bluuebunny/crossref_metadata_2025 huggingface dataset:
Measurement of ocean-bottom compressional and shear speeds in the Canadian high Arctic
Authors: Dosso S. E. , Brooke G. H.
Published: October 2005

Geoacoustic properties of the ocean bottom can be a significant factor influencing acoustic propagation in shallow-water environments. Knowledge of these properties is required for reliable acoustic propagation modeling and matched-field processing. This paper presents the preliminary analysis of a sea-floor seismic experiment carried out on the continental shelf of the Lincoln Sea north of Ellesmere Island, Canada, to determine compressional- and shear-speed models for the ocean bottom. A three-component ocean-bottom seismometer (OBS) was deployed through ∼4 m of multiyear sea ice in 540 m of water. Broadband explosive sources were detonated on the sea floor at ranges from 50 to 900 m. Source–receiver ranges and shot instants were verified by timing the direct and surface-reflected arrivals through the water. Subsequently, a layered compressional-speed model for the ocean bottom was determined from a first-break analysis of the head-wave arrivals. Also, interface (Sholte) wave arrivals have been identified on many of the seismograms and exhibit characteristic elliptical particle motions. Waveform modeling of these interface waves is discussed with a view to extracting shear-speed properties in the uppermost bottom layers.
Elastic normal modes in multilayered heterostructures
Authors: Esquivel-Sirvent R. , Cocoletzi G. H.
Published: October 2005

A simple matrix approach is presented to study the elastic properties of multilayered systems. Periodic superlattices are considered to calculate the dispersion relations of the collective normal modes, and finite and semi-infinite structures to obtain the reflectivity of longitudinal modes. The dispersion relations for a two layer period superlattice show band structure. For transverse waves, it is considered that the single layer may support surface modes and found that their interaction with those of the adjacent layers yield also a band structure. The reflectivity of normal incident longitudinal waves for the semi-infinite superlattices resembles the band and gap regions of the dispersion relations. The comparisons between theory and experiment of the reflectivity of sound waves for the three layer systems show good agreement.
Shear horizontal waves in layered structure with 180° ferroelectric domain
Authors: Cai Dingguo , Zhu Youxin , Wu Zhiming , Li Xingjiao
Published: October 2005

Theoretical analyses and numerical results describing the propagation characteristics of shear horizontal interface waves in layered structure with 180° ferroelectric domain are presented. A sandwich structure and a multilayered structure with finite layers are considered. It is found that there exist universally strong dispersion and multimode in a multilayered structure. There are N−1 dispersion curves in a structure with N 180° domain layers. For the multimode, some symmetry properties are found. The fundamental mode is a symmetrical one and the higher-order modes are symmetry or antisymmetry. The multimodes are arranged as symmetrical–antisymmetrical alternatively. There exist cutoff frequencies. It is noted that the B–G wave and the M–T wave may be obtained as special cases of the present theory. Three kinds of dispersion curves are found: ferroelectric domain layer wave, ferroelectric domain layer flexure wave, and lovelike wave. The dispersion relations and the multimode characteristics of the SH interface waves in 180° domain finite multilayered structure were verified by experiments. [Work supported by NNSF.]
Perception of vowel height: Effects of varying F1–F0 difference
Authors: Hoemeke Kathryn A. , Diehl Randy L.
Published: October 2005

Perceived vowel height has been reported to vary inversely with the difference (in Bark) between the first formant frequency (F1) and the fundamental frequency (F0) [H. Traunmüller, J. Acoust. Soc. Am. 69, 1465–1475 (1981)]. Syrdal and Gopal [J. Acoust. Soc. Am. 79, 1086–1100 (1986)] observed that naturally produced [+high] and [−high] vowels tend to divide at a critical F1–F0 difference of 3–3.5 Bark, corresponding to the range of the spectral center of gravity effect [L. Chistovich and V. Lublinskaja, Hear. Res. 1, 185–195 (1979)]. In the present study, listeners identified three sets of synthetic vowels varying orthogonally in F1 and F0 and ranging from /i/–/i/, /i/–/eh/, and /eh/–/æ/. For the /i/–/eh/ set, which spanned the [+high]/[−high] distinction, there was a steep identification boundary located at an F1–F0 difference of 3–3.5 Bark. However, for the /eh/–/æ/ and /i/–/i/ sets, which occupied regions where F1–F0 was always greater than or less than 3 Bark, vowel labeling varied more gradually as a function of F1–F0. In fact, for the /i/–/i/ set, F1–F0 was actually a poorer predictor of identification performance than F1 alone. [Work supported by NIDCD.]
Formant transitions and the identification of concurrent vowels
Authors: Assmann Peter
Published: October 2005

