bluuebunny/crossref_metadata_2025 huggingface dataset:
Detection thresholds for isolated vowels
Authors: Kewley-Port Diane
Published: October 2005

A series of experiments on the detectability of vowels in isolation has been completed. Stimuli consisted of three sets often vowels: one synthetic, one from a male talker, and one from a female talker. Vowel durations ranged from 20–160 ms for each of the sets. Thresholds for detecting the vowels in isolation were obtained from well-trained, normal-hearing listeners using an adaptive-tracking paradigm. For a given duration, detection thresholds for vowels calibrated for equal rms sound pressure at the earphones differed by 22 dB across the 30 vowels. In addition, an orderly decrease in vowel thresholds was obtained for increased duration, as predicted from previous data on temporal integration. Several different analyses were performed in an attempt to explain the differential detectability across the 30 vowels. Analyses accounting for audibility reduced threshold variability significantly, but vowel thresholds still ranged over 15 dB. Vowel spectra were subsequently modeled as excitation patterns, and several detection hypotheses were examined. A simple average of excitation levels across excited critical bands provided the best prediction of the level variations needed to maintain threshold-level loudness across all vowels.
The echolocation ability of the beluga (Delphinapterus leucas) to detect targets in clutter
Authors: Turl Charles W. , Skaar Debra J. , Au Whitlow W.
Published: October 2005

A beluga (Delphinapterus leucas) was trained to detect different length cylinders in front of a clutter screen at five separation distances. The clutter screen consisted of 300, 5.1-cm-diam cork spheres located behind the targets. The clutter screen was used in two positions: perpendicular to the body axis of the beluga (90° grazing angle) and rotated 22° (68° grazing angle). Five hollow stainless-steel cylinders (3.2-cm-diameter and 0.38-cm wall thickness) of different lengths (14, 10, 7, 5, and 3 cm) were used as targets. Detection data were collected on the beluga's performance as a function of the separation between targets and the clutter screen. The beluga's performance was above 80% correct detection for the 14- and 10-cm cylinders as the separation distance decreased from 10.1 to 5.1 cm for both grazing angles. For all targets except the 3-cm cylinder, the beluga's detection was higher at 0-cm separation than at 2.5-cm separation. The beluga's detection performance was higher at 68° than at 90° grazing angle. The target strength of the cylinders and the clutter screen was measured in peak-to-peak amplitude and the energy of the incident and reflected echoes. For 0-cm target-to-clutter screen separation at 90° grazing angle, the echo-to-reverberation ratio at 50% correct detection was − 1.0 dB, based on the peak-to-peak measurements, and − 5.1 dB based on the energy measurement. The results of this experiment suggest that a beluga can detect targets in 3.6 to 5.4 dB more reverberation than previously reported for a bottlenose dolphin (Tursiops truncatus).
A magnetohydrodynamic underwater acoustic transducer
Authors: Schreppler Stephen C. , Busch-Vishniac Ilene J.
Published: October 2005

The radiation characteristics of a magnetohydrodynamic (MHD) acoustic transmitter radiating into a free medium are examined. Two source mechanisms are associated with the transducer: an electromagnetic force associated with the MHD operation, and a density fluctuation associated with heating of the medium by an electrical current. The wave equation is derived and a solution is found for impedance boundary conditions using a Green's function approach. Measurements obtained in a large water tank are shown to compare favorably with the predictions.
Influence of the sharpness of a wedge on the diffraction pattern
Authors: de Billy M. , Piet J. F.
Published: October 2005

The effect of the sharpness of a wedge on the diffraction pattern of an acoustic beam incident at 90 deg on a metallic sample immersed in water is studied in this letter.
The shortcomings of pure circuit models of transducers
Authors: Busch-Vishniac Ilene J. , Paynter Henry M.
Published: October 2005

Circuit models of transducers have been used for decades and have proven to be useful tools for prediction of transducer behavior. In this approach, the transduction mechanism is typically modeled using ideal two-port transformers or gyrators that neither dissipate nor store energy. Energy loss and storage are taken into account using one-port elements connected to the transducing gyrator or transformer. An alternative approach to transducer modeling recognizes that the transduction mechanism itself generally is associated with energy storage or dissipation. In this formulation, the transduction mechanism is typically modeled using a two-port energic element (such as a two-port capacitance). A comparison of the two formulations for piezoelectric transducers shows that there are two fundamental shortcomings introduced in the circuit model approach that are not present in the energic multiport element approach: The transformer modulus must be made a function of the strain of the piezoelectric material, and it is not possible to discuss the piezoelectric material response without imposing a load on the transducer. It is concluded from this analysis that circuit models fail to represent accurately energy exchange in transducers.
Microsurface acoustic wave resonator
Authors: Golan G. , Griffel G. , Seidman A. , Croitoru N.
Published: October 2005

