bluuebunny/crossref_metadata_2025 huggingface dataset:
Determination of compressional and shear wave speed profiles and attenuation in sea ice
Authors: Rajah Suhramaniam D. , Doutt James A. , Frisk George V.
Published: October 2005

Sea ice is a very heterogeneous material with brine channels, layers of varying density and crystal size, and a porous skeletal layer at the ice-water interface. Acoustic propagation under ice in the Arctic has been studied by many investigators, and efforts to predict the measured field by theory has met with limited success. One of the likely causes is the lack of adequate information on the anelastic properties of sea ice. A cross-hole tomography experiment designed to measure the spatial and temporal structure of the anelastic properties of sea ice will be described. Results of a lake experiment and an experiment conducted in the Canadian Arctic during 1989 will be presented.
Active control of NITINOL-reinforced composite beams
Authors: Baz A.
Published: October 2005

The static and dynamic characteristics of flexible fiberglass composite beams are controlled by activating optimal sets of a shape-memory alloy (NITINOL) wires, which are embedded along the neutral axes of these beams. The underlying phenomena influencing the behavior of this class of composite structural members are presented. The individual contributions of the fiberglass-resin laminate, the NITINOL wires, and the shape-memory effect to the overall performance of the composite beam are determined at different operating temperatures and initial preloads of the wires. The static deflection, stiffness, and buckling characteristics of the fiberglass beams are measured with and without the NITINOL reinforcement. Also, the modes of vibration and modal damping ratios are monitored for various operating conditions. With properly designed NITINOL reinforcement, it is shown that the beams can become stiffer and less susceptible to buckling. The modes of vibrations of the activated NITINOL-reinforced composite beams can also be shifted, to higher frequency bands, relative to those of the unactivated or unreinforced beams. Finite-element models are developed to describe the static, dynamic, and thermal interaction between the NITINOL wires and the fiberglass-resin laminates. Close agreement is obtained between theoretical predictions and experimental results. With such tunable characteristics, the NITINOL-reinforced composite beams have proven successful in attenuating the vibrations induced by cross flows. Experiments are conducted on 30-cm-long and 0.156-cm-thick reinforced beams mounted in a clamped-clamped arrangement inside a wind tunnel. The effect of varying the flow speed or the induced vibration is determined with and without the activation of the NITINOL wires. The results indicate that vibration attenuations in excess of 60% were attained near and at resonance. [Work supported under grant from ARO.]
A frequency-tracking dynamic absorber with and without feedback
Authors: Hudimac Albert A.
Published: October 2005

A dynamic absorber can be combined with a transducer to shift the antiresonant frequency by inserting an electrical element across the transducer terminals. For a disturbance with a slowly varying frequency, tracking can be effected by means of, e.g., a digital logic circuit that estimates the frequency, assigns it to a certain frequency interval, selects a corresponding interval, and selects a corresponding electrical termination to put across the transducer. At any given frequency setting, the total Q is high to obtain good suppression, thus running the chance of creating an “ear” in the frequency response. This effect can be reduced by suitable filters, shunting all or part of the electrical termination. A means, involving a special “broadband” termination, can extend the range of suppression without sacrificing Q at any frequency in the range. A composite device utilizes both a dynamic absorber (even the frequency-following kind) and a negative feedback velocity control. A brief comparison will be made among the several types indicated, as well as with standard active vibration controls.
StanisLAX: A speech processing environment to design and simulate hybrid automatic speech recognizers
Authors: Junqua I. C.
Published: October 2005

Today, a wide variety of mathematical, statistical, and artificial intelligence (AI) tools are available for speech processing. Means of storing and representing data graphically or in computers are evolving. The main purpose of this work is to integrate all these tools in a user friendly framework using AI technology. In this paper, the system StanisLAX, which is an “X-window based artificial laboratory for the simulation and interactive speech processing,” is proposed. Such a system is designed to help the speech scientist to find and develop a model for the problem, simulate the model, evaluate the model versus other models, interact with the simulation, graphically represent the model, process databases, facilitate encoding and acquisition of speech knowledge, and build hybrid automatic speech recognizers. More than an assembly of library routines, StanisLAX is a fully integrated environment designed to be an expert workstation that facilitates the transfer of high abstraction levels of speech knowledge from the speech programmer to the computer. The conception of such a system arose from four ideas: (1) the necessity of using reusable components for more productivity and flexibility in speech related software design; (2) the modular decomposability of the speech signal and speech processing algorithms; (3) the need for tools to manipulate databases and help the evaluation of automatic speech recognizers; and (4) the belief that speech-processing algorithms can best be handled through the effective and efficient cooperation of multiple sources of knowledge. In this paper, the details of a prototype system will be presented.
Toward an automatic system for distinctive feature extraction from speech
Authors: Howitt Andrew Wilson
Published: October 2005

