual organization of speech is distinct from other auditory organization. Because classic evidence and arguments discourage schema-based conceptualizations of phonetic processes, the two- stage model of organization appears to be implausible for speech. Perceptual organization of speech can better be rationalized by a phonetically specific facility that organizes and analyzes utterances. [Research supported by NIDCD and NICHD.]
The case for uniformity in the computer storage of room impulse response binary files
Authors: Campbell Richard H.
Published: October 2005

The proliferation of computer-based acoustical data acquisition instruments, which store room impulse responses on magnetic media, suggests that some uniformity and interchangeability of these data would be beneficial. For example, a central, accessible public-domain library of room impulse responses would allow practitioners, researchers, and educators the opportunity of applying their respective post-processing techniques to any of these uniformly stored data from venues around the world. Suggestions are given for headers, data structures, and trailers based on current practice for two systems. An example of a data translation exercise from Ram-Soft-II to MLSSA is used to emphasize the need for a standard.
Articulatory underspecification as a concept for controlling a speech synthesizer
Authors: Kröger Bernd J. , Heike Georg , Opgen-Rhein Claudia
Published: October 2005

A segmental model for generating articulator movements based on the principle of articulatory underspecification has been developed. This model is part of a computer-implemented speech production model controlling an articulatory speech synthesizer and capable of producing any utterance [B. J. Kröger, in SignalProcessingVI: TheoriesandApplications, edited by J. Vandewalle etal. (Elsevier, New York, 1992), pp. 331–334]. Articulatory/phonatory control parameter time functions generated by this segmental model are compared to those generated by a gestural model. Qualitative comparison shows that the concept of segmental underspecification and the concept of cooccurrence (or overlap) of articulatory gestures result in similar articulator movements. [Work supported in part by DFG Grant No. He 434/21-1.]
Measurements of low-frequency, low-grazing-angle sea-surface scattering and feature strengths
Authors: Gauss Roger C. , Fialkowski Joseph M.
Published: October 2005

Direct-path measurements of low-frequency (200–1000 Hz) and low-grazing-angle (≤10 deg) acoustic surface scattering were made in the Gulf of Alaska in April of 1990 and in March of 1992. Both narrow-band (cw) and broadband (FM,PRN) transmissions were used to examine the strength of sea-surface backscatter as a function of frequency and environmental conditions. Results have revealed that in addition to a continuously distributed reverberation component, there are often strong spatially and temporally discrete scattering events. Scattering and feature strengths will be presented to quantify the relative contributions of these components to mean backscatter levels (such as were used in deriving the Ogden–Erskine scattering-strength curves), and to extend the scattering-strength results of Ogden and Erskine to lower grazing angles. Furthermore, the presented results have strong implications as to the types of bubble clouds that are responsible for each component (as will be discussed).
Measurements of the spectral characteristics of low-frequency, low-grazing-angle surface reverberation
Authors: Gauss Roger C. , Fialkowski Joseph M.
Published: October 2005

Direct-path measurements of low-frequency (200–1000 Hz) and low-grazing-angle (≤10 deg) acoustic surface scattering were made in the Gulf of Alaska in April of 1990 and in March of 1992. Short-duration (0.6–2.4 s) cw and PRN waveforms were used to quantify the spectral character of surface/near-surface reverberation as a function of frequency and environmental conditions. Measures include spreading and peak-Doppler-shift statistics versus azimuth. Results over the range of wind speeds (10–35 kns) have revealed a dominant zero-Doppler component and a weaker-than-expected dependence of spread on both sea conditions (wind speed and relative direction of the seas) and frequency. The results are consistent with sub-surface bubbles as the primary mechanism for surface reverberation when white caps are present, and give insight as to the nature of the bubble clouds that could give rise to the observed acoustic scattering.
Analysis of low-frequency, broadband measured reverberation levels at Blake Escarpment
Authors: Kalra A. K. , Fulford J. K. , Showalter J. A.
Published: October 2005

Acoustic reverberation data were acquired at the Blake Escarpment using Naval Research Laboratory’s (NRL’s) deep towed acoustic/geophysics system (DTAGS). DTAGS is similar to a seismic acquisition streamer system with its own omnidirectional source (LFM, from 250 to 650 Hz) but consists of two collinear receiver arrays (an acoustic and a geophysics) and can be towed in deep waters (to 5000 m). Although DTAGS acquires acoustic and geophysics data simultaneously, only the acoustic array data are analyzed here. The acoustic (field) data consist of backscattered reverberations and/or specular reflection events from all angles. These data are first transformed into p-t (ray-parameter versus time-intercept) domain and studied to retain the grazing angles (intermediate and high) of interest. These selected angles of interest are further analyzed to determine the reverberations versus frequency relationships, and also transformed back to space-time domain to determine the locations of the scattering patches that produced the reverberations. [Work supported by ONR, ARSRP Program.]
Application of numerical grid generation to scattering from rough penetrable ocean bottoms
Authors: Gilbert Kenneth E. , Kulbago Timothy J. , Di Xiao
Published: October 2005

