bluuebunny/crossref_metadata_2025 huggingface dataset:
Characterization of High-Frequency Echoes by Orthonormal Filtering
Authors: Huff Lloyd C. , Middleton F. H.
Published: September 2005

A five-section orthonormal filter was synthesized, using an analog computer. The transfer function for each of the filter sections was expressible as a ratio of real root exponential polynomials. This orthonormal filter was employed in the characterization of high-frequency echoes arising from a solid aluminum prolate spheroid. Characterization was achieved by the coefficients determined for the least-squared error in a series approximation of weighted orthonormal exponential polynomials. Echo envelopes obtained for several different unique transmit envelope waveforms were processed as a function of target aspect. Data were taken for each one-half degree of target aspect. An explanation of the experimental effort is accompanied by a discussion of the filtering process. A short film demonstrating the effect of target aspect on the high-frequency echoes arising from a prolate spheroidal model target is presented. [Investigation was supported by the U. S. Office of Naval Research.]
Transmission Rates for a Binary Data Acoustic Telemetry System
Authors: Rowlands R. O. , Quinn F. G.
Published: September 2005

A discussion of the error rates in AM, PM, and FM telegraphy as a function of the signal-to-noise ratio is followed by the derivation of an equation relating the optimum acoustic carrier frequency to the distance separating the transmitter from the receiver. It is then shown how the maximum transmission rate over a given distance may be calculated when the maximum permissible error rate is specified.
Observations of Acoustic Waves Generated in a Neon Glow Discharge
Authors: Wilson O. B.
Published: September 2005

Using two calibrated condenser microphones located within a discharge tube, measurements were made of the sounds generated by phenomena associated with moving striations in a dc discharge in neon at pressures of several Torr and at currents ranging from 5 to 100 mA, thus confirming observations reported in 1965 by W. A. Saxton. Acoustic pressures up to several microbars in amplitude existed at the end of the tube during naturally occurring oscillations. Oscillation frequencies from a few hundred to over 1000 were common. Based on phase measurements at different frequencies, it is concluded that the sounds are generated by fluctuations in heating processes at or very near the cathode owing to oscillations in the tube current. It appeared that, in this apparatus, the sounds generated during naturally occurring oscillations are secondary effects and do not appreciably affect the striation phenomenon. External modulation of the electrode potentials increases the sound pressures and appears to affect the striations. Experimental data are presented and discussed. [Work supported by the U. Office of Naval Research.]
Infrasound from the Epicenter of the 1964 Alaskan Earthquake
Authors: Cook Richard K.
Published: September 2005

Data are available on the surface motions at the epicenter caused by the main shock of the earthquake. These show that the permanent vertical displacements were of the order of 1 m, over an area of about 250 000 sq km. The vertical motions occurred in a period of time short as in comparison to the time for sound propagation over the strained area. The earth's surface thus became a source for an impulsive infrasonic wave in the atmosphere. The radiated-sound field has been computed from the known surface motions, and the total radiated energy has been estimated. The results depend substantially on the structure of the atmosphere, whose scale height is much less than the linear extent of the sound source.
Instrumental Method for Improved Intensity-Discrimination Data
Authors: Campbell Richard A. , Lasky Elaine Z.
Published: September 2005

When obtaining auditory intensity-difference limens from human subjects, the limitations typical psychoacoustic laboratory apparatus and measuring instruments can easily combine to yield random intensity errors (±0.1 dB) that are the same order of magnitude as the smaller possible difference limens (0.2–0.3 dB). In the proposed method, the required small stimulus-intensity changes are obtained by combining two waveforms for the variable stimulus with only one waveform constituting the standard. The intensity difference between the two stimuli, or the increment in the variable stimulus owing to the addition of the second signal, can then be calculated from the voltage or sound-pressure ratio in decibels between the two waveforms. As an example of the improvement of precisions: if this ratio is maintained within 0.1 dB, the precision of the resulting increment can be 0.002 dB for an intensity increment of 0.2 dB. Tables and formulas for calculating the increments or decrements resulting from the in or 180° out-of-phase combination of identical waveforms or increments from the combination of independent waveforms are outlined. [Work supported in part by the National Science Foundation.]
External-Ear Replica for Acoustical Testing
Authors: Bauer B. B. , Rosenheck A. , Abbagnaro Louis A.
Published: September 2005

An artificial ear was designed to simulate the mechanical and acoustical properties of the external ear, up to and including the impedance of the eardrum. The sensing element is a B & K 4132 electrostatic microphone terminating a simulated earcanal with an acoustical impedance-matching network that, combined with the microphone, furnishes the eardrum impedance. The canal proper has dimensions approximating those of the real ear and is surrounded by a plastisol pinna of realistic dimensions and texture. The new artificial ear is suitable for testing all types of receivers and ear enclosures under realistic conditions. The inner portion of the artificial ear is made of reproducible metallic components, making it suitable for consideration as an artificial-ear standard. [Work supported by NASA-Manned Spacecraft Center, Houston, Texas.]
Distribution of Reverberant Sound in Large Rooms
Authors: Lubman David
Published: September 2005