A novel representation of acoustic information leading to the extraction of distinctive features is proposed. This representation is hierarchical and nonsegmental, and it is constructed in a three-stage process. First, frame-based parameters are calculated, which represent the spectral characteristics of the signal, with attention to the auditory representation of these characteristics. Second, events or discontinuities in the acoustic signal are detected (with the aid of the parameters from the first stage), which represent information about certain manner features. Third, information about laryngeal and place features is extracted via processing in regions based on the events from the second stage. Examples of the process of feature extraction by this method will be presented and comparison with other methods will be discussed. The proposed representation should be useful both as a front end for speech recognizers and as a research tool for the study of models of lexical access by human listeners.
Speaker normalization using neural networks
Authors: Krishnamurthy Ashok K. , Riegelsberger Edward L. , Anderson Timothy R.
Published: October 2005

This paper describes a method of speaker normalization using neural networks. The spectral feature vectors from a speaker's (speaker A) spectral space are mapped to the spectral space of a reference speaker (speaker R) using a feedforward multilayer perceptron (MLP) neural network. The MLP is trained using phoentically aligned speech data from the two speakers. The original and mapped feature vectors from speaker A are used as input to a speaker-dependent speech recognizer trained for speaker R. The recognition task is to recognize 13 vowels from the TIMIT database. The improvement in recognition accuracy is then taken as a measure of the effectiveness of the normalization scheme. The effectiveness of the speaker normalization technique is compared for a number of input representations: line spectrum frequencies, weighted cepstral coefficients, and the outputs of the Payton and Patterson-Holdsworth auditory models. [Work performed under the AFOSR Summer Faculty Research Program.]
A co-articulatory speech model based on HMM representation of phonetic features
Authors: Erler Kevin , Deng Li
Published: October 2005

Rather than the traditional approach to speech recognition of separating the acoustic observations into disjoint segments such as phonemes, this paper describes a structural approach to speech recognition that attempts to incorporate speech knowledge into the representation model. Here, the acoustic observations at any given moment are represented by a set of independent characteristics of the underlying physical system that produced the observations. The coarticulation problem can be viewed as the horizontal alignment of the characteristics as the underlying production system changes from one particular configuration to the next. Because the characteristics are independent, a flexible horizontal alignment is permitted, substantially reducing the coarticulation problem. This model is implemented using a large ergodic HMM in which each state represents one particular combination of phonetic features. The model is constrained to ensure that the feature values change in a manner consistent with the physical laws governing the mechanical generation system. Recognition results from the new system show a marked increase in recognition rate over the more conventional approach of modeling each phoneme separately. Analysis of the recognition errors also shows a substantially smaller frame error during the coarticulation period. This speech recognition model demonstrates a significantly different approach than has been previously pursued. Several issues raised by this model are dealt with including the coarticulation problem, model constraint mechanisms, selection of appropriate features, and model implementation. Results obtained so far are very positive, indicating that this model can actually improve on existing recognition rates. The reduced error during the coarticulation period points to numerous paths for further improvement of the model.
Basin-wide, range-dependent reverberation modeling on the Connection Machine
Authors: Dacol Dalcio K. , Jennings E. C. P. , Baer R. N.
Published: October 2005

A model for rapid computation of acoustic fields in three-dimensional, range-dependent, ocean environments [Kuperman et al., J. Acoust. Soc. Am. 88 (1990)] has been implemented on the Connection Machine and it has been extended to incorporate bottom reverberation. Bottom reverberation is treated by means of a Kirchhoff approximation to the Helmholtz-Kirchhoff integral expression for the exact Green's function using the approximate Green's function computed by the adiabatic normal-mode model as a starting point. This entails computing the integral over the bottom of the entire basin. A gridded acoustic environment is used that is mapped onto the Connection Machine's processor hypercube. This is crucial in allowing the treatment of the bottom interaction in a manner fully consistent with the wave nature of the problem without having to resort to the ad hoc introduction of plane-wave scattering amplitudes. Numerical results are shown. [Work supported by ONR, Code 1125OA.]
Measured estimates of monostatic and bistatic reverberant scattering strength using low-frequency linear FM pulses
Authors: Preston J. , Kinney W. , Roderick W. , Jespers S.
Published: October 2005

Measured estimates of ocean-bottom reverberant scattering strength are presented as a function of range and bearing for a number of monostatic and bistatic geometries achieved during two separate sea thais. One trial was performed off the west coast of Sardinia in approximately 2800 m of water during February of 1990, and the other was performed in the North Sea in approximately 1000 m of water during October 1988. The receiver in both cases was a horizontal array of 64 elements spaced at 0.5 m. Source and receiver depths ranged from 50–100 m. The signals used consisted of linear FM pulses centered near 1 kHz, with durations ranging from 4–16 s and bandwidths ranging from 32–64 Hz. Scattering strength values are estimated using uncorrelated (i.e., not match-filtered) pulses. The values presented vary from − 30 to − 45 dB for distances up to 20 km. Also presented are plots that show the difference between uncorrelated and correlated received levels. These plots demonstrate the extent of correlator gain against noise and reverberation.
Quantitative estimation of seafloor backscattering strength using long-range sidescan sonar data
Authors: Anderson Aubrey L. , Dwan Fa S.
Published: October 2005

