bluuebunny/crossref_metadata_2025 huggingface dataset:
Descriptive characterization of kinematic effects of changes in speaking rate
Authors: Westbury John R. , Moy Gloria
Published: October 2005

Speakers change how they move their articulators when they change the rate at which they speak. Some speakers move different distances, and/or at different rates. Others change how multiple movements are coordinated in time. Still others appear to change the syllabic structure of utterances, and thus, the number and kinds of necessary movements. Up to now, most of what has been learned about speaking rate and movement has been derived from small speaker samples, and limited comparisons involving ‘‘magic’’ measures of short intervals from unnatural speech tasks. This new investigation seeks to describe and relate patterns of sagittal-plane movements of tongue, jaw, and lip fleshpoints accompanying an eight-word sentence, read 4–6 times, at three rates (nominally, double-speed, habitual, and half-speed), by 22 normal young adult speakers of American English. Summary statistics have been computed within and across speakers and fleshpoints, for distributions of fleshpoint positions and speeds, position-history spectra, and durations of selected subintervals. These data provide an improved estimate of the nature and variability of rate-modifying strategies, and useful working ideas about control objectives in speech, and the role of the articulator plant in determining the temporal and spatial character of speech. [Work supported by NIH Grant No. DC00820.]
High-definition speech synthesis
Authors: Campbell Nick
Published: October 2005

This paper describes a method for producing high-quality speech synthesis, without signal processing, using indexing and resequencing of phone-sized segments from a prerecorded speech corpus for the purpose of reproducing the voice characteristics and speaking style of the original speaker to create novel utterances. It describes procedures for indexing and retrieval using pointers into an external speech corpus that enable the synthesizer to be both language- and speaker independent. The prosody-based synthesis unit selection process does not itself produce speech sounds, but yields an index for a ‘‘random-access’’ retrieval sequence into the original speech to produce the closest approximation to a desired utterance from the segments available in a given speech corpus. To find the optimal sequence of segments for concatenation, the synthesizer first creates an inventory of phones and their acoustic and prosodic characteristics, and then selects from among these by a weighted combination of features to give an index of the segment sequences that best match the target prosody while minimizing intersegment acoustic discontinuities. It has, to date, been tested on English, Japanese, Korean, and German using more than 20 speech corpora.
Modeling the perception of prosody: P
centers
Authors: McAngus Todd Neil P. , El-Leighy S. , Bull J.
Published: October 2005

Previously, a multiscale mechanism [N. Todd and G. Brown, J. Acoust. Soc. Am. 94(A) (1994)] has been shown to be effective in locating significant events in a speech signal and in estimating their relative stress. Two experiments were carried out in order to determine whether major stress points identified by the model correspond to so-called ‘‘p centers’’ [Morton et al., Psychol. Rev. 83(5), 405–408 (1976)]. Recordings were made of (1) two subjects reciting the digits ‘‘one’’ to ‘‘eight’’ and (2) a number of subjects reciting metrical verse in synchrony with a metronome. The speech was recorded on one channel, the metronome on the other. The speech was then extracted without the metronome and fed into the model and the predicted syllabic stress points recovered and compared with the metronome. In the case of the digits the mean error of prediction was −9 ms (N=48). Given that the model has an internal delay of about 50 ms (20 Hz), this is good agreement. The variance was quite high, 1609 ms, but actually within the range of variation typically reported for such a task, e.g., G. Allen [Lang. Speech 15, 72–100 (1972)].
Broadband matched-field processing in shallow-water environments overlying elastic sediments
Authors: Orris Gregory J. , Perkins John S.
Published: October 2005

It has been demonstrated that the additional complexities associated with range and azimuthally dependent environments can be used to break the symmetries of the ambiguity functions of matched-field processors. Additional symmetry breaking can also be accomplished though the inclusion of more information in the form of better propagation models, increased environmental parametrization or broadband processors. In many shallow-water environments, a downward-refracting sound-speed profile and variable sedimentation suggest that acoustic propagation (and consequently matched-field processing) in these areas should be dominated by the acoustic characteristics of the bottom. This suggests that the propagation models used to generate replica fields should not only be range-dependent, but should also include the physics of acoustic propagation in the presence of elastic material (i.e., the ocean bottom). A broadband matched-field processor which utilizes the elastic parabolic equation as the replica generator has been implemented. The effectiveness of this new processor on real experimental data will be discussed.
Sector-focusing enhancement of shallow-water matched field localization data from Panama City, FL
Authors: Chandler H. A. , Feuillade C. , Smith G. B.
Published: October 2005

