bluuebunny/crossref_metadata_2025 huggingface dataset:
Low-frequency velocity structure of continental shelves
Authors: Houtz R.
Published: October 2005

Broadly speaking, shelves are cut back into basement rocks by wave action and the erosion products are then transported to the seaward edge of the shelf. Seafloor sound velocities therefore range from more than 6.0 km/s in the “cutting” mode (SW Australian margin) to 1.5 km/s in the rapidly “filling” mode (northern Sunda Shelf). Intermediate seafloor sound velocities can result from the erosion of the original sediment cover (Barents Sea), from very slow sediment accumulation rates, and/or the addition of pelagic materials. Variable-density analog recordings of Sonobuoy data (18 Hz) from different shelf environments reveal outstanding differences in the character of the traveltime graphs. Rapidly accumulating sediments commonly have low water/sediment velocity contrasts and produce curved-path refracted arrivals. In this case, the seafloor reflections are weak but are augmented by strong curved path multiples at the minimum range, where the rays are focused at a velocity cusp. Records from areas with a high seafloor velocity yield late-arriving, energetic bands developed from multiple critical reflection events, whereas first-arrival refractions form straight lines. Curved path data in shallow water can be inverted to provide very accurate velocity/depth distributions. Because there are no masked refraction layers and no major sub-bottom reflectors, spreading losses can be computed with rather more precision than is possible with deep water data (source and receiver at the surface).
Compressional and shear wave gradients in marine sediments
Authors: Hamilton E. L.
Published: October 2005

Compressional wave velocity (Vp, km/s) versus depth (D, km) in three major sediment types are, for 20 turbidite areas: Vp = 1.511 ± 1.304D − 0.741D2 + 0.257D3; siliceous sediments (Bering Sea): Vp = 1.509 + 0.869D − 0.267D2; calcareous sediments (Ontong-Java Plateau): Vp = 1.559 + 1.713D − 0.374D2. Vp gradients, surface, and 0 to 1000 m, average 1.3 to 0.8 s−1 (turbidites); 0.9 to 0.6 s−1 (siliceous); 1.9 to 1.3 s−1 (calcareous). In sands Vp gradients vary from 18 s−1 (1 to 2 m) to about 4 s−1 between 1 and 20 m. Shear wave velocity (Vs) gradients in turbidites average about 4.7 s−1 near surface to 0.6 s−1 from 120 to 650 m. Excellent investigations in Japan have yielded Vs to 100 m in sands; gradients are 25 s−1 (1 to 2 m); 2.3 s−1 (40 to 50 m). Major factors in Vp gradients in silt-clays and turbidites are pressure-induced porosity reductions and effects on the sediment mineral frame (to 500 m: 65% of the gradient), temperature increases due to heat flow (19%), pore-water pressure increases (2%), and increases in rigidity caused by lithification (14%). These percentages vary with sediment depth. Velocity ratios and gradients may cause a small sediment sound channel.
Velocity and attenuation of ultrasound in hedoro (soft mud layer)
Authors: Kaya A. , Tsuchiya A , Nishimura M.
Published: October 2005

Hedoro that appears on the bottom of harbor is a low concentration sediment containing organic matters. To clarify the sound scattering or reflection by such a layer, the relationship between vertical profile of sound velocity and attenuation, and physical properties of a layer should be obtained. In order to simplify the measurements, a sediment layer was reformed artificially in laboratory tank by settling down the hedoro collected from harbor bottom. Velocity and attenuation were measured by pulse transmission method at 200–800 kHz on each sample extracted from several depths in this layer. Analyses of density, volume concentration, and grain size of sample were also made. Results of analyses show that these density, concentration, and mean grain size increase slightly and gradually with depth. Even though concentration of sample was enough low in this experiment, the measured velocity increases with concentration, acoustic frequency, and depth, but these tendencies agree with results of calculation by theoretical formula of suspensions taking acount of a factor of grain size distribution.
Acoustical imaging of buried objects in bottom sediment by using focused ultrasonic transducer
Authors: Ohgaki M. , Shimizu R. , Miyazawa H. , Okujima M.
Published: October 2005

The surveillance of the bottom sediment condition, especially detecting buried objects in sediment is important to assure safety in harbor construction work. Conventionally, however, a slant cylindrical object is usually detected as if there were two objects because the image is separated into two—each corresponds to its end—when using a usual plane type transducer. A test has been conducted in which, using a focused array transducer having elements arranged on a concaved surface, the detection characteristic of slant object has been found to be improved to a great extent as long as the object gradient is within a half of the transducer aperture angle. The frequencies used are 150 kHz when the object is scaled down to 1/10 and 15 kHz in the actual scale test. Two displays are used, one for the vertical plane, the other for horizontal plane. In the actual test, the cylindrical iron tube and others are buried in a sand-box of 2 m deep having 4 × 3-m2 area and apparent improvement has been observed.
Discrimination between similar sounding noiselike stimuli
Authors: Janota Claus P.
Published: October 2005

