bluuebunny/crossref_metadata_2025 huggingface dataset:
Lightweight piezoelectric polymer hydrophones
Authors: Powers J. M. , Corcella A. T. , Crooks R. E.
Published: October 2005

Low-density piezoelectric polymer hydrophones were designed for towed arrays where weight is important. Flexural mode designs consisted of two strips of polymer glued to air-backed rectangular flexural plates made of plastic or metal capable of hydrostatic pressures to 4 MPa. They had sensitivities of −200 dB//1V/μPa and capacitance values of 1000 pF. Another design consisted of double layers of polymer wrapped in three separate sections around a thick-walled, air-filled, plastic cylinder capable of hydrostatic pressures to 12.4 MPa and was positively buoyant. The sensitivity was −212 dB//1V/μPa and the capacitance was 13 000 pF. A final design used the hydrostatic coupling mode and should have great pressure capability. It consisted of 32 layers of polymer, electrically connected in series, and held together by a tight-fitting, oil-filled, polyurethane boot. This hydrophone is semiflexible. The sensitivity was −203 dB//1V/μPa and the capacitance was 350 pF. Details of the designs are discussed and sensitivity variations with pressure and temperature are presented. [Work supported by NAVMAT and NAVELEX.]
PVDF infrared detector and microphone for monitoring
Authors: Shigiyama K. , Miura K.
Published: October 2005

PVDF has both piezoelectric and pyroelectric characteristics and is easy to fabricate as a polymer. Microphones, headphones, and tweeters using PVDF have already been developed and are used in practice. However, the PVDF infrared detector is still under development. We have now developed the sensor having characteristics suitable for both an infrared detector and the microphone with PVDF. The sensing film of the microphone is a sheet of 10–30-μm thickness of PVDF thin-film, a slightly curved arch at right angles to the stretched direction. This sensor is so constructed, having an electrode of thin-coated Ni or Ni-Cr metal film on both surfaces of PVDF film, that it is sensitive to infrared rays and sound pressures at the same time. Piezoelectric output due to the sound pressure is picked up and amplified by the audio-frequency amplifier, and pyroelectoric output due to the infrared ray is also picked up and amplified by a low-frequency amplifier of less than 20 Hz. The sensitivity of the infrared detector is nearly 108 cm Hz1/2/W (D* sensitivity), and it can detect far-infrared rays from, for example, the human body and discern someone approaching. The infrared signal drives the microphone monitoring circuit.
PVF
2
dual mode sensor technology
Authors: Huffnagle N. P.
Published: October 2005

In continuation of papers presented to the Society in the area of automatic vehicle ranging and target classification, the current effort is to prove the viability of using a polyvinylidine fluoride (PVF2) thin-film detector to detect both airborne acoustic- and infrared-emitted vehicle signatures. In brief, vehicle detection and discrimination have been performed using a single-point detector to sample both information spectra. Infrared detectors of PVF2 material have been known for some time. Our work in combining these effects to form a viable dual-mode sensor has led to some novel and interesting potential applications. Using a single detection element in this manner leads to intrusion techniques that can classify and locate intruders, sort out nonalarms, such as animals and spurious events, and open up the possibility of triple-mode operation (where the third sensor input could be seismic in nature). Vehicle signature separation is accomplished by noting that the PVF2 detector has a frequency response that is virtually flat from a few to a few hundred hertz. Typical vehicle acoustic responses lie between 100 and 400 Hz, while the infrared response occurs only during transit through a particular field of view. Although transit time is dependent on vehicle speed, this transit spectrum only has frequency components from approximately 4 to 15 Hz. The implementation to be shown uses a set of parallel-channel, bandpass filters to separate the acoustic signature from the infrared signature. Combinational logic decides vehicle type (in the acoustic domain) and vehicle presence (in the infrared domain). System outputs, then, are both vehicle type and presence.
Highway noise impact
Authors: Ostergaard Paul B.
Published: October 2005

Environmental noise, either existing or proposed, can stimulate much overt response in a community. The community can range, however, from one or two people to a neighborhood, or to an entire municipality. In some cases the community might be defined as highway designers who must meet certain statutory requirements. Various cases will be discussed where a community response has occurred as a result of environmental noise, proposed or actual. In some cases it can be seen that noise might be the focus of response, but indications are that other factors may have been at least as important.
A study for rating fluctuating noise on the basis of the additivity of annoyance
Authors: Yamamoto T. , Hiramatsu K. , Takagi K.
Published: October 2005

A method for predicting ESL (effective sound level)—the sound level of steady noise which is considered to be as annoying as fluctuating noise—has been developed. The method is based on a kind of integration of ψ, where ψ is the perceived magnitude of the sound and proportional to the sound pressure to the power 2α. By applying this method, the ESL of a fluctuating noise is calculated from the probability density function of the level of the noise. ESLs of 118 fluctuating noises with various forms of the level distribution were measured by means of the method of magnitude estimation. Good agreement was found between the calculated and observed ESLs of the noises except for those whose sound level continuously increased during the sound duration.
Research needs in community noise
Authors: Yaniv Simone L.
Published: October 2005

