bluuebunny/crossref_metadata_2025 huggingface dataset:
Acoustic inspection of inertial confinement fusion targets
Authors: Asaki Thomas J.
Published: July 2002

Acoustic cavity resonances were used to examine the interior surface geometry and gas fill pressures of various prototype inertial confinement fusion targets. A representative target consists of a millimeter-sized spherical beryllium shell filled to high pressure with deuterium/tritium (DT) gas at room temperature. Below the triple point of DT the gas forms a solid and redistributes itself symmetrically within the shell through a process known as beta-layering. The thickness of the solid DT layer, and the symmetry and smoothness of each surface must adhere to strict specifications for ignition to occur. Sound-speed measurements at room temperature provide the gas fill pressure and thus the solid DT layer thickness upon cooldown. These measurements rely upon an accurate equation of state. Degenerate-mode splitting of the cavity resonances provides interior surface geometry information. This technique was applied to a variety of deuterium and helium filled shells at pressures up to 356 atm. Several of these shells were manufactured with known interior surface perturbations. Measured mode splitting is compared with theory and the utility of the technique for cryogenic targets is examined.
Individual differences in perceptual closure at the phoneme level
Authors: Boothroyd Arthur
Published: July 2002

The amount of acoustic information available from the speech signal was reduced by removing all regions of the waveform in which the rms amplitude fell below a threshold value. This removal was accomplished by envelope center clipping followed by attenuation and envelope expansion. The processing was carried out independently in three-frequency bands, which were then recombined. The result was intended to simulate the effects of hearing loss with recruitment. Phoneme recognition in consonant–vowel–consonant words was measured in eight normally hearing subjects as a function of threshold. Subjects, essentially, heard the top x dB of the speech signal, where x varied from 3–30 dB. Significant individual differences of performance were found, covering a range of ±3.6 dB when measured in terms of the threshold required for a score of 50%. Because all subjects in this experiment had access to the same sensory evidence, the findings support a perceptual closure explanation of individual differences. It may be, therefore, that some of the individual differences of phoneme-level speech perception by the hearing impaired, or by normals listening in noise, also reflect differences in the ability to complete perceptual closure. [Research supported by NIDCD.]
Voice offset characteristics for voiceless fricatives and stops
Authors: Ni Chasaide Ailbhe , Gobl Christer
Published: July 2002

This paper is concerned with whether the nature and timing of the glottal abduction gesture might differ for voiceless fricatives and stops. The preconsonantal vowel in VCV sequences were examined, where C =/p(:) b(:) f(:) v(:)/ in French (single consonants), Italian, and Swedish (geminates). The speech pressure waveform was inverse filtered using interactive software and the LF voice source model [Fant et al., STL-QPSR 4/1985, 1–13 (1985)] was matched to the source signal. Results show a long breathy-voiced transition from a preceding vowel to the voiceless fricative, suggesting that glottal abduction occurs early. For the voiceless stops, however, results confirm a previous study [A. Ni Chasaide and C. Gobl, Lang. Speech 36, 303–330 (1993)] indicating that the timing of glottal abduction is quite variable and language/dialect dependent. A tentative conclusion is that early glottal abduction in voiceless fricatives may be a universal feature, necessitated by the dual production requirements of effecting devoicing with an open vocal tract and achieving the high oral pressure required for generating oral frication. This clearly is not the case for voiceless stops where the precise control appears to depend on language-specific rules. [Work supported by ESPRIT Project No. 6975.]
Low bandwidth aids to speechreading based on conveying speech band amplitude envelopes
Authors: Mansour Sharieff A. , Nadeau Philip M. , Braida Louis D.
Published: July 2002

The amplitude envelopes of filtered bands of speech can improve speechreading by normal hearing listeners substantially [Grant et al., J. Acoust. Soc. Am. 89, 2952–2960 (1991)]. Acoustic signals that convey these envelopes to listeners with severe hearing impairments are currently being evaluated in field and laboratory studies. An initial field study used SiVo aids to derive the amplitude envelope of the octave band of speech centered at 500 Hz and modulate a 200-Hz tone. Of two listeners who used the aids for two months, one benefitted substantially from the amplitude modulated signal and continues to use the aid in preference to conventional hearing aids. In the laboratory study, four normal-hearing listeners compared speechreading supplements derived from one and two filtered bands of speech. Presenting the envelopes of octave bands at 500 and 3300 Hz by modulating the amplitudes of 200- and 240-Hz tones led to keyword scores in IEEE sentences of 74% correct, compared to 76% corrrect when 200- and 500-Hz tones were modulated, and 63% when only a single 200-Hz tone was modulated. Modulating pairs of tones with closely spaced frequencies may thus assist speechreading by listeners with limited hearing. [Work supported by NIH.]
Lateralization effects in distant shadowing
Authors: Lyberg Bertil , Hällgren Åsa
Published: July 2002