Many of the noiselike sounds to which man is exposed serve to convey useful information and, under proper circumstances, allow the listener to make inferences about the source. A model of this discrimination process as one of feature extraction and decision making about the presence of salient features is proposed. The model and the signal patterns used in the experiment are restricted to features of the sounds which are dichotomous, i.e., present in one sound pattern and absent in the other of a pair of sounds to discriminate. A modified version of the classical threshold technique was employed in which the SNR of the signal increased through the course of a trial. Subjects were required to respond with their decision regarding which of two signals was presented with the broadband background noise. Results were analysed in terms of mean SNR to respond and probability correct. In general, the findings supported the proposed model.
Effect of the bandwidth and temporal characteristics of background noise on speech quality
Authors: Pierce Linda , Pavel Michael
Published: October 2005

Previously [L. Pierce and M. Pavel, J. Acoust. Soc. Am. 62, S75(A) (1977)] we reported that for speech transmission quality the effects of background and circuit noises with identical spectral and temporal characteristics are different but additive. In this study we have examined the effects of the bandwidth and the temporal characteristics of the background noise. Our observers rated the quality of transmission for PB sentences presented monaurally together with different levels of circuit noise on each trial. The background noise was presented diotically and was kept constant over a block of trials. The background noises employed were narrow band Gaussian (300–3300 Hz), narrow band Gaussian (300–3300 Hz) modulated by a slow varying noise (4–15 Hz), wide band Gaussian (100–6100 Hz), and a noise recorded in a large office. We confirmed that the energy within the speech band is the most important determinant of speech quality. There were small effects when the temporally modulated noise was compared to the Gaussian which varied as a function of the background noise level.
Towards quantification of adaptation effects on RT
Authors: Weiler Ernest M. , Davis James M.
Published: October 2005

Chocholle (1945) proposed equations to account for the typical decline of RT with increased acoustic intensities. By altering the constant in his equation to account for Davis and Weiler's (1976) greater means, at various test frequencies, the Chocholle equation for the middle intensities accounts for preadaptation data within 3 ms at two of three intensities and three frequencies. RTs following 7 min of adaptation at two of three frequencies and three intensities may be accounted for within 5 ms. Analysis of variance had revealed a significant interaction between frequency and intensity. The increase in RT between pre and post tests, representing adaptation effects after 7 min of exposure, declined from the baseline value (50 dB) to the +10 test level (60 dB). The ratio between baseline and +10 adaptation was about 1/3 at two frequencies, which appeared to agree with the change in SDLB values reported by Weiler and Hood (1977). However, more detailed examination seems to support previous suggestions that RT and SDLB procedures measure different aspects of the effects of auditory adaptation on intensity coding in the ear.
History of architectural acoustics and noise control in Hawaii
Authors: Miyake Iwao
Published: October 2005

This paper traces the history of architectural acoustics and noise control from ancient to modern Hawaii. It begins with Hawaiian folklores on noise and concludes with suggestions on future studies on natural ventilation system with high noise transmission loss, building design and community planning for noise control. Between these limits, the paper describes an indoctrination program which eventually led the City and County of Honolulu Building Superintendent to require architects to include acoustic and noise control measures in their design of public schools and municipal buildings. Some of the changes in architectural design which followed are described. The paper also considers laws and ordinances regulating noise at the state and municipal levels, and related problems involving enforcement and conviction in the court.
Propagation of spherical acoustic noise of finite amplitude
Authors: Kamakura T. , Ikegaya K.
Published: October 2005

Some theoretical and experimental aspects of a practical problem concerning the wave motion of spherically spreading acoustic noise of finite amplitude in freefield have been treated. This deals with the changes of noise spectrum with increasing propagation distance. The theoretical analysis is based on Burgers' equation and Banta's expansion, and a stationary Gaussian noise of zero mean is assumed as an initial signal. The power spectrum of sound pressure is considered on the acoustic axis of a loudspeaker (tweeter) in an anechoic room. The source radiates 13-octave band noise with center frequency 5 kHz, and noise level at 1m from the source is 111 dB (re 20 μPa). Regarding the spectrum envelope, the experimental data obtained by digital computational technique agree well with the theoretical results.
Nonlinear acoustics in aerosols and dusty gases
Authors: Crighton D. G.
Published: October 2005

Considerable attention has been given to finite-amplitude waves in aerosols, principally by G. A. Davidson and D. S. Scott [(J. Acoust. Soc. Am. 54, 1331 (1973) and J. Aerosol Sci. 5, 55 (1974)], who derived and analyzed a single model equation for unidirectional plane waves. This note seeks to correct an error in their analysis for the case in which the thermal and momentum relaxation times of the particular phase are equal. The correct result is that the medium in that case is a relaxing gas of precisely the kind considered by a number of other authors, with equilibrium and frozen sound speeds and a bulk viscosity all capable of simple interpretation. Possibilities not previously recognized are then opened up; specifically that partly dispersed shock waves will occur under appropriate conditions at low frequencies, while at high frequencies the model equation may be reduced to the familiar Varley—Rogers equation, which demonstrates that at a given frequency shock formation may be altogether precluded if the properties of the aerosol are suitably chosen.
Dynamic Gamut
Authors: Sábat E.
Published: October 2005