In October 1985, a shallow-water localization experiment was performed 19 miles south of Panama City, FL [J. Acoust. Soc. Am. 88, 423–433 (1990)]. The experiment involved a 450-Hz source placed 2.2 km from a vertical array of hydrophones in 33 m of water. Passive localization of the source using MVDR processing was successful after extensive ‘‘fine tuning’’ of the environmental parameters. In recent years the high-resolution sector-focusing (SF) processor has been developed for robust estimation in a mismatched environment, and has shown promising results when applied to simulated data. This work reports the application of the SF method to the Panama City data. Used for localization, greater tolerance to mismatch is evident. Furthermore, sector focusing may be applied as a parameter subspace technique for attaining high-resolution inversion of environmental parameters.
Source localization in a range-dependent environment
Authors: Miklovic D. W. , Abawi Ahmad T.
Published: October 2005

Source localization using matched-field processing in a range-depependent environment is evaluated with experimental data taken in the Gulf of Mexico. The bathymetry in this experiment is a moderately sloping flat bottom, resulting in mode coupling. The conventional adiabatic normal-mode method for the solution of the wave equation is not expected to be accurate here. A new range-dependent coupled-mode method for the evaluation of the replica fields is used. This is a one-way method in which the energy is propagated from one point in range to the next by a first-order differential equation. This approach allows for the relatively efficient calculation of replicas for many source positions. The ability to localize known sources at various ranges and frequencies with a fixed vertical array is quantified. Results from both the conventional adiabatic normal-mode method and the new range-dependent coupled-mode method are compared.
Fluctuation based processing to enhance the detection of low-fluctuation amplitude signals in a background of high-clutter and noise
Authors: Wagstaff Ronald A.
Published: October 2005

A new nonlinear method of processing digital time series data is presented. The method is sensitive to fluctuations in the amplitude of the signals, attenuating them more severely as the amplitudes of the fluctuations increase. In the case of a background noise that has high amplitude fluctuations and a signal that has small amplitude fluctuations, the new method results in significant enhancements in signal-to-noise ratio (SNR) from 5–15 dB and other highly desirable characteristics including increased spatial and spectral resolution, unalerted detection and identification of stable tonals from submerged sources, and reduction in tonal clutter. Results are presented for two fluctuation sensitive processors designated WISPR and AWSUM. It is concluded that through exploiting amplitude fluctuations, these new processors access a new source of gain and are thus able to achieve higher levels of performance than had previously been imagined. Furthermore, they do it with very little additional processing overhead. [Work supported by ONR and NRL.]
Computing the environment of an artificial neuronal network for underwater signal classification
Authors: Zhang Jinwen
Published: October 2005

An environment for ANN signal processing has been designed and developed, which is written with Turbo Vision’s CLASS in Borland C++ 3. Algorithms for mapping a BP network on a machine and task assign tactics are devised to implement ANN parallel algorithms. To use the PC as controller of the whole parallel system, some problems have to be solved. So the floating-point conversion between TMS320C30 and IEEE formats, executable program transforms, as well as data-loading techniques, are developed and presented in this paper. This application system has been used to quickly classify three kinds of underwater signals with a three-layer BP network in the ANN. Results show that the above classification can be finished in 28 μs. The FFT algorithm with 1024 points is also simulated on the system and it takes 2.3 ms. The other advantage of this software is that it is easily expanded to include new functions.
A novel approach for identifying polyphonic piano signals
Authors: Rossi L. , Girolami G.
Published: October 2005

The complexity of polyphonic sound signals and octave errors are the two main difficulties encountered in the identification of piano sounds [Chafe and Jaffe, Proc. ICASSP, Tokyo, 1289–1292 (1986)] [E. R. S. Pearson, Ph.D. thesis, University of Warwick (1991)]. Inharmonicity, which can be important in the case of the piano, is also often an obstacle to the identification of the fundamental frequency. The aim of this paper is to present a spectral identification method which takes advantage of inharmonicity to distinguish notes and avoid octave errors. Two independent steps are required to identify piano notes with this method. First, the distribution of the partials of each note must be determined, and a database containing this information must be, once and for all, created. Then, the notes are identified by comparing the incoming spectrum with the database. Frequential fluctuations on the positions of the partials are allowed during the identification procedure. This method identifies the notes of octaves C1 and B5 and treats monophonic as well as polyphonic signals (two, three, or even four notes played with a comparable hammer velocity).
Non-native speech perception as a window on the native phonological system and its development
Authors: Best Catherine T.
Published: October 2005

The perceptual assimilation model (PAM) [Best etal ., JEP: HPP 14, 345–360 (1988)] posits that perception of non-native contrasts is constrained by experience with both the phonological functions and the phonetic details of familiar native contrasts. Adults’ discrimination of an unfamiliar contrast depends not only on whether its members resemble a known phonological contrast, but also on perceived ‘‘goodness of fit’’ to native phonetic categories. Adult findings support PAM’s prediction that perceptual assimilation of contrasting non-native consonants or vowels to a single native category yields poorer discrimination than assimilation to two categories (TC contrasts). Discrimination is worst if both phones are equally similar to a single category (SC), substantially better if they differ in category goodness (CG), near ceiling for TC assimilations. PAM predictions have focused on adults’ first encounters with unfamiliar contrasts. However, non-native speech perception also provides a window on infants’ emerging knowledge about phonetic details and phonological organization in native speech. Native phonetic experience influences discrimination of unfamiliar non-native contrasts by 10 months, but systematic phonological knowledge is not evident until later. PAM can also be extended to phonetic and phonological aspects of second language (L2) learning developmentally. [Work supported by NICHHD and NIDCD.]
Interlanguage differences and L2-related change in audio–visual speech processing
Authors: Sekiyama Kaoru
Published: October 2005