If reverberant sound-energy density in a large room is sampled by moving a probe from point to point, the limiting distribution is chi-square of m degrees of freedom and variance is 2/m. For pure tone or very narrow-band excitation, m is 2. For wider bands, m is approximately twice the product of bandwidth by the time for a 10-dB decay. Results seem independent of the number of modes, provided that number is large, and apply to rooms with fixed boundaries and to regions remote from these boundaries. This theory (due to Waterhouse) implies a limitation on the uniformity of reverberant sound that was verified by measurements in a reverberant water tank. However, the limitation was overcome by using a rotating diffuser that, in these experiments, reduced variance by an order of magnitude. The distribution continued to match chi-square but with larger m than given by formulas above. The factor by which m increases is proposed as a measure of effectiveuess of a rotating diffuser. [Work supported by the U. S. Office of Naval Research.]
Experimental Investigation of Feedback in a Sound System—Room Closed Loop
Authors: Connor W. K.
Published: September 2005

A sound system in a room, consisting of single or multiple microphones and loudspeakers connected by an active electrical link, forms a closed loop that may oscillate, or howl, at frequencies for which the magnitude of the loop gain is unity or greater and the phase shift an integral multiple of 2π. This paper gives preliminary results of detailed measurements of 10017 gain and phase, from which howl points can be predicted. The number of howl points versus gain, and the effects of air temperature, room absorption, and transducer configuration on the feedback frequencies are considered. The results have a practical bearing upon techniques of sound-system equalization.
Some Detailed Aspects of Sound Reinforcing Systems in Reverberant Churches
Authors: Boner Charles R.
Published: September 2005

Three case histories of sound-system corrections in reverberant churches by the narrow-band filtering process are given, following an earlier paper on the general subject [J. Acoust. Soc. Am. 37, 131–135 (1965)]. Reverberation characteristics are given for two of these rooms. The discussion emphasizes the two opposing aspects of the problem of high reverberation in the church: (1) the problem of masking of speech in itself, and (2) the question of the corrected environment's acceptability to the listener.
Existence of Transverse Vibration of Thermo-Elastic Plates owing to Heating
Authors: Lyons W. C.
Published: September 2005

A refinement of the dynamic-thermoelastic-plate theory is accomplished by making use of the exact three-dimensional solution for an infinite layer in operational form. From an expanded form of the solution, high-order equations for the deflection of the thermoelastic plate and their associated moment and shear force expressions are derived from the coupled and uncoupled three-dimensional thermoelastic equations. From an analysis of the operators on w(x,y,t), the plate deflection, it is concluded that there exists, in theory, some restriction on the dynamic-thermoelastic-plate theory: namely, (2h)2ω<γk, in which ω is the forcing frequency of an applied heat input, k is the thermal diffusivity of the plate material, 2h is the plate thickness, and γ is a constant that depends on the temperature input conditions on the plate surfaces (i.e., γ=2 for flux-prescribed forcing functions, and γ=6 for temperature-prescribed forcing functions). The dynamic-elastic-plate inequality is (2h)2ω̄2≪c2, in which c2=D(ρ2h)−1, D is the flexural rigidity of the plate, ρ is the density of the plate material, and ω̄ is the forcing frequency of an applied mechanical load.
Coupled-Mode Approach to Elastic-Vibration Analysis
Authors: Eer Nisse Errol P.
Published: September 2005

A variational approximation technique for elastic-vibration problems that may be interpreted as a coupled-mode theory is derived. The theory is valid for the strong coupling occurring in isotropic and anisotropic elastic material owing to Poisson's ratio, shear-shear interaction, and/or shear-extensional interaction effects. The technique permits simultaneous analysis of vibrational modes with different relative phases between the displacement components. As an example, the coupled-mode theory is briefly applied to the two-dimensional extensional vibrations of a thin rectangular plate. Calculations for several of the lower branches of the frequency spectrum compare very closely to both published experimental results and new experimental results obtained with hot-pressed ferroelectric ceramics. The theory predicts the existence of so-called edge modes that have been related to past analyses to the complex branches of the dispersion relation. The present technique calculates the resonant frequencies and displacement distributions for these edge modes, using real propagation constants and a few zero-order approximations for extensional vibrations. The technique produces accurate results, yet physical insight is not obscured by mathematical complexities. [Work supported by the U. S. Atomic Energy Commission.]
Preliminary Results of Study of Effect of Pulse Duration on TTS from Impulse Noise
Authors: Fletcher J. L. , Loeb M.
Published: September 2005

Human subjects were exposed to high-intensity impulses with the peak sound-pressure level (SPL) of about 170 dB held constant and duration of the pulses varied. Growth of temporary threshold shift (TTS) as a function of number and duration of pulses was determined. The frequency point of maximum loss was also noted. Correlations between the numbers of rounds necessary to obtain TTS for the different durations were calculated. Wide variability in response to the pulses was noted and discussed.
Answer to Central Factors in Auditory Fatigue
Authors: Price G. Richard , Oatman Lynn C.
Published: September 2005

