bluuebunny/crossref_metadata_2025 huggingface dataset:
Nonlinear propagation effects on broadband attenuation measurements and its implications for ultrasonic tissue characterization
Authors: D’hooge Jan , Bijnens Bart , Nuyts Johan , Gorce Jean-Marie , Friboulet Denis , Thoen Jan , Van de Werf Frans , Suetens Paul
Published: July 2002

A study is presented in which the influence of the pressure amplitude of the incident pulse on the estimated frequency dependency of the attenuation coefficient is shown. First, the effect is demonstrated with a simple theoretical model for both transmission and reflection measurements. Simulations show that for both measurement techniques a high-amplitude incident pulse results in a biased estimate of the attenuation coefficient due to nonlinear interaction of the different frequency components of the incident pulse. It is shown that in transmission and reflection measurements the biases have opposite signs. The effect of bandwidth, central frequency, and phase of the incident pulse on this bias is investigated. Second, the effect is demonstrated both in vitro, using a broadband through-transmission substitution technique on a tissue mimicking gelatine phantom, and in vivo, using reflection measurements with standard clinical equipment. The experimental results agree well with the theoretical model. The relevance of this study for ultrasonic tissue characterization is shown.
Relationships between DPOAE and TEOAE amplitude and phase characteristics
Authors: Knight Richard D. , Kemp David T.
Published: July 2002

Most published data comparing the amplitudes of transient evoked otoacoustic emissions (TEOAEs) and bi-tonally evoked otoacoustic emissions (DPOAEs) indicate a low level of correlation, raising the question to what extent do the two responses share the same relationship with hearing function. However, DPOAE intensities are sensitive to stimulus parameters and comparisons with TEOAE have mostly been made with the specific range of DPOAE parameters found to optimize DP output level. To determine if other DPOAE stimulus parameter domains give closer correspondence between TEOAE and DPOAE characteristics, 2 f1−f2 and 2 f2−f1 DPOAE intensity and phase measurements were made across a sample 1/2-octave frequency range centered on 2 kHz in nine normally hearing subjects using a wide range of stimulus parameter configurations. The DP fine structure was resolved by detailed measurements and the mean DP levels were compared to those in the corresponding frequency range from TEOAE measurements obtained with the same probe fitting. The closest relationships between TEOAE and DPOAE amplitude were obtained with the smallest DPOAE stimulus frequency ratios and with lower DPOAE stimulus levels. For these conditions, the DPOAE intensity for individuals in this subject group could be predicted from TEOAE with a standard deviation of 1 dB, similar to the test–retest difference. The 2 f1−f2 DPOAE phase versus frequency gradients for fixed f2/f1 corresponded closely with phase gradients in TEOAE data when a small DP primary stimulus frequency ratio was used. They differed markedly at wider frequency ratios. In contrast, 2 f2−f1 DP phase agreement with TEOAE was good for all stimulus parameters, where measurable. These data suggest that the detailed mechanism of TEOAE and all DPOAEs is very similar when close stimulus tones are used to stimulate DP's. Significant divergence exists with the 2 f1−f2 DP with the wider stimulus ratios typically employed for clinical testing. The reasons are discussed.
Constant “segmental anchoring” of F movements under changes in speech rate
Authors: Ladd D. R. , Faulkner Dan , Faulkner Hanneke , Schepman Astrid
Published: July 2002

This study develops the notion of the “segmental anchoring” of F0 movements, specifically, the finding that both the beginning and the end of a rising pitch accent are anchored to specific points in segmental structure [Arvaniti et al., J. Phonetics 26, 3–25 (1998)]. If there are segmental anchors for rising accents (and if, as shown by Arvaniti et al., the F0 level of the beginning and ending points is unaffected by the experimental manipulations), the anchors should be closer together as the speaking rate increases, and the rises should therefore be shorter and steeper. This was tested in two experiments in which speakers were asked to modify rate. The first experiment confirmed the basic prediction that the duration of accentual F0 rises is affected by changes in rate. The second experiment studied the alignment of the beginning and ending points with their hypothesized segmental anchors. For all speakers there was a strong correlation between the duration of the rise and the time interval between the anchors, but only small and inconsistent effects of rate on alignment. Effects of rate on F0 excursion size were likewise small and inconsistent. The results support a model in which pitch accents consist of “tonal targets,” and in which the alignment and F0 level of tonal targets are what determine a pitch accent’s shape.
Bistatic scattering: A new way to improve sonar detection capabilities?
Authors: Schmitt Francoise , Daout Franck
Published: July 2002

In the field of seabed sonar imagery, it is necessary to establish local scattering models to improve the performances of the detection or recognition algorithms. In this paper, we present the Probability Density Function (PDF) of the acoustic intensity scattered by a natural profile as a function of the bistatic angle. To do this we have developed a 1-D bistatic scattering model called NEWS (Numerical Estimation for Waves Scattering) that incorporates physical phenomenons like multiple reflections, shadow and the reflection coefficient of the profile. Moreover, NEWS takes into account acquisition parameters like sensors characteristics and their positions in relation to the center of the illuminated area. Gaussian spectra for the profile height fluctuation are considered. Five hundred profiles are generated. For each profile NEWS’s algorithm gives the angular distribution of the scattered field in amplitude and in phase for all geometries and as a function of incident and scattered wave. The acoustic intensity is then treated as a random variable, and histograms are established. The PDF of the scattered intensity is compared to the K, Weibull and lognormal distributions and we examine the statistical informations providing by bistatic sonar.
Evaluation of multitrack recording practices in rock and roll
Authors: Case Alexander U.
Published: July 2002

