bluuebunny/crossref_metadata_2025 huggingface dataset:
Design and evaluation of underwater hydrophones with self-noise suppressing structures
Authors: Im Jong-in , Roh Youngrae
Published: October 2005

Underwater hydrophones are mounted on a vibrating surface in common applications and their performance is usually degraded by the interference of various external noises. This paper describes the design and evaluation of a self-noise suppressing underwater hydrophone without sacrificing their performance as a receiver of external acoustic signals. The hydrophone under consideration is an array sensor composed of several tonpilz transducers. To increase the insensitivity to external noises, first, effects of transfer paths of the external noises on the performance of the hydrophone is checked through computer simulation with finite-element methods (FEM). Second, several geometrical variations are imposed on the original structure of a hydrophone. The variations include additional air pockets and acoustic walls that work as acoustic shields or scatterers of the noises. Results show that the transverse noise induced by the outside water flow is isolated most effectively when a thin compliant (damping) layer combined with two air pockets is inserted into the circumference of the hydrophone nose. The noise level is reduced to about 40% of that of the original structure. When the transducer array is surrounded by two thin sandwich compliant layers with an acoustic wall the noise level is as low as 25% of that of the original model. Further, the optimum acoustic impedence of the compliant layer is estimated to be either less than 1 Mrayl or more than 4 Mrayl. The structure developed in this paper allows the hydrophone to be mounted on vibrating surfaces with a better signal-to-noise ratio.
Beating in propeller singing
Authors: Okajima Masahiko , Toshio Maeda
Published: October 2005

The unsteady nature of the shear flow leaving the trailing edge of a propeller blade may give rise to a correlated discrete vortex structure that is also a source of a significant noise, so-called propeller singing. It is well known that beating phenomena occur during the propeller singing. To understand this phenomena, the acoustic field generated by the cavitation-free propeller was investigated. The work started with an analysis of elementary kinematics of sound radiation by the propeller, considering the rotational motion of the blades relative to acoustic media. Then the acoustic spectrum due to the blade force resulting by inhomogeneous inflow was considered. Using the analysis, frequency domain expressions were obtained to examine the beating. Directivity was also calculated to study the nature of the propeller sound field.
Seawater effects on cavitation noise
Authors: Kamiirisa Hikaru , Miike Toshiyuki
Published: October 2005

Through the research on an air-mixed water injector for reducing the cavitation noise of a ship’s propeller it had been noticed that the difference of the bubble radius between fresh water and seawater might have an influence on its radiated noise. In order to investigate the influence, an experimental apparatus was newly manufactured, and then the cavitation noise was measured with both fresh water and seawater. As a result of the measurement, it was found that the inception cavitaiton number in seawater was 15% lower than that in fresh water, and also that the noise spectra in fresh water had the same characteristics as those obtained at a 15% lower cavitation number in seawater. An ordinary simulation test with fresh water tended to overestimate the full scale propeller cavitation noise in the sea by about 20%. Simulating the motion of bubbles by the Rayleigh–Plesset equation by considering the surface tension, viscous effect, and the pressure inside the bubble, it was predicted that the ratio of gas content would cause the different characteristics of the cavitation noise between fresh water and seawater.
Application of the active control to improve the sound insulation of double-pane windows
Authors: Watanabe Naoki , Yano Hiroo , Tachibana Hideki
Published: October 2005

Although double-pane windows are being widely used for airborne sound insulation of building facades, they generally have sound insulation deficiencies at low frequencies due to the resonance between two panes coupled by air space. As a method to improve this very serious defect, the application of active control technology has been investigated experimentally. At first, the applicability was examined by scale model experiments in which, by setting a primary noise source in a model room, the possibility of reducing the sound transmitting through a double-pane window was examined by using the usual adaptive active control technique based on the filtered-x LMS algorithm. Regarding the secondary sources, the method using loudspeakers (inside of the source room and in the air space between the double panes) and that using vibration actuators attached on the pane were investigated. As a result, it has been found that the sound insulation deficiencies in the frequency range of the resonance can be recovered by use of the active control. In order to examine the practical applicability of this control method, field experiments were performed by using real buildings.
Flanking paths of a floor–ceiling assembly in wood frame constructions: Measurements and predictions using statistical energy analysis
Authors: Nightingale T. R. T. , Steel J. A.
Published: October 2005

This paper examines the net sound isolation of a floor–ceiling assembly and the supporting walls that might be found in wood frame apartment constructions. The specimen was constructed in the flanking transmission facilities of the National Research Council of Canada. Measured results show that the presence of two load-bearing walls introduce flanking paths that can reduce the net sound isolation of an FSTC 59 floor–ceiling assembly by 6 FSTC points. Various methods of modeling the structural transmission at the floor–wall intersection are presented. The models indicate that beams at the joint (i.e., joist header, head, and sole plates) introduce significant high-frequency joint attenuation due to rotational and shear inertia. Comparison between measured and predicted velocity level differences indicates that floor–wall intersection in standard wood frame constructions should be modeled as two corner joints [(1) upper party wall to floor decking, (2) floor decking to lower party wall] sharing a common plate (the floor decking). A statistical energy analysis model is used to show that the dominant flanking path is the floor decking to the load bearing wall below, while the wall–wall flanking path is much less important. The predictions are shown to be in good agreement with measured results. a)Present address: Shuller International, 10100 West Ute Ave., Littleton, CO 80127.
Degradation of wood stud wall sound isolation by electrical outlet boxes
Authors: Nightingale T. R. T. , Quirt J. D.
Published: October 2005