Three experiments were conducted to study further the influence of central factors on pure-tone auditory fatigue noted by Wernick and Tobias [J. Acoust. Soc. Am. 35, 1967–1971 (1963)]. Experiment 1 replicated the Wernick and Tobias study and produced similar data. On the basis of observations made during Expt. 1, several changes in procedure were made. Experiments 2 and 3 were concerned with the effects of the changes. The results indicated that if S could resume postexposure threshold tracking without being required to do something else simultaneously (such as write an answer to a problem), the differences between the experimental groups disappeared. The effect interpreted earlier as being the influence of a central factor thus seems to be a procedural artifact. This interpretation is consistent with the procedure and data of previous experiments.
Pitch of Complex Tones
Authors: Plomp R.
Published: September 2005

The question was studied whether the pitch of complex tones is based on the frequency of the fundamental or on the periodicity of the sound as a whole. Pitch-judgment experiments in which the fundamental of a complex tone was shifted to a 10% lower frequency and all other harmonics to a 10% higher frequency demonstrated that, for fundamental frequencies up to about 1400 cps, the pitch changed according to the harmonics, whereas, beyond this frequency, the pitch followed the fundamental. This holds both for tones with harmonics of equal amplitude and tones with harmonics, of which the amplitudes decrease with 6 dB/oct. These results support the view that the pitch of complex tones is based on periodicity rather than on frequency. It is reasonable to suppose that this also holds for simple tones.
Experimental Cepstrum Pitch Detector for Use in a 2400-bit/sec Channel Vocoder
Authors: Kelly James M. , Kennedy Richard N.
Published: September 2005

A vocal-pitch detector for use in a channel vocoder has been designed and implemented at Bell Telephone Laboratories. A vital part of the pitch detector consists of two short-time spectrum analyzers. These are used to produce the spectrum of the log-spectrum, or cepstrum, of input speech. The cepstrum analyzer produces, with high accuracy, rapidly changing pitch information for a wide range of voices. The detector was implemented to verify the theoretical and simulated technique described by Noll [J. Acoust. Soc. Am. 36, 296–302 (1964)]. Noll calculated the short-time spectra using sampled-data Fourier-transform techniques. The implementation employs modified DELTIC time compressors and heterodyne sweep analyzers to determine spectra. The basic functions of the implementation and differences between the simulation and implementation are described. Problems due to sweep analysis of segmented time functions are discussed. The cepstrum pitch detector has been operated as part of an experimental channel vocoder. Taped performance of the vocoder's operation are presented. Among the recordings is a demonstration of a 2400-bit/sec cepstrum pitch-excited channel vocoder.
Time Compression and Expansion of Recorded Speech
Authors: Golden Roger M.
Published: September 2005

Mechanical systems for time compression and expansion of recorded speech generally suffer from noticeable distortion. Distortion occurs because compression or expansion is achieved by discarding or repeating portions of the speech waveform. Two electronic systems that preserve all of the input information and therefore introduce little distortion are the harmonic compressor-harmonic expander (HARCO-HAREX) and the phase vocoder. The HARCO-HAREX separates the harmonics in a speech signal by using a bank of contiguous narrow-band bandpass filters. The signal output from each of the filters is processed appropriately to achieve time compression or time expansion. The design of the original HARCO-HAREX has been improved and simplified since it was first described by Schroeder, Logan, and Presti. The phase vocoder [J. L. Flanagan, D. I. S. Meinhart, R. M. Golden, and M. M. Sondhi, J. Acoust. Soc. Am., 38, 939(A) (1965)] determines the instantaneous amplitudes and phase time derivative in a number of contiguous bands. Compression or expansion is accomplished by scaling the phase time-derivative signals. The phase vocoder allows compression and expansion by non-integer multiples, whereas the HARCO-HAREX was specifically designed to achieve compression and expansion by a factor of two. Both systems have been studied by means of digital-computer simulation.
Shallow-Water Propagation under Downward-Refraction Conditions
Authors: Cole B. F. , Podeszwa E. M.
Published: September 2005

Propagation-loss results obtained in 30-f (fathom) water under downward-refraction conditions are interpreted in terms of ray theory, and are found to depend directly upon the velocity profile and the bottom loss. A bottom loss value of 1.3 dB per bounce was inferred from the measurements, and is shown to compare favorably with theoretical values computed from bottom-impedance information. Observed pulse shapes are then compared with theoretical pulse shapes synthesized from the combined ray and bottom-loss models. In addition, the computed propagation-loss values for the various paths are presented as a function of source angle in order to illustrate the vertical directivity introduced by the environment. Finally, the effect of this vertical directivity on shallow-water propagation measurements is demonstrated by comparing propagation loss results for directional and explosive sources.
Determination of Bottom Slope and Roughness from Travel-Time Differences
Authors: Jacques A. T.
Published: September 2005