Listeners exploit several sources of information to segregate the speech sounds produced by two competing talkers. When two synthetic vowels are presented simultaneously and monaurally, listeners can take advantage of a difference in fundamental frequency (f0) to identify the vowels. This study compared the contributions of f0 differences and formant-frequency transitions to the identification of concurrent vowels. Accuracy improved when a two-semitone difference in f0 was introduced, compared to the case where both f0’s were the same. In some conditions a small, independent improvement was observed when one of the vowels was preceded by a 100-ms gliding formant pattern. However, the presence of the glide did not help to identify the vowel which is preceded; its presence helped by making the competing vowel more identifiable. One explanation for improvement in the glide condition is a formant-tracking process which groups together the formants of each voice on the basis of good continuity. However, this account predicts improvement for both vowels which was not observed. An alternative account is provided by a model that (i) performs a series of analyses using a brief, sliding temporal window, and (ii) determines which region of the signal provides the strongest evidence of each vowel constituent. [Work supported by NIDCD.]
Acoustic characterization of ice vibrations
Authors: Giellis G. R. , Yang T. C.
Published: October 2005

Ice cracking/breaking induces ice vibrations, hence measurements of acoustic waves in ice can be used to probe the forcing function which cracks/breaks the ice. Previous measurements using ice-implanted geophone sensors (array) have revealed the existence of longitudinal (plate), flexural, and horizontally polarized shear waves at low (<100 Hz) frequencies. Applying frequency-wave number analysis to hammer blow data received on a line array of three-axis geophones, the phase velocities of these waves were measured. As expected, the longitudinal and horizontal-shear waves travel with constant phase velocities whereas the phase velocity of the flexural wave is frequency dependent. The frequency wave number analysis also reveals the existence of a new wave, heretofore not noticed, with a different frequency dispersion characteristic compared with the flexural wave. This wave travels with a phase velocity comparable to the flexural wave, consequently identifying it in the time series data is difficult. This wave is hypothesized as the forced wave excited by a point force. Measurements of group velocities and attenuation for each type of wave are also discussed.
Identification and change/no-change discrimination of [i] stimuli: Further tests of the ‘‘magnet’’ effect
Authors: Lauckner-Morano Valerie J. , Sussman Joan E.
Published: October 2005

In original work Kuhl [Percept. Psychophys. 50, 93–107 (1991)] used a stimulus that was 120 mels away from the prototype (P) stimulus for demonstration of the ‘‘magnet’’ effect. Listeners in the current study who phonetically transcribed P and nonprototype (NP) [i] stimuli showed that 92% of responses for the 120-mel P token were not [i]. Further, another group of listeners rated the 64 stimuli as either the [i] in ‘‘peep’’ or ‘‘not [i].’’ Results showed that only 20% of responses rated the 120-mel token as [i] suggesting that this token along the −1 diagonal was not considreed an [i] member. Thus three other groups of listeners participated in ‘‘change/no-change’’ discrimination testing with 500-ms interstimulus intervals for stimuli along the −1 diagonal common to the P and NP distributions. Each group had a different fixed standard in each four-item trial: the Kuhl P token, the 120-mel token, and an 85-mel token. Results will be discussed according to the effects of different standards on discrimination sensitivity and prototype theories in phonetic perception.
Influence of dynamic spectral information on rate-dependent vowel perception
Authors: Gottfried Terry L. , Triesch Susanne K.
Published: October 2005

Earlier research has shown that listeners adjust their identification of spectrally adjacent tense and lax vowels according to the speaking rate of the surrounding sentence. This study attempted to determine whether these rate effects could be reduced if test stimuli more closely resembled natural speech. Two synthetic /æ/–/eh/ series were created: (1) covarying F1 and F2 with syllable duration (i.e., stimuli with more /æ/-like formants were also longer in duration); (2) covarying formant frequencies with syllable duration and with formant frequency contours (i.e., stimuli with more /æ/-like formants and duration had more symmetrical formant contours.) Stimuli from these series were embedded in fast and slow sentence contexts. In series (1), listeners identified more stimuli as /æ/ in fast sentences, showing rate-dependent perception of this tense-lax contrast, replicating Gottfried etal. [Phonetica 47, 155–172 (1990)]. However, in series (2) (with appropriate formant contours—symmetrical for /æ/ and longer offglide for /eh/), listeners showed no significant effect of rate. These results suggest that sentence rate may primarily affect vowel identification when important dynamic acoustic information is absent from the syllable itself.
Measuring articulatory movements with an electro-magnetic midsagittal articulometer (EMMA) system
Authors: Perkell J. S. , Svirsky M. A. , Matthies M. L. , Manzella J.
Published: October 2005

It is difficult to find experimental evidence for underlying mechanisms in speech production, largely because speech acoustics and kinematics are very variable, within and across speakers. Thus one approach to exploring speech production is with statistical analyses of large quantities of articulatory and acoustic data, gathered in a systematic and efficient fashion. If used carefully, EMMA systems can transduce large amounts of accurate movement data simultaneously from midsagittal points on the maxilla, mandible, tongue, lips, and possibly the velum. The use of such a system in combination with appropriate analysis techniques can make it possible to investigate speech kinematics while taking variability into account. This presentation will cover a number of aspects of the use of an EMMA system including: the theory of operation, subject safety, calibrations, preparing for and running an experiment, signal processing, data extraction, checking and using the data, and limitations of this kind of data. [Work supported by N.I.D.C.D.]
Magnetic resonance imaging (MRI) in speech research
Authors: Gracco Carol , Tiede Mark , Browman Cathe , Goldstein Louis , Gore John
Published: October 2005