Numerical grid generation methods have recently been applied with success to atmospheric sound propagation over irregular terrain [Di etal., J. Acoust. Soc. Am 92, 2431 (A) (1992)]. Here, the method was applied to scattering from rough ocean bottoms that allow sound to penetrate into the sediment. Considered are some simple geometries for which analytic results exist as well as more complex geometries that are treated numerically. Of interest are the relative contributions from surface scattering and volume scattering to the total acoustic backscatter. The ultimate objective of the investigation is to arrive at a simple, physically intuitive formulation that will aid in the interpretation of backscatter data from complex coastal environments. [Work supported by ONR.]
Scattering of superpositions of localized waves from spheres
Authors: Power Des , Donnelly Rod
Published: October 2005

Localized wave (LW) solutions of the homogeneous wave equation can represent either focused or extended pulses, depending on the choice of a free parameter present in the solution. One can form finite energy superpositions, with respect to this parameter, and one such superposition, known as a modifiedpowerspectrum (MPS) pulse, has been extensively studied [R. W. Ziolkowski, Phys. Rev. A 39, 2005–2032 (1989)]. In particular, the enhanced near-field LW propagation effect has been demonstrated by others, in the context of multi-time-derivative transmit/receive systems [R. W. Ziolkowski and D. K. Lewis, J. Appl. Phys. 68, 6083–6086 (1990)]. The LW solutions are broadband, so it is perhaps to be expected that one could use this property, along with the enhanced focusing characteristic, to extract target parameters when scattering LW superpositions from objects. The acoustic MPS pulse backscattered spectra from a number of different sized steel and aluminum spheres will be presented. It will be demonstrated that these spectra can be used to extract target size information. [Work supported by NSERC.]
Scattering from a rigid target in shallow water
Authors: Sarkissian Angie
Published: October 2005

An algorithm is developed to compute the field scattered from a rigid target placed in shallow water. The method of superposition is applied where the scattered field is approximated to be equivalent to the field produced by a large number of point sources placed on a surface inside the target. The source strengths are determined by minimizing the least-squares error of the normal velocity on the surface of the target. This algorithm is applicable even when the source or the receiver is close to the target as well as in the case where the target is close to the surface. Scattering computations are made for a cylinder with hemispherical endcaps in an isovelocity fluid with soft boundary conditions on top and rigid boundary conditions on the bottom. They are compared to the scattered field computed by using an algorithm discussed by Ingenito [F. Ingenito, J. Acoust. Soc. Am. 82, 2051–2059 (1987)] for the case when both the source and the receiver are distant from the target and when the target is distant from the surface. Results from both methods agree in the range of applicability of Ingenito’s method.
Effects of ambient noise on the scattering from cylinders and spheres
Authors: Honeycutt Rebecca L. , Johnson Steven J.
Published: October 2005

The sound scattering due to an ambient noise field, approximated by a squared cosine function, is considered for infinite rigid and elastic cylinders and rigid spheres. For the cylinders, it is assumed that the acoustic wave front is parallel to the axis of the cylinder (normally incident). For this assumption, a closed-form expression for the scattered sound field-to-incident ambient noise field (signal-to-noise) ratio is obtained not only for the cosine squared directivity, but for any arbitrary directivity that can be expressed in terms of a Fourier series. For the sphere, it is assumed that the noise is circumferentially symmetric, which leads to a closed-form expression for the signal-to-noise ratio due to a cosine-squared directivity.
Phonetic observations on tone and stress in Mandarin Chinese
Authors: Moore Corinne B.
Published: October 2005

This study reports results of an experiment on Beijing Mandarin tones and stress. There are four lexical tones in Mandarin, plus a pitch pattern referred to as the ‘‘neutral tone,’’ which occurs on atonic or unstressed syllables. The literature has been vague, however, about whether syllables that have undergone tone deletion as a result of destressing and underlyingly atonic syllables show the same surface pitch pattern. This experiment examined these lexical tones in stressed and unstressed contexts, and then compared unstressed tone patterns to that of the neutral tone. The experiment focused on duration and pitch range as effective correlates of stress. Unstressed syllables were found to be significantly shorter in duration than stressed syllables, corroborating previous studies. Pitch range was also significantly reduced in unstressed contexts in all cases except one. The data unexpectedly showed that changes in pitch range are a function of the pitch peak only; pitch valleys did not significantly change. Finally, the results provide evidence that unstressed syllables with underlying tones do not neutralize completely, but remain distinct from the neutral tone.
VOT values of voiceless and voiced stop contrasts in Hindi and Kannada
Authors: Gopal H. S.
Published: October 2005

