bluuebunny/crossref_metadata_2025 huggingface dataset:
Use of one-dimensional photodiode arrays to detect string vibratiom
Authors: Nakamura Takashi , Dirkes Lawrence L. , Hanson Roger J.
Published: October 2005

A device has been developed to measure the x-y amplitudes of vibration of a string and to facilitate waveform and frequency analysis of vibrations below about 500 Hz. This device can be used to detect amplitudes approximately ten times as great as is possible with photon-coupled interrupter modules [C. E. Gough, J. Acoust. Soc. Am. 75, 1770–1776 (1984); R. J. Hanson, Phys. Teacher 25, 165–166 (1987)]. The device uses a pair of synchronized self-scanning diode arrays mounted perpendicularly to each other such that each array responds to a shadow of the vibrating string. The output of each scan is two series of pulses. The pulses are converted to zeros and ones that are used to produce a pair of numbers which correspond to the absolute location in each coordinate component. The numbers are stored in the memory of the Apple II microcomputer in which the device is resident and are also routed to a pair of DAC's to regenerate the waveforms.
Triggering algorithm for the Boie radio drum
Authors: Mathews Max , Bole R. A.
Published: October 2005

The Boie radio drum is used to play electronic music synthesizers. The drum is a sensitive surface that tracks the position and velocity of two or more drum sticks in a shallow volume above the surface in three dimensions. Positions and velocities are determined by measuring the strength and distribution of electrical capacitance between the tips of the drum sticks and the surface. The drum is made force sensitive in the surface normal direction by coating either the sticks or surface with an elastic foam layer. Position information of a stick in contact then corresponds to two-dimensional position on the surface and force into the surface. One of the drums functions is to generate triggers that start notes. The trigger algorithm must be very reliable in that one and only one trigger is produced no matter how gently or how hard the drum is struck. An algorithm, patterned after the action of a piano key has been implemented. A trigger plane and a reset plane are defined above the drum surface. Motion through the trigger plane produces a trigger. Motion back through the higher reset plane enables the next trigger. Loudness is controlled by the stick velocity as it crosses the trigger plane. The algorithm gives very reliable triggers over a bigger dynamic range than existing synthesizer keyboards.
Chaotic dynamics of woodwind multiphonics
Authors: Keefe Douglas H. , Laden Bernice
Published: October 2005

Experimental evidence for chaos in multiphonic tones derives from measured power spectra, and more directly from the measured correlation dimension of the reconstructed phase space of the dynamical system. The frequencies fl,m of the line spectral components of a reed-driven woodwind multiphonic fit a biperiodic spectrum of low- to mid-playing levels; i.e., there exist base frequencies f1 < f2 such that fl,m = lf1 + mf2, for non-negative integers l and m. For an alto saxophone multiphonic, these base frequencies are phase locked, namely, their ratio is equal to a ratio of small integers. Using the same nominal saxophone fingering, this measured ratio equals 8 : 5 or 7 : 4 (within 0.1%) depending upon player's adjustment of embouchure and playing level. A broadband spectrum is present in the saxophone multiphonic spectra at all but the lowest levels, which exceeds instrumentation noise and window leakage associated with signal processing. A phase-locked biperiodic spectrum superposed on a broadband background spectrum is characteristic of low-dimensional chaotic attractors in other fluid mechanical systems. At the highest playing levels, period doubling of the f1 component occurs, the broadband level significantly increases, and phase locking is not observed. The dimension of the attractor is measured by embedding the time series in a higher space. The correlation dimension D (Grassberger and Procaccia, 1983) is measured by embedding a single measured time series in a higher-dimensional space, so to reconstruct the phase space of the dynamical system. For the saxophone multiphonic at the lowest and highest playing levels, D = 2.85 ± 0.06 and 2.84 ± 0.12, respectively. These fractal dimensions suggest that multiphonics are strange attractors. Data for other woodwinds will be presented.
Perception of a musical construct by professional musicians
Authors: Baczewski Philip , Killam Rosemary N.
Published: October 2005

This study examined 15 highly trained listeners' response to a stimulus consisting of a 16-measure piece of music. Subjects were music performance graduate teaching fellows. They were presented with a taped performance (prepared by professional musicians) of three repetitions of a Mozart duet, together with the experiment's instructions. Experimental design allowed measurement of response to each stimulus repetition. Subject responses were recorded in one of two differing formats: eight subjects recorded the musical notation of each of the three stimulus repetitions in a different color of pen; seven subjects recorded the musical notation of each of the three stimulus repetitions on a different translucent paper overlay. Subjects were allowed 3 min to record each response. Data were encoded and analyzed to provide measurement of cumulative response arising from all three repetitions. Effects of pitch boundary and constituent groupings were obtained. A model emerges of complex perceptual strategies employed by musically sophisticated subjects. The strategies incorporate pitch relationships, contour, and grouping. [Work supported, in part, by an Apple Education Foundation Grant.]
Vocal range for speech determines perception of a musical pattern
Authors: Deutsch Diana , North Tom , Ray Lee
Published: October 2005