Data on vocal tract shape and dimensions acquired using magnetic resonance imaging techniques have been useful in understanding the articulatory and acoustical processes involved in speech production. This video demonstrates the application of magnetic resonance imaging (MRI) to the study of vocal tract anatomy and application to the investigation of normal and abnormal aspects of the speech production. MRI techniques used to gather basic data to apply in computational models of speech articulation will be illustrated. Recent studies of the vocal tract in normal subjects used MRI to examine changes in vocal tract shape during production of different vowels and illustrated the correspondence between vocal tract shape and the acoustic product (Baer etal., 1990). Subsequent studies have correlated pre- and post-operative assessment of vocal tract dimensions with acoustic data to determine the effects of surgical alteration of specific vocal tract cites on speech production. Other studies have examined the effects of neurological disease on speech and voice production. Images will be used to illustrate nature of these changes on the speech mechanism. Applications of basic MRI techniques as well as specific limitations of this instrumentation will be discussed. [Work supported by NIH Grants Nos. DC-000444, DC-00121 to the Haskins Labs.]
Using a new model of vocal-tract midsagittal profile to area-function conversion to constrain an optimization algorithm for the articulatory-acoustic inversion of fricative consonants
Authors: Badin Pierre , Beautemps Denis , Laboissière Rafael
Published: October 2005

An extension of Heinz and Stevens’ αβ model [Proc. of 5th Int. Congr. Acoust. A44 (1965)], where α continuously depends on both midsagittal distance and VT position (along the midline), has been optimized from x-ray cineradiographs and formant frequencies obtained on a set of vowels and fricatives uttered by a subject. The purpose of the study was to recover the midsagittal function (a vector of 50 midsagittal distances) of fricative consonants from the measured formant frequencies, in [VCV] contexts. In order to reduce the excess degrees of freedom of this ill-posed problem, the proposed model has been used as a major way to constrain an optimization algorithm based on a gradient descent technique. The gradient of the error between actual and desired formant values is computed through a back-propagation network implementing both sagittal-to-area conversion and acoustic wave propagation. Preliminary results are presented and discussed. [Work partly supported by CNRS & CEC-ESPRIT project SPEECH MAPS.]
Bubble clouds as sources of underwater noise
Authors: Oguz Hasan N.
Published: October 2005

Wave breaking and subsequent formation of whitecaps are known to be the major contributor to the wind-dependent ambient noise levels in the ocean in the low-frequency range (100 Hz–1 kHz). A theoretical model that can account for the noise emissions from randomly distributed bubble clouds is developed in this study. The model assumes that individual whitecaps produce bubble plumes that grow as a result of air entrainment at the ocean surface. The injection of bubbles at the base of this plume excites the bubble cloud. With the only experimental input of the whitecap coverage ratio, the underwater ambient noise level is calculated, without the need of empirical constants, by integrating contributions from bubble plumes of all sizes. The results are found to be in good agreement with the available field measurements. [Work supported by the ONR.]
Endoscopy, stroboscopy, and transillumination in speech research
Authors: Löfqvist Anders , Oshima Kiyoshi
Published: October 2005

This tape demonstrates the application of endoscopy, stroboscopy, and transillumination to the study of laryngeal function in speech. Stroboscopy involves intermittent illumination of the vibrating glottis using short flashes of light with a frequency of slightly less than that of the glottal vibrations. Thus it has the effect of apparently slowing down the vibrations so that they can be visualized and measured. It has to be remembered, however, that one vibratory cycle recorded using stroboscopy is the composite of several actual glottal cycles. Recordings can be made using either a rigid endoscope or a flexible fiberscope; the latter allows analysis of running speech. Images are recorded on video tape for subsequent processing. Transillumination records the amount of light passing through the glottis and thus variations in the size of the glottal opening. The light is provided by a fiberscope and a transducer is placed on the neck. Transillumination is particularly useful for recording glottal abduction and adduction for voiceless consonants. Stroboscopic recordings using both rigid and flexible endoscopes are demonstrated. Applications are discussed, and the usefulness of simultaneously recording transillumination with other signals is exemplified. [Work supported by NIH.]
Spectral representation of envelope beat frequency in the discharge of cochlear nucleus units in the chinchilla
Authors: Shofner W. P. , Sheft Stanley
Published: October 2005

Nonlinear processing of stimulus envelope was investigated in the discharge of cochlear nucleus (CN) neurons. A standard method of investigating nonlinearities is to examine the output of a system for distortion products when the input consists of two tones. For a tone that is amplitude modulated by a two-tone complex, there is no spectral energy at the envelope beat frequency for either the stimulus or the half-wave rectified version of the stimulus. Post-stimulus time histograms (PSTHs) were generated for CN units recorded from barbiturate anesthetized chincillas to amplitude modulated best frequency (BF) tones modulated by a one- or two-tone complex. Modulation frequencies were between 4–64 Hz. Fourier analysis of the PSTHs shows spectral peaks corresponding to envelope beat frequencies. The magnitudes of the spectral peaks at the beat frequencies are less than the magnitudes of the spectral peaks at the primary modulator frequencies. Spectral peaks in neural spectra at envelope beat frequencies presumably arise from a compressive nonlinearity. Psychophysically, low-frequency envelope beating can be an effective masker of AM detection at the beat frequency (Sheft and Yost, in review). The preliminary results suggest that a spectral representation of the envelope beat frequency does exist at the level of the CN. [Work supported by a PPG grant from NIDCD.]
Array measurements of backscattered data from internally loaded cylindrical shells
Authors: Mackovjak David P. , Baggeroer Arthur B.
Published: October 2005