A number of languages from the subcontinent of India have a four-way voicing contrast—the aspirated voiceless, the unaspirated voiceless, the unaspirated voiced, and the breathy voiced stops (for example, [ph, p, b, bh]). It has been suggested that the primary difference between the unaspirated voiceless and the unaspirated voiced stop consonants (for example, [p] vs [b]) is one of voice onset time (VOT). This study investigated whether VOT adequately separates the unaspirated voiceless and the unaspirated voiced stop consonants in two languages, Hindi and Kannada. Hindi is primarily a north Indian language and Kannada is primarily a South Indian language. Using four speakers, VOT values were measured for these two stop consonant series at four places of articulation—bilabial, dental, retroflex, and velar. The results will be discussed with reference to the adequacy of VOT as the primary acoustic feature distinguishing these two stop contrasts in the two languages.
Segmental duration changes due to variations in stress, vowel, place of articulation, and voicing of stop consonants in Greek
Authors: Kollia H. B.
Published: October 2005

Although Lisker and Abramson (1967) found no effect of the following vowel on the VOT of a stop consonant, Port and Rotunno (1979) found VOT to have greater values for voiceless stops followed by tense than by lax vowels. The purpose of the present study was to obtain a complete database on the VOT characteristics of voiced and voiceless initial stop consonants in Greek, and to examine the contextual effects on the VOT of the post-consonantal vowel, the stress pattern, and the distance of the stress from the initial stop consonant. The question here was whether the vowel effects found by Port and Rotunno for English would be seen in Greek, a language whose two stop categories have voicing lead and medium lag. Speakers read isolated disyllabic and trisyllabic words of four stress patterns. The utterance-initial stops /p, t, k, b, d, g/ were followed by the five vowels of Greek, /ɑ , eh, i, o, u/. Results indicated that both voicing lead and voicing lag increased for stops followed by higher than by lower vowels. Fourakis (1986) found that stress changes did not affect VOT of /p,t/, only of /k/. In this study, stress variations affected VOT for all stops. Furthermore, these effects were vowel dependent. [Work supported by NIH Grant Nos. DC-00121 and DC-00594 to Haskins Laboratories.]
Do women listen to men’s voices with men’s mouths in mind?
Authors: Johnson Keith , Ladefoged Peter , McDonough Joyce
Published: October 2005

This paper reports the results of an acoustic study of vowels produced by Navajo speakers, which was supplemented by perception data using the method of adjustment. Analyses of vowels produced by male and female speakers showed these two groups distinguished long and short vowels differently. Male speakers had about the same formant values for both members of the long/short vowel pairs, while female speakers tended to distinguish long and short vowels by means of both a length and a vowel quality distinction. The perception data showed that when listening to a ‘‘male’’ synthetic voice all listeners (irrespective of sex) collapsed long and short vowels in the perceptual vowel space, selecting vowels that were similar to those in the male production data, but with more extreme formant values than those produced by the male speakers. One hypothesis is that Navajo women produce different long and short vowels, but when listening to a male speaker’s vowels select as matches long and short vowels that do not differ significantly in quality. It is not known whether the men would listen to women’s voices with women’s mouths in mind.
Spectral shape cues for vowels that predict the perception of multiple-tone steady-state stimuli
Authors: Zahorian Stephen A. , Zhang Zhong-Jiang , Nossair Zaki B.
Published: October 2005

In previous experiments for which multiple tone stimuli were synthesized such that either the formants or global spectral shape were matched to that of naturally spoken vowel tokens, it was found for both cases that vowel identity and quality was not well preserved [S. A. Zahorian and Z.-J. Zhong, J. Acoust. Soc. Am. 92, 2414–2415 (1992)]. In the present study several additional criteria were tested for selecting the amplitudes and frequencies of sinusoids with the objective that stimuli synthesized from these sinusoids would be perceived as most similar to original ‘‘target’’ vowel tokens. Of the methods investigated, vowel quality from stimuli synthesized from N sinusoids was best preserved if these sinusoids match the N largest peaks in the magnitude spectrum of the original vowels. Depending on the vowel, between 5 and 10 sinusoids are required such that the synthesized token is perceived as sounding nearly identical to the original token. A new metric for spectral shape that yields acoustically invariant cues to vowel perception in a manner consistent with the results of these experiments has been developed and will be presented. This new metric also predicts the number of sinusoids that are required to synthesize each vowel.
Modeling of precedence-effect behavior in single neurons and in human listeners
Authors: Colburn H. Steven , Ibrahim Hany
Published: October 2005