It is widely accepted that the presence of electrical outlets in party walls can degrade the fire resistance and sound isolation. However, a systematic study of the factors affecting the sound isolation have not been reported. This paper will summarize the key factors identified in a recent study conducted at the National Research Council of Canada which considered single, staggered, and double wood stud walls. The paper will show that box placement was very important for standard metal boxes (having many penetrations), but was not an important factor when the boxes were airtight (i.e., vapor barrier type). Wall cavity absorption and how it was installed in the wall cavity at an electrical box played a key role. Cavity absorption greatly reduced the effect of poorly located metal boxes. The benefit could be maximized by placing the absorption so that any direct line-of-sight was obstructed. It was also shown that a septum, separating back-to-back boxes, was not effective unless cavity absorption was present. The paper will also present simple and effective retrofit techniques that do not involve removal or relocation of either electrical box. These include draft stoppers, inserts, caulking, etc.a) Present address: Shuller International, 10100 West Ute Ave., Littleton, CO 80127.
Analysis of the sound–structure interaction problem for a partially opened cavity structure
Authors: Oh Jae-Eung , Rhee Ding-Ik , Ahn Ji-hoon , Do Jung-Seok , Kim Tae-Bong
Published: October 2005

In order to control the noise in the cavity, a careful analysis of the vibration of the plate and sound field in the cavity are necessary. The deformation of the flat plate was approximated by the orthogonal polynomials and inserted into the vibration equation of the plate and the velocity potential equation in the cavity. As the boundary conditions, the velocity of the plate is assumed to be equal to the velocity of the adjacent particles to the plate. For the partially opened cavity structure, the above results were superimposed on the sound transmitted through the cutout. The experimental system was constructed as a rectangular cavity having a 1-mm-thick steel plate, which transmits the external sound. In the case of the flat plate having a cut-out, the theoretical study appeared to agree with the experiment. While the area of cutout is less than 1% to the whole surface area of the flat plate, no significant change was observed in the vibration characteristics of it. For the case of a small cut-out, the acoustic modes show no significant difference for the case of no cutout, in spite of the large difference of the sound-pressure level.
Sound transmission through a thin baffled plate: Comparison of a light fluid approximation with the numerical solution of the exact equations and with experimental results
Authors: Filippi Paul J. T. , Mattei Pierre-Olivier , van der Burgh Adriaan H. P. , de Jong Koen
Published: October 2005

A thin elastic rectangular plate, with clamped boundaries, extended by an infinite perfectly rigid plane baffle is considered. Both half-spaces are occupied by a gas. A sound source is located on one side of the plate. One is interested by the sound field transmitted through the plate. The boundary value problem governing the system is solved by two methods: A boundary element method, which gives a numerical solution of the exact equations; and a series representation of the solution in terms of the fluid-loaded plate resonance modes, an analytical approximation which is obtained by a perturbation technique (the ratio of the gas density to the surface mass of the plate being the small parameter). The discrepency between the two methods does exceed 1 dB or so: This shows that the perturbation method, which is much less time consuming than the BEM, is a very powerful prediction tool. The Laboratoire de Mécanique et d’Acoustique in Marseille has two anechoic rooms, which are connected by an aperture. A steel plate has been built inside the aperture. The measured acoustic pressure transfer function between points located on both sides of the plate is well predicted by the numerical methods. [This research has been partly sponsored by the Ministère de l’Environnement and by the Programme International de Coopération Scientifique Vibro-Acoustique dans le Domaine des Transports initiated by the CNRS.]
Effects of a back cavity on the acoustic properties of a stage floor
Authors: Nakanishi Shinsuke , Daido Masafumi , Sakagami Kimihiro , Morimoto Masayuki , Takahashi Daiji
Published: October 2005

To gain an insight into the acoustic properties of a stage floor, the sound reflection of a spherical sound wave from an infinite elastic plate, which is driven by a point force, is theoretically analyzed. According to a survey of a stage floor’s constructions, most stage floors have an air back cavity. Therefore, effects of an air back cavity on the reflection from stage floors should be clarified. Furthermore, the analysis is generalized to extend the theory to a back cavity filled with absorbent media. The resultant field takes the form of the sum of the field reflected by the plate without a point force excitation and the radiated field caused by a point force excitation. The total energy fluctuates around the sum of the reflected energy and the radiated energy, and the fluctuation range is theoretically determined by the reflected pressure and the radiated pressure. The cavity causes a dip which is similar to the one seen in the panel absorbers, and some periodic small dips and peaks due to the resonances of the acoustic mode in the cavity.
Passive imaging of targets with ambient noise: Experimental results
Authors: Buckingham Michael J. , Epifanio Chad L. , Readhead Mark L.
Published: October 2005