Surface acoustic wave (SAW) resonators are the modern equivalents of the old bulk crystal resonators, used for high-Q frequency filtering, or very accurate frequency sources. A conventional resonator consists of a transmitting/receiving transducer, bounded from both sides by grid reflectors, creating a “Fabry-Perot” cavity structure. A major disadvantage of conventional resonators is their large dimension, which makes them inconvenient for integrated applications. In order to overcome this size limitation, new types of microresonators were designed and tested. These guided SAW resonators are able to guide and control the propagating component of the acoustic energy upon them, almost without losses. Based on theoretical calculations, two kinds of such devices were developed, namely, a corrugated waveguide filter and a microresonator structure. The main component in each of those devices is a metallic ΔV/V channel, which confines the propagating wave underneath and small, x-direction, transverse, short metallic strips in order to obtain periodic perturbations. Based on the “coupled mode theory,” it was found that by using such periodic perturbations, a complete Fabry-Perot reflecting mirror for surface acoustic waves can be realized. The reflection coefficient of such a micrograting mirror is almost total and its performances are similar to the conventional resonators. [This work was supported by the Israeli Ministry of Communications.]
A new window function to synthesize a surface acoustic wave bandpass filter
Authors: Kelin Wu
Published: October 2005

A new window function is proposed, which is written as w(t) = [1 − th a(t/τ)2], where a > 0 and is a parameter to be selected according to the given response specification, and τ is the width of the window, namely, the length of the pulse response. The results of synthesizing SAWF by means of the new window function are reported. In synthesizing, one unapodized transducer and one length-weighted transducer are used. The transducers have a double electrode structure with each half of the double electrode individually weighted.
A study on howling suppression by automatic directivity control of a microphone array
Authors: Makino Tadayori , Miyagawa Takeshi
Published: October 2005

Howling is one of the most annoying problems for public-address systems in a reverberant sound field. A new technique to suppress howling has been developed, in which the direction of the strongest coupling between the microphone and the loudspeaker is constantly detected to lower the sensitivity of the microphone in that particular direction. For detecting the direction of coupling, a reference signal, whose spectrum is modulated to spread over the frequency range of the public-address system, is combined with the acoustic input to be reproduced from the loudspeaker. This combined reference signal is eventually picked up by an array of microphone units. The direction of the coupling is detected from the difference in the arrival time of the reference signal between these microphone units. To lower the sensitivity of the microphone array for the particular direction, the directivity of the microphone array is controlled by individually delaying the outputs of each of the microphone units and then combining them as the output of the microphone array.
Acoustic wave field analysis of ultrasonic transducers
Authors: Sung Shung H.
Published: October 2005

A numerical method was developed for predicting the radiated acoustic field from arbitrarily shaped ultrasonic transducers. The numerical method was first verified for a spherical transducer, and it was found that the numerical results agreed very well with the exact analytical results. The numerical method was then applied to other transducer shapes where exact analytical results are not available: toroidal, elliptical, conical, and parabolic transducers. Among these transducers, the elliptical and the parabolic ones have never been studied in regard to their ultrasonic radiation characteristics. As expected, an ultrasonic transducer's geometrical shape affects the radiated acoustic field and acoustic beam pattern, and thus characterizes its specific application. For both the focusing-and non-focusing-type transducers, the ultrasonic characteristics of focal length, depth of field/depth of penetration, beam radius, sidelobe level, and its radial location were calculated and are discussed.
Analysis and synthesis of sounds using waveform and source-filter models
Authors: Rodet Xavier
Published: October 2005

Sound analysis and synthesis for music requires accurate analysis and high-quality synthesis of sounds. Some of the methods that were developed at the Institut de Recherches et de Communication Acoustique/Musique (IRCAM) for contemporary music research and production are described. The following analysis techniques have been implemented and used: FFT and phase-vocoder, wavelets, linear prediction algorithms (Levinson, global covariance, Burg, lattice covariance, lattice recursive), prony analysis, pitch detection, temporal envelope tracking (power tracking), detection of harmonic/noisy components, modal analysis (resonance modeling), formant-waveforms (FOF) coding, and formant trajectory coding. The phase-vocoder is used extensively for analysis and processing (filtering, cross-synthesis, time warping, etc.) of speech and music. Different LPC analysis algorithms have been studied and are used for precise estimation of spectral envelopes. These envelopes can be coded into (as well as recalculated from) formant parameters without loss of precision. Sounds and LPC residuals are coded into a harmonic/noise index for each of the frequency bands centered around harmonic partials of the signal, thus improving the quality of synthetic sounds. For sound processing, the following algorithms have been used: cascade filtering (LPC-vocoder), parallel filtering (in the program CHANT), frequency domain filtering (phase-vocoder), time-warping (phase-vocoder and LPC-vocoder), time-varying and arbitrary sample-rate conversion. Several synthesis methods have been developed, including LPC-derived synthesis filter, formant synthesis, formant-waveform (CHANT), parallel filtering, phase-vocoder, and additive synthesis. Synthesis-by-rule uses sophisticated object-oriented languages (PREFORM) for calculation, manipulation, and real-time control of synthesis parameters. For these studies, a real-time work-station is under development, based on SUN-3, Mercury array processor, and Sony PCM AD/DA.
Conservation and compensation for noise-induced hearing loss in Japan
Authors: Zusho Hiroyuki
Published: October 2005

