bluuebunny/crossref_metadata_2025 huggingface dataset:
Preliminary results from at-sea tests of the Autonomous Buoyed Environmental Sensor System (ABES)
Authors: Stevenson J. Mark , Newton John L.
Published: July 2002

ABES is an autonomous oceanographic measurement system which provides relatively dense spatial and temporal sampling of the highly variable shallow-water ocean environment. Compared to more traditional sampling methods, ABES is low cost; albeit providing slightly lower measurement accuracy and resolution. ABES is not intended to replace existing high-resolution oceanographic instrumentation, but fill the niche for a low-cost, lower-resolution system that would provide higher spatial sampling. The instrument is a sensor string connected to a cylindrical electronics canister. The prototype sensor string contains one electromagnetic current meter, three tilt sensors, three magnetoresistive compasses, and ten conductivity, temperature, and depth sensors. These sensors are distributed along a 40-m cable. A system controller, GPS receiver, and spread-spectrum radio are housed in the 1-m-long electronics canister. Together, these components weigh 10 kg in air and occupy a volume of 5 l. Data collected during three at-sea tests in 1998 and compared to traditional CTD measurements suggest that ABES can effectively define the temporal and spatial characteristics of the shallow-water environment. [Work supported by ONR 321-SS and 322-PO.]
Bottom reflection phase inversion using modal dispersion curves
Authors: Cox B. T. , Joseph P.
Published: July 2002

A technique to obtain an estimate of the seabed reflection coefficient phase from modal dispersion curves is presented. The technique is demonstrated on experimental data with a frequency range of 4 to 100 kHz in a 10-cm-deep water waveguide. Thirteen modal dispersion curves, defined as the pressure amplitude as a function of frequency and horizontal wave number, were obtained by measuring the acoustic pressure as a function of range and performing a Hankel transform. Initial attempts have produced estimates of the phase that are in close agreement with an independent estimate obtained by measuring the seabed wave speeds and density. [Work sponsored by EPSRC and DERA (Winfrith).]
The acoustical design of a new open air theater in Thessaloniki, Greece
Authors: Tzekakis Emmanuel G. , Schubert Gottfried
Published: July 2002

The international experience in the design of open air theaters is limited. Most new theaters are used exclusively with electroacoustic installations, because they are very large or in very noisy surroundings. As a result, no new architectural or acoustical breakthroughs are available. The new theater reported in this paper is a 5000 seat open air theater, with a shape derived from the ancient Greek prototypes, optimized for best listening conditions without electroacoustic installations. The background noise level reduction and the geometry of the theater were the main concerns of the acoustical design. Modeling tools were used to predict the final acoustical conditions. Measurements made in the almost ready theater indicate that using known and reliable prototypes in a careful way results in high quality acoustical conditions.
Restoration by means of sound reinforcement of a church of the Renaissance
Authors: Romero-Faus Jose , Gimenez-Perez Alicia , Marin-Sanchis Albert , Sanchis-Sabater Antonio
Published: July 2002

This work summarizes the project of the acoustic restoration of a Church of the Renaissance, XV Century, after the architectural restoration carried out in it throughout the last 10 years, with volume, 9000 m3; total surface, 3000 m2; room constant, R=1110 m2; reverberation time to 500 Hz, 1.8 s; inteligibility from the altar to the last banks, 0.34; loss of sound level, 7 dB. Since the acoustic response for the music and the song was good, the one which should improve above all was the intelligibility. Specifically, this project was undertaken to provide a uniform sound coverage to all the hearing zone, an adequate sound level, the nonexistence of perceptible late reflections, and the adequate intelligibility characteristic of the room. Bearing in mind, furthermore, that the acoustics of the enclosure could not be improved by means of acoustic materials, the adopted solution was to use an electroacoustic system, putting a centralized cluster of loudspeakers over the altar, to 8 m of height, that assured the sufficient sound reinforcement. This solution improved the intelligibility, which became 0.53, as well as the coverage field. These results have been validated by means of a ray tracing program.
Listener perception of spatial impression and auditory source width and their influence on overall impression in real rooms and headphone studies
Authors: Gold Martin A.
Published: July 2002

Comparisons of listener evaluations of music in real room situations and in headphone listening studies were used to assess the contribution of reverberance, loudness, auditory source width (ASW), and spatial impression on the overall impression of music listening and to evaluate their role as independent acoustic variables. Room studies were conducted using large numbers of average listeners at designated locations with a live sound source. Headphone listening studies were conducted using anechoic recordings with simulated acoustic environments (early reflections and reverberation) having different configurations of early and late sound energy. Directional components of the early and late energy were independently modeled as part of the experiment (Gold, 1994). Regression models were conducted to assess the interrelationship of independent variables on listener overall impression. In many cases, independent elements could be identified as contributing to overall impression; however, these elements were not the same in each case. For example, in cases where reverberance is not present, loudness and ASW become greater contributors. In addition, an attempt was made to find out if there is a listener distinction between ASW (a widening of the source sound) and spatial impression (the quality of being immersed in the sound).
Controls of a modulated delay time of reflection enhancing subjective preference
Authors: Atagi Junko , Ando Yoichi , Ueda Yasutaka
Published: July 2002