Many of the physiological observations of Yin (reported in this session) are consistent with the precedence effect observed in human psychophysical data. In an attempt to understand the mechanisms involved in the generation of these observations, models for the observed neural interactions are being investigated. Since precedence-effect phenomena are fundamentally related to transient parts of stimuli, the models of these phenomena incorporate descriptions of responses to transient stimuli. In contrast to models for the observed binaural interaction in the medial superior olive (MSO) for sinusoidal stimuli, models for responses to click stimuli in the inferior colliculus (IC) must include inhibitory effects. These effects are long-lasting (tens of milliseconds) and asymmetric, consistent with earlier data from the IC with binaural clicks [L. H. Carney and T. C. T. Yin, J. Neurophys. 62, 144–161 (1989)]. The location of this inhibitory interaction is not clear since physiological measurements in lower levels of the brainstem (e.g., MSO) have not included the appropriate transient stimuli. The physiological responses from the IC will also be discussed in the context of recent psychophysical observations which suggest that the precedence effect may be a very central phenomenon. [Work supported by NIDCD.]
Neurophysiological correlates of the pitch of complex tones
Authors: Delgutte Bertrand , Cariani Peter A. , Tramo Mark J.
Published: October 2005

The pitch of complex tones plays an important role not only in speech and music, but also for perceptual segregation of sound sources. In order to study physiological correlates of these phenomena, autocorrelation histograms were recorded and computed from auditory-nerve fibers in anesthetized cats to estimate the aggregate interspike interval distribution for the entire auditory nerve. Results for many types of complex stimuli (including those used in classical pitch shift and pitch dominance experiments) suggest that the perceived pitch corresponds to the most frequent interspike interval in the auditory nerve. Also investigated was how interspike interval information might be used to separate simultaneously presented vowels. Psychophysical experiments show that both constituents in a pair of concurrent vowels are better identified when they have slight different pitches. These results show that concurrent vowels can usually be segregated from the information in the aggregate interspike interval distribution, and that peformance improves when the vowels have different pitches even though the method used does not rely on explicit pitch identification. [Work supported by NIH Grant Nos. DC00119 and DC00038.]
Detecting changes in spectral shape of complex auditory signals: Profile analysis
Authors: Green David M.
Published: October 2005

This is a review of current theories and recent experimental data on how human listeners detect changes in the spectrum of a complex sound. The effects of parameters such as the number of components in the complex, their density, bandwidth, and whether the components are harmonic or equally spaced in logarithmic frequency will be described. Some new data on how signal duration affects the ability to detect spectral change will also be reviewed. It is believed that three distinct sets of cues mediate the detection of changes in the spectral shape depending on the total bandwidth of the complex. For wide bandwidths, the primary cue is a simultaneous comparison of level over different frequency regions. For spectra that are one or two critical bands in width, the primary cue is a change in pitch. For spectra less than one critical band, the primary cue is a change in the temporal fluctuation or roughness of the sound related to the power spectrum of the envelope.
Nonlinearity of spectra processing in the dorsal cochlear nucleus (DCN)
Authors: Young Eric D. , Nelken Israel , Spirou George A.
Published: October 2005

Most studies of the neural representation of complex stimuli assume a parallel filterband form of representation in which neural response is plotted against position on the basilar membrane (or best frequency, BF). Such a representation assumes that the neural response reflects the energy in the signal within the neuron’s tuning curve and makes most sense if neural tuning properties are linear or only weakly nonlinear. By contrast, the principal cells of the DCN integrate energy across their response areas in a highly nonlinear fashion. These neurons give predominantly inhibitory responses to narrow-band stimuli and excitatory responses to broadband stimuli. Their responses to broadband stimuli cannot be predicted by a filtering model, in which the response equals the sum of narrow-band responses, weighted by the stimulus spectrum. Similarly, a narrow notch in a noise band centered on a unit’s BF also gives an inhibitory response. This inhibitory response also cannot be predicted from narrow-band responses and may represent a specialization for detecting spectral notches produced by the directional filtering properties of the external ear. [Work supported by NIH, Keck Foundation, and Fulbright Foundation.]
Jaw velocity events as correlates for the control of quantity contrast in Arabic
Authors: Jomaa Mounir , Abry Christian
Published: October 2005

