bluuebunny/crossref_metadata_2025 huggingface dataset:
A neural-counting model based on physiological characteristics of the peripheral auditory system: Application to loudness estimation and intensity discrimination
Authors: Lachs G. , Al-Shaikh R. , Bi Q. , Teich M. C.
Published: January 2006

We have previously used an energy-based neural-counting model, incorporating spread of excitation, receptor saturation, spontaneous neural activity, and refractoriness in the primary auditory fibers, to predict the outcome of a number of neurophysiological and psychoacoustical experiments [M. C. Teich, et al., J. Acoust. Soc. Am. Suppl. 1 71, S18 (1982); G. Lachs et al., IEEE Trans. Syst., Man. Cybern. SMC-13, No. 5 (1983)]. We wish also to incorporate the frequency characteristics of the middle-ear transmission function, and the cochlear mapping function. This places a number of constraints on the allowed parameters of the theory. We discuss the parameter values required for our theoretical system to effectively predict the outcome of loudness-estimation and intensity-discrimination experiments, both for pure-tone and variable-bandwidth noise stimuli. [Work supported by NSF.]
The relation between loudness growth and intensity discrimination
Authors: Schlauch Robert S. , Wier Craig C. , Huckins Elizabeth
Published: January 2006

The notion that the intensity-difference limen can be used as an indirect test of loudness recruitment is an old idea. However, to date, a firm quantitative relation between the two measures has not been established. We consider the problem as one in which loudness is related to the overall stimulus magnitude and intensity discrimination is a reflection of the accuracy with which a loudness judgment can be made. Since for pure tones loudness and intensity discrimination covary as functions of intensity, for a specific intensity one quantity can be inferred from the other. To test this relation, adaptive ABLB loudness-growth functions were obtained for a 2000-Hz, 500-ms pure tone from a normally hearing listener (partially masked monaurally) and a listener with a moderate, unilateral sensorineural hearing loss. Intensity-discrimination performance was measured for the same 2000-Hz tone, at several levels, in both ears of each subject. Intensity-recruitment functions derived from these data are very similar to the obtained loudness-recruitment functions. [Work supported by DRF and NINCDS.]
The application of stepwise coupled modes to a refracting ocean using Galerkin's method
Authors: Evans Richard B.
Published: January 2006

The method of coupled modes developed [R. B. Evans, J. Acoust. Soc. Am. 74, 188–195 (1983)] for a homogeneous water column with stepwise depth variations of a penetrable attenuating bottom is applied to a realistic ocean environment with general sound-speed profiles. The main complication associated with this application is caused by having to find the complex eigenvalues for the range independent sections and not by the range dependence. To handle this problem, the Galerkin method is employed with an appropriately chosen orthonormal basis set which replaces the search for the complex eigenvalues with a more tractable matrix eigenvalue problem. Numerical results based on this application are presented. [Work supported by NORDA.]
Shallow water waveguide characterization using Hankel transforms
Authors: Lynch James F. , Frisk George V.
Published: January 2006

A coherent processing technique is proposed for the characterization of shallow water channels. For a horizontally stratified ocean and bottom, the method consists of measuring the magnitude and phase versus range of the pressure due to a cw point source and Hankel transforming this data to obtain the depth-dependent Green's function versus horizontal wave-number. The Green's function contains all the information about the channel necessary to solve the forward problem, including the nature of the discrete and continuous spectra and the plane-wave reflection coefficient of the bottom. Characteristics of the Green's function can also be used to infer acoustic properties of the bottom. Results for simple examples are presented using synthetic data and implications for the nonstratified case are discussed.
Wave reflection from a sediment layer with depth-dependent properties
Authors: Stern M. , Bedford A. , Millwater H. R.
Published: January 2006

The prediction of reflection coefficients for the water-sediment interface is strongly influenced by the physical nature of the sediment (a viscoelastic porous material saturated by a liquid) and the depth dependence of the mechanical properties [E. L. Hamilton, J. Acoust. Soc. Am. 68, 1313–1340 (1980)]. An efficient computational scheme for calculating reflection coefficients has been developed for a sediment layer using the Biot equations with depth-dependent coefficients. The qualitative effects of the inhomogeneous sediment properties (in comparison with uniform properties) are studied numerically. [Work supported by the Office of Naval Research.]
Acoustic reflection from ocean abyssal plains
Authors: Freese Herbert , Besieris Ioannis , Kohler Werner
Published: January 2006

Existing bottom loss and pulse response data, measured at a variety of grazing angles, frequencies, and locations (Sohm and Hatteras abyssal plains, Blake Plateau, Florida shelf, and Gulf of Mexico), will be reviewed. A comparison of data with existing theoretical models will be made.
A WKB-numerical integration method for computing bottom loss
Authors: Foreman Terry , Pitre Richard
Published: January 2006