A synthesis and review of the literature dealing with community noise and its impact upon people is given. Important research needs, which promote the development of comprehensive noise control programs for community noise, are identified and discussed. The implications of some of these research needs upon everyday living conditions including outdoor activities are also addressed.
Noise exposure of U. S. population caused by refuse vehicles
Authors: Mintz Fred , Goldstein Jeffrey
Published: October 2005

In connection with preregulatory studies of refuse truck noise, an estimate has been made of the contribution of compacting refuse vehicles to the overall noise exposure of the U. S. population. The calculation, programmed for computer, combines the model for environmental noise contribution of refuse vehicles with a model of U. S. population distribution by land-use categories. The latter model is based on Bureau of Census data. This computerized model for noise exposure was applied in estimating baseline noise exposure due to typical vehicles currently in use and projected noise exposures for a range of possible regulatory scenarios entailing projected noise emission limits and various effective dates of regulation. Results are summarized for a selected group of regulatory scenarios.
On the magnitude of ultrasonic volume viscosity
Authors: Madigosky W. M.
Published: October 2005

Measurements of the absorption coefficient of ultrasonic waves is in all cases larger than can be explained by viscosity and heat conduction. The benchmark paper by Herzfeld and Rice [Phys. Rev. 31, 691 (1928)] introduced a third factor, namely vibrational relaxation, which for many substances accounted for most of the excess absorption. The remaining excess absorption has led to the concept of a volume viscosity, which has been historically associated with structural and other relaxation mechanisms. We have observed that when accurate absorption data is available for such diverse substances as gases, liquids, and polymers in which there is an absence of vibrational and structural (e.g., water) relaxation mechanisms, the ratio of volume to shear viscosity is exactly 2/3. We present numerous data to support this. The consistency of this ratio, and its temperature independence, suggests that the viscous counterpart to the Lamé constant λ″ is negligibly small. That is, ηv = λ″ + 2ηs/3, ηv/ηs ≈ 2/3, therefore λ″ ≈ 0. Furthermore, if λ″ ≈ 0, α/αclass = 1.5 is to be expected, and therefore, in the many substances where this is observed there is no need to postulate a structural relaxation mechanism beyond that which leads to shear viscosity. Finally, when ηv/ηs > 2/3, as in the case of water, λ″ is greater than zero. The meaining of λ″ as a measure of structural relaxation is explored.
Vibrational relaxation in binary mixtures
Authors: Choi P.-K. , Takagi K. , Negishi K.
Published: October 2005

Ultrasonic and hypersonic studies were made for vibrational relaxations in two systems of liquid mixtures, furan-cyclohexane and dichloromethane-benzene. High-resolution Bragg reflection method and Brillouin scattering were used to measure the velocity and absorption at 20 °C over a frequency range from 100 MHz to 7 GHz. In furan-cyclohexane mixtures, vibrational relaxation time for furan molecule was determined from velocity dispersion curve and found to decrease strongly with the addition of cyclohexane. This decrease was interpreted in terms of the large efficiency in vibration-vibration (V-V) energy transfer between the lowest vibrational mode of furan (605 cm−1 and the nearest mode of cyclohexane (522 cm−1). The rate constant of the V-V transfer was estimated from the concentration dependence of the relaxation time. Vibrational relaxation time for cyclohexane was found to decrease slightly with the addition of furan. Also in dichloromethane-benzene mixtures, it appears that V-V transfer is responsible for the change in the relaxation time.
Acoustochemical relaxations in the ocean
Authors: Fisher F. H.
Published: October 2005

This paper is a revue of sound absorption due to pressure-dependent chemical relaxations in the ocean. There is acoustic experimental evidence for four chemical relaxations; the salts responsible for the absorption, their relaxation frequencies and values of (αλ)max in a pH − 8 ocean at 4° are the following: MgSO4, ∼70 kHz, 4.5 × 10−6; B(OH)3, ∼0.8 kHz, 8 × 10−6; CaSO4, ∼100 kHz, 1.1 × 10−6; unknown, ∼3 kHz, 2.8 × 10−6. The existence of the CaSO4 relaxation is based on the acoustic work of Kurtze and Tamm in MgCrO4 and CaCrO4 solutions. The unknown relaxation at 3 kHz is based on data of Lovett and Morris from independent propagation experiments in the ocean. Problems remaining in understanding the boric acid and magnesium sulfate relaxations are discussed. Laboratory experiments to resolve boric acid and magnesium sulfate problems will be discussed. Experiments to confirm the 3-kHz and CaSO4 relaxations will also be discussed. [This work was supported by the Office of Naval Research and the National Science Foundation.]
Wave propagation in bubbly liquids
Authors: Drumheller D. S. , Bedford A.
Published: October 2005