Previously, a two-tone musical pattern was described that possesses some highly unexpected properties [D. Deutsch, J. Acoust. Soc. Am. Suppl. 1 80, S93 (1986)]. For any given listener, it is heard as ascending when played in one key, and yet as descending when played in a different key. As a further paradox, when the pattern is played in any key it is heard as ascending by some listeners but as descending by others. Since then, the behavior of this musical paradox has been explored under parametric variation [e.g., D. Deutsch, Percept. Psychophys. 42, 563–575 (1987)]; however, the basis for the individual differences remained elusive. Here, it is demonstrated that a striking correlate exists between the way this pattern is perceived and position of the listener's vocal range for speech. To the authors' knowledge, this is the first demonstration of a close connection between the perception of a musical pattern on the one hand and the listener's speech characteristics on the other. Implications of these findings for both music and speech are discussed. [Work supported by the Digital Equipment Corporation, and by the UCSD Biomedical Research Fund.]
Native and non-native music perception by children
Authors: Lynch Michael P.
Published: October 2005

The development of culturally specific perceptual knowledge for music was investigated. Using a background/contrast paradigm originally developed for speech perception [Eilers et al., J. Speech Hear. Res. 20, 766–780 (1977)], Western 10–13 year olds and adults were tested in detection of mistunings in melodies based on native Western major and Western minor scales and a non-native Javanese pelog scale. Each age group was divided equally into musically inexperienced and musical subjects. The musically inexperienced adults had a perceptual bias for the melodies based on their native scales, but the adult musicians did not appear to share this bias, since they were able to generalize their knowledge of Western scales to the perception of a melody based on the Javanese scale. In contrast to the adults, the musician 10–13 yeat olds had a perceptual bias for the melodies based on Western scales, and the nonmusician 10–13 year olds did not share this bias. The musically inexperienced adults' performance was significantly better than that of the musically inexperienced children on both the Western and Javanese scales, and the same was true for the musician adults and children, indicating development of perceptual ability that is independent of musical training. Perception of Western mistunings was significantly related to the childrens' total years of musical experience and the amount of time devoted to practicing a musical instrument or voice. [Work supported by the American Physchological Association.]
Prosody as a cue to syntactic structure: The role of prosodic information in disambiguating phonetically ambiguous sentences
Authors: Price Patti J. , Ostendorf Marl , Shattuck-Hufnagel Stefanie , Fong Cynthia
Published: October 2005

To go beyond dictation tasks, speech recognition needs to be integrated with an understanding component. Natural language understanding research, however, has typically considered text input, and has not had to face the massive phonetic ambiguity arising from speech input: Even with no phonemic errors, word boundary ambiguities can yield thousands of parses for a substantial portion of moderate length sentences. This study assesses the potential role of prosody in aiding speech understanding. Using phonetically similar sentence pairs with and without a major structural break, seven types of structural ambiguity were investigated. Professional speakers read ten sentences of each type after reading diambiguating contexts. Naive listeners selected the appropriate context for each sentence. For recording and listening, only one member of each pair was used in each session; sentences were otherwise randomized. The results indicate substantial differences across structural types: e.g., appositions and tags were nearly always correctly identified, whereas main-main versus main-subordinate pairs were at the chance level. This suggests that prosody can disambiguate some, but not all, structural ambiguities. [Work supported by NSF.]
Phonetic evidence for a three-vowel system in Kabardian
Authors: Choi John-Dongwook
Published: October 2005

The Kabardian vowel system has been the focus of much debate. Central to this debate is whether or not the low vowel ɑ is quantitatively distinct from the mid vowel ə. This study examines phonetic evidence that bears on this issue. Spectrographic and durational analyses were conducted for 118 vowels taken from a Kabardian text read twice, once slowly and once at a normal rate, by three adult male speakers. Each vowel was measured for its first three formant frequencies and duration. The data reveal that the mean duration of ə is 62.5% that of ɑ, compared to the mean duration of i which is 67.5% that of ə. These figures represent ratios characteristic of intrinsic durational differences rather than distinctive length and support a three-vowel analysis. Qualitatively, the formant frequency measurements support the claims in the literature that Kabardian vowels contrast uniquely along the height parameter with some frontback allophony for the two higher vowels. Moreover, ɑ has a mean F 1 of 690 Hz as compared to ə which exhibits a mean F 1 of 510 Hz, further supporting a three-vowel analysis. [Work supported by NSF.]
More on domain-final lengthening and foot-level shortening in spoken English
Authors: Rakerd Brad , Sennett William , Cooke Paul
Published: October 2005

