bluuebunny/crossref_metadata_2025 huggingface dataset:
Self-calibrating apparatus for the measurement of dynamic compressibility
Authors: Edmonds David E. , Nolle A. W.
Published: October 2005

An apparatus has been developed to measure the complex dynamic compressibility for a sample without depressurizing the test chamber. Traditional measuring devices [McKinney et al., J. Appl. Phys. 27, 425–430 (1956)] require three measurement conditions per data point: chamber empty, containing an unknown sample, and containing a calibration sample. These chambers must be reassembled and repressurized for each condition, thus introducing two sources of error: the reassembly process, and the effect of pressure history on the internal piezoelectric transducers. The new apparatus employs a partition that allows the sample to be acoustically “added to” and “taken out of” the chamber. The chamber can be used to test organic solids in a range of 0–70 MPa (10 000 psi), below 0–60 °C, and up to 1200 Hz. Measurements at higher frequencies are feasible when corrections are made for acoustical inertances and resistances. Results for poly (vinyl acetate) have the same pressure dependence, and are the same in magnitude (within 7%) as those for a sample (though of different molecular weight distribution) measured by McKinney and Belcher [J. E. McKinney and H. V. Belcher, J. Res. Natl. Bur. Stand. Sect. A 67, 43 (1962)].
Measurement of loss factor by observation of phase shift and noise reduction of panels by damping
Authors: Kanazawa Jun'ichi , Yoshimura Jun'ichi
Published: October 2005

The measurement of a decay curve or a frequency response is usually used to measure the loss factor of panels. Using an FFT analyzer, the phase shift was observed near the resonant frequency in this study. The loss factors measured from the phase shift and other methods were in good agreement except for the cases of higher loss factors at frequencies of higher modes of the panel. Theoretically, the loss factor is inversely proportional to the radiated sound power if the signal is random. In this study, the sound-pressure levels radiated from various panels of the same size and different loss factors were measured. They agreed fairly well with the predicted values, but the results for the panel with very low damping exhibited discrepancies from the theoretical values. This was due to the influence of the internal impedance of the panel as a vibrational source.
Acoustic coatings for a large water-filled tank
Authors: Corsaro Robert D. , Covey Joel , Houston Brian , Spryn Gregory , Bednarchik Paul , Weaver Duane
Published: October 2005

This paper describes four sound absorbing coatings designed to reduce extraneous wall echoes in water-filled pools in the frequency range 7.5–25 kHz. These coatings were specifically developed for rise in the NRL Acoustic Pool Facility, where acoustic studies have occasionally been hampered by reflections from the concrete pool walls. At frequencies above 25 kHz, the pool is sufficiently large (11.5 × 8.0 m and 6.1 m deep) that wall echoes can generally be excluded from the data record by using familiar time-domain filtering; however, this filtering is not helpful for many interesting test geometries at frequencies below 25 kHz. Hence, to extend the usable frequency range of this facility, echo reducing wall coatings were considered. The most economical treatment meeting these requirements was found to require the use of four anechoic (antireflective) coatings, each with different characteristics and each covering selected regions of the pool. These four coatings were subsequently developed from classical design principles, using either layers or wedges of absorptive rubber. The acoustic characteristics of the materials used are also discussed.
Ultralow- and very-low-frequency seismic and acoustic noise in the Pacific
Authors: Orcutt John A.
Published: October 2005

A discussion of the sources of seafloor noise is most conveniently broken into four frequency bands since the noise is dominated by different physics in each of these bands. The first, and most familiar, band is from 3–50 Hz. This band is known as the very-low-frequency (VLF) or infrasonic band but is termed high-frequency noise for these puposes. The best documented mechanisms for the generation of ambient noise in this band is shipping. The next lower band, from 80 mHz to 3 Hz, is commonly called the microseism band after the high-level microseismic noise that is clearly recorded at all sites on the Earth's surface and results from nonlinear wave-wave interactions. The third band, the noise notch (20–80 mHz), has a variable bandwidth and is observed on both the continents and in the ocean. Noise levels within this notch appear to be controlled largely by currents and turbulence in the seafloor boundary layer. The final ultralow-frequency (ULF) band extends from dc to 20 mHz and the levels can be attributed to surface gravity waves. Very limited pressure and inertial displacement measurements have been made in the Pacific within these bands.
Estimation of sediment shear Q using horizontal component OBS refraction data
Authors: Bromirski Peter D. , Duennebier Frederick K. , Frazer L. Neil
Published: October 2005

