bluuebunny/crossref_metadata_2025 huggingface dataset:
Adaptive Receiver for Synchronous Detection with Finite Memory
Authors: Spooner Ronald L.
Published: September 2005

The detection of synchronous signals in noise in the presence of unknown parameters implies the possibility of adaptation in an attempt to learn the true parameter values. The adaptive process depends on the availability of past observations and results in an improvement in detection performance during future detection intervals. For the synchronous case, however, the optimum-likelihood-ratio receiver design results in the impractical requirement of a receiver that possesses a time-growing memory. The problem considered here is the design of a limited-memory receiver with adaptive characteristics for the detection of synchronous signals of unknown amplitude. The class of receivers considered perform the operations of cross correlation and ideal limiting. The optimum-likelihood receiver is chosen from this class, and its adaptive characteristics are exhibited. The performance of this locally optimum receiver is compared with the performance of the optimum full-memory receiver and with the performance of a locally optimum receiver derived for the class that use no adaptation.
Statistical Model Relating Underwater Acoustic Phase to Environmental Parameters
Authors: Yacoub Kamal
Published: September 2005

As a result of recent experiments in the Straits of Florida, the dependence of underwater-sound propagation parameters on environmental parameters has been pointed out [J. Steinberg, J. Acoust. Soc. Am. 41, 1617(A) (1967) and Clark et al., Proc. U. S. Navy Symp. Military Oceanogr., 4th (May 1967)]. In this paper, these relationships are evaluated through the analysis of the cross-spectra and coherence functions. A multi-input statistical model is suggested and defined through its transfer-function matrix, with water-temperature, tide-height, and wind components serving as its inputs while two phases measured at hydrophones placed at distances of 3 and 43 NM serve as its outputs. The analysis of this model is based on measurements of these parameters conducted for a period of 5 mo, and in most cases, without interruption except for six days during Hurricane Inez in October 1966. [Work supported by a contract from the Office of Naval Research.]
Averaging to Reduce Variance in Bearing Estimates
Authors: Schalow R. D.
Published: September 2005

When computing θ = arctan (Y/X) from k independent samples of each of the random variables X and Y, the variance of θ, σθ2, may be reduced by averaging in either of two ways: (1) averaging X and Y individually, then computing θ from the resulting averages; (2) averaging the individual values of θ obtained from each pair of values of X and Y. Theoretically, if X and Y are normally distributed and minimum σθ2 is desired, Method (1) is best if S/N ratio >0 dB and method (2) is best if S/N ratio <−5 dB. For other values of S/N ratio, the choice depends upon k. Practical considerations cause .Method (1) to be preferred in virtually all situations.
Model for Predicting the Effects of Normalization on Sonar Performance
Authors: Wilkinson John , Dow Jerry
Published: September 2005

Curves giving (S/N)R, the required input (S/N), for a specified detection probability versus probability of noise marking are commonly used for describing processor performance. These curves are commonly produced by assuming stationary background noise. However, backgrounds are generally nonstationary, so that normalization techniques are utilized to produce a more stationary processor-output waveform. In order to predict the effects of various normalization techniques on sonar performance, a model has been developed that defines ideal normalization and gives a method for describing the effects of normalization in terms of (S/N)R for fixed clutter probability. This method involves measurement of degradation in (S/N)R due to nonideal normalization. The results are given in the form of two curves and an average degradation over the display. One curve gives a plot of Δ(S/N)R as a function of time where Δ(S/N)R is the increase in (S/N)R due to nonideal normalization relative to (S/N)R required with ideal normalization. The other curve gives a plot of the percentage of the display in which Δ(S/N)R is greater than a specified Δ(S/N)R. The average degradation is the average of Δ(S/N)R over the display.
Measured and Computed Convergence-Zone Transmission Anomalies
Authors: Cole B. F. , Einstein L. T. , Weinberg H. , Podeszwa E. M.
Published: September 2005

