bluuebunny/crossref_metadata_2025 huggingface dataset:
Acoustical design of the Tokyo Metropolitan Art and Cultural Hall
Authors: Nagata Minoru
Published: October 2005

The Tokyo Metropolitan Art and Cultural Hall is planned as the second Bunka-Kaikan by the Tokyo Metropolitan Government and is scheduled to open in October 1990. The facilities have a major hall (1887 seats) for classical concerts, a medium hall (850 seats) for stage performances, and two minor halls for drama performances and musical events. The basic problems of the acoustical design were: (1) noise control of subway traffic; (2) sound insulation between halls; and room acoustical design of the major hall. The allocation of halls was studied considering both subway noise and sound insulation between halls. The major hall is placed on the top floor above the medium hall. The estimated subway noise in the major hall is below NC-20 and sound insulation between the major hall and the medium hall is 80 dB at 500 Hz. The acoustics of the major hall is designed to reproduce the outstanding features of Bunka-Kaikan sound. At an early stage of the design, the room shape was studied with computer simulation and at the construction stage, the details of the hall were studied by 1/10 scale model experiments. The acoustic characteristics will be presented at the meeting.
Phase response measurement with the three-port two-microphone coupler
Authors: Wong George S. K.
Published: October 2005

With the advent of sound intensity measurement, the need for a reliable and precise method for the assessment of phase response of microphones is evident. For national laboratories, a direct method is preferred. There are less time-consuming signal-processing techniques that are capable of measuring the phase response of microphones. However, for high-precision measurement, it is relatively difficult for national laboratories to verify or to substantiate the uncertainty of phase measuring techniques that involve inaccesible computational algorithms. The theory on precision phase measurement with the three-port two-microphone coupler [J. Acoust. Soc. Am. 71, 1276–1277 (1982)] is presented with experimental verification. The uncertainty of the method depends on the uncertainty of the phase measuring instrument. With a calibrated commercial phase meter, the uncertainty is typically 0.02 deg.
Reflection coefficient measurement techniques for underwater acoustic panels
Authors: Létiche Michel , Giangréco Christian
Published: October 2005

A new measurement method for evaluating underwater acoustic materials is proposed. It is based on a two-hydrophones technique and allows the determination of the complex reflection coefficient for size-limited acoustical panels. The results are shown to be independent of the incident plane wave amplitude and the distance between the hydrophones and the panel. Therefore, this technique involves direct and quick measurements so that it is possible to reduce the effects of scattered waves from the edges of the sample by averaging. Simulations have been made to check the stability of the method versus noise. The experiment covered the frequency range 1–12 kHz. Techniques (intensity, impedance, and pressure measurements) tested at G.E.R.D.S.M. (Groupe d'Etudes et de Recherches en Détection Sous-Marine) using surfacic sensors are also presented. Their accuracies are discussed through a mathematical model based on Helmholtz integral equation. Results obtained by the various methods on finite panels are compared.
Effect of microphone distribution on estimation of sound source by multiple microphones
Authors: Kido K. , Abe M. , Hongo S. , Chubachi N. , Suzuki H.
Published: October 2005

The effect of the shape, size, and interval of microphone array on the accuracy in estimation of the position and spectra of sound sources is investigated after the explanation of the principle of the estimation method. It is shown that the spatial distribution of sound intensity can be mapped according to the estimation of the position and the strength of each sound source. By the proposed method, the mapping of the sound intensity distribution due to a specified sound source can be carried out even in the case of noisy environment eliminating the effect of the other sound sources. The conditions to obtain an exact result are as follows: The sound field is a free field or a free field with an infinite baffle board, the intervals between the microphone elements are closer compared with the wavelength, and the extent of the array is broad enough. But it is usually difficult to satisfy these conditions in a practical case. This paper investigates how the estimated results are affected by the defects in the conditions and the use of some weightings on the outputs of the microphones.
Extensive pulsed ultrasonic Doppler method
Authors: Ohtsuki Shigeo
Published: October 2005

Recently, the pulsed ultrasonic Doppler method has been widely applied to the field of medicine. The color flow mapping system is helpful in a clinical decision. But the system is not sufficient for quantitative measurement of blood flow in the heart. The reason for this is the limitation of the measurable speed range due to aliasing of the pulsed Doppler system. On the other hand, there is a need for the observation of low-speed flow in organs such as the liver. Now, an extensive pulsed ultrasonic Doppler method is proposed here. The Doppler frequency can be expanded or compressed in order to observe the Doppler frequency within a limited frequency range due to a pulse repetition frequency. The Doppler signal is processed into an extensive Doppler signal with an estimation technique. Especially, Doppler frequency compression is free from aliasing of the pulsed ultrasonic Doppler system.
Simulation of the effects of high-frequency hearing impairment in noise and estimation of the effects of hearing aids incorporating filtering of low-frequency components in the presence of background noise
Authors: Rawool Vishakha W.
Published: October 2005