Shear Q was estimated by amplitude comparison of horizontal component reflectivity synthetic seismograms with exceptionally high-quality horizontal component ocean bottom seismometer (OBS) refraction data collected at DSDP site 581C in the Northwest Pacific using a Soviet air gun source. Computations were performed on a CRAY X-MP/48 using an algorithm developed by Mallick and Frazer (1987). Sediment modeling was constrained by well log data and two-way VP travel time from a normal incidence reflection profile. The velocity structure of the sediments and the upper 2 km of the basement was refined using ray tracing and reflectivity modeling. A strong shear conversion at the top of layer 2 constrains the shear-wave travel time through the sediments to about 1.8 s, giving an average VS of about 0.2 km/s. Additional constraints on the modeling were imposed by matching the amplitudes of the primary as well as later arrivals. Preliminary results of a sensitivity analysis of the modeling indicate a minimum sediment shear Q of about 200 (α = 0.0136 N/wavelength) at 9 Hz is necessary to match amplitudes of later arrivals. A stability analysis to determine the sensitivity of the modeling to QP, QS, density, and velocity will be presented. [Work supported by ONR.]
Observation of impulse response for ultrasonic transducers by optical computed tomography
Authors: Obara Hiroyuki , Shibayama Kimio
Published: October 2005

The present study is aimed at developing accurate and simple techniques to observe the behavior of ultrasonic transducers and their holders. To know the impulse response of wideband type transducers driven in water, it is important to investigate the ultrasonic field pattern under actual performance. One solution would be given by the combination of the optical CT and the impulse response analysis by the M-sequence correlation method. The experimental arrangement of the Raman-Nath diffraction was used. The signal was given by a PIN-photodiode, which included the responses of all the components of the Fresnel diffraction. The optical measurement system using the M-sequence correlation method gives high stability and sensitivity over a wide frequency range. The ultrasonic signal in the megahertz region was processed by a low-cost slow A/D converter with 12-bit resolution. The experiment was conducted with a 13-mm concave transducer, and the instantaneous wave front was reconstructed every 50 ns. The effect of the electrodes could be clearly observed.
Two-dimensional optical beam scanning using a leaky surface acoustic wave device
Authors: Urabe Heijiro , Toda Kohji
Published: October 2005

Two-dimensional (2-D) optical beam scanning is obtained by using a leaky surface acoustic wave device. The device consists of two interdigital transducers (IDTs) on one of the surfaces of a piezoelectric substrate, operating at a liquid-solid boundary. Ultrasound radiated into the liquid from each of two IDTs is valid for acousto-optic (AO) deflection gratings, one for horizontal and the other for vertical deflection. The fundamental performance of the AO deflector is characterized in the relation of the input electrical admittance of the leaky wave transducer with the specific acoustic impedance of the liquid by an equivalent circuit model analysis. The device has the linearity essential for a laser scanner in a wide frequency range. The 3-dB intensity area of the 2-D deflected spot is 0.8 deg for a frequency bandwidth of 30 MHz. The access time is 0.66 μs, and the reasonable spot numbers are 20×20 for an optical beam diameter of 1 mm.
A multilmth calculation of surface-generated underwater acoustic ambient vertical directivity
Authors: Kennedy Robert M. , Szlyk Thomas K.
Published: October 2005

A general calculation of the spatial correlation function due to a distributed acoustic source near the water surface has been made [W. A. Kuperman and F. Ingenito, J. Acoust. Soc. Am. 67, 1988-1996 (1980)] using normal mode theory. In this paper, a multiple ray path analysis is used to calculate the acoustic ambient vertical directivity function. Multipath propagation and ambient directional functions are physically intuitive concepts to underwater acoustic theory and direct application of these notions has advantages in the interpretation of data. Existing procedures and computer code [H. Weinberg, J. Acoust. Soc. Am. 58, 97-109 (1975)] were used to expand the received pressure field, from a distributed source, into a sum of terms interpreted as multiple propagation paths. Standard forms for the source function and a geometric transformation were used to convert the pressure field to a solid angle density function. Surface roughness, bottom geoacoustic parameters, and sound velocity-depth profiles measured in the Tongue of the Ocean were used in the calculations to predict hydrophone array performance in that area. Directivity function versus elevation angle and frequency were displayed for observers above and below a surface duct. A path-by-path contribution to the vertical distribution of energy is discussed.
Koto scales and tunings
Authors: Andô Masateru
Published: October 2005

The koto, an instrument consisting of 13 strings stretched over a long sounding board, is tuned by a set of movable bridges, one for each string. The koto's musical range normally extends from A1 to A7. Within this range, two and a half octaves are most commonly utilized. Excluding modern pieces, most koto pieces are constructed from a five-tone scale. Generally classified, there are two types of scales that can begin on any tone; a minor-sounding scale (mi-fa-la-si-do-mi), and a major sounding scale (re-mi-sol-la-si-re). The most common tuning is known as Hira-jôshi, which begins on the fifth string of the koto and follows the minor-sounding scale noted above. Tuning on the koto is accomplished by first setting string No. 1 to the desired pitch. Then the fifths, fourths, octaves, firsts, and minor seconds are tuned, creating a scale close to the Pythagorean scale. Like piano tuning, inharmonicity is observed. Several of the traditional tunings for koto music have the same scale as the minor-sounding Hira-jôshi except that they begin on a different string; No. 6, 7, 8, or 9, rather than string No. 5. Likewise, the major-sounding scales can begin on any of the same five strings.
Sound reproduction in a car cabin using a digital filter network
Authors: Hanba Michio , Hamada Hareo , Miura Tanetoshi , Kiryu Yoshinori
Published: October 2005