Pop and rock music reaches its listeners using loudspeakers as the interface between the recording medium and the acoustic environment. The art of creating pop music recordings relies on a terrific amount of audio equipment to not just capture but also enhance the musical instruments recorded. In the never-ending search for a more compelling loudspeaker playback experience, the equipment of the recording studio is often used in unexpected and creative ways. The application of signal processing to multitrack pop music occurs using approaches that are accessible, but not necessarily intuitive to the acoustician. In this presentation the snare drum is studied to provide a view of some modern multitrack music recording and mixing techniques. The sound of the drum is analyzed both before and after some typical signal processing in an attempt to quantify the motivation behind these recording practices. Microphone selection and placement priorities are discussed. The application of equalization, compression, and ambience effects is analyzed. The perceptual significance of some processes is demonstrated through the playback of audio examples, both before and after signal processing. A live multitrack mixdown in which discreet elements of the mix can be isolated for evaluation is presented.
Calibration of a two-dimensional array system for ultrasonic aberration correction
Authors: Lacefield James C. , Phillips Daniel B. , Waag Robert C.
Published: July 2002

Two procedures for electronic compensation of array spatial and temporal nonidealities are compared. The motivations for array calibration are to account for signal variations due to differences in the impulse responses of the elements and to reduce beam degradation caused by nonideal element directivities. Both methods calculate inverse filters to equalize the signal at each element. One approach equalizes the array response to a planar reflector, while the other equalizes the response at a specified focal point. The point calibration method reduces the standard deviation of arrival time fluctuations in the measured wave front from a point reflector from 21 to 2 ns and reduces the standard deviation of energy level fluctuations from 2.5 to 1.7 dB. However, the point method also diverts more energy outside the main peak of the focused beam, which causes the −10-dB peripheral energy ratio to increase from 0.30 to 0.32. Point calibration of the receive aperture is nevertheless desirable for aberration correction using backpropagation followed by time-shift compensation because the fidelity of the correlation algorithm is dependent upon unbiased measurement of the echo wave front at the receive aperture.
Assessing arterial stenoses by tracking turbulence with Doppler ultrasound
Authors: Miller Megan M. , Holland Christy K. , Disimile Peter J.
Published: July 2002

When a stenosis, or narrowing, causes a significant area reduction of a blood vessel, turbulence in the post-stenotic jet can be detected with Doppler ultrasound distal to the stenosis. A technique is investigated for assessing the severity of a stenosis, defined as the pressure drop across the lesion, by extracting the streamwise turbulence intensity (or the normalized square root of the velocity variance) from the Doppler ultrasound in an arterial flow model. The model consists of an optically and acoustically transparent polyurethane tube that mimics femoral artery compliance, a pump capable of continuous and pulsatile flow, 10-μm glass spheres as an ultrasound and laser scatterer, and both blunt and rounded inlet stenoses. The flow field through three axisymmetric, Plexiglas stenoses with diameter reductions from 60% to 95% mH were investigated with an ATL HDI 3000 using the L7-4 linear array in Doppler mode (1.0-mm spatial pulse length) and a Dantec two-color 55X laser Doppler anemometer (0.7-mm major axis). To validate the ultrasound technique, correlation of Doppler ultrasound and laser Doppler anemometry flow measurements was examined. The correlation of the peak velocity, the maximum turbulence intensity, and the pressure drop across each stenosis was also investigated. This Doppler ultrasound technique could be sensitive to more subtle alterations in hemodynamics and therefore could aid early detection of atherosclerosis.
Suitcase lab for measuring digital cellphone interference in hearing aids
Authors: Killion Mead C. , Teder Harry
Published: July 2002

A low-cost, ‘‘real-life’’ method for measuring the interference caused by cellular telephones in hearing aids is proposed. Data would be valid for specific phone and hearing aid models. Estimated equipment cost is about $500. Real−ear recordings will demonstrate the audible interference.
Hearing aid measurement and consumer counseling
Authors: Victorian Thomas A.
Published: July 2002

The relationship between digital cellular telephones and hearing aid devices requires specific measurement techniques to determine system compatibility between the two technologies. The ANSI C63.19 working group has developed such measurement techniques related to the actual usage of a digital phone while wearing a hearing aid—known as ‘‘near field’’ or ‘‘user’’ condition. The hearing aid rf near-field immunity test requirements involve hearing aid orientation and bias as well as rf near-field simulated test conditions. The testing is performed in both the audio and telephone-coil modes. The measurement figure of merit is defined as IRIL (input referred interference level) which normalizes the hearing aid acoustic gain. It can be shown that hearing aids will be required to withstand digital phone E-field emission levels greater than 50 dB V/m and H-field emission levels greater than −2 dB A/m. By characterizing the simulated phone antenna properly this test enables hearing aid immunity testing within a controlled rf laboratory. The outcome of these measurements will be used to classify hearing aids for compatibility with categorized digital phones. This classification process will assist the hearing aid dispenser in patient counseling.
Acoustics of Baltic psaltery
Authors: Peekna Andres , Rossing Thomas D.
Published: July 2002