A theory of bubbly liquids is developed in which the liquid and the bubbles are treated as separate continua. The equations of motion are obtained from a variational principle. The effects of slip between the liquid and the bubble are considered as well as heat conduction and surface tension. We also consider the effects of condensation and evaporation in two-phase liquid-vapor systems. The resulting equations are linearized and applied to plane harmonic wave propagation. The dispersion and attenuation solutions are compared with experimental data for air-bubbles in water. Calculations for steam bubbles in water are also presented and the effects of condensation and evaporation are discussed. In general, we conclude that slip between the bubbles and the liquid does not produce a noticeable effect on acoustic wave propagation. However, the presence of even small amounts of condensation and evaporation between the bubbles and the liquid produce changes of as large as 102 in attenuation. Also, for low concentrations of vapor bubbles the approach to thermal equilibrium is very slow and occurs only at frequencies well below 0.01 Hz. [This article sponsored by the U. S. Department of Energy under Contract AT (29-1)-789.]
Extension of Sato's mass-spring model for numerical analysis ultrasonic field
Authors: Onoe Morio , Okamoto Yukihiro
Published: October 2005

The finite-difference approximation of the equations of motion for elastic waves is a powerful tool for numerical analysis of propagating ultrasonic field [K. Harumi, F. Suzuki, and Y. Sato, J. Acoust. Soc. Am. 53, 660–664 (1973)]. It lacks, however, clear correspondences with boundary conditions, which makes difficult an analysis of a body with complicated shapes or inner defects. Recently Sato proposed a mass-spring model of an elastic medium [Y. Sato, Japanese J. Non-destruct. Test. 27, 180–185 (1978)]. The use of his model solves the above-mentioned difficulties, but is limited only to a medium, which satisfies the Cauthy's condition. (Both Lame constants are equal.) Sato's model contains only longitudinal springs. This paper shows that the addition of shear springs extends Sato's model to cover an anisotropic medium as well as a general isotropic medium and a fluid. The three-dimensional propagation, which involves both dilatational and equivoluminal waves, can also be handled. Several examples including the wave propagation in a plate will be shown. The results are useful for ultrasonic nondestructing testing.
Central denervation hypersensitivity in the auditory system of the cat
Authors: Gerken G. M.
Published: October 2005

The effect of deafness on the sensitivity of brain stem auditory mechanisms to electrical stimulation was measured using operant methodology. Seven cats were used with a total of 29 electrodes permanently placed in or adjacent to cochlear nucleus, superior olivary complex, nucleus of the inferior colliculus, and medial geniculate body. Electrical stimulation thresholds were measured prior to and following bilateral cochlear destruction. Four of the animals were also tested using a site-of-stimulation discrimination. Hearing loss was evaluated in all cats after completion of the experiments. As a result of deafness, mean stimulation threshold across all subjects, including one animal with some residual hearing, was reduced by 7.9 dB. The site-of-stimulation discrimination was not permanently impaired. It is concluded that deafness leads, on a long term basis, to widespread central hypersensitivity in auditory neurons. An alteration is predicted in the functional properties of the auditory system when it is accessed either by direct stimulation or by peripheral stimulation using residual hearing. [Work supported by NIH grant NS 08612.]
Plastic changes in the mouse inferior colliculus studied by [14C] deoxyglucose technique
Authors: Taniguchi I. , Saito N.
Published: October 2005

Plastic reorganization in the inferior colliculus after unilateral destruction of the cochlea in the mouse was studied by means of the autoradiographic [14C]deoxyglucose (DG) technique. Autoradiographs of the brains of mice exposed to white noise were obtained successively on different days after unilateral cochlear destruction. Autoradiographs obtained on the fourth day after the destruction showed an asymmetric uptake of DG into the inferior colliculi, i.e., the uptake on the ipsilateral side to the intact cochlea remarkably decreased. This result is similar to the previous observations by the neuroanatomical [K. K. Osen, J. Comp. Neur. 144, 355–372 (1972)], and by the DG techniques [L. Sokoloff, Neurosci. Res. Prog. Bull. 14, 489 (1976)]. On the 18th day, however, the DG uptake into the inferior colliculus ipsilateral to the intact cochlea increased and practically the symmetrical uptake was attained. Such plastic changes suggest that additional neural connections may be formed via the commissure of the inferior colliculi or ipsilaterally via the lateral lemniscus.
Acoustic brainstem responses in adults and infants
Authors: Klein A. J. , Kramer S. J.
Published: October 2005

BSRs to filtered clicks (FCs) with center frequencies at audiometric values were recorded from infants and adults. The latencies of Wave V for the respective FCs and intensities were compared for the two groups. The latencies are strongly determined by stimulus parameters: Low frequencies and weak intensities elicit responses with long latencies while high frequencies and strong intensities produce responses with short latencies. The strong stimulus control of response latency is most apparent in adults and is diminished in infants. For low frequency FC's, response latency in infants latencies lag adult values. As the infant matures, latency for the high frequencies decreases. Since Wave V is relatively late in the auditory chain, following cochlear mechanisms and, perhaps, four or more levels of neural processing, interpretation of our data is still uncertain. However, the differentiation of response latency re frequency suggests that one site of auditory maturation is within the cochlear system. [Supported by NINCDS contract NS-5-2313.]
How short are long waves in the cochlea?
Authors: van Dijk J. S. C. , Viergever M. A.
Published: October 2005