Computation of bottom loss at many frequencies and grazing angles is a computationally intensive process. At high frequencies and large grazing angles, the numerical integration of the wave equation can be very slow. When shear wave generation is an important process, the problem is exacerbated because low shear wave speeds imply short wavelengths and therefore small integration step sizes. The WKB approximation method is very rapid and accurate at short wavelengths, but is inaccurate near turning points. The hybrid integration method described here combines both methods by dividing the integration domain into WKB and numerical integration (turning point) regions. Due to the frequency dependent size of the turning point region, the accuracy and computational effort of this hybrid method are only weakly dependent on frequency. [Work supported by Naval Ocean Research and Development Activity.]
Optimum frequency of acoustic propagation in shallow water
Authors: Gershfeld D. A. , Eller A. I.
Published: January 2006

The effect of various basic physical and environmental parameters on the optimum frequency of propagation in the ocean is investigated through the use of a model that is computationally simple to evaluate. A closed form expression for predicting transmission loss (for the Pekeris case) is obtained by applying the Poisson sum formula to transform a slowly converging transmission loss expression (consisting of a sum over the normal modes) into a rapidly converging series where only a few terms are needed to predict reasonable loss levels. A comparison of spectrum levels between the derived loss expression and a sophisticated normal mode transmission loss model shows close agreement for the frequencies of interest. The results show that the optimum frequency is strongly dependent upon the respective depths and separation distance of the source-receiver pair. The model is used to explore the dependence of the optimal frequency on several parameters. [Work supported by ONR Code 425UA and NAVSEA Code 63R.]
Highway identification of exhaust system noise problems
Authors: Kessler F. M. , DiPolvere E. , Van Ouwerkerk M.
Published: January 2006

To reduce the number of excessively noisy vehicles on the road, a simplified noise control enforcement procedure was developed and tested. This procedure utilizes the “trained” ear of enforcement officials in detecting faulty, modified, or absent mufflers and exhaust systems. The purpose of this study was to evaluate whether or not the trained ear can be utilized as a screening device for “probable cause” to stop a vehicle for inspection of the exhaust system resulting in the citation of the driver for a mechanically defective exhaust system. Two teams conducted these tests. The first team included the “spotting officer” and a person who recorded the pass-by sound level of all vehicles. The second team consisted of an inspection officer and a person making stationary sound level measurements. A majority of the vehicles called loud by the “spotting” officer had pass-by sound levels 5–10 dB below the pass-by standard for that class of vehicle. This indicates that the quality of the sound emitted by such vehicles is distinctive and that the officers are able to detect defective exhausts prior to the defects causing the excessive noise to exceed the pass-by standard. Numerous additional findings are discussed.
The effects of community noise on modular and manufactured/mobile homes
Authors: Hawley Mones E. , Back Geoffrey M.
Published: January 2006

Dwellings which are built in factories and transported to their sites account for an increasingly important fraction of new housing construction in the United States. About half of these houses are placed in clusters with other factory-built homes. The interior sources of noise and the insulation against noise from exterior sources were investigated and compared with those for new site-built houses. Manufactured houses (formerly called mobile homes), which have an integral chassis for transportation, must be built in conformance with federal standards. Modular homes, which are removable from their transporters, are built to state and local building codes. The result is that factory-built houses have internal noise levels and external-to-internal noise level reductions equivalent to those of comparable site-built houses. However, exterior community noise levels frequently are higher for factory-built homes than for site-built homes because of the way the sites are chosen. [Work supported in part by the U.S. Environmental Protection Agency under contract number 68-01-6159.]
An overground excess sound attenuation (ESA) study
Authors: Powell Robert G.
Published: January 2006

In order to better predict community noise from Air Force operations an extensive study on overground excess sound attenuation (all attenuation in excess of inverse square and air absorption losses) was conducted at Wright-Patterson Air Force Base. The unique aspects of this study were the methods used to acquire the reference noise levels, procedures used to obtain up and downwind propagated noise, and the all-weather, radio-controlled microphone recording systems. To be presented are the experimental design and samples of ESA values calculated from the noise of 400 plus ground runups of a KC-135A aircraft propagating along a 2400-m microphone array. Also to be discussed are the availability and format of the basic acoustic and meteorological data from this study.
Analysis of overground excess attenuation
Authors: Bishop Dwight E.
Published: January 2006

The noise and meteorological data acquired by the Air Force AMRL at Wright-Patterson Air Force Base has been analyzed to determine the factors most influencing the excess sound attenuation at different frequencies and distances. Multiple regression analyses of the excess sound attenuation (the difference in noise levels measured at different positions referred to the reference position levels after adjustment for inverse square and air absorption losses) indicate that the strongest variables influencing the excess sound attenuation are the wind component in the direction of sound propagation, the temperature gradient, and the ground cover (grass versus snow). Regression coefficients ranged up to the order of r2 = 0.8. Plots of the variance accounted for (r2) values as a function of frequency show two distinct peaks which shift downwards in frequency as distance increases. At low frequencies (below about 200 Hz), the most significant variable is ground cover. For the higher frequencies the most significant variable is the wind component. Curves showing the measured excess attenuation values as a function of wind component and temperature gradient will be compared to results from other studies and theoretical models.
Sound radiation from a concave or a convex dome in a semi-infinite tube
Authors: Suzuki Hideo
Published: January 2006