The Baltic psaltery, a plucked-string instrument, is known in Finland as ‘‘kantele,’’ in Estonia as ‘‘kannel,’’ in Latvia as ‘‘kokle,’’ in Lithuania as ‘‘kankles,’’ and in northwestern Russia as the ‘‘wing-shaped gusli.’’ In its archaic form, it had a limited range, ∼5−13 strings, usually tuned diatonically, though the lower strings were sometimes tuned as bourdons, several tones lower than the other strings. Results of experimental investigations on several instruments are presented. Methods used for calculating the lowest resonance of the air cavity are described. Investigations include determining the modes of vibration using electronic TV holography and recording sound spectra throughout the tonal range. Results indicate that the more successful instruments have many resonances close together in frequency, within their limited tonal range. This may account for why the archaic Baltic psalteries had several small soundholes rather than the single soundhole found on guitars and mandolins, which have much wider tonal ranges. Several instruments will be demonstrated.
Simplified model of the seismic reverberation observed due to the high-resolution transient acoustic signal
Authors: Dorfman Yevgeniy Y. , Krumhansl Peter A. , Goldsmith Michael
Published: July 2002

Unexploded ordinance (UXO) presents a lethal hazard to individuals worldwide. To predict and/or simulate the performance of a seismic sonar for detecting UXO, one needs to develop a model of propagation and reverberation in the inhomogeneous seismic media. An underground seismic detection experiment was performed during fall 1998 as a part of the BBN Innovative Seismic System for Buried Unexploded Ordinance Detection and Classification program. A physics-based model of monostatic reverberation due to the low-directivity shear wave source pointed down was postulated. This model provides a way to (a) simulate the time series for different source-receiver geometries and source signals; and (b) to invert the field measurements for ground parameters. Exercising the model, it was found that the observed signal due to reverberation is a function of shear wave speed (usually known), exponential attenuation, and assumed reverberation strength (often not well-known). It was further found that under realistic assumptions for UXO detection scenario frequency-dependent attenuation must be accounted for in the modeling. A numerical code was developed based on the same physical assumptions in order to account for nonmonostatic source-receiver configuration. It was found that collected data are matched well by the numerical model predictions. [Work supported by SERDP.]
Daytime sound levels and coherence as a function of height: Experiment and theory
Authors: Di Xiao , Gilbert Kenneth E. , Clark Richard D.
Published: July 2002

Daytime ground-to-ground acoustic propagation has been measured by a number of researchers over the years. Measurements of daytime sound levels as a function of height, however, have been made only in a limited number of near-ground experiments and apparently never at a considerable height. This paper reports measurements using a tethered balloon to measure the mean sound level and coherence from the ground up to 210 m at three separate ranges (300 m, 600 m, and 900 m) and at three frequencies (210 Hz, 380 Hz, and 600 Hz). A meteorological instrument package on the balloon measured the mean temperature, wind and humidity as a function of height. Fluctuation spectra were estimated using a stationary sonic anemometer at 5-m height which sampled the temperature and vector wind at 10 Hz. Acoustical inputs computed from the meteorological data were used with a three-dimensional parabolic equation model to predict the mean sound level and coherence as a function of height and range. The predictions and measurements agree reasonably well, both showing a clear shadow-zone boundary that is a function of the meteorological conditions. Descending across the shadow boundary, the predicted and measured mean sound levels and coherence decrease dramatically.
Hearing discrete elements of an auditory temporal sequence: Evidence for a central mechanism
Authors: Ries Dennis T. , DiGiovanni Jeffrey J. , Schlauch Robert S.
Published: July 2002

The ability to hear discrete elements of a temporal sequence was assessed. In one experiment, just-noticeable differences (jnds) in level between adjacent pulses of a 3- or 4-pulse temporal sequence that alternated in level was measured. The stimuli were pure tones (0.5-, 1.0- and 4.0-kHz tones), frozen noise, and white noise. All of the stimuli produced identical results. The jnd was about 5 dB for pulse rates less than 15 Hz and increased dramatically for higher rates. In a second experiment, listeners judged the initial direction change (high or low) of a frequency-modulated pure-tone sweep and a sinusoidal amplitude-modulated broadband noise. As in the first experiment, subjects performed poorly for rates higher than about 15 Hz. The similarity of the results for white noise, frozen noise, tones of different frequencies, AM noise, and FM sweeps is consistent with the notion that the factor-limiting performance is not in the auditory periphery. It is interesting to note that the approximately 15-Hz upper limit of performance on these tasks is consistent with the highest rates observed for phase locking in the auditory cortex. [Work supported by NIH-NIDCD R29 DC01542.]
Effect of syllable structure on syllable counting in English by Japanese
Authors: Yamada Tsuneo , Erickson Donna , Tajima Keiichi
Published: July 2002