In the present investigation the lateralization effects for distant shadowers are studied. The subjects listen to spoken utterances presented to them in either the right or the left ear and repeat the utterances simultaneously. The subjects are from previous shadowing experiments known to be distant shadowers, i.e., they repeat stimuli with mean delays of 400 ms or more [Marslen-Wilson, Speech Commun. 4, 55–73 (1985)]. The time difference between the incoming and the repeated speech is measured at the vowel onsets and the mean value of these differences are used as a measure of shadowing delay. Results from a series of experiments with natural speech with normal fundamental frequency and with constant fundamental frequency will be presented.
The locus of the verbal transformation effect: Two tests of the predictions of the Node Structure Theory
Authors: Shoaf Lisa C. , Pitt Mark A.
Published: July 2002

This study explored the verbal transformation effect, a perceptual phenomenon in which participants report hearing illusory words while listening to a recycling word. The Node Structure Theory, or NST [D. G. MacKay, G. Wulf, C. Yin, and L. Abrams, J. Mem. Lang. 32, 624–646 (1993)], an interactive model of speech perception and production, has been proposed to account for this phenomenon. Two experiments tested NST’s claim that verbal transformations have a lexical origin. Experiment 1 examined the influence that lexical information has on the perception of transformations by comparing words and pseudowords. Experiment 2 used the standard (perception-only) task to explore NST’s concept of lexical node stability (the idea that lexical items cause fewer VTs than nonlexical items) and its relationship to the number and types of transformations participants perceive. The results suggest that while the locus of the VTE is not lexical, lexical information does influence the types of transformations participants perceive.
A comparison of the acoustic structure of the vowels /i/, /u/, and /a/ in American and Russian mothers’ speech to infants and adults
Authors: Andruski Jean E. , Kuhl Patricia K. , Chistovich Inna A. , Chistovich Ludmilla A. , Kozhevnikova Elena V. , Ryskina Victoria L. , Stolyarova Elvira I.
Published: July 2002

Speech input to infants may exert an important influence on the development of language-specific perception by providing infants with information regarding the phonetic and phonological properties of their language. This study compares the acoustic structure of vowels produced by Russian and American mothers conversing with their infants, with the structure of vowels produced by the same women conversing with an adult. Mothers in both countries were asked to play with their infants (aged 2–5 months) using sets of toys whose names contained the target vowels in at least two different consonant contexts (e.g., sheep, bead). They were asked to use the same words in conversation with an adult. All mothers increased the degree of acoustic separation between vowel categories in infant-directed speech by selectively increasing or decreasing F1 and F2 values. Relative to American mothers [P. K. Kuhl et al., Science 277, 684–686 (1997)], Russian mothers showed smaller increases in fundamental frequency and less variation in vocal pitch while speaking to their infants. Nevertheless, mothers in both countries produced highly similar changes in acoustic structure, and comparable increases in the size of the acoustic space delimited by the vowels /i/, /a/, and /u/, when speaking to their infants.
Effects of language experience on speech perception: American and Japanese infants’ perception of /ra/ and /la/
Authors: Kuhl Patricia K. , Kiritani Shigeru , Deguchi Toshisada , Hayashi Akiko , Stevens Erica B. , Dugger Charmaine D. , Iverson Paul
Published: July 2002

Listening to language during the first year of life has a dramatic effect on infants’ perception of speech. With increasing exposure to a particular language, infants begin to ignore phonetic variations that are irrelevant in their native language. To examine these effects, 72 American and Japanese infants were tested at two ages, 6–8 months and 10–12 months, with synthetic versions of the American English /r/ and /l/ consonants. The /r–l/ contrast is not phonemic in Japanese. In both countries, the same experimenters, technique (head-turn conditioning), and stimuli were used. The results revealed two significant effects. The first shows the impact of language experience on speech perception. At 6–8 months of age, American and Japanese infants did not differ. Both groups performed significantly above chance (American M=63.7%; Japanese M=64.7%). By 10–12 months of age, American infants demonstrated significant improvement relative to performance at 6–8 months (M=73.8%), while Japanese infants declined (M=59.9%). Second, performance varied significantly as a function of the direction of stimulus change (/l/ to /r/ easier than the reverse), regardless of age or language experience. Discussion will focus on separating effects attributable to linguistic and psychoacoustic factors. [Work supported by NIH.]
Effects of audio-visual training on the identification of English /r/ and /l/ by Japanese speakers
Authors: Akahane-Yamada Reiko , Bradlow Ann R. , Pisoni David B. , Tohkura Yoh’ichi
Published: July 2002