In ocean water depths of a few kilometers, long-range sidescan sonar systems generate “acoustic brightness” images of seafloor swaths 10 km or more in width using energy acoustic energy returned from the sea floor by monostatic scattering. When simultaneously acquired seafloor bathymetry is available, as it is for some modern systems, the acoustic received level versus time (imagery) data from the towed sonar “fish” can be inverted to extract estimates of seafloor backscattering strength and associated seafloor grazing angle. These estimates can be made for the entire “swath” of seafloor for which imagery is usually generated. Such inversion has been performed for sidescan imagery data from a portion of the Monterey Fan off the coast of California. Published sidescan imagery for the same region [Atlas of the Exclusive Economic Zone, Western Conterminous United States (U. S. Geological Survey Miscellaneous Investigations Series 1–1792, 1986)] indicates a strongly scattering seafloor for much of the region investigated. Consistent with this, the backscattering strength values estimated from data inversion are higher than typical values from the literature in the bright image regions, though consistent with typical values over much of the local seafloor. [Work supported in part by the Geodynamics Research Inst., Texas A&M Univ.]
Scattering from a stochastically modeled seafloor approximated by a collection of discrete point scatterers
Authors: Haralabus G. , Alexandrou D.
Published: October 2005

The ability to simulate a continuous, three-dimensional seafloor with a discrete point distribution could potentially allow the modeling of the scattered field generated by complex, realistic seafloor topography. In order to evaluate this approach, a one-dimensional seafloor profile was modeled as a stationary, zero-mean, Gaussian process. It was generated using the spectral method whereby the equation of seafloor height is derived as a function of the second-order statistics and the correlation length, under the assumption of Gaussian spectral roughness. In addition, this simulated seafloor was designed to be locally flat so that the Kirchhoff approximation can be safely invoked. A discrete version of this profile was simulated with point scatterers, using the simulation program REVGEN/SST. Low-frequency (150-Hz) synthetic reverberation was created by insonifying the collection of scatterers. Measurements were made in the near-specular direction and care was taken to avoid shadowing effects. The mean pressure of the scattered field was used to obtain an estimate of the mean-squared (m.s.) roughness of the seafloor according to a well-known result of Kirchhoff theory. The success of the simulation is measured in terms of the proximity of this estimate to the true m.s. roughness, which is known a priori. [Work supported by ONR.]
Slowness-time and tau-slowness spectra for signals scattered for boundary reverberation from the seafloor
Authors: Baggeroer A. B. , Schmidt Henrik
Published: October 2005

The spatial properties of reverberation from the seafloor is often characterized both theoretically and experimentally in terms of a directional, or slowness, spectrum. These slowness spectra of signals scattered from the seafloor determine the power that propagates to become long-range reverberation. Slowness-time spectra are essentially short-time-duration directional spectrum estimates and are used when the excitation signals are transient such as a shot or a short pulse. Slowness-time spectra and their transformation to τ − p (intercept time-slowness) spectra are very powerful analysis methods for a array data since they incorporate both slowness and travel time estimates of the scattered energy and identify scattering sources. To illustrate the power of τ − p spectra in the interpretation of boundary reverberation, two different numerical approaches are applied. A finite-difference approach is applied to generate the time-domain array response, which is then subsequently transformed into the τ − p domain and compared to direct simulations of the τ − p response using a hybrid boundary element-wave-number integration algorithm. This comparison clearly illustrates the signal-processing issues involved in estimating the τ − p representation of real reverberation data. The τ − p spectra for signals scattered from (i) a rough seafloor and (ii) a buried inclusion in a stratified seabed have been estimated and used to analyze the energy partitioning from the various propagation modes and the locus of the dominant scattering centers.
Reverberation modeling for horizontal and sloping ocean bottoms
Authors: Barker Barry I. , Ioup Juliette W. , Rayborn Grayson H. , Ioup George E. , Slater Richard R. , Fulford James K. , Showalter James A.
Published: October 2005

A ray-tracing algorithm was used to model the bottom reverberation for a Pekeris waveguide with a variety of marine bottoms including sands, silts, and clays. Monostatic and quasimonostatic cases were studied. Effects of attenuation in the bottom were also considered. In addition to the horizontal case, sloping bottoms between 3 and 15 deg were investigated. Results from the model were used to draw conclusions about the magnitude and importance of these and related parameters in calculations of bottom reverberation. [Work supported by NOARL and the Navy/ASEE Summer Faculty Fellowship Program.]
The Wigner-Ville distribution: Properties, efficient computation, and application to blosonar signal analysis
Authors: Yu Biao , Lin Zhen-biao
Published: October 2005