The ‘‘McGurk effect’’ demonstrates that visual lip-read information is incorporated into speech perception even when it is conflicting with auditory information. While the McGurk effect has been shown to be robust in English speaking cultures, interlanguage differences between native speakers of Japanese and American English have been found. Native speakers of Japanese tend to show a weaker McGurk effect than native speakers of American English when listening to clear speech of their native language, although these Japanese subjects show a highly increased McGurk effect for stimuli with auditory noise or for stimuli of their non-native language. This auditory reliance of the Japanese may be due to their cultural habit of avoiding looking at a speaker’s face. The Chinese who are believed to be similar with respect to the face avoidance also showed a weaker McGurk effect than American subjects. More intriguing about the Chinese subjects was that the subjects who had stayed in Japan longer tended to show a stronger visual effect (r=0.723). These results imply that people who are seriously learning a second language also learn to use any cues including lip-read information to improve their listening comprehension.
Propagation characteristics of a leaky surface acoustic wave on proton-exchanged 36 ° Y–X LiTaO3
Authors: Kakio Shoji , Nakagawa Yasuhiko
Published: October 2005

The propagation characteristics of a leaky surface acoustic wave on proton-exchanged 36 ° Y−X LiTaO3 were experimentally investigated to evaluate the propagation loss. To measure the insertion loss, a pair of interdigital transducers was fabricated on a 36 ° Y−X LiTaO3 substrate with the proton-exchanged layer, which was formed by immersion in phosphoric acid at 240 °C. The period of a pair of fingers (λ) was 20 μm. Each transducer was composed of eight finger pairs with an overlap length of 100λ. The distance of the free surface between two transducers was 250λ. The measured insertion loss at the center frequency decreased from 24.4 dB for the unexchanged sample to 16.8 dB for the 4-h proton-exchanged one, though the reflection coefficient of the transducer increased from 0.27 to 0.36. Therefore, the above result means that the propagation signal is improved at least 0.03 dB/λ by the proton-exchange process. This phenomenon is probably due to the field confinement in the vicinity of the surface, which is caused by the formation of the proton-exchanged layer with a slow phase velocity as compared with an unexchanged substrate.
Methods of calculating coupling coefficients for electrostrictive materials
Authors: Robinson Harold C.
Published: October 2005

The definition of the coupling constant for piezoelectric ceramics is straightforward and depends directly on the square of the piezoelectric constant and inversely on the compliance and dielectric constants appropriate for the boundary conditions. However, these definitions are not directly applicable to electrostrictive materials because the material is nonlinear and therefore the equivalent piezoelectric and dielectric constants are not universally defined. Also, the presence of bias voltages about which the material would operate complicates the situation dramatically. In this presentation, energy methods [Hom et al., IEEE Trans. Sonics Ultrason. and Freq. Cont. UFFC-41, 541–545 (1994)] are used to calculate k33 for electrostrictive materials using several different constitutive models for the material. Experimental data on PMN taken at NUWC will be used to characterize the materials. The behavior of the electrostrictive coupling constants in the limit of small drive amplitudes about the bias voltage will be discussed. The effects of prestress, bias voltage, and drive level on the coupling coefficient will be examined for each model. The results will be compared with the equivalent piezoelectric definitions for k33.
Fields of sound emitted from circular transducers
Authors: Ohtsuki Shigeo
Published: October 2005

Sound fields of circular transducers are basic in acoustics. The ring function method for the calculation of an emitted sound field by a plane or spherical transducer was introduced in 1974. Now, computer technology is developing rapidly. An event-driven program is available in personal computer systems such as ‘‘Windows 95.’’ Then, a personal computational program for sound fields produced by circular transducers is developed. Computed images of sound fields are useful for understanding the features of emission of sound.
Theoretical model to describe the coupling between a vibrating membrane and thin gas layers
Authors: Lotton Pierrick , Bruneau Michel
Published: October 2005

A theoretical model to describe the behavior of a membrane loaded by gas films has recently been developed. This analytical model is required by the current level of miniaturization of acoustical transducers and by new devices and applications, especially in the high-frequency range (up to 100 kHz and more). It is based on the fundamental acoustical and mechanical equations (the Navier–Stokes equation, the conservation of mass equation, the Fourier equation for heat conduction, the equation of motion of the membrane) and includes realistic boundary conditions. These equations are simplified by making assumptions based on the fact that the thicknesses of fluid films are very small. Then, conventional models, based on equivalent electrical networks, generally used to describe the behavior of electrostatic or piezoelectric microphones or loudspeakers, are improved. For example, the influence of the viscosity, the heat conduction, and the deformation of the membrane in the high-frequency range can be accurately taken into account.
Ultrasonic sound-pressure measurement from microparticle locus observation in a periodically perturbed ultrasonic standing wave
Authors: Yamakoshi Yoshiki
Published: October 2005