Two groups of Japanese speakers were trained to identify AE /r/ and /l/ using two different types of training: audio visual and auditory only. In audiovisual training, a movie of the talker’s face was presented together with the auditory stimuli, whereas only auditory stimuli were presented in audio-only training. Improvement in /r/–/l/ identification from pretest to post-test on three types of tests (audio-only, visual only and audiovisual) did not differ substantially across the two training groups. Interestingly, the audio-only group showed improved identification in the visual-only tests, suggesting that training in the auditory domain transferred to the visual domain. A McGurk-type test using /r/ and /l/ stimuli with conflicting audio and visual information was also conducted. Identification accuracies on this test showed a greater effect of conflicting visual information at post-test than at pretest for the audio–visual training group, but not for the audio-only training group, suggesting that audio-visual training facilitated integration of auditory and visual information. Taken together, these results suggest that the internal representation of second-language phonetic categories incorporates both auditory and visual information. Implications for theories of perceptual learning and phonological development will be discussed.
Listeners’ representations of within-word structure for bilingual Japanese speakers of English and monolingual speakers of Japanese and English
Authors: Otake Takashi , Yamamoto Keiko
Published: July 2002

Assuming that listeners may form conscious representations of potential within-word structure, it was investigated what phonological units could be exploited by monolingual speakers of Japanese and English [Otake et al., Proc. EUROSPEECH 95, 1703–1706 (1995)]. The present study investigated how bilingual Japanese speakers of English and monolingual speakers of Japanese and English could form conscious representations of potential within-word structure in the two languages. The three groups of subjects were presented with spoken words in both languages and asked to mark on a written transcript of each word (e.g., buranko for Japanese and veranda for English) the second natural division point from the onset in the word. The statistical analysis showed that both the Japanese and English monolinguals exploited morae and syllables, respectively, and that the bilinguals exploited syllables. These results suggest that the exploitation of phonological units to represent within-word structure may depend upon the degree to which the structure of the language being acquired encourages the use of the units and that the system of the representation may be highly compatible with speech segmentation procedures [Cutler et al., Cogn. Psychol. 24, 381–410 (1992)]. [Work supported by a Fulbright senior research grant and TAF.]
Human localization of sound signals with reduced bandwidth
Authors: Christensen Flemming , Lydolf Morten , Soerensen Michael F.
Published: July 2002

In multimedia systems, the sound is often presented with reduced bandwith. As 3-D sound is introduced, this might give rise to localization problems, a consequence which is partly confirmed by this investigation. A listening experiment was made, in which 12 subjects participated. The subjects were placed in an anechoic room in a setup with 17 loudspeakers placed in different directions. As a reference experiment the subjects listened to a pink noise signal played from one loudspeaker at a time. The subjects were asked to point out the loudspeaker from which they perceived the sound. In eight other experiments the sound was low-pass filtered at frequencies of 1, 2, 4, or 8 kHz, or high-pass filtered at 0.5, 1, 2, or 4 kHz. The procedure was the same as in the reference experiment. For each subject the order of the tests was randomized. The experiment showed that all four low-pass filtered signals gave an increase in localization errors. This was not the case with the high-pass filtered signals.
Auditory motion aftereffect and its position-transfer characteristics
Authors: Dong C. J. , Swindale N. V. , Zakarauskas P. , Haywood V. , Cynader M. S.
Published: July 2002

To date, there have been only a few studies of the auditory motion aftereffect (aMAE). All of these studies used simulated moving sound as adapting or test stimuli. In the present study, real moving sound was used as both the adapting and test stimuli. The real moving sound was generated with a loudspeaker mounted on a robot arm which was able to move smoothly and quietly in three-dimensional space. Six subjects were tested in an anechoic chamber with white noise as both the adapting and test stimuli. Results from experiment 1 showed a clear aMAE in all subjects involved. The magnitude of the aMAE tended to increase up to the highest velocity tested (<30°/s). In experiment 2, the adapting and test stimuli were presented across two separate spatial ranges, and the magnitude of the aMAE was measured as a function of the distance between the two separate loci. Results show that when the adapting and test ranges are 20° apart, the aMAE is weak. However, when the two ranges abut, there is a noticeable aMAE, although its magnitude is smaller compared to that when adapting and test ranges overlap. The detailed space constant is under investigation.
Coupled wave-number integration approaches to range-dependent, seismoacoustic modeling
Authors: Schmidt Henrik
Published: July 2002

Wave-number integration (WI) approaches directly implement the ‘‘exact’’ spectral formulation for the seismoacoustic field in ocean waveguides. Therefore, these modeling approaches are inherently limited to range-independent, or horizontally stratified, environments. However, WI possesses a number of unique seismic capabilities, and various wave-field coupling approaches have recently been combined with WI to extend its applicability to two-way, seismoacoustic propagation in strongly range-dependent environments. These hybrid modeling approaches use a stepwise discretization of the environment similar to the one used in coupled-mode models. The spectral super-element approach [J. T. Goh and H. Schmidt, J. Acoust. Soc. Am. 100, 1409–1420 (1996)] then uses a Galerkin boundary-element formulation for the coupling between the range-independent sectors (super elements), while the virtual source approach [H. Schmidt, Proc. SWAC-97, Beijing (Apr. 1997)] uses an approximate, but highly efficient, local solution to the coupling problem. The performance of these new approaches has been demonstrated using a new class of two-way seismoacoustic benchmark problems with both weak and strong range dependence. Here, the new modeling approaches will be described, and their unique seismoacoustic capabilities will be discussed and illustrated by application to, e.g., the deep-ocean waveguide coupling of near-shore tomography sources, and analysis of potential seismoacoustic coupling mechanisms for oceanic T phases. [Work supported in part by the Office of Naval Research.]
High-definition temporal/frequency analysis of sampled data sets
Authors: Frye George J.
Published: July 2002