Previously, the most preferred delay time of reflection for time-invariant sound fields corresponds to the effective duration of the autocorrelation function (τe) of source signals [Ando, Concert Hall Acoustics (Springer-Verlag, New York, 1985)]. However, the SPL fluctuation in a real room is fluctuated and the model of such a time-variant system is proposed by the direct sound and the varying delay time of reflection [Ueda and Ando, J. Acoust. Soc. Am. 102, 2771–2775 (1997)]. To realize more excellent sound fields than time-invariant sound fields, a method of controlling a modulation of the delay time of reflection is discussed. Experiments are performed on the modulation methods to enhance subjective preference in a sound field. The following methods are examined to control the delay time of reflection in the experiment: (1) sinusoidally modulated delay time, (2) random modulated delay time, (3) delay time corresponding to the change of the running τe of the source signal, and (4) fixed delay time of reflection at the minimum value of the running τes.
On interfacing physics with electrophysiology in echolocation in bats
Authors: Raichel Daniel R. , Palakal Mathew J.
Published: July 2002

In animal echolocation, reflected signals carry far more information than is generally realized. The real, nonlinear nature of sound propagation and the sensitivity of reflected sound to the effects of the target and its environs provide a matrix M representative of both probe and reflected signals that lends itself to real-time spectral processing and formation of acoustical images. This process necessitates a sensitivity to signals at least two or more octaves beyond the fundamentals of the probe signals, combined with a high degree of resolution in the frequency, signal strength, and phase domains. The information feeds into the neurological system at the rate of dM/dt, enabling neurological processing of the high-frequency downward pulses ranging from 20 to 100 kHz used by echolocation bats as well as pure FM pulses and CF-FM combination pulses used by certain species of bats. The multi-dimensional matrix also accommodates the duration and repetition rates of these pulses which are continuously altered according to the perceptual demand during echolocation. Electro-physiological data suggests that the specialized delay-sensitive neurons that are present in various regions of the auditory pathway allow the animal to obtain fine delay acuity and target discrimination ability in cluttered environments, with nanosecond resolution.
Comparing predictions from backscatter models to in situ measurements of a dual-chambered, swimbladdered fish
Authors: Horne John K. , Jech J. Michael , Walline Paul D.
Published: July 2002

Most acoustic backscatter models predict echo amplitudes of single or aggregations of fish with single-chambered swimbladders. Lavnun (Acanthobrama terraesanctae), a planktivorous fish in Lake Kinneret, Israel, possesses a dual-chambered swimbladder. Target strength, length, frequency, or aspect relationships do not exist for this commercially important species. What is the most appropriate way to model backscatter amplitudes from single and aggregations of fish with multi-chambered swimbladders? Carrier frequency, organism length, organism aspect, and swimbladder shape all influence amplitudes of returned echoes. Predicted backscatter amplitudes and intra-species variance predictions from an anatomically-based Kirchhoff-ray mode model were compared to in situ measures of lavnun backscatter at 120 and 420 kHz. The lavnun swimbladder was modeled as a single chamber (with equivalent volume of both chambers), as two independent chambers, and as two acoustically interacting chambers. Wider application and comparison of theoretical models to field measurements increases understanding of factors influencing backscatter amplitude and variability for all fish species, and should improve methods for identifying organisms. [Work supported by ONR and US–Israel BSF.]
Simulation of ultrasonic fields in composite media with planar boundaries
Authors: Mažeika Liudas , Kažys Rymantas J.
Published: July 2002

In this paper the method and fast algorithm for simulation of ultrasonic fields exited by the direct or angular transducers in media with a few thick layers are presented. The method exploits regularities of the disk-shaped transducer field in a homogeneous medium, adapting it to a multilayered medium, the boundaries of which are not necessarily parallel. It was assumed that at any point the pulse response of the transducer must be of the similar shape, consisting of the fragments corresponding to the plane and the edge waves. The additional parameters of the pulse response are evaluated solving a system of nonlinear equations. These parameters are used to find some equivalent point in pulse response for a homogeneous medium with adequate values of parameters. The method proposed allows the evaluation of ultrasonic field structure in the solids when the immersion technique is used or inside tanks when ultrasonic waves pass through a wall. The simulation and experimental results presented show a good correspondence with the field structure calculated using other methods.
Wideband ultrasonic piezoelectric microphones
Authors: Augutis Vygantas , Gailius Darius
Published: July 2002