The structural acoustic interaction of model cylindrical shells with different structural discontinuities and internal loading configurations is studied using the maximum-likelihood method (MLM). Three different finite fluid-loaded cylindrical shells are insonified with a midfrequency broadband pulse. The original MLM algorithm is modified using time windowing and slowness steering to exploit the transient nature of the backscattered field. This allows for a detailed investigation of different propagating waves along the shell surface and internal subsystem. Results show that propagating helical membrane waves are important contributors to the scattered field and are frequency dependent. Frequencies below ka=5.5 result in a coupled system where the shell and the internal structures interact dispersively. Above this frequency, the membrane waves interact with the shell’s discontinuities creating almost continuously radiating localized point scatters. The source location proves to be an important parameter in the creation of these localized radiation fields. The MLM array processing appears to identify weak flexural resonances established by the shell’s physical geometry. [Work supported by ONR and Charles Stark Draper Laboratory, Inc.]
Acoustic radiation from a coupled planar semicomplex structure in heavy fluid
Authors: Atalla Noureddine , Berry Alain
Published: October 2005

A mathematical model is derived to address the vibrations and sound radiations into a dense fluid, of a coupled system consisting of two semicomplex plates (i.e., supporting added mass, stiffeners, and having arbitrary elastic boundary conditions) linked in four points through multistage suspensions elements. The top plate is subjected to point, line, or surface harmonic excitation, while the radiating plate is excited through the suspensions. Both plates are assumed to be baffled, and the radiating plate is fluid loaded. The model is based on a variational approach for the plates, and a matrix transfer approach is used to handle the coupling between the two plates. The solution is sought using a Rayleigh–Ritz expansion in terms of polynomial trial functions which are shown to allow for the arbitrary elastic boundary conditions and to facilitate the calculation of the radiation impedance matrix. The vibrations and noise design of the system is discussed. The main design indicators are the force transmissibilities between the different excitation and attachment points, the ratio of mean-square velocities of the two plates, the mean-square velocity, the radiated power, and the radiation efficiency of the radiating plate. Numerical examples are presented to show the effects of different design parameters (plates thickness, boundary conditions, added mass, stiffeners,...) on the radiated sound. [Work supported by DREA.]
Scattering from a finite width panel with attached internal resonances in an infinite baffle
Authors: Russell Daniel A. , Rochat Judith L. , Pierce Allan D. , Sparrow Victor W.
Published: October 2005

Recent research has shown that the vibration response of a finite panel in an infinite baffle may be strongly affected by attaching structures which have internal resonances. In another study ongoing at Penn State, it has been demonstrated that the mass of a fuzzy attachment has a significant effect on the response of a system to an incident acoustic plane wave. In the present research a formalism is developed for predicting the acoustic scattering from such panels with attached masses. A plane wave is incident on a panel, finite in one direction and infinite in the other, with light fluid loading assumed. The backscattering cross section and target strength are calculated for the panel alone. Attachments, treated as lumped element masses, are then added to the panel producing internal resonances of the system. In this paper, the attachments are assumed as well known, they are not fuzzy. The backscattering and target strength are calculated and compared to those of the panel without attachments. An effective scattering cross-section length then may be determined. [Work supported by ONR.]
Experimental investigation of enhancements in scattered pressure field of fluid-loaded shell with periodic structure
Authors: Photiadis Douglas M. , Houston Brian H. , Spanier Jonathan E.
Published: October 2005

The scattered field from an irregularly ribbed, fluid-loaded plate has been the subject of several recent investigations; one has proposed the existence of dispersive wave phenomena that is due to interference between Bragg diffraction and locally propagating Bloch waves [D. M. Photiadis, J. Acoust. Soc. Am. 91, 1897–1903 (1992)]. Recent measurements at NRL have confirmed the existence of such interference patterns in the scattered pressure field of a fluid-loaded cylinder having nearly periodically spaced ribs in the frequency region of 4.0<ka<30. It is shown that the theory for the two-dimensional case can be applied to the three-dimensional cylinder problem with moderate first-order agreement with experimental data. Possible approaches to better approximations of solving the cylinder problem are proposed and discussed, including using the developed (2-D) plate theory with a derived analytic form of the bending wave dispersion relation from the thin shell equations. The far-field behavior is shown to be determined by the dispersion relation of the free waves on the structure, the rib spacing, the transmission (T) and reflection (R) coefficients of a flexural wave incident upon a rib, and the deviation from periodicity of the ribs. The feasibility of applying this theory to experimental data for the purpose of determining T and R is also considered.
Design of a hearing test probe using active acoustic noise attenuation
Authors: Rafaely Boaz , Furst Miriam
Published: October 2005