The stressed syllables of spoken English tend to be lengthened when they occur before major syntactic boundaries (domain-final lengthening) and to be shortened when they are followed by unstressed syllables within the same metrical foot (foot-level shortening). The nature of the interaction between these two factors has been investigated. A first study [B. Rakerd, W. Sennett, and C. A. Fowler, Phonetica 44, 147–155 (1987)] suggested that their effects are independent. In that study, the syllabic structure of metrical feet spanning word boundaries were manipulated, where those boundaries either coincided with noun-phrase/verb-phrase boundaries or did not, and we compared stressed syllable durations. A limitation on the comparison was that the phonetic details of the metrical feet of interest were matched but not held constant across syntactic conditions. In the present study, the procedure to control phonetic details throughout has been refined. With this modification, a significant interaction was observed between domain-final lengthening and foot-level shortening; specifically, it was found that the intervals between lengthenings delimited the domains over which substantial shortening could occur. The whole pattern of results will be discussed in light of current theories of foot-level shortening.
The syllabic attachment of post-vocalic liquids and nasals: Error elicitation evidence
Authors: Laubstein Ann Stuart
Published: October 2005

Inferences based on spontaneous speech error corpora regarding the syllabic attachment of post-vocalic liquids and nasals are experimentally tested using the error elicitation technique described in Motley, Baars, and Camden (1983). In the first experiment, 44 subjects were tested on “Vr” and “Vl” sequences. As predicted Vr sequences were not split up, whereas Vl sequences were. Subjects moved the vowel significantly more often in Vl sequences than in Vr sequences (Wilcoxon, p < 0.001); in addition “l” was moved significantly more often than “r” (Wilcoxon, p <0.0001). Here, Vl did, however, move as a unit, lending support to the Halle and Vergnaud (1980) sonorant constituent. The second experiment used 45 subjects to test the viability of such a constituent by comparing Vl to vowel nasal (“Vn”) sequences, and nasal and liquid obstruent (SO) sequences to obstruent obstruent (OO) sequences. As predicted Vl and Vn sequences were indistinguishable from each other and SO and OO were significantly different. This convergence of evidence from separate domains supports the generative assumption that constructs of competence underlie the capacity for use.
Classification of Japanese syllables including the speech sound found in loan words
Authors: Hiki Shizuo
Published: October 2005

Methods of classifying Japanese syllables are discussed with regard to their use in evaluating performance of speech processing techniques for both synthesis/recognition systems and user's speaking/listening characteristics. First, by taking into account the initial/intervocalic contrast of utterance, 16 intervocalic syllables are introduced to the traditional 50 Japanese syllables chart or the list of 100 Japanese syllables that has been used in articulation test of speech transmission channels. Then, by rearranging those syllables based on place/manner of articulation and by supplementing missing kind of combinations of the preceding consonant and following vowel, more than 60 new syllables which can be pronounced in the loan words are derived. Also discussed in this paper is rationale for various levels of deriving new syllables that correspond to the special Kana letters used in writing loan words in Japanese.
Architectural acoustics: The forgotten dimension
Authors: Wetherill Ewart A.
Published: October 2005

The basic considerations of architectural acoustics—isolation from unwanted noise and vibration, control of mechanical system noise, and room acoustics design—are all clearly exemplified in Sabine's design for Boston Symphony Hall. Opened in 1900, this hall is one of the outstanding successes in musical acoustics. Yet, as we approach the hundredth anniversary of Sabine's first experiments, acoustical characteristics remain one of the least considered aspects of building design. This is due, in part, to the difficulty of visualizing the acoustical outcome of design decisions, complicated by individual judgment as to what constitutes good acoustics. However, the lack of a comprehensive teaching program remains the dominant problem. Significant advances over the past 2 or 3 decades in measurement and evaluation have refined the ability to design predictability and to demonstrate acoustical concerns to others. New techniques such as sound intensity measurements, new descriptors for room acoustics phenomena, and the refinement of recording, analysis, and amplification techniques provide fresh insights into the behavior of sound in air and other media. These topics are reviewed with particular emphasis on the need for a comparable advance in translation of acoustic principles into building technologies.
Tapered-thickness piezoelectric ceramics: Models and characteristics
Authors: Barthé Peter G. , Benkeser Paul J.
Published: October 2005

Tapered-thickness piezoelectrics have been studied for use in novel devices and imaging techniques, such as wideband transducers and frequency-controlled beam translation. However, no adequate theoretical model exists. In this work, a staircase model is developed in which the thickness gradient is represented as discrete steps. Each step is modeled as a constant-thickness section, which contributes to the net response. This approach allows the results and techniques developed for conventional devices to be readily applied to those with tapered thickness; it emphasizes important similarities and differences, and provides physical insight into the dynamics of tapered piezoelectrics. The model shows that the distributed half-wavelength resonances dominate its response. In addition, however, a taper-dependent, transverse interaction exists among steps, which adds to the primary response. A coupled oscillator approach is used to model this transverse cross talk with an acoustic transmission line. The theoretical electric input impedance, spectral content, pulse, and continuous-wave pressure profile responses obtained with the model are shown to be in close agreement with the measured responses from several tapered piezoelectric transducers.
Computation of the performance of an electret microphone diaphragm
Authors: Madaffari Peter L.
Published: October 2005