In November 1995 an experiment was performed in San Diego harbor, in which the Acoustic Daylight Ocean Noise Imaging System (ADONIS) was used to produce images of a variety of targets in the ocean. ADONIS operates in the frequency range 8 – 80 kHz, and relies on ambient noise to provide the acoustic contrast between look angles on- and off-target. No dedicated sound source is used to provide illumination. Thirty-gallon PVC oil drums filled with sand and syntactic foam were imaged at a range of 40 m in the water column, and at a range of 20 m when lying on the seafloor (silt). The primary sources of illumination were snapping shrimp (high frequency) and small-boat noise (low frequency), with the directionality and intensity of the latter varying greatly over time. The spectral, temporal, and azimuthal variations in the noise field will be discussed in relation to shadowing features in the images. Small-scale fluctuations in the medium gave rise to scintillation effects, with spatial and temporal scales that can be investigated with the ADONIS receiver. [Work supported by ONR.]
Array processing methods for imaging ambient noise sources
Authors: Olivieri Marc P. , Glegg Stewart A. L.
Published: October 2005

This paper describes array processing procedures which have been developed to investigate the ambient noise in the ocean. Most previous studies of ambient noise have been concerned with its detrimental effect on array performance. More recently however the ambient noise is being used to study oceanographic features and locate passive objects. Array processing methods which have been developed for active imaging or passive narrow-band detection are not necessarily optimal for this application. In this paper procedures for imaging using very broadband signals are described and it is shown that good resolution can be obtained using a volume array with a small number of transducers (four to six), by processing the cross-correlation matrix of the signals. Resolution is enhanced by an order of magnitude by using a deconvolution procedure combined with a Hilbert transform. The optimum processing procedure is found to be a multiplicative method which only uses the cross correlations between a reference transducer and the others in the array. Experimental results are shown which verify the predicted array performance. [Work supported by ONR.]
High-frequency backscatter measurements of coastal dynamics in Haro Strait
Authors: Pawlowicz Rich , Farmer David M. , Bellingham James G.
Published: October 2005

The Straits of Georgia and Juan de Fuca near the British Columbia and Washington state coasts form an estuarine system through which Fraser River runoff empties into the Pacific. Most mixing between the fresh surface water and deeper Pacific inflow occurs during spring tides in the narrow connecting channel of Haro Strait, leading to a 2-week periodicity in mean stratification. However, coastal and estuarine environments are characterized by complex and rapidly changing structures with scales of meters to kilometers which are difficult to observe with conventional techniques. Thus little progress has been made in understanding the details of mixing in Haro Strait. However, the rapid surveying capabilities of acoustic Doppler current profilers, volume imaging echo sounders, and surface scattering side-scan sonars enable the formation of nearly instantaneous snapshots of various stratified flow effects. These include frontal formation and decay, shear flow instability and sill hydraulics. In June/July 1996 these instruments were deployed from ships, drifters, and autonomous underwater vehicles to better understand the mixing processes in this area. [Work supported by ONR.]
Three-dimensional effects on broadband pulse propagation near the mid-Atlantic Bight
Authors: Smith Kevin B. , Chiu Ching-Sang
Published: October 2005

With the emphasis of USW shifting to littoral environments, the understanding and, ultimately, prediction of acoustic propagation in the vicinity of the shelf break become increasingly important. The sloping bathymetry, the extreme seasonal changes in the vertical sound-speed structure, and the significant horizontal variability generated by the shelf-break front are just a few of the environmental factors that make this problem both interesting and complex. An ONR-sponsored field study known as PRIMER is designed to examine the influence of both the shelf-break front and the sloping bathymetry on propagation onto the shelf in the vicinity of the mid-Atlantic Bight. Recent results using a three-dimensional (3-D) acoustic ray model suggest that 3-D effects are not observable directly upslope but could be significant along tracks oblique to the continental slope. A more thorough investigation of these influences on pulse propagation has been performed using an advanced, full-wave, 3-D parabolic equation model. Specifically, this analysis will focus on the predictability of the 3-D, broadband acoustic wave fields in a complex shallow-water environment, and the dependence of those effects on acoustic frequency. A description of the data to be examined will also be provided. [Work supported by ONR Code 321OA.]
Behavioral observations of zooplankton in the Red Sea using the three-dimensional acoustical imaging system Fish TV
Authors: McGehee Duncan E. , Jaffe Jules S. , Genin Amatzia
Published: October 2005