A previous study [Erickson, Akahane-Yamada, Tajima, and Matsumoto, Proceedings of ICPhS (1999)] showed that Japanese listeners have difficulty correctly counting the number of syllables in spoken English words, presumably because Japanese syllable structure is more restricted than English. A complex syllable like ‘‘stress’’ is not allowed in Japanese, and tends to get mapped onto many syllables, such as ‘‘su.to.re.su.’’ To test what factors contribute to Japanese listeners’ perception of syllables, a set of nonsense words varying in number and voicing of initial and final consonants, vowel type, and number of syllables, was presented auditorily to Japanese and American listeners. The task was to count the syllables in the words. American speakers counted the syllables with nearly 100% accuracy; Japanese listeners performed at about 50% accuracy. The number of initial consonants seems to have a stronger negative effect on the syllable counting performance by Japanese listeners than the number of final consonants, or vowel type. These findings suggest that onsets may play a stronger role than codas in Japanese listeners’ perception of the psychological weight of syllables, contrary to current notions of phonological weight which are sensitive to codas and ignore onsets. Implications for second-language speech perception/production learning will be discussed.
Acoustics of Spanish vowels as spoken in two regions of Colombia
Authors: Urueta Mazzilli Dolly , Bahr Ruth , Strange Winifred
Published: July 2002

It has long been believed that Spanish dialects do not differ with respect to vowels. However, acoustic differences between Spanish dialects have been reported anecdotally. A reasonable assumption then is that, like English, Spanish dialects would vary both within and across countries. In fact, dialect differences have been found within regions of Panama [M. C. McNair, ‘‘An acoustical analysis of Panamanian vowels,’’ unpublished Master’s Thesis (1996), University of South Florida]. Therefore, the following analysis was conducted to determine if acoustic differences exist within Colombian Spanish. Nine male monolingual Spanish speakers from two cities of Colombia (Barranquilla and Santa Fe Bogota) produced three tokens of each of the five Spanish vowels in bilabial, alveolar, and velar environments within the frame, ‘‘Escribe CVCV bien.’’ Spectral and temporal measurements were calculated for each vowel in the first syllable of the word. A repeated measures ANOVA revealed a significant four way interaction suggesting that differences between dialects were dependent upon specific contexts, vowels, and formants. While vowel differences between dialects were subtle, perceptual differences in prosody and consonant production emerged.
Gender differences in fundamental frequency in focused words: A case from Japanese
Authors: Nagao Kyoko
Published: July 2002

When a word is focused in some context in Japanese, its fundamental frequency (f0) range is usually increased [Pierrehumbert and Beckman (1988)]. It is also well known that Japanese females tend to employ higher frequencies when they speak in Japanese than Caucasian females do [Yamazawa and Hollien (1990)]. However, previous studies have not examined the interactions between these two factors, focus and gender. The present study was done to determine whether the same strategy for realizing focus is employed by female and male speakers, in terms of the f0 and temporal patterns. From the analysis of 252 utterances produced by six Japanese speakers, almost the same amount of increase in absolute f0 values was observed in the speech of both sexes. Therefore, when viewed in proportion to the speakers’ normal f0 range, females showed smaller increases in the proportional f0 values on the focused words than males did. The results suggest that Japanese females have less room for focus expansion because they normally use higher frequencies within their pitch range. Since temporal focus effects do not occur in female speech, it is considered that the perception of intended words as emphasized will be more ambiguous in female than in male speech.
Effect of computer-assisted training on production of English /r/ and /l/ by Japanese
Authors: Akahane-Yamada Reiko , McDermott Erik , Adachi Takahiro , Takada Tomoko
Published: July 2002

Japanese speakers were trained to produce American English (AE) /r/ and /l/ using a computer-assisted learning system which was developed to investigate how to provide useful and effective feedback to second-language learners regarding the goodness of their production in an automatic way. In experiment 1, two groups of Japanese speakers were trained to produce /r/ and /l/ in a reproduction task. In one group, spectrographic representations with formant-tracking results overlayed were used as feedback, and in another group, acoustic scores produced by an HMM-based speech recognition system were used as feedback. Learners in both groups significantly improved from pretest to post-test in their production ability as evaluated by AE judges. Experiment 2 investigated the order effect of production training and perception training. One group of trainees received production training before perception training, and a different group received training in the opposite order. Both groups improved in their production ability by an equal amount from pretest to post-test. However, the production-to-perception training group improved more in perception ability than did the perception-to-production training group, suggesting the relevance of training order in perception ability.
Principal features for nonstationary signals from moments of the singular value decomposition of Cohen–Posch (positive time-frequency) distributions
Authors: Groutage Dale
Published: July 2002