A method for determining bottom slope together with a measure of bottom roughness, based only on measurements of time differences of arrival between various bottoms and surface-reflected paths, is presented. The analytical approach consists of expressing various appropriate time differences by means of binomial expansions in order to obtain precision by manageable working equations. The method has the advantage of yielding bottom-slope characteristics that result from sound energy arriving at oblique angles of incidence. Thus, the reflection process is that associated with an effective bottom reflecting area (Fresnel-zone configuration) comparable to that which exists for many practical applications. A sample calculation based on experimental data is included and results are compared with other means of determining bottom features.
Spherical Radiation
Authors: Molloy C. T.
Published: September 2005

This paper presents formulas that permit the calculation of the radial velocity at the surface of a spherical transducer such that a prescribed pressure distribution will be achieved over the surface of an arbitrary sphere. Both unsymmetrical and symmetrical pressure distributions are considered. Also the corresponding pressure and particle-velocity functions are given that describe the acoustic field throughout all space exterior to the transducer surface. Special symmetrical pressure distributions described by Tchebycheff polynomials are discussed. Formulas are presented that permit the independent specification of beam width and lobe suppression for this class of pressure distributions. Formulas for their directivity index are also given. Computation of some specific cases is presently being performed on an IBM-7094. These numerical results are discussed. [Work supported by U. S. Navy Ordnance Systems Command.]
Investigation of Microphone Nonlinearities, Using the High-Intensity Pistonphone
Authors: Dahlke H. E. , Kantarges George T.
Published: September 2005

Increased emphasis toward the development of new microphones for measurements at high acoustic pressure occurring in sonic-boom phenomena, flight-vehicle noise, and shock-wave studies has necessitated continued development of devices and methods to measure and calibrate these transducers. Since each method has its limitations with regard to pressure, frequency, and accuracy, it is necessary to apply several calibration methods for evaluation and calibration of pressure transducers. From the great number of calibration devices available, the pistonphone is the apparatus that delivers an absolute calibration of high accuracy in the pressure range up to 185 dB (re 10−4 dyn/cm2). Its application at high pressure and frequency is limited owing to nonlinearity characteristics of the gas, which leads to a distortion of the waveform applied. A high-intensity pistonphone has been developed; its theoretical aspects are described and compared with experimental results. It operates in the range from 5 to 200 cps at sound-pressure levels from 134 to 185 dB. A number of measurements showing nonlinearity, hysteresis effects, and repeatability of sensitivity are described.
Acoustics of High-Speed Turbine-Driven Trains
Authors: Mull Harold R.
Published: September 2005

A revolution in intercity transportation will take place with the introduction of high-speed turbine-driven trains. The speed and efficiency resulting from the application of aerospace technology to rail transportation can bring the intercity passenger back to the railroad. However, he must experience a noise and vibration environment equal to that of other means of transportation. Important noise sources are rail joints, turbine inlet and exhaust, and boundary-layer noise. Estimates of these noise sources are presented, along with acoustic control measures. Criteria for external and internal noise are presented.
Time-Varying Energy-Detection Model for Discrimination and Detection
Authors: Henning G. Bruce
Published: September 2005

One frequency-encountered difficulty with contemporary analytic models of hearing is their prediction of errorless discrimination performance in conditions in which human observers perform poorly. In this paper, a modified energy-detection model is discussed. The model, consisting of a Gaussian filter (the center frequency of which is a time-varying random variable) followed by a squarer and an integrator, predicts some of the common features of both amplitude- and frequency-discrimination and of detection data. [Research conducted primarily at The University of Pennsylvania and was supported, in part, by the Public Health Service, U. S. Department of Health, Education, and Welfare.]
Discrimination of Direction of Auditory Pulsed Doppler
Authors: Pollack Irwin
Published: September 2005

Trains of pulses were presented to the earphones of 17 trained listeners. In a given trial, two trains of pulses were presented: in one, the interval between pulses successively increased; in the other, the interval between pulses successively decreased about a constant center interval. The task of the listener was to identify the decreasing train. A wide range of conditions (center interpulse intervals from 0.17 to 45 msec; number of intervals from 3 to 255) yielded a wide range of detectable threshold slopes (from about 0.5 to 10 000 μsec per interpulse interval). Abrupt changes in Doppler sensitivity are observed in the vicinity of 60 pps for pulse trains with a small number of intervals, possibly suggesting a transition from rhythm to pitch-discrimination. The center frequency for minimum threshold slope systematically increased with a larger number of intervals. The result supports the proposition that a greater number of time markers are required for accurate frequency resolution at the higher frequencies. The minimal threshold time change from the first interpulse interval to the center interval is about 8 μsec in the vicinity of 1000 pps. [Research supported in part by the National Sciences Foundation.]
Two-Point Thresholds in the Ear
Authors: Weiss David J.
Published: September 2005