A previous study [H. Suzuki and J. Tichy, J. Acoust. Soc. Am. 69, 41–49 (1981)] showed that a concave or a convex dome in an infinite baffle has radiation characteristics that are different from those of the flat piston. These results indicated that a concave or a convex dome in a lossless tube may also have different radiation characteristics than a similarly mounted flat piston. This problem was investigated using the same method as the previous studies. The results show that the farfield sound pressure response rises by about 0.5 to 0.7 dB in the frequency range ka = 2.2 to 2.8 for both concave and convex domes having the height-to-radius (a) ratio of 0.5 to 1.0. As expected, the radiation resistance increases by about 13% to 17% over the radiation resistance in the low-frequency region. The radiation reactance is very small at low frequencies but becomes comparable to the resistance at frequencies above ka = 2.5. The reactance is always negative below the cutoff frequency.
Corona characteristics of transducer ceramics
Authors: Timme R. W. , Browder L. P. , Montgomery R. E.
Published: January 2006

Transducers are frequently plagued by the presence of corona which can adversely affect the performance and lifetime of the unit. Sites at which corona can occur include the wiring, spacing, configuration, sharp electrode edges, etc., but these can be minimized with proper design and engineering. A fundamental limit below which one cannot reduce the appearance of corona lies with the transduction material, that is, the piezoelectric ceramic itself. The corona characteristics of over 2500 lead zirconate-lead titanate ceramic rings have been determined and the results used to model a much larger ring population from which predictions can be made about the corona performance of transducers assembled from these rings. A summary of the information to be presented is that: (1) Populations of “as-fired” ceramic rings have significantly higher corona inception voltages than rings with surfaces finished by machine grinding. (2) Two distinctively different types of corona have been identified, one of which can be partially “cured” by sulfur hexafluoride. (3) The corona data for the rings have been fit to a Weibull distribution model and applied to a “weak-link” model to predict the best performance that can be expected from the assembled transducer. [Work sponsored by the Sonar Transducer Reliability Improvement Program, NAVSEA.]
Field experience with sound and vibration measurements using an FFT analyzer
Authors: Brittain Frank H.
Published: January 2006

The theory upon which FFT analyzers are based promises a very powerful and useful measurement tool. In the field, a variety of practical difficulties often affect the ability to obtain valid results using an FFT analyzer. These difficulties fall into five categories: (1) inadequate knowledge on the part of the user, (2) details of the analyzer as implemented by the manufacturers, (3) bugs in the hardware and software of the analyzer, (4) conditions under which measurements are made (including the individual characteristics of the equipment tested and the background vibration), and (5) ability of the user to interpret data from the analyzer. Each of these five categories is discussed, with emphasis on two-channel measurement applications. Practical examples of field problems encountered and methods used to overcome them are discussed.
Acoustic impedance measurement via FFT
Authors: Richarz Werner G.
Published: January 2006

The measurement of acoustic impedance of acoustic materials and networks often requires a rather laborious impedance tube measurement. Other methods have been evolved, but these usually rely on two microphone measurements. The single microphone system reported here utilizes the incident and reflected pulse in a circular impedance tube. The pulse is generated by a suitably damped loudspeaker driven by a low-frequency square wave. A two-channel Fast Fourier Transform analyzer captures the pressure-time history. The signal is then conditioned leaving only replicates of the incident and the reflected pulses suitably shifted in time to simulate conditions at the sample surface. Using the various features of modern FFT analyzers, the reflection (absorption) coefficient as well as the complex valued impedance are readily available from a single measurement. At present the usable frequency range is 5 kHz. [Work supported by the Natural Science and Engineering Council, Canada.]
Optimizing pulse amplitudes in multipulse excitation
Authors: Singhal Sharad
Published: January 2006

Multipulse excitation [B. S. Atal and J. R. Remde, Proc. ICASSP 82] is an attractive way of modeling the input to the LPC synthesis filter at low bit rates. In the method described by Atal and Remde, the amplitudes and locations of the pulses are determined in successive stages—one pulse at a time. At each stage, one pulse is determined to achieve maximum reduction in the weighted mean-squared error between the original and synthetic speech waveforms. For closely spaced pulses, successive optimization of individual pulses is inaccurate, and often requires additional pulses to compensate for inaccuracies introduced earlier. This problem can be only partly corrected by reoptimizing all pulse amplitudes at the end. This paper describes a method which locates s pulse at any stage by jointly optimizing its amplitude together with the amplitudes of all pulses located up to that stage. As a result the amplitudes of all pulses are kept optimal at every stage, and very little inaccuracy is introduced even if the pulses are closely spaced. The new method provides significant improvements in the SNR—by as much as 10 dB in some sections of speech.
Violin sound synthesis from first principles
Authors: Weinreich Gabriel
Published: January 2006