Microphones with an air coupling multiple matching layer and fiber damper were developed. The method for calculation of the characteristics of the transducers is presented. The signal-to-noise ratio (SNR) of these microphones is about 10 dB lower than the SNR ratio of capacitive microphones, but they are more suitable for use in dusty and wet conditions. The frequency range of these microphones is 10–100 kHz. The system for the measurement of the characteristic of microphones was developed. Short acoustic pulses, generated by the electical spark, were used as a wideband probing acoustic signal. Serious attention was paid to the estimation of instability of the spark-generated acoustic pulses. Several modifications of developed microphones, their characteristics, and fields of application are presented.
A simple device for flow resistance measurements
Authors: Holmberg Dag , Nilsson Erling
Published: July 2002

A simple device for static measurements of flow resistance on thin porous materials is developed and evaluated. The device consists of a duct with a piston which is driven at a steady low speed. This generates a flow of air with constant volume velocity. The duct ends with a cavity with an interface to the outside where the sample is mounted. The differential pressure between the cavity and the outside is measured. In order to prevent nonlinear effects, the speed of the piston is kept so low that the constant velocity of the air is of the same order of magnitude at the sample as the rms velocity of sound waves at approximately 100 dB SPL. The device is so constructed that it is possible to measure samples of standard dimensions, e.g., cloth mounted at different perforated frames, if the speed of the moving piston is accordingly adjusted to match the open area. The performance of the device is tested and discussed.
Musical acoustics, the music student, and the music teacher
Authors: Guettler Knut
Published: July 2002

Dialogs between musicians and acousticians are not always successful. A lack of common language combined with mutual suspicion of the other person trespassing on ones own professional authority, often restricts what ought to have been a fruitful exchange of knowledge. For the music student, there is, however, a third matter that seriously impedes the possibility of gaining acoustical insight. As for all arts, yearning and intuition are the driving forces in the development of musical and instrumental skills. Focus on purely technical matters, or even worse, the feeling of being overruled by such, is hardly combinable with a desirable musical progress of the young student. Based on the author’s experience as a principal professional player and teacher of music for more than 30 years, this paper gives some suggestions on which and how topics in musical acoustics may be introduced.
Combined effects of noise and whole body vibration in the working environment
Authors: Schust Marianne , Seidel Helmut
Published: July 2002

The simultaneous exposure to noise and whole-body vibration (WBV) is the most frequent combination of physical factors at workplaces. Both factors probably act not-independently of each other. The results of the studies regarding the aural effects indicate that additional exposure to WBV amplifies the temporary threshold shift as well as of the permanent threshold shift. Various parameters, i.e., postural sway, vision, and cardiovascular changes, were investigated in experimental studies to clarify the extraaural physiological effects. The results demonstrate a dependence on the intensity and the frequency spectrum of the noise as well as of the WBV. Mutually amplifying interactions of noise and WBV were observed. The events of experimental research regarding the subjective assessment of exposure conditions suggest the following conclusions: an increase of noise intensity causes stronger sensations of the WBV intensity except for WBV with main frequencies around 2.5 Hz. Changes of WBV intensity have no effect on the subjectively experienced loudness and noise-related annoyance. Compound judgments of the entire situation are influenced by an interaction of noise and WBV. The results are discussed to derive proposals for guidelines considering combined effects.
Assessment and remediation of perceived handicap in subjects with acquired hearing impairment
Authors: Hallberg Lillemor R-M.
Published: July 2002

In a treatment-comparison group design, a group-rehabilitation program developed for hearing-impaired individuals and their close relatives was evaluated. The majority of subjects had hearing loss from noise exposure. The aim of the program was to give adequate information, psychosocial support, and training for effective coping strategies. Sixty-six consecutive patients at the hearing clinic were included in a pilot study. Standardized questionnaires, intended to measure ‘‘perceived handicap,’’ ‘‘social support,’’ ‘‘perceived negative attitudes,’’ ‘‘acceptance of hearing loss,’’ and ‘‘communication strategies’’ (maladaptive behavior, verbal strategies, and nonverbal strategies), were used at the end of the last group session (short-term evaluation) and with postal questionnaires four months later (long-term evaluation). The evaluation showed that the use of maladaptive behavior (e.g., guessing, pretending to hear, and avoiding interactions) was significantly decreased in the treatment group in a short-term as well as in a long-term perspective (p=0.000 and p=0.032, respectively). The use of verbal strategies was significantly increased in the treatment group after the rehabilitation (p=0.020), but this increase did not reach a significant level four months later. Self-reported handicap was decreased in some conditions. Presently, data from ongoing rehabilitation groups are assessed and statistical data analysis will follow.
Common problems in noise control projects
Authors: Beristain Sergio
Published: July 2002