Designs to reproduce stereo sound fields in a car cabin have many problems, such as the asymmetrical positions of loudspeakers (for the listener) and the complex characteristics of the sound transmission. Therefore, a technique is investigated in which the sound field in a car cabin is transformed using a digital filter network, keeping in mind the interaural sound pressure in both ears of the listener. The digital filter network can be realized from the transfer functions measured by a head and torso simulator in the car cabin and in the desired sound field. The purpose of using the digital filter network is to equalize the sound pressure in both ears of the listener (when the digital network is not used) to that obtained in the desired sound field. The realizability of the desired sound field in a car cabin was confirmed by computer simulation.
Cockpit noise generation mechanisms in light aircraft
Authors: De Metz Fred C.
Published: October 2005

The high noise levels occurring in the cockpits of light, powered aircraft can significantly contribute to pilot fatigue, interfere with pilot communications with passengers and air traffic control, cause permanent hearing loss, and detract from an otherwise pleasurable flight experience. The levels and spectral features of the cockpit noise field are presented for a number of popular types of light aircraft, under different operating conditions. The dominant noise generation mechanisms are identified and the effectiveness of practical and impractical noise reduction techniques for existing light aircraft are estimated. The noise control design features for a new generation of light aircraft are explored.
Inhomogeneous plane waves: Energy conservation and reflection coefficient
Authors: Leroy O. , Claeys J. M. , Quentin G.
Published: October 2005

In studying the reflection and transmission of inhomogeneous plane waves at liquid-solid interfaces, it is found that the theory predicts a minimum in the reflection coefficient at the Rayleigh angle. This phenomenon was not predicted previously using homogeneous plane waves. Even surprising is the fact that the modulus of the reflection coefficient of inhomogeneous plane waves becomes greater than unity for angles larger than the Rayleigh angle. It is shown that this phenomenon is consistent with the principle of conservation of energy and a physical interpretation leads to explanation of the shift of those waves along the surface (Schoch displacement).
Calculation of sound diffraction by means of the integral equation
Authors: Furue Yoshihiro
Published: October 2005

The Helmholtz integral equation and its normal derivative form have been widely used to calculate the acoustic diffraction by any shaped objects. A method for calculating the matrix of the simultaneous algebraic equations derived from the above integral equations is proposed to save computation. All of the four kinds of surface integral should be calculated to determine the elements of the matrix and converted to the marginal integral of the tangential surface elements of the object. The velocity potential, or its normal derivative on each surface element is assumed to be constant and the calculation point is located at its center. Numerical examples for the sound diffraction by a rigid thin pipe of finite length by means of the above method are presented with measurements. In this case, the nonderivative Helmholtz integral equation cannot be used because of degeneration. Only the derivative form is applicable, in which the unknowns should be the difference in the velocity potential between the front surface and the back.
The difference between acoustic and auditory parameter signals as a cue for phonetic segmentation and categorization
Authors: Tillmann Hans G. , Pompino-Marschall Bernd
Published: October 2005

Normally, speech recognition systems are based on purely acoustic or on auditorily modeled acoustic speech signals, respectively. The difficulties in segmentation and categorization encountered by those models may partly be overcome by a technique that exploits the difference signal between acoustic and auditory parameters. The difference signal tested here was between the sound-pressure envelope and the loundness time series. Sound-pressure level (dB-scaled) was calculated as the mean value of absolute amplitude within a rectangular widow of 300 sample points (= 15 ms). Loudness was computed every 15 ms according to Paulus and Zwicker [Acustica 27, 253–266 (1972)] by using critical band amplitudes approximated by averaged DFT values (with frequency-dependent differences in length of the Hanning window; < 500 Hz: 60 ms; 500–1500 Hz: 30 ms; > 1500 Hz: 15 ms). For calculation of the difference, signal loudness and sound-pressure level were normalized to 0 < × < 1 and subtracted from one another.
Word recognition using synthesized reference templates
Authors: Blomberg Mats , Carlson Rolf , Elenius Kjell , Granström Björn , Hunnicutt Sheri
Published: October 2005

A major problem in large-vocabulary speech recognition is the collection of reference data and speaker normalization. In this paper, the use of synthetic speech is proposed as a means of handling this problem. An experimental scheme for such a speech recognition system will be described. A rule-based speech synthesis procedure is used for generating the reference data. Ten male subjects participated in an experiment using a 26-word test vocabulary recorded in a normal office room. The subjects were asked to read the words from a list with little instruction except to pronounce each word separately. The synthesis was used to build the references. No adjustments were done to the synthesis in this first stage. All the human speakers served better as reference than the synthesis. Differences between natural and synthetic speech have been analyzed in detail at the segmental level. Methods for updating the synthetic speech parameters from natural speech templates will be described. [This work has been supported by the Swedish Board of Technical Development.]
A telephone number information system for private branch exchanges
Authors: Hiraoka Shoji , Morii Toshiyuki , Hoshimi Masakatsu , Niyada Katsuyuki
Published: October 2005