The FFT is a successful steady-state analysis tool, but falls short when temporal analysis is needed. A class of filters designed around a Gaussian windowed DFT is described that, when applied to signal data sets in a way similar to that of a wavelet analysis, yields high-definition, fractional octave bandwidth, temporal analysis. The filter set’s desired properties are temporal resolution that is inversely proportional to bandwidth and directly proportional to frequency, no assumption of signal periodicity, and the trade-off of accuracy against temporal window width. The main undesirable property is the computation needed to perform an analysis. Examples are chosen to show the need of a good display technique. They include analysis of a mathematical impulse, a hearing aid output as driven by a transient tone burst, and a room reflection with a broadband chirp signal. The author wishes to acknowledge Mike Day and Ilker Gungor of Frye Electronics for help in basic concepts and software programming.
On the modeling of sound propagation through capillary tubes with mean flow
Authors: Ih Jeong-Guon
Published: July 2002

In order to predict the transmission loss of catalytic converters in automotive exhaust systems, efforts have been devoted recently to wave propagation through a capillary tube with a steady flow. Several theoretical models also exist. In this paper, general linearized governing equations are discussed and the characteristic features of each model are shown in conjunction with methods for solving the equations. Characteristics of the forward and backward propagating acoustic waves are investigated under the influence of steady flow and the dependency on the shear wave numbers is checked. Additionally, the presence of hydrodynamic wave components is examined under the steady flow condition. The application limits of the concurrent acoustic models are discussed as the shear wave number varies. Several prediction methods are applied to the calculation of the transmission loss of a stack of monoliths with and without mean flow considering the effects of radial components.
Robert W. Young’s eminence in musical acoustics
Authors: Martin Daniel W.
Published: July 2002

Interest in musical sound led Robert W. Young into physics graduate study at the University of Washington, where his 1934 doctoral thesis reported on standing waves within the Boehm flute. During subsequent research at C. G. Conn Ltd., Young and co-inventor Hugo Schuck developed the Conn Chromatic Stroboscope. It was the most precise frequency measuring instrument for acoustical research in the audio frequency range. Young’s book of tables for instrument users, From Frequency to Cents [1939; 2nd ed. (1952)], related frequency to cents deviation from the equally tempered musical scale. Research in musical acoustics, musical instruments, and pitch perception was especially benefited for decades, before computers became available. Young’s 63 research publications in musical acoustics concerned tuning of wind instruments, vibration of piano strings, temperature and humidity effects upon timing, inharmonicity of string vibration modes, and the resulting stretched scale of piano tuning. He was the associate editor for musical acoustics for the Journal of the Acoustical Society of America for 45 years, exerting a positive, and often creative, influence on research by authors around the world. He also served as associate editor for the acoustical patent review section of this Journal from 1943 to 1977, and personally reviewed thousands of patents, many on musical sound.
Robert Young and acoustics during the War
Authors: Gales Robert S.
Published: July 2002

This paper will discuss Bob Young’s contributions to the field of acoustics through his work in San Diego, CA during World War II. The contributions will be highlighted by anecdotes from personal experiences.
Algorithm for BEM calculations to simulate sonic boom noise penetration into the ocean
Authors: Sparrow Victor W.
Published: July 2002

Robert W. Young has made many contributions to our understanding of how sound can get transmitted from air to water. One important contribution is Bob’s measurements of sonic boom noise penetration into the ocean from supersonic airplanes [J. Acoust. Soc. Am. 44, 392 (1968)]. Bob’s last words in this paper were ‘‘I would voice the hope that this experimental evidence will inspire some further theorizing to explain the observations.’’ Now almost 30 years after this statement, some of us are still theorizing. The present paper gives a quick overview of the state of knowledge in sonic boom penetration into the ocean and introduces a new boundary element method (BEM) approach to solving the problem of evanescent sound penetrating the ocean surface. The method is based on a mesh made from discretizing the surface of the air–water interface. Perhaps along with analytical formulations, finite difference simulations, and other experimental observations already available for determining the sonic boom noise penetration into the ocean, such new BEM simulations will help us answer the questions which Bob Young has posed. [Work supported by NASA Langley Research Center.]
Coherence of acoustic modes passing through internal solitary waves
Authors: Preisig James C. , Duda Timothy F.
Published: July 2002