This paper presents a new method for determining the principal features of a nonstationary time series process based on the singular value decomposition (SVD) of the Cohen–Posch positive time-frequency distribution. This new method uses density functions derived from the SVD singular vectors to generate moments that associate with the principal features of the nonstationary process. Since the SVD singular vectors are orthonormal, the vectors whose elements are composed of the squared-elements of the SVD vectors are discrete density functions. Moments generated from these density functions are the principal features of the nonstationary time series process. These descriptive features can be used in a variety of ways to gain information about the specific aspects of the time series process. This new technique has application to a broad area of fields and applications such as the medical field (nonintrusive diagnostics and condition classification), the defense industry (classification of acoustics transient signals), the field of machinery diagnostics, the field of marine mannal acoustics, and the manufacturing industry. An example is presented which illustrates how the new technique can be used to classify acoustic transient signals from underwater vehicles.
Time-varying coherent AM–FM demodulation and denoising of acoustic signals
Authors: Loughlin Patrick J. , Cakrak Ferhat
Published: July 2002

Noise removal via linear time-invariant (LTI) filtering is most effective when the signal and noise spectra have minimal overlap in frequency. In particular, it can be difficult to extract, via LTI filtering, broadband signals from broadband noise, because often their spectra overlap. However, many broadband signals are locally narrow band (e.g., AM–FM signals with large FM and moderate to small AM), and this characteristic can be exploited to improve noise suppression for such signals. We present a method for extracting locally narrow band signals from broadband noise, based on an AM–FM decomposition of the signal and time-varying filtering. The center frequency and passband of a linear time-varying filter are determined from estimates of the instantaneous frequency and instantaneous bandwidth of the signal. Results on both synthetic signals and recorded whale sounds in ambient noise demonstrate a significant improvement in SNR compared to LTI-based filtering. [Supported by ONR Grant N00014-98-1-0680.]
Anderson (1950) revisited
Authors: Feuillade C. , Clay C. S.
Published: July 2002

The Anderson fluid sphere scattering model [J. Acoust. Soc. Am. 22, 426–431 (1950)] has been reexamined to clarify three issues which have been the source of misunderstanding among underwater acousticians. First, the accuracy of the Morse large-range approximation for the spherical Hankel functions is investigated. It is shown that the minimum range for use of the approximation is strongly mode-number dependent, and should be carefully evaluated in short-range and/or high-frequency applications. Second, the precise characterization of the forward scatter region is studied. When the scattered field and the incident plane wave are combined, it is shown that little advantage is obtained in detection and localization applications by using forward scattering, rather than backscattering. Third, the translational response, or ‘‘rebound,’’ of the sphere under the action of the incident field is examined. By demonstrating that Anderson’s theory is a limiting case of Faran’s scattering model [J. Acoust. Soc. Am. 23, 405–418 (1951)] for an elastic sphere, which contains the rebound response, it is shown that the response is completely explainable within Anderson’s theory, and is consistent with a description which uses a normal mode expansion around a fixed origin. [Work supported by ONR.]
Inexpensive technique for accurate determination of elastic properties by free decay in circular plates
Authors: Gabrielson Thomas B.
Published: July 2002

Rapid determinations of elastic modulus are often made in the lab by measuring the resonances of regular bars of the material. When this is done, the most accurate results are obtained by suspending the specimen with as close to free conditions as possible. A circular plate supported by a taut wire through a central hole behaves accurately as an annular plate with both edges free. For low-loss materials, 10–20 modes are easily detected with an inexpensive microphone when such a plate is struck. Furthermore, the analytical solution for a thick, annular plate can be programmed readily in computational tools such as mathcad, matlab, or mathematica. (Even for plates traditionally considered thin—thickness-to-diameter ratios less than 0.1—thin-plate theory can produce significant errors for higher-order modes.) The first two resonance frequencies can be used to determine the elastic modulus and Poisson’s ratio and the higher modes can be used to assess the uncertainty or to uncover anisotropy. The resonance frequencies can be measured with a fast Fourier transform spectrum analyzer and an understanding of the response of such an analyzer to a slowly decaying sinusoid. [Work supported by the Naval Sea Systems Command.]
The history of physical acoustics: The worldwide scene
Authors: Beyer Robert T.
Published: July 2002

The term physical acoustics, used by von Helmholtz in his book in 1862 to distinguish the subject from physiological acoustics and music, did not become popular until after World War II. The term covers the production, transmission, and reception of sound, as well as the interaction of sound with matter. With such a broad definition, we shall be able here to delineate only a few major threads. The production of sound moved in the early nineteenth century from musical instruments, explosions, and the human voice, to the ingenious devices of Helmholtz, into the field of ultrasonics through the work of the Curie brothers, and into underwater sound from the researches of Boyle and Langevin. The attenuation of sound includes the theoretical work of Stokes and Kirchhoff, the experiments of Neklepaev, the further theory of Einstein, and the experiments of Kneser. Nonlinear acoustics began with the theories of Poisson and Riemann, was pushed along by Rayleigh, and developed in this century by Fubini, Lighthill, and Khoklov. The optical effect, discovered by Debye and, independently, by Lucas and Biquard, received its theoretical basis in the work of Raman and Nath, and was confirmed in detail experimentally by Nomoto.
Contribution of spectra of input signals to two ears to sound localization in the sagittal plane
Authors: Iida Kazuhiro , Rin Eigo , Kuroki Yasuko , Morimoto Masayuki
Published: July 2002