An attempt was made to simulate the performance of a tensioned-diaphragm, electret-based, microphone. Use of commercial finite-element analysis programs were unsuccessful because of the computational system used. An alternate method was developed using a relaxation approach. A third-order equation for the balance of forces appeared to give approximate solutions for both tensioned and multiply supported diaphragms. Contours of physical deformation and acoustic compliance were generated and mapped. From these computations, an improved electret backplate and diaphragm system were designed.
Edge effects in standard shapes piezoelectric transducers—A new method for calculating elastic coefficients
Authors: Hazony Dov , Ozkaya Birol
Published: October 2005

An electrical network similar to the Mason model is constructed to conform with the electroelastic properties of thin isotropic piezoelectric ceramics of standard shapes under the condition of short stress pulses in this thickness mode. It will be seen that these pulses travel close to the speed of sound in an unbounded medium and that it is possible to calculate the elastic coefficients by setting the two active faces free. Specifically, both λ and μ, the Lamé constants, may be calculated from one measurement.
A high-frequency omnidirectional source for a robot end effector
Authors: Leifer Jack , Busch-Vishniac Ilene J.
Published: October 2005

A high-frequency omnidirection source has been designed that enables three-dimensional position monitoring in air to be performed. The source consists of a piezoelectric disk sandwiched between the large ends of two solid exponential Webster horns. When appropriately housed and driven to longitudinal resonance, the narrow end of one of the horns can be modeled as a baffled piston. Because of the high driving frequency (301 kHz) and the small radius of the horn, the directivity pattern of the source contains no nodes. Preliminary data regarding the sound field produced by the source were presented previously [J. Leifer and I. J. Busch-Vishniac, J. Acoust. Soc. Am. Suppl. I 85, S115 (1988)]. Since that time, a more quantitative study of the source field has been performed, the results of which are presented here. It was found that the horns did not resonate exactly at the theoretically calculated frequency so a mechanical tuning process had to be employed. The predicted and achieved resonant frequency of the source differs by about 7%. [Work supported in part by General Motors Research Laboratories.]
Self- and mutual radiation resistance measurement by use of the sound intensity technique
Authors: Suzuki Hideo , Anzai Masazou , Ono Takahiko , Fossaert Frank
Published: October 2005

Radiated sound power from sources can be measured by use of the sound intensity technique. The radiated power is obtained by integrating the normal intensity over the surface surrounding a sound source. Since the total radiated power from sound sources is determined by self- and mutual radiation resistances, it is possible to determine the radiation resistances from the measurement of radiated power. For a sound reinforcement system, it is useful to know the radiation resistances of loud speakers in a listening room. Acknowledge of self- and mutual radiation resistances is also important for an active sound power or noise control. As an example, self- and mutual radiation resistances of two closely located loudspeakers measured in lecture room are presented. Due to reflections, the resistance curve of a piston in an infinite baffle. The effect of the loudspeaker placement on the self and mutual resistances is very significant.
Some calibrations and applications of the aerodynamic fan reference sound source
Authors: Campanella Angelo J.
Published: October 2005

Introduced by H. C. Hardy [Noise Control (May 1959)], the mechanical reference sound source (RSS) provides a broadband sound field by turbulent air shear at the perimeter of a rotating open centrifugal fan wheel. For a given wheel rpm, geometry, and air density, the emitted sound level is constant within less than 1 dB when averaged over a few seconds. The sound power is relatively uniform in octave or third-octave bands from 125 to 8000 Hz. Usable levels of sound power are emitted above and below these frequencies, with field measurements in the 63-Hz octave band feasible in large rooms. Small motor-driven units have endured for decades. Hemi-anechoic, reverberation room and intensity methods JR. Peppin et al., NOISE-CON 88, and W. D. Gallagher (private communication)] for RSS calibration will be described. Low-frequency discrepancies between free-field and reverberation room calibration methods [P. Brnel, B & K Review (1978, No. 3)] have diminished as the reverberation room method is improved (better sound diffusion, space averaging, and decay analysis) [P. K. Baade, 1989 ASHRAE Trans. (January 1989)]. The portability and reliability of this device suit it especially to worldwide use in hostile environments such as factory assembly rooms and construction sites. Some recent experiences in application of the Acculab RSS to small and large device sound power measurement and room sound absorption will be described.
Diffraction about acoustically soft panels
Authors: Radlinski Ronald P.
Published: October 2005

The use of two-dimensional shaded and unshaded arrays to discriminate against diffraction about acoustically soft panels is investigated numerically by the use of a two-dimensional boundary element integral method. The calculated predictions for an infinitely long strip are used to estimate diffraction at a hydrophone array behind a square baffle. The diffraction floor for insertion loss of hydrophone arrays is determined by weighting the hydrophone responses with various shading coefficients. Comparisons of diffraction discrimination by a hydrophone array versus a single hydrophone at various positions are presented.
A generalized approach to random noise synthesis: Hardware implementation and performance characteristics
Authors: Hsueh K. D. , Hamernik R. P.
Published: October 2005