In Japan there are workplaces with a high level of noise exceeding 100 dB(A) in various industries, such as shipbuilding, metal mining. metal manufacturing, and ear manufacturing. In the past several years, the number of new beneficiaries of workers' compensation insurance for noise-induced hearing loss has ranged from 400 to 800 yearly. According to Japanese laws concerning conservation of hearing in workplaces, workplaces producing noise are designated as such. These places are obligated to measure the noise level periodically and also to carry out the control of the work environment, hearing conservation, and education on labor hygiene. Recently, a new guideline for noise control was published and hearing conservation in these designated workplaces will be carried out in accordance with this guideline in the future. As for workers' compensation insurance, various types of benefits are paid upon the worker's retirement in accordance with the extent of the hearing loss suffered by a worker at a workplace with a level of noise exceeding 85 dB(A) over a long period of time.
Procedures evaluated by the ANSI S12.12 Working Group for determining the effectiveness of hearing conservation programs (HCPs)
Authors: Royster Larry H. , Royster Julia Doswell
Published: October 2005

The ANSI S12.12 Working Group has evaluated several procedures for conducting audiometric database analysis (ADBA), including threshold shift criteria and indices of threshold measurement variability. Findings from applying these procedures to a large audiometric database created by the committee have been used to establish trial guideline ranges for judging HCP effectiveness. The S12.12 Working Group will publish its recommendations for trial utilization and evaluation. This paper presents the ADBA procedures evaluated by S12.12 and the trial guideline ranges under consideration. A recommended format for applying ADBA to evaluate the effectiveness of HCPs includes: (1) restricting the database to a known controlled population, (2) checking the database for non-noise-related contaminations such as calibration errors, (3) applying the procedures to the total database to judge overall program acceptability, and (4) using the procedures to compare the relative effectiveness of the HCP in protecting subpopulations such as groups in different production units or groups using different hearing protectors.
Workplace evaluation of earmuff-type hearing protectors
Authors: Durkt George , Giardino Dennis A.
Published: October 2005

The measurement of attenuation values for hearing protectors as used by personnel at the work site give a more realistic estimation of the protection afforded the worker. Head movement, sweating, chewing, and other physical factors may affect the integrity of the hearing protector resulting in different attenuation values than those obtained from the standard ANSI test method. A miniature, two-channel, frequency-modulated, telemetry system has been developed to measure attenuation values for earmuff-type protectors. Data from small microphones located inside and outside the hearing protector are transmitted to a remote receiver (up to 200 ft) and tape recorder for later analysis. The wireless link between the worker and observer allows the gathering of data while leaving the worker unencumbered to perform normal duties. Repetitive laboratory tests using combinations of nine subjects and eleven different muffs have provided correction factors for the system relative to the ANSI S3.19-1974 psychoacoustic standard method for measurement of earmuff attenuation. The results of numerous field trials in terms of NRR, dBA reduction and octave-band attenuation values are presented for various muffs, occupations, and noise sources. These results are compared to those stated by the manufacturer.
Exposure time versus noise level: A 3-, 5-, or 6-dB rule?
Authors: Bies David A. , Hansen Colin H.
Published: October 2005

The relationship between noise exposure and noise level represented in long-accepted published data [A. Glorig, in Noise and Vibration Control, edited by L. Beranek, Chap. 17, table 17.15; W. Burns and D. W. Robinson, Hearing and Noise in Industry (Her Majesty's Stationary Office, London, 1970), Appendices 10 and 11] is examined in a new way. It is shown that the data support the hypothesis that, in any situation involving a fixed time of exposure to a constant noise level, the same noise exposure (or hearing damage) is experienced if the noise level is increased by 6 dB and the exposure time is halved. This is contrary to regulations in Europe and the U.S.A. where the quantity 6 dB is replaced by 3 and 5 dB, respectively. An integral of pressure with time is consistent with the 6-dB observation, which also becomes the logical choice if the mechanism by which hearing damage accumulates is properly understood. A quantity identified as a hearing deterioration index (HDI) is introduced as a single scaling parameter for the data [D. A. Bios and C. H. Hansen, Engineering Noise Control (Unwin Hyman, London, 1988), Chap. 4].
Anomalous ultrasound propagation in high-Tc superconductors
Authors: Bhattacharya S.
Published: October 2005

Propagation of longitudinal and transverse ultrasound in high-Tc superconductors reveals unusual behavior at and below Tc. The superconducting order parameter is found to possess a large coupling to some volume-preserving shear distortions. Unlike in conventional superconductors, the behavior below Tc, is inconsistent with a condensation of the carriers alone. The elastic modulus behavior below Tc is reminiscent of charge density wave/spin density wave transitions. The attenuation below Tc correlates with thermal transport and is related to either a strong phonon-phonon interaction or the Zener effect of intergranular heat transport in polycrystalline systems. Several anomalies in both the velocity and the damping are observed between Tc,. and room temperature, indicating the presence of other phase transitions of yet unknown origins. Magnetic field dependence of sound velocity shows fluctuation effects above Tc and a large stiffening below Tc probably related to an unusual flux-lattice dynamics. Implications of these results on the nature of superconductivity will be discussed.
Resonant ultrasound measurements in single crystals of intrinsic and superconducting copper-oxygen plane compounds
Authors: Migliori Albert , Brown Stuart E. , Fisk Zachary , Ahrens Eric T. , Maynard J. D. , Mather J. H.
Published: October 2005