An attempt was made to simulate the effects of high-frequency hearing impairment in “noisy” situations. It was assumed that the difficulties experienced by individuals with high-frequency hearing impairment in noise can be simulated, if in addition to the spectrum shaping of speech, the noise was also filtered. Such filtered noise, containing mainly low-frequency elements was expected to effectively mask the only low-frequency cues available after filtering the high-frequency elements from the speech spectrum. The speech discrimination ability of normal individuals was also assessed in a condition where both the speech and noise were low pass filtered at 500 Hz. This condition was presented to simulate the effects of hearing aids incorporating noise-suppressor switches that filter out both the noise and the low-frequency speed cues in the presence of background noise. Detailed analyses of the results will be discussed.
Slow waves, boundary conditions, and reflection and transmission coefficients
Authors: Plona Thomas J. , Johnson David Linton , D'Angelo Ralph
Published: October 2005

Biot's theory of acoustics in fluid-saturated porous media predicts a “fast” compressional wave, a unique “slow” compressional wave, and a shear wave. Experimental work by the authors demonstrated that Biot theory correctly predicts the measured speeds of all three waves. The boundary conditions, however, which must include an additional condition beyond those postulated by elasticity theory to describe the “slow” wave, have not been adequately studied experimentally. Measurements have been made of reflection and transmission coefficients (including velocities and attenuations) at the plane interfaces between water and a water-filled porous slab at ultrasonic frequencies. At these frequencies, the slow wave is a propagating mode and is readily measurable, but only if the porous medium has a carefully chosen pore structure. By using either an open pore surface or a sealed (thin layer of epoxy) surface, the excitation of the slow wave can be controlled. Slow waves are generated in the solid even when the interface is sealed. These measurements are compared to predictions using the “open” and “sealed” interface boundary conditions of Deresiewicz and Skalak.
Coherent potential approximation for Biot's equations and generalized Gassmann's equations
Authors: Berryman James G.
Published: October 2005

After calculating the scattering coefficients for a spherical scatterer in a poroelastic medium, the coherent potential approximation (CPA) must be used to compute estimates of the effective coefficients in Biot's equations when the medium is more inhomogeneous than is usually assumed in the standard derivations such as that due to Gassmann. The resulting formulas are generalized Gassmann's equations within the single scattering coherent potential approximation. These results have been shown to agree with the most general forms of such coefficients as established by Brown and Korringa. [Work supported by DOE Office of Basic Energy Sciences.]
A comparison between wave propagation in water-saturated and air-saturated porous materials
Authors: Albert Donald G.
Published: October 2005

The classical Biot theory predicts the existence of three waves that can propagate in a fluid-saturated porous material: a fast compressional wave, a slow compressional wave, and a shear wave. Through use of this theory, propagation characteristics within water-filled and air-filled materials were compared in the 10-Hz to 10-kHz band. Numerical calculations show that the ratio of fluid to solid motion for the slow compressional wave is around 1 in water-filled sand, but greater than 100 in air-filled sand. In addition, calculations of plane-wave transmission from a fluid into a fluid-saturated porous solid show that when the fluid is water, the displacement potential transmission coefficient for fast compressional waves is about 100 times larger than the coefficient for slow compressional waves. When the fluid is air, however, the fast wave transmission coefficient is about 100 times smaller than the slow wave coefficient. These results justify the approximate modeling of saturated undersea sediments as viscoelastic materials (ignoring motion in the pores) and air-filled soils as modified fluids (ignoring motion in the frame). [Work supported by the Directorate of Research and Development, U. S. Army Corps of Engineers, Project 4A161102AT24.]
Experimental investigation of models of sound wave propagation in air-saturated porous media
Authors: Champoux Yvan , Stinson Michael R.
Published: October 2005

Several theoretical models of sound wave propagation in air-saturated porous material are studied making use of precise measurements of both physical (porosity, flow resistivity, tortuosity) and acoustical (characteristic impedance, propagation constant) parameters. The aspects covered include the use of shape factors and the adequacy of frequency-dependent dynamic viscosity functions. Experimental results on both model materials and real material are used, with special attention to the low-frequency/high-flow resistivity regime for solid frame porous materials.
Role of fluids on ultrasonic attenuation in porous materials
Authors: Nihei Kurt T. , Cook Paul J. , Myer Larry R. , Suárez-Rivera Roberto
Published: October 2005

Measurements of P- and S-wave attenuation in saturated alundum and Berea sandstone were made under confinement with various pore fluids: water (1 cP, 72 mJ/m2), n-decane (0.92 cP, 28 mJ/m2), glycerol (1490 cP, 64 mJ/m2), and silicone oil (913 cP, 21 mJ/m2). The results for alundum indicate that attenuation for both P and S waves is controlled by viscosity: Qglycerol−1 > Qoil−1 > Qwater−1 > Qdecane−1. Attenuation in alundum is qualitatively consistent with Biot dissipation and is much less sensitive to confinement, presumably because it is composed of well-sintered contacts. P- and S-wave attenuation in Berea sandstone is not purely viscosity dependent: Qwater−1 > Qglycerol−1 > Qdecane−1 > Qoil−1. Glycerol at 25 °C has the same viscosity as silicone oil but results in P- and S-wave amplitudes that are 55% and 30%, respectively, smaller than those for silicone oil. This result is not consistent with Biot dissipation but may be explained by differences in surface tension or hydrogen bonding at the contacts between grains. [Work supported by Dept. of Energy.]
Juergen Tonndorf: The last great hearing scientist/otologist
Authors: Abramson Maxwell
Published: October 2005