A theoretical model and a computer simulation of a generalized approach to the synthesis of random noises were presented in an earlier paper [K. D. Hsueh and R. P. Hamernik, J. Acoust. Soc. Am. Suppl. 1 85, S40 (1989)]. In this paper, the hardware implementation for this new synthesis technique is described. The digitally generated analog signals transduced through a high-intensity acoustic driver show that manipulations in the phase domain can be advantageous in creating acoustic signals of widely varying descriptors (e.g., skewness, kurtosis, etc.) while maintaining identical amplitude spectra. In essence, entire families of random waveform having identical amplitude spectra but variable statistical properties can be created. The influence of the acoustic driver on the spectra and the statistical properties of the synthesized waveforms will be shown. Some examples of the flexibility of phase manipulations will be presented, e.g., (1) amplitude-modulated random phase and (2) convolutions of the synthesized waveforms with any arbitrary function. These results illustrate that a variety of phase manipulations can be performed to produce interesting families of complex random waveforms. The application of this methodology to acoustic or vibration testing, environmental stress screening, psychoacoustic studies, etc., will be discussed. [Work supported by NIOSH.]
Respect for the variables—James P. Egan remembered
Authors: Miller James D.
Published: October 2005

In Egan's laboratory a variety of scientific eccentricities were forgivable or even desirable, but there was an absolute requirement that one “respect the variables.” Respect for the variables included their careful measurement, accurate recording, their unconfounded control, and beyond that a certain reverence. Egan's precept of “respect for the variables” has had a special impact on hearing-aid research at CID. Since 1966 with the first attempts to understand why a simple microphone-amplifier-earphone combination performed better than a hearing aid [Miller and Neimoeller, J. Sound Hear. Res. (1967)] to the current work on fitting digital hearing aids, this work has been guided by the “respect-the-variables” precept. Methods developed at CID and elsewhere have led to increasing accuracy in specifying the acoustic performances of hearing aids in relation to the listener's auditory area. Progress and current issues will be reviewed.
Spectral interference and the masking-level difference
Authors: Bernstein Leslie R.
Published: October 2005

Several investigations suggest that sensitivity to changes in interaural disparities within select spectral regions may be degraded by the presence of energy at other, even remote, spectral regions. It seemed interesting to determine whether similar degradations would be observed in an MLD paradigm. Detection thresholds were measured for NoSo and NoSπ. The signal, and 800-Hz tone (100 ms), was presented in continuous, broadband noise. Thresholds were also measured in the presence of an antiphasic, 400-Hz tone (the interrefer) gated simultaneously with the signal or presented continuously. NoSπ thresholds increased by about 7 dB with the gated interferer at 80 dB SPL. Smaller increases were observed with lower levels of the interferer. Slightly increasing or decreasing the frequency of the interrefer away from the octave reduced substantially its effect, as did presenting it continuously. NoSo thresholds were affected only slightly by the interferer. Reversing the roles of the signal and interferer (400-Hz signal, 800-Hz interferer) led to smaller, but reliable degradations in performance. Homophasic interferers had, in general, smaller effects on performance. [Work supported by NIH and AFOSR.]
Spectral tuning for a temporal decline of masking
Authors: McFadden Dennis
Published: October 2005

When a tonal signal of 1250-Hz and 5O-ms duration is masked by a spectrally nonoverlapping, narrow-band noise centered at 2050 Hz, the amount of masking observed depends upon the duration of this flanking noise band. Specifically, when the flanking band is gated on only for the duration of the observation interval (burst), detectability can be 20–25 dB worse for some subjects than when the flanking band is gated on a few hundred milliseconds prior to the observation interval (fringed). This temporal decline of masking as the fringe duration increases has the appearance of an adaptation effect—or what Egan used to call perstimulatory fatigue. When the fringe consists of a single tone of 2050 Hz preceding the flanking band, a substantial temporal decline of masking can still be obtained. As the frequency of this tonal fringe is varied around 2050 Hz, the magnitude of the improvement relative to the burst condition diminishes, and, at least for most listeners, a “tuned” existence region can be mapped for this temporal deline of masking. Possible explanations include temporal development of the critical-band filter, and within-channel adaptation followed by across-channel masking. [Work supported by NINCDS.]
Sociophonetic aspects of creaky voice
Authors: Henton Caroline G.
Published: October 2005