Subjects were asked to discriminate whether one tone or two tones, closely spaced in frequency, had been presented. When two tones were presented, their total intensity equalled that of the single tone. The tones were presented in noise; performance was evaluated in terms of the ratio of signal energy to noise-power density at which the subjects were able to distinguish one tone from two 75% of the time. Center frequencies of 500, 1000, and 4000 cps and durations of 25, 100, and 500 msec were employed; frequency separation was varied for each condition. The number of beats in the observation interval proved to be the important variable. Minimum frequency separations for which the discrimination was possible were 40 cps for the 25-msec conditions, 5 cps for the 100-msec conditions, and 1 cps for the 500-msec conditions. Center frequency had no effect on these minimum-frequency separations. A model based upon an energy detector and simple integrator is proposed. [Research supported in part by the National Institutes of Health, Public Health Service, U. S. Department of Health, Education, and Welfare.]
Graphical Language for the Scores of Computer-Generated Sounds
Authors: Mathews M. , Rosler Lawrence
Published: September 2005

Conventional musical scores are an insufficient and inconvenient way of describing sound sequences to computers. A procedure is described for drawing scores as graphical functions of time, using a light pen on a cathode-ray tube attached to a small computer. The information is transmited digitally to a larger computer, which synthesizes the sound and reproduces it immediately with a loudspeaker. Typically, functions for amplitude, frequency, and duration of a sequence of notes are drawn. An algebra allows combining functions by addition and multiplication. In this way, certain compositional processes may be performed by the computer. For example, the time-varying weighted average between two melodic or rhythmic sequences may be synthesized. The graphical programs provide great flexibility for drawing, copying, erasing, and altering functions. Thus it is easy to develop a sound sequence by a succession of trials. Microfilm and punched-card versions of the score are automatically provided. In addition to being compositional tools, the graphical scores are effective representations of the sound to a listener. In many ways, they are easier to follow than conventional scores.
Experiments on the Musical-Interval Sense
Authors: Pikler A. G.
Published: September 2005

Mersenne (1636), Huygens (1661), Mercator (1667?), Sauveur (1701), and Chladni (1802) estimated by introspection the ear's tolerance for mistuned musical intervals in terms of comma fractions. Delezenne (1826), using a monochord and methods of adjustment and forced choice, was the first to study the ear's accuracy on the unison, octave, natural third, fifth, and sixth, expressing the error in terms of comma fractions. Other, isolated investigations were made by Weber (1829; monochord), Scheibler (1834; forks), Seebeck (1846; monochord and fork), and Cornu and Mercadier (1869; organ pipe and the autograph). Preyer (1876) employed a specially designed reed harmonium, a method of forced choice, and an index of sensitivity. Schischmanov (1889) used forks with Wundt's method of minimal changes. Stumpf and Meyer (1898) used adjustable screw forks and Fechner's technique of forced choice. Seashore, Jenner, and Miles (1905–1914) studied vocally produced intervals with the tonoscope. Sokolovsky (1911) used motion pictures of these sung tones. Moran and Pratt (1926) used a tone-variator, 12 intervals, and the method of adjustment. Psychoacousticians in the electronic age have largely negelected the psychophysics of the interval sense. Data have been reported by Greene (1937), Bolt (1947), Nickerson (1949),Ward (1954), Corso (1954), and Lichte (1955). Pikler (1960) mapped the interval sense over the five central octaves (61 tones), using the method of adjustment, oscillators, and a Stroboconn. The equal temperament of 12 tones was reproduced with standard errors ranging from 1.2 to 38.7 cent. Intra- and inter-octave analysis did not show significant differences. Sample data taken on the major third confirmed that the syntonic comma is supraliminal, whereas enharmonic differentiation proved only better than chance. Pikler and Harris (1961) employed forced-choice and tape-recorded mistuned octaves.
Piezoelectric Analysis of the Monolithic Crystal Filter
Authors: Beaver W. D.
Published: September 2005

The “monolithic crystal filter” consists of a quartz-crystal plate with an array of two or more electrode pairs placed on its major surfaces and with each pair establishing a resonator system. The resonators are acoustically coupled via the elastic quartz plate. A theoretical analysis of such a piezoelectric system is presented. Specific calculations to determine the coupled resonant frequencies are made for the two and three coupled resonator filters. Such calculations consist of the solution of the differential equations of the piezoelectric system with the appropriate boundary conditions. The frequency equations are derived giving the natural resonances of the device, which are also the zeros of the short-circuit input impedance of the filter. The extension of this analysis technique to a general filter of n acoustically coupled resonators is presented.
Microphonic Response from the Inner Ear of the Bullfrog
Authors: Capranica Robert R. , Flock Åke , Frishkopf Lawrence S.
Published: September 2005

The labyrinth of the bullfrog contains two auditory sense oragns, the basilar and the amphibian papillae. The microphonic response of this system was recorded with a silver/silver-chloride wire electrode placed in the perilymph above the common crus or within the crus itself. Pure-tone stimuli were presented and the spectral content of the electrical response from the labyrinth was measured with a narrow-band wave analyzer. This analysis has shown—First, the response is largest over the range of frequency sensitivity of the amphibian papilla and drops sharply in the range of the basilar papilla; therefore, the response derives primarily from the amphibian papilla. Second, both fundamental and second-harmonic components are prominent in the response, even at moderate intensities; this result was predicted on anatomical grounds, based on the complex orientation of hair cells in the amphibian papilla. Third, both fundamental and second-harmonic components in the response to one tone can be reduced in amplitude by the addition of a second tone of appropriate frequency; the range of frequencies and relative stimulus magnitudes over which this effect occurs suggests that the suppression of the microphonic response from the amphibian papilla and the well-known tone-on-tone inhibition of primary neurons derived from that organ have a common genesis.
Auditory Responses in the Medulla of the Bullfrog: Comparison with Eighth-Nerve Responses
Authors: Frishkopf Lawrence S. , Capranica Robert R.
Published: September 2005