An asymptotic long wave technique is applied to a two-dimensional cochlear model. The method leads to an ordinary differential equation for either the pressure difference across the cochlear partition or the partition velocity. The lowest order approximation appears to be identical to the equation describing one-dimensional models; the complete approach is equivalent to that of Viergever and Kalker [J. Eng. Math. 9, 353–365 (1975) and 11, 11–28 (1977)] and that of Sondhi [Proc. 9th ICA, Madrid, ( 1977)], which thus are shown to be long wave methods as well. The validity of the one-dimensional solution is confined to waves along the basilar membrane whose wavelength λ satisfies λ >> 2πh; h is the scalaheight. The presented technique is applicable when λ > 2h, which is not only a factor of π, but also an order of magnitude better. The asymptotic approach is especially useful since it makes the physical principles in the cochlea comprehensible. The limited validity however asks for a comparison of the results of the present method with exact numerical results of the two-dimensional cochlear model, which indeed will be made in the lecture.
Peripheral vasoconstrictor reflex for whole body transient vibrations
Authors: Yonekawa Y. , Miwa T.
Published: October 2005

Photo-plethysmographic blood volume pulse amplitude was observed on five male subjects on the middle finger phalanx of the right hand under whole body impression of transient vibrations of three types, such as, one shot shock motion (SSM), one period sinusoidal vibration with an ultralow frequency (OPV), and one short burst random vibration (SBR) on a vertical vibration table. The pattern of the plethysmogram, of which amplitude decreased abruptly as called vasoconstriction and recovered gradually until certain constant amplitude observed at rest condition, was not affected by the transient vibrations of the three types. The vasoconstriction was commonly enforced with increasing the amplitude of the transient vibrations and in addition the vasoconstriction was influenced by the factors for the transient vibrations, that is, the duration for SSM, the fundamental frequency for the OPV, and the built-up time for the SBR. The relation between the vasoconstriction and these factors was quantified from the variance analysis in the experimental design method.
Field measurement of forces on the leg during skiing
Authors: Mote C. D.
Published: October 2005

Design of snow ski release bindings, intended to protect the leg from injury, has been hindered by the uncertainties surrounding the loading environment of skiing and the mechanisms of injury. An instrumentation system, utilizing a test ski with imbedded dynamometers and an FM-PCM telemetry system, was developed to monitor the detailed force transmitted to the leg during actual skiing. Using this test ski subjects skied aggressively on test courses purposely set to increase the probability of falling. Unstaged falls occasionally occurred permitting measurement of the leg excitation during conditions of release and no release of the ski from the boot, as well as the excitation typical of cruising skiing. The data show the excitation of skiing to be notably higher than previously reported. The force magnitude often exceeds the typical ultimate strength of the tibial shaft in bending and torsion. The specification of design and performance criteria for release bindings are notably complicated by this observation. The contributions of muscle activity in the ski binding release problem are significant.
Damping characteristics of golfing equipment
Authors: Curreri J. R. , Abrami P.
Published: October 2005

The dynamic properties of golf shafts and golf balls were measured. This includes natural frequencies and damping factors. Both steady-state and transient procedures were used in the test evaluations. Different types of golf shafts were used, including different steel alloys, aluminum, and graphite. Damping values are reported for the entire golf shaft portion, with and without the clubhead. Damping is also reported for smaller sections of the golf shaft by cutting the shaft into three parts and measuring each part. This was done to examine the influence of the manufacturing process since the small diameter end is processed differently from the large diameter end. Golf-ball measurements were made of a coefficient of restitution which is converted to an equivalent viscous damping factor. This value is reported for different impact velocities and for different types of manufactured balls. This includes the solid ball with and without a covering layer as well as the core wound ball. The significance of the golf shaft and golf-ball properties is discussed in the light of the entire golf swing and ball impact process. The aim is to assess the possibility of designing and manufacturing a better double impedance match between the initiation of the swing through the final stages of impact with the ball.
The dynamic response of the human leg in snow skiing
Authors: Piziali R. L. , Hight T. K. , Nagel D. A.
Published: October 2005