We discuss mathematical studies of a novel structure that simultaneously support the Pythagorean Gamut as well as the Ptolomeic. The fundamental reason that unites both classical gamuts is that the two Ptolomeic ration units “10/9, 5/3, 5/4, 15/8” and “16/15, 8/5, 6/5, 9/5”, have their structural elements separated by a perfect fifth, through which it can be affirmed that they have a parallel structure to the Pythagorean. Dynamic Gamut must be obtained by contracting the Pythagoras Gamut. One of the best solutions consists of contracting each structural Pythagorean fifth in Ninth root of the musical atom, i.e., 3/2 divided by 1,0001254. The Dynamic Gamut, so formed, of 53 dynamic comas per octave possesses 33 tonics for all the greek modes and their mirrored equivalents of John Redfield, and for all combined Pythagorean and Ptolomeic dodecaphonics, and therefore it is very modulable. Almost all the classical important ratios untune in 1,0001254, which is completely inaudible, and 16/15, 9/8, 16/9, 15/8, twice 1,0001254.
A fundamental study of railway train squeal noise
Authors: Yokoi M. , Nakai M.
Published: October 2005

Railway train squeal noise on a curve of small radius consists of specific frequencies which correspond to the natural frequencies of the axial vibration of a wheel. Those vibrational modes are nodal modes, such as (4,0) or (7,0), and the damping coefficients of a wheel are small in those particular kinds of modes. To investigate the specific frequencies of squeal noise, we made an experimental apparatus which has a thin steel disk and a steel rod. For the purpose of our experiment we will assume that the disk serves as a wheel and the rod serves as a rail. The circumferential face of the disk is rubbed in the axial direction with the rod. The frequencies of squeal noise generated by this fundamental experiment are coincident with those of the nodal modes. Frequencies with a small damping coefficient tend to occur in the squeal noise. Thus, the results of the numerical model concerning the main frequency of squeal noise generated by the coupled vibration of a disk and a rod almost agreed with the results of our experimental model.
Transmission across, radiation from, and reflection by ribs on a panel
Authors: Maidanik G. , Crighton D. G. , Eisler T.
Published: October 2005

A formalism of the response of an infinite, plane, and uniform panel is derived in terms of the impulse response function. The modification of the impulse response function caused by attaching parallel line mechanical constraints is considered. It is shown that the impulse response function of the so constrained panel can be cast in the form of two terms. The first is simply the impulse response of the unconstrained panel. The second is a functional of the impulse response function of the unconstrained panel and the impedances of the line mechanical constraints. It is argued that the formalism so cast is particularly suitable for ascertaining the modification to the response introduced by line mechanical constraints (ribs). The argument is exemplified by deriving the expressions for the transmission coefficient across the ribs, the radiation to the farfield generated by the presence of the ribs, and the change in the reflective properties of the panel introduced by the ribs.
Evaluation of lifters by frequency response
Authors: Ooyama Gen , Katagiri Shigeru , Kido Ken'iti
Published: October 2005

This paper describes the evaluation of lifters with respect to their frequency spectra. A lifter is the quefrency window for the cepstrum and is used for eliminating the pitch components of the speech signal to obtain the spectral envelope. We have already suggested that comb lifters are useful for separating the close formants even if the pitch frequency is high [G. Oyama et al., IEEE-ICASSP 78, 19–22 (April 1978)]. It is desirable that the comb lifters should have a sharp main lobe and small side lobes in the frequency spectrum. To make the main lobe sharp, it is necessary that the quefrency of the first stopband of the comb lifter should be high, though the quefrency depends upon the pitch frequency. The lifter should have gentle slope along the quefrency axis to reduce the side lobes in the frequency spectrum. Therefore the envelope of the comb lifter should decrease gradually with an increase in the quefrency. To suppress the peaks due to the pitch, the stop bands in the comb lifter should have certain width, because the pitch frequency of natural speech usually fluctuates. Frequency spectra of various types of comb lifters are investigated on the basis of these considerations.
Displaying speech as vocal tract area function pictures
Authors: Nichol D. G.
Published: October 2005

Various linear predictive models of the speech process enable estimates of the vocal tract area function (VTAF) to be obtained throughout a speech utterance. This paper examines the benefits and difficulties of displaying the VTAFs as intensity-modulated digital pictures. In this display, distance from the glottis is plotted along the vertical axis, time is plotted as the horizontal axis, and the area is shown as intensity. The uses proposed for this display include speech and phonetic studies, automatic recognition algorithm testing and invariant feature enhancement and extraction. The major problem associated with the VTAF picture is caused by the breakdown of linear prediction models for nonvoiced speech, particularly the fricatives. A pattern recognition algorithm to detect fricatives and compensate for this weakness is described. [Work supported by DRCS.]
Some experiments with an area function dyad synthesizer
Authors: Gillmann Richard A.
Published: October 2005