A simple computer model of a violin has been constructed. It consists of two arrays carrying the forward and backward waves on the string, appropriate reflection rules at the bridge and finger, a bow of finite width, and a combination of parallel two-pole filters to simulate the radiativity of the violin body. The string is assumed to be nondispersive, although an equivalent amount of dispersion can be included in one or both of the boundary conditions. The instantaneous force exerted by the bow on the string is formulated as a single-valued function of their relative velocity. The output signal can be converted into sound. The model has been used to study a number of phenomena, including the dynamic changes in the Helmholtz motion when the bow velocity changes, and in particular the starting transients.
Characterizing guitar sound using time-evolving spectra
Authors: Palmer E. Paul , Zuvich Linda
Published: January 2006

The sound of a guitar or similar instrument can be characterized by the time evolution of its partials and its noiselike sound components. This includes the establishment or generation of noise, families of partials, and formants and resonances by the nature of the plucking and the characteristics of the instrument. It particularly includes the decay and interchange of energy among all the sound components. The investigation of these components is facilitated by using linear and logarithmic-frequency spectral analysis and by using variable-length time windows in acquiring data to be analyzed. Displays of time-evolving spectra demonstrate the different characteristics of classes of guitars such as classical and steel string and the differences between members of each class. The relationships among measurable sound parameters and guitar quality and guitar design and construction will be discussed.
Compatibility of ships and whales in Glacier Bay, Alaska
Authors: Johnson James H.
Published: January 2006

Glacier Bay, southeast Alaska, normally a summer feeding ground for humpback whales, was abruptly abandoned in midsummer 1978, prompting charges by environmentalists that increased vessel traffic had driven them out. Subsequent research focused on whale-prey relationships, vessel effects on whales, and the overall acoustic environment of Glacier Bay compared to nearby areas where whales continue to feed in undiminished numbers. Prey studies showed a high correlation between whale distribution and prey abundance; in 1982 a group of whales chose to spend most of the summer feeding in the busiest, noisiest part of the Bay. Behavior studies showed whales are affected by vessels up to 3-km away, but reactions were of short duration, and meaning in terms of possible physiological stress and cumulative effect is unknown. Acoustic studies revealed minor differences between the Bay and comparison areas (e.g., “white sound” from effervescence of melting glacier ice, and sporadic seismic noise), which presumably are longstanding phenomena that do not explain the recent reduction of whales. Humpbacks are gradually reoccupying the Bay, possibly the result of current traffic regulations and restrictions on numbers of vessels allowed entrance. However, a “crash” of whale prey stocks in 1978 still cannot be discounted. [Work supported by National Park Service, National Marine Fisheries Service, and Alaska cruise ship industry.]
Behavioral responses of wild beluga whales (Delphinapterus leucas) to noise from oil drilling
Authors: Awbrey Frank T. , Stewart Brent S.
Published: January 2006

Two seasons of field observations and playback experiments with southwest Alaskan beluga whales has established that wild whales in their natural habitat respond more negatively to sudden changes in sound level than to sustained sounds. Recordings of noise from SEDCO 708 drilling platform were projected, underwater at 163 dB re: μPa source level. Whales less than 1.5 km from the source usually reacted to onset of the noise by swimming away. In two instances, groups of whales that were at least 3.5 km from the noise source when playback started continued to approach. One group approached within 300 m before all or part turned back. The other group submerged and passed within 15 m of the projector. Results were consistent with observations of captive belugas.
Preliminary measurements of sound attenuation by snow over a model seal lair
Authors: Cummings William C. , Holliday D. V.
Published: January 2006

Because of their winter habit of denning in snow above Arctic ice, ringed seals, Phoca hispida, may be exposed to airborne man-made noise. Incidental to acoustical studies of winter seismic exploration [D. V. Holliday et al., J. Acoust. Soc. Am. Suppl. 1, this meeting] a series of measurements was undertaken of airborne tones received in a model seal lair. Five successive layers of snow blocks were added around the sensor. Seven tones from 105 to 10 510 Hz were played from a loudspeaker at two distances, 0.8 and 11.5 m, through the accumulative five layers (maximum thickness, 0.9 m). This study was undertaken off Prudhoe Bay, Alaska, April 1983). The results indicated an increasing transmission loss through the “lair” wall of about 6 dB/doubled thickness, an increasing dependence with frequency (Hz), and evidence of significant propagation through the ground path. It is tentatively concluded that, in this situation, snow may be an effective barrier for higher frequency airborne noise, but attenuation is limited by flanking propagation through the underlying medium. [Work sponsored by NOAAs National Ocean Survey, Office of Oceanography and Marine Services.]
Source levels of bowhead whale sounds determined by acoustical array localization
Authors: Cummings William C. , Holliday D. V.
Published: January 2006