Most hearing tests today are done in soundproof rooms. A method is proposed for conducting tests in a regular room while using a hearing test probe with active acoustic noise attenuation. The system consists of passive and active attenuation parts. The passive attenuation is achieved by a foam plug placed in the ear canal. The active attenuation is achieved by using two mini-microphones, a receiver and a DSP board. The first microphone is placed at the ear canal entrance and receives the acoustic noise. It then transfers the noise to the DSP board (with the TMS320, placed in a PC), which by using an adaptive algorithm transmits the antinoise signal to the receiver placed in the ear canal. The second microphone is placed near the receiver and produces the error signal to the adaptive process. The system was analyzed on a larger scale acoustic model and a 20- to 40-dB noise attenuation was achieved for tones up to 4 kHz (not including the plug passive attenuation). Theoretical analysis of the acoustic filter transfer function (from the outer microphone to the inner microphone) was done using computer simulation, and showed good agreement with measurement. Implementation of the system for broadband noise attenuation will be discussed. Preliminary results have shown that the characteristics and configuration of the acoustic probe and the DSP controller specifications have to be considered.
Real-ear attenuation provided by noise-reducing earphone enclosures
Authors: Frank Tom , Ritter Amy C.
Published: October 2005

Real-ear attenuation at threshold values were obtained for Audiocup, AudioMate, Auraldome II, and Madsen ME-70 noise-reducing earphone enclosures (N-REEs) housing a TDH-type earphone mounted in supra-aural cushion using ten subjects following the procedures specified in ANSI S12.6-1984. Overall, the attenuation provided by each N-REE increased as frequency increased and was about 8 dB from 125 to 250 Hz, 21 dB from 500 to 1000 Hz, and 35 dB from 2000 to 8000 Hz. The Audiocup provided the least (3 dB) and the AudioMate, the most (16 dB) attenuation from 125 to 250 Hz. The Auraldome II provided the least (18 dB) and the AudioMate provided the most (25 dB) attenuation from 500 to 1000 Hz. Each N-REE provided similar attenuation (35 dB) from 2000 to 8000 Hz. Compared with the attenuation values for a TDH-type earphone in a supra-aural cushion (re: ANSI S3.1-1991), each N-REE provided significantly more attenuation except for the Audiocup, Auraldome II, and Madsen ME-70 at 125 Hz and the Audiocup at 250 Hz. Implications for hearing testing using N-REEs as a means to reduce high levels of ambient noise will be discussed.
Ambient noise levels in audiometric test rooms
Authors: Frank Tom , Williams Dennis L.
Published: October 2005

Octave-band ambient noise levels measured in 136 diagnostic and 202 industrial audiometric test rooms were compared with the maximum permissible ambient noise levels (MPANLs) specified in ANSI S3.1-1991. The mean octave-band ambient noise levels for the diagnostic rooms were less than the ears covered but higher than the ears not covered MPANLs for each test frequency range specified in ANSI S3.1-1991. Specifically, 68 (50%), 69 (51%), and 112 (82%) of the diagnostic rooms met the ANSI ears covered MPANLs for the 125- to 8000-Hz, 500- to 8000-Hz test frequency ranges, respectively; however, only 14 (11%) for the 125- to 8000-Hz and 250- to 8000-Hz test frequency ranges and 39 (29%) for the 500- to 8000-Hz test range met the ANSI ears not covered MPANLs. The mean octave-band noise levels for industrial rooms were lower than the ANSI ears covered MPANLs for the 500- to 8000-Hz test frequency range except at 250 and 500 Hz where the levels exceeded the ANSI MPANLs by 2.4 and 6.4 dB, respectively. For the industrial rooms, 66 (33%) met the ANSI ears covered MPANLs for the 500- to 8000-Hz test range. Overall, both the diagnostic and industrial test rooms most often failed to meet the ANSI MPANLs at 250 and 500 Hz. Implications for diagnostic and industrial hearing testing will be discussed.
On the placement of microphones for outdoor measurements
Authors: Morgan Scott , Bass Henry E. , Raspet Richard , Klugerman Majorie , Wood Stephen
Published: October 2005

Acoustic measurements outdoors typically involve microphones located either on the surface or on stands which hold the microphone approximately 1.0 m above the ground. Arguments given for placing microphones on the surface include lower wind noise and ease of installation. Quite often microphones on the surface are mounted above a large solid sheet. In this talk, the effects of wind noise and ground impedance on the optimum design of measurement systems for different frequency and source position are considered.
Micrometeorologically forced intermittency in atmospherically propagated signals
Authors: Hansen Scott D. , Thomson Dennis W.
Published: October 2005