Juergen Tonndorf was born in Goettingen, Germany and received his medical education and training in otolaryngology at the University of Keil. He served as captain in the submarine corps of the German Navy during World War II, beginning his scientific career studying the effects of increased atmospheric pressure on hearing. His career in the United States began in 1947 at the school of Aviation Medicine, Randolph Field, Texas, then as research professor in the department of Otolaryngology at the University of Iowa. He joined the department of Otolaryngology at Columbia University in 1962, establishing an auditory physiology laboratory which he directed until his retirement in 1982. Dr. Tonndorf combined a uniquely broad scientific curiosity and an extensive knowledge of both scientific and clinical subjects with an enormous capacity for hard work to produce a rich body of scientific achievements that included 157 publications on subjects that ranged from cochlear mechanics to the inhibition of tinnitus. He collaborated with a varied and distinguished group of hearing scientists, engineers, and physicians including many students with no previous research experience. Juergen Tonndorf was the scientist's clinician as well as the clinicians's scientist. He brought us together and inspired us. He is very much needed today.
Cochlear pathways
Authors: Duvall Arndt J.
Published: October 2005

Active and passive barriers and pathways into and within the cochlear duct were of interest to Tonndorf. He investigated them with vital stains. Subsequent studies with ultramicroscopy, tracers, immunohistochemistry, and biochemistry have resulted in few changes in our concepts of cochlear duct compartmentalization. The basilar membrane presents almost no barrier, whereas the tight junctions between cells at the endolymphatic surface do. The inner hair cells are less penetrable than are the outer hair cells. The stria vascularis is the most isolated cell grouping within the cochlear duct. Selective active transport makes many general statements suspect. The differences in the penetration rates of Gentamicin and the more cochleotoxic Neomycin may, in part, explain their relative cochleotoxicity. The rate of penetration of Neomycin into the cochlear duct is more rapid than that of Gentamicin. However, the ultimate distribution of both antibiotics is similar and does not contradict Tonndorf's observations.
Frequency dependence of auditory temporal resolution
Authors: Kishon-Rabin Liat , Boothroyd Arthur
Published: October 2005

Thresholds for pure tones, embedded in a 600-ms white noise burst containing a brief gap, were measured as a function of tone frequency and gap duration in five normal adults. This paradigm avoided both spectral splatter, associated with brief tones, and temporal artefacts, associated with external filtering. The data relating tone threshold to gap duration were modeled by two processes occurring during the gap: (a) persistence of sound energy in the cochlea (analogous to room reverberation); and (b) partial recovery from perstimulatory adaptation. The first process is dominant for short gaps and causes true masking—threshold recovery being exponential in the energy domain (i.e., linear on a decibel scale). The second process is dominant for longer gaps and is assumed to represent electrochemical activities at cochlear or retrocochlear levels, threshold recovery being exponential on a decibel scale. Time constants for the first process were a few ms and were highly dependent on frequency. Time constant for the second process was of the order of 90 ms and was independent of frequency. [Work partially supported by NIH Grant No. 17764.]
Automatic sentential vowel stress and sentence final pitch slope labeling
Authors: Hieronymus James L. , Williams Briony J.
Published: October 2005

Vowel stress in English is marked by pitch rise-falls, energy, and duration. After studying the stress marking strategies of 15 talkers of American English, an algorithm was devised that labels vowels in continuous speech with three levels of stress. The algorithm is based on a model for combinations of pitch rise-falls, relative energy, and duration. The pitch of voiced regions is characterized as rising, falling, or steady. Sequences of three regions are examined to find the pitch rise-fall patterns that signal stress. If the energy of the vowel is within 8 dB of the maximum in the utterance, it is considered energy stressed. Duration is corrected for pre-pausal effects. If two out of three cues are present, then the vowel is labeled stressed. The algorithm was tested on sentences of American and British English and found to perform very well. Detailed results will be presented, and performance of the final pitch slope determination will be discussed relative to a perceptually labeled database.
The role of acoustic phase in syllable categorization and discrimination
Authors: Harsin C. A. , Porter R. J.
Published: October 2005