A telephone speech-recognition information system has been developed using a personal computer, a speech-recognizer, and a voice synthesizer. This speech recognizer is speaker independent and has a large vocabulary [S. Hiraoka et al., IEEE ICASSP'86, pp. 2.8.1–4]. The user calls the system and gives the surname and division name of the addressee; both words are recognized and the extension number is supplied. Previously developed microphone phoneme templates were exchanged and made telephone compatible. The speech-sample data, which were used for making the previous templates were reused and converted to sound via a mouth simulator. The data were then returned via a handset to the computer through a PBX line. The new phoneme templates were validated using 18 individuals' calls of 212 words each, giving a recognition rate of 91.2%. The new effective templates were easily made because the data had been previously labeled. The recognizer produces 15 candidates for each of the two words and the personal computer combines them in order of priority to improve the effective recognition performance.
Benchmark test procedures for automatic speech recognition systems
Authors: Pallett David S.
Published: October 2005

Benchmark test procedures have been developed and implemented in the DARPA-sponsored program in automatic recognition of continuous speech. These tests make use of a database of recorded speech that includes material for both speaker-dependent and speaker-independent technologies. In implementing these uniform tests, it is necessary to define a standard lexicon of units to be scored and a standard or reference orthographic representation for each sentence, and to select a scoring procedure and error taxonomy. Considerations taken in identifying the reference lexicon and representations will be outlined. For the DARPA program, a dynamic-programming string alignment procedure was implemented, and errors were defined in terms of substitutions, insertions, and deletions. This scoring software is publicly available. Known limitations of this procedure will be discussed. [Work supported by DARPA.]
Acoustic properties for the distinction of Japanese stop consonants
Authors: Hirose Keikichi
Published: October 2005

Of all the sounds of speech. stop consonants present the greatest obstacles to automatic speech recognition. A study has been conducted to find acoustic properties that can be exploited for reliable distinction among the stop consonants of Japanese in CV utterances produced by five male speakers of Japanese. Short-time spectra were measured at the stop burst and at the onset of the following vowel. It was found that the most consistent property for distinguishing velar stops from other consonants is the presence of a dominant peak in the burst spectrum in the vicinity of the second/third formant of the following back/front vowel. On the other hand, the distinction between alveolar and labial stops can be made on the basis of the second formant transition when the stops are followed by a back vowel. This transition is concave for alveolars and almost level or convex for labials. When these stops are followed by a front vowel, however, the overall slope of the burst spectrum can be more reliably used for their distinction than the second formant transition. A similar analysis was also made on the nasal consonants of Japanese. [Work conducted at Research Laboratory of Electronics, Massachusetts Institute of Technology while the author was on leave from University of Tokyo.]
On the use of triphone models for continuous speech recognition
Authors: Lee Kai-Fu
Published: October 2005

This paper describes both completed and future work with triphone models for speech recognition. Triphone models are a powerful subword modeling technique because they account for the left and right phonetic contexts. One problem with triphone models is that there are too many models for the amount of training data available. Moreover, many triphone contexts are very similar. In view of this, generalized triphones are introduced, which are created by clustering similar triphones together using an information theoretic criterion. This technique provides a way of finding the “right” number of models for any task. With generalized triphone models, a 96% accuracy with a 1000-word speaker-independent continuous task was achieved. There are only 2500 triphones for the above task. For a different task, there will be new triphones; for a natural English task, the number of triphones will be much larger. In the future, improvement of the applicability of triphone modeling could be done by (1) training a large set of generalized triphone models for English, and (2) exploring techniques to deal with new triphones. [Work supported by DARPA.]
Automatic discovery of acoustic measurements for phonetic classification
Authors: Phillips Michael S.
Published: October 2005

One approach to phonetic classification is to rely on the use of well-motivated acoustic measurements such as formant frequencies. However, the extraction and the use of these measurements are often heuristically based, thus resulting in fragile classification performance. Alternatively, one can rely completely on learning procedures to automatically discover the phonetic regularities. But the requirement on computation and training data may become prohibitive. This paper describes an approach in which the determination of the acoustic measurements is formulated as a constrained search problem. In the present investigation, the constraints are provided through a set of generic measurement primitives based on knowledge of acoustic phonetics. These primitives, such as spectral moments or energy changes, have free parameters that control the specifics of the measurements. Primitives may be combined to form more complex measurements according to a set of constraints. The set of primitives, along with their free parameters and the combination constraints, specify a space of allowable measurements. This space can be searched using a measure of phonetic discrimination performance on a large body of training data to obtain an optimal set of measurements for a particular task. Results of classification experiments using these automatically generated measurements will be presented for various phonetic discrimination tasks. [Work supported by DARPA-ISTO under contract N00039-85-C-0254.]
Recursive partitioning and parametric transformation for isolated digit recognition
Authors: Atlas Les , Rivera Carlos , Martin R. Douglas
Published: October 2005