Responses to acoustic stimuli of neural units in the medullae of DIAL-anesthetized bullfrogs (Rana catesbeiana) have been recorded with metal microelectrodes. In a number of ways, these units appear to preserve the peripherally encoded information conveyed in the auditory portion of the eighth nerve. As in the periphery, two classes of units can be identified on the basis of frequency sensitivity, a low-frequency group that can be inhibited and a high-frequency group that cannot. The range of frequency sensitivity of these units for excitation and inhibition is the same as for the corresponding primary units. In other ways, however, the neurons of the medulla transform the eighth-nerve input. Weak facilitation of the response of a high-frequency unit by a low-frequency stimulus has been found in several units. Many low-frequency units exhibit very rapid adaptation at their frequencies of maximum sensitivity, responding only to the onset of a tonal stimulus. The maximum repetition rate at which such NO units can respond synchronously to pulses is much lower than for the corresponding peripheral units. The ON response has been found only among low-frequency units. High-frequency units respond, with adaptation, throughout a stimulus and show synchronous responses to pulses at rates as high as, or higher than, their peripheral counterparts.
Flow-Noise Measurements in Water
Authors: Franz Gerald J.
Published: September 2005

Available measurements of flow-induced noise and closely related turbulent-boundary-layer pressure fluctuations in water are reviewed and nominal-spectral-density and cross-spectral-density data for the fluctuating pressures are presented in dimensionless form. Included are data from pipe-flow facilities, gravity-propelled and towed bodies, and self-propelled vehicles. The effects of size, shape, smoothness, type, and location of hydrophones; curvature, roughness, fairness, thickness, homogeneity, and impedance of domes; Reynolds number of the flow; and various active boundary-layer control schemes are examined. Experimental difficulties encountered in water measurements are enumerated and compared to similar difficulties encountered in air. Finally, needed future experiments on large gravity-propelled models and on full-scale submarines and surface ships are described.
Mechanical-Equipment Capabilities for Environmental Control in an Acoustics Laboratory
Authors: Ramer L. G. , Harold R. G.
Published: September 2005

The Trane Acoustics Laboratory was designed to provide for a variety of environments for thermal loading of air-conditioning equipment. This permits showing the effect of load on sound generation. This was accomplished by providing a number of services to the anechoic and reverberant rooms. Among these were piped water, compressed air, gas, and steam, with valves terminated for ready flexible connections to the room interiors. More significant was the ducting of heated or cooled air through the test rooms in sufficient quantity to provide 180 000 Btu/h of heating or cooling but without introducing noise to interfere with acoustical measurements. Details of these services are shown. The control of relative humidity and its effect on acoustical measurement are discussed.
Sound Absorption and Velocity in Sulfur Dioxide/Argon Mixtures
Authors: Shields F. Douglas
Published: September 2005

Sound-absorption and -velocity measurements have been made in pure SO2 and mixtures of 25% and 50% Ar with SO2 at room temperature, 400° and 500°K. The results confirm earlier reports of a multiple relaxation process. The data have been fit with both series and parallel relaxation curves and the temperature dependence of the individual relaxation times determined. The difficulty in distinguishing between the two processes is discussed.
Origin and Characteristics of Sound Generation in Compressors and Fans
Authors: Powell Alan
Published: September 2005

The physical significance of the basic sound sources—the monopole, dipole, and quadrupole—are described in relation to their occurrence in compressors and fans, and to their consequent radiation characteristics. This makes clear the principles by which the various types of source may (sometimes) be identified and controlled.
Vocal Pitch and Intensity Regulation at the Glottis
Authors: Koyama T. , Kawasaki M. , Ogura J. H.
Published: September 2005

Voice production and regulation at the glottis requires a coordination of a number of factors. These include glottal air flow, subglottic air pressure, vocal-cord muscle contraction, acoustic coupling of the cavities above and below the glottis, etc. In order to explore this complex mechanism, a series of experiments were performed on anesthetized dogs producing phonation artificially by adducting and tensing the vocal cords via electrical stimulation of laryngeal motor nerves and controlled air blast to the subglottic area. Objective measurements of sound pressure, sound pitch, air flow, subglottic air pressure, and electrical stimuli were made simultaneously. Precise interrelationship among these parameters and efficiency for voice production induced from the parameters will be discussed. In particular, significance of the cricothyroid muscle on pitch and intensity regulation will be stressed.
Color Speech
Authors: Beninghof William J.
Published: September 2005