Relative timing in speech has been shown to underlie the perceptual stability of syllable structure over transformations in speech rate and stress. To investigate this relationship in the acoustic domain, synthetic speech signals corresponding to the V + C + V sequence /ipi/ were created. The durations of the initial and final V's and the position of a burst corresponding to a /p/ were manipulated. Subjects were asked to judge whether particular signals had the structure /i pi/ or /ip i/. Subjects also were asked to perform a discrimination task among the same signals. A clear monotonic relationship was found between burst position phasing in the final-vowel + intervocalic interval continuum and syllable categorization, but the discrimination function does not show a peak at the category “boundary.” Thus, whereas the burst-phase relationship appears to underlie syllable categorization, it does not seem to support discrimination: Syllabic judgments and discrimination functions may be based on different psychoacoustic aspects of the signals. [Work supported in part by Louisiana Lions' Foundation and UNO Chancellor's Fellowship.]
Relationships between duration and variability measures of children's speech
Authors: Smith Bruce L.
Published: October 2005

In recent years, several studies have found that young children often have longer speech segment durations than older children and adults do and that young children are also typically more variable across repetitions of stimuli. It has generally been suggested that both of these findings are indications of increased speech motor control as children become older. If both measures are reflections of neuromotor maturation of the speech production system, it seems that they should be highly correlated with one another and that they should lead to similar conclusions about the speech production abilities of individual subjects. However, data from children ranging from about 3–10 years of age indicate that duration and variability measures often are not very strongly correlated. For example, children who have shorter segment durations do not always show less variability than children with longer durations, and children with greater variability do not necessarily have longer segment durations. In addition, it appears that children may achieve adultlike duration values before measures of their variability become adultlike. Possible implications of these findings regarding children's speech timing control will be discussed.
Sound reduction of panels and walls using small-scale models
Authors: Walker Keith W.
Published: October 2005

The ability to make sound reduction and transmission loss measurements on small-scale wall models is an attractive proposition for R&D work. Using the impulse response technology offered by the Grozier GTS-800 acoustical measurement system, measurements of insertion loss were made on single free-standing panels at normal incidence and at 45 deg. In the range 4 to 40 kHz, the results were in good agreement with mass law theory. Additional measurements of a wall between two 18-scale offices also showed characteristics similar to full-scale walls, but in the range 1 to 40 kHz.
A link between music perception and speech production
Authors: Deutsch Diana
Published: October 2005

A two-tone pattern has recently been described that has certain paradoxical properties. For a given listener, it is heard as ascending when played in one key, but as descending when played in a different key. Further, when the pattern is played in any one key, it is heard as ascending by some listeners but as descending by others [D. Deutsch, J. Acoust. Soc. Am. Suppl. 1 80, S93 (1986); Mus. Percept. 3, 275–280 (1986); Percept. Psychophys. 42, 563–575 (1987)]. This paper demonstrates a link between perception of this pattern and the speech production of the listener. First, a significant correlation is shown between the way this pattern is perceived and the range of fundamental frequencies in the listener's spontaneous speech. Second, a significant difference in perception of this pattern is shown between a group of listeners who grew up in the south of England and those who grew up in California: Where the English group tended to hear the pattern as ascending, the Californian group tended to hear it as descending, and vice versa. These findings indicate that the same, culturally acquired, pitch template influences both speech production and also perception of this musical pattern.
An experiment in state-space acoustic control of a plate excited by steady disturbances
Authors: Rubenstein S. P. , Saunders W. R. , Robertshaw H. H.
Published: October 2005

A demonstration of active acoustic control of radiation from a simply supported plate is presented. First, open-loop characteristics of a 0.5×0.6-m rectangular plate are shown to validate the assumption of nearly simply supported boundary conditions. Next, a state-space design method for closed-loop control of acoustic radiation is discussed. The design procedure is used to generate controllers that reject either narrowband or broadband point-force disturbances of the plate's first four resonant modes. Simulation and experimental results are provided to illustrate the difference between active vibration control and active acoustic control. Radiation filters, derived from the radiation efficiencies of the modal basis, are implemented for these subsonic acoustic control experiments. During the acoustic cost experiments, the controller places a priority on actively damping only those modes that are contributing significantly to the acoustic radiation. The results of this active acoustic control experiment validate the built-in selectivity of this control approach for suppression of structural acoustic radiation. The theoretical foundation for these experimental results is given in a companion paper, “Active acoustical control of broadband structural disturbances” [J. Acoust. Soc. Am. Suppl. 1 88, S 149 (1990)]. [Work supported by DARPA/ONR.]
Visualizing the effect of active control on structural vibrations using vein diagrams
Authors: Covey Joel F. , Yen Nai-chyuan , Corsaro Robert D. , Ng Kin W. , Ross Chris , Yoder Tim
Published: October 2005

The vein diagram is a new method of visualizing the time-frequency characteristics of acoutic signals [Yen et al., J. Acoust. Soc. Am. 87, 2359–2370 (1990)]. It is based on the modified Wigner distribution function. Previously this technique has been used for analysis of acoustic scattering including the inverse scattering problem. Here, the vein diagram, is employed as a visualization tool to display the effect of a surface mounted projector on acoustic scattering from a submerged complex structure. The projector is part of an active vibration control system. The important feature of this technique is that is allows distinct observation of the influence of the control signal on each of the various types of structural vibrations.
Generation and decay of the acoustically significant sub-surface bubble plumes
Authors: Monahan Edward C.
Published: October 2005