Fish TV is a three-dimensional acoustical imaging system that has been used successfully to measure the swimming trajectories of individual zooplankters in situ [Jaffeet al., Deep Sea Res. 42, 1495–1512 (1995); McGehee and Jaffe, ICES J. Mar. Sci.53 , 363–369 (1996)]. In order to observe zooplankton behavior at various times of day and night, Fish TV was deployed in July 1994 on an open-water mooring (water depth 280 m) in the Gulf of Aqaba, approximately 1 km from Eilat, Israel, at 27 m depth. Data were collected over several days. Analysis of these three-dimensional acoustical images has resulted in the measurement of thousands of animal (mostly euphausiid) trajectories. The trajectories reveal the behavior of these individuals as they forage during the night in the photic zone, and as they make their downward vertical migration before dawn. [Work supported by the Office of Naval Research, the U.S.–Israel Binational Science Foundation and the National Science Foundation.]
Multifrequency, multimodel inverse methods
Authors: Holliday D. V.
Published: October 2005

Many marine assemblages of planktonic organisms are dominated, at least in narrow depth intervals or small volumes, by a single species or at least by animals with similar sound scattering properties. Situations arise, however, in which there are two or more acoustically dissimilar species present in the same volume examined acoustically. As the abilities to model sound scattering from a variety of animals becomes more sophisticated, a method is needed that allows one to take advantage of differences in acoustical signatures of different genera or species. Progress toward automatic classification of zooplankton and micronekton by acoustical methods is discussed, while information on size structure in the assemblage is retained. [Work supported by ONR Oceanic Biology and NSF Biological Oceanography.]
A wideband, single-channel acoustical probe for observations of vertical fine structure in zooplankton
Authors: Greenlaw Charles F. , Player RoniSue , Stokes Robert H.
Published: October 2005

Single-frequency acoustical estimates of zooplankton biomass are notoriously inaccurate, varying 10–30 dB with variations in sizes of the scatterers. Multifrequency acoustical estimates are much more accurate but require substantial data processing. A compromise system, utilizing wideband acoustical signals (2.5–3.5 MHz) and broadband processing to reduce the sensitivity to scatterer size variations, has been developed. This system offers the advantages of rapid single-channel data rates and high spatial resolution while providing satisfactory accuracy in biomass estimation. System design required development of several novel subsystems: a broadband transducer with an acrylic lens for obtaining reasonable acoustic sample volumes from physically practical elements; a programmable broadband signal generator; and a wideband square-law receiver/detector. [Work supported by ONR Oceanic Biology and NSF Biological Oceanography.]
Acoustical beamforming and targeting using phase conjugation of ultrasonic fields
Authors: Roy Ronald A. , Hallaj Ibrahim M. , Kargl Steven G.
Published: October 2005

Phased arrays were used to locate waterborne targets, and to focus ultrasonic fields onto the targets using phase conjugation in computer simulations. Array performance was evaluated in the presence of small uncertainties added to the amplitudes and phases of the array elements. For the cw case, the free-space Green’s function was used to calculate the conjugated field in space. The propagation of a wideband pulse utilizing time reversal concepts is simulated using a finite-difference code that allows the introduction of inhomogeneous media. Again, the performance of the array with uncertainties in the phase, amplitude, and pulse-shape information is studied. The focusing capability of a high-intensity phase conjugation array operating in the nonlinear regime will suffer from the nonlinear response of the propagating medium and transducer devices. This study examines the performance penalties of uncertainties in the linear regime as a first step to implementing a digital time-reversal system with high-intensity, nonlinear arrays in the laboratory. [Work supported by the ONR.]
Real-time 3-D imaging pulse-echo acoustical imaging with coded excitation
Authors: Ebbini Emad S. , Shen Jian
Published: October 2005

A novel algorithm for high-speed parallel processing of ultrasound pulse-echo data for real-time 3-D imaging will be presented. The approach utilizes a discretized model of the received data from the region of interest (ROI) using a conventional beamformer. The transmitter array elements are fed with binary codes designed to produce distinct impulse responses from different directions in ROI. The discretization of the model leads to a matrix operator that can be inverted using a regularized pseudoinverse. Under realistic simplifying assumptions, the pseudoinverse leads to a transversal filter bank with every filter in the bank designed to extract echoes from a specific direction in ROI. The number of filters in the bank determines the number of image lines acquired simultaneously. Experimental results have shown that the new approach is valid for phased array pulse-echo imaging of speckle generating phantoms typically used in characterizing medical imaging systems. Furthermore, it was demonstrated that the new approach has lateral resolution exceeding that of conventional imaging with the same array. A full description of the algorithm along with representative image reconstructions will be given. In addition, operator design criteria for optimal design for SONAR, NDE, and medical imaging will be discussed.
An experimental test of the concept of Fresnel zones for modes
Authors: Gurbatov Sergey N. , Kurin Vasily V. , Pronchatov-Rubtsov Nikolai V. , Simdyankin Sergei I. , Virovlyansky Anatoly L.
Published: October 2005