Use of a spectrogram is the common way to perform time-frequency analysis, but it suffers from inherent trade-off of time resolution versus frequency resolution that does not provide necessary resolution in the time-frequency domain for bat pursuit signals and dolphin signals. The Wigner distribution (WD) appears ideally suited for nonstationary signal analysis. The main limitations of the WD are the interference terms, which are discussed in detail in this paper. A pair of values x(tt), x(tj) at times tt and tj produce interference oscillatory with “frequency” |tj − tt| at the position of (tt + tj)/2. The appropriate time (frequency) window length is the trade-off for the smooth and frequency (time) resolution. An efficient algorithm is proposed. This algorithm employs the fast Fourier transform (FFT) to compute two sequences of smooth pseudo-Wigner distribution; 40% of the number of arithmetic operations are saved. Twenty-one signals emitted by a Chinese Pipistrellus abramus temmink while hunting an insect over a fish pond were recorded. The 2-D time-frequency structures of bat sonar signals for searching, identification, pursuit, and capture are systematically investigated. Using the first-order moment of the WD, the instantaneous frequencies of bat signals are also obtained. As a bat is approaching an insect, the FM components are rather stationary, the CF components are shortened, and the instantaneous frequency tends to be an ideally hyperbolic one, which means the bat is keeping good range resolution, is saving energy, is providing good Doppler tolerance, and is eliminating the coupling error of Doppler delay from search to capture. In terminal phase, very short-duration and harmonic structures are significant techniques to improve range resolution. [Work supported by NSFC 1881155 of China.]
A study of shoe-box hall's room shape in consideration of directional characteristics of orchestral instruments
Authors: Toyota Yasuhisa , Oguchi Keiji , Nagata Minoru
Published: October 2005

It was reported in a previous paper [Toyota et al., J. Acoust. Soc. Am. Suppl. 1 84, S130 (1988)] that the early reflections within 80 ms after a direct sound arrival were especially important in concert hall acoustics. In this paper, the importance of taking the directional characteristics of the sound source into account in the acoustical design for concert halls is discussed. First, the idea that the upper directional part of orchestral sound is very important for concert hall acoustics was introduced. Then, the result of the study on the proportion of shoe-box hall's transverse section with the upper directional sound source is reported. It was found that shoe-box halls with the transverse section of more than 0.8 can get early reflections over a whole seating area. It was also found that the ceiling and the upper area of both side walls played an important role in the halls. These results are supported by the proportions found in existing halls such as Vienna (proportion of 0.92) and Boston (0.80).
Computer simulation technique regarding distribution of early reflections for room acoustical design
Authors: Oguchi Keiji , Toyota Yasuhisa , Nagata Minoru
Published: October 2005

The computer simulation technique is used to study the fundamental room shape of auditoriums, especially of concert halls, during the conceptual and schematic design phases in our acoustical design. The hardware of the computer simulation system consists of HP 9816 (CPU:68000, Memory: 1.25MB) and some output devices. The simulation software is based on the ray-tracing algorithm. In our computer simulation study of the room shape, the density, and the distribution of the early reflections on the audience area are observed. The omnidirectional and upper directional sound sources are assumed, and the early reflections after the arrival of the direct sound are calculated in following three time intervals, such as 0–30, 30–60, and 60–90 ms, respectively. This method has the advantage of easy modification of the room shape and visual explanation of the sound behavior in an auditorium. Then it becomes a very useful tool of our acoustical design, particularly during early design phases. On the other hand, it has the disadvantage of difficulty of treating curved surfaces.
The use of an acoustic scale model for the design of the Escondido Civic Center 1500-seat multi-use theatre
Authors: Talaske Richard , Siebein Gary
Published: October 2005

The use of a 1/24th scale physical model will be discussed regarding how it was used to aid the acoustic design of the new 1500-seat Lyric Theatre for the Escondido Civic Center of North San Diego County. The discussion will focus on the acoustic goals of the project, how the model was tested to determine if goals were accomplished, the modifications that were performed, and final results. Impulse responses along with energy ratios were the primary measurement performed. During model testing, the emphasis was to determine how the shape of the room could be optimized within a schematic design scheme developed in concert with the project architect and theatre consultant. Numerous ceiling designs and subtle changes to the shape of the room in plan were tested for their acoustic merit. The evaluation of flat and diffuse surfaces in specific locations was evaluated to determine their influence on achieving the desired impulse response.
Study of parameters linked to sound recording and replaying for virtual acoustics applications
Authors: Warusfel Olivier , Jullien Jean-Pascal
Published: October 2005