The generation of the bubble plumes found just beneath the sea surface is identified with spilling wave crests (stage A whitecaps). The decay of the initial, acoustically radiating, α plumes into β plumes (with their accompanying stage B white caps), and then into γ plumes, and, finally, the dispersion of the bubbles in these diffuse plumes into the background bubble layer is described, and bubble spectra are associated with each stage of plume evolution. An explicit model relating the flux of bubbles into the sea to the fraction of the sea surface covered by whitecaps is provided. The near-surface concentration of large bubbles (radii greater than 100 μm) inferred from this model for 11- to 13-m s−1 winds is in general agreement with the published field observations [P. A. Kolovayev, Oceanol. 15, 659–661 (1976); B. Johnson and R. C. Cooke, J. Geophys. Res. 84, 3761–3766 (1979); A. L. Walsh and P. J. Mulhearn, J. Geophys. Res. 92, 14553–14565 (1987)] obtained using optical bubble detectors under similar wind conditions. [Work supported by ONR.]
Effect of vocal fold radii on pressure distributions in the glottis
Authors: Scherer Ronald C. , Guo Chwen G.
Published: October 2005

Modeling laryngeal aerodynamics requires specification of glottal geometry. This study examines primary effects of glottal inlet and outlet radii. A velocity-pressure integrated finite-element method produced glottal pressure profiles for three radius conditions: completely rounded vocal folds (the edematous condition}; vocal folds with sharp corners; and vocal folds with realistic radii. Converging, uniform, and diverging glottal shapes were used with a Reynolds number of 800 and a minimum glottal diameter of 0.02 cm. For the diverging glottis, sharp corners produce unrealistically large local negative pressure spikes due to convective acceleration. For convergence, the expansion curvature of realistic radii creates pressure recovery hear the glottal exit, producing a location of negative pressure. This may reinforce the motion phase difference between the upper and lower glottal regions due to a net positive pressure upstream and a net negative pressure downstream. For the completely rounded vocal fold, both positive and negative pressures appear upstream in the glottis, with a potential tendency to inhibit vocal fold motion. [Work supported by NIH R01 DC00157 to The University of Iowa.]
A new algorithm for speech synthesis based on vocal tract modeling
Authors: Lin Qiguang , Fant Gunnar
Published: October 2005

A new algorithm for articulatory speech synthesis is described in this paper. The algorithm constitutes two main parts: a detailed frequency domain modeling of the vocal tract and a data transform from the frequency domain to the time domain. A computer model was developed to simulate the vocal tract acoustics. The model incorporates all known important components of the vocal system and computes the transfer function between the lip/nostril output and the acoustic source. By decomposing the obtained transfer function into its numerator and denominator, frequencies and bandwidths of resonances (and of antiresonances, if any) can be determined. The transfer function can next be written as a partial fraction expansion series in terms of calculated residues at the poles and can be approximated by retaining the first few terms in the series. Usually, each of these terms is a second-order module and corresponds to an elementary formant resonance. Formants are thus connected in parallel. The time domain output is obtained by the inverse Laplace transform. Compared with other synthesis methods, this vocal-tract-oriented synthesis strategy has a number of advantages. For instance, the frequency dependency of loss elements of the vocal tract is preserved and accurate frequency responses can be reproduced. It is also computationally efficient relative to a direct convolution method. Examples of spectrum matching are presented to discuss the properties of the proposed algorithm. The algorithm has been incorporated in an articulatory-based speech synthesis system currently under development at KTH. [Work supported by grants form the Swedish Board for Technical Development (STU).]
Laryngeal intramuscular pressures
Authors: Cooper Donald S. , Pinczower Eric , Rice Dale H.
Published: October 2005

Simultaneous measurements of isometric force, intramuscular fluid pressure, and EMG were made in the vocal fold muscles of anesthetized dogs in vivo, in situ during supramaximal stimulation of recurrent nerve. Resting interstitial fluid pressure was usually slightly positive, with some local variation within the muscle. Covariation of isometric force and pressure was studied in twitch and tetanic contractions and over a range of subtetanic stimulation frequencies. Most often tetanic pressure rises were smaller than in limb muscles, probably because of vocal fold geometry and tissue compliance. The intramuscular pressure rises produced during tetanic contraction were of a magnitude known to obstruct blood flow and produce peripheral fatigue. Analysis of relations between pressure and force records demonstrated that, unlike raw EMG, intramuscular pressure variations correlated closely to isometric force in their time course. Relation of findings to vocal fold tissue mechanics and phonation, and possible application as a measurement of muscle activity in speech are considered. [Work supported by Voice Foundation.]
A study of the variation of glottal pulse shape with fundamental frequency by inverse filtering of the speech wave
Authors: Lobo Arthur P. , Ainsworth William A.
Published: October 2005