In actual industrial noise control, it is common to face obstacles before being able to analyze the noise, in addition to others arising in the process of analysis and designing the proper solution: usually it is not a new installation, which produces big savings in the future, but noise due to bad installations or irregular growth of industrial plants. This paper discusses the most common problems found in practice, and presents some ideas for facing some of them. Among them are economics, legislation, lack of knowledge or interest, lack of trained personnel for making the installation, etc., and the noise control project itself, which indeed is very important. This means that the noise control specialist should be ready to face conditions fully outside his or her expertise, but completely related to the problem to be solved. There are some additional aspects not related with technique, standards, or economics, which are opposed to the implementation of the designed solution, like culture, education or ‘‘the expectation that nobody will get deaf in the future if they did not get deaf today.’’ Knowledge of all of the above limitations is essential to avoid failure in noise control in new or rebuilt installations or in hearing conservation programs.
Auditory damage due to impulses or continuous noise
Authors: Fleischer Gerald
Published: July 2002

In order to determine the detrimental effect of continuous noise versus impulsive noise, a number of groups were examined: school children, university students, soldiers, workers, pilots, and amateur and professional musicians. Auditory performance was determined by means of pure tone audiometry up to 16 kHz. Auditory load throughout life was estimated, using a detailed questionnaire that included the aspects of profession, hobbies, sports, and consumption of music. Occurrence and severity of tinnitus were also determined. It can be shown that impulsive noise is more damaging to the ear than continuous noise of the same Leq. Analyzing the situation at the workplace, it can be concluded that the damaging effects of acoustic impulses are not properly measured by the Leq.
Measured and computed effect of meteorological condition and ground surface on propagation of impulsive sound to 20 km
Authors: White Michael J.
Published: July 2002

The cumulative propagation effects of weather and ground surface on sound exposure levels at long distances from an impulsive sound source are not well established. Measurements of sound exposure and peak pressure at distances up to about 20 km from explosions were made by the U.S. Army for three types of ground conditions. These data, along with measured weather profiles and ground impedance parameters, were used as inputs to the FFP model, for comparison of measurement to prediction. The 1/3-octave band attenuations were calculated for band center frequencies from 1 Hz to 1 kHz, for the measurement microphone locations in the study. A great deal of the variation in signal level from event to event was predicted by the FFP model, without resort to nonlinear propagation effects or to turbulent scattering. Groups of cases with similar signal attenuation with distance and similar weather profiles may provide a reasonable basis for classifying long-distance propagation conditions. Such a classification would have merit as an efficient way to assess long-term noise averages or to estimate the error in short-term predictions of noise levels. [Work supported by USACERL and AMSAA.]
A comparison of engineering methods for predicting ground effect
Authors: Attenborough Keith
Published: July 2002

Methods for calculating ground effect for broadband noise are reviewed and predictions of these methods are compared with recent data for propagation from a fixed jet engine and measurements over farmland in various states of cultivation. The review includes empirical, standard, and approximate analytical methods. The comparison leads to a discussion and evaluation of their relative merits. [Work supported by EPSRC and BBSRC (UK).]
Noise computation of a mixing layer using LES simulations
Authors: Bogey Christophe , Bailly Christophe , Juvé Daniel
Published: July 2002

A study of noise computation of a two-dimensional subsonic mixing layer is performed. The first concern is to develop a large eddy simulation (LES) code solving the filtered Navier–Stokes equations to obtain the nonlinear generation of sound. The compact DRP scheme of Tam and Webb is used in space, and the time integration is carried out by a four-step Runge–Kutta algorithm. The boundary conditions of Dong and Tam are implemented. Then, a two-dimensional mixing layer is simulated. The spatial growth, turbulence rates, and spectra are in good agreement with other available studies. The goal is now to use the LES data to calculate the acoustic field generated by the mixing layer. Various methodologies have been investigated. First, the direct exploitation of the LES pressure field reveals tremendous difficulties due to the different orders of magnitude between aerodynamic and acoustic fluctuations. The other approaches consist of using source terms calculated from the LES velocity fluctuations. Two formulations are tested: Lighthill’s analogy based on a volume integral over the region encompassing sound sources, and linearized Euler’s equations where a source term is added. Numerical results and future developments will be discussed.
Incremental responses in the auditory cortex and the auditory periphery
Authors: Smith Robert L. , Baptiste Dyanne , Lutkenhoner Bernd
Published: July 2002

When probe tones are added to background sounds, the perception or detectability of the probe is influenced by the time delay from the onset of the background to the onset of the probe. In general, the shorter the time delay, the harder it is to detect or perceive the probe as a separate signal. Possible physiological counterparts of these phenomena were investigated by adding 40-ms-long probe tones to 200-ms-long background tones. The corresponding increment in response was measured in the auditory nerve of anesthetized gerbils, using single-unit and modulation following responses, and in the auditory cortex of awake humans, using a 37-channel Magnetoencephalogram (MEG) and a single equivalent current-dipole model. The auditory nerve responses appear to obey superposition such that the increment in response is independent of the time delay. In contrast, the increment in the MEG response was significantly smaller for small delays than for long delays. The results suggest that the cortical responses behave differently than the peripheral responses, and more closely resemble those obtained psychophysically, although differences in experimental paradigms may also affect comparisons. Hence the difference between peripheral and central effects may reflect a functionally significant change in temporal processing occurring along the auditory pathway.
A modal synthesis approach to investigate the influence of viscoelastic boundary conditions on the sound radiated from a plate backed cavity
Authors: Chiello Olivier , Sgard Franck C. , Atalla Noureddine
Published: July 2002