There exists a functional analogy between the perception of speech and that of color. Loudness, spectral mean, and spectral spread of speech are analogous to the brightness, hue, and saturation in the color environment. It is believed that sensory replacement based upon perceptual analysis is preferable to other methods based upon the nature of the physical stimulus. Equipment that maps a sequence of phonemes in a connected utterance into the corresponding sequence of color strips has been constructed. The color strips are displayed on the screen of a TV set. The equipment includes storage that permits the pattern corresponding to the most recent 4 sec of the utterance to be displayed at each instant. Theoretical justification for the analogy and the effectiveness of this particular implementation is discussed. [This research was supported by a NASA Fellowship and the Electrical Engineering Department at Northeastern University.]
Ultrasonic Velocity at High Pressures in a Solution of Urea
Authors: Barrett R. E.
Published: September 2005

The ultrasonic velocity was measured in an 8m aqueous solution of urea at pressures up to 2000 lb/in.2 and at temperatures between 28° and −5°C. Within this temperature range, the velocity-vs-temperature curve is nearly linear, having a slope of about 0.182 m/sec⋅atm (not varying with temperature). Water at high temperatures has about the same value of (∂c/∂p)T. When considered along with ultrasonic-absorption data for urea solutions, as previously reported [cf. Bull. Am. Phys. Soc. 13, 222 (1968)], this suggests that urea preferentially breaks the open-packed state in water. In particular, for an 8m solution the clusters of the open-packed state seem to be nearly completely broken up for temperatures down to −5°C. However, the existence at high concentrations of an absorption in excess of the viscous absorption indicates that there are still two states present. [The high-pressure measurements were performed at the Physics Department of Brown University. This work was supported in part by the U. S. Department of Health, Education, and Welfare.]
Determination of Intensity Discrimination and Detection Capabilities by Absolute Identification Procedures
Authors: Farrell R. M. , Pynn C. T. , Braida L. D.
Published: September 2005

Results of classical absolute-identification experiments on signal intensity indicate that when the number of stimuli is large, subjects make more identification errors than would be expected on the basis of the pairwise discriminability of the stimuli. These experiments, however, all employed interstimulus intensity increments that are large relative to discrimination ability. Recent work suggests that identification performance should be directly related to discrimination performance when the increments are small. Experiments using small increments have been conducted with tone pulses (discrimination) and with tone pulses masked by noise (detection). The results not only confirm the expected relationship (operating characteristics and sensitivity measures obtained with identification are the same as those obtained with conventional procedures), but indicate that the identification paradigm is very efficient for evaluating detection and discrimination capabilities. [Work supported principally by the National Institutes of Health and in part by the Joint Services Electronics Program.]
Instruction and Application of Voiceprint Identification to Law Enforcement
Authors: Kersta L. G.
Published: September 2005

The experience and success of Voiceprint Identification as applied to law-enforcement applications and the training of law-enforcement officers in the applied art are the dual purpose of this paper. The general method of application of Voiceprint Identification to law enforcement and the preparation of the exhibits for criminal trial presentation will be described. Examples drawn from the over 170 cases processed for over 35 law-enforcement agencies will be presented including conclusions for both probative and trial applications. A description of the Voiceprint Identification training course for qualified law-enforcement personnel and their success scores will be given. The two-week in-residence training has so far resulted in identification success scores exceeding 97% for this relatively short training period. Officers who were qualified fingerprint experts achieved success scores notably better than those without fingerprint-identification background.
Simulated Vocoder Analysis
Authors: Rothauser Ernst , Maiwald Dietrich
Published: September 2005

An all-digital spectrum-analyzer technique for a pitch-excited channel vocoder evaluated by computer simulation is presented. The “fast Fourier transform” algorithm is utilized as the basis for estimating the power-density spectrum of fixed segments of sampled input speech. This approach is a convenient way to investigate system requirements for a high-quality vocoder-analyzer without the usual restrictions due to hardware and real-time analysis. The design of such an analyzer system depends extensively on the statistical properties of the input signal, i.e., speech. The chosen approach allows processing in both time and frequency domains with possibilities far exceeding the capabilities of hardware systems. The well-known difficulty in designing a speech-analysis system by exact mathematical considerations alone has been overcome in some critical points by using the experience gained from processing actual speech data. Several system parameters were experimentally adjusted, based on evidence from printouts of intermediate numerical results and also by listening to speech signals synthesized from computed speech-analysis results. The proposed approach will be discussed, and analysis results obtained with our experimental system will be demonstrated.
Reciprocity Calibration of Altec 21BR Series Microphones
Authors: Leinbach Henry J.
Published: September 2005

In calibrating Altec 21BR series microphones, much trouble was experienced with repeatability of results at frequencies above 2000 Hz. Also, it was desired to have an absolute basis for results above 200 Hz. Hence reciprocity apparatus for calibration of Altec 21BR series microphones was designed and built. Measurements were made that showed that this method is workable with 21BR series microphones with precision on the order of that obtainable with W.E. 640AA or Brüel & Kjær 4131 microphones.
Acoustical Facilities for Loudspeaker Evaluation
Authors: Allen Harry E.
Published: September 2005