One effect of man-made underwater noise is possible masking or other interference with sounds of whales requiring knowledge of the source level. In conjunction with a technical feasibility study of acoustical location in northern Alaska [W. C. Cummings, Tracor Doc. No. T-83-06-002-U (1983)] numerous whale sound source levels were determined based on accurate distances using a three-hydrophone array baseline of 2.4 km. In terms of dB re: 1μPa at 1 m (peak spectrum level over the duration of each signal), 24 localized song elements ranged from 158 to 189, median 176.5 dB. The 33 localized moans ranged from 129 to 170, median of 158 dB. At a distance of 10 km, this implies a theoretical received signal-to-noise ratio for songs of 36 dB, based upon the observed median ambient level at 500 Hz, or 16 dB at maximum noise level. However, based upon our experience with the prevailing shallow water propagation losses and noise characteristics, the S/N ratio of songs can be expected to approach 0 at about 15 km, with moans, 5–10 km. [Work sponsored by the North Slope Borough.]
The absorption of sound by anisotropic porous layers
Authors: Burke Shawn
Published: January 2006

An analytical model which describes the effect of a directional anisotropy of acoustical properties in porous sound absorbers is presented. The oblique incidence impedance of an anisotropic porous layer is derived in terms of the material's flow resistance, effective mass, and phase velocity measured in two directions. The impedance boundary condition is shown to be strongly dependent upon the absorber's transverse complex velocity, as well as on the angle of incidence. Values of the impedance and oblique incidence absorption coefficient for a layer of Fiberglas are calculated using experimental anisotropic parameter data. The random incidence sound absorption coefficient is then calculated for three boundary condition models of the Fiberglas, over frequencies ranging from 300–1200 Hz. The anisotropic model predicts absorption coefficients that agree better with reverberation chamber data than either an isotropic bulk reacting model or a locally reacting model. The absorption coefficient is shown to be insensitive to weak anisotropies.
Propagation of intense sound in fibrous bulk porous materials
Authors: Nelson D. A. , Blackstock D. T. , Perreira N. D.
Published: January 2006

The work reported here is an extension of that by Kuntz [NASA CR 167979, September 1982; see also J. Acoust. Soc. Am. Suppl. 1 69, S80 (1981)], who investigated the nonlinear behavior of intense sound propagating through air-filled bulk porous materials. The primary purpose of the current work is to explain some of Kuntz's experimental observations. Our model is based on the assumption that the nonlinear flow resistivity of the material is far more important than the ordinary hydrodynamic nonlinearities that affect an open fluid. It is further assumed that the nonlinear resistance term is proportional to either u2 sgn(u) or u3, where u is particle velocity. Both forms lead to harmonic distortion of the “cubic” type. In this case the distortion components of an initially sinusoidal wave are exclusively odd harmonics. This and other predictions are supported by Kuntz's data. The model is thus verified. [Work supported by NASA and ONR.]
Data averaging and FFT devices
Authors: Holger David K.
Published: January 2006

FFT devices often include a data smoothing capability that is based upon averaging over a selected number of adjacent points in a data array. Averaging time domain data in this way is equivalent to low-pass filtering the data. The resulting inaccuracies associated with frequency domain representations of the averaged data are examined. Predicted and experimental results are presented for the FFT's of averaged sinusoidal, periodic, and filtered random waveforms. The effects of sample interval and number of averaging points on the equivalent filter characteristics of the averaging process are discussed.
Acoustics of teleconferencing
Authors: Stockbridge Chris
Published: January 2006

Room acoustics is an element of teleconferencing too often neglected. Ideally, the room should not be symmetrical and the walls should be nonparallel, covered irregularly with sound scattering and noise absorbing panels. At the very least, a place for teleconferencing must be quiet and not reverberant. Technical specification of various sizes of desirable acoustical spaces which match both people's behavior and equipment constraints will be set forth in this talk. Equipment and techniques will be discussed which select acoustics for a natural feel within the room while generating signals that create clear and recognizable voice sounds when heard at remote location(s). These include phased array directional microphones, carefully positioned loudspeakers, simple acoustic absorption, and for the more sophisticated, nonparallel walls and irregular ceiling clouds. The real challenge is to implement this technology in an aesthetically pleasing fashion, tailored and matched to the culture of the expected user population. A variety of teleconferencing rooms will be presented in photographs, showing evolution in the design of rooms for audio, audiographic, and full motion video teleconferencing.
The Quorum TM teleconferencing microphone
Authors: Snyder James H.
Published: January 2006

Arrays of microphones have long been used in underwater acoustics, but only recently has it been appreciated that the directivity of microphone arrays makes them superior to conventional microphones in many audio applications as well. In this paper we describe a particular microphone array, the Quorum TM teleconferencing microphone, whose directivity is achieved by the nonuniform spacing of the transducers. We present equations for the array sensitivity assuming plane and spherical wave radiation, and compare these results with measurements.
RTSKED, a real-time scheduled language for controlling a music synthesizer
Authors: Mathews M. V.
Published: January 2006