An area function dyad synthesis procedure is described which is based on an area function model representing the vocal tract as a set of 14 cross-sectional areas. Area functions representing phoneme steady states (nuclei) and transitions between any ordered pair of phonemes (dyads) are stored in a dyad table. Given an input phoneme string, the synthesizer selects the corresponding sequence of nuclei and transitions and interpolates between each of the 14 cross-sectional areas, producing a model of the shape of the vocal tract changing in time. The filtering process of this vocal tract model is identical to the optimum inverse filter of linear prediction analysis, allowing direct conversion to linear predictive coding (LPC) synthesis. Experiments were performed which tested the effect of varying certain attributes of the synthesizer, viz., linear versus spline interpolation, one point versus three point transitions, nucleus versus no nucleus, and different gain roles. Results will be discussed and a tape played. [Work supported by Rome Air Development Center under Contract No. F30602-77-C-0056.]
The intelligibility of simulated underwater voice communication as measured by the NSMRL Tri-Word Test of Intelligibility (TTI)
Authors: Atkinson James E. , Lacroix Paul G.
Published: October 2005

Our previous paper suggested the TTI for use in assessing the effect of temporal distortions on speech intelligibility. Here, the TTI is applied in an investigation of the phenomenon of multipath, a temporal distortion which can severely limit the intelligibility of underwater voice communication. Evidence suggests that intelligibility may be improved if speech is time-reversed prior to underwater transmission. Results are presented for simulated forward and time-reversed speech conditions across an echo delay range of 50–500 ms as a function of the amplitude of an echo relative to that of the original signal. Asymmetric intelligibility functions are demonstrated which are dependent on the length of echo delay interval. These results are discussed in terms of temporal masking and perceptual fusion effects.
Lateral excitation of a vocal tract analog
Authors: Sekey Andrew
Published: October 2005

A schematic mechanical analog of the vocal tract was constructed, consisting of a cascade of 21 adjustable-width rectangular cross-sections of 8-mm length each. Side openings spaced at 25.4-mm intervals along the tube permitted the excitation of the tract by a standard hearing aid driver, mounted flush with the internal wall, at any one of seven locations from “glottis” to “lips.” The tract was adjusted to simulate the area functions of several American vowel sounds estimated directly from the speech wave [H. Wakita, Speech Commun. Res. Lab. Monogr. No. 9. (1972)]. The periodic excitation waveform resembled a glottal pulse train; the radiated sound was picked up by a condenser microphone in front of the “lip opening.” Waveshapes and spectra were recorded and formant characteristics measured. Fairly good agreement with the norm was noted in the frequency region up to about 800 Hz. Deviations at higher frequencies can be attributed to the violation of the plane wave propagation usually assumed for the acoustic tube model. This work is part of a collaborative effort (with Dr. Robert J. Hanson of the UCSB Speech Department) to develop a speech support system for laryngectomees. [Project supported by the American Cancer Society, Inc.]
A digital speech synthesizer
Authors: Miki N. , Nomura T. , Nagai N. , Yoshimoto C.
Published: October 2005

A method for constructing a speech synthesizer using microprocessors (AMD 2900) is described. A set of time-varying digital filters which produces speech signals was realized by a program written using only microinstructions. After each interval of a few ms, if the filter coefficients are given, the activated filter can produce output speech signals. The synthesizer consists of a 16-bit ALU with a cycle time of 270 ns, a 16-bit parallel multiplier with a calculating speed of 500 ns, and an interface controller. As for the filter configuration of the synthesizer, a new lattice form is proposed based on the production process of speech. In the synthesis of consonants, the location of the noise source can be simulated easily by the filter. The filter coefficients can be transferred from a CPU through the selector channel. The channel interface uses a microcontroller which executes the microroutine in ROM. As rewriting of the microprogram can be implemented easily, the system is flexible.
Phonetic transcription, stress, and segment durations from spelled proper names
Authors: Liberman M. Y.
Published: October 2005

As one application for a general text-to-speech project, we have implemented a system for pronouncing names and addresses. This talk sketches the portion of the system which translates orthographic input into a phonetic transcription, assigns stress, and determines durations. The spelling-decoding involves a limited morphological analysis and a set of context-sensitive rewriting rules. Stress is assigned by a simplified version of the principles described in Liberman and Prince [Linguistic Inquiry 8(2), 189–423]. A small dictionary of orthographically exceptional words is maintained. Durations are assigned according to a set of rules which take into account the segment, its segmental context, the local stress pattern, and constituent structure. All the rules for this system are written in a quasiphonological form and maintained on files separate from the program, for ease in modification. The processing described here runs several times faster than the associated speech time, on a PDP-11/45. In tests on 1000 randomly selected telephone directory entries, 91% trail entries were given a “correct” (possible) phonetic transcription and stress pattern.
Experimental text-to-speech system for the handicapped
Authors: Carlson Rolf , Granström Björn
Published: October 2005