Analysis of mode amplitude variations due to scattering by large-scaled inhomogeneities in waveguides can be performed by an approach based on using the notion of the Fresnel zones for modes (generalization of the well-known Fresnel zones for rays). This approach can be applied to describe sound scattering by an opaque screen, by inhomogeneities of refractive index, and by those of a rough waveguide boundary. In order to test several theoretical predictions made in the framework of the above approach, a tank experiment on sound diffraction by an opaque screen in a waveguide has been carried out. The waveguide was a water layer over a rubber bottom. The opaque screen crossed two thirds of a waveguide vertical section. The field was excited by a point source and registered behind the screen by a uniformly moving point receiver. The spectrum of the received signal was used to study the modal structure variation due to the screen. The spatial spectrum of the sound field and the field intensity as a function of range were measured behind the screen and compared with the theoretical predictions. The comparison showed a good agreement between theory and experiment. [Work supported by RFBR Grant No. 95-02-04565.]
An algorithm for enhancement of binaural hearing with hearing aids
Authors: Soli Sigfrid D. , Gao Shawn X. , Nilsson Michael J.
Published: October 2005

An algorithm for a wearable prototype digital hearing aid processor will be described. The algorithm enables the hearing aid wearer to better utilize their binaural directional hearing ability. Fitting of the algorithm is accomplished in two steps: Hearing aid equalization (HAE) and hearing loss compensation (HLC). HAE equalizes the amplitude and phase insertion effects of the aids and maintains the binaural cues with the aids in place. The HAE digital filters for each ear are obtained from in situ probe tube measurements of aided and unaided test signals. HLC for each ear is also achieved with digital filters and associated gain. The target response for each HLC filter is determined from auditory thresholds and sound field reference signals, and is based on an adaptation of articulation theory. The response of the HLC filter can be modified to prevent acoustic feedback. The HAE and HLC filters are combined to produce a single digital filter. A description of the processor, fitting algorithm, and the hardware and software comprising the fitting system will be given. Results from field trials with hearing-impaired individuals will also be reported.
Time varying frequency-domain Wiener filtering based on binaural cues: A method of hearing aid noise reduction
Authors: Lindemann Eric
Published: October 2005

Environmental noises such as cocktail party babble, traffic, and restaurant clatter are a continuing source of irritation to hearing aid wearers. One approach to reducing this noise is to provide the wearer with a directional field of hearing. Sounds coming from the back and sides are attenuated, while sounds coming from the front, or look direction, are unattenuated. Several approaches to designing such systems have been attempted: directional microphones, passive microphone arrays, and classical adaptive filter microphone array beamformers. These techniques are compromised by poor performance in reverberant environments, undesirable packaging constraints, and large computational requirements. This paper describes an efficient algorithm for a two-microphone directional hearing system based on time-varying frequency-domain Wiener filtering. The frequency-dependent SNR estimates necessary to determine the time-varying frequency-dependent attenuation of the Wiener filter are based on interaural time-of-arrival and interaural magnitude differences. Two alternative formulas are presented for calculating these SNR estimates. A study is made of the frequency resolution requirements needed to achieve good results using the Wiener filtering approach. Results from clinical trials showing improvements in both objective intelligibility scores and subjective SRT scores are presented.
A new technology to compensate degeneration of hearing intelligibility for elderly individuals: Development of a portable real-time speech rate conversion system
Authors: Miyasaka Eiichi , Nakamura Akira , Seiyama Nobumasa , Imai Atsushi , Takagi Tohru
Published: October 2005

This paper presents a portable real-time speech rate conversion system for compensating degeneration of hearing intelligibility of elderly listeners who are suffering from listening to rapid speech. This system enables an elderly user to convert a speech rate as desired by him/herself on real time, with invariance in pitch as well as small impairments in quality. Conventional hearing aids are focusing on compensation of decreased hearing abilities of peripheral auditory pathways in the frequency area, while the new system tries to compensate for them with central auditory pathways in the temporal area. When this system is applied to speech accompanied with a picture, temporal discrepancy between the converted voice and the picture can be absorbed by changing the speech rate at every pitch period along a monotone decreasing function from slow to fast. The system is 180×130×65 mm in size. Another feature allows a user to change the speech rate quickly whenever he/she wants to convert at any stage of conversion. This system can be applied to both the Japanese spoken language and foreign spoken languages, and can be used as a listening aid to learn foreign languages.
An audio–visual system for telephonic speech to improve hearing ability of the hearing impaired
Authors: Ikeda Takashi , Watanabe Akira , Yamazoe Kiminari , Ueda Yuichi
Published: October 2005

A hearing aid was proposed previously using single resonant analysis for telephonic speech. First, using this hearing aid, experiments by normal-hearing persons with masking noise as a simulation of a hearing threshold level were done and showed satisfactory results. Next, experiments by two hearing-impaired persons were done to examine the effectiveness of the hearing aid. The usefulness of this hearing aid for vowels and consonants was ascertained. In some consonant samples, in spite of different consonants, the subjects felt they were the same as in the case of the linearly amplified hearing aid. In the case of a hearing aid by single resonant analysis, however, the samples tend to sound different to the subjects. Since the subjects were not used to the sounds of the hearing aid, an assistant system was necessary. An audio–visual system is proposed that can make phoneme cues for the hearing-impaired person by using vision together with hearing. It will help subjects to grasp correspondence between sound image and phonemes. The audio–visual system and training by the system will be discussed.
Device characterization techniques for digital hearing aids
Authors: Edwards Brent W. , Struck Christopher J.
Published: October 2005