Jaw vertical position of VCC vs VVC quantity contrasts in Arabic uttered at two speech rates were monitored. Events detected are: jaw lowering onset (VCO), and maximum velocity (MVV), jaw raising onset (CCO), and maximum velocity (MCV). Results show that at conversational rate, contrasts depend on differences in absolute timing along only one cycle (VCO–VCO). At fast rate, relative timing values no longer carry quantity contrasts for two speakers (out of four) within this cycle, whereas the contrast is still maintained in the MVV–MVV cycle. Velocity profiles for the raising gesture show that speakers who lose contrasts in relative timing in the VCO–VCO cycle, are also speakers who lose them within this gesture at fast rate: in fact, they lose the absolute lead that geminates have over nongeminates in the acceleration phase of this gesture. Consequently, it appears that contrasts rely mainly on the timing of raising gestures that follow the vowel, thus hiding it faster when the consonant is a long one.
Proposal of a wavelet model for vocalic speech coarticulation
Authors: Lange Robert C.
Published: October 2005

A recent development in the theory of wavelets is wavelet system characterization [R. K. Young, WaveletTheoryanditsApplications (Kluwer, Boston, 1993)]. In wavelet system theory, the wavelet transform can be used to describe the time-frequency behavior of a transmission channel, by virtue of its ability to describe the time-frequency content of the system’s input and output signals. The present research proposes a wavelet-system model of speech production; wherein, the system is the process of transformation from one speechstate to another. The transmission channel generates an effected utterance from a control utterance. A speech production effect (such as voice quality, or the influence of the speaker) is thereby characterized in a time-frequency manner. The effect of CVC coarticulation is investigated in particular, using samples of real speech. In this case, an isolated vowel utterance (the control) is contrasted with an utterance of the same vowel imbedded within a /C–C/ context (the effected). The channel thus describes (in terms of wavelets) the CVC coarticulation. [Work supported by Appl. Res. Lab.]
Effects of frequency scale on inter-age comparisons of first and second formant transitions
Authors: Hodge Megan M. , Nearey Terrance M.
Published: October 2005

Comparison of spectral measures from a wide age range of talkers, to draw inferences about inter-age differences in articulatory behavior, is confounded by the absolute resonant frequency differences that result from vocal tract size differences. Normalization procedures are required so that spectral differences due to vocal tract size differences are not misinterpreted as age differences in speech production behavior. Based on vowel (steady-state) data, the log Hz scale has been favored over several other frequency scales that have been used to normalize spectra for analysis of production data [T. M. Nearey, Proc. Int. Conf. Spok. Lang. Process. 1, 583–586 (1992)]. The magnitudes of first and second formant transition extents of male talkers, who ranged from infancy to adulthood, were compared under three different normalization conditions (log Hz, ERB, and Bark scales). Each formant was measured in Hz at the onset and offset of the consonant–vowel transition in wideband spectrographic displays of the syllables /bɑi /, /bi/, and /dæ/. These values were transformed to each scale and formant transition extents and rates were determined. The results are discussed relative to the potential of these scales to lead to artifactual differences in inferred inter-age articulatory differences.
Resistance of bilabials /p, b/ to anticipatory labial and mandibular coarticulation from vowel types /i, a, u/
Authors: Sock Rudolph , Löfqvist Anders
Published: October 2005

The timing of lip and jaw movements and acoustic events was studied in one subject producing the six pseudowords /pipi,papa,pupu,bibi,baba,bubu/ at two speaking rates. Lip and jaw movements were recorded optoelectrically. Acoustic durations of bilabial closures served as a temporal base for analyzing consonantal resistance to vowel coarticulation. Within each temporal base, a phase was defined as the interval between peak raising or lowering velocity of an articulator and the acoustic onset of the following vowel. The proportion of time taken by each phase in the temporal base provides a measure of anticipatory coarticulation of the vowel in the consonant. Results suggest that the rounded vowel /u/ has a higher coarticulatory influence on the labial consonant that the unrounded vowels. The voiced consonant /b/ has less resistance to coarticulation than its unvoiced cognate. Both domains investigated show comparable phasing patterns, suggesting a general strategy of coarticulation for the successive consonants. Generally, all three articulators show similar gestural kinematic patterns. The results are discussed in terms of biomechanical constraints on speech production tightly coupled with the perceptual requirements of the linguistic code. [Work supported by the Fyssen Foundation and NIH.]
Effects of prosodic context on Taiwanese tones
Authors: Peng Shu-hui
Published: October 2005