The previous studies show that the amplitude spectra of input signals to two ears contribute to sound localization in the sagittal plane [e.g., M. B. Gardner, J. Acoust. Soc. Am. 54, 1489–1495 (1973)]. Furthermore, it is known that the ear on the source side has more influence than the ear on the opposite side on localization [M. Morimoto, Dissertation, Univ. of Tokyo (1982); R. A. Humanski and R. A. Butler, J. Acoust. Soc. Am. 83, 2300–2310 (1988)]. It is, however, not clarified how each ear spectrum contributes to localization. The present paper builds up some hypotheses on this issue and examines them for their validity by some psychoacoustical experiments.
The relative contribution of onset asynchrony, harmonic ratios and angular separation of sound sources to the cross-spectral grouping of complex tones in a free field
Authors: Turgeon Martine , Bregman Albert S.
Published: July 2002

The Rhythmic Masking Release paradigm (RMR) was used to study how onset asynchrony, harmonic ratios and speakers separation interact toward the cross-spectral grouping of concurrent complex tones. In RMR, a regular sequence is perceptually masked by some irregularly spaced masking sounds, which are acoustically identical to those of the rhythm. The identification of the rhythm becomes contingent upon the perceptual grouping of the masking sounds with some flanking sounds of different frequencies. The rhythm-detection accuracy in a 2-AFC procedure was used to estimate the degree of perceptual grouping of 48-ms long masking and flanking tones, each composed of four harmonics. For each of 18 listeners, synchrony of onset and offset between the masking and flanking tones yielded an almost perfect rhythm-detection performance, independent of whether or not they shared a common fundamental frequency and of the spatial separation of their sources. Rhythm-detection accuracy strongly diminished with onset asynchrony and weakly diminished with inharmonicity and speakers separation. Our results suggest that onset asynchrony is the determining factor for the perception of concurrent sounds of a short duration as separate events; inharmonicity and the spatial separation of sources are weak factors, although they appear to reinforce each other toward perceptual segregation.
Measurement of vocal fold depth and thickness in frozen and thawed canine larynges
Authors: Tayama Niro , Chan Roger , Kaga Kimitaka , Titze Ingo
Published: July 2002

Anatomical data on vocal fold dimensions are necessary for defining the vocal fold boundaries in biomechanical modeling of vocal fold vibration. In the mid-membranous coronal section, vocal fold depth can be defined as the distance from the vocal fold medial surface to the thyroid cartilage, whereas thickness can be defined as the distance from the inferior border of the thyroarytenoid muscle to the vocal fold superior surface. Unfortunately, reliable geometric data from histological sections can be obtained only if the effects of sample preparation are quantified. For instance, tissue deformations are often induced by fixation and dehydration, sometimes producing shrinkages around 30%. In this study, reliable geometric data of the canine vocal fold were obtained by comparing frozen and thawed larynges. Coronal sections of frozen larynges were thawed gradually in saline solution. Images of the mid-membranous coronal sections at various thawing stages were captured by a digital camera. Measurements of vocal fold depth and thickness were made using a graphic software package (NIH image). Results showed that geometric changes of the vocal fold induced by freezing were likely reversed by thawing, such that the vocal fold depth and thickness measured on thawed larynges were representative of the pre-freezing state. [Work supported by NIH Grant No. P60 DC00976.]
Vocal rise time and perception of a hard glottal attack
Authors: Shrivastav Rahul
Published: July 2002

Three modes of voice onset are commonly recognized—breathy, normal and hard [Moore (1938)]. These differ in terms of the pulmonary airflow and the degree of medial compression force associated with vocal fold adduction [Koike et al. (1967)]. Perceptually, this is characterized by a ‘‘transient rise and fall in pitch and loudness at the onset of voicing’’ [Orlikoff and Kahane (1996)]. Perceptual identification of hard glottal attacks (HGA) is an important part of assessment and management of voice disorders. The relation between frequency and intensity rise times and perception of HGA was studied using a continuum of synthesized monosyllabic words varying in the frequency rise times. These tokens were recorded from a normal talker, modified and re-synthesized using straight [Kawahara (1997)]. Five listeners were asked to rate the perceived strength of the HGA on a five-point scale. Preliminary results with tokens varying in frequency rise time suggest an increase in the perceived strength of HGA when the frequency rise time increases from 20 ms to 50 ms. The perceived strength of HGA appears to plateau beyond rise times of 50 ms.
Pressure profiles on the walls of the glottis for oblique glottal ducts
Authors: Scherer Ronald C. , Shinwari Daoud
Published: July 2002

Oblique angles of the glottis occur during both normal and abnormal phonation. This study examined pressures on the glottal walls for a variety of oblique angle conditions. A Plexiglas model of the larynx was used to obtain wall pressures at 14 taps along the glottal surfaces. The model is 7.5 times larger than real life. A minimal diameter of 0.04 cm and duct axial length of 0.3 cm were used for all cases. Results indicated, for example, that for an included angle of 10 deg (divergence) and a transglottal pressure of 5-cm H2O, an oblique glottal angle of 15 deg compared to the symmetric glottis: (1) displaced the minimal wall pressure location on the divergent side a short distance downstream and on the convergent side to mid-glottis; (2) increased the pressure drop on the divergent side by about 15% and decreased the pressure drop on the convergent side by about 13%; and (3) broadened the minimum pressure dip on both the divergent and convergent sides. These results suggest paradoxical (opposite) driving pressures on the inferior glottal walls, which would enhance the nontypical phase differences between the two vocal folds. [Research support: NIH grant 1R01DC03577.]
Articulatory and acoustic data for vowels in isolation and in an /sVd/ context: An x-ray microbeam study
Authors: Slama Angela , Weismer Gary , Bunton Kate
Published: July 2002