Recursive partitioning [Breiman, Friedman, Olshen, and Stone, Classification and Regression Trees (Wadsworth, Belmont, CA, 1984)] is a trainable classification technique that has proven useful in a variety of applications. Some advantages of this technique include very fast recognition (due to tree architecture), the flexibility of a nonparametric estimator, and the ability to self-organize partitions that may illuminate the underlying structure of the data. Due to the special nature of the speech recognition problem, it was necessary to adapt the basic concepts of recursive partitioning. For example, in order to add invariance to differences in speaking rate, a parametric transformation [Atlas et al., Proc. IEEE Conf. Neural Inf. Proc. Sys. (1988)] was used to preprocess the speech signal. This parametric transformation converted a sequence of order N LPC parameters into a trajectory in N-dimensional space, where a sequence of samples of this trajectory was used to train the classifier. Observations of these trajectories confirmed the insensitivity to changes in the time axis and also showed that the greatest changes in the trajectory occurred during transient portions of the speech. These techniques are currently being integrated into a study of comparative performance for recognition of isolated digits. [Research supported by NSF and U. S. West Advanced Technologies.]
An isolated spoken word database using CD-ROMs
Authors: Makino Shozo , Abe Masato , Kido Ken'iti
Published: October 2005

A database of isolated spoken words has a 212-word vocabulary uttered by 40 male and 30 female speakers and further a 3000-word vocabulary uttered by 10 male and 20 female speakers. The 212-word vocabulary was selected to contain most of the Japanese phoneme contexts. The 3000-word vocabulary is a set of Japanese railroad station names. The speech data were sampled at 24 kHz and digitized to 12 bits. About 1000 magnetic tapes were used for the recording of the speech data. However, because of the huge number of the magnetic tapes, it is very difficult to deliver these tapes to speech researchers, and there are many problems in maintenance. For building up a database, the CD-ROM has many merits such as compactness and price. Furthermore, the so-called High Sierra Format was published last year and most computers can access CD-ROMs. The magnetic tapes are being converted to CD-ROMs in that format. By April 1989, two CD-ROMs will be available that will contain the 212-word vocabulary uttered by 50 speakers.
Formant contour extraction by a temporally constrained search of the spectral resonance space
Authors: Clermont Frantz
Published: October 2005

An algorithm for extracting the first three formant-frequency contours of vowel sounds is based on properties of the linear prediction phase spectrum derived by Yegnanarayana [J. Acoust. Soc. Am. 63, 1638–1640 (1978)], and Yegnanarayana and Reddy [Int. Conf. Acoust. Speech Signal Process., Conf. Record, 744–747 (1979)]. The former study has shown that the negative derivative of the linear prediction phase spectrum (NDPS) behaves like a formant-enhancing filter. The latter study has shown that the Euclidean distance between pairs of NDPS emphasizes differences only around formant peak regions, and that it is easily computed as an index-weighted cepstral distance. In stage 1 of the proposed algorithm, all posible sets of four candidate peaks from the original spectrum are used to synthesize four-formant spectra. The index-weighted cepstral distances between these simplified spectra and the original spectrum become row entries in a matrix of intraframe distances. In stage 2, a dynamic programming (DP) procedure imposes continuity constraints across frame spectra. The DP-cost function is the cumulative sum of the intraframe distances plus the minimum of interframe cepstral distances. Extracting temporally constrained contours then consists of backtracking the optimum path through the spectral distance matrix.
Tests of Wiener-Hermite approximations for rough-surface scattering
Authors: Jackson Darrell R. , Thorsos Eric I.
Published: October 2005

The Wiener-Hermite expansion has been used [Nakayama et al., Radio Sci. 15, 1049–1057 (1980)] to obtain a first-order approximation for wave scattering from randomly rough surfaces. Recently, a somewhat different Wiener-Hermite approximation has been developed [Meecham and Lin, J. Wave-Mater. Interaction 1, 4–15 (1986)] and extended to second order [Meecham and Lin, IEEE J. Oceanic Eng., 357–361 (1987)]. This work is closely related to the smoothing method [Watson and Keller, J. Acoust. Soc. Am. 75, 1705–1708 (1984)] and, like the latter, evidently includes multiple scattering effects while violating reciprocity. First-and second-order Wiener-Hermite approximations that satisfy reciprocity have been developed. The second-order approximation requires solution of an integral equation for the coherent reflection coefficient. This result is then inserted in an integral expression for the incoherent scattering cross section. Various Wiener-Hermite approximations have been tested by comparing them with exact numerical calculations. The numerical calculations were performed using a Monte-Carlo method [E. I. Thorsos, J. Acoust. Soc. Am. 83, 78–92 (1988)], and the comparisons were restricted to the case of pressure release surfaces with Gaussian roughness spectra. [Work supported by ONR and ARO.]
Diffracted sound field behind a noise screen with horizontal slits
Authors: Maekawa Zyun-iti , Tsuchiyama Kouhei
Published: October 2005