Packets of internal solitary waves can affect coherence between acoustic modes propagating in shallow water. Shallow water is characterized as having less than 10 or 15 propagating modes, including the influence of an attenuating bottom. Previous work has shown that individual internal waves or packets of waves can scatter low-mode energy into higher modes. Prior results and intuition indicate that the higher modes attenuate more quickly through boundary interaction. The propagation of the internal waves, which couple energy between the modes, can rapidly influence the relative amplitudes and phases of the modes. Numerical simulations are used here to quantify the effects which the relative positions of the source, receiver, and wave packet (or packets) have on the mode coherence. Of particular interest are packets of solitary waves near the source, which can amplify the source. Frequencies of a few hundred Hz and propagation over tens of km are investigated. The results are compared with experimental data. [Work supported by ONR Ocean Acoustics.]
Flux-transfer amplitudes and the one-way coupled normal mode procedure
Authors: Werby Michael F. , Sidorovskaia Natalia A.
Published: July 2002

In a range-dependent environment, it is of interest to determine the way in which energy or flux proceeds in range. In the normal mode para- digm of propagation, it is possible to trace the energy flow according to specific modes. For a stratified environment there is no mechanism to alter the energy in each mode. It remains the same as long as the environment is range independent. However, for the range-dependent case, for which the coupled mode theory is adequate, it is possible to derive ‘‘coupled mode amplitudes.’’ The amplitude structure may then be studied in range to determine the manner in which energy traverses in the waveguide. This work illustrates what is meant by mode coupling amplitudes and the interpretative value in which the variation of these quantities in range adds insight in understanding a variety of mechanisms that alter or cause energy to proceed in a specific manner. [Work sponsored by ONR, the Naval Research Laboratory, and the University of New Orleans.]
The ecological approach toward ultrasonic perception
Authors: Peremans Herbert , Hallam John
Published: July 2002

Navigation seems to automatically entail the construction and use of environment models. In this paper it is shown how, in certain situations, the information provided by an echolocation system contains invariants that can be used directly to control the navigation behavior of an agent without the need for an explicit environment model. In particular, the case of an echolocating agent moving through an opening between two reflecting objects is analyzed. The transmit signal, duration 2 ms, is a frequency modulated sinusoid, hyperbolic frequency sweep from 100 down to 20 kHz. The processing performed on the raw measurement results, i.e., the signals picked up by the microphones, is modeled on the peripheral auditory processing of the mammal brain. The output of the model is an approximation of the short-time amplitude spectrum of the received signal. It will be shown that the control of the agent can be expressed solely in terms of invariant properties of the patterns appearing on the outputs of this model. This approach not only predicts the flight path of bats flying between vertical wires but it also significantly simplifies the control of a mobile robot moving between obstacles as will be shown.
The effects of aging and rate on tongue shape variability and on preservation of position in glides
Authors: Jaberi Maryam , Stone Maureen , Gray Suzanne , Slawinski Elsbieta
Published: July 2002

Slowing of neural processes due to aging may cause effects in motor speech production. In a study of /w/, Slawinski [E. B. Slawinski, J. Acoust. Soc. Am. 95, 2221–2230 (1994)] found a slower prevoicing stage, and a smaller off-glide frequency range for older speakers than younger. The latter finding suggests that older speakers centralize their articulatory gestures. Articulatory gestures may also centralize at faster speech rates. The present study compared midsagittal tongue gestures for /w/, for older versus younger subjects, matched for rate of speech. Ten female subjects (five aged 21–29, five aged 64–71) repeated: /■wVp/ and /■gwVp/ using the vowels /i,a,u/, five times each, while ultrasound video recordings were made. The hypothesis was that coarticulationwould cause different /w/ positions, (higher in /g/ context). If older subjects centralize tongue position for /w/, they might produce an even greater difference, if the high /g/ tongue position facilitated a more extreme /w/. Target frames were digitized and tongue surface edges were extracted [M. Unser and M. Stone, J. Acoust. Soc. Am. 91, 3001−3007 (1992)]. The distances between /w/ and /(g)w/ were determined using L2 norms. For two older and two younger speakers, rate effects were observed and age effects were not.
A dynamic neural network model of speech production in the developing child
Authors: Callan Daniel E. , Kent Ray D. , Vorperian Houri K.
Published: July 2002