Resonant ultrasound measurements in very pure single crystals of La2CuO4 have been made. The material was prepared such that its resistivity at 300 K is extremely high (170 Ω cm), and untwinned crystals of about 0.1×0.1×0.1 cm were used. Oxygen was then introduced, eventually leading to superconductivity. Here, data on the three longitudinal sound velocities and attenuations as a function of temperature for different oxygen dopings are presented and the effects of magnetic and superconducting interactions are discussed. All data were taken with a resonant ultrasound method developed by us in which the transducers are flexible and are less than 1% of the sample mass, minimizing transducer loading effects.
Can you hear the elastic tensor?
Authors: Visscher William M.
Published: October 2005

The measured response of an object of known size and shape composed of a homogeneous but anisotropic elastic material as a function of frequency of excitation is, in principle, enough to determine the linear elastic constants, 21 of them in the most general case. A method for computing the resonant frequencies is explained and illustrated for an object of simple geometry (a×b×c rectangular parallelepiped) with free or clamped faces of a material whose elastic modulus tensor reflects the symmetry of an orthorhombic lattice (nine independent constants). The method is a variant of MOOT [method of optimal truncation, see J. L. Opsal and W. M. Visscher, J. Appl. Phys. 58, 1102 (1985)], which is a boundary residual method developed to compute elastic wave scattering. The calculation was undertaken as a part of the analysis of measurements of ultrasonic response of single-crystal, high-temperature superconductor materials (reported elsewhere at this meeting). Results will be presented, and the practicality of this approach to determine unknown elastic constants will be discussed. [Work supported by USDOE.]
Relations between formant slopes and speech intelligibility in neurologically impaired talkers
Authors: Kent R. D. , Weismer G. , Kent J. F. , Martin R. E. , Rosenbek J. C.
Published: October 2005

The relation between F1 and F2 slopes measured from monosyllabic words and speech intelligibility scores for these words was studied for 25 men with amyotrophic lateral sclerosis. The formant slopes were estimated from digitized formant tracks for each test word. The intelligibility scores were computed as the percentage of words correctly identified for each talker by a listener panel of ten young women. The intelligibility scores for the group of 25 talkers ranged from 41%–99%. The correlation coefficient computed between the F2 slopes and the intelligibility scores was 0.76. The F1 slope was not as predictive of intelligibility, perhaps because the dysarthric talkers compensated for impaired tongue control with large jaw movcmcnta (and, hence, large F1 shifts). It is concluded that F2 slope, a dynamic measure of acoustic structure, bears a moderately high correlation with intelligibility in these subjects. Examples of formant patterns are shown for different degrees of speech impairment. [Work supported, in part, by NINCDS.]
Articulatory dynamics of deaf speakers during plosive production: Aerodynamic and kinematic evidence
Authors: Mahshie James J. , Yadev Pradeep
Published: October 2005

The production of speech requires precise movement and timing of the articulators to accurately valve the egressive airstream. While evident that deaf speakers often experience difficulty controlling such valves for speech production, the nature of such valving difficulties remains elusive. Previous aerodynamic studies have been primarily descriptive, characterizing steady-state aspects of speech production. The present study described and compared dynamic aspects of deaf and normal-hearing speakers' speech production using aerodynamic and kinematic measures. Five profoundly deaf and two normal-hearing young adults produced multiple tokens of voiced and voiceless plosive segments in varied vowel contexts, while simultaneous measures were obtained of oral and nasal airflow, oral pressure, and electroglottograph and acoustic signals. Conductance of oral and nasal vocal tract constrictions were used to describe and compare the magnitude and relative timing of articulatory gestures for deaf and hearing subjects. Computer simulation of the supraglottal air pressure and airflow waveforms was used to further investigate how these waveform variations may be related to articulatory maneuvers. Implications for speech training will be discussed. [Research supported by the Whitaker Foundation.]
Quantitative evaluation of hypernasality in cleft palate patients
Authors: Kataoka Ryuta , Takahashi Koji , Yamashita Yukari , Imai Satoko , Michi Ken-ichi , Okabe Kaoru , Hamada Hareo , Miura Tanctoshi
Published: October 2005

To quantitatively evaluate hypernasality in cleft palate patients, the Japanese vowel /i/ pronounced by six cleft palate patients and four normal children (controls) of similar ages was analyzed acoustically by cepstrum analysis. Spectrum envelopes obtained by the cepstrum method were evaluated every 1/3 octave to obtain the mean level in each band. Ten listeners evaluated a speech sample from each subject for degree of nasality on an equal interval scale ranging from 0 (no nasality) to 4 (strongest nasality). Two factors were obtained from the factor analysis of the judged scores. The first factor, which accounted for 77% of the total variance, was the consensus perception of nasality. The second factor, which accounted for 9%, was the difference among the individual listeners. The levels in two 1/3 octave bands were highly correlated with the first factor. The central frequencies of these two bands were 1 and 5 kHz.
Noise radiation from a jet impinging on a flat plate
Authors: Shen Jianping , Meecham William C.
Published: October 2005