An ultrasonic wave sound-pressure meassurement which uses a reference ultrasonic wave and a microparticle suspended liquid cell is proposed. In the method, microparticles are held in the liquid cell which is filled with the saline solution whose specific gravity is the same as the microparticles. If the cell is placed inside the standing wave which is produced by the ultrasonic wave to be measured and the reference ultrasonic wave, the microparticles are trapped in the standing wave and they are arranged at equal spaces. Then, the phase of the reference ultrasonic wave changes periodically; the microparticles reciprocate within the adequate range because the time-varying standing wave applies alternating acoustic radiation force to the microparticles. Since the displacement amplitude of the microparticle motion is proportional to the acoustic power, the sound pressure is estimated by measuring the displacement amplitude. In this method, the displacement amplitude is measured by observing the microparticles’ loci when the cell is in the ultrasonic standing wave with adequate phase shift. Basic experiments are carried out by using polystyrene particles with a mean diameter of 40 μm. Results almost agree with those measured by the hydrophone.
Development of an omnidirectional, low-frequency, high-power, underwater sound projector standard
Authors: Ivey Larry E. , Hugus George D.
Published: October 2005

This paper presents the design and development of a truly omnidirectional, low-frequency, high-power, underwater sound projector standard. The transducer is operational within the frequency range from 1 Hz to 10 kHz, with a transmitting response which is stable with hydrostatic pressures up to 1524 kPa (152-m depth) and temperatures from 5 to 35°C, and can be used as a hydrophone or a projector. The sensitive element is a 22.9-cm (9.0 in.) o.d. piezoelectric lead zirconate–lead titanate (MIL-STD 1376, Type 1) hollow sphere encapsulated in polyurethane. This transducer, designated the USRD type F80, produces an undistorted omnidirectional source level that is linear with driving voltages up to 1000 V rms, and can also be electrically tuned for a higher response at predetermined, discrete frequencies. This paper includes an electroacoustic equivalent circuit model of the transducer, compares the predicted to the measured transmitting response, and discusses the mechanical and electrical design.
The DIET WISPR and DIET AWSUM high-resolution and high-gain deconvolution techniques
Authors: Gardner Susan D.
Published: October 2005

Two closely related deconvolution techniques, called DIET WISPR (directivity improved estimation technique combined with Wagstaff’s integration silencing processor) and DIET AWSUM (directivity improved estimation technique combined with the advanced wispr SUM) are designed to achieve higher resolution results and signal-to-noise ratio (SNR) gain for stable amplitude signals embedded in fluctuating signals and noise. These objectives are achieved by combining a deconvolution technique (producing higher resolution and associated gain) with the WISPR and AWSUM filters (fluctuation sensitive signal processing algorithms with high SNR gain capabilities). The DIET WISPR and DIET AWSUM techniques and autoregressive parametric modeling techniques with parameters generated by the Burg algorithm, commonly known as the maximum entropy method (MEM), are applied to measured ocean acoustic data. MEM is used to provide a reference for the DIET WISPR and DIET AWSUM estimates. The MEM, DIET WISPR, and DIET AWSUM methods all demonstrate an ability to increase the resolution. However, the DIET WISPR and DIET AWSUM techniques provide a robust method for generating both SNR gain for stable signals and higher resolution simultaneously. [Work supported by ONR and NRL.]
Noise analysis of the optical fiber hydrophone array
Authors: Arai Hiroshi , Sato Ryotaku , Dobashi Koujl , Kohama Yoshikazu
Published: October 2005

In this paper, noise analysis of the optical fiber sensor array system based on the phase generated carrier (PGC) technique with time division multiplexing (TDM) will be discussed. In this system, the laser beam with sinusoidal frequency modulation from a laser diode is gated by a LN(LiNbO3) intensity modulator, amplified by a booster erbium-doped fiber amplifier (EDFA) sent into array. The returned pulses from the sensor array are amplified by pre-EDFA, converted to electric pulses, and demultiplexed into each channel by the analog switch followed by the interpolation filter, and finally demodulated. Results of both noise analysis and experimental data will also be discussed.
Sound insulation at low frequencies
Authors: Kihlman Tor , Pietrzyk Andrzej
Published: October 2005