Duration and F0 patterns for the high and high-falling tones of Taiwanese Amoy in different contexts were investigated in two experiments. The first examined the target tones in four prosodic positions: initial, medial, and final to a prosodic phrase, and utterance final. The nonutterance-final targets also varied in tonal context, being followed by high or mid tone. Words at the end of a prosodic phrase were substantially longer than in the other prosodic positions. The target tones were higher in frequency phrase-initially and phrase-medially than phrase- or utterance-finally. Tonal coarticulation occurred between the target tone and its following tone. When followed by high tone, the offset of the high target tone did not change, while that of the high-falling target tone increased. The second experiment investigated tonal coarticulation further by examining the high tone followed by five different tones in the middle and the end of a prosodic phrase. It is predicted that the tone height and the tone pattern of the target tone will vary depending on the following tone.
Interpreting complex articulations: The backward tongue movement in Ewe doubly articulated stops
Authors: Maddieson Ian , Perkell J. S. , Matthies M. L. , Svirsky Mario A.
Published: October 2005

In West African languages, /kp/, /gb/ are described as plosives with simultaneous bilabial and velar closures. From auditory and acoustic evidence Maddieson and Ladefoged (1989) argued that the two articulations are slightly offset in time, with the velar leading the labial. This claim has now been directly tested. Two Ewe speakers provided data using the EMMA system (Perkell etal., 1992). Temporal asymmetry of the two articulations was found, as well as broad similarities to gestures in simple velar and labial stops. In labial velars there is more backward movement of the tongue dorsum than in plain velars; in particular, the tongue back retracts during the closed phase of the stop before releasing forwards. This retraction might result from (a) movements to create suction in the front cavity, (b) increased air pressure in the front cavity pushing the tongue backwards, (c) greater fortition of the velar contact resulting in more compression and displacement of the tongue, or (d) larynx lowering, creating suction in the pharyngeal cavity and pulling the tongue dorsum backwards. Comparison between labial velars and plain velars with respect to the height reached by the tongue back, coordination between movements of the tongue front and back, and voicing features in the acoustic signal suggest that, in these data, pharyngeal cavity expansion is the most likely explanation for retraction.
Approximate evaluation of the spectral density integral for a large planar array of rectangular sensors excited by turbulent flow
Authors: Thompson William , Montgomery Robert E.
Published: October 2005

An approximate numerical procedure has been developed for rapidly evaluating the spectral density integral that predicts the output of a planar array of sensors excited by turbulent boundary layer pressure fluctuations. This procedure is particularly useful in cases where the transfer function factor of the integrand is not a simple function of the wave numbers in the flow and transverse directions. The procedure exploits the facts that the entire integrand is a separable function of these two wave numbers and, for large arrays, the array function factor of the integrand is a rapidly varying function of wave number, characterized by many similar-shaped lobes. In addition, a model for multilayered media is employed to provide the transfer function for boundary conditions that closely correspond to reality. Results generated by this procedure were compared to those from an exact evaluation of the integral that is possible if the transfer function is taken to be constant; there was agreement to within 0.2 dB or better over a broad frequency interval. Some results for a realistic transfer function are presented, such as the case of an elastomeric layer backed by an elastic plate with the sensors embedded at an arbitrary position within the layer. [Work supported by NAVSEASYSCOM 06UR.]
Fundamentals of outer decoupler design for flow noise reduction
Authors: Martin Nathan C.
Published: October 2005

Turbulent boundary-layer (TBL) wall-pressure fluctuations due to the flow of water over a sonar array produces significant levels of array self-noise. This flow noise background reduces the ability of the array to detect acoustic waves propagating from distant sources. These flow noise levels can be reduced by various methods including hydrophone area averaging and the separation of the array from the flow interface by structures generically referred to as outer decouplers. Examples of outer decouplers (O.D.) include sonar domes and elastomer-covered arrays. The fundamental objective of outer decoupler design is to provide a low pass filter in wave number k. Analytic modeling of the structural acoustic behavior of O.D. structures provides a basis for both understanding the wave-number filtering process and designing effective O.D.’s for specific applications. This paper reviews the basic structural acoustics of various O.D. classes to illustrate the differences in wave-number filtering performance at various frequencies and the application of these filter characteristics to array flow noise reduction. [Work supported by Office of Naval Research.]
Effect of various array structures on flow noise reduction
Authors: Ko Sung H. , Schloemer Howard H.
Published: October 2005