The acoustics of speech production is based on the concept of a source and a filter function. In current models the source of voiced sounds is represented by a quasiperiodic succession of pulses of air emitted through the glottis, as the vocal cords open and close, and the filter function is assumed to be linear and short-time invariant. The source term or the acoustic excitation produced by the larynx is subject to variation in speech produced by a natural speaker. The analysis described here forms part of an investigation into variation of the shape of the glottal volume velocity waveform as a function of fundamental frequency. The deconvolution of the speech wave, i.e., separation of the glottal pulse signal from the vocal tract impulse response, is done by inverse filtering the speech signal using a filter derived in the closed phase of the glottal cycle. Linear predictive covariance analysis is used in estimating the inverse filter coefficients. A linear relationship between the glottal parameters, open phase, opening duration, and closing duration with time period is shown. A model is then proposed for the variation of these glottal parameters with fundamental frequency.
Multidimensional perceptual spaces for vocal breathiness and roughness
Authors: Kreiman Jody , Gerratt Bruce R.
Published: October 2005

Although many authors have suggested breathiness and roughness are multivariate characteristics of voices, these qualities are always measured using unidimensional scales, and their perceptual structure has not been formally investigated. In the present research, five experienced listeners heard all possible pairs of 18 pathological voices and judged how similar voices were in their breathiness levels. At a separate session, they judged how similar the same voices were in roughness. Although breathiness and roughness ratings were not highly correlated, multidimensional scaling analyses suggest both qualities may be represented by nearly identical two-dimensional spaces in which stimuli are arranged in a circle. These findings suggest breathiness and roughness are not independent unidimensional qualities, but instead are better treated as a single multidimensional “feature” of voices. Use of unidimensional scales to measure a multidimensional construct may be responsible for some of the unreliability often associated with unidimensional ratings of breathiness and roughhess.
Analysis of thickness “resonances” when scattering from submerged elastic shells at high frequency
Authors: Werby M. F. , Gaunaurd G. C.
Published: October 2005

When sound scatters from an elastic spherical shell submerged in water at suitably high frequencies, it is noted that backscattered returns, when plotted as a function of frequency, display unusually large signals over a broad frequency range. The point at which this occurs is a function of both shell thickness and material properties. This effect is illustrated for five elastic materials and three shell thicknesses (15 distinct examples). Simple expressions that predict the location of the strong returns in ka (the ka values, where k is the wave number in the fluid and a is the radius of the sphere) for all 15 examples are then given. The results are also examined in partial wave space and are explained within the context of flexural waves and half-integral wavelengths that correspond to the thickness of the sphere.
Flow through the reed channel of a single-reed wind instrument
Authors: Hirschberg A. , Houtsma A. J. M. , Wijnands A. P. J. , Dane H. J. , Gilbert J.
Published: October 2005

An analytical model of the stationary flow in the aperture of a single reed woodwind instrument is presented [A. Hirschberg et al., Acustica 70, 146-154 (1990)]. For large reed apertures, flow separation occurs on the entire channel length and the volume flow can be calculated by applying Bernoulli's equation with a yens contracta factor of 0.5. For ratios of reed channel length/height of four or more, reattachment occurs and the volume flow is predicted within 5% by using a modification of the theory of Ishizaka and Matsudaira [“Fluid mechanics consideration of vocal chord vibration,” SCRL Monograph 8, Speech Comm. Res. Lab., Inc., Santa Barbara, CA (1972)]. The force on the reed due to the low pressure in the separation region increases with the channel height because the length of the separation region increases with the height. Hysteresis (due to nonlinearity or unsteadiness) in the transition from a fully separated flow to a reattached flow is responsible for oscillations observed in the absence of acoustical feedback. In a similar way, hysteresis in the Bernoulli force may explain oscillations of lips or vocal chords.
Noise in environmental assessment and licensing
Authors: Hale Marlund E.
Published: October 2005

An increasing emphasis on environmental issues, including noise, results in numerous requirements that must be addressed during planning for new or expanded facilities. Not only must the federal and state agency permitting and environmental assessment requirements be included, but the many new and upgraded local requirements also need careful attention. A systematic approach to licensing will avoid many of the headaches and delays often encountered. By knowing the lead agencies' permitting process and by allowing adequate time and resources, it is often possible to both prepare the needed documents and comply with the applicable environmental noise requirements more efficiently. While permits are rarely required specifically for noise, the sources of noise requirements in permitting are discussed. The types and extent of information typically needed in environmental impact statements and the trend to monitor for compliance with commitments are reviewed.
Cavitation from diagnostic ultrasound
Authors: Roy R. A. , Holland C. K. , Apfel R. E. , Crum L. A.
Published: October 2005