In this paper an efficient formulation is proposed to calculate the sound radiation from a plate with edges viscoelastically restrained against translation and rotation and coupled to a cavity. The viscoelastic constants can be varied arbitrarily along the plate boundary and reproduce simply supported, clamped, free or guided edges as limiting cases. The formulation classically couples a modal expansion for the structure in vacuo and the cavity, using a coordinate transformation to calculate the fluid-structure coupled modes. The modal basis associated with the structure in vacuo is determined using a free component mode synthesis technique considering the viscoelastic parameters as components with a residual flexibility only. This approach is appropriate to compute the coupled modes efficiently and to perform a parameter study. Numerical examples are presented in the case of a plate excited either mechanically or acoustically. The aim of the study is to determine which boundary parameters have an influence on the acoustic behavior of plates and to develop tools for the pratical design of plate mounting conditions.
The structural acoustic optimization of an unstiffened cylinder finite-element model
Authors: De Rosa Sergio , Franco Francesco , Fontana Mauro , Paino Romualdo , Sollo Antonio
Published: July 2002

The presentation will show some results of an ongoing research, aimed to the tuning of a structural-acoustic optimization procedure. The target is the definition of a design tool in which it will be possible to include, at the same time, the weight and noise limits, together with the stress constraints, for getting reliable aircraft fuselage models. All the new capabilities included in the actual version of the MSC/NASTRAN have been tested. They allow analyzing the structural dynamics, the coupled acoustic response, to evaluate the structural-acoustic sensitivity and the global optimization process. Several similar attempts have been reported in the specific literature but, typically, different solvers have been used for each part of the problem (the structural degrees of freedom, the acoustic ones, and the optimization algorithm). Here, all results concern the application of the same finite-element model for a simplified (unstiffened) cylinder. The thicknesses of the plate elements have been grouped longitudinally and circumferentially, so that it has been possible to evaluate the predictive trends for fixed excitation frequency. Several optimization strategies have also been compared and discussed. They have also verified the optimized structural-acoustic models in output by evaluating the structural-acoustic geometric and dynamic coupling coefficients.
VCV synthesis from muscle commands
Authors: Pascal Perrier , Payan Yohan , Perkell Joseph , Zandipour Majid , Matthies Melanie
Published: July 2002

The biomechanical model of the tongue developed by Payan and Perrier [Speech Commun. 22, 185–205 (1997)] is used to generate tongue movements in VCV sequences, where V is [i], [a], or [u] and C is the stop consonant [k]. Tissues’ elastic properties are accounted for in finite-element modeling, and the mechanisms underlying the production of muscle forces are modeled according to Feldman’s Equilibrium Point Hypothesis. Each elementary sound is associated with a target that is described as a static equilibrium position of the tongue, which is likely to vary with the phonemic context. In vowel production, the target is actually supposed to be reached, while the production of stop consonants consists of movements toward virtual targets located beyond the palate, which therefore cannot be reached. The collision of the tongue against the palate is modeled according to a penalty method, based on a nonlinear relationship between contact force and position/velocity of points located on the tongue surface [Marhefka and Orin, IEEE Conf. Robotics and Automation, 1662–1668 (1996)]. The acoustic signal is generated with a one-dimensional low-frequency approximation based on the Kelly–Lochbaum model. Synthetic kinematic and acoustic signals are compared with human speakers’ data. [Work supported by CNRS, NSF, and NIH.]
The FUL speech recognition system
Authors: Lahiri Aditi , Reetz Henning
Published: July 2002

The featurally underspecified lexicon (FUL) speech recognition system is based on a lexicon where words are represented by radically underspecified phonological features. Features are extracted from the speech stream and are mapped with a ternary logic (match, no-mismatch, mismatch) to the entries in the 100000-word lexicon without building a segmental representation. The mapping process operates from left to right and allows for certain insertions and deletions of segments of the words in the lexicon. Mismatching features are deactivated from the search. Word candidates remaining after this selection undergo the parallel active prosodic and syntactic parsing of phrase hypothesis. The system is speaker independent and not sensitive to microphone or line conditions and does not need a high signal-to-noise ratio.
Resolution capability of reassigned Fourier spectrum
Authors: Yang Dekun , Ainsworth William A. , Meyer Georg F.
Published: July 2002