Since the intensity of the sound path, which is horizontal at both the acoustic source and receiver, cannot be determined by ray acoustics, this path is of particular interest in attempting to describe the nature of convergence-zone sound fields. The present paper describes some specific 3.5-kHz convergence-zone propagation-loss results for this path and contrasts these results with other data obtained when this path was not present. Finally, both sets of measurements are compared with theoretical loss values obtained by combining the sea-water attenuation loss with the spreading loss given by ray theory, and the relative differences presented.
Prediction Model for the Oceanographic Influences on Bottom-Reflected Sound
Authors: Davidson R. O.
Published: September 2005

A prediction system is developed for computing geometrical spreading loss. The system is applicable to all underwater-sound transmissions generated by nonvertexing rays. The prediction system has the advantage of being able to: (1) generate the complete spectrum of values for geometrical spreading loss in a large oceanographic area over an annual time period; (2) index the predicted contours with two indices; (3) provide a maximum error bound on the difference between a predicted value and the value that would be obtained by a detailed sound-ray calculation; and (4) use the computed errors to delineate the oceanographic area in which the predictions are valid. The prediction system, or model, is used to obtain all contours of geometrical spreading loss that can be associated with the North American Basin. The errors in these predictions are generally less than ±1 dB for ranges out to 55 kyd.
Noise Generation by Fluid Flow in Pipes
Authors: Gordon Colin G. , Sawley Roger J.
Published: September 2005

Many industrial processes involve high-velocity and high-pressure fluid flow in piping systems. Such systems can be a source of significant process noise. In this paper, the major sources of noise associated with straight pipes, pipe turns, valves, and other flow discontinuities are identified. Procedures are developed whereby the noise radiation into the surrounding environment may be estimated. These procedures also enable the influence of changes in system design to be predicted. Some comparisons of these procedures with field data are presented.
Response of a Honeycomb Panel to the Pseudosound Field of a Jet
Authors: Maestrello L. , Gedge M. R.
Published: September 2005

Prediction techniques are applied to aluminum honeycomb panel of size 12×7×0.10 in. with core depth of 0.08 in. mounted along, and just outside of, the wake of a model jet. From measurements made, a functional representation of the pseudosound wall-pressure correlation is obtained. The response of the panel is then determined by employing the superposition of normal modes for the case of simply supported boundaries. The measured mean-square displacement is closely approximated, with the largest discrepancy occurring at the center of the panel. The predicted modal energy distribution has the maximum deviation in the fundamental mode corresponding to the region in which the pressure field model is most inaccurate. For the higher modes, better agreement is obtained with predicted levels lying within a factor of 3 of the measured level.
Preliminary Studies of Voice Tracking Behavior
Authors: Elliott Lois L. , Niemoeller Arthur F.
Published: September 2005

Each of a sample of 21 normally hearing adults attempted to match his vocal fundamental frequency to the frequency of tonal signals presented either before or during vocalization under a variety of conditions. He was never informed of his accuracy. The vocalizations were sustained for about 2 sec and were processed through a specially designed “pitch meter”and strip recorder with continuous output and fast rise time. In general, during the first 12-sec of vocalization, the fundamental frequency shifted either up or down toward the target frequency. Subjects were within about 1% of the target frequency on approximately half the trials, although they showed a tendency to be “on target” for as many as 80% of the trials if they had been “on target” in the preceding trial. None of several measures of vocal pitch matching was related to the more classical auditory pitch matching.] This research was supported in full by a U. S. Public Health Service, Department of Health, Education, and Welfare research grant from the National Institute of Neurological Diseases and Blindness.]
Automatic Recognition of Spoken Words
Authors: von Keller T. G.
Published: September 2005