Systems for generating speech from unrestricted text have recently been brought to a level appropriate to several applications. Since some applications, e.g., in the rehabilitation area, are radically new, user system requirements remain to be specified. To that end, a stand-alone text-to-speech system has been implemented with enough flexibility as to programming and interface to allow high speech quality and easy adaptation to different needs. The speech is generated from text by a set of explicit rules [Carlson and Granström, IEEE ICASSP (1976)]. There is a branching capability, so that different sets of rules are applied to different items, e.g., numbers. The rules are augmented by a user programmable lexicon in which the user could introduce codes for word sequences. Several system applications will be discussed: as a general text-to-speech converter in the telephone network, in automatic recording of talking books, as a vocal output for computerized information systems. In particular, experience in the use of the system as a personal speech prosthesis for a nonvocal CP patient will be reported.
Audio teleconferencing: An experimental bridge
Authors: Botros R.
Published: October 2005

The growth pattern of business, governments, and educational institutions has stimulated an increasing need for more efficient communication among several locations simultaneously via the switched telephone network. Such “conference” calls incur problems of impedance mismatch, low signal levels as well as high line noise and generally require costly and complex bridges. Other problems are added when loudspeaking or conference telephones are used because of the growth and decay rates of sound in a room, background noise and the “rain-barrel” effect. To demonstrate an efficient and inexpensive solution, an experimental bridge has been developed, using the technique of voice-actuated gain switching. The design takes account of network, room-acoustic, as well as psychoacoustic parameters. In addition, SAGC (Scanning Automatic Gain Control) has been used to reduce the disparity of signal levels from different terminals due to different line conditions. Another aspect of the bridge is that it can be made to receive calls automatically, providing for a “meet-me” feature. The compatibility of the bridge with room-acoustic, as well as network conditions is described, together with subjective results obtained.
The effect of nearby hull structure on the low-frequency radiation from a near surface volume source
Authors: Gray Leslie M.
Published: October 2005

The effect of a nearby ship hull structure on the low-frequency acoustic radiation from propeller cavitation volume modulation was investigated in model scale. Reciprocal measurements were made of the pressure in the propeller plane below a structural scale model of a 600-ft tanker (1:100 scale), a similar nonstructural pressure release model, and in the absence of any model. These measurements were conducted both in a reverberant tank, showing the effect of hull structure on the total radiated acoustic power; and in a free-field condition, showing the directive effects of the hull structure on the radiated pressure at shallow angles. The results of these measurements show that at higher frequencies (≳30 Hz for a 600-ft tanker), the hull acts as a local free surface anomaly. At very low frequencies and shallow angles, hull structural reradiation may augment the direct acoustic radiation. The impact of these considerations on modeling of ocean ambient noise are briefly discussed.
The vertical directivity and wind speed dependence of ambient noise
Authors: Buehler Barry G. , Freese Herbert A.
Published: October 2005

The results of an eighteen month ambient noise measurement program at the Azores Fixed Acoustic Range are discussed. Measurements were made over the decade of acoustic frequencies from 500 to 5000 Hz using mechanically steered, directive receiving antennas. Received levels have been correlated with wind speed and wave height at each center frequency. The angular dependence of noise level has been compared to the model of Liggett and Jacobson. The comparison is good at lower acoustic frequency but degrades as center frequency increases. The large quantity of available data, simultaneous measurements on two independent receiving systems, and the mechanical steering capability have provided a reliable set of ambient noise data. [Work supported by NAVSEA.]
Handbook of hydrophone element design technology
Authors: LeBlanc C. L.
Published: October 2005

This handbook consolidates existing information regarding hydrophone element design. The handbook presents useful working equations and includes graphical and schematic information necessary for selecting designs once system constraints have been specified. The major effort involves designs using piezoelectric ceramics since these are the primary materials used in Navy sonar systems. The handbook contains (1) Introduction, (2) Basic Hydrophone Criteria—electrical, mechanical, and environmental criteria, (3) Energy Conversion Materials—material parameters and stability characteristics of stress and time, (4) Basic Hydrophone Design Considerations—analysis of simple configurations such as cylinders, spheres, and flexural disks; hydrophone characteristics versus physical dimensions; and performance tradeoff criteria, (5) Receiving System Noise Analysis—ambient noise, receiver noise, system optimization, signal-to-noise ratio, effects of cable, magnetic hydrophones, and hydrophone arrays, (6) Special Hydrophone Design Considerations-pressure compensation features, directional hydrophones (pressure gradient and multimode hydrophones), amplifiers, acceleration cancellation features, and shock protection features, and (7) Complementary Materials—end cap materials and fill fluids. [Work supported by NAVELEX.]
Effect of bond lines on piezoelectric longitudinal stacks
Authors: Wilder Wm. Doug
Published: October 2005