The development of digital hearing aids will allow for more sophisticated signal-processing techniques than have previously been implemented in commercially available analog devices. The effects of applying noise reduction or spectral enhancement, for example, must be quantified in a reasonable and consistent manner. Methods for characterizing linear aids using pure tones or stationary broadband noise are no longer sufficient for this task. Indeed, current standard techniques fail to satisfactorily describe the most sophisticated of the currently available analog devices. Techniques for characterizing nonlinear devices must be developed that capture and quantify the abilities of these new processing schemes. This is also important so that the performance differences between alternative processing schemes are readily comparable and understandable by hearing health care specialists. A set of tests is presented that characterizes the performance of DSP aids and the results are related to standard psychoacoustic measures.
A prescription for a nonlinear hearing aid using speech intelligibility and loudness models
Authors: Hou Zezhang , Struck Christopher J. , Pavlovic Chas V.
Published: October 2005

A method for prescribing a nonlinear hearing aid using a simplified loudness model and the speech intelligibility index (SII) model [ANSI draft (1993)] is described. The specific loudness of a signal is calculated as a linear function of the sound-pressure level bounded between threshold and UCL. An integrated-SII model is constructed by computing the weighted sum of the SII values across several listening conditions [Hou and Thornton, Ear Hear. 15, 105–112 (1994)]. The fundamental goal of the method is to adjust the hearing aid parameters so as to maximize the integrated SII. The specific loudness of the aided speech and noise for a hearing-impaired subject is constrained so as not to exceed the specific loudness of the unaided signals for normal-hearing listeners in each listening condition. The method is applied to a database of patients who wear multiband wide dynamic range compression hearing devices. Resulting hearing aid program parameters are compared with the patients’ existing programs.
Predicting speech intelligibility of hearing-impaired listeners from audibility
Authors: Ching Teresa Y. C. , Dillon Harvey , Byrne Denis
Published: October 2005

Speech intelligibility of mild and moderately hearing-impaired listeners can be successfully accounted for in terms of audibility, as quantified by the articulation index theory. For people with more severe losses, performance is often overestimated by audibility. It is hypothesized that deficits in frequency and temporal resolution may be responsible for the reduced ability of these listeners in using an audible signal for speech intelligibility. This study aimed to measure the amount of information that could be extracted from an audible signal by listeners with different degrees of hearing loss at different frequency regions. Twelve normally hearing and 40 hearing-impaired listeners were tested using filtered sentence material presented at different levels. The results showed that some listeners performed better than normal at low sensation levels but much poorer at high sensation levels. The conventional method of modifying the index by a multiplicative factor would not be appropriate because for many subjects, a smaller factor would be needed at high than at low sensation levels. A modification procedure which incorporates a level-dependent and frequency-dependent correction is proposed, and is verified using speech results obtained from sentences and nonsense syllables tests.
The mechanism of vocal registration in folk singing
Authors: Titze Ingo R.
Published: October 2005

The use of falsetto register in alternation with chest register is common in folk music, in particular, Hawaiian song and Alpine yodeling. There is no special effort to merge or equalize the registers; rather, each register is cultivated in its own frequency range and some attempt is made to balance the intensity. An update on the acoustics and physiology of voice registration is given. Current thinking is that the thyroarytenoid muscle in the vocal fold regulates the tension and shape of the vocal fold. Chest register is associated with a more rectangular vocal fold shape (in the vertical direction), a more sudden interruption of airflow in the glottal cycle, and a strengthening of the second and third partials. Falsetto register, on the other hand, is associated with a more convergent glottis (in the vertical direction), a more gradual interruption of airflow, and a concentration of energy in the fundamental. [Work supported by NIDCD, Grant No. P60-DC00976.]
Vocal timbre and meaning: An ethnomusicological approach
Authors: Szego C. Kati
Published: October 2005

This paper is based on results of research undertaken at the Kamehameha Schools, Hawaii’s only private school for native Hawaiians. The Kamehameha Schools are renowned for their Hawaiian-language choral singing tradition and their unique choral timbre, which is often described as ‘‘dark’’ and ‘‘sensual.’’ The first part of the paper addresses features of the Hawaiian language that may contribute to this choral sound and the symbolic significance of Kamehameha students singing in their ancestral language. The second part addresses how Kamehameha students apprehend music. They prefer vocal music to purely instrumental music, and focus much of their attention on timbre. In addition to describing vocal timbre with adjectives like ‘‘warm’’ or ‘‘harsh,’’ they interpret timbre socially, i.e., they impute personal attributes to singers on the basis of their sound quality. This phenomenon parallels other kinds of Hawaiian communicative practice, where individuals interpret expressive cues to determine others’ commitment to them [Howard, 1974]. These cues are considered reliable indicators for developing and maintaining harmonious relations—a cornerstone of the Hawaiian social ethos. Listening to vocal music is treated with all the delicacy of a social encounter: Listeners read vocal timbre with a view of assessing singers intersubjectively.
A lecture-demonstration of native Hawaiian music
Authors: Fong Randie K.
Published: October 2005