Line of sight, continuous acoustic signal propagation experiments were conducted as a part of the Joint acoustic Propagation experiment (JAPE) performed during July and August of 1991 at White Sands Missile Range. One of the purposes of JAPE was to examine the effects of the micrometerological phenomena on low-frequency sound. During one experiment, pure tones of 80, 200, and 500-Hz were propagated simultaneously for a period of 1 h over distances up to 1000 m. Tower-based measurements included both microphones and turbulence sensors spaced logarithmically to a height of 32 m. Frequency, height-, and range-dependent statistics of the recorded acoustic transmission loss (TL) are interpreted in terms of the ambient surface and boundry layer micrometeorological processes. In addition, nonlinear attractor-based parameterizations of signal variability are evaluated.
A fast phase-screen method for sound propagation through a turbulent atmosphere
Authors: Di Xiao , Gilbert Kenneth E. , Ameling Lucy J.
Published: October 2005

An ‘‘integrated-turbulence’’ phase screen method has been developed to efficiently treat sound propagation through turbulence having both small-scale and large-scale structure [X. Di and K. E. Gilbert, J. Acoust. Soc. Am. 92, 2404 (A) (1992)]. In this paper, the method is used with two different parabolic equation (PE) algorithms: a PE based on a Fourier transform, split-step range step and a PE based on a Crank–Nicolson range step. It is shown that the phase-screen approach allows one to take advantage of the much longer range step possible with the Fourier method and thereby speed up the calculation by a factor of approximately 100 relative to the Crank–Nicolson result. Consequently, with the phase-screen method and the Fourier PE, it is practical to directly compute the statistics of the fluctuations in the acoustic field. Results are shown from such calculations and comparisons are made with available experimental results.
Sound propagation through atmospheric turbulence: Multifractal phase fluctuations
Authors: Mellen R. H. , Siling G.
Published: October 2005

Measurements of phase fluctuations in sound propagation through turbulent air show frequency spectra that approximate the Kolmogorov β=5/3 scaling law over several decades. This suggests a fractal-like process for which the apparent fractal dimension would be D=(5−β)/2≊5/3. However, most natural phenomena are found to be multifractal rather than monofractal. Multifractal theory treats, not only first and second moments of the process, but also the relation between moments (including nonintegral). For monofractals, this relation is simply a linear function of the moment number and a codimension 0≤C1≤1. The degree of multifractality 0≤α≤2 is determined from systematic deviations from linearity. The parameters α, β, and C1 are all indicators of the nature of the nonlinear energy cascade. Analysis of experimental data obtained at 5 kHz, range ≊4 m and wind speed ≊12 m/s is reported here. Results indicate that phase fluctuations are ‘‘hard multifractal’’ (α≳1), which is comparable to measures of velocity fluctuations in atmospheric turbulence.
Segregation of voiced and whispered concurrent vowels in English and Japanese
Authors: Lea Andrew P. , Tsuzaki Minoru
Published: October 2005

This paper presents new evidence that listeners can use the difference between voiced and whispered speech to segregate both voiced and whispered members of concurrent vowels consisting of voiced/whispered pairings. Previous experiments performed by Lea and Summerfield [Acoust. Soc. Jpn. Tech. Rep. H-92-31] required English listeners to identify the members of pairs of steady-state synthetic English vowels, called concurrent vowels. Accuracy of identification was higher when one vowel was voiced and the other whispered compared to control conditions. Surprisingly, the improvement in accuracy was restricted to the whispered member of the voiced/whispered pair, thus suggesting that the whispered vowel was segregated, but not the voiced vowel. Lea and Summerfield’s study was replicated in this paper using Japanese vowels and listeners, however, the results suggest that both the voiced and whispered members of the voiced/whispered pair were segregated. These incompatible results either indicate a difference in segregational ability between English and Japanese listeners, or a difference in procedure between the experiments. Correspondingly, further experiments were performed using Lea and Summerfield’s English vowels and English and Japanese listeners which show that the incompatible results are mainly due to the different methods of matching the levels of the voiced and whispered vowels between the two experiments. Thus the evidence suggests that the Japanese results indicate the true segregational ability of listeners.
Testing the response of personal noise exposure meters (noise dosimeters) to impulse-like signals using a laboratory reference noise dosimeter and a theoretical model (after ANSI S1.25-1978)
Authors: Evans David J.
Published: October 2005

A laboratory reference noise dosimeter consisting of a measuring amplifier and instrument controller was assembled and characterized. A theoretical expression was obtained for the A-weighted, rms-detected output of a noise dosimeter assuming an input signal equal to that of a tone-burst sequence containing bursts comprised of an integral number of cycles of a sine wave. The output of the theoretical detector was numerically integrated to obtain percentage criterion exposure or time-weighted equivalent level. Equivalent levels were measured and calculated for each sequence in a set of tone-burst sequences with burst durations of 1 to 1000 cycles and crest factors of 4.8 to 30 dB. The agreement between equivalent levels obtained using the reference dosimeter and theoretical calculations was typically +0.1 dB. The results of these measurements and calculations were then used as a basis of comparison for testing the response of commercially available noise dosimeters to a subset of the tone-burst sequences. [Work supported by OSHA.]
Reducing instability through dimensionality reduction, with application to matched-field processing
Authors: Byrne Charles L.
Published: October 2005