In this study, a search is made for high-noise-source regions for a jet impinging on a large flat plate. By using cross-correlation techniques for the pressure fluctuation on the plate and the farfield acoustic radiation, the location of the noise source using time delays (phase shifts) can be determined. Different jet speeds (up to Mach number 0.4), jet-to-plate distances (5 to 10 jet diameters), and impinging angles (0° to 80° from the jet axis) were used in an experiment in the UCLA Aeroacoustics Laboratory anechoic room. A full digitization treatment was used in analysis. The results show that the source is located on or near the plate in a circular region centered on the jet center, with a radius of approximately one jet diameter. The directivity pattern of the radiation will also be discussed. Standard theory indicates that there should be little surface, dipole sound from a large flat plate. The question will be discussed.
On sound radiated by inhomogeneities convected through a space-varying flow
Authors: Powell Alan
Published: October 2005

It was pointed out in Powell [J. Acoust. Soc. Am. 36, 1032 (1964)] that inhomogeneities in composition, entropy, or vorticity give rise to sound radiation when convected through a steady space-varying flow as in a nozzle or past a curved surface. The one-dimensional case was extended by Cuadra [J. Acoust. Soc. Am. 42, 725–732 (1967)] and more completely by Ffowes-Williams and Howe (1975), but the initial “scattering approach” presented appears to have been published only qualitatively, e.g., Powell [Noise Control Eng. J. 8, 108–119 (1977)]; but note related developments by Crighton and Leppington (1971), and Morfey (1973). This is simply Rayleigh scattering with an axis change so that the inhomogeneities sense time-varying changes as they are swept supersonically through weak oblique standing waves springing from a fixed sinusoidal wall. The physical arguments carry over for evanescent waves (complex wavenumber), the corresponding steady flow being subsonic. It is readily inferred, e.g., that for subsonic and supersonic flows (away from M ≃ 1), the sound power depends on the sixth power of the velocity.
Turbulent boundary layer simulation setup on a sonar dome
Authors: Audet Jean , Lagier Michel , Marin-Curtoud Pierre , Rohan Thierry
Published: October 2005

An experimental-analytical technique was developed to evaluate the noise produced inside various types of sonar domes under the effect of a turbulent flow, in a frequency range where the structure has a high modal density. The transfer functions between a punctual external force applied to the dome and the acoustic pressure at a point inside the cavity was measured on a scale model. This measure was made using a reciprocity technique with emitting hydrophones within the cavity and with accelerometers all over the dome's external surface. The noise level at a point inside the cavity is calculated from these experimental transfer functions and a turbulent boundary layer wall pressure model. The wave vector frequency spectrum model used is Chase's model [D. M. Chase, J. Sound Vib. 70, 29–67 (1980)], adjusted according to the local parameters of the flow. The results obtained with this method agree with experimental results. In order to test various types of structure, this technique on a scale model has considerable advantages compared with experimentations on a real structure: limited expense, experimental ease, and reliability.
Wall pressure fluctuations beneath a transitional boundary layer
Authors: Audet J. , Dufoureq Ph.
Published: October 2005

Up to now, different models have been developed to describe wall pressure fluctuations beneath a fully developed turbulent boundary layer [G. M. Corcos, J. Acoust. Soc. Am. 35, 192 (1963); D. M. Chase, J. Sound Vib. 70, (1)]. Recent]y, the total field pressure in the transition zone was described in different papers [G. C. Lauchle, J. Acoust. Soc. Am. 67, 158–168 (1980); M. Laglet and D. Sornette, Acustica 61, 116–124]. These works show that intermittent processes induce monopole sources whose radiation could be dominant. The present paper is devoted to the development and numerical analysis of a formulation based on these new models and fitted to wall pressure fluctuations calculation (local field). In this case, the wall pressure fluctuation is the sum of two terms: one associated with the turbulent content of intermittent spots and the other associated with the spot boundaries (monopole radiation). The relative contribution of each kind of sources to the transitional boundary layer wall pressure fluctuation can be investigated. Successful comparisons wth experimental results [Ph. Dufoureq, these Docteur Ingénieur, Ecole Centrale de Lyon (1984)] prove, in many cases, the dominance of monopole sources induced by intermittency. Noise generated by the transition region could be dominant at low frequency and could even disturb the upstream laminar flow.
Nonlinearities seen in hair-cell data from in vivo and in vitro preparations
Authors: Dallos Peter , Evans Burr N. , Cheatham Mary Ann
Published: October 2005