An approach to automatic word recognition based on simple segmentation rules will be described. The speech sound wave is segmented where significant changes in articulatory configuration are reflected in easily measurable acoustic features. Examples of such articulatory changes are closures or near closures of the vocal tract, nasal coupling, and changes in excitation. The segments thus obtained do not necessarily correspond to individual phonemes or to syllables. The actual recognition program consists of three main parts. Certain features of the acoustic wave are measured first and are then used to segment the utterance. Finally, the word is identified by the sequence of these segments. This program was used to recognize words spoken in isolation by a number of male and female speakers. The results will be presented and evaluated.
Automatic Detection of Intonational Signals in American English by Computer Simulation
Authors: Takefuta Yukio
Published: September 2005

A study of automatic detection of intonational signals in American English has been carried out by a digital-computer simulation. Using a Trans Pitchmeter, the fundamental frequency was measured at every 1/40 sec of each utterance. The utterance was segmented into constituent units of half-syllables. The pattern of frequency modulation in each of these constituent units was quantified from the slope of the linear regression line fitted to the raw data of fundamental frequencies in each unit. The level of fundamental frequency was obtained likewise by calculating the mean of the measured fundamental frequencies. After further processing, the output of the analysis was compared with the dictionary data in the computer storage. Detection was performed by adopting a decision-tree method based upon experimentally derived rules. When tested on a sample of 500 sentences recorded by five native female speakers, the signal detectability of this system was as good as the average detectability of 20 native listeners. [This research was supported by a contract between The Ohio State University Research Foundation and the Office of Naval Research.]
Ultrasonic Absorption in Dimethyl Sulfoxide Solutions of Alkali Halides
Authors: Dickson D. R. , Kruus P.
Published: September 2005

The ultrasonic absorption of solutions of alkali halides in dimethyl sulfoxides is studied by the pulse technique over the frequency range of 3.5 to 60 MHz. A relaxation is observed near the low-frequency end of the range for each salt studied. This relaxation is assigned to the second equilibrium in the two-step formation of a contact ion pair from free ions in the solution: [M+]sol.+[X−]sol.⇄[M+(DMSO)X−]sol.⇄[M+X−]sol.. There is evidence that a relaxation is also occurring at a frequency higher than the range of the apparatus. This may be due to the first step.
Effect of Pressure on Ultrasonic Absorption and Chemical Equilibrium in MnSO4 Solutions
Authors: Fisher F. H. , Wright W. M.
Published: September 2005

Measurements were made on the effect of pressure on ultrasonic absorption in a 0.5M aqueous solution of manganese sulfate at 25°. A cylindrical resonant cavity was used for obtaining absorption in the frequency region 300–500 kHz at pressures up to 20 000 psi. Results are similar to those obtained for MgSO4 in that absorption decreases by a factor of about 3 for a pressure increase of 15 000 psi. Independent conductivity data show only a decrease of about 14.5% in the concentration of dissociated ions at 1000 atm, about the same as for MgSO4. The data support the existence of a low-frequency relaxation in the ∼200-kHz region. [Work sponsored by the Office of Naval Research.]
Ultrasonic Propagation in Polyethylene Near the Melting Point
Authors: Hartmann Bruce , Jarzynski Jacek
Published: September 2005

The absorption and velocity of ultrasound in high-density polyethylene have been measured at a frequency of 53 kHz from room temperature to above the melting point. The acoustic attenuation was found to have a very pronounced maximum at the melting point, in excess of 10 dB/cm relative to the attenuation at room temperature. Writing the thermodynamic parameters in the vicinity of the melting point as the sum of a frozen part and a relaxational part, a simple equation for the relaxation strength can be derived from a two-state model for the melting process. Using the best available data for the,thermodynamic parameters, a relaxation strength of 0.6 was calculated at the melting point. This relatively large value for the relaxation strength is in agreement with the observed maximum in acoustic attenuation.
Apparatus and Technique for Auditory Stimulus Sorting
Authors: Bricker P. D. , Johnson S. C. , Mattke C. F.
Published: September 2005