The sound insulation of building partitions at low frequencies depends on the particular configuration of rooms that are separated by a given partition, not only on the properties of the partition itself. When the modes of the rooms match (e.g., when rooms are identical), the sound insulation can be reduced substantially. If the partition responds in a resonant way, the reduction can also be due to the matching of modes of the partition with those of the rooms. In the current paper the phenomena involved are studied by means of the numerical simulation. Both the modal approach and the finite/boundary element method are used for the calculation of the difference of the averaged sound-pressure level in the 3-D case, yielding both the narrow-band results and band-averaged ones. The modal approach offers faster calculations, but is less flexible regarding rooms’ geometry and partition model. The study of the changes of sound insulation due to slight changes to the original dimensions has been carried out for a great variety of combinations of original rooms’ dimensions, and for several wall types (single leaf nonresonant, single leaf resonant, double leaf).
Sound absorption performance of recycled porous aluminum sheet
Authors: Oh Jae-Eung , Rhee Ding-Ik , Ahn Ji-hoon , Do Jung-Seok , Kim Tae-Bong
Published: October 2005

The goal of this work was to evaluate the performance of the sound absorption for the recycled porous aluminum sheet instead of the traditional glass wool panel. A rectangular cavity had a wall to provide sound-pressure excitation, and its facing wall was covered with a porous aluminum sheet. The noise reduction effect was observed by the comparison of the sound between the cases with and without the cover. The impedance in the perpendicular direction to the surface in the experiment was employed as a boundary condition of the wave equation for numerical computation. The frequency characteristic and noise reduction characteristic for the two cases above was measured empirically. Both the calculation and the experiment showed the porous aluminum absorption sheet provided considerable noise reduction. Thus this material has many merits: It can be employed in a high-temperature environment, such as an automobile engine room, producing no pollution by dust as the case of glass wool; it makes it convenient to cover any surface due to its stiffness; and it is cheaper because it can be recycled. All of these applications ensure porous aluminum a promising future.
Acoustic properties of permeable double-leaf membranes
Authors: Sakagami Kimihiro , Kiyama Masakazu , Morimoto Masayuki
Published: October 2005

The acoustical properties of double-leaf membranes, which are usually used as roofs in actual membrane-structure buildings, were theoretically analyzed. The double-leaf membranes were modeled, in the analysis, as of infinite extent. One of these leaves, which faces an incident sound, was assumed to have permeability, which is often the case in actual buildings. The permeability makes the acoustical performance of the double-leaf membranes totally different from that of impermeable double-leaf membranes in some cases. The effect on the absorptivity mainly appears at high frequencies: The optimal permeability can significantly increase the absorptivity at high frequencies, and makes the characteristics similar to those of cavity-backed porous absorbents. At low frequencies the effect also appears in a difference of absorption and transmission coefficients, which are consistently used in this paper to measure the absorptivity as the true energy loss caused in the structure. The transition of those two conditions, where a mass–spring peak appears or vanishes depending on the permeability, is at middle frequencies. These tendencies are drastically changed with changes in parameters such as mass of leaves, flow resistivity, cavity depth, etc. A detailed analysis was also carried out to interpret the mechanism and behavior of these effects.
Tomographic imaging of the ocean volume
Authors: Spindel Robert C. , Worcester Peter F.
Published: October 2005

Ocean acoustic tomography has become a highly developed tool for observing the ocean’s interior. It has been used in the Atlantic, Pacific, and Indian Oceans, the Mediterranean, Greenland, and Norwegian Seas, and in a number of smaller bodies of water including Puget Sound. There are active ocean tomography research programs in the U.S., France, Germany, Japan, and India. In the same sense that x-ray, or ultrasonic acoustic tomography is used to image parts of the human body, ocean tomography provides images of the ocean’s interior. This presentation will describe the methodology and present examples of tomographically derived ocean images.
Imaging of ocean noise
Authors: Coulson Robert K. , Glegg Stewart A. L.
Published: October 2005

A passive underwater imaging system has been developed using a volume array of only 6 elements and time average product (TAP) broadband array processing techniques to create images consisting of thousands of pixels. This system was developed to image objects using ambient noise as illumination. Sea trials have been conducted where targets were passively detected at ranges up to 40 m. The system has also been used to image the global directivity of ambient noise, identifying the dominant noise sources, and to visually track the bearing and range of boat traffic and submerged divers. This system is capable of generating a 360° image of the ambient noise field approximately every second (with 1.6° resolution at 25-kHz bandwidth). The small number of transducers and relatively low data volume produced in the image processing make this system ideal for long-term monitoring of ocean noise. An autonomous version has been designed for offshore deployment. Initially it will consist of an autonomous stationary platform but will subsequently be incorporated into an autonomous underwater vehicle (AUV) for mobile noise surveillance missions. [Work supported by ONR.]
Volumetric visualization of zooplankton sound scattering in the ocean
Authors: Greene Charles H. , Wiebe Peter H.
Published: October 2005

Acoustic visualization combines the techniques of acoustic remote sensing and scientific data visualization. It provides scientists with the means for interactively exploring three-dimensional data sets collected with conventional acoustic survey methods. Here, investigations are described in which acoustic visualization has provided new insights into the volumetric distribution of zooplankton sound scattering in the ocean. The presentation will document the steps involved with survey design as well as data analysis and interpretation. It will focus on attempts to derive quantitative, objective inferences from sparse, irregularly spaced data collected during acoustic surveys of patchily distributed zooplankton. [Work supported by NOAA, NSF, and ONR.]
High-resolution maximum-likelihood estimator for ambient noise imaging
Authors: Potter John R. , Ong Sim H. , Ye Qin Z. , Wu Fang N.
Published: October 2005