The room acoustics laboratory of IRCAM is conducting a research project on virtual acoustics, which intends to control, by electroacoustic means, the different perceptual aspects linked to room acoustic quality. Part of this project deals with the minimization of the perceptual distance between real instruments and the replaying of musical instrument signals through loudspeakers. These applications have pointed out the importance of the effects linked to input and output processes of an electroacoustic chain: the recording and replaying of sound. Aside from the specific processing of each application (e.g., reinforcement, variable acoustics, broadcasting), parameters like the directivity of transducers, their position, and orientation with regard to the instrument or listener will govern the final quality. Optimization of such systems requires better knowledge of how the electroacoustic parameters will interfere with the different aspects of the acoustic quality. This study uses data collected through computer simulation and analyzed with objective criteria whose relevance to the perceptual domain has been validated in previous studies [Jullien et al., Proc. [CA 2, 161–165 and 169–176 (1989)]. Additional measurements have been performed on musical instruments to characterize their directivity.
Acoustic measurement of progressive laser damage in composite material
Authors: Knollman G. C.
Published: October 2005

An acoustic instrumentation system has been developed for continuously monitoring and measuring physical “damage” occurring at a laser-irradiated site of a composite material. With either continuous or pulsed laser radiation of fixed intensity, the thermoelastic response of the material at the beam impact point is recorded during the time of exposure. Simultaneously, sound waves are transmitted through the laser impact region and the modulated signal is measured beyond. Such ultrasonic data reveal the “onset” of internal microstructural damage and provide a temporal history of cumulative damage effects. Also, the spectral composition of the acoustic signals is seen to change as a function of damage evolution. As an example of laser damage assessment, measurement of the progressive depth and volume of pyrolysis at a laser-exposed site on a graphite-epoxy composite will be discussed based on through-transmission acoustic signal-amplitude data.
Detection of thin layers of frost and ice formation on an aircraft
Authors: Wu Junru , Zhu Zhemin
Published: October 2005

The principles of the detection of thin layers of frost and ice formation on an aircraft will be discussed. The theoretical and experimental results of the influence of liquid layers on the propagation of Lamb waves in a plate of finite thickness will be presented. The dispersion equations of Lamb waves in a plate bordered with layers of liquids were derived. Numerical solutions of the equations show the phase velocity of Lamb waves changes with the thickness of the liquid layers. In the vicinity where the thickness of the liquid is equal to a quarter of a wavelength, there is no real solution of the equations. The experiment qualitatively agrees with the theory. [Work supported by Hercules Aerospace Co., Simmons Precision Aircraft System Div.]
Determination of the piezoelectric properties of piezoelectric composite materials by laser Doppler velocimetry
Authors: Rittenmyer Kurt M. , Dubbelday Pieter S.
Published: October 2005

Measurement of the piezoelectric coefficients of a commercial piezoelectric composite material and their temperature dependence was made using a complex electrical admittance-fitting technique. The results were reported previously [J. Acoust. Soc. Am. Suppl. 1 83, S81 (1988)]. In order to extrapolate these measurements to lower frequencies, the time-temperature superposition technique as described by Ferry was successfully applied to direct measurements of the dielectric and elastic constants [Proc. 6th Int. Symp. Appl. Ferroelectr.]. In order to measure the piezoelectric coefficients over an extended frequency range, laser Doppler velocimetry was employed. The technique is shown to be useful for nonresonant determination of piezoelectric coefficients over extended ranges of frequency and temperature. [Work supported by the Office of Naval Research.]
Acoustic methods for determining the acoustical properties of rigid-structure materials
Authors: Hodgson Murray , Woodcock Roland
Published: October 2005

The prediction of the acoustic performance (impedance, absorption, etc.) of “rigid-structure” materials requires a suitable prediction model and a knowledge of its parameters describing the material's acoustical properties (e.g., flow resistivity, porosity). Usually these parameters are determined by nonacoustic methods or are simply estimated; often, they are assumed to be frequency invariant. In this paper new acoustic methods for determining these parameters are proposed. The methods rely on a suitable prediction model and on measurements of the acoustic surface impedance of the materials backed by various impedances. They are first illustrated in the case of fibrous materials and the determination of their flow resistivities. Then, their extension to the general case of any rigid-structure material described by the Zwicker and Kosten four-parameter (flow resistivity, porosity, compressibility, structure factor) model is discussed and illustrated for fibrous materials.
North Pacific gyre-scale reciprocal acoustic transmission experiment
Authors: Worcester Peter F. , Dushaw Brian D. , Howe Bruce M.
Published: October 2005

Three acoustic transceivers moored in a triangle approximately 1000 km on a side in the North Pacific simultaneously transmitted broadband signals to one another for 4 months during summer 1987. All resolvable ray paths were surface reflected; the travel times represent both range and depth averages. Sound-speed profiles were calculated from CTD data taken on the three legs using both the Chen and Millero and Del Grosso sound-speed algorithms. The absolute travel times and measured arrival patterns agree best with predictions made using the Del Grosso algorithm. Sum travel times decreased as the summer thermocline formed during the experiment. Inverting the sum travel times yields range- and depth-averaged sound speeds consistent with those computed from CTD and XBT data obtained during transceiver deployment and recovery. Differential travel items are dominated by tidal curents since the experimental area has both a low-mesoscale energy level and low mean currents. Tidal currents determined acoustically agree well with Schwiderski's tidal model and with barotropic tidal currents determined from a current meter mooring located on the northern leg of the triangle. Inverting the differential travel times gives low-frequency depth- and range-averaged currents of a few mm/s, with an eastward current along the northern leg in agreement with the barotropic current determined from bottom-mounted electric field measurements. The circulation around the triangle corresponds to relative vorticities of about 10−8 s−1, of the same magnitude as expected from Sverdrup dynamics. [Work supported by ONR and NSF.]
A single slice tomography experiment using a long vertical array of receivers
Authors: Duda Timothy F. , Flatté Stanley M. , Worcester Peter F. , Cornuelle Bruce D. , Hildebrand John A. , Hodgkiss William S. , Howe Bruce M. , Mercer James A. , Spindel Rober C.
Published: October 2005