High-resolution spectral estimation is important for speech analysis. The conventional Fourier transform technique cannot offer sufficient resolution for speech signals of short durations. The reassignment method has been recently developed to improve the resolution of the Fourier spectrum. The basic idea of the reassignment method is to assign the value of the Fourier spectrum to the gravity center of the region rather than the geometric center of the region. However, the characteristics of the reassigned Fourier spectrum have not been fully understood. This paper is concerned with a detailed investigation of the resolution capability of the reassigned Fourier spectrum. The factors affecting resolution, such as frequency separation and relative amplitude, are studied. The minimal frequency separation which can be resolved in the reassigned Fourier spectrum is determined. As a result, it shows that the reassigned Fourier spectrum has better resolution capability in comparison to the Fourier spectrum. Furthermore, the study shows that the frequency estimator of closely spaced sinusoids has a bias. The bias is due to the interaction of the two sinusoids under the assignment operation. The expression of the bias is derived and shows that the bias can be ignored for most speech signals.
Query language for access to speech corpora
Authors: Mengel Andreas , Heid Ulrich
Published: July 2002

Typically, speech corpora are designed for the study of speech, its patterns, and their influences. Corpora for synthesis support the extraction and reuse of units in concatenative speech synthesis systems. Recent approaches consider speech segments of variable length, to enhance naturalness. The development of spoken language dialog systems (SLDSs) and similar applications requires dialog corpora with multiple information, such as prosodic labeling, grammatical annotations, dialog acts, etc. For constructing such multilevel corpora and, particularly, for retrieving useful information from there, three aspects are crucial: A theory for the description of the corpora, an effective markup and methods for specifying, and efficiently retrieving relevant portions of the data. MATE (EU Telematics Project LE4-8370) proposes standards for an integrated and consistent multilevel annotation based on the existing TEI standard (Text Encoding Initiative). For the access to massively marked-up speech data, a query language and a query engine have been developed. This allows the re-trieval of any combination of data, with linguistic annotations from different levels. The presentation will describe and demonstrate the query language.
Acoustic-phonetic analysis of the acquisition of American English tense-lax contrasts by native speakers of Chinese
Authors: Rogers Catherine L.
Published: July 2002

In a study of the perception of Chinese-accented English by native American English-speaking listeners [C. Rogers and J. Dalby, J. Acoust. Soc. Am. 100, 2725(A) (1996)], in which performance on seven segmental error category variables was correlated with intelligibility at the sentence level, vowel tenseness was found to be most strongly correlated with connected-speech intelligibility (r=0.76, p<0.001). Analyses of these results across the eight Chinese speakers showed that the average performance of the four most intelligible speakers (measured in terms of sentence percent-correct performance) on words targeting tense-lax contrasts surpassed that of the four least intelligible speakers by more than 30%. Analyses of both durational and spectral properties of the target lax phonemes /ɪ, ε, and ᴜ/ produced by the Chinese speakers will (1) explore the acoustic properties responsible for the difference in performance on the target lax phonemes across the two groups of speakers, (2) compare results across the three target phonemes, and (3) investigate the hypothesis that acquisition of the tense-lax contrasts is associated with more nativelike implementation of phenomena realized only at the connected-speech level (e.g., phrase-final lengthening). [Work supported by NIDCD.]
Perceived vowel quantity in Swedish: Native and British listeners
Authors: Behne Dawn M. , Sullivan Kirk P. H. , Czigler Peter E.
Published: July 2002

In many languages, vowels are characterized by their use of contrastive phonological vowel quantity and vowel quality. In Swedish, vowels have traditionally been described as being distinct in quality as well as having a phonological distinction between short and long vowel quantities. In English, however, phonological distinctions among vowels are described as primarily qualitative. This investigation examines the perceptual use of vowel duration and the first two vowel formant frequencies in distinguishing Swedish vowel pairs by three groups of listeners: native Swedish listeners (SS), British English listeners who do not know Swedish (EE), and British listeners who know Swedish well (ES). For each of three pairs of Swedish vowels (IPAKiel[i=DC]IPAKiel-[I], IPAKiel[o=DC]-IPAKiel[O], IPAKiel[A=DC]IPAKiel-[a]), /kVt/ words were resynthesized having ten degrees of vowel duration and ten degrees of F1 and F2 adjustment. Listeners’ responses and reaction times in a rhyming task show that unlike native listeners, L2 (i.e., ES) listeners distinguish Swedish vowel quantity using duration almost exclusively, which is consistent with what they have been taught, whereas TimesEE listeners respond as expected for a comparable English vowel continuum, using both vowel duration and spectra. These results are discussed in terms of the different stategies used by the different listeners.
Training American listeners to perceive Mandarin tones: Transfer to production
Authors: Wang Yue , Spence Michelle M. , Jongman Allard , Sereno Joan A.
Published: July 2002