The raw image data from the Acoustic Daylight Ocean Noise Imaging System (ADONIS) consist of only 126 pixels, arranged in an elliptical shape of major and minor axes 14×11 pixels. An estimator is presented which produces higher spatial resolution from these raw images using a maximum-likelihood estimator (MLE) coupled with a point-spread function model of the imaging system. The technique converges iteratively on a constrained MLE of the intensity distribution in each image frame. The MLE performs better than traditional interpolation methods. The estimator is also extended to constructing improved images from a sequence of image frames, from which the relative values and convergence rates of different frequency components becomes apparent. Examples of the technique are presented for both simulated and experimental data taken from a deployment of the ADONIS system. [Work funded by NUS and DSO, Singapore. ADONIS data provided by Scripps Institution of Oceanography.]
Statistical models for ambient noise imaging in temperate and tropical waters
Authors: Potter John R. , Chitre Mandar
Published: October 2005

The data obtained from the Acoustic Daylight Ocean Noise Imaging System (ADONIS) in San Diego, California, is dominated by ambient noise from snapping shrimp. This is expected in all warm and shallow waters, except with very high winds or precipitation. It is shown that these data exhibit a log-normal intensity distribution. This exactly determines the number of independent statistical parameters that can be used to create images. Further, ambient noise data taken near Singapore are also shown to be log-normal in intensity. This distribution is common for unsaturated, multiply-scattered acoustic fields, but it is shown that it is a property of the sources and not the propagation. It is also demonstrated that good ambient noise images can be created from cross correlations of pixel intensities, independently of mean or other statistical measures. Image examples will be presented and some implications for the next-generation ambient noise imaging system will be discussed. [This work was funded by NUS and DSO, Singapore. ADONIS data provided by Scripps Institution of Oceanography.]
Relationship between acoustic backscattering field and suspended matter distribution in the Kara Sea and Ob and Yenisei estuaries
Authors: Vinodradov Michail E. , Shevchenko Vladimir P. , Muyakshin Sergey L. , Nuchlov Vladimir N. , Lisitzin Alexander P. , Lebedeva Ludmila P.
Published: October 2005

During the 49th cruise of the RV DMITRY MENDELEEV, remote ADCP measurements of ultrasonic backscattering coefficients (BC) at 215 kHz were made together with biological sampling and sedimentological station deployment. It was found that the structure of the BC field on the Ob–Yenisei Delta is closely connected with the processes of fast sedimentation in the region of fresh-water influence. The BC field in the west basin of the Kara Sea is formed on the background of weakly sound scattering Arctic water under the influence of the intrinsic mezoplankton distribution, cost erosion, tidal mixing processes, and refreshed water transport from the Ob–Yenisei Delta. The present study demonstrates that ADCP is a very powerful tool for investigation of suspended matter distribution and transport in the marginal seas with river plumes.
A digital hearing aid based on a composite filtering compression and its assessment
Authors: Ueda Yuichi , Aoyama Masazumi , Watanabe Akira
Published: October 2005

In this paper, a hearing aid process is described, based on a composite filtering compressor, and its implementation, which is based on a digital hearing aid system. In the compressor, a relatively smooth compression curve similar to the FFT method is calculated using outputs of a conventional filter bank. Consequently, it is expected that the qualitative degradations in compressed speech are reduced and, as a result of computational reductions, the compression is suitable for DSP operations. Moreover, it is possible to realize an adaptive compression where the flame lengths for spectral estimation may be varied according to speech features. In order to implement the compression, a digital hearing aid system has been designed with some external adjustments, and the syllable articulation tests and the quality assessments have been carried out. As a result, it was confirmed that the appropriate compressions could be realized by the subjects’ adjustments of the parameters.
Array signal processing applicable to hearing aids
Authors: Asano Futoshi , Hayamizu Satoru , Suzuki Yôiti , Tsukui Shinji , Sone Toshio
Published: October 2005

For the use of hearing aids in noisy environments, noise reduction or speech enhancement techniques are being investigated by many researchers. In this report, array signal-processing techniques and systems applicable to hearing aids, which are being developed by the authors, are introduced. A delay-and-sum beamformer, in which the microphone-array outputs are summed with appropriate delays and weights, can focus on a signal coming from a certain direction and can reduce environmental noise. Moreover, by scanning the beam by controlling the delays and the weights, the directions of the sound sources can also be known. By combining these techniques, a noise reduction system, which tracks the movement of a talker and suppresses the environmental noise, can be constructed. In this report, hardware equipped with two DSPs (TMS320C40), which realizes the above method, is introduced. Moreover, by extending the beamformer to stereo outputs, a listener can perceive the direction from which the sound is coming. In this report, simulation results of the stereo beamformer are also briefly described.
DSP algorithms for hearing aids
Authors: Greenberg Julie E. , Zurek Patrick M. , Rabinowitz William M.
Published: October 2005