The nonlinear load-displacement properties of human cadaver knees were measured by applying single displacements to the tibia and fibula while the femur was held fixed. Six degree of freedom dynamometers measured all of the forces and moments. Data was digitized and fit with a four section, least squares, smoothed cubic spline. This data, the geometry and material properties of the bones, and estimates of muscle forces were used in a whole leg analytical model. Matrix condensation reduced the upper and lower leg to two elastic beam elements. General large displacements and rotations are permitted for these elements consistent with the nonlinear load-displacement characteristics of the hip, knee, and ankle joints. Numerical integration consisted of combining an energy convergence criterion for the Newmark operator and an iterative equilibrium scheme. Large rotations utilized updated direction cosines, and the forces on the bones were determined by separating rigid body and elastic displacements. The leg-ski impulse response, the response to forces measured during skiing [M. L. Hull “Skiing Injuries: Field _______”, Ph.D. thesis (U. C. Berkeley, 1975)], and the stresses in the bones have been determined.
CCD channel vocoder
Authors: Hewes C. R. , Buss D. D. , Eversole W. L. , Mayer D. J. , Hui Tit-Kwan , Hester R. K. , Pettengill R. C.
Published: October 2005

CCD NMOS technology is being applied to the development of a channel vocoder. The objective is to demonstrate a low cost, low power vocoder for 2.4 kbps speech encoding. The system is based on the Belgard algorithm and comprises two special purpose integrated circuits, one for analysis and one for synthesis, along with conventional low cost microcomputer IC's. The analyzer vocoder IC consists of a filter bank having 19 bandpass filters covering the speech band from 180 to 4100 Hz. Each bandpass filter output is rectified and lowpass filtered to yield a low resolution, short-term spectral analysis of the input speech. During each speech frame interval of 20 ms, the 19 channels are multiplexed through an on-chip logarithmic A/D converter so that the amplitude of each channel is represented by a 5 bit log PCM code. The analyzer bandpass filters are each 100 tap transversal CCD filters while the lowpass filters are implemented with switched capacitor filter technology. The speech synthesis of this system is done with a second IC having a similar filter bank. The gain of each channel in the synthesizer is controlled by the output of the corresponding analyzer channel. Excitation by a periodic pulse train or a random pulse sequence is self-contained in the synthesis IC. [Work supported by DARPA Contract No. N00173-77-C-0100.]
The TMC0280 speech synthesizer (Part I)
Authors: Wiggins Richard
Published: October 2005

The TMC0280 is a one-chip speech synthesizer based on the technique of linear predictive coding. The considerations leading to the choice of LPC-10 for a high quality, low cost speech synthesizer are outlined and the resulting synthesis algorithm is discussed. The key computational innovations introduced in this chip include a pipelined multiplier and adder. Combined with the proper computational ordering, this allows a monolithic PMOS implementation of the all-pole lattice filter. The input to the 0280 is a coded bit stream representing the parameters of energy, pitch, and reflection coefficients. Using variable data length frames, the chip is capable of generating good quality speech from a data rate of 1200 bits per second. The synthesizer operates at up to a 10 kHz sample rate. Further capabilities of the chip are discussed and a demonstration of the speech quality will be presented. [Part II of this paper will be given by Larry Brantingham and concerns the hardware implementation.]
Japanese IC technology in speech recognition and audio response
Authors: Kato Y.
Published: October 2005

Present status and some potentiality of IC technology in Japan are overviewed. Application strategies of those IC's to each functional block of speech recognizer and audio response system are discussed. The technology level appears close to that of the U. S. as for plain standard IC's, while it is far behind for those for special purpose applications. This has led us to microprocessor approach rather than developing dedicated LSI's. Further, this approach gives us the capability of system evolution. Actual examples of recently developed systems are described. One is a multimicroprocessor spoken word recognizer based upon the dynamic programming algorithm optimization [H. Sakoe and S. Chiba, IEEE Trans. ASSP-26, No. 1 (1978)]. The central processor which controls the whole system and makes final decision consists of bipolar bitslice microprocessors. The I/O interface processor is 8080. The other example is a microprocessor controlled singleboard speech synthesizer [K. Fushikida and K. Ochiai, IECE of Japan National Meeting (March 1978)].
Microprocessor based automatic speech recognition system for flight applications
Authors: Hitchcock M. H. , Coler C. R.
Published: October 2005

Advances in the semi-conductor industry have made possible significant improvements in speech recognition hardware especially in the areas of size, power consumption, and cost. A microprocessor based discrete word recognizer to be used in various flight applications is described and preliminary performance results given. The system utilizes a time warping algorithm based on spectral change to normalize the duration of spoken inputs and dynamic updating of the stored patterns to reduce the effects of slow changes in the speakers voice. The entire preprocessor including shaping circuits, filter bank, multiplexer, A/D and all digital processing including CPU, random access memory, program memory, and input/output circuitry is contained on a single 612 × 912 in. printed circuit board. Word prototype pattern storage and recognition algorithm logic is implemented externally, either through another microprocessor based system or a standard minicomputer. In this way, one slow and relatively inexpensive computer may handle several simultaneous speech inputs in real time and at a fraction of the cost of currently available equipment.
Comparison of three propagation models with detailed convergence zone transmission loss data
Authors: Boyles C. A. , Joice G. W.
Published: October 2005