In a wide-reaching study of creaky voice in British English (Henton and Bladon, 1988) it was indicated that use of creaky voice varied significantly according to sex and accent of speakers, as well as with utterance position. Specifically, males used between three and ten times as much creaky voice as did females. Accentual differences were also found: Speakers of Modified Northern were more polarized in the overall amounts of creak used than were RP speakers. Individual speakers varied considerably in their use of creak, with a number of males being persistent creakers. In addition, creaky voice was shown to be an “end-of-utterance” phenomenon, used by females and males in a demarcative function. Further research using data from American English and from French indicates that creaky voice may be employed as a sociophonetic marker across cultures. Indeed, its role as a marker of male speech may be enhanced in certain cultures where gender-role expectations are more pronounced. The demarcative intonational function is observed again cross linguistically. This phenomenon, coupled with Hyman's (1988) recent results for West African languages, may make intonational creaky voice a candidate for a phonetic universal.
Acoustic scattering from transient, micron-sized cavitation bubbles
Authors: Apfel Robert E. , Madenshetty Sameer , Roy Ronald A. , Xu Qihong
Published: October 2005

If cavitation occurs in vivo due to the application of medical ultrasound, it is likely to be difficult to observe. Concern rests less with the destruction of biological cells and more with the more subtle subcellular injuries that could result in damage to components responsible for cell reproduction. Cavitation on the scale of single biological cells—microcavitation—can be observed in vitro by scattering high-frequency ultrasound (30 MHz) off of single transient cavitation bubbles. An active cavitation detection (ACD) has been used to determine the onset conditions (e.g., cavitation thresholds) in vitro for water with various artificial cavitation nuclei. More recently, the same scattering technique has been employed to observe the radius versus time characteristics of transient cavitation bubbles of one or more cycles. Reports on both threshold measurements and attempts to measure quantitative radius versus time histories of individual bubbles will be presented. The results will be presented in the general context of the joint Yale-National Center for Physical Acoustics collaborative effort to assess the safety of diagnostic ultrasound. [Work supported by the National Institutes of Health through Grant 4R01-CA39374.]
Bubble penetration—A major mechanism causing cell lysis?
Authors: Williams A. R.
Published: October 2005

The dramatic increase in the amount of cell lysis observed when the exposure chamber is rotated has been shown to be the result of recycling cavitation bubbles through the suspension [D. L. Miller and A. R. Williams, Ultrasound Med. Biol. (in press)]. In addition, the destructive effects of these bubbles have been shown to be reduced when the number of cells within the suspension was increased, with fixed cells being much more inhibitory than fresh cells [Ellwart et al., Ultrasound Med. Biol. 14, 43–50 (1988)]. It is proposed that under these exposure conditions cell lysis is caused by collision between cells and rapidly moving gas bubbles. If the bubble velocity is high enough then the cell is penetrated and lysed, but this interaction also slows down the bubble. Below a certain critical velocity it is presumed that the bubble attaches to (or is trapped within) a cell and ceases to interact with other cells. Additional support for this proposed mechanism has been obtained from studies utilizing erythrocytes embedded in agar gels and from histological observations in vivo [C. J. Martin et al., Ultrasound Med. Biol. 7, 235–265 (1981)].
The study of cavitation in tissuelike gels
Authors: Mortimer A. J. , Trollope B. J. , Birnbaum E.
Published: October 2005

A study of the effects of cavitation in agar gels was undertaken as an extension of studies in this laboratory of cavitation-induced free radicals in aqueous media using the free radical scavenger terphthalic acid. Agarose gels were made at a concentration of 1.5% agarose and 1% terphthalic acid in distilled, deionized water. Gels were exposed to ultrasound for times of up to 3.5 min in both pulsed (1:1, 100–1000 ms) and continuous modes, at intensities from 1–2 W/cm2. The liquid was then separated from the gel by dialysis and the fluorescence counted using a spectrofluorimeter. In all cases no fluorescence was measured; however, at the higher intensities, visible bubbles could be observed after treatment. At the same intensities and exposure conditions, significant levels of free radical production were observed in aqueous solution and in gels made from previously irradiated terphthalic acid. These results indicate that either cavitation thresholds in aqueous gels are significantly higher than would be expected from studies in aqueous solutions, or that the medium acts as an effective free radical scavenger, thereby limiting the possible biological effect of acoustic cavitation.
Loudness of sonic boom waveforms with atmosphere-induced rounding
Authors: Sohn R. A. , Schowe D. W.
Published: October 2005

The effect of atmosphere-induced rounding on the loudness of sonic boom waveforms is investigated. Waveforms with varying degrees of roundness representative of measured flight test data are considered. Simple equations are developed by which the perceived loudness and C-weighted sound exposure levels of these sonic booms can be calculated directly from their waveform parameters. This method shows excellent agreement with more complex loudness calculations (e.g., by Johnson and Robinson) as well as the full C-weighted sound exposure level calculation. A new set of flight test data is used to correlate shock strength with rise time. The results are compared to a semi-empirical model proposed by Dr. Kenneth Plotkin from earlier work.
Environmental impulse sound evaluation by LDI index: Physical and subjective tests
Authors: Brambilla G. , Cannelli G. B. , Santoboni S.
Published: October 2005