Advances in semiconductor technology are making digital signal-processing (DSP) hearing aids a viable alternative to conventional analog devices. Some of the most promising DSP algorithms for hearing aids are found in the areas of (1) microphone-array processing for interference reduction, (2) feedback reduction for increased gain margin and stability, and (3) automatic gain control for improved audibility and comfort. Typical common elements of such algorithms are a tapped-delay line, which is used to store a sequence of input values sampled over time, and a time-varying response, which is computed based on the input values. For example, adaptive filters are required by feedback reduction techniques that estimate the feedback path, as well as by adaptive microphone arrays, which attenuate spatially distinct interference sources. Additional benefits in DSP hearing aids are expected from their ability to use sophisticated fitting methods, particularly when the fitting procedure includes selection of algorithm-specific parameter values as well as the frequency gain characteristic. Finally, DSP hardware platforms designed for the hearing aid application provide a powerful research tool, facilitating field trials designed to evaluate algorithms and determine optimal parameter settings. In particular, such devices allow side-by-side comparison of multiple algorithms, including both DSP algorithms and those that could ultimately be implemented using analog components. [Work supported by NIH.]
Evaluation of the perception of input-referred noise in hearing aids
Authors: Lee Lidia W. , Geddes Earl R.
Published: October 2005

Hearing-aid circuitry noise has been a major complaint among hearing-aid users, especially among those who have normal hearing at low-frequencies. The purpose of this study is to examine the spectral distribution of the noise-tolerance range for hearing-impaired listeners. The primary purpose of this study is to determine the upper and lower fences of the permissible equivalent input noise (in dB SPL) of a hearing aid. First the audiometric threshold is evaluated, then the just-audible noise level (JAL) and the just-objectionable noise levels (JOL), measured in dB SPL at the eardrum, are determined in quiet using normal hearing and hearing impaired listeners. Given the audiometric threshold, various prescriptions can then be applied to obtain the target gain values. The JOL minus the prescribed gain values (Rx) and the JAL minus Rx represent the maximum input noise level an individual will tolerate and the just-noticeable input noise level, respectively. These data will be useful as specifications of input referred noise for hearing-aid components. The mean and standard deviation of input referred JOL and JAL, in one-third octave bands, for normal-hearing and hearing-impaired listeners will be reported. [Work supported by Knowles Electronics, Inc.]
A wearable digital master hearing aid with transient reduction software
Authors: Gingsjö Anders L.
Published: October 2005

A low-cost, wearable digital master hearing aid has been designed for research purposes. It is comprised of a 20 million instructions per second (MIPS) digital signal processor (DSP) and a two-channel analog interface allowing a 75-dB signal-to-noise ratio and a 17-kHz bandwidth. The digital master hearing aid is powered by rechargeable batteries, giving an operating time between 12 and 20 h. It is housed in a 125×70×29-mm box equipped with three controls allowing the user to change up to ten parameters or switch between ten different programs. Also, a software implementation of an algorithm for detection and reduction of transient noise is presented.
Analysis of sound generation mechanisms of a Samisen: Comparison between its sound waveform and vibration waveforms of the skin, bridge, and plectrum at the beginning of the sound
Authors: Horiuchi Ryuzo , Kikuchi Tsuneo , Sato Sojun , Miura Hajime
Published: October 2005

A Samisen is one of the typical plucked string instruments in traditional Japanese music. In this work, the sound generation mechanism of the Samisen will be discussed by analyzing how each part of the Samisen works. The Samisen sound is classified into four time regions by considering the characteristics of the waveform. The time–frequency analysis of the vibration waveform of each part in each time region is compared with that of the sound. Furthermore, the sound generation mechanism will be discussed, including the time region by the transient touch of the plectrum to the strings and the skin. It is concluded that both the cause of the vibration of the skin and the generation of the Samisen original sound are mainly due to two mechanisms: the transmission of the continuous vibration of strings through the bridge and the transient and direct touch of the plectrum.
The ‘‘Satsuma’’ biwa
Authors: Kishi Kenshi , Tohnai Yoshimasa , Yamada Masashi
Published: October 2005

The modern biwas used in Japanese music can be classified into two groups, ‘‘Satsuma’’ biwa and ‘‘Chikuzen’’ biwa. In this paper, the structural difference between Chinese biwas (PiPa) and Satsuma biwas [especially the fret height and ‘‘torikuchi’’ (nut), related to ‘‘sawari’’], which have the same origin, are discussed by one of the authors, Y. Tohnai. Tohnai is a skilled player and a researcher of modern biwas. Three Satsuma biwas, with approximately the same size and shape but from different generations, were selected from Tohnai’s collection. The open strings of all three were played three times by Tohnai in an anechoic chamber; the tones were recorded on DAT and analyzed using a sound spectrograph and FFT analyzer. The most prominent result, common to all the tones, is that the sound-pressure level of the fundamental tone and lower overtones is very low compared with 6–12th overtones (from −40 to −30 dB). Psychological experiments must be carried out in order to study its influence on the sensation of ‘‘pitch.’’
The American hammered dulcimer: Its acoustical properties, role in traditional culture, and current design developments
Authors: Peterson David R.
Published: October 2005