Many theories of speech processing propose the existence of motor control systems that utilize invariant vocal tract configurations to specify particular goals of speech production. One problem that challenges these theories is the fact that the associated structures involved with speech production go through a considerable amount of change during development. The same speech goals continue to be achieved during the course of development despite the changes that the vocal tract configuration undergoes. In this paper, speech production in the developing child is modeled by a dynamic neural network that incorporates value-dependent learning based on self-produced auditory stimulation across self-organizing sensorimotor maps. The conversion from articulatory configuration to acoustic signal is worked out by a modified version of the Maeda articulatory model that utilizes developmental parameters. The performance of the neural network was assessed at different points during development by determining the articulation-acoustic output needed to produce various segments of speech in relation to the corresponding trajectory of activation across the neural network. The results are discussed in relation to a dynamic theory of speech processing that can account for developmental change as well as some speech disorders. [Work supported by NIDCD.]
On-line measurement of error-path impulse response for an adaptive ANR headset
Authors: Pan George J. , Brammer Anthony J.
Published: July 2002

A digital, adaptive ANR headset system has been developed and tested in the laboratory, using a filtered-X LMS adaptive control algorithm and a floating-point digital signal processor (DSP). The implementation of the filtered-X LMS adaptation algorithm requires an impulse response (IR) model of the so-called ‘‘error-path,’’ which consists of the serial connection of an earphone, earcup acoustics, microphone, and electronic amplifiers and filters. Though the IR model can be measured off-line for laboratory studies using standard measurement procedure, on-line measurement or identification, using the DSP engine and digital signal processing techniques, is considered necessary and advantageous for practical applications. This paper discusses a DSP-based identification technique for the error-path IR model, which employs the well-known pseudorandom binary signal (PRBS), also called the maximum length sequence (MLS) signal, as the input to the error path. In this paper, the measurement method and system will be discussed. As an example, the error-path impulse response and the corresponding ANR results obtained in a laboratory setup will be presented. [Work supported by DCIEM, Toronto, Canada.]
Measurements of normal incidence surface impedance using the single transfer function method with multitone periodic random noise signal
Authors: Olny Xavier , Sgard Franck C.
Published: July 2002

Commonly used signals for determining the transfer function of a linear filter are pure tone excitation, white noise, or a pseudorandom noise (MLS) signal. However, it is known that the accuracy of this measurement depends strongly on the kind of signal used. In order to try to improve the quality of the measurement, multitone periodic random noise signal properties are investigated in the case of the normal incidence surface impedance measurement. Kundt’s tubes and free-field impedance results using a single microphone transfer function method are given in a large frequency range (20–4000 Hz) for a multitone periodic random noise signal. These results are compared with those obtained with pure tone, white noise, and MLS signal excitation. Also, the influence of noise immunity is investigated for each type of signal.
Comparisons of computed and exact time-reversing array retrofocusing in a shallow-water sound channel
Authors: Dungan Michael R. , Dowling David R.
Published: July 2002

A time-reversing array (TRA) can retrofocus acoustic energy to the location of its original source even when the acoustic environment is complex and unknown. However, predictions of array retrofocusing performance require an acoustic propagation model for the environment. Results are presented from a study of the sensitivity of TRA retrofocusing to acoustic propagation modeling. Results from a modal sum propagation model and a parabolic equation (PE) code (RAM by M. D. Collins) are compared for a perfect waveguide. The influence of waveguide complexity on retrofocusing is determined by adding range dependence and realistic bottom properties to the computed PE fields. Variations in acoustic frequency and sound source depth are also explored. It is found that acoustic penetration into the bottom primarily influences sound pressure levels but does not strongly influence the size of the TRA retrofocus. In addition, small changes in the waveguide’s range dependence are found to have little effect on retrofocus properties. [Work supported by the Office of Naval Research.]
Variable range focusing in a time-reversal mirror
Authors: Song Hee Chun , Kuperman William A. , Hodgkiss William S. , Akal T. , Ferla C. , Jackson Darrell R.
Published: July 2002

Recently, a time-reversal mirror (or phase conjugate array) was demonstrated experimentally in the ocean, which spatially and temporally re-focused an incident acoustic field back to the original position of the probe source. [Kuperman et al., accepted for publication in J. Acoust. Soc. Am.]. Here, this waveguide time-reversal mirror technique is extended to refocus at ranges other than that of the probe source. This procedure is based on the acoustic-field invariant property in the coordinates of frequency and range in an oceanic waveguide [G. A. Grachev, Acoust. Phys. 39(1), 33–35 (1993)]. This technique has been verified in a second phase conjugate experiment conducted in the Mediterranean Sea in May 1997. The theory behind variable range focusing will be presented along with experimental results demonstrating the ability to shift the focal range ±1 km for a probe source in the vicinity of 15 km from the source–receive array.
Paul S. Veneklasen, contributions to electroacoustic system and design
Authors: Ortega Jose C.
Published: July 2002