Broadband signals were transmitted from a near-axial source to a long vertical receiving array 1000 km distant in the North Pacific for 9 days in July 1989. The signals were received at 50 hydrophones spaced 60 m apart in the vertical, starting near the surface. Pulses were recorded at 10-min intervals for two periods, totaling 24 h, while hourly pulses were recorded for the duration, giving a total of about 320 pulses. Individual ray arrivals were resolved and their arrival times measured with a precision of a few milliseconds using phase-coded transmissions centered at 250 Hz with a bandwidth of 83 Hz (12-ms resolution). The pulse arrivals as a function of time and depth correspond well with predictions based on geometric (ray) and physical (WKBJ) optics. In each transmission, 600 to 900 ray arrivals can be identified with rays from numerical analysis, distributed along up to 24 distinct wave fronts. The ray arrivals evolve at the time scales of ocean processes. The longest period travel time changes can be inverted to infer changes in the mean sound-speed field between the source and receiver and in high-wave-number ocean features with wavelengths correspondng to ray loop lengths and their harmonics (70 km and shorter wavelengths). Simultaneous CTD and XBT surveys are available for comparison. Temporal and vertically lagged fluctuation statistics of the individual arrivals quantify internal wave and fine-structure effects. [Work supported by ONT.]
The 1989–90 Gulf Steam tomographic experiment at 55°W
Authors: Malanotte-Rizzoli Paola , Chester David , Wunsch Carl , Lynch James
Published: October 2005

Mesoscale variability of the Gulf Stream southern recirculation gyre has been investigated via acoustic tomography. A reciprocal tomographic experiment was carried out in the region just south of the Gulf Stream mean path at 37N, 55W. A pentagonal array of transceivers, with a nominal leg separation of 100 km, transmitted a pseudorandom sequence at a frequency of 400 Hz, with a bandwidth of 100 Hz. The experiment lasted for 10 months from October 1988 through August 1989. Travel-time inversions for temperature, current velocity, and vorticity have been calculated for the array. The oceanography of the region is shown to be largely influenced by local meandering of the GulfStream and its associated Rossby wave radiation field. The estimated time series of area-averaged relative vorticity shows the most important fluctuations to occupy the mesoscale frequency band.
Results of a pilot interlaboratory comparison of REAT measurements of hearing protectors, coordinated by ANSI S12/WG11
Authors: Berger Elliott H. , Royster Julia D. , Casali John G. , Merry Carol J. , Mozo Ben T. , Royster Larry H.
Published: October 2005

ANSI working group S12/WG11 has been tasked with developing laboratory and/or field procedures that yield useful estimates of the real-world attenuation of hearing protection devices. The first goal selected was the development of a laboratory-based real-car attenuation at threshold (REAT) protocol that would estimate the protection that can be, or is being, obtained in the top 10%–20% of today's hearing conservation programs. A protocol was developed based upon ANSI S 12.6-1984, but with modified subject selection, fitting, and training procedures that were described explicitly and in substantially greater detail than in the standard. Pilot testing of two earplugs in four laboratories with ten naive subjects per facility showed subject-fit attenuation results comparable to available real-world studies. Although subjects achieved significantly greater attenuation after the experimenter demonstrated how to insert the devices (informed user fit), the interlaboratory reproducibility of the group data was not significantly improved. The indications are that a subject-fit protocol, in which experimenter involvement is minimized, yields the best estimates of real-world attenuation, and still provides acceptable reproducibility. The pilot results suggested refinements in the design of a full-scale interlaboratory comparison which should begin in the later part of 1990.
Significant threshold shifts and follow-up audiograms in the Army Hearing Conservation Program
Authors: Donahue Amy M. , Resta Eileen K.
Published: October 2005

This paper examines the Department of the Army's (DA) audiometric follow-up program subsequent to identifying standard threshold shift (STS). The Occupational Safety and Health Administration's current policy has provisions for one optional follow-up audiometric test to be completed within 30 days of identifying an STS. Current DA policy requires two follow-up audiometric tests to be conducted within 60 days of identifying the STS. Successes and failures of the DA policy are presented and discussed using 387 000 annual and follow-up audiograms obtained on noise-exposed civilians and soldiers and entered into the Hearing Evaluations Automated Registry System (HEARS) audiometric database. Program participation rates, elapsed time for completion of the follow-up audiograms, and noted changes in STS identified in the follow-up will be presented. The results have implications for hearing conservation policy decisions on follow-up audiometric requirements.
Sound attenuation evaluation of four prototype helmet-integrated ANR systems
Authors: Williams C. E. , Maxwell D. W. , Thomas G. B.
Published: October 2005