H. F. Olson helped create the RCA synthesizer. This machine was the first to use sophisticated digital control. Control data were punched into a wide roll of paper tape using a binary code. New technology has superseded paper tape, but control remains the principal problem for modern synthesizers. Precise control of the timing of events in music is difficult for stored program computers. A control language is described in which the computer executes a schedule. A schedule differs from a normal program in that commands and the times at which these commands are executed are separately specified, thus making a clean separation between what the computer does and when it is done. Times can be specified in absolute terms (wait so many milliseconds) or be specified in relative terms (wait until a performer presses a particular key). The language is intended to control real-time performance. It allows the flexible combination of information from a score in the computer memory with information generated by the performer playing on a keyboard or on other sensors. Early experiments indicate that RTSKED is easy to learn, pleasant to use, and powerful.
Theory, ingenuity, and wishful wizardry in loudspeaker design—A half-century of progress?
Authors: Augspurger George L.
Published: January 2006

During the past 50 years the course of professional loudspeaker design has been well-documented. The history of high-fidelity consumer products is less familiar and more erratic. Loudspeaker designers seem to have embodied three archetypes in varying proportions: the elegant theoretician, the inspired tinkerer, and the wishful wizard. A brief examination of some of their enduring successes, ephemeral favorites and resounding failures is presented.
Effect of the sound field structure on the measurement of acoustic intensity
Authors: Elko G. W. , Tichy Jiri
Published: January 2006

The effect of instrumentation phase and magnitude error on the measurement of the active and reactive acoustic intensity for different sound fields will be discussed. We will show experimental results of measurements made for the active and reactive intensity in some semireverberant fields. An easy method to calculate the effects of phase bias will be presented and demonstrated for the experimentally measured data. The measured results will be compared to theoretically predicted values and the measuring system will be described.
Acquisition and analysis of architectural acoustic data using a microcomputer
Authors: Palmer E. Paul , Price Rodney
Published: January 2006

A microcomputer is easily adapted to the task of acquiring up to 32-k samples of 8-bit digitized acoustic data at rates up to 20 k bytes/s. This makes it suitable for acquiring and analyzing architectural data such as impulse response, reverberation time, speech intelligibility, etc. We will compare computer-simulated and experimentally obtained impulse-response data for a simple rectangular room. Comparisons will be made for data acquired using an integrated-signal spark sound source, a pseudorandom-sequence, white-noise source, and a swept-sine source. Difficulties met in using a pseudorandom sequence of length different from a power-of-two will be discussed. Some limitations met in using low-resolution, 8-bit data will be indicated. System variations and improvements will be noted.
Perception of sine-wave analogs to individual formants of CV syllables
Authors: Szczesiul Rosemary , Pastore Richard E. , Rosenblum Lawrence
Published: January 2006

The immediate goal of our research was to delineate the nature of the perceptual consequences of the types of acoustic changes typical of phoneme contrasts. In this study we used sine-wave analogs of individual formant components of CV syllables. All of our sine-wave stimuli were dynamic in that they were characterized by various changes in frequency as a function of time which mimicked characteristic formant transitions. Naive listeners performed both labeling and discrimination tasks in response to individual formant equivalents. Preliminary results suggest some similarity between the perception of speech and sine-wave analogs to formant components both in the demonstration of categorical perception and the location of category boundaries. In general, the relatively distinct category boundaries seldom corresponded to those which would be predicted by simple inspection of visual representations of the physical stimulus characteristics. [Work supported by NSF grant BNS 8003704 to the second author.]
Effect of voice type and message format on speed and accuracy of response to voice warnings
Authors: Anderson Timothy R. , Freedman Jay , Rumbaugh Andy
Published: January 2006

The effect of LPC voice type and of message format on the speed and accuracy of response to voice warning signals were investigated at various levels of a simulated military inflight noise environment. LPC voice types were male, female, and machine, and the message formats were voice warning, repeat voice warning, and tone-voice warning. The primary task was a manual tracking task that required continuous attention whereas the secondary task required individual manual responses to randomly presented warning messages. A background signal of inflight aircraft voice communications was presented via the subject's earphones throughout the experiment. The voice warning messages were also presented via the earphones at levels of 0, 5, and 10 dB above this background communication level. Ten subjects completed the experimental tasks while wearing standard Air Force flight helmets and oxygen masks in the presence of the emulated aircraft noise at levels of 105 and 115 dB (SPL). The effects of LPC voice type and message format on voice warning effectiveness will be presented and discussed.
Power transmission across a complex plate branch
Authors: Haberman Robert C. , Pan Donald T. , Graham Thomas S.
Published: January 2006