The growth patterns of fundamental, dc, and harmonic receptor potential components recorded in vivo from the low-frequency region of the guinea pig cochlea as a function of stimulus frequency versus best frequency of the cell were examined. Changes in these components with natural variations in membrane voltage and with electrical polarization of the cell are also considered. A report is also given on nonlinear growth patterns of outer hair-cell motile responses as a function of cell length (best frequency).
High-frequency sea surface scattering: Recent progress
Authors: McDaniel Suzanne T.
Published: October 2005

Recent progress in understanding the processes governing high-frequency acoustic scattering from the sea surface is reviewed. Theories for scattering due to surface roughness and from resonant microbubbles are presented, and the status of environmental inputs to these theories, including ocean roughness spectra and independently measured bubble populations, is discussed. Recently acquired acoustic data are shown to clearly demonstrate that resonant scattering from subsurface bubbles constitutes an important contribution for backscattering geometries at frequencies as low as 3 kHz. The theoretical predictions are then extended to lower frequencies, where resonant bubbles are not a viable scattering mechanism, and compared with backscatter data acquired using explosive charges. Whereas good agreement is obtained with certain data sets, other data lie consistently above the predictions. [Research supported by ONT with NORDA management.]
The effect of monomolecular films on the underlying ocean ambient noise field
Authors: Rohr Jim , Glass Ray , Castile Brett D.
Published: October 2005

A series of at-sea tests has established unequivocally that the presence of a monomolecular surface film results in a pronounced reduction of ambient noise beneath the film. This reduction was observed over the wide range of surface conditions associated with sea states 1–6. Possible mechanisms through which the film affects the underlying ambient noise field are explored. [Work supported by DARPA and ONT.]
Characteristics of the SeaMARC II phase data
Authors: Matsumoto Haruyoshi , Shor A. N. , Blackinton J. G.
Published: October 2005

SeaMARC II is an interferometric, shallow-tow, long-range side-scan sonar system that is capable of producing bathymetry as well as side-scan images of the bottom in real time. To produce the bathymetry, phase differences between the echoes received by two linear acoustic arrays are measured. The phase difference data, sampled in travel time, show a slightly skewed, near-Gaussian distribution. The ambient noise and signal interference from the sea surface seem to cause increase of the phase data spreading. For SeaMARC II, the following are true: (1) Both the surface reverberation and the indirect echo path via water surface cause significant phase dispersion throughout the echo return time; (2) ambient acoustic noise is the main limiting factor for the angular resolution in a deep-water survey; and (3) the phase sample distribution is near-Gaussian and skewed in time. The relationships of these factors to the induced spread of the phase data are discussed. The current scheme for producing bathymetry from the phase difference data is discussed in the light of signal interference and phase sample distribution. Based on the Gaussian and isotropic ambient noise assumption, a simple analytical equation for the SeaMARC II phase data spreading is derived as a function of sonar parameters including the bottom backscattering coefficient. [Work supported by ONR.]
Speech production by a vocal cords-vocal tract-vocal tract wall vibration model
Authors: Suzuki Hisayoshi , Nakai Takayoshi
Published: October 2005

This research was aimed at finding the effects of finitehess in the mechanical impedance of the vocal organs on speech sounds. A speech synthesis program has been developed considering the vocal tract wall vibration that causes sound leakage from the vocal cavity to the nasal cavity even when the velum is closed. The radiation of sound from the outer surface of the vocal tract is not always negligibly small, and its effect is enhanced by the nasal cavity. The synthesis program has the parameters of the physical properties and pressure of ambient gases, as well as the mechanical impedance of a yielding vocal tract wall. The wall vibration is treated as a small perturbation in the area function of the vocal tract. The following items will be discussed in the paper: comparisons between vowels with and without vocal tract wall vibration, sounds from the mouth opening, nostrils, and vocal tract wall, and the distribution of the strength of the wall vibration along the vocal tract. The effect of the pressure of ambient gases on these items will also be discussed.
Measurements of articulatory dynamics using optoelectric and magnetic devices
Authors: Sonoda Yorinobu , Konomi Toshiharu , Mori Keisuke
Published: October 2005

This paper introduces two methods for measuring articulatory movements. Optical and magnetic sensing techniques were applied to recording tongue movements. The optical recording principle is based on the measurement of the phase shift of modulated light reflected from the tongue surface with less sensitivity to its reflection factor, and not on the measurement of the intensity of reflected light, which has been the usual practice. The sensor unit was composed of a small phototransistor and two LEDs, mounted together onto an artificial palate, and was attached to the hard palate. The prototype system consisted of three sensor units. The magnetic field sensing system was further developed to monitor tongue articulations. Design improvements were made on an earlier prototype system [Y. Sonoda, IEEE Trans. Magn. 10, 954–957 (1974)]. The sensor unit was composed of a magnetic multivibrator bridge circuit with amorphous alloy cores. Using these sensing systems together with the previously reported PSD system [Y. Sonoda, J. Acoust. Soc. Am. 72, 700–704 (1982)], experiments are now being conducted to observe simultaneously tongue, lip, and jaw movements.
Pellet placement in x-ray microbeam studies
Authors: Lindau Mona , Ladefoged Peter
Published: October 2005