The possibility of cavitation as a consequence of short-pulse diagnostic ultrasound was predicted theoretically in 1982 by Apfel [R. E. Apfel, Br. J. Cancer 45 (Suppl. V), 140–146 (1982)] and Flynn [H. G. Flynn, J. Acoust. Soc. Am. 72, 1926–1932 (1982)]. Experiments were performed in ultraclean water to test this hypothesis under conditions comparable to those found in clinical ultrasound examinations. The development of an ultrasensitive acoustic backscattering technique that utilizes 30-MHz pulsed ultrasound has made it possible to detect the presence of transient cavitation due to microsecond-length pulses [R. A. Roy et al., J. Acoust. Soc. Am. 87, 2451–2458 (1990)]. This “active cavitation detector” has been employed in an in vitro study to probe the potential of cavitation production by a clinical ultrasound imaging system, specifically, an HP 77020A. Two calibrated, phased-array HP imaging transducers with and 5.0-MHz operating frequencies were driven in M-mode (single cycle) and Doppler mode (4 cycles) by the HP imaging system. Hydrophobic polystyrene spheres with an average diameter of 0.245 μm served as potential cavitation nuclei. Cavitation was detected in the water at 2.5 MHz in both M-mode and Doppler mode at a peak negative acoustic pressure of 1.1 MPa or greater. Insonfication at 5.0 MHz in either mode did not produce a detectable amount of cavitation, even with peak negative pressures as high as 1.2 MPa. [Work supported by the National Institutes of Health, Grant No. 5-RO1-CA-39374-05.]
Localization and binaural detection with monaural and binaural amplification
Authors: Koehnke Janet , Zurek Patrick M.
Published: October 2005

As part of a study comparing the effects of monaural and binaural amplification, localization and binaural detection was measured for two groups of listeners with bilateral, sensorineural hearing losses. The subjects in one group have moderate-to-severe, flat losses; the subjects in the other group have moderate-to-severe, high-frequency losses. Masking level differences (MLDs) were obtained using speech-spectrum noise maskers and octave-band speech-spectrum noise targets. Localization in the horizontal plane (nine locations from − 90 to + 90 deg) was measured using three-word phrases in an anechoic and a reverberant environment. Both binaural detection and localization performance vary between subject groups and within subject groups, and all subjects generally have smaller MLDs and poorer localization than normal-hearing listeners. The subjects with flat hearing losses usually have better localization and detection with binaural amplification than monaural amplification, and the high-frequency hearing-loss subjects tend to have better overall performance than the flat hearing-loss listeners. The data for all the subjects combined show a tendency for RMS errors in localization to decrease as MLDs increase. [Work supported by NIH.]
A digital hearing aid that compensates loudness for impaired listeners and its performance
Authors: Suzuki Yoiti , Asano Futoshi , Hayashi Tetsuya , Sone Toshio , Kakehata Seiji , Satake Mitsuaki , Ohyama Kenji , Kobayashi Toshimitsu , Takasaka Tomonori
Published: October 2005

A method of designing a digital hearing aid for speech signal suitable for sensorineural hearing loss with a narrow audible range is proposed. First, the input signal is divided into short time blocks (8 ms) and its spectrum is analyzed using short-time FFT. Then the frequency-gain characteristic of the digital filter for the block is determined using loudness compensation functions that describe the relation between the loudness of normal listeners and that of impaired listeners, so that the input signal within the block is projected on to the dynamic range of the subject. This algorithm is realized using a digital signal processor (DSP). The results of monosyllabic intelligibility tests to evaluate the performance of the system for six sensorineural impaired listeners are also shown. [Work supported by Grant-in-Aid for Scientific Research from the Ministry of Education and the Ministry of Public Welfare.]
Effect of the time constant on the performance of a multichannel compression hearing aid
Authors: Yund E. William , Buckles Krista M.
Published: October 2005

Previous research [E. W. Yund, H. J. Simon, and R. Efron, J. Rehab. Res. Dev. 24, 161–180 (1987)] demonstrated the effectiveness of an eight-channel compression hearing aid for individuals with sensorineural hearing loss (SNHL) for speech discrimination in a background of speech-band noise. In the present study, the averaging time window of the Robinson and Huntington [J. Acoust. Soc. Am. 54, 314 (1973)] compression was varied between 5 and 100 ms. Constant intensity, sinusoidally modulated, and square wave modulated speech-band noise were used. Nonsense syllable discrimination of 15 subjects with SNHL showed no significant change with averaging time in constant intensity noise. Only the use of the square-wave modulated noise revealed an effect of time window on discrimination, better discrimination with the two longer time windows (36.8 and 100 ms) as compared to the two shorter time windows (5 and 13.6 ms). These results indicate that a hearing-impaired patient's success with such a multichannel compression hearing aid should not be strongly dependent on the time window parameter. [Work supported by VA Rehab. Res. and Der. Ser.]
Listening bands in adults: Implications for studying auditory selective attention in infants
Authors: Bargones Jill Y. , Werner Lynne A.
Published: October 2005