Techniques that allow subjects to sort stimuli categorically or order them according to some attribute have been widely used with visual stimuli but are generally inapplicable to auitory stimuli. An apparatus is described that allows subjects to control the presentation of up to 30 auditory stimuli and to arrange the control buttons, which also serve as visual symbols for the stimuli, in spatial relationships to each other according to instructions. When the number of stimuli is between 20 and 30, the essential dominance or similarity relations among them may be obtained from a subject in a fraction of the time required for paired comparison presentation. This brevity of task and facility with which subjects can and do recheck their arrangements reduce hazards from shifting criteria and boredom. Techniques for treating data from various tasks performed on the same stimuli are briefly described and illustrated by data from experiments on subjective evaluation of signaling tones. Potential applications to such problems as language acquisition and voice-quality analysis are discussed.
Sine and Cosine Masking
Authors: Green David M.
Published: September 2005

Results of the masking of one sinusoid by brief pulses of another sinusoid are reported as a function of the phase of the masking pulse. In one case, the brief masker (10 msec) is presented as a sine pulse; in the other case, the masker is presented as a cosine pulse. The sine pulse produces more masking than the cosine pulse at frequencies below the frequency of the masker and less masking at frequencies above the frequency of the masker. This inversion is explained on the basis of the power spectra of the two pulses. Details of the results are compared with a simple filter model and indicate that the ear's filter is at least a second-order system with a greater rate of attenuation above the center frequency of the filter than below. [This research was supported in part by the National Institutes of Health, Public Health Service, U. S. Department of Health, Education, and Welfare.]
Residual Masking by Pure Tones Measured on the Békésy Audiometer
Authors: Ehmer Richard H. , Ehmer Barbara J. , Seamon John , Cohen H. Harvey
Published: September 2005

A Békésy audiogram for a 5-msec tone signal repeated once per second was obtained over the frequency range 100–10 000 Hz. The audiogram was repeated with a 100-msec masking tone preceding the signal by 10 msec. The masking tones were presented at 20–100-dB SL's. The results show that 250 and 500 Hz produce no masking at 20 dB but the greatest over-all masking, without sharply defined maxima, at higher sensation levels. An 8000-Hz tone produces the least masking and attains a limit before the highest masking level is reached. Maskers of 1000, 2000, and 4000 Hz produce masking even at the 20-dB level. At the low levels, there is a clear peak at the masking frequency; but as the masker level is raised, this peak begins to shift to increasingly higher frequencies, although it never reaches the second harmonic. [Work supported by the National Institute of Neurological Diseases and Blindness.]
Air Conduction, Bone Conduction, and Underwater Hearing Thresholds in Man
Authors: Smith Paul F.
Published: September 2005

Air conduction (AC), bone conduction (BC), and underwater-hearing thresholds were obtained on three samples of underwater swimmers. In Sample I, AC and BC hearing levels varied over a considerable range at the higher frequencies. Underwater hearing thresholds were positively correlated with BC thresholds. The loss of sensitivity upon immersion was negatively correlated with AC hearing levels. In Sample 2, underwater hearing thresholds of three divers having depressed AC hearing levels at 6 kHz and two divers having depressed AC and BC hearing levels were compared with underwater hearing levels of eight normal-hearing divers. Depressed underwater hearing levels occurred for the divers having combined AC and BC losses. Divers exhibiting only AC losses heard as well underwater as normal hearing divers. Maximum sensitivity for the normal hearing divers over the frequency range 1–8 kHz was −20 dB/μbar at 1 kHz. In a third sample, data were obtained on threshold sensitivity over the 125-Hz to 8-kHz frequency range. Maximum sensitivity of −23 dB/μbar was observed at 1 kHz.
Equivalent Peak Level: A Threshold-Independent Speech-Level Measure
Authors: Brady Paul T.
Published: September 2005