A Pacific sea test was performed wherein a CW source was towed over a distance of 100 nm going out to the third convergence zone (CZ). During the test detailed transmission loss (TL) measurements were made with an omnidirectional hydrophone, the TL being measured with 2-s samples. The multipath structure was also measured using LFM techniques. The TL data analysis was done by A.M. Diamant and the LFM data by L. H. Wallman both of APL. Comparison of one set of these measurements with three theoretical propagation models was performed for the first three CZ's. Two of the models, viz., the Parabolic Equation Model (PE) and the FACT model are in widespread use. The third model is a version of H. Weinberg's CONGRATS ray trace program which was modified to handle smooth caustics. TL versus range plots were generated by each model for comparison with the experimental data. Also, probability density and distribution functions for TL were used for comparison with PE and modified CONGRATS. The multipath structure was also compared with CONGRATS.
Multipath effects on intersensor correlation of signals from a narrowband moving source
Authors: Smith P. W. , Sullivan F. J. M.
Published: October 2005

The output of some correlation processors is equivalent to the time average of the complex product of two received signal envelopes, which have been delayed and frequency-shifted relative to each other in order to compensate for differences in travel time and Doppler between the two receivers. However, in oceanic transmission, the signal is spread both in Doppler and in travel time, because of the simultaneous existence of many paths to each receiver. In the present analysis, the effect of time-spread is neglected on the hypothesis that the source signal is sufficiently narrowband. A statistical analysis for the mean squared output of the processor is carried out in terms of statistical metrics of the propagation paths, developed from physical principles in previous studies. Results of numerical calculations for some typical ocean environments will be shown. [Work supported by DARPA.]
Different point of view on the role of attenuation as a component of total propagation loss in underwater acoustics
Authors: Maples Louis C.
Published: October 2005

Propagation along the axis of a depressed sound channel characterized by a bilinear sound-speed gradient, invariant with range, is considered. The attenuation coefficient is taken as constant for the medium region of interest. A pulsed sinusoidal source is assumed, at the frequency of which the attenuation coefficient is designated α0. Using only this information and simple propagation theory, a propagation loss-versus-range table is constructed. Then, the table is taken as a set of experimental data and analyzed to evaluate the attenuation coefficient α by the method originally outlined by Urick and since employed by Thorp, Browning, etc. The value of α so determined is compared with α0 and the results are examined critically, with recommendations for an alternative measurement and analysis approach. [Work supported by the Naval Underwater Systems Center.]
Source extraction in a random medium
Authors: Vasholz D. P.
Published: October 2005

A general formulation of the problem of extracting information about nontransient sources immersed in a randomly fluctuating medium is presented. The formulation is based upon a Green function approach in which the medium is treated as a random linear system. It is shown that the problem may be cast into two possible forms, each involving an integral equation the solution of which describes the locations and spectral content of all sources. One of these integral equations is time independent and consists exclusively of deterministic quantities. The other equation is time dependent, consists of stochastic as well as deterministic quantities, and contains considerably more detail. Approximate solutions to the second equation enable one to study image undulations as explicit functions of time. Examples are presented which illustrate some of the difficulties induced by the medium fluctuations when one attempts to solve the basic integral equations.
Sound separation by generalized inverse
Authors: Yanagida M. , Kakusho O.
Published: October 2005

A statistical method to separate the sound waves from several sources is presented. The method is based on the generalized inverse of matrices in the frequency domain. The time domain expression for receiving the sound waves from n point-sources with m microphones is formulated in the system of m linear equations with n unknowns under the condition that the field is linear and homogenious, and that the sound pressure is inversely proportional to the distance. The system of equations is converted into that of the frequency domain by Fourier transform. Then, the least square solution is calculated using the generalized inverse technique and is again converted into the required solution in the time domain by inverse Fourier transform. This method is quite perfect under the ideal condition, but it has some difficulties under practical situations. Some investigations on such difficulties are studied on computer simulation.
Effect of ocean frontal extent on sound transmission
Authors: Eberhardt W. B. , Hashimoto E. , Davis J. A. , Anderson A. L.
Published: October 2005

Oceanographic fronts associated with transitions between water masses such as the Labrador Shelf Water, the Gulf Stream, and the Sargasso Sea are experimentally not well defined, and their acoustical effects are not well understood. Numerical simulation of the acoustical effects employing simple environmental representations for such water masses and associated frontal regions may shed some light on the importance of frontal extent or frontal width. Sound transmission calculations employing a two-dimensional parabolic equation (P.E.) approximation model have been made for two simple environmental representations of different tracks through the Labrador-Gulf Stream-Sargasso provinces (E. Hashimoto, Environmental Considerations Applicable to Numerical Acoustic Modeling Studies Across Ocean Fronts, this meeting). In particular, the acoustical significance of the fronts as a function of frontal width was investigated through numerical simulation. The results of this study along with the limitations imposed by the environmental representation and method of solution will be presented and discussed. [Work supported by CND/NORDA.]
Monsoon effects on bottom conjugate depth in the northern Indian Ocean
Authors: Colborn J. G.
Published: October 2005