Underwater sound transducers often incorporate a stack of identical, longitudinally polarized, piezoelectric ceramic elements. An exact electromechanical equivalent circuit has been derived for this type system when the number of segments is large and bond lines are negligible [G. E. Martin, J. Acoust. Soc. Am. 36, 1366–1370 (1961)]. It has been found experimentally that with bond lines on the order of 1% of the element length, significant discrepancies exist between measured and calculated parameters such as resonant frequency. In this paper an approximate electromechanical equivalent circuit is derived which accounts for finite bond lines. Measurements were taken on several stacks and the results are compared with theory.
Low-frequency underwater moving coil projector fluids
Authors: Ivey L. E.
Published: October 2005

Castor oil (dB grade) has been used as a coupling fluid in underwater transducers for many years. There are several good reasons for this, but not all of the characteristics of castor oil are best suited for this purpose. The viscosity for instance changes widely in the temperature range 0°–30°C. Since viscous damping affects the output of the moving coil projectors, the output is reduced at low frequencies as the temperature is lowered. FS 1265 fluoro silicone fluid (300 centistokes) has replaced castor oil in USRD moving coil projectors when extreme low temperatures are expected. Another fluid, monoalkyl terminated poly (1,2-propylcne oxide) (PAG) has been evaluated as a replacement for castor oil and FS 1265 in transducer application. PAG has an even lower viscosity than FS 1265 and changes less with temperature. However, the volume resistivity of PAG is lower than FS 1265 or castor oil. These three fluids were, simultaneously, tested and compared in a USRD J15-3 projector at the USRD Leesburg Facility, and then at the NUSC Seneca Lake Facility. Each of the three fluids was also tested in a J13 moving coil projector in the USRD/ATF Faculty at two temperatures. The results of the measurement demonstrate that PAG can replace FS 1265 and castor oil in moving coil transducers if a lower volume resisitivity can be tolerated.
Transient responses of piezoelectric transducer for longitudinal mode using transversal effect
Authors: Okuyama Daitaro , Kumagai Toshio
Published: October 2005

When a step voltage is applied to an electrostrictive transducer, the decay of the terminal current is not monotone. Although this phenomena caused by piezoelectric reaction has already been confirmed for the thickness mode, [C. Kasai, D. Okuyama and Y. Kikuchi, “Transient Response Analysis of Piezoelectric Transducers Using Reflection and Transmission Coefficients at Their Surfaces,” Electron. Commun. Jpn 56-A(3), 19 (1973)] it has not been confirmed for the longitudinal mode using transversal effect. This paper presents an analytical method for longitudinal mode that is similar to that used for the thickness mode, and also shows that the numerical calculations based on this method approximately agrees with the experimental results for the samples of electrostrictive columnar and rectangular plate. By these results, it became clear that the current waveform depends on the electromechanical coupling factor of the transducer material and the time constant of the driving circuit of the transducer.
Acoustical adjustments to improve tone and playing qualities of child size violins
Authors: Hutchins Carleen M. , Esty Mary Lee
Published: October 2005

Two sets of acoustical adjustments are being successfully applied to improving the tone and playing qualities of child-size violins—3/4, 1/2, 1/4, 1/8, 1/10, and 1/16: (1) Tuning a pair of free plates of a given instrument using the Chladni pattern method to indicate where and how much wood to remove from each plate for desired frequency and damping characteristics of mode 2 and mode 5; (2) Applying the principle of second harmonic reinforcement to the low G string range of each instument by adjusting the Helmholtz air resonance to be more than an octave above the open G196 Hz—between 400 and 450 Hz—so that strong tones occur over a range of two or three notes an octave below the Helmholtz resonance as the broadband input from the bowed string activates the instrument. Methods of plate tuning and frequency adjustment of the Helmholtz air resonance will be discussed and illustrated and small violins will be played to demonstrate results.
On the sound level measurements for musical-tones
Authors: Shirasuna S. , Nakamura S.
Published: October 2005

It has been noted that among the tones of a musical instrument, the same loudness does not always produce the same values of sound level measured in dBA. Therefore, the values of the readings were studied for musical tones of various pitch which had been heard to be the same in loudness. The piano, trombone, violin and some other instruments were chosen as examples. The level indicators used in the experiment were as follows: a VU-meter, a precision sound level meter (fast. impulse) (A,B,C,D) and a high-speed level recorder (A, B, C, D). The results show that it is not easy to find a suitable and simple level indicator for our purpose. In the case of the piano, for instance, the readings in dBD are closer to the loudness than are those in dBA, but there is still a noticeable discrepancy. To alleviate this problem, a diagram of values is presented—as a temporary means—which may be added to the dBC reading to obtain a reconciliation of perceived and measured sound. This diagram is specific to each pitch and musical instrument.
Atmospheric absorption adjustments for bands of noise
Authors: Marsh Alan H.
Published: October 2005