Direct measurements of the LDI index were performed by a new digital apparatus [G. B. Cannelli and S. Santoboni, Italian Patent No. 48694 A/87 and International Patent in progress] on a group of sounds having different impulsivity and recorded in the environment. The measurements showed that the LDI index accounts for fast fluctuations of sound level better than other noise indexes can do. In addition, the results indicated that the time constant is a critical parameter in measuring the impulse component of a sound. The sound recordings were presented in the laboratory at different sound levels and in balanced order to a group of naive subjects. They were asked to rate annoyance and impulsivity of each sound stimulus using a ten-point numerical scale [G. Brambilla, M. R. Carretti, and S. Santoboni, Proc. Noise 88, Vol. 3, p. 159, Stockholm (1988)]. Correlation between the LDI values and the subjective ratings is in progress in order to test the validity of this index in the evaluation of both impulsivity and annoyance of noise. [Work partially supported by CEC.]
The STRESS sonar experiment
Authors: Jackson Darrell R. , Dworski J. George
Published: October 2005

A high-frequency scanning sonar experiment has been conducted as part of the STRESS Program (sediment transport events: slopes and shelves). An autonomous platform was used to scan a circular region 100 m in radius with 5° angular resolution. The acoustic frequency was 40 kHz and the transmitted signal was an FM pulse of 2-kHz bandwidth. The objective was to quantify changes in bottom scattering due to storm-induced sediment transport events. The acoustic experiment was conducted in water of depth 90 m at a silty site off the California coast. Other experiments measured environmental parameters such as sediment properties, bottom currents, and suspended sediment concentration. Scans were made at a rate of ten per day over a period of 13 weeks during the winter of 1988–1989. The acoustic data have been processed to determine scattering strength and the correlation between scans as a function of time separation. These results are presented in the form of images showing the evolution of the scattering process during quiet and stormy periods. Very little change is seen in the scattering strength, but the correlation magnitude exhibits a monotonic decrease that accelerated during a storm. The phase of the complex correlation provides a very sensitive indicator of small changes. Two applications of these phase data will be presented. [Work supported by ONR.]
Verification of Bergeron's analytical method applied to room acoustics
Authors: Shimoda Hidemaro , Yoshida Norinobu , Fukai Ichiro
Published: October 2005

Applicability of the Bergeron's method to room acoustics has been studied, and was found useful for a transient analysis in the time domain [J. Acoust. Soc. Am. Suppl. 1 84, S64 (1988)]. In this paper, a verification study of the proposed analytical method is presented. Transient time responses and reverberation times measured in a rectangular scale model were compared to the analytical results. The time responses at six positions in the room were observed by pulse excitation with 1/3-octave-band tone burst, and reverberation times were evaluated from these responses using Schroeder's integrated impulse method. Each time response at various points agreed very well with that of the scale-model experiment and the average of calculated reveberation times also agreed favorably with the experiment.
Temporal change in interaural cross-correlation functions due to signal dynamics
Authors: Yanagawa H.
Published: October 2005

Maximal values of transient interaural cross-correlation functions (TRICC) for simple burst signals were used earlier to analyze temporal changes in spatial impressions such as sound image localization and subjective diffuseness. These temporal changes were found to correspond with subjective experiences [Yanagawa et al., J. Acoust. Soc. Am. 84, 1728–1733 (1988)], for the present study, a sound signal was chosen that is closer to transient sources such as music or voice. It consists of two bursts of white noise with a silent pause inbetween that is shorter than the reverberation time of the room. TRICC values at the rise of the second white noise signal were analyzed to see how they changed with the length of the pause and with room volume. It was found that a shorter pause causes the fall (reverberation) of the initial sound burst to overlap the second, so that TRICC values are smaller, but less so when the room is smaller. These results clarify how temporal changes in TRICC closely reflect differences in rooms with regard to ease of sound image localization.
Fundamental frequency and perceived prominence of accented syllables
Authors: Terken J. , Collier R.
Published: October 2005

In natural speech, accented words may differ as to the degree of perceived prominence. At the acoustic level, two aspects of fundamental frequency (F0) variation may be responsible for the perceived prominence differences: the magnitude of F0 changes and the relative frequencies of F0 maxima. Two experiments, with the same group of subjects, addressed the question of which aspect of F0 better predicts perceived prominence. Both experiments used reiterant speech with synthesized F0 contours. The speech materials consisted of ‘mamamamamamama’ utterances with F0 maxima on the second and penultimate syllables (“P1” and “P2,” respectively). In one experiment, subjects adjusted the frequency of P2 so that it was judged to have the same pitch as P1, for different rates of baseline declination. In the second experiment, subjects adjusted P2 so that it was judged to have the same prominence as P1, again for different declination rates. The results to be presented are relevant for refining the theory of pitch accentuation. For instance, if perceived prominence is predictable from F0 maxima, both experiments should give the same results.
(Semi-)automatic pitch-synchronous computation of glottal flow
Authors: Koreman Jacques , Cranen Ben
Published: October 2005