Polynesian seafarers are thought to have arrived in Hawaii some two millennia ago. Over centuries, a socially and politically complex civilization developed, which may have supported as many as a million inhabitants. The music of pre-Western Hawaii was primarily a vocal tradition that placed great emphasis on poetic verse. A variety of chant styles and settings were used for purposes of prayer, incantations, genealogical recitations, praise songs, hula (dance) as well as spontaneous expressions of emotion. A diverse inventory of implements made from various materials including stone, wood, bamboo, gourds, dog teeth, shells, and shark skin were developed and used to accompany solo and ensemble chant and dance. Generally, Hawaiian chant is categorized as either mele (strictly metered) or oli (extemporaneous, nonmetered), and typically utilizes from one to four tones with some quarter-tones, as well as several types of vocal ornamentations. From initial Western contact in the late 18th century to the present, indigenous Hawaiian music has evolved significantly from early syncretic ‘‘Hawaiianized’’ hymns, Euro-Hawaiian waltzes, marches and ballads, ragtime and swing arrangements, pop, rock, and reggae (‘‘Jawaiian,’’ or Jamaican–Hawaiian) to a contemporary neotraditional return to native sounds and styles spurred by cultural renaissance.
Simultaneous laryngeal and respiratory measures during varying sound-pressure level and F0 in singers
Authors: Stathopoulos Elaine T. , Huber Jessica , Carroll Linda M. , Ramig Lorraine O. , Johnson Antonia , Titze Ingo R.
Published: October 2005

The purpose of the present investigation was to observe simultaneous laryngeal and respiratory function during singing. Five professionally trained classical singers (three men and two women) performed singing tasks including crescendo–decrescendo at two register transitions. Laryngeal aerodynamic measures of estimated subglottal pressure, open quotient, speed quotient, maximum flow declination rate, and alternating glottal airflow were made as well as acoustic measures of sound-pressure level and fundamental frequency. Lung, rib cage, and abdominal volume measures were also made for each breath group. Aerodynamic data were collected with a Glottal Enterprises airflow mask. Oral airflow was inverse-filtered using CSpeech. Respiratory data were collected using Respitrace. Aerodynamic, acoustic, and respiratory kinematic data were used to describe the respiratory and laryngeal mechanisms used to control register transitions. Individual subject data show respiratory and laryngeal variation when comparing register transitions. A brief video will be shown to demonstrate the technology used. [Work supported by NIH.]
Acoustical studies of Japanese traditional drums (Tsuzumi)—Vibration, analysis, and synthesis
Authors: Ando Shigeo
Published: October 2005

First, properties of vibrations on the membranes of Tsuzumi (Japanese traditional drums) are reviewed. There are two kinds of Tsuzumi, Ootsuzumi and Kotsuzumi, that have features different from each other. It is mentioned that the vibrational frequencies of Ootsuzumi may approximate ratios of the roots of Bessel’s function and those of Kotsuzumi may indicate harmonic properties. Then the Ootsuzumi and Kotsuzumi tones are analyzed and their various physical parameters are extracted by means of discrete Fourier transform (DFT) techniques, and the synthesized tones using these parameters are produced and studied with regard to the features of these parameters contributing to the tone quality of Tsuzumi. Physical parameters for several playing styles of each Tsuzumi are classified in the following: (1) waveforms, (2) spectra, (3) amplitude envelopes of partials, and (4) time variances of the frequencies of partials. Finally, a subjective test for the above synthesized tones with the original ones is performed. It is verified that the synthesized tones have good similarity with the original ones using the following methods: (a) partial synthesis for Ootsuzumi tones, (b) additive synthesis for Kotsuzumi tones on Otsu and Hodo styles, and (c) additive synthesis which includes partial synthesis on the parts of the spectrum for Kotsuzumi tones on Kashira and Kan styles.
Acoustics of Caribbean steelpans
Authors: Rossing Thomas D. , Hansen Uwe J.
Published: October 2005

Although it is scarcely 50 years old, the Caribbean steel pan has already created a rich tradition in its home country of Trinidad and Tobago and throughout the Caribbean islands. It has become a versatile and popular instrument, used worldwide in musical styles ranging from symphonic to calypso. Modern steel bands include pans that span five octaves from a lead pan (typically with 28 notes) to a bass, consisting of six or more drums with three notes each. The acoustical behavior of these instruments is dominated by nonlinear mechanical behavior which leads to a rich spectrum of partials in the radiated sound.
Gourds of the Kiowa Tia Piah Society
Authors: Duncan Stephen F.
Published: October 2005