A weakness of the x-ray microbeam system for recording articulatory movements is the difficulty of placing pellets on the pharyngeal part of the tongue. It is, however, possible to use the system to obtain static position of the tongue-root in another way. Image scans of sustained English vowels of one subject, in which a chain of 15 pellets 5 mm apart had been placed on the midline of the tongue, were recorded and analyzed. The results enabled the determination of the best locations for placing pellets and the extent to which tongue shapes can be determined from a small number of pellets. If only two pellets are used, one should be placed in the range of 30–35 mm and the other 60–65 mm from the tip. The tongue shapes predicted from pellets in these locations were highly correlated (r2 = >0.98) with the observed shapes as specified by three factors that themselves determine the tongue shapes within 0.5 ram. The same high correlations (r2 = >0.98) can be achieved with several different combinations of three pellets. Most (75%) of these combinations involve one point at the root of the tongue. However, by selecting points near the tongue tip (5–10 mm back), the middle (approximately 35 mm from the tip) and the back (45–50 mm from the tip), equally good correlations are obtained.
Tongue pellet data analysis by singular-value decomposition approac
Authors: Xue Qiuzhen , Nadler Robert , Abbs James
Published: October 2005

In order to obtain tongue trajectory information during speech, three or four pellets attached to the tongue of subjects are tracked in real time by the University of Wisconsin x-ray microbeam system. For each pellet, two time series are produced that correspond to x and y pellet coordinates, respectively. These data are being used to build a model of vocal tract acoustics. The singular-value decomposition (SVD) method has been found to be a good way to approach the following questions (1) Are the data redundant? (2) Can we find a better way to represent the data set? (3) Can relationships among the different pellets be discerned? In this experiment, six data series are organized into a data matrix and decomposed by SVD into two orthogonal matrices and one diagonal matrix. The diagonal matrix, called the singular-value matrix, indicates the redundancy of the original data set. Different deductions of the singular-value matrix were tested for five data sets. It is found that three transformed orthogonal data vectors can represent the original six data series at a 98% level of approximation. This work indicates that the orthogonal data matrices are well suited for model construction because: (1) the data size is decreased; (2) ambiguity is reduced for mapping work (e.g., to the terminal acoustic characteristics); and (3) relationships among the pellets can be found from the orthogonal matrices. These results and some other related issues will be discussed. [Work supported by NIH Grant NS-16373.]
Vowels formants in different modes of speaking
Authors: Umeda Noriko , Wallace Karen
Published: October 2005

Articulatory differences between maximally articulated situations and fluent speaking are said to lie in the degree of articulatory overlapping between neighboring phonemes, and target undershooting due to lack of execution time when the phoneme becomes short. In addition to these segmental factors, however, global factors may also cause a certain mode of speaking distinctively different articulatorily from other situations. For example, more limited jaw movements in fluent speaking may result in smaller F1 movements than words in isolation. The effect of corner cutting while concentrating efforts on selected items may be at work in fluent speaking. In order to capture comprehensive articulatory strategies, a study on formant frequencies of vowels in various modes of speaking has begun. A preliminary result with one speaker shows that, at the onset of vowels from consonants in the stressed syllables of content words, F1, representing roughly jaw movements, varies far greater a range in carrier phrases than in fluent situations; F2, which represents roughly back-and-forth movements of the tongue varies least in sentence-reading and greatest in conversation.
Kinematic analysis of articulatory declination
Authors: Vatikiotis-Bateson Eric , Fowler Carol A.
Published: October 2005

Previous investigations of declination—the progressive reduction of some measure over the course of an utterance—have focused primarily on F0, with occasional mention of acoustic amplitude. Recently, F0 declination has been shown to track a decline in subglottal pressure. To our knowledge, whether declination occurs in other subsystems associated with speech production has not been investigated. One such subsystem currently being examined for speakers of English, French, and Japanese is that of supralaryngeal articulator motion. Kinematic variables of displacement, duration, and peak velocity, associated with motion of the lower lip/jaw complex during reiterant speech, were tested for declination effects. Initial findings show that there is articulatory declination for some speakers, but there is much interspeaker variability when kinematic variables are examined individually. However, as demonstrated previously for stress and speaking rate contrasts, much of the spatiotemporal variability may be accounted for when the relations among kinematic variables are considered, thereby making it easier to determine whether articulatory declination is present. Furthermore, for the data examined thus far, articulator motion appears to adhere to the constraints of a damped, second-order dynamical system, whether articulatory declination is observed or not. [Work supported by NIH.]
Multimode coding: A novel approach to narrow- and medium-band coding
Authors: Taniguchi Tomohiko , Unagami Shigeyuki , Gray Robert M.
Published: October 2005