Large-scale investigations of the aerodynamics of voice production are most easily performed using noninvasive measurement methods. Glottal airflow can be estimated by inverse filtering the mouth flow, measured with a pneumotachograph mask. A semi-automatic pitch-synchronous inverse filtering method was developed that combines the advantages of interactive inverse filtering and inverse filtering with fixed-filter settings: Our method yields accurate and calibrated estimates of glottal flow during VC and CV transitions, which are amenable for parametrization and can be used for bulk processing. Using glottal dosing and opening moments derived from the electroglottogram, covariance LPC analysis on the closed glottis interval is employed to compute the optimal inverse filter for each period. For male speakers, the method yields quite stable waveforms. These will be parametrized in order to develop rules for a physiologically interpretable, time-varying synthetic voice source. The control parameters used are F0, duty cycle, top-top flow amplitude, glottal leak area, and vertical phasing. A number of these parameters can be derived from both the EGG and the glottal flow. Comparison of parameter values from the two signals will give an indication of their reliability. The results will be discussed.
Fundamental frequency database with linguistic and phonetic information
Authors: Abe Masanobu , Sagisaka Yoshinori , Kuwabara Hisao
Published: October 2005

An important problem for speech science is the relationship between syntactic information, prosodic information, and fundamental frequency contour. To facilitate the study of the interaction among these three factors, all three have been coordinated in a continuous speech database. Specifications of the database are as follows. (1) Speech samples consist of 503 phoneme-balanced Japanese sentences spoken by a male professional announcer [Kuwabara et al., ICASSP '89, 560–563 (1989)]. (2) Phonetic transcriptions at several levels of detail are provided [Takeda et al., Euro. Conf. Speech Technol. 2, 13–16 (1987)]. (3) Fundamental frequency is automatically extracted every 2.5 ms and extraction errors are corrected by hand. (4) The corresponding sentence is decomposed into constituent words and morphemes with lexical information such as inflectional categories and is assigned a tree structure. This information is semiautomatically generated from input texts. (5) Each utterance is segmented into minor phrases and each accent position is marked by listening to each utterance. This fundamental frequency database has been used to quantify fundamental frequency control factors and to show the effectiveness of this information.
Vocal fundamental frequency: Variation by language, language group, and sex
Authors: Wardrip-Fruin Carolyn
Published: October 2005

Vocal fundamental frequency was measured for speakers of five languages under three conditions (reading English, reading native language, and spontaneous speaking-native language). The samples were recorded in a sound-treated booth and analyzed by a Visipitch (Kay Elemetrics) frequency analyzer interfaced to an IBMxt computer. Preliminary analysis suggests that mean fundamental frequency was surprisingly similar across languages for the various speaking conditions, and that the mean fundamental was higher for reading than for speaking (as has been found in studies of English), but that there were significant differences between languages and by sex in standard deviation of the fundamental under the various speaking conditions (reading English, etc.). The results suggest that fundamental frequency is determined primarily by physiological factors with some linguistic variations.
On the mechanical properties of laryngeal muscles
Authors: Alipour Fariborz , Najafi Abdolali
Published: October 2005

Active properties of canine laryngeal muscles were investigated through a series of experiments conducted in vitro. Samples of the thyroarytenoid medial and lateral muscles, also, cricothyroid pars recta and pars oblique, were dissected from dog larynges excised a few minutes before death and kept in Krebs solution at a temperature of 37 ± 1 °C and a pH of 7.4 ± 0.05. Field stimulation with parallel-plate platinum electrodes was applied to study twitch and tetanic responses of these muscles in isometric conditions. The active force of each sample was recorded electronically with a dual servo system (ergometer). Results are reported on the twitch contraction times and half relaxation times of laryngeal muscles. Tetanic responses of these muscles were normalized and compared and results reported on tetanic 90% contraction and 90% relaxation times. It was found that the thyroarytenoid lateral muscle was the fastest muscle in the group with the mean twitch contraction time of 12.3 ms and mean half relaxation time of 11.9 ms, and thyroarytenoid medial muscle had mean twitch contraction time of 21.2 ms and mean half relaxation time of 18.6 ms. [Work supported by NINCDS Grant No. NS 16320-07.]
Monitoring noise exposure using employee movement analysis techniques
Authors: Bragg Thomas S.
Published: October 2005

In most industrial activity, employee work routines vary from one day to another or from week to week. The United States Occupational Safety and Health Administration (OSHA), recognizing this fact, has specified that an employee's long-term noise exposure should be expressed as a summation of partial exposures to varying sound levels. A technique has been developed that analyzes employee movement, breaking an average day into a series of locations or activities, each with an associated time duration. Sound level data are then acquired for each location or activity. A computer is then used to calculate the partial exposure, as defined by any criterion, for each location or activity and these are summed to express the overall exposure level for each employee job function. The computer also generates a list of noise control priorities and the minimum amount of noise reduction required at each location to achieve compliance with the selected criterion. Maximum priority is given to the location which, when attenuated, will give most benefit to most employees. Determining these minimum reduction levels is most important, as the cost and feasibility of noise control are directly related to the amount of reduction required.