Paul Veneklasen’s interest in acoustics was wide ranging. Research and development in electroacoustics occupied his endeavors in the early years. This paper will present a survey of his work. Topics to be presented include development of a series of condenser microphones and loud- speaker systems at the Altec Lansing Corporation, and development of instrumentation to measure the sound pressure level using the Western Electric 640 AA condenser microphone. Projects of a more unusual nature included the development of a Fabric Acoustimeter or ‘‘Fuzz Feeler’’ to measure by acoustic means the ‘‘handle’’ of a fabric, development of the tooth and forehead microphones for detecting and transmitting voice signals in very high-level noise enviroments and research and development of the electronic stethoscope. This instrument extended the frequency range of heart sound analysis. Paul’s interest and pioneering work in auditorium acoustics led to contributions to the development of instrumentation for use in acoustical scale modeling research.
Effects of fixed-wing military aircraft noise on California gnatcatcher reproduction
Authors: Awbrey Frank T. , Hunsaker Don
Published: July 2002

To test the assumption that high levels of aircraft noise impede bird reproduction, noise analyzers were placed for 1 week in the nesting territory of each of 39 California gnatcatcher pairs on Naval Air Station Miramar. The 1-week average sound levels (7DL) recorded in those nesting territories were then related to the number of nest attempts; number of eggs laid; number of chicks hatched; number of chicks fledged; and number of eggs, chicks, and fledglings per nest attempt. Nest attempts and eggs laid have weak negative correlations (p=0.14 and 0.28) with 7DL. That is, the birds may tend to build fewer nests and lay fewer eggs in noisier areas, which is consistent with the common observation that bird nesting is more easily disturbed before eggs are laid than after. None of the other indicators is correlated with sound levels. Once a nest is established, with eggs in it, military aircraft noise has no detectable influence on reproductive performance. Gnatcatchers reproduced in places where 1 HL exceeds 80 dB for several hours every day. If fixed-wing aircraft noise impedes California gnatcatcher reproduction, it is overwhelmed by such factors as disturbance, predation, weather, edge effects, and differences in quality of habitat.
Acoustic interaction of humpback whales and whale-watching boats off Maui, Hawaii
Authors: Au Whitlow W. L. , Green Marsha
Published: July 2002

Whale-watching activities during the humpback whale annual winter sojourn in Hawaii have increased steadily over the past years. A concern of many is the possible detrimental effects of whale-watching boats on humpback whales; including the possibility of damage to their auditory system. To address this concern, the noise generated by five different types of whale-watching boats was measured, two with outboard engines, two with inboard diesel engines, and a seagoing vessel, in the near shore waters of west Maui. The first series of measurements was conducted at the peak of the whale season, and the boat noises were partially masked by humpback whales’ chorusing (boat noises and whale chorus levels were on the same order of magnitude). A 1/3 octave band spectral analysis indicated whale chorus levels as high as 136 dB re: 1 μPa with no whales in sight. Throughout the day humpback whale chorusing noise was present at similar levels for miles along the same coastline. Measurements taken close to the end of the season had about a 15-dB decrease in the humpback whale chorusing levels. Considering the high level of chorusing by humpback whales, the acoustic effects of the whale-watching boats should be relatively small or negligible.
Representing groove: Rhythmic structure in interactive music performance
Authors: Iyer Vijay , Bilmes Jeff , Wright Matt , Wessel David
Published: July 2002

The concept of ‘‘groove,’’ a musical element particularly operative in African-derived musics such as jazz, may be described as a collectively determined and relatively isochronous pulse that can be inferred from the interaction of a number of interlocking rhythmic groups. A powerful representation for musical rhythm as it appears in groove-oriented contexts has been developed. Implemented in the MAX music-programming environment, the representation encompasses pitch, accent, rhythmic deviations, tempo variation, note durations, and probabilistic processes. The design facilitates ‘‘bottom-up’’ combination techniques such as the construction of large musical objects by assembling small musical ‘‘cells’’ in series or parallel, and thus it allows improvisatory manipulation of simple rhythmic structures. The richness of control over many meaningful musical quantities distinguishes the representation from those in more common usage, such as music notation programs, sequencers, and drum machines. The implementation supports a variety of creative applications in improvised performance. Rather than attempting to make the system ‘‘sound human’’ by forcing it to simulate specific human activities, two alternative approaches are taken: the development of novel, sensitive control structures, and/or the emphasis on the system’s active role as a situated participant in an interactive musical environment.
A novice’s interface to programming digital synthesizers based on genetic algorithms
Authors: Sandnes Frode E. , Robertson Craig
Published: July 2002

The last decade has seen vast advances in commercial synthesizer developments, both in cost and features. Programming of synthesizers requires patience and detailed knowledge about sound synthesis. Also, the low-cost interfaces to these units are time consuming to access compared to the sliders of analogue synthesizers. Many musicians resort to costly off-the-shelf preprogrammed sound libraries, often without the actual characteristics and textures required. Therefore, a novel approach to programming of digital synthesizers is proposed. The task of programming is defined as an optimization problem, where the optimization function is the musician’s rating of the sound, and the function arguments are the actual parameters of the sound synthesis algorithm, such as ADSR, LFO’s, etc. These parameters are coded as chromosomes, and a genetic algorithm is used to search for the desired sounds. The interface works by presenting an initially random palette of sounds to the musician, for him/her to evaluate. Based on these evaluations, the genetic algorithm applies standard genetic operators to the chromosomes and over time breeds the desired sounds. Genetic algorithms ensure rapid convergence. This makes the technical details of a synthesizer transparent to the musician while it still allows him/her to explore all of its advanced features.
Numerical research of the influence of external conditions on the sonoluminescence process
Authors: Nogin Vladimir N. , Karlykhanov Nikolay G. , Kovalenko Gennady V. , Simonenko Vadim A. , Moss William C.
Published: July 2002