The original listening band experiments [G. W. Greenberg and W. D. Larkin, J. Acoust. Soc. Am 44, 1513–1523 (1968)] were replicated using a one-interval paradigm. Modifications designed to adapt the procedure for use in studying auditory selective attention during infancy were also implemented. A 1000-Hz primary tone was presented on 75% of signal trials at a level that allowed for approximately 80% correct detection. Probe tones of 880 Hz and 1120 Hz were presented equally often on the remaining 25% of the trials at the same level as the primary tone. All signals were 300-ms in duration with 16-ms rise/fall times; signal and no-signal trials were each presented on 50% of the trials. Signals were presented in quiet or in the presence of a continuous background noise (300–1800 Hz; 10 dB N0). Trials were presented with or without a visual cue signalling trial onset and with or without feedback to the observer concerning the correctness of the response. Results showed that adults detected the primary tones about 80% of the time but probe tones were usually detected at near chance levels. When probe tones were subsequently presented as the sole signal frequency for a block of trials, they were in general detected near 80% of the time. The results suggest that adults listen selectively for the primary frequency. Complications of the procedure include controlling for response bias and in quiet, controlling for threshold microstructure. [Work supported by NIH DC00396.]
Recent developments in acoustical imaging applied to vibrating structures
Authors: Wetta P. , Beguet B.
Published: October 2005

METRAVIB R.D.S., the French leader in the field of dynamics in mechanics (vibrations-materials-acoustics) is one of the pioneers of the development and application of acoustical imaging techniques in France (intensitymetry-holography-backpropagation). Associated today with the main French and European industrial groups, METRAVIB R.D.S. has validated its methods in various real industrial applications. This paper presents an overview of these applications either on fixed or moving sources, focusing on theoretical foundations of the selected resolution methods, and knowledge acquired through these techniques. Moreover, this paper deals with the most recent applications to related fields such as structural intensitymetry techniques, and experimental investigation of periodic stiffened structures. The conclusions are devoted to developments in progress, concerning phenomena studies of fluid-structures coupling by means of radiation or scattering fields exploration.
Acoustic imaging of ribbed plates
Authors: Pate Anna L. , Zhang Yong , Wei Ruey-Chang
Published: October 2005

In recent years, a number of measurement techniques have been developed for structural intensity in various systems such as beams and plates. These techniques are based on motion measurements by either contact sensors (e.g., accelerometers) or noncontact methods (e.g., near-field acoustic holography). It has been shown that it is possible to calculate the two-dimensional structural intensity vector in plates on the basis of acoustic imaging. However, the method is very sensitive to errors occurring in the image of the reconstructed surface motion. Thus the goal of this paper is to investigate the accuracy of the surface velocity when calculated from two different acoustic imaging methods: near-field acoustic holography and a new acoustic imaging method based on surface motion expansion in a set of orthonormal basis functions. The imaging is performed by using a singular value decomposition method along with additional constraints to calculate appropriate coefficients in this expansion. Comparisons between imaging of thin homogeneous plates and ribbed plates will be discussed. [Work supported by ONR.]
The measurement of plane-wave reflection coefficients by using two-dimensional spatial transforms
Authors: Hu Ziqiang , Bolton J. Stuart
Published: October 2005

Recently, Tamura has proposed a method for measuring plane-wave reflection coefficients at arbitrary angles of incidence [M. Tamura, Proceedings of INTER-NOISE 89, 1085–1088 (1989)]. The procedure is based on measuring the complex pressure distribution on two parallel planes located between a compact source and the reflecting surface. By spatially transforming the measured data into the wave-number domain, the sound field may be decomposed into upward- and downward-going plane-wave components. The reflection coefficient is then simply the ratio of the complex amplitudes of the upward- and downward-going plane-wave components at a particular angle of incidence. In this presentation, the fundamentals of the technique will be reviewed followed by the presentation of new results that illustrate the effect of source type and location on measurement accuracy and practicality. In addition, it will be shown that acceptable accuracy may be achieved using “reasonable” spatial aperture sizes and sample rates only if the spatial data are transformed using a Hankel transform based on a cylindrical geometry rather than a two-dimensional Fourier transform based on a rectangular geometry. Finally, it will be shown that the use of a generalized Hankel transform can eliminate the usual requirement that the source radiate a cylindrically symmetric sound field. Both simulated and model-scale experiments will be used to show that the technique yields accurate results when the source type and transform geometry are chosen appropriately.
An AXB discrimination test of McGurk effect stimuli
Authors: Rosenblum Lawrence D. , Saldaña Helena M.
Published: October 2005

In the McGurk effect, visual information for articulation affects auditory judgments of speech syllables. Recently, questions have arisen concerning whether this effect is perceptual or post-perceptual in nature [L. Rosenblum, J. Acoust. Soc. Am. Suppl. 1 84, S159 (1988)]. One way of exploring this issue is to determine the degree to which a visually influenced syllable sounds like its auditory-alone counterpart. The hypothesis was made that, if the McGurk effect is truly perceptual in nature, then observers should have difficulty discriminating the sound of a visually influenced syllable from an audio-alone syllable. Two experiments were conducted that implemented an AXB matching procedure to determine if a visually influenced /va/ (audio /ba/; video /va/) and an audio-visual consistent /va/ (audio /va/; video /va/) sounded equally like an audio-alone /va/, as is predicted by the above hypothesis. The results of the first experiment indicate that observers are more likely to match the audio /va/ to the audio-visual consistent /va/ than to the visually influenced /va/. However, the results of the second experiment suggest that this matching might be based simply on similarities in acoustic characteristics and not on differential phonetic categorizations of the stimuli.