Critical depth of the northern Indian Ocean is generally greater than the bottom depth. The more useful parameter of bottom conjugate depth has been defined as the depth above the main channel axis where the sound speed at the bottom is repeated. Seasonal charts provide information on acoustic blockage and indicate the minimum source depth required to avoid bottom interaction and provide potential convergence zone propagation. A most interesting and unusual seasonal variation is indicated in the bottom conjugate depth for the Somali Basin. Intense local upwelling produced by the Southwest Monsoon during June through September causes very low surface sound speeds. The area ceases to be bottom limited during the summer months, a season which classically should produce the most extreme bottom limited conditions. This unusual event can be shown to be a predictable annual situation. Propagation modes are analyzed using ray tracing models for the premonsoon and monsoon periods. Qualitative evaluations of acoustic propagation indicate the enhanced conditions for long-range propagation during the summer in the region affected by the upwelling.
Sound propagation in the Straits of Bab-el-Mandeb
Authors: Powers M. R. , Browning D. G.
Published: October 2005

The Straits of Bab-el-Mandeb is a natural constriction between the highly saline Red Sea and the Gulf of Aden (Indian Ocean). The merging of these two distinct water masses in the Straits results in a highly stratified sound-speed profile consisting of a high sound-speed layer overlying a low speed layer. A sound propagation experiment utilizing shots detonated at 18 m was conducted across the Straits along a relatively flat sandy bottom with an average water depth of 60 m. Results are compared with modeling predictions using the Fast Field Program (FFP).
New anechoic chamber with slits between wedges for airconditioning at Kobe University
Authors: Maekawa Z. , Morimoto M.
Published: October 2005

A new anechoic chamber was built in the Environmental Acoustics Laboratory at Kobe University in 1978. The chamber has slits between units of sound absorbing wedges installed on its ceiling and bottom serving as air inlet and outlet, respectively, so that air conditioning can be on while acoustic measurements are going. Effects of the slits on the characteristics of sound absorbing wedges were measured by the standing wave method with an acoustic tube of 30 × 30 cm. In comparison with those which have no slits, the results show that the cutoff frequency shifts a little higher. However, the characteristics in the frequency range lower than the cutoff frequency are greatly improved. The results of measurements reveal that the reason for such improvement should be a resonance effect caused by the combination of the slits and the air space behind the wedges.
Measurements of sound power, sound absorption, and sound transmission loss in the reverberant rooms of the Royal Melbourne Institute of Technology
Authors: Buntine John D. , Cook Kenneth R.
Published: October 2005

The new reverberant rooms are described. They consist of a single room of 200 m3 volume which is employed for sound power and sound absorption measurements and a smaller coupled pair of rooms each of approximately 110 m3 volume used for measurement of sound transmission loss. Each room is of 30 cm reinforced concrete and vibration isolated from the main building structure. The floor plans are pentagonal, the ceilings are inclined, and some of the walls are tilted from the vertical. For the large room without absorbers or any type of diffusing element, the reverberation time is 10 s at 500 Hz. The spatial variations of the sound field and of the reverberation times are described, together with preliminary results on the measurement of sound power, sound absorption, and sound transmission loss.
New method of measuring reverberation time by Fourier transforms
Authors: Aoshima N.
Published: October 2005

Reverberation time can be measured from pulse response by integrating squared signal reversely [M. R. Schroeder, J. Acoust. Soc. Am. 37, 409–412 (1965)]. By this method, however, effects of filters cannot be separated from room characteristics. The author devised new method by which reverberation responses of rooms only can be obtained. It is shown that “reverberation” of linear system at time t (ensemble average of squared output at t) is given by integrating |G(f,t)|2 over certain frequency range, where G(f,t) is the Fourier transform of the inpulse response of the linear system from t to infinity. In this method, integrations on time axis are made not for squared responses as Schroeder's method but for complex numbers derived from inpulse responses. Computations, are made by mini-computer using FFT alogorithm and obtained reverberation decays are displayed on CRT.
Acoustical characteristics of outlet with glass wool chamber and duct
Authors: Furukawa S. , Miyano H.
Published: October 2005

Low-frequency sound generated from air-duct systems is difficult to reduce by a terminal device. We have, therefore, studied the use of an outlet with glass wool chamber and duct to increase the effective insertion loss. As the transmission loss of the chamber walls is small, low-frequency sound energy equivalent to the sound insulation power of ceiling board can be passed into the space above the ceiling by using these units. The acoustical characteristics of steel plate chambers and flexible ducts were also studied for comparison. According to the results obtained, the power level of transmitted sound at 125 Hz is lowered by about 1 dB by the use of a glass wool chamber, and by about 12 dB in the case of a steel plate chamber. The sound pressure level difference of an auditorium ceiling is usually about 15 dB.
Optical fiber hydrophones employing incoherent light sources
Authors: Carome E. F. , Schmidt P. B. , Satyshur M. P.
Published: October 2005