A new procedure has been developed for determining the absorption experienced by pure-tone sound waves propagating through the atmosphere. The procedure, which accounts for classical absorption processes (viscosity and heat conduction) as well as molecular energy transfer mechanisms, has a firm theoretical basis, has been extensively verified by laboratory experiment, and has been issued recently as an American National Standard ANS 1.26-1978. To apply the procedure of ANS 1.26 to determination of atmospheric absorption losses experienced by wideband random noise required development of a method that took into account the slope of the power spectrum of the sound pressure at the microphone. An important application of the method would be to adjust a measured 13-octave-band sound-pressure-level spectrum for differences in atmospheric absorption occurring under test and reference meterological conditions. Development and verification of the method are described for application to measurements of aircraft flyover noise. [Work supported by the Federal Aviation Administration.]
The development of noise abatement configurations adaptable to amphibious life science studies scheduled for Spacelab projects
Authors: Hale M. E.
Published: October 2005

Due to the adverse intensity of the acoustic noise within Spacelab, which will be generated during the launch of Space Shuttle, it has been determined that mitigating measures must be implemented in order to preserve the integrity of certain data to be collected during in-orbit amphibious life sciences experiments. Because of dynamic experimentation constraints, the preliminary design of the noise abatement system required the utilization of performance trade-offs resulting in the development of a multicomponent system to achieve effective noise control capabilities. A double-walled life support container fully filled with water and a specimen cadaver has been experimentally modelled in full scale. Its design is intended to merge noise control capabilities with life support, data acquisition, and dynamic requirements in order to achieve the significant noise reduction required by the project.
Sound radiation due to rapid deformation of an impacted plate
Authors: Akay Adnan , Cook Reginald O.
Published: October 2005

Sound radiation due to forced deformation of an impacted plate was investigated. An initial sound pressure pulse due to rapid local deformation of a plate was generated before the radiation from natural modes of the plate occurred. On the axis of impact, near the plate, a distinct pressure pulse is observed to reproduce the velocity waveform of the plate at the opposite side of the impact point. Data from experiments involving impact of spheres on plates differing in size and material properties are presented to show the plate behavior and the radiated sound pressure. Theoretical results were obtained from an analytical model in which impulsive acceleration of a plate with Gaussian spatial distribution is convoluted with the acceleration-time history of the actual impact. Theoretical results for the on-axis pressure are compared with the experimental results.
Ultrasonic propagation properties of tissues
Authors: Dunn F.
Published: October 2005

A selected review of ultrasonic propagation in mammalian tissues, organs, and aqueous solutions of biologically pertinent macromolecules is presented. The influence of water and proteinaceous constituents on absorption, attenuation, scattering, and velocity is discussed. It is observed that the ultrasonic propagation properties and organ function form a correspondence allowing for tissue characterization. [Portions of the research reported were supported by the NIH.]
Ultrasonic characterization of turbulent flow
Authors: Lee P. P. K. , Waag R. C. , Gramink R.
Published: October 2005

A model of acoustic wave propagation and scattering in an inhomogeneous moving medium is developed to describe weak scattering of a plane wave by inhomogeneities in the medium and velocity variations for the characterization of flow. When attenuation in the medium is neglected, an expression relating the scattering as a function of angle and frequency to the medium acoustic refractive index variations and spatial motion characteristics is obtained by assuming single scattering and farfield measurement. The results are intended for noninvasive characterization of blood flow but are also applicable to measuring flows of other opaque or conducting media. The method offers a quantitative characterization of turbulent flow in terms of its energy spectrum, with the range of motion inhomogeneities measured on the order of the ultrasound wavelength employed. This approach provides a more complete characterization than Doppler shift methods which do not take into account scattering due to motion-induced inhomogeneities. Measurements of scattering from turbulence behind a grid in a water tank with an ultrasound diffraction apparatus using frequency sweeping and angle scanning produced data yielding a decay characteristic of turbulence in good agreement with theoretical predictions and the results of hot wire techniques in the literature. [Supported by NIH Grant 5 RO1 HL 15016 and NSF Grant APR 75-14890.]
Computer studies of linear and nonlinear processing for reflection tomographic data: Extension of synthetic focus imaging methods for medicine
Authors: Johnson S. A. , Greenleaf J. F. , Rajagopalan B. , Tanaka M.
Published: October 2005

Data and simulated data from circumscribing and linear sampled arrays has been used to form high resolution reflection images by two techniques. The first technique (synthetic focusing) uses the mathematical equivalent of a lense to focus the image at each point by adding either linear (uncompressed) or nonlinear (compressed) transforms of the data. Logarithmic and square root compression is studied. The lense effect is obtained by backprojecting convolved or other linear transformed versions of the data along equal time propagation surfaces in the image with appropriate corrections for refraction and frequency dependent attenuation. The choice of apodizing function and different wave forms can greatly improve image resolution. The second technique (reflection ART) is not based on a focusing model but rather on direct inversion of a matrix (linear transform) model of the object-acoustic field interaction and data acquisition process by a Kacmarz iterative inversion method. The two methods are compared.
Performance of an expert spectrogram reader
Authors: Cole I. R. A. , Rudnicky A. I. , Reddy D. R. , Zue V.
Published: October 2005