Current Navy flight helmets do not provide sufficient passive sound attenuation of the high-intensity, low-frequency noise present in some naval aircraft. As part of a program to provide improved hearing protection for aviators and aircrew personnel, the Naval Aerospace Medical Research Laboratory conducted sound attenuation evaluations of four helmet-integrated active noise reduction (ANR) systems. Sound attenuation measurements (passive attenuation and combined passive/active attenuation) were obtained on Navy and Marine Corps student aviators wearing the prototype helmets in a laboratory noise environment. A Knowles miniature microphone was used to obtain sound levels at the subject's ear. Three sets of measurements were obtained on each subject for each of the prototype systems. A comparison of the combined passive/active attenuation measurements with the passive attenuation measurements reveal increased sound attenuation at 125,250, and 500 Hz and decreased attenuation at 1000 and 2000 Hz. Speech intelligibility measurements obtained on one of the prototypes demonstrated improved speech intelligibility at high sound pressure levels (105 dB and 115 dB) in the ANR “on” mode. [Work supported by the Naval Air Development Center.]
Effect of a resonance of the frame on the surface impedance of a porous materal
Authors: Depollier Claude , Guignouard Philippe , Rebillard Pascal , Allard Jean F.
Published: October 2005

Biot has shown that two compressional waves can propagate in a porous material with an elastic frame. For a material with a heavy frame and a medium flow resistivity, one of these waves propagates mainly in air, and the other is created by frame oscillations. The amplitude of this wave, which is generally negligible, can be important at the λ/4 resonance of the frame. A fast variation of the surface impedance appears at this resonance. This effect, which is predictible with the Biot theory, has been observed on a glass wool of high stiffness.
Nonlinear voltage-dependent capacitance of the outer hair cell (OHC) membrane
Authors: Santos-Sacchi J.
Published: October 2005

Nonlinear, voltage-dependent charge movement within the OHC plasma membrane, presumably associated with the movement of the motility voltage sensor, was studied by ac and step analyses using the whole cell voltage clamp technique. Ionic currents in the cells were blocked. Qualitatively, the two analyses agree; however, there are some discrepancies. The ac analyses indicate a maximum nonlinear capacitance (about 18 pF) at a holding potential near −40 mV, whereas step analyses indicate a maximum capacitance (about 20 pF) at −17-mV step potential from a holding potential of about −120 mV (p/ − 4 procedure). Measures of OHC motility magnitude versus voltage indicate that half-maximal response occurs at about − 40 mV, and follows the ac nonlinear capacitance function. It is possible that step analyses may be adversely affected by the series resistance of the whole cell patch pipette. Gadolinium ion treatment (0.5 to 1 mM) produces some membrane surface charge screening; however, maximum capacitance was reduced as well, and may correspond to the simultaneous reduction or abolition of OHC motility. It is interesting to note that the nonlinear capcitance associated with the movement of the OHC motility voltage sensor will dynamically influence the time constant of the OHC during acoustic stimulation. [Work supported by an RCDA from NIDCD and NIH grant DC00273.]
The effect of tension, stiffness, and air flow on laryngeal resistance in the in vivo canine model
Authors: Natividad Manuel , Berke Gerald S. , Green David C.
Published: October 2005

Recent studies have indicated that glottal resistance varies inversely as a function of flow rate [M. E. Smith and G. S. Berke, J. Acoust. Soc. Am. Suppl. 1 85, S147 (1989)]. This study used an in vivo canine model to determine the effect of air flow on glottal resistance at low, medium, and high levels of recurrent laryngeal nerve (RLN) and superior laryngeal nerve (SLN) stimulation. The data indicate that glottal resistance is linearly and inversely proportional to the log of air flow for low and medium levels of SLN stimulation. At air flow greater than 500 cc/s, the glottal resistances approached 0.1 mm Hg/cc/s for all levels of RLN and SLN stimulation. For low levels of air flow less than 100 cc/s, the glottal resistance was proportional to the degree of RLN and SLN stimulation.
The effect of gas density on glottal vibration and exit jet particle velocity
Authors: Green David C. , Smith Marshall E. , Berke Gerald S. , McGowan Richard S.
Published: October 2005