Construction techniques used in making gourd rattles for the Kiowa Tia Piah Society (a hereditary warrior society) are examined. As a member of the Memphis Tia Piah Society (a clan of gourd dancers which can trace its foundation to the Kiowa nation), this writer has been privileged to observe firsthand many of the current customs. Tribal elders, headmen, and friends have provided much information about those ancient customs from which these are derived. This work draws upon both written and oral tradition as well as research done as a participant–observer for a number of years. A brief history of the Kiowa Tia Piah Society is provided. Gourd rattles in their various forms are some of the oldest musical instruments found on the Great Plains of North America. Currently, gourds are made from traditional bottle gourds, small tin cans, large (military) salt shakers, and metal tea balls. An anthropological emic model has been followed throughout. [Work supported by LECAPSR.]
Acoustical properties of free reeds: A study of reeds from American reed organs
Authors: Koopman Philip D. , Hanzelka Chad D. , Cottingham James P.
Published: October 2005

The frequency and amplitude of vibration of a representative sample of reeds from American reed organs have been studied as functions of the (negative) static pressure difference between the interior and exterior of the windchest, along with the spectrum of the near-field sound. Over a pressure range that includes the normal playing pressure of the instrument, the frequency initially tends to decrease in an approximately linear fashion with increasing blowing pressure, but some anomalous effects are observed at higher pressures, including an increase in frequency at very high blowing pressures. Some hysteresis effects are also observed in the frequency–pressure relationships. The effects of constrictions to the airflow, simulating the effect of a partially opened pallet, have also been studied.
Scattering cross section and variances of log-amplitude and phase fluctuations of a sound wave in a moving random medium with the von Karman spectrum of medium inhomogeneities
Authors: Ostashev V. E. , Goedecke G. , Auverman H. , Noble John M.
Published: October 2005

The scattering cross section, σ, and the variances of log-amplitude and phase fluctuations, 〈χ2〉 and 〈φ2〉, of a sound wave propagating in a moving random medium (turbulent atmosphere or ocean, etc.) with the von Karman spectra of temperature and medium velocity fluctuations are calculated. The derived equation for σ differs from that obtained in the paper [Baikalova et al., J. Acoust. Soc. Am. 83, 1332–1335 (1988)]. The difference is due to the fact that in this paper the von Karman spectra of temperature and medium velocity fluctuations are assumed to have the same form, while it is known in the theory of turbulence that these spectra have different forms [J. Hinze, Turbulence (McGraw-Hill, New York, 1975)]. The derived equations for 〈χ2〉 and 〈φ2〉 are new results. It is shown that 〈χ2〉 and 〈φ2〉 depend on the wave parameter D, widely used in the theory of waves in random media. For D≪1 and D≫1, 〈χ2〉 and 〈φ2〉 coincide with the values previously calculated for the cases of sound propagation in media with the Kolmogorov and Gaussian spectra of temperature and medium velocity fluctuations. [This material is based upon work supported by the U.S. Army Research Office under Contract No. DAAH04-95-1-0593.]
Acoustic propagation over rough soil surfaces
Authors: Chambers James P. , Sabatier James M.
Published: October 2005

The propagation of sound over finite impedance rough surfaces is investigated. Laboratory experiments have illustrated that roughness can be treated as an effective impedance in propagation models. This finding holds for high- and low-impedance surfaces and is valid for propagation at grazing and nongrazing incidence. The purpose of this present work is to find out if these trends persist in the propagation of sound over rough soil surfaces outdoors. Of specific interest is the inverse problem of determining the roughness scale from the acoustic propagation. Previous field measurements were unable to separate the attenuation due to roughness and that due to the changes in the ground impedance caused by creating the roughness. Future experiments will examine surfaces that have been sufficiently weathered so that the parameters that influence impedance, such as porosity and flow resistivity, are comparable between rough surfaces and nearby smooth surfaces. [Work supported by USDA.]
Computation methods for sound speed in air
Authors: Wong George S. K.
Published: October 2005

Sound speeds in air are computed with the traditional method and compare with those obtained with a simplified method. Data for both methods are derived from the constituents of standard air. With the traditional method, the virial coefficients of the constituents of air and their temperature derivatives are utilized. For the simplified method, only the real-gas specific heats of the constituents at constant pressure are used. It can be shown that with the simplified method, the computed sound speeds for nitrogen which is the major constituent in air, are in better agreement with published nitrogen sound-speed experimental data. Uncertainties of both methods are discussed.
Speed of sound in real gases. II. Comparison with experiment
Authors: Zuckerwar Allan J.
Published: October 2005

The data needed to evaluate the specific heat, virial, and relaxation corrections for the sound speed have been compiled from the literature for over 40 pure gases. Theoretical and published experimental sound speeds are compared in several case studies. (1) In precision measurements in a spherical resonator (argon, nitrogen), agreement between theory and experiment falls within 0.04% and 0.02%, respectively, over the entire ranges of temperature and pressure. (2) In high-pressure measurements (argon, helium, methane), agreement remains within 1% or better up to a certain pressure, typically 20 to 50 times the critical pressure, beyond which the increasing disagreement is attributed to truncation of the virial expansion. (3) In a measurement with large dispersion due to molecular relaxation (sulfur hexafluoride), agreement is well within 0.5%. The specific heat, virial, and relaxation corrections are analyzed in detail.