Turbulent boundary layer pressure fluctuations can be reduced by filtering of its wave-number response with either a finite hydrophone or a hydrophone array, or by filtering the wave-number response through a layer of elastomer. In general practice, hydrophone arrays are embedded within a layer of elastomer to reduce the turbulent boundary layer pressure fluctuations and then mounted to an elastic structure. To understand the role of the mounting structure, this work will examine the performance of idealized and realistic array mounting configurations. Three configurations are modeled in this paper. The first one is a plane elastomer layer backed by a rigid surface. The second one is backed by an elastic plate with a finite thickness. Finally, the third one is backed by a layer of water. The front side of the elastomer layer is exposed to turbulent flow. The transfer functions for the three different configurations will be discussed. The results presented here are the numerically calculated noise reductions relative to the calculated noise level for a flush-mounted point hydrophone. In addition, an equivalent-plane-wave flow noise level is derived and compared among the three configurations to directly show any array performance similarities or differences.
Interaction of TBL-induced flexural waves, element spacing, and steering on the self-noise of hydrophone arrays
Authors: Jameson Paul W. , Brown Richard N.
Published: October 2005

Most ship-mounted planar hydrophone array systems employ stiff structures for isolation from local noise. One type of structure is a plate (‘‘window’’), which isolates the hydrophones from high wave-number turbulence on the outer surface. Most arrays also use a heavy steel plate behind the array to achieve isolation from the inner surface. These plates support bending waves, which can be induced by TBL forces. Even though these waves are subsonic over the frequency region of interest, the noise stress can enter the array output through an aliasing lobe when the array is steered. This noise increase is frequency dependent, and increases with frequency after a critical value. This talk will indicate how this noise contaminates the array response and discuss the interplay of element spacing, steering, and frequency performance of the array.
A planar eight-element flexural disk fiber optic acoustic sensor
Authors: Brown D. A. , Netzorg G. B. , White B. , Kapolka D. , Gardner David L. , Garrett Steven L.
Published: October 2005

Interferometric fiber optic flexural disk hydrophones made of aluminum, castable epoxy, and composite materials have previously been constructed and reported [D. Kapolka etal., J. Acoust. Soc. Am. 91, 2325(A) (1992)]. These hydrophones consist of a closed cylinder with fiber coils bonded to the sides of the two end caps in a push–pull fashion in order to increase acoustic sensitivity and allow for cancelation of acceleration induced signals. It is possible to sum in series the phase modulations induced in multiple transducers (e.g., flexural disks) using the two legs of the interferometer. This will enhance the spatial averaging capability of the sensor due to its increased size. The sensor self-noise arising from flow-induced pressure fluctuations can then be reduced resulting in improved detectability. The fabrication and performance of a sensor comprising 8 flexural disks and 16 coils of optical fiber will be discussed. [Work supported by the Naval Sea Systems Command.] a)LT., USN.
Improved integration methods for poles in transfer functions
Authors: Zwillinger Daniel , W. Jameson Paul
Published: October 2005

When analyzing stress transmission in layered media as a function of in-plane wave number, the transfer function may exhibit pole-like behavior. These poles are associated with trace matching to bending waves in high stiffness layers. However, the exact interrelationship of multiple poles for the case of multiple various layers is not well understood. This condition presents difficulties to routines that integrate over the wave number. Hence, most users employ adaptive integration schemes that compensate for the poles by increasing the number of points at which the transfer function is evaluated. An improved approach, which involves singularity isolation, is being used in software being developed at BBN. This approach is to describe the pole in an analytical form and perform the integral analytically in the region in which the pole occurs. In this presentation, how to construct the analytic function for a single layer is illustrated, and the generalization to multiple layers is described. This approach leads to a reduction in computer processing time.
Application of wavelet transform to the perception of reproduced sound
Authors: Schuck Peter L.
Published: October 2005

Various time frequency distributions including spectrograms, Wigner distributions and smoothings thereof, and more recently scalograms, have been used to analyze speech. Very little work has been presented on attempts to use these methods to study sound reproduction, that is, loudspeaker/listening room systems. The loudspeaker/room system alters the reproduced sound mainly through resonances, delayed resonances, and reflections. The wavelet transform offers another potential avenue to visualize loudspeaker/room interaction with the highly desirable property that it is ‘‘constant Q’’ or constant percentage bandwidth as opposed to the other transforms that are constant bandwidth. Visual displays of the scalogram or squared modulus of the continuous wavelet transform are presented of resonances, delayed resonances, and echoes at, above, and below the threshold of audibility. In this way, the viability of the scalogram as a tool for the analysis of loudspeaker/room interactions is explored.
Wavelet-like transforms in the auditory system of the bat
Authors: Haresign Tim , Simmons James A. , Ferragamo Michael
Published: October 2005