Highway noise screens can have some adverse effects on the human environment such as blocking sunshine, breeze, and view. In order to mitigate these adverse effects, the authors tried to create slits in such screens, and the effect of the slits on the noise reduction by the screen has been studied. Although the result for the vertical slits was reported at Inter-Noise 86, it has become clear that using horizontal slits is more convenient to satisfy the purpose mentioned above. A parametric study by calculation of the diffracted sound field using the line integral along the periphery of the screen under Kirchhoff's boundary condition was carried out with a personal computer. The results show that there are two zones behind the screen—that is, one increasing noise reduction and the other decreasing it—and that the shielding effect for highway noise is possible to be increased up to 7 dBA by the suitable design of a slitted screen.
The ultimate community noise monitoring system
Authors: Brickson Ellwyn G.
Published: October 2005

Community noise measurements of intruding noise sources in Orange County have usually been conducted with hand-held or integrating sound-level metering systems. The ambient noise is measured either before or after the test period. When the intruding noise cannot be turned off or vehicular traffic is a major contributor to the problem, the noise must be controlled to meet a community noise standard. A more sophisticated system must be utilized to achieve compliance during an outdoor concert. One system proposed for Pacific Amphitheatre concerts would use a real time analyzer to transmit bursts of octave band pink noise from the speaker system prior to each concert. The signal would be monitored at three locations in the residental area to determine attenuation and make the correct assessment of such intruding noise. A lap-top computer can then be used to assess the noise dose and to prescribe the amphitheatre music sound level reduction so as not to exceed noise ordinance standards.
Formant frequency estimation by moment calculation of the speech spectrum
Authors: Takagi Kazuyuki , Itahashi Shuichi
Published: October 2005

Moment calculation is applied to extract the formant frequencies of a speech spectrum. Three kinds of first-order moments divide a spectrum into four frequency regions. The centers of gravity of the first three regions are calculated to give the 0th order estimation of the 1st, 2nd, and 3rd formant frequencies. Then the upper and the lower bounds of each region are modified so that the estimated frequency comes closer to the major peak of the spectrum, utilizing the second-order and the third-order moments that represent the variance and skewness of the spectral pattern. The process repeats until the k th estimation equals the (k − 1) th estimation. This modification improves the estimation precision significantly. An experiment with model spectra generated by an all-pole model gave estimation precision of 3% using formant frequencies typical of the five Japanese vowels. Speech materials spoken by five male and five female speakers were used for this experiment. The speech waveform was sampled at a rate of 10 kHz through a 5 kHz LPF, quantized into 12 bits; then the spectrum envelope was calculated with the first 24 cepstra of a 256-point FFT spectrum. The results give acceptable precision, compared with visually determined formant frequencies.
Analysis and synthesis of CV syllables in Hindi
Authors: Agrawal S. S.
Published: October 2005

Twenty-nine consonants of frequent occurrence in Hindi were combined with a cardinal vowel /ɑ/ to make CV-type syllables. These were spoken by a standard male speaker. The spectral analysis was done using a sound spectrograph and the covariance method of LPC analysis. The formant frequencies and their bandwidths were obtained in segments of varying duration from 8 to 20 ms depending upon the nature of acoustic information in the syllable. A program called “SNDSYS” was used to estimate the fundamental frequency and overall amplitude values at every 10-ms duration. These parameters were used to give a basic acoustic description and frame rules for consonant-vowel combinations. A P.C. version of Klatt's synthesizer called “KLPC” was used to synthesize the CV syllables. There are over 40 parameters and constants that are used in the default synthesizer configuration. Based on the acoustic description of different sounds, various parameters were updated at an interval of 5 to 10 ms to generate different consonant-vowel combinations. The configuration file obtained for each syllable (named as documentation file) was further used to update and change the parameters to improve the quality of synthesized speech. Special considerations related to the synthesis of voiced and unvoiced aspirated consonants of Hindi are also discussed.
Free-field acoustic calibration of long underwater acoustic arrays in a closed tube
Authors: Luker L. D. , Zalesak J. F. , Brown C. K.
Published: October 2005

A technique is described for measuring the complex free-field sensitivity and directivity patterns of long underwater acoustic arrays in a closed tube. The tube contains sound projectors and monitoring hydrophones located along its length. Peforming these free-field calibrations normally requires a large body of water to insure that the reflections from the sides, surface, and bottom have a negligible effect on the measurement. The projectors are driven to produce an acoustic pressure in the tube that is equivalent to a free-field plane acoustic wave propagating from any angle of incidence. The required projector drives, both amplitude and phase, are determined by measuring the transfer function for every projector and every hydrophone location and inverting the resulting transfer matrix. A prototype 10-m-long system was developed. The system operates over the frequency range from 100–2000 Hz and is capable of simulating depths to 600 m. Directivity patterns, obtained using the prototype, for a test line array 5 m long were in good agreement with theoretically predicted patterns. [Work supported by ONT.]
Avoiding pitfalls in mastering sound intensity measurements
Authors: Brittain Frank H.
Published: October 2005