Finite element and classical techniques were combined to compute the transmission and reflection of vibratory energy across a plate branch with structural connections too complex to be modeled with classical techniques alone. The procedure for combining the models is outlined and numerical results are presented. Results are given for a simple plate connection (which agrees with classical theory) and several complex welded and riveted plate connections. For complex connections, results are shown to be very sensitive to structural details such as mass and contact area at the branch. Since contact area is in general unknown, this suggests that power transfer coefficients should be averaged over frequency bands. The method described may be used to compute coupling coefficients for SEA models.
Target detection: Beluga whale and bottlenose dolphin echolocation abilities compared
Authors: Turl Charles W. , Penner Ralph H.
Published: January 2006

An experiment compared the echolocation abilities of a beluga whale and a bottlenose dolphin to detect five targets of the same size and target strength ( + / − 2 dB) at distances from 40 to 120 m. The performance for both animals at 40, 60, 80, and 100 m exceeded 90% correct detection; however, performance at 120 m ranged between 60% to 62% correct detection. The beluga whale and the bottlenose dolphin were tested with the same targets at 5-m increments at distances from 100 to 120 m. These test results indicated: (1) no significantly different performance (p < 0.05) between the beluga whale and bottlenose dolphin at the five test distances, (2) both animals differentially reported three of the five targets, independent of target distance, and (3) the performance of both animals at 120 m ranged between 60% to 90% correct detection depending on which target was presented at 120 m. There may be several reasons why both animals differentially reported three of the targets at ranges beyond 100 m, irrespective of absolute distance: (1) variable target strength, (2) the manner in which the targets were suspended plus rotational motion about the suspension point may have induced target echo variability, and (3) for targets under 100 m, target echo levels were large and both the beluga whale and bottlenose dolphin were able to differentiate the signal from the background noise. Beyond 100 m, subtle features of the targets determined their detectibility. On two targets, the salient features were apparently unavailable at ranges beyond 100 m, and differential detection occurred. Because the range differences were small, the significant target feature that was absent was not solely a function of spherical spreading loss.
Bottlenose dolphin (Tursiops truncatus): Difference in the pattern of interpulse intervals
Authors: Penner Ralph H. , Turl Charles W.
Published: January 2006

When the echolocation detection abilities of a beluga whale and a bottlenose dolphin were tested on identical cylindrical targets and distances, they had interpulse interval (IPI) distributions which were different, while detection accuracy showed no significant difference. Bottlenose dolphin pulse trains show a predictable and systematic relationship between the distance to a detected target and time between echolocation pulses, thus permitting an inference of attending distance [R. H. Penner and J. Kadane, J. Acoust. Soc. Am. Suppl. I 68, S97 (1980)]. The beluga whale pulse trains contained, at all target distances, a first IPI component starting around 30 ms and increasing to 60 ms and a second IPI component between 200 to 220 ms which became more numerous as target distance increased. The transition zone between the first and second components contained very few interval counts. The interpulse intervals within the transition zone increased as target distance increased. If the bottlenose dolphin IPI patterns are accepted as models, and similar assumptions regarding attending distance are applied to the beluga whale pulse train, the picture is one of always scanning out to 30 m; followed by short scan out to 160 m. Subsequent target detection work using spheres instead of cylinders as targets, showed the same IPI distribution patterns for the beluga whale.
A qualitative model of the dolphin sonar
Authors: Fitzgerald James W.
Published: January 2006

By means of a complex, functional anatomy, the bioacoustic apparatus of the dolphin enables him to operate four separate sonar systems more or less simultaneously, using some of the anatomical components in common. These systems are: (1) a broadband high-frequency (∼30–150 kHz) echolocation “attack” sonar, with a narrow (∼10°) forward-looking beam pattern, that is used for the location and tracking of such targets as food-fish; (2) a broadband, low-frequency (<30 kHz) echolocation “search” sonar, with a broad (∼120°) forward-looking beam pattern, that is used for orientation and intruder detection; (3) an audio-frequency (∼0.2–15 kHz), frequency-contoured “communication” sonar, with a substantially nondirectional beam pattern, that is used for communicating with other dolphins; and, finally, (4) a wideband (0.15–150 kHz), binaural “listening” sonar that is used as the sonar receiver for all three of the preceding active sonars, as well as a passive listening system for target detection, bearing; and classification. It is shown that echolocation “clicks” originating in the larynx, which enter the melon through the closed nasal plug, are totally internally reflected, radiating only from the forward end, and form the attack sonar. A logarithmic spiral of the form R = A exp[(−tan βc)θ], where βc is the critical angle, fits the shape of the melon. For the open nasal plug condition, echolocation “clicks” from the larynx propagate up the nasal duct into the premaxillary sac, which acts as a “bubble” transducer for the search sonar. The source of the whistle-like communicative phonations is identified as lip-modulations excited by exhaust air expelled from the premaxillary sac into the vestibular sac. Experimental confirmation, drawn in part from the bioacoustic literature on Odontocetes, is proffered in support of the principal qualitative features of this new model.
A processing system for acoustic reciprocal transmission experiments
Authors: Nguyen H. , DeFerrari H. A.
Published: January 2006