Most research on narrow-band coding is concentrated on how to transmit excitation parameters efficiently. However, the important thing in obtaining a good reproduced speech quality is how to control the balance of the transmission bit rate between the excitation and the LPC parameters. Multimode coding, which is proposed here, has two coding modes: One is a mode that transmits the LPC parameters in every frame, as conventional coders (A mode). The other is a mode that avoids the transmission of the LPC parameters by using the same coefficients as the previous frame and increases the bits allocated to the residual quantization instead of to the LPC transmission (B mode). In each frame, the mode selection takes place based on an evaluation of the reproduced speech quality, and the assignment of transmission information is dynamically controlled by switching between the two modes. This coding algorithm is applied to a 7.2 kb/s CELP coder, and approximately 3 dB of improvement is achieved in SNR compared with a conventional CELP coder. The B mode was used 78%–82% of the time.
Realization of a multirate speech codec utilizing spectrum peak emphasis
Authors: Taguchi Tetsu
Published: October 2005

One of the most promising means of realizing multirate speech coding is multipulse excited linear predictive coding (MPELPC) adopting an auditory weighting filter (AWF). This AWF effectively shapes quantization noise to be masked by the speech signal and ameliorates speech quality at approximately 8 kb/s or more. However, below this rate, the AWF is not effective because the SNR is so low that the signals scarcely mask the noise. When the SNR is rather low, only the high-power frequency components of speech signals (whose levels are higher than those of the noise) should be encoded. In order to emphasize these components effectively upon encoding, a spectrum peak emphasis (SPE), which is performed by a prefilter and utilized in MPELPC, has been proposed. Below 8 kb/s, it has been shown by computer simulation that compared to the AWF, the SPE improves SNR at least by 2.2 dB. Subsequently, a multirate code from 16 to 4.8 kb/s, basically utilizing the AWF but with the SPE below 8 kb/s, has been realized on a DSP, NECμPD77230. This coder is applicable to mobile satellite communication.
Real-time 3-D ultrasonic diagnostic imaging
Authors: von Ramm Olaf T. , Smith Stephen W.
Published: October 2005

Real-time phased-array two-dimensional ultrasound imaging has gained worldwide acceptance as a valuable diagnostic tool. In the past years, a parallel receive processing scheme for phased-array systems, called Explososcanning, which permits imaging of several B-mode lines for each transmitted acoustic burst, has been developed. Now this receive processing is being utilized in conjunction with newly developed two-dimensional arrays to electrically scan the ultrasound beam in three dimensions. The current 2-D transducers consist of 256 elements, operate at approximately 2 MHz and measure 16 by 16 mm at the site of skin contact. The 3-D echo data obtained from the pyramidal scan volume can be displayed as multiple sectors at preselected azimuth and elevation directions, as a two-dimensional projectional image with perspective or as stereoscopic pairs. Parallel processing will permit the formation of approximately 9000 look directions within the pyramidal scan volume at TV rates.
An historical review: Two demonstrations of restoring old acoustic recordings
Authors: Stockham Thomas G.
Published: October 2005

This presentation demonstrates two complementary approaches used in the early 1970s for the experimental restoration of acoustic recordings of Enrico Caruso. The first approach employs a method of blind deconvolution that attempts to eliminate the reverberant and resonant characteristics of such recordings. The second approach attempts to remove the orchestra] background using an analysis-synthesis method.
Measurement of spatial information in sound fields by a closely located four-point microphone method
Authors: Yamasaki Yoshio , Itow Takeshi
Published: October 2005

When an estimation is made of the sound field in a room, it is important to grasp spatial information, especially for the early reverberation periods. Here, a way to grasp spatial information of sound fields from impulse responses measured at four closely located points, the origin, and three points the same distance (3∼5 cm) from the origin on the rectangular coordinate axes will be discussed. From these four impulse responses the position of virtual image sources are calculated by the correlation and/or intensity technique. Concert halls in Europe, the United States, and Japan and some scale models are measured by this method. The three-dimensional distribution of virtual image sources, directivity patterns, and impulse responses from several directions are shown. Calculation of signals from special directions by convolving these impulse responses with dry music to estimate the sound fields with ears or to help the acoustic design, and the introduction of the Wignet distribution are also discussed.
Angular dependence of backscattered intensity of rough surfaces at different frequencies
Authors: Soetanto K. , Ohtsuki S. , Okujima M.
Published: October 2005

The relation between the scattered intensity and the roughness of a surface was measured by the angle of incidence at several frequencies with a typical pulse-echo system. Four types of phantoms with roughness, (a) smooth, (b) 0.03 mm, (c) 0.1 mm, and (d) 0.3 mm, are made of agar with one of the surfaces made rough for the observations. For these tests, 2.25-, 3.5-, 5.0-, and 7.5-MHz transducers are used. The angle of incidence [θ] ranges from 0° to 60°. The scattered intensity of the (a) and (b) phantoms was found to rapidly decrease by − 25 to − 38 dB as the angle increased. The decrease of scattered intensity with angle was less for phantoms with rougher surfaces. Smaller variation in scattered intensity at 7.5 MHz, compared to that at the low frequency of 2.25 MHz, is because of stronger scattering occuring from the rough surface when the wavelength is shorter. The rougher surface exhibits the smaller variation in scattered intensity with changing angle of incidence. The results of these measurements agree with earlier results [Reid, in Diagnostic Ultrasound, edited by C. C. Grossman et al., (Plenum, New York, 1966), pp. 1–12].