The most important facility for loudspeaker design is the anechoic chamber. The usual practice is to “build in” baffle arrangements in the chamber for measuring direct radiators. The refraction and reflection characteristics of these baffles vary from chamber to chamber making it difficult to compare measurements between facilities. In the facilities to be described, baffled loudspeaker tests are performed in a separate chamber making it possible to perform free-field measurements in a truly anechoic room. The baffled tests are performed in a chamber with five treated surfaces, the loudspeaker being mounted in the untreated wall producing an environment very close to the “infinite-baffle” ideal standard conditions. Other facilities include a 7440-ft3 reverberation chamber, terminated acoustic transmission lines, accelerated life-test chamber, and subjective evaluation laboratory.
New NRL Acoustic Research Tank Facility
Authors: Chervenak J. , Hanish S.
Published: September 2005

A new underwater acoustic research facility for precision mapping of the acoustic fields of model arrays of sonar transducers has been installed at the Naval Research Laboratory. The facility consists of a 30-ft diam and 22-ft deep cypress wood tank with associated pulsed electronic equipment for testing underwater sound transducers operating over a 3–100-kHz range. A vital component of unique construction is a mechanical scanner in which a tiny hydrophone is programmed to automatically scan the radiating face of a model transducer. The output of the hydrophone is transmitted to a radiation distribution printer, converted to numerical decibel, values and automatically typed as a numerical pattern. An acoustic evaluation of the tank is presented together with samples of three-dimensional acoustic pressure field plots for a model array of free-flooded ceramic ring transducers operating at a frequency of 26 kHz.
Sound Transmission in Variable-Area Ducts
Authors: Morfey C. L.
Published: September 2005

The effects of (1) density gradients and (2) compressible fluid flow are introduced into the classical problem of one-dimensional sound transmission along a variable-area pipe. An admittance equation is developed, along the lines of Mawardi's horn theory [J. Acoust. Soc. Am. 21, 323 (1949)], which relates the acoustic properties of the pipe at different sections. Once the admittance is known, the acoustic-energy equation provides the remaining information needed to solve the transmission problem. In each case, the admittance obeys an equation of the type found for horns by Mawardi, suggesting the concept of an equivalent horn. The effects of Mach number in Case (2) are discussed and compared with the low-frequency analysis of Powell [J. Acoust. Soc. Am. 31, 1527–1535 (1959)].
Lithium Niobate and Lithium Tantalate Piezoelectric Resonators in the Medium-Frequency Range with Low Ratios of Capacitance
Authors: Hannon J. J. , Lloyd P. , Smith R. T.
Published: September 2005

Lithium niobate and lithium tantalate crystal units in the 400–600-kHz range, operating in the length extensional mode have been designed and fabricated. Ratios of capacitance as low as 12, and Q's of 70 000 to 80 000 have been attained. The electrical characteristics of these units have been measured using a new method suitable for high-coupling materials. This paper describes the resonator geometry and the method used to determine the electrical properties. The frequency-vs-temperature characteristics and the unwanted mode spectra are also included.
Individual Differences in Speech-Quality Perception
Authors: McGee Victor E.
Published: September 2005

Consider one voice, bandlimited in K different ways. A complete paired-comparison schedule is presented to N listeners to obtain “similarity-with-respect-to-quality” judgments. From each listener's judgments, a “perceptual space” is determined and the question is: “Are there individual differences between these N perceptual spaces?” In order to answer this, we compare perceptual spaces, obtain some index of similarity, and determine a “points-of-view” space containing N points. Suppose this is a two-space. Typically we now argue for differences on the basis of how the N points spread out on a sheet of graph paper, but a very real question is rarely considered. How big can the points-of-view space be? By introducing “random listeners” (random perceptual spaces), we can answer this question; and the results so far indicate that “real listeners” cluster so tightly in this “real-plus-random” space that no individual differences can be found for the “real listeners.” [This research was supported by the Office of Naval Research.]
Pitch of Short Tone Bursts of Changing Frequency
Authors: Nabelek I. , Nabelek A. , Hirsh I. J.
Published: September 2005

Tone bursts with linearly changing frequency (upward or downward glides) were matched in pitch by listeners who adjusted the frequency of tone bursts with fixed frequency and the same duration. Stimuli included tone bursts of 12, 40, and 120 msec, with frequency changes of 50, 200, and 580 Hz around a geometric mean of 715 Hz. In those bursts where the frequency changed throughout 100% of the duration, when the frequency change was small, pitch was matched by a steady frequency near the geometric mean with very little dispersion. For large frequency changes, pitch was matched by a frequency between the mean and the terminal frequency but with large dispersion. In those bursts where the frequency changed during only 25% of the burst duration, such transient appeared at several delays from the start of the burst. When the bursts were short and the frequency changes small, pitch was matched by frequencies within the interval of change. When the bursts were short or long and the frequency changes large, however, pitch was matched by either the initial or the terminal frequency, depending on when the transient occurred. [Supported by a grant from NINDB, National Institutes of Health.]