Previous research showed that the logarithm of the absolute voltage of a speech waveform is nearly uniformly distributed between some arbitrary threshold and a peak. In the present study, the equivalent peak level (epl) of a speech sample is defined as that peak of a log-uniformly distributed random variable that would have produced the same rms above threshold as was measured from the speech sample. Measurements on 116 speech samples show that, if threshold is varied over a range of at least 35 dB on repeated measurements of the same speech sample, the epl varies ±1 dB. Equivalently, if the measuring threshold is held constant, the epl will follow known speech level changes on a dB-for-dB basis to within ±1 dB over a 35-dB range. The epl may be measured with any facility having means of obtaining rms above threshold.
Rules for Synthesizing Prosodic Features of Speech—Preliminary Investigation
Authors: Rabiner L. R. , Levitt H. , Rosenberg A. E.
Published: September 2005

In an earlier investigation [L. R. Rabiner, Bell System Tech. J. 47, 17–37 (1968)], a procedure for synthesizing speech by rule was described. The speech generated by this procedure had good intelligibility, but a machinelike quality. This was due, in part, to inadequate rules for controlling the prosodic features of the utterances (e.g., stressed vowel duration, fundamental frequency contour). In order to improve the quality of the synthesized speech, a preliminary investigation was carried out using three simple, declarative sentences. The stress pattern for each sentence was defined by assigning one of four possible stress levels to each vowel. Increments in duration and fundamental frequency were used as the correlates of stress, and various combinations of these factors were evaluated using a paired-comparison technique. One set of conditions was found to be highly favored for each of the three test sentences by a crew of six subjects. Using the results of this experiment, a set of rules was determined for controlling prosodic features in synthesizing simple, declarative sentences. Examples of synthetic speech demonstrating these rules will be played.
Parallel Resonance Synthesizer for Speech Research
Authors: Glace Donald A.
Published: September 2005

This paper describes the development of a parallel terminal analog speech synthesizer designed for the needs of a nontechnically oriented user who has minimal engineering support. Salient features considered are reasonableness of cost, ease of maintenance and calibration, flexibility of configuration, and compatibility with various input devices. Four vowel resonances, a nasal resonance, and two fricative resonances are included. Resonance bandwidths are manually adjustable over a wide range. The excitation may be either pulses, noise, or a mixture of both. The synthesizer requires 10 time-varying parameter inputs and one two-bit state control. A parameter switch steers the parameter inputs to 14 synthesizer control points according to the synthesizer state. Integrated circuit operational amplifiers are used extensively to provide linear control functions.
Effects of Speech Combining within Analysis-Synthesis Processes
Authors: Helms Stanley W.
Published: September 2005

Qualitative results of various combining techniques for pitch-excited and voice-excited vocoders have been obtained using the Diagnostic Rhyme Test (DRT). Evaluation results of the two techniques of ORing and Summing of respective vocoder channels is presented in detail based on the discriminability of the six consonant attributes of the DRT over a range of test-speech to masking-speech (TS/MS) ratios. ORing is shown to perform as good or better than Summing over the range of TS/MS ratios of greater than −6 dB. Relative results of PEV and VEV combining are given, as well as some all-digital techniques, which are compromises between ORing and Summing of respective channels. [This research was conducted under contract with AFCRL, Office of Aerospace Research, U. S. Air Force.]
Near-Surface Underwater Acoustic Duct as a Communication Channel
Authors: Bright C. W. , Thomas R. S.
Published: September 2005

The characteristics of the surface duct as a communication channel determine the rate of acoustic transmission of information from one point to another. A deep-water acoustic-propagation experiment has been performed in which relatively stable transmitting and receiving platforms were placed in the surface duct in an attempt to obtain a measure of the channel stability and multipath structure. In the results, the channel stability is expressed as a frequency spreading on CW tones and multipath as the time spreading of short pulses. Both of these parameters depend upon the physical geometry of the experiment and upon the roughness of the ocean surface. Sound-ray plots show that energy is propagated between two relatively close points by means of surface paths and a direct path through the water alone, and between widely separated points by means of a series of surface paths, illustrating the increasing dependence of channel characteristics on surface conditions as the propagation distance increases.
CW Phase Measurements across Block Island Sound
Authors: Huff Lloyd C.
Published: September 2005