Sound or acoustic intensity measurements yield results that were previously impractical or, in some cases, not feasible to obtain from measurements of sound-pressure level. Unfortunately, these techniques are also more sophisticated and present more opportunities to get erroneous results. Errors of 10 dB and more are not uncommon, nor is it rare to obtain irrelevant data. Because intensity is measured indirectly, these errors are often not obvious. To get results sufficiently close for engineering purposes, a considerable investment in instrumentation, education, and practice is required. This paper suggests how to begin the process of mastering intensity technology, including suggestions for initial reference materials. A series of experiments is suggested not only to develop proficiency, but to verify validity of the results determined. Examples are given to illustrate limitations of intensity techniques.
Multidimensional large deformation waves by the theory of characteristics
Authors: Ziv M.
Published: October 2005

A presentation is given of solutions to multidimensional wave propagation in solids. These solutions are obtained by a method based on the theory of characteristics. While for one-dimensional analysis a characteristics formulation suffices, it must be realized that multidimensional analysis necessitates two additional steps. First, the characteristics formulation must be extended to accommodate strong discontinuities. Second, the characteristics numerical integration must be confined to each region of influence encountered in the solution domain. For nonlinear material motions a third step is essential consisting of an all-events method of integration that mutates along the actual existing waves. Only then is adequate resolution reached for multidimensional wave propagation problems; the numerical results are then intrinsically stable and computer usage is minimal. Since, in the resultant deformation, the various wave fronts are explicitly revealed, physical interpretation is ensured. The characteristics approach, well posed, readily lends itself in a compatible manner to nonlinear transient deformation where strong shock waves occur.
Experimental measurements of rectified diffusion in oscillating bubbles
Authors: Holt R. G. , Crum L. A.
Published: October 2005

An optical scattering technique previously reported on [J. Acoust. Soc. Am. Suppl. 1 82, S12 (1987)] has been used to measure growth rates and pressure thresholds for rectified diffusion in oscillating bubbles as a function of the equilibrium radius of the bubbles. Using an Ar-I laser, the scattered light intensity from bubbles with sizes ranging from 20–80 μ and driven by a 25-kHz sound field is measured and used to determine the bubble radii as a function of time. The measurements are compared to previous experimental measurements [J. Acoust. Soc. Am. 68, 203–211 (1980)], and also to recent numerical results. [Work supported by ONR.]
A comparison of the thresholds for free-radical generation and transient cavitation activity induced by short pulses of ultrasound
Authors: Roy Ronald A. , Fowlkes J. B.
Published: October 2005

Insonification of relatively dirty water with short pulses (order 10 μs) of high-frequency (1-MHz) focused ultrasound can lead to acoustic cavitation. If the bubble activity is sufficiently energetic, free radicals are produced that can be detected as sono-or chemiluminescence in the fluid. The question remains as to how the thresholds for light production compare to measured thresholds for transient bubble activity. The threshold for chemiluminescence in gassy water doped with luminol using photon counting techniques was measured. In addition, thresholds were simultaneously monitored for bubble activity using a high-frequency (30-MHz) acoustic backscattering technique. All measurements were carried out for various combinations of acoustic pulse length and repetition frequency, and were compared with previous investigations as well as theoretical predictions. [Work supported in part by ONR and NIH.]
Seawater as a working fluid for thermoacoustic engines
Authors: Swift G. W. , Fusco A. M.
Published: October 2005

In thermoacoustic prime movers (the modern embodiment of the Sondhauss tube), a temperature gradient imposed on the fluid in a resonator produces acoustic power at the resonance frequency. Such sound sources have been built using air, helium gas, and liquid sodium as working fluids. A complete lack of moving parts makes thermoacoustic sound sources simple and reliable. The thermodynamic and transport properties of water are presented with emphasis on water's virtues as a thermoacoustic working fluid, and the possibility of using a water thermoacoustic engine as a high power SONAR projector is discussed. [Work supported by DOE/BES.]
Performance of a liquid-sodium thermoacoustic engine
Authors: Migliori A. , Swift G. W.
Published: October 2005

A thermoacoustic engine that uses liquid sodium as its working substance has been constructed. The engine generates acoustic power in liquid sodium, using heat flowing from a high-temperature source to a low-temperature sink. The measured performance of this engine disagrees significantly with numerical calculations based on a linear theory of thermoacoustic engines. The efficiency of the engine is a substantial fraction of Carnot's efficiency, and its power density is comparable to that of the conventional heat engines in widespread use. Thus this type of engine may ultimately be of practical, economic importance. [Work supported by DOE/BES.]
Chemical asphyxiants as sources of ototoxicity
Authors: Fechter Laurence D.
Published: October 2005