The subject in our study has spent between 2000 and 2500 h learning to read speech spectrograms. We tested the subject's ability to identify the phonetic content of broadband speech spectrograms of unknown utterances during eight separate sessions of about 4 h each. The expert was presented with 23 spectrograms of English sentences and sequences of words and nonsense words, and 50 English words embedded in a known carrier phrase. The phonetic labels produced by the expert agreed with the phonetic labels produced by trained phoneticians (who listened to the speech) between 80% and 90% of the time, depending upon the scoring method used. When presented with words in a known carrier phrase, labeling performance improved to about 95%. A linguist who was presented with the phonetic transcriptions produced by the spectrogram reader was able to identify all of the words in 10 of 15 utterances, and missed only a single word in each of the remaining five. A film will be shown of the expert segmenting and labeling two spectrograms of unknown utterances.
An evaluation of segmentation for speech recognition
Authors: Edwards T. J. , Li K. P.
Published: October 2005

To evaluate TRW's automatic segmentation algorithm as a gross classifier of phones in continuous speech, we have performed two separate experiments: (1) verification of segmentation with hand-transcribed phonemic labels and (2) evaluation of the capability of the segmentation to determine the segmental and syllabic structures of languages. In the first experiment, hand-transcribed segment labels were compared with the segmentation output to determine reliability and confusability of segment classes. In the second experiment, segmental and syllabic models were developed to extract language-specific structures from a variety of talkers for each language. These models were of a finite-state (Markov-chain) type, though not necessarily segment-synchronous. Both inter- and intra-syllable structures, suprasegmental factors (e.g., duration), and parsing strategies were studied. The language description results were then evaluated linguistically using statistical methods. Current results for both experiments will be presented.
Segmentation of speech utilizing the time-domain properties of speech signals
Authors: Akamatsu N. , Niki N. , Takahashi Y.
Published: October 2005

One of the most difficult problems in speech analysis is the segmentation of speech signals. In this paper, we describe a pitch-detection algorithm which mainly utilizes the time-domain properties of speech signals and a segmentation method based on the pitch-detection algorithm. The segmentation is carried out in three phrases. First, by using the zero-crossing rate and the log energy of speech signals, the voiced-unvoiced-silence detection is approximately made. Second, the pitch period of the candidate for the voiced section can be determined by the pitch-detection algorithm. A computational algorithm for estimating pitch periods untilizes the time-domain properties of speech signals. The acoustic boundary is determined by utilizing the time variation of the similarity used in the pitch detector. Finally, according to the amount of speech energy, the remaining sections are successively classified into transitional regions and regions of voiced speech. In order to evaluate our method, we compare its performance with that of L. Rabiner's pitch-detection algorithm. Quantitative results and several examples illustrating the performance of these methods are included.
Ground vibration caused by road traffic
Authors: Yokota A. , Kimura K. , Nakagawa S. , Tokita Y.
Published: October 2005

The ground vibration caused by road traffic has been evaluated by the statistical value L10 of vibration level in Japan. L10 is usually a function of traffic flow, and moreover is influenced by road construction, characteristics of soils, and so on. However, the present study concentrates on the effect of traffic flow by model simulation. In this model, variables are the number of vehicles in two categories (heavy and light vehicles) and running velocity. The vibration levels of heavy and light vehicles and the propagation characteristics of the ground are determined experimentally. The distribution of vehicles is assumed to be an exponential probability density function (which was confirmed by field measurements), and heavy and light vehicles are located in random order. We can then get a model of the time series of vibration levels. The instantaneous levels observed at a roadside point are sampled, and L10 can be calculated accordingly. These results were compared with the measured ones. L10 measured is 2.5 dB less than the calculated level, with a standard deviation of 1.5 dB. [Work supported by The Tokyo Metropolitan Construction Bureau.]
Acoustic harmonic generation at unbonded interfaces and fatigue cracks
Authors: Buck O. , Morris W. L.
Published: October 2005

The generation of a second-harmonic signal at a flat unbonded interface between two solids has been observed. This signal is caused by passage of a longitudinal acoustic wave across the interface. The harmonic amplitude depends upon the pressure applied normal to the interface and is largest close to zero pressure, as expected theoretically. The effect has also been used to detect the presence of microcracks on the surface of Al 2024, and Al 7075, developed during fatigue with an acoustic Rayleigh wave propagating along the fatigued surface. An alteration of the harmonic amplitude of the alloy has been observed after 5% of the expended fatigue lifetime, corresponding to the formation of microcracks on the alloy surface. The harmonic signal is observed to become progressively stronger as the microcracks propagate with additional fatigue loading. The results are especially encouraging in regard to future use of the technique to measure the remaining fatigue life of a component long before large cracks, detectable with other NDT techniques, develop prior to component failure.