Extensive research is in progress on the use of optical fibers as direct acoustic sensors. Results reported to date have emphasized acoustically induced optical phase modulation effects [e.g., J. A. Bucaro, H. D. Dardy, and E. F. Carome, J. Acoust. Soc. Am. 62, 1302 (1977)]; high coherence laser light sources are required for their detection. We also have detected acoustically induced signals on incoherent optical beams propagating in acoustically irradiated fiber elements and attribute these to intensity modulation processes [E. F. Carome and M. P. Satyshur, J. Acoust. Soc. Am. 63, (S)10 (A) (1978)]. We are continuing studies of this latter phenomenon and will report results of our recent experiments. Detailed data on the sensitivity and other characteristics of several different multimode optical fiber sensors will be presented. [Work supported in part by OWR.]
A nearfield, underwater measurement system
Authors: Marciniak R. D.
Published: October 2005

A nearfield acoustic measurement technique for accurately computing the farfield radiation characteristics of underwater sound transducers was developed, implemented, and evaluated. The analysis was based on evaluation of a form of the Helmholtz integral which utilizes a Green's function that vanished over the surface of integration, thereby requiring only knowledge of the nearfield acoustic pressure for computing the farfield radiation. An experimental program was conducted to provide corroboration of the technique. Details of the automated, digital, nearfield measurement instrumentation are presented. Comparisons between conventionally measured farfield properties of four different types of test transducers and the results obtained by using the nearfield technique are given. The results showed that excellent agreement was achieved for the radiation properties compared. For the directional responses, the nearfield technique matched the conventionally measured results over the minor lobe structure as well as over the major lobe structure of the pattern. The results also showed that absolute sound pressure levels and directivity indices could be ascertained accurately and with relative ease. [Work supported by NAVSEA 034.]
AN/FQM-12(V) sonar test set. II: signal generation and processing for an automated acoustical measurement system
Authors: Ashby L. Wayne , Truchard James J.
Published: October 2005

This paper describes the signal generation and processing capabilities of an automated acoustical measurement system. The basic design goals for the signal generation and timing capabilities of the system were to provide the maximum flexibility for signal generation and reception while using sampling techniques that allow fast signal processing algorithms to be used in the data processing and analysis. The waveform and timing generator was designed to generate the conventional sinusoidal pulse waveforms used in many acoustical measurements along with special purpose arbitrary waveforms which can either be used directly as the transmit signal or to modulate the sinusoidal transmit pulses. The system timing technique allows a user to make simple single pulse measurements or to make multiple pulse measurements where data samples are summed either coherently or incoherently to maximize the signal-to-noise ratio. [Work sponsored by NAVSEA 06H4D.]
Study on detection of steel bar in concrete with electromagnetic impact driving method
Authors: Motooka S. , Okujima M.
Published: October 2005

The authors devised a new detection method of measuring the location of a steel bar within concrete on the concrete surface through an electromagnetic impact driving method. Through this method, a steel bar within concrete is driven directly into vibration with occurred impulsive magnetic flux at a flat spiral coil set on the concrete surface. That is, a steel bar within concrete is regarded as a sound source. The location of a steel bar within concrete is determined by the measured propagation time within concrete of radiated sound from a steel bar and the sound velocity in the concrete block. The experiment on the detection of a steel bar at a depth of less than 25 cm under the concrete surface was performed. From these experimental results, this method is expected to be useful for the detection of a steel bar within concrete. Furthermore, the propagation characteristics of sound in the concrete block and driving conditions for a steel bar within concrete were also examined.
On the sensitivity of the fiber-optic hydrophone
Authors: Shajenko Peter , Flatley James P. , Moffett Mark B.
Published: October 2005

Previous treatments of the sensitivity of the fiber-optic hydrophone have neglected the effect of the fiber's dimensional changes in response to the acoustic pressure. It is shown that the changes in length actually contribute more to the hydrophone sensitivity than the refractive index changes. The two contributions are of opposite sign. An experiment verifying the sensitivity calculation was performed with 60-kHz pulses from a spherical source. [Work was supported by NUSC Independent Research Program.]
Applications of the finite element method to the vibration of a loudspeaker diaphragm
Authors: Yashima O. , Suzuki H. , Shindo T.
Published: October 2005

The vibration modes and eigenfrequencies of a cone-type diaphragm were investigated making use of the finite element method. First, the equations to predict the first and second eigenfrequencies of a straight cone with free boundary were derived from the numerical results. The eigenfrequencies become smaller as the cone angle and the ratio of outer to inner radius increase. The cone thickness has relatively small effect on the eigenfrequencies. The vibration modes and eigenfrequencies of a cone diaphragm with a voicecoil at the inner perimeter and a surround at the outer perimeter are different from those with free boundary. The surround has its own resonant frequencies decided by its physical properties and shape, which are independent on the diaphragm parameters. Young's modulus and mass density of the surround are most effective on the first mode of the diaphragm. Higher modes are much less sensitive to these parameters. The modes, eigenfrequencies, and the frequency response of a loudspeaker with an aluminum diaphragm were compared with the numerical results with good agreement.