Broadband signals such as linear maximal pseudorandom sequences (LMPRS) find many applications in acoustic remote sensing in the ocean. We have designed a system which both transmits and receives LMPRS entirely under microprocessor control. Timing of the experiments, generations of the transmitted signals, and coherence sampling of the received signal are accomplished with a low-power-consuming RCA 1802 processor. Parameters of the experiment can be set by software making possible a variety of experiments without hardware modifications. The system is battery powered and is designed to operate for periods of months at sea. Three systems have been constructed for acoustic reciprocal transmission experiments in the Florida Straits and are now being tested. The system will be described along with preliminary results from experiments. [Work supported by ONR and NOAA.]
Dichotic listening to musical sequences: Relationship to hemispheric specialization of function
Authors: Deutsch Diana
Published: January 2006

It is generally assumed that patterns of ear advantage in dichotic listening reflect greater involvement of the hemisphere contralateral to the preferred ear. This assumption is critically reviewed with respect to musical materials. A set of experiments is described which demonstrate the presence of a left-right anisotropy in the perception of frequency combinations. When two frequencies are simultaneously presented in a sequential setting, perception of these frequencies is superior when the higher is to the right and the lower to the left than when the higher is to the left and the lower to the right. Localization is also more accurate when the higher frequency is to the right and the lower to the left. These phenomena are shown to give rise to apparent patterns of ear advantage, whose direction may vary depending on such factors as overall error rate and listening strategy. Such apparent ear advantages should not be taken as reflecting greater involvement of the contralateral hemisphere. Implications for patterns of ear advantage reported for speech materials are also discussed. [Work supported by NIMH.]
Three-dimensional directional sensitivity of goldfish 8th nerve fibers
Authors: Fay Richard , Coombs Sheryl , Baumann Joseph
Published: January 2006

Three-dimensional maps of directional sensitivity were constructed for single fibers of the saccular, lagenar, and utricular nerves by measuring acceleration thresholds to sinusoidal motion oriented along 40 axes in three orthogonal planes. Stimuli were generated by adding three mutually perpendicular vibration inputs to a rigid cylinder containing water (and the fish). The phases and amplitudes of the inputs were computed to achieve linear motion vectors with particular orientations, as determined by an accelerometer array. Phase-locking was used to define a cell's acceleration sensitivity, and the polarity of the response (the excitatory direction of motion). In general, the 3-D directional sensitivity of cells (in dB) is described by a solid composed of two tangent spheroids, with an axis of greatest sensitivity passing through their centers. In the saccule, these axes are oriented primarily dorsal-ventrally with deviations from vertical of up to 40°. Lagenar fibers show similar patterns but with greater variation of orientation in the mid-saggital plane. Utricular units show greater variation in all planes. Any given axis of particle motion is represented in the profile of response across arrays of fibers of different orientation, and is best defined by fibers oriented with a null plane parallel to the axis of motion. [Supported by the NSF.]
Whole-nerve response to tone bursts in the alligator lizard
Authors: Turner Robert G. , Shepard Neil T. , Nielsen Donald W.
Published: January 2006

The peripheral auditory system of the alligator lizard (Gerrhonotus multicarinatus) has been studied extensively; however, limited data are available for the whole-nerve (AP) response. Could the AP serve as a convenient monitor of the condition of the ear during experimentation? AP responses to tone burst acoustic stimuli of various frequencies were recorded from the alligator lizard and three measures using the AP response were obtained. (1) Averaged suprathreshold AP responses were related to primary auditory nerve fiber activity. (2) AP thresholds were compared to single-unit thresholds. (3) AP excitatory tuning curves were compared to single-unit tuning curves. The good agreement between AP responses and single-unit data indicate that the AP is a useful tool for measuring the physiology of the peripheral auditory system in the alligator lizard.
Behavior of the pulse-number distribution for the neural spike train in the cat's auditory nerve
Authors: Teich Malvin Carl , Khanna Shyam M.
Published: January 2006

Pulse-number distributions (PNDs) were recorded from primary afferent fibers in the auditory nerve of the cat, using standard microelectrode techniques. Pure-tone and broadband-noise stimuli were used. The number of neural spikes (pulses) n was measured in a set of contiguous intervals, each of duration T seconds. The quantity n varies from one interval to another. These data were then used to determine the PND, which is the probability p(n,T) of occurrence of n spikes in the time T, versus the number n. The estimated mean and variance of p(n,T) were obtained. Two different values of T were used. We observe that the count mean-to-variance ratio R is relatively constant and independent of the stimulus intensity. The PND readily exhibits the existence of spike pairs in the underlying point process for some units. A study of the scaled and unscaled pulse-interval distributions (PIDs), under conditions of spontaneous firing, demonstrates that the occurrences of neural events are sometimes not describable by a renewal process. [Work supported by NIH Grant 2-R01-NS 03654.]