Training American listeners to perceive the four Mandarin lexical tones has been shown to be effective [Y. Wang, A. Jongman, and J. A. Sereno, J. Acoust. Soc. Am. 103, 3090(A) (1998)]. Eight trainees participated in a two-week training program, after which their tone identification accuracy showed an overall 21% increase from the pretest to the posttest, and the improvement was generalized to new stimuli and new talkers. Further studies have now been carried out to investigate whether the tone contrasts gained perceptually can be transferred to the production domain without additional training. Before their perception pretest and after their posttest, the same trainees were recorded, producing a list of 80 Mandarin words bearing the four tones. Their productions were judged by 40 native Mandarin listeners in an identification task. Trainees’ posttest tone productions were on average 16% more accurately identified than their pretest productions, indicating significant tone production improvement after perceptual training. Acoustic analyses of the pre- and posttraining productions further reveal the nature of the improvement. Results are discussed in terms of the relationship between non-native tone perception and production as well as learning at the suprasegmental level.
Spatial voice activity detection using a microphone array noise canceller
Authors: Hoffman Michael W. , Li Zhao
Published: July 2002

An array of microphones is used for both noise suppression and spatial voice activity detection (SVAD). A generalized sidelobe canceller (GSC) is employed for adaptive interference rejection and the signals in the noise canceller are also exploited to provide desired signal activity detection. The SVAD shows promising results for environments with moderate reverberation and single or multiple interfering sources. For direct-to-reverberant energy ratios of 4 to 11 dB and signal-to-interference power ratios of −10 to 10 dB, speech detection miss rates are generally below 5%, while false alarm rates range from 10% to 40%. Most scenarios generally resulted in the false alarm rates between 10% and 20%. A key advantage of the GSC-based SVAD is its computational simplicity. In addition, the proposed technique is quite suitable for multiple interfering sources as well as nonstationary interference such as competing speakers. The system has been tested with a standard CELP speech coder.
Improvements in teleconferencing sound quality and gain before feedback through the use of DML technology
Authors: Mapp Peter , Azima Henry
Published: July 2002

One of the primary limitations of teleconferencing systems is the premature onset of acoustic feedback. The current trend toward the use of desktop personal computers as the teleconferencing platform, rather than dedicated and acoustically treated video conferencing rooms, further exacerbates the problem. Local, sound reflecting surfaces including the desktop, conference table, or computer screen, etc., can cause undesirable specular reflections from the loudspeakers to occur. These are then picked up by the system microphone(s), causing the formation of strong comb filtering and resultant sound coloration, reduced speech clarity, and reduced gain before feedback. Historically, a number of approaches have been adopted to help overcome such effects, including echo cancelling, phase shifting and ‘‘smart’’ microphone techniques. The paper illustrates the effects of such reflections and shows that distributed mode loudspeakers can produce less boundary interactions and comb-filtering effects. Experimental data are presented to show that DML technology can passively offer a 4–7-dB improved gain before feedback advantage over conventional cone devices in typical teleconferencing applications. The fact that significant gains in feedback margin and reductions in potential coloration are achieved without the need for additional DSP or active techniques implies a radically different and advantageous acoustical radiation mechanism.
Bats, insects, and the evolution of hearing
Authors: Miller Lee A.
Published: July 2002

Insectivorous bats appeared in the fossil record about 50 million years ago, or long after certain insects had adopted nocturnal habits. Many of these nocturnal insect species already had complex ears used mainly for intra-specific communication. Due to their physically small size, insects can produce high-frequency sounds most efficiently. Thus many species were ‘‘preadapted’’ for hearing bats. Examples would be certain beetles and bush crickets. Some nonhearing insects presumably evolved the capacity to hear the ultrasonic signals of bats. One well-documented case is that of the green lacewing, where the first fossil records of ears appeared after bats evolved. Many species of moths have ears tuned to bat signals, an adaptation that has evolved independently in different families and presumably in direct response to bat predation. All evidence points to a substantial selection pressure applied by bats on nocturnal insects. With the preoccupation of hearing mechanisms to avoid predation, other responses of nocturnal insects to echolocating bats might be overlooked. One example could be the advantage for nocturnal insects to occupy the ‘‘clutter zone’’ near or in vegetation, which would mix echoes from the prey with those of clutter. [Work supported by the Danish National Research Foundation.]
Broad-scale volume backscattering strength measurements in the Black Sea
Authors: Mutlu Erhan , Gucu Ali C. , Bingel Ferit
Published: July 2002

Enhanced echograms of high-frequency sound (120 and 200 kHz) from entire tracklines surveyed during summer and winter of 1991–1994 were evaluated to observe volume backscattering in the Black Sea. Echo intensities were filtered through two thresholds for distinguishing suspended scatterers from active swimmers (fish). It was postulated that the backscattering by the suspended matters was associated with bathymetry, seasonal stratification and different levels of mixing of water, and daily migration of zooplankton. The scattering was layered in association with the physical structure in the open water. As bottom depth was shoaled, the vertical structure became more homogenous. Biological scatterers were diversified with zooplanktoners and pelagic fish (anchovy and sprat). Copepod and gelatinous organisms (comb-jelly and moon jellyfish) appeared to be responsible for the zooplanktonic scatterers. Changes in seasonal and diurnal migrational patterns of fish were investigated. During the summer, pelagic fish schools were observed in an offshore area, whereas in winter the schools were confined to the coastal area. In regard to the schooling density, higher concentrations were found in winter.
Resonant diurnal internal tides in the North Atlantic
Authors: Dushaw Brian D. , Worcester Peter F.
Published: July 2002