Traditional views of neural processing of auditory signals in the nervous system have been shaped by conceptual models based on Fourier analysis—in the frequency domain for spectral selectivity or in the time domain for periodicity analysis. However, in recent years wavelet analysis has been proven to be a powerful new technique for signal processing. Evidence has been found for a wavelet-like representation of ultrasonic auditory signals in the inferior colliculus of the big brown bat, Eptesicusfuscus. The inferior colliculus is a laminar structure, with the neurons in each layer being sensitive to a specific, narrow frequency band. In response to a broadband stimulus the averaged activity of small groups of neurons exhibits a structured temporal pattern in its response. The temporal response pattern is dependent on the spectral-temporal structure of the input sound, and varies as a function of the frequency lamina from which the measurements are taken. Based on these results, it seems likely that information about ultrasonic signals is being segregated according to frequency and then transformed onto a different time scale within each frequency lamina as a wavelet-like burst, so that the time and phase relations within ultrasonic signals can be displayed by time and phase relations on a corresponding expanded physiological scale. [Work supported by ONR Grant No. N00014-89-J-3055, NIMH Grant Nos. MH-00521 and MH-19118, and NIH Grant No. DC-00511.]
Improvement of the spatial Hankel transform method to determine the impedance of outdoor ground surfaces
Authors: Verhaegen C. , Lauriks W. , Cops A.
Published: October 2005

The method of measuring the impedance of a ground surface based on the spatial Hankel transform, can be divided into three parts. First the pressure field is sampled on two horizontal lines near to the surface, from beneath a cylindrical symmetric sound source till infinity. Next a Hankel transform is performed on the data of each horizontal line. Finally the impedance of the surface for an incident angle ranging from 0 to 90 deg, is calculated. In practice, due to the finite maximum, measuring distance oscillations occur in the impedance curve as a function of angle of incidence. This effect can be avoided by using a spatial window before transforming the data. A Kaiser–Bessel window is selected to give the best results. A side effect of using a window is a shift from the real impedance values. Further improvement can be obtained by calculating more data points for a horizontal value larger than the maximum measuring distance. The method and its improvements will be illustrated with measurements on outdoor ground surfaces.
Comparison of STFT, Gabor, and wavelet transforms in transient vibration analysis of mechanical systems
Authors: Önsay Taner , Haddow Alan G.
Published: October 2005

Short-time Fourier (STFT), Gabor, and wavelet transforms are applied to the analysis of vibration signals obtained from mechanical systems. Nonstationary vibration signals resulting from impacts, friction, and transient dynamic interactions are studied by employing the transforms to demonstrate the time-evolution of their spectra. In the application of the STFT, the time-window length and the overlap in window translation are varied and their effect on the resolution of the time-frequency information is discussed. Due to its optimal time-frequency localization characteristics, the Gaussian window used in Gabor transform is found to be particularly useful in the analysis of time-decaying sinusoidal signals, whereas for vibration signals having sharp variations the logarithmic zooming properties of wavelets result in better localization in the time-frequency domain. The use of different analyzing wavelets are considered for detection of transient vibrations and characterization of the associated dynamic behaviors. Based on applications and examples of synthetic signals, the advantages and the limitations of each transform technique are discussed and their conceptual similarities are emphasized.
Wavelet analysis of rhythm
Authors: Todd Neil P. McAngus
Published: October 2005

The application of wavelet analysis to rhythm is described. Rather than carry out the multiscale decomposition on the sound signal itself this method decomposes the sound energy flux [N. P. McAngus Todd, J. Acoust. Soc. Am. 92 (A), 2380 (1992)], This enables the analysis of rhythmic phenomena that have frequencies several orders of magnitude below pitch and are not represented in the sound signal. The decomposition is carried out by a logarithmically spaced Laguerre series approximation Gaussian filter bank. The analog design was discretized to a cascaded IIR structure using a bilinear transformation. This has the following advantages over the usual FIR approach: (a) plausibility of interpretation as a perceptual model since the ideal Gaussian is not physically realizable; (b) a smaller number of parameters; (c) smaller delay times than the Gaussian ideal; (d) avoidance of the need for down/up sampling. Two complementary structural components are obtained from the projection in the frequency-time plane of loci of zero crossings in the rate of change of energy: (a) segmentationstructure, corresponding to positive second derivatives; (b) a stressstructure, corresponding to negative second derivatives. A compact tree coding of rhythm is possible from points of convergence of the loci of zero crossings.