Using a 1-D approximation, processes are numerically simulated that take place during oscillations of a single gas bubble located in the center of a spherical flask with liquid exposed to a periodical acoustic wave. Real equations of state for water, gases (air, argon), surface tension, and heat conduction (molecular and radiative) are taken into account in the simulation. The role of shock-wave processes at bubble compression is investigated. Because of the peculiarities in the behavior of the gas, the equation of state is shown to be the most important at densities greater than 1 g/cm3, with the parameters of bubble compression being weakly affected by the water equation of state. Time dependencies of the bubble radius during one acoustic cycle are obtained for different ambient conditions: acoustic-wave amplitude and frequency, liquid temperature, and magnitude of surface tension. Comparison with experimental data is performed. [Work supported by CRDF.]
Auditory processing is related to reading ability
Authors: Protopapas Athanassios , Ahissar Merav , Merzenich Michael M.
Published: July 2002

Aspects of auditory processing related to temporal and spectral resolution were investigated in 50 adult subjects (aged 18-58) with varying reading abilities. The tests administered included tone detection, tone frequency discrimination, tone sequencing, interval discrimination, and gap detection. Tones were either long (250 ms) or short (20 ms) and ranged in frequency between 600 and 1400 Hz. Detection and frequency discrimination tasks were given in the clean and in a masking context, in which tones were followed by 300-ms bandpass noise. Thresholds were determined using an adaptive procedure. Backward masking interferences for short tones, as well as interval and frequency discrimination limens were substantially elevated for most poor readers, and were correlated with one another. Three-tone sequencing and short-tone frequency discrimination in a masking context were especially strongly correlated with single-word reading ability. Gap detection thresholds did not correlate with other tasks or with reading ability. In sum, spectral-temporal auditory processing resolution appears to be related to reading ability, possibly via an acoustically based deficit in phonetic development. These findings raise the question of whether nonspeech acoustic training can be used to improve adults’ reading ability. [Work supported by Scientific Learning Corporation.]
Chairs’ introduction to special session on phonetic perception and word recognition
Authors: Auer Edward T. , Bernstein Lynne E.
Published: July 2002

A classic statement of the speech perception problem is how to account for the perception of phonemes based on the acoustic stimulus. A classic statement of the word recognition problem is how to account for the access and selection of lexical representations, with little concern for phoneme/phonetic perception. Over the past two decades, research in spoken-word recognition has begun to explicitly model the stages of word recognition in terms of phonetic/phonemic perception and later stages involving lexical knowledge. These different models incorporate very specific proposals for the interaction of information from the bottom-up stimulus and the lexicon. They also raise issues concerning the appropriate model for phoneme perception, that is, for example, does it precede or follow the lexical processing stage? It would be convenient if theories/models of phoneme perception could ignore effects of the lexicon, and therefore, autonomously account for the ‘‘front-end’’ of word recognition. But can they, and be adequate theories/models of either phoneme perception or the front-end to word recognition? Alternatively, are there phoneme identification effects that are irrelevant to word recognition? In this session, researchers representing a variety of perspectives on the true relationship between lexical processes and phoneme/phonetic perception are brought together.
Artificial neural network processing of time-varying loudness data to model annoyance of noise
Authors: Laux Peter C. , Davies Patricia
Published: July 2002

In this talk, a procedure is presented for the development of an artificial neural network model of human annoyance to noise. This model development is based on using available sound quality metrics (loudness, sharpness, fluctuation strength, and roughness) that were calculated using post-process temporal filtering and numerical formulation operations on measured time-varying critical band loudness data obtained from a Bruel and Kjaer loudness analyzer. The artificial neural network (ANN) was initially developed to model Zwicker’s equation for unbiased annoyance (UBA). Subsequent training of the artificial neural network was done by using a series of 700 noise signals that were evaluated for their annoyance by a pool of normal-hearing young adult U.S. born subjects in a parameter estimation subjective test procedure. The noise stimuli were varied in seven different properties, which included: peak loudness, modulation frequency, modulation envelope shape, modulation depth, frequency band of modulated noise, addition of continuous complex tone structures, and the relative level of complex tone set to the underlying noise. The response data were averaged across subjects and the UBA-ANN model was retrained using the average subjective responses. The results of this new ANN–annoyance model are compared to other metrics for sound measurement/rating.