Interspeaker variability in articulatory processes has been of great interest in the past decade, but studies addressing this issue have typically been limited to a relatively small group of speakers. In the present study we report on articulatory coordinates and formant frequencies for corner vowels spoken by 50 young adults. The vowels were produced in isolation, and in an /sVd/ context. Articulatory coordinates for the upper and lower lips and four tongue pellets were obtained at temporal midpoint of the vowels using the x-ray microbeam database. Formant frequencies, measured at the same temporal location, were derived from a well-known speech analysis package. Representations of both kinematic and acoustic interspeaker variability will be described and discussed, and relations between the kinematic and acoustic working space for vowels will be examined. A goal of this project is to develop some standards for kinematic/acoustic representations for vowels that can be applied to speakers with dysarthria. [Work supported by NIH DC00820.]
Children’s and adults’ perception of speech in noise
Authors: Fallon Marianne , Trehub Sandra E. , Schneider Bruce A.
Published: July 2002

Age-related differences in identifying speech in noise were investigated by means of a computerized four-alternative forced-choice task. In experiment 1, children 5 to 11 and adults heard prerecorded instructions (e.g., ‘‘Touch the X’’) mixed with multitalker babble. Selections were made on a touch-sensitive screen and corrective feedback was provided. To control for age-related differences in sensitivity, signal-to-noise ratios yielding 85% correct were determined for each age group. The S/N difference between the youngest and oldest listeners was 5 dB, consistent with previous research. Equivalent additions of background noise had comparable effects for all age groups, which implies that previously reported age differences in noise were attributable to sensitivity differences rather than noise effects. Experiment 2 investigated the degree to which sentential context affected speech identification in similar noise. Children 5 and 9 years of age and adults identified the final word in high-context (e.g., ‘‘I carried the water in a pail’’) and low-context sentences (e.g., ‘‘He looked at the pail’’), with differences in auditory sensitivity controlled. Adults benefited ore from high sentential context than did younger children, indicating that limitations in cognitive processing adversely affect children’s perception of speech in noise. [Work supported by the Medical Research Council of Canada.]
A new wavelet basis for extracting periodic acoustic signals in noise
Authors: Miller Antonio J. , Reichard Karl M.
Published: July 2002

A new wavelet basis is constructed to extract periodic acoustic signals in noise. A wavelet based comb filter is constructed from the linear superposition of narrow-band, harmonically related wavelet components. The best choice of wavelet components has proven to be a hybrid design incorporating the flat passband of the Shannon wavelet with the transition band of the Morlet wavelet. There are several benefits to taking a wavelet approach to comb filtering when dealing with periodic acoustic signals. Good noise rejection using a finite impulse response filter requires a precise measurement of the signal period and an adaptive sample rate to capture an integral number of samples per period. The wavelet approach relaxes the perfect frequency localization requirement and gives the comb filter ‘‘teeth’’ a small, but finite passband. This passband allows for measurement error in the signal period and for the signal to be sampled at an arbitrary rate. The new approach has proved to be an attractive alternative to the time domain averaging techniques used in accessing mechanical fault conditions from vibration measurements of rotating machines.
Characteristics of a set of noises used to test effects of underwater explosions on sea mammal hearing
Authors: Clark Joseph A. , Moore Paul M. , Young Jane A. , Gaspin Joel B.
Published: July 2002

As part of a study of temporary threshold shifts (TTS) expected to occur in sea mammals near underwater explosions, a set of ten explosion-like noises were synthesized [J. A. Clark, J. A. Young, and J. B. Gaspin, J. Acoust. Soc. Am. 105, 1048 (1999)]. The ten transient noises differed principally in maximum peak-to-peak pressure and were used in the TTS study as an ascending series of levels presented to two beluga whales and two bottlenose dolphins. The set of levels were selected from a larger set of predicted signatures of underwater explosions corresponding to a variety of charge weights and ranges in an underwater environment of interest which was characterized by charge depth, receiver depth, water depth, and sound-speed profile. This talk describes the ten transients used for the TTS studies. Time histories, frequency spectra, time-frequency distributions, and a variety of characteristics derived from these graphical descriptions will be presented.
The ASA standards program and underwater sound, are they incompatible?
Authors: Johnson Daniel L.
Published: July 2002