High-resolution estimators of source parameters, such as Capon’s maximum-likelihood (ML) method, become unstable in the presence of nonwhite ambient noise and can degrade rapidly when the data includes perturbations due to model mismatch, finite averaging, system phase errors and the like. To combat the instability, some have suggested the use of multiple mainlobe constraints or derivative constraints (Vural, Steele). Stability through beamspace processing was proposed by Gray and was suggested as a way of avoiding nonwhite noise by Bienvenu and Kopp; dimensionality reduction (DR) is an extension of this idea. In matched-field processing (MFP), DR can be introduced through mode-space processing (Yang) or through ‘‘reduced ML’’ (Byrne). When the number of propagating modes approximates the number of phones more dramatic improvement over ML can be achieved through ‘‘sector-focusing’’ (SF) (Byrne and Steele). The basic idea of SF is to project the data vectors onto a subspace of lower dimension. Application of SF to MFP is considered by Frichter etal. [J. Acoust. Soc. Am. 88, 2843–2851 (1990)].
Inversion for bottom sound speed and bathymetry in a 3-D shallow-water Arctic environment
Authors: Ozard John M. , Dawson Trevor W. , Dosso S. E.
Published: October 2005

Propagation modeling and matched-field processing (MFP) require a good knowledge of ocean-bottom properties on a grid of points covering the search region. These parameters can be difficult to measure in range-dependent Arctic shallow waters. An effective environmental model based on MFP inversion can be easier to obtain than extensive geophysical measurements and more effective for MFP if the same propagation model is used both for inversion and MFP. An even more effective model for MFP is obtained if inversion is carried out for several consecutive source positions. In developing MFP-based inversion for this scenario, recovering the bathymetry alone was investigated first, then both sound speed and bathymetry, for sources at a number of known ranges. After these successful inversions, inversion for recovering the unknown position of one source and the bathymetry was investigated. Both of these cases employed two-dimensional modeling but it is recognized that significant changes in the predicted field can occur when the third dimension is taken into account. Examples of seamounts and realistic shelf slope environments were modeled in three dimensions and the effect of inverting in two dimensions for bottom properties in the shelf slope case was investigated.
Global inversion using genetic algorithms
Authors: Gerstoft P.
Published: October 2005

An inversion of sound fields for determining unknown environmental parameters can be separated into four parts: (1) discretization of the environment and discretization or transformation of the data, (2) efficient and accurate forward modeling, (3) efficient optimization procedures, and (4) uncertainty analysis. While much work had been done on the first two, much less has been done on the latter two, especially for object functions with several local minima. Global optimization methods accept the multiple minima and try to find the global minimum, without doing an exhaustive search. These methods are based on directed Monte Carlo search, and two promising methods are simulated annealing and genetic algorithms (GA). These have been compared and it has been found that GA’s often are superior. The global methods are time demanding and in order to speed up the convergence, gradient steps can be taken during the iteration process. The examples here will be based on a horizontally stratified environment where all the material and geometrical parameters can be taken as unknown parameters. The solution is presented in terms of aposteriori probability function describing the parameters. From this the most likely model parameters can be found and their uncertainty and importance can be assessed.
Matched-field inversion for range-dependent bottom properties and source location
Authors: Dosso S. E. , Ozard John M.
Published: October 2005

Ocean-bottom properties such as bathymetry and bottom sound speed can be difficult to measure directly in ice-covered Arctic waters, particularly if they vary with range. However, inversion procedures hold much promise for determining range-dependent bottom properties from acoustic measurements. The inversion is formulated as an optimization problem by assuming a discrete model of unknown parameters and a bounded search space for each parameter. The optimization then involves finding the set of parameter values which minimizes the mismatch between the measured and modeled acoustic fields. Two practical problems are considered. The first problem involves determining range-dependent bathymetry and bottom sound speed using a number of sources at known ranges. The second problem involves source localization when available knowledge of the bathymetry is inadequate. In this case, the inversion is carried out for source range and depth as well as bathymetry. Although the bathymetry cannot be determined uniquely in this manner, an effective bathymetry model can be computed which allows accurate source localization. This latter case is essentially an extension of the method of ‘‘focalization’’ [M. D. Collins and W. A. Kuperman, J. Acoust. Soc. Am. 90, 1410–1422 (1991)] to bottom parameters.
Sensitivity to duration and frequency of occurrence of tones in music
Authors: Cuddy Lola L.
Published: October 2005

Understanding music in the Western idiom includes perceptual sensitivity to the tone relationships and hierarchy of stabilities contained in the tonal-harmonic system. These features of the tonal-harmonic system are correlated with statistical properties of tonal music. For example, the frequency with which each tone is sounded has been shown to be closely related to its relative prominence in the tonal hierarchy. Listeners may be sensitive to such regularities, and, it has been argued, may use this information to develop, or elaborate, a sense of tonality for the music. This experimental work has pursued the issue of sensitivity to regularities in music by examining listener’s sensitivity to duration and frequency of occurrence in tone sequences for which construction rules deviated from the rules of Western tonality. Sensitivity will be described both for abstract sequences of pure tones and for musical materials provided by contemporary composers. It is contrasted with an alternative perceptual strategy—that of assimilation to the tonal hierarchy. [Work supported by NSERC.]
Stimulus properties critical to normalization of instrument timbre
Authors: Cho Jennifer L. , Hall Michael D. , Pastore Richard E.
Published: October 2005