The modern hammered dulcimer is a trapezoidal-shaped instrument with 24 to 30 courses of treble and bass strings with a partially chromatic range from D3 to D6, which is played with light wooden hammers. The treble strings are divided by a middle bridge in a ratio of 2:3 so that each course provides two notes a musical fifth apart. Although acoustically similar to the piano, the distinctly different bridge/soundboard placement and hammer/string interactions give the hammered dulcimer a unique percussive sound. Historically in the United States, the hammered dulcimer has remained almost exclusively a folk instrument, most often used as a lead or rhythm instrument in traditional dance bands, but more recently as a concert instrument as well. Instrument builders, almost exclusively folk craft people, and players themselves, have responded with new designs that have changed the sound characteristics and musical and physical flexibility of the instrument.
The reed organ in North America and Japan
Authors: Cottingham James P.
Published: October 2005

It is believed that free reed instruments originated in Asia and were subsequently introduced to the West, where eventually instruments such as the harmonica, accordion, and reed organ were developed. The reed organ enjoyed a period of great popularity in North America for about a century, beginning around 1835, with thousands of instruments per year being manufactured. The reed organ was introduced to Japan and also became successful there, with Japanese reed organ production beginning in the 1880s. The instrument often called the ‘‘American’’ reed organ differs from the ‘‘European’’ harmonium in the use of a vacuum exhauster rather than a pressure bellows to drive the free reed vibration. The reed organ does not use a resonating pipe, but reed cell dimensions and other characteristics of the reed environment as well as the shaping and scaling of the reeds, can modify the spectrum of the sound produced. Recent investigations have explored, in some detail, the variation of the frequency and amplitude of reed vibration as a function of blowing pressure.
Progress report and discussion on the continuing activity on ASA’s role in noise and its control
Authors: Sutherland Louis C. , Koyasu Masaru
Published: October 2005

This discussion meeting is being sponsored by the Technical Committee on Noise (TCN) to review progress made to date on actions initiated by the TCN. The results of the successful outreach seminar held in Indianapolis in 1996 May will be discussed, and topics for future seminars will be presented. The possibilities for a self-administered hearing test in public places will be discussed, and the feasibility of such a program examined. Reprints have been obtained of articles based on papers presented at a special noise session held in Washington, DC on 1995 June 01. The session was titled Noise: State ofthe Art—Noise Effects andControl . Two thousand copies of a 64-page booklet have been printed, and will be distributed by ASA to those interested in noise control. Cooperation with the ASA Standards Secretariat has resulted in an electronic version of the latest standards catalog with links to the abstracts of all standards, including noise standards. The information will be posted on the World Wide Web. The distribution of a floppy disk containing the same information will be discussed. Dr. Koyasu will also discuss some of the recent progress made in Japan in noise control activity.
Approximation of theoretical wind turbulence spectra from wind-speed measurements
Authors: Di Xiao , Gilbert Kenneth E.
Published: October 2005

Theoretical calculations of ground-to-ground sound propagation in a turbulent atmosphere require the power spectrum for horizontal wind-speed fluctuations, Φxx. Experimentally, however, the longitudinal spectrum, Φrr, is usually measured. In propagation predictions, a frequently used approximation is Φxx≊Φrr. Under the assumption of homogeneous, isotropic turbulence, an exact relation between Φxx and Φrr is derived and several limiting cases are discussed. For a Kolmogorov spectrum and small-angle scattering, for example, Φxx≊(11/6)Φrr is found. The practical significance of the relation between Φxx and Φrr for analyzing experimental data is discussed. It is shown that, for many situations, the approximation Φxx=constant ×Φrr is reasonable for comparing predicted and measured sound levels. [Work supported by the Army Research Laboratory and the Applied Research Laboratory, Penn State.]
Atmospheric turbulence effects on acoustic angle-of-arrival estimations
Authors: Wilson D. Keith
Published: October 2005

Atmospheric turbulence degrades angle-of-arrival (AOA) estimates derived from acoustic receiver arrays by reducing mutual coherence between the array elements. This turbulent degradation becomes most important for large propagation distances and high frequencies. Theoretical array performance is assessed by calculating Cramer–Rao lower bounds (CRLB) on the AOA estimates. CRLB calculations are presented for several different turbulence models and propagation environments. The reduction in mutual coherence is found to be caused almost exclusively by wind velocity fluctuations generated by near-ground wind-shear instability (i.e., small-scale turbulence). Large-scale eddies may play an indirect role by creating regions of locally high shear, and hence locally high C2V (the structure-function parameter for velocity).