Total asphyxiation has repeatedly been shown to disrupt cochlear function, but moderate reduction in oxygen delivery has also been suggested as a general mechanism responsible for auditory dysfunction. Several recent experiments in this laboratory using environmental contaminants to produce moderate hypoxia have been used to study the parameters under which reductions in oxygen delivery can disrupt cochlear function. Both transient and permanent impairments have been studied. Acute carbon monoxide exposure increases cochlear blood flow and at higher doses produces transient disruption of the compound action potential. This loss occurs specifically at high frequencies. When carbon monoxide and noise are presented simultaneously, permanent auditory loss results, which is substantially greater than the effect of noise alone. The impairment is greatest at high frequencies and injury to hair cells occurs in the basal turn when white noise is employed. Evidence of ototoxicity by such chemical asphyxiants as butyl nitrite and trimethyltin will also be presented. Taken as a whole, the data do support the hypothesis that environmental agents which impair oxygen delivery and utilization are potent ototoxicants and suggest that hypoxia may be an important mechanism for disrupted auditory function.
Effects of lead acetate on guinea pig—Electrophysiological study of the cochlear function
Authors: Yamamura K. , Ohno H. , Terayama K. , Yamamoto N. , Kishi R.
Published: October 2005

Segmental demyelination and axonal degeneration of motor nerves induced by lead exposure are well known in man and animals. The effect of lead acetate exposure to man may involve the cranial nerves, since vertigo and sensory neuronal deafness have been reported among lead workers. However, there are only a few reports concerning the dose effects of lead acetate both on the cochlear and the cranial VIII nerve. The effects of lead acetate on the cochlea and the VIII nerve using CM (cochlear microphonics), AP (action potential), and EP (endocochlear potential) of the guinea pigs are investigated. The results of this experiment were as follows: No changes of CM and EP were found by lead acetate exposure. On the other hand, there were an elevation of threshold but a decrease of maximum output voltage on AP.
System for monitoring peripheral circulation in the diagnosis of vibration hazards
Authors: Nohara Seiichi , Nakamura Hideki
Published: October 2005

Peripheral circulatory disorders such as Raynaud's phenomenon are thought to be basal symptoms of vibration hazards. The elucidation of their pathogenesis and their diagnosis between attacks is regarded as important. A system for monitoring the body surface circulation noninvasively and continuously has been developed, which enables the quantitative dynamic assessment of skin blood flow. This apparatus is based on the thermal diffusion method and uses a probe incorporating a Peltier stack. After fundamental studies on this system, the peripheral circulatory functions of workers using vibrating tools were studied. The workers were divided into three groups: group A, without any symptoms; group B, with numbness and/or pain but without Raynaud's phenomenon; and group C, with Raynaud's phenomenon. The finger blood flow (FBF) of group C at rest was significantly lower than that of group A (p < 0.01). In a handgrip test and with exposure to local vibration (60 Hz, 50 m/s2), the FBF of group A significantly changed compared to its value at rest, but that of group C showed no significant change.
A sentence-level speech training system for hearing-impaired children
Authors: Shigenaga Minoru , Muramatsu Hirooka
Published: October 2005

Hearing-impaired children, or even adults, cannot necessarily pronounce words or sentences fluently, even though they can clearly articulate individual vowels. The system described here aims at helping teachers train hearing-impaired children by showing the differences between the children's pronunciations and the correct ones given by teachers. The system displays the speech waveform uttered by a trainee, together with several useful correlates including frequency spectra, vocal tract area functions, and the phonetic symbols for estimated vowels. The lateral shapes of the vocal tract are also estimated for vowel segments and displayed together with the correct shapes so that the trainee can practice the correct articulation of vowels referring to the place of articulation and the lip opening. The fundamental frequency contour and the waveform envelope of the trainee's speech are also displayed together with the teacher's, so that the trainee can practice pronunciations with correct prosody. This system is designed so as to be used also for self-training by older children with hearing impairment.
Active control of simple modes of structural vibration
Authors: Varadan Vasundara V. , Bao X.-Q. , Jeng J.-H. , Varadan Vijay K.
Published: October 2005

An active feedback control system is developed for control of beam vibration and torsional vibration. The experimental system consists of piezoceramic sensors and actuators bonded on a 32-cm aluminum beam as well as electronic circuits incorporating a negative feedback loop. Detailed analysis of the system is presented, and the obtained data are analyzed. [Work supported by the Research Center for the Engineering of Electronic and Acoustic Materials.]
Active control of impact vibration using an active damper with a preview action
Authors: Tanaka Nobuo
Published: October 2005

In order to suppress impact vibration, an active control method using an active damper with a preview action is presented. First, from the viewpoint of the preview control method, the system equations of the preview control system are derived. Then, using a parameter optimization technique, the necessary conditions for optimality are obtained. Next, by solving the optimal conditions, the design parameters of the active damper system are determined. Based upon the design procedure, the analytical results of the control effect are shown, and the roles of both the preview control and the feedback control on the control effect are investigated. With a view to clarifying the mechanism of the vibration control with the preview action, a control chart is introduced. Using the chart, the superiority of the preview control method over other control methods is discussed. Moreover, a hybrid control compensator is presented, which consists of a digital and analog component, that is, one to control the preview time and the other to calculate the control signal, respectively. Finally, an experiment is carried out, verifying the possibility of suppressing an impact vibration.