Although theoretical studies include a term for gas density in their mathematical descriptions of glottal aerodynamics, there are no studies that have measured the effect of gas density on glottal vibration. This study used an in vivo canine model to evaluate the effect of gas density on glottal vibration by comparing phonation with air and helium. Phonation with helium at a constant flow rate resulted in an increased exit jet particle velocity, and an increased open phase, compared to phonation with air. Phonation with helium also resulted in a decrease in maximum glottal width, glottal area per period, frequency, and subglottic pressure when compared to phonation with air. These results were compared to those obtained with air and helium in a four-mass computer model of phonation.
Morphologic changes in laryngeal vibration associated with increasing air flow
Authors: Sercarz Joel A. , Natividad Manuel , Berke Gerald S.
Published: October 2005

Data from the in vivo canine model have indicated that laryngeal resistance is not constant during constant levels of vocal fold stiffness, but varies inversely as a function of air flow. This study was undertaken to determine what morphologic changes in the glottis account for the resistance measurements. Three animals were examined stroboscopically during increasing air flow with constant vocal fold stiffness. Glottal dynamics were measured including photoglottography, electroglottography, and subglottic pressure. Analysis of computerized stroboscopic images indicated that increasing air flow had no effect on the glottal chink (vocal press contact). Increasing air flow was associated with an increase in the amplitude of peak opening and an increase in the glottal area integral.
Regulation of intraoral air pressure during speech
Authors: Moon Jerald B. , Folkins John W. , Luschei Erich S.
Published: October 2005

Speakers with velopharyngeal dysfunction produce higher intraoral air pressure than that obtained by simulating velopharyngeal openings using a passive mechanical model of the vocal tract and a compressed air source. It has been inferred that, in order to maintain air pressures within a regulated range, speakers with velopharyngeal dysfunction compensate for the leakage of air through the velopharyngeal port by altering their respiratory or vocal tract movements. An alternative explanation may be that, instead of depending on an active compensation, the pressure is maintained by the characteristics of the respiratory system. Previous modeling experiments have used an air source with constant flow into the vocal tract. However, it may be that the respiratory system acts more like a constant pressure source than a constant flow source, in which case the intraoral pressures would not be changed by the presence of a velopharyngeal leak. Experiments are reported comparing mechanical models of speech aerodynamics with constant flow and constant pressure inputs. Also reported on are experiments comparing vocal tract aerodynamics measured during speech with data from constant-flow and constant-pressure models.
A glottal waveform model for high-quality speech synthesis
Authors: Tenpaku Seiichi , Hirahara Tatsuya
Published: October 2005

A new glottal waveform model for high-quality speech synthesis is proposed and the results of the perceptual evaluations for synthesized speech using the proposed model and other models are compared. The proposed glottal waveform model consists of two parts: a waveform generator and a spectrum shaping filter. A third-order polynomial, of which coefficients are determined by combinations of OQ (open quotient), SQ (speed quotient), AV (amplitude of voicing), and F0, is used for the waveform generator. A second-order IIR filter, which is designed to control the spectral tilt and the relative amplitudes of lower harmonics components by two parameters, serves as the spectrum shaping filter. Thus the parameters have a direct effect on the waveform and its spectral shape. Using the three kinds of information (F0, power, and formant) extracted from the eight different Japanese words pronounced by a male and a female announcer, 80 synthesized speech stimuli were prepared for the preference test. The stimuli were generated by a cascade formant synthesizer using five different glottal waveform models: the proposed model, Fant's model [STL-QPSR 4, 1–13 (1985)], Fujisaki's model [ICASSP 86, 1605–1608 (1986)], Klatt's model [J. Acoust. Soc. Am. 67, 971–995 (1980)], and Rosenberg's model [J. Acoust. Soc. Am. 49, 583–590 (1971)]. Results of the preference tests with 20 subjects show that the naturalness of the synthesized speech generated by the proposed model is as good as those of the Fant and Fujisaki models.
Toward a voice source characterization of voice quality
Authors: Chasaide Ailbhe Ni , Gobl Christer
Published: October 2005

Voice source characteristics for voice quality variations in an individual nonpathological speaker are presented. This forms part of an ongoing analysis that is being carried out within the framework of Laver's classification [J. Laver, The Phonetic Description of Voice Quality (Cambridge U. P., Cambridge, 1980)] and is based on multiple repetitions by a trained phonetician of an English prose passage and nonsense materials, with modal (neutral), creaky, breathy, whispery, tense, and lax voice qualities. The objective is to provide an acoustic characterization of different “cardinal” qualities in the terms of both time and frequency domain parameters. Time domain measures are based on inverse filtered data to which a model of glottal flow (the LF model) has been matched. Frequency domain measures are based on spectral sections of the speech waveform and of the glottal waveform. As an initial strategy, templates of the different voice qualities are presented, based on the stressed vowel [æ]. However, data are also presented to demonstrate the considerable modulation of source quality even within what would traditionally be considered a single voice quality. For example, in creaky voice, the degree of creakiness is not uniform, but tends to be most extreme in unstressed syllables and at specific locations related to the prosodic profile of the utterance. Thus the differences between qualities may be more or less great depending on the environment compared. [Work supported by ACCOR, Esprit, BRA, No. 3279, and the Swedish Institute.]