Using an acoustic tomography array located midway between Puerto Rico and Bermuda, enhanced diurnal tidal signals associated with the lowest internal-wave mode have been observed. These signals result from a diurnal internal wave with O(500-km) wavelength that is resonantly trapped between the shelf just north of Puerto Rico and the turning latitude, a distance of about 1100 km. The data obtained using the large acoustical array are consistent with the predicted Airy-function variation with latitude of diurnal internal waves near the turning latitude. The existence of this wave is a striking demonstration of the long spatial and temporal coherence of oceanic internal tides.
Numerical simulation of the enhanced heat production in tissue due to the nonlinear character of high-intensity focused ultrasound (HIFU)
Authors: Ginter Siegfried , Steiger Eckard , Riedlinger Rainer
Published: July 2002

HIFU is used in thermotherapy to destroy tumors located deep in human tissue by heat. It is well known that HIFU is connected with the generation of higher-frequency components due to nonlinear steepening during propagation. An over-proportional heat generation in tissue occurs with enhanced power input. To obtain fundamental information about an optimal dosage of the applied ultrasound, simulations are necessary. A hot spot in healthy tissue must be avoided and a sufficient heat delivery in the tumor region must be guaranteed. Until now most numerical investigations of thermotherapy with HIFU have been carried out under the assumption of linearity. This means an essential underestimation of heat production. Here an improved model is introduced which enables accurate simulations, including nonlinearities. It consists of a set of nonlinear acoustic equations which ensure an accurate full-wave propagation modeling. A broadband absorption model of typical soft-tissue frequency-power-law character is included to obtain a nonlinear modeled heat source. The thermal behavior is described by a bio-heat-transfers-equation. The complete model is solved by means of a high-order FDTD scheme. Exemplary simulations illustrate the efficiency of this method.
Noninvasive estimation of tissue temperature via ultrasound frequency shift using a narrow-band transducer
Authors: Song Seong-Wook , Sung Koeng-Mo , Choi Min-Joo , Lim Jun-Seok
Published: July 2002

When applying the high power focused ultrasound to medical therapy, it is necessary to monitor the temperature change in tissue. Some methods have been proposed to measure this change noninvasively. One of those techniques measures the frequency changes in harmonics which are linear with the temperature changes. The linear coefficient has been defined with the fundamental frequency changes and the temperature changes. This measurement requires at least three harmonics, so that it requires a wideband transducer to estimate the fundamental frequency efficiently. This paper suggests a method to measure the temperature changes noninvasively with only one component of harmonics, under the condition that the proportional constant is defined such as the thermal expansion coefficient. It does not require the wideband transducer. Additionally, it shows the mutual relationship between thermal expansion coefficient, thermal frequency shift coefficient, and thermal speed change coefficient, which are defined in similar form.
A model of primary sensations, pitch, loudness, and timbre, of sound signals
Authors: Ando Y.
Published: July 2002

Primary sensations, pitch, loudness, and timbre, of a given source signal and sound field are described based on the model of the auditory–brain system [Ando, Architectural Acoustics (AIP/Springer-Verlag, New York, 1998)]. The model consists of the autocorrelation and interaural cross-correlation mechanisms. (Note that the power density spectrum is identical with the ACF.) In order to describe timbre, the human cerebral hemisphere specialization for the temporal and spatial factors of sound fields is taken into consideration in a similar manner as the subjective preference. Here, timbre is defined by a remaining primitive sensation that cannot be expressed by only pitch and loudness. [Work was partially supported by the Ministry of Education, Grant-in-Aid for Scientific Research (C), 9838022, 1998.]
The mechanisms underlying temporal pitch perception studied with sequences of filtered clicks
Authors: Bering Christian , Kaernbach Christian
Published: July 2002

In order to probe the mechanisms underlying temporal pitch perception it is necessary to eliminate all spectral cues from the stimulus, i.e., to use stimuli that cannot be resolved by the cochlea. High-pass filtered clicks combined with the corresponding low-pass filtered noise prove to be especially suitable for the construction of stimuli with specific temporal properties. These stimuli allow for the testing of specific hypotheses concerning temporal processing in the auditory system. The present study compares psychophysical results, obtained with periodic and aperiodic click sequences with single-cell simulations of chopper cells in the ventral cochlear nucleus. The latter are thought to explain the observed sensitivity to temporal regularity.