In the ASA National Standards Catalog, there are over 70 standards that concern the measurement and evaluation of sound in air. There are only a few standards, such as ANSI S1.11-1986 (filters) or ANSI S1.22-1992 (underwater transducer calibration) that concern underwater sound. This latter standard is the only standard of the 111 current standards that specifically relates to underwater acoustics. Standards such as ANSI S1.26-1995 (absorption of sound by the atmosphere) or ANSI S12.9-1996 (measurement of occupational noise exposure) just do not have a counterpart for water. Is this because there is no need? Is there no value in standardizing some common practices for underwater sound measurement and evaluation? Is it indeed true that underwater sound needs few standards? Comments from the audience are expected.
Evaluation of a FitCheck hearing protector test system
Authors: Murphy William J. , Franks John R. , Harris Dave A.
Published: July 2002

The measurement of attenuation provided by a hearing protector is an important element in the development of an effective hearing-loss prevention program. Three systems were evaluated for accuracy compared with a sound-field test of real-ear attenuation at threshold (REAT) [J. R. Franks, J. Acoust. Soc. Am. 105, 1129 (1999)]. The FitCheck method used with headphones was the most feasible system to use for field-test situations. However, the FitCheck method produced REAT values in the low frequency that were less than those produced by the sound-field method when used with the EAR Express earplug. A subsequent study has been conducted which evaluated the EAR Classic and the Howard Leight MAX earplugs with the FitCheck and sound-field methods. Twenty-four subjects were instructed to fit the devices without experimenter intervention (re: ANSI S12.6 Method B subject-fit protocol). Subjects were not allowed to adjust the earplugs between test methods. Mean REAT data from both methods will be compared. Preliminary data suggest that the FitCheck method yields lower REAT estimates for frequencies below 1 kHz, but the differences were not statistically significant.
Perceptual effects of place of articulation on voicing for audiovisually discrepant stimuli
Authors: Brancazio Lawrence , Miller Joanne L. , Paré Matthew A.
Published: July 2002

This research investigated effects of the visually specified place of articulation on perceived voicing. It is known that the /b/–/p/ boundary along a voice-onset-time (VOT) continuum falls at a shorter VOT than the /d/–/t/ boundary. Green and Kuhl [Percept. Psychophys. 45, 34–42 (1989)] demonstrated that tokens from an auditory /ibi/–/ipi/ continuum dubbed with an /igi/ video and perceived as /idi/ and /iti/ due to the ‘‘McGurk effect’’ had a voicing boundary at a longer VOT than when presented only auditorily and perceived as /ibi/ and /ipi/. We extended this finding in two directions. First, using an auditory /bi/–/pi/ series with a video /ti/ for which the McGurk effect did not always occur, we compared visually influenced (/d,t/) and visually uninfluenced (/b,p/) responses for these audiovisually discrepant stimuli. The /d/–/t/ boundary was at a longer VOT than the /b/–/p/ boundary, affirming the boundary shift’s perceptual, not stimulus-based, origin. Second, we tested the generalizability of Green and Kuhl’s findings using auditory /di/–/ti/ tokens with /ti/ and /pi/ videos. Preliminary results suggest that boundaries for /b/–/p/ and /bd/–/pt/ percepts (with video /pi/) may occur at shorter VOTs than for /d/–/t/ percepts (with video /ti/). Thus, the visually induced voicing boundary shift apparently replicates across different stimulus configurations. [Work supported by NIH/NIDCD.]
Formant modification of sung vowels in four languages
Authors: Watson Catherine I. , Thorpe C. William
Published: July 2002

Formant frequencies of sung vowels are modified at higher pitches because of formant tuning, leading to a reduction in intelligibility as the formants converge with increasing F0. Formant frequencies are also dependent on language-specific pronunciation requirements. The question of whether language-specific characteristics of sung vowels are maintained through pitch-effected formant tuning has not previously been examined. Sustained vowels from five similar positions on the vowel quadrilateral in four languages, English, Dutch, French, and Hungarian, were sung in a similar singing style over a range of pitches by a professional female singer who was proficient in singing each of these languages. Formant frequencies were extracted and the magnitudes of the voice partials computed from the average spectrum. The results showed an increase in F1 for the high vowels of all languages, but in the mid–low vowels there were some language differences in the direction of vowel shift as F0 increased. In addition, there were language differences in the relative magnitudes of the partials of all five vowels, particularly at higher frequencies. This suggests that some of the language-specific characteristics of sung vowels may be due to differences in style or voice characteristics.
Experimental investigation of air-driven free reeds using a laser vibrometer system
Authors: Busha Michael , Cottingham James P.
Published: July 2002

Previously reported measurements of free-reed vibration using a variable impedance transducer (VIT) gave a picture of the motion of a point on the reed throughout its cycle [J. P. Cottingham, J. Acoust. Soc. Am. 105, 940 (1999)]. A more detailed picture of the reed motion throughout its vibrational cycle has been obtained by measuring reed velocity using a laser vibrometer system. Unlike the VIT sensor, the laser vibrometer can be scanned along the reed, allowing reed profiles to be obtained. The laser system is also usable on reeds of magnetic material (e.g., steel accordion reeds) as well as reeds too small and inaccessible for the VIT. In this investigation the emphasis has been on comparison of the vibrational motion of accordion reeds with that of the reed organ reeds which had been measured in earlier studies. Some of the differences observed between the two types of reeds can be related to the structure of the reed tongues and frames.