The perceptual system appears to engage in active, time-consuming processes called normalization that maintain perceptual constancy by adjusting for source differences [e.g., Mullennix and Pisoni, J. Acoust. Soc. Am. 85, 365–378 (1989)]. Prior music research [Cho etal., J. Acoust. Soc. Am. 89, 1988(A) (1991)] demonstrated normalization for irrelevant timbre variability in chord identification. The present investigation identified important global components of instrument timbre that may be differentially subject to normalization. Using intact and physically manipulated natural stimuli, experiment 1 used similarity scaling to directly assess contributions of previously identified temporal and spectral properties of timbre. Contributions of attack and upper harmonics were evaluated. Results indicated that similarity was based primarily on the upper harmonics, with little contribution from attack functions. The relevance of these stimulus properties to normalization was evaluated in experiment 2 using an AX chord discrimination task. Normalization, as indicated by RT, was inversely related to similarity. Information present in higher harmonics appears to be the major contributor to instrument timbre and thus is most relevant to normalization. [Research supported by NSF and AFOSR.]
Abstraction of musical tonality
Authors: Steinke Willi R. , Cuddy Lola L. , Holden Ronald R.
Published: October 2005

Data were collected from 100 subjects for a series of music tests and standardized tests of intelligence and cognitive abstraction. Music tests focused on abstraction of tonality. Three tests employed the probe-tone technique [C. L. Krumhansl and E. J. Kessler, Psych. Rev. 89, 334–368 (1982)] for both harmonic and melodic materials. Four other tests collecting tonality ratings with melodic sequences and chord progressions were constructed for this study. The subject sample comprised the following ranges: age (18–40 years); general education (7–22 years); music education (none to professional). Results confirmed the reliability of the probe-tone technique. Results for all music tests were significantly intercorrelated and significantly correlated with music theoretic predictions. Amount of music training, but not level of general intellectual functioning or level of general education, was implicated in the subject’s responses to music tests. [Work supported by NSERC.]
Early musical acculturation
Authors: Lynch Michael P.
Published: October 2005

Humans can universally, under typical circumstances, appreciate music. This appreciation is presumably dependent on the ability to process musical structure, but it is not clear how this processing develops. The present study was designed to contribute toward clarification of this issue. Western infants were tested in detection of mistunings (increase of 3.2% in frequency of a randomly selected melody note) in melodies based on either native or nonnative musical scales. Infants repeatedly heard the well-tuned versions of the melodies and learned to turn their heads toward the sound source when they heard a mistuning. If infants, like adults, have developed schematic knowledge of musical tuning, then their performance in this task would be expected to be better in testing conditions involving native scales than in conditions involving nonnative scales. Although infants at 6, 9, and 12 months of age all performed the task with reliably better than chance success, statistically better native than nonnative mistuning detection was evident only at 12 months of age. This finding suggests that perceptual reorganization for musical tuning occurs in an age range that is similar to that observed for speech perception in infants. [Work supported by NICHD Grant No. HD28527.]
The perception and processing of foreign-accented speech: Contributions of various phonetic parameters
Authors: Magen Harriet S.
Published: October 2005

This research investigates the relative importance of various parameters contributing to the perception and processing of foreign accented speech. A native Spanish speaker of fluent heavily English read phrases designed to elicit prosodic and segmental deviations from American English. Parameters investigated were: prosodic (phrasal intonation and lexical stress); vowel segmental (vowel epenthesis, tense-laxness, reduction); consonant segmental [VOT, final /s/ deletion, fricative voicing (/z–s/), manner (š–č)]. English-speaking listeners rated extent of foreign accent of the Spanish phrases as actually produced and as edited by computer. Results showed that listeners are sensitive to manipulations, even those involving a brief portion (some as short as 60 ms) of a 3-s phrase, if the manipulation involves a phonological rather than subhonemic category. Results of a word monitoring experiment on the same stimuli will be discussed in terms of the contribution to perception and processing of subphonemic versus phonological and segmental versus prosodic parameters. [Work supported by NIH.]
Methods for testing vowel perception in the field
Authors: Ladefoged Peter , Johnson Keith
Published: October 2005

Based on procedures developed by Johnson etal. [J. Acoust. Soc. Am. 91, 2387(A) (1992)] programs have been written for Macintosh and MS-DOS portable computers for experiments on the perception of vowel quality in the field. In one version, vowels determined by 15 different values of F1 and 22 values of F2, with F3 determined by algorithm, were digitized and stored. In the basic experiment listeners try to match one of these stimuli to a selection of their own vowels. Listeners can hear any of the stored stimuli by clicking with a mouse on a square in a 22×15 matrix. In other experiments, listeners estimate the distance between pairs of stimuli, or select the better match to one of their own vowels from pairs of vowels that are algorithmically varied from trial to trial in accordance with the previously selected better match. A later version of these programs, not yet tested in the field, generates vowels in real time. The advantage of this system is that the fundamental frequency, the vowel duration, and the third formant can be manipulated by the subject. This is important when matching vowels in languages in which tone or length or rhotacization is contrastive.