A shallow-water acoustic range 40 m deep has been investigated at a range of 33 km to determine the magnitude and temporal character of phase variations in the received signals. Variations sinusoidal in nature with peak-to-peak amplitude on the order of 20 cycles of the acoustic carrier were observed at the tidal periods. The phase of the received signal retarded during flood tide and advanced during ebb tide. Carrier frequencies of 1700 and 1704 Hz were employed in separate experiments conducted under mean sound velocities of 1470 and 1450 m/sec, respectively. The results of both tests demonstrate that there is a temporal scale of phase variability closely related to the tidal phenomena.
Experimental-Theoretical Model of Directional Ambient Ocean Noise
Authors: Klaus Pat , Cerceo M.
Published: September 2005

A mathematical-experimental model for the prediction of the directional characteristics of deep ocean ambient noise for an arbitrary environment is developed from ray acoustics. The surface disturbance is assumed to be an invariant for all deep-ocean environments and is calculated from directional noise data for several Beaufort wind speeds and frequencies between 500 and 1500 Hz. Utilizing the calculated surface disturbance, dependence of the anisotropic noise on depth (from 150 to 12 000 ft) bottom loss and velocity profile were investigated. Comparison of Beckens averaged data with data predicted from the model agree to within 1° for the location of the null, and to within 1.5 dB for the magnitude at a particular angle. Results indicate the feasibility of predicting the anisotropic noise and show that the shape of the directional noise field is dependent upon receiver depth, the presence of shadow zones, and bottom loss. The predicted change with depth of the noise level as would be measured by an omnidirectional hydrophone was from 0.5 1.5 dB for depths from 150 to 12 000 ft.
On the Design and Calibration of Infrasonic Gradient Microphones
Authors: Bauer Benjamin B.
Published: September 2005

Until recently, infrasonic phenomena have been explored mainly with pressure microphones. Gradient microphones, especially when arranged in an orthogonal bigradient array, have the advantage of providing information about the direction, as well as the intensity and time, of sound arrival. A low-frequency bigradient array is described. Windscreens appear to affect the response of gradient microphones to a greater extent than that of pressure microphones, and some factors affecting the windscreen performance are described. The calibration of infrasonic gradient microphones poses a special problem because of the difficulty of obtaining a low-frequency plane progressive sound field. A calibrator for infrasonic gradient microphone with internal dimensions of 4×4×8 ft has been developed along the principles of the laboratory calibrator for gradient hydrophones described by the present author [J. Acoust. Soc. Am. 39, 585–586 (1966)].
Tube-Cavity Resonance
Authors: Troke Robert W.
Published: September 2005

Mathematical definitions of all tube-cavity vibration modes are developed from experimental results. The crossover from one mode to another is defined in terms of pertinent design parameters. A simple, accurate, and first-time-presented solution is developed for the spring-mass vibration mode. The higher harmonics associated with the spring-mass resonators, commonly called Helmholtz resonators, are identified from experimental data. Tube-cavity limit conditions of zero length as well as zero and infinite volumes are included in this study. The resonance modes observed are spring-mass, half-wave, and quarter-wave. Using the new solution for the spring-mass mode, all natural resonances are found to be bracketed by the limit conditions of the adiabatic and isothermal processes.
Mobile Hydrophone Suspension System for the Measurement of Low-Frequency Sound
Authors: Franklin J. B. , Toye J. R.
Published: September 2005

The investigation of the effect of environmental factors on low-frequency sound propagation and ambient noise in the deep ocean requires measurement equipment that is mobile, flexible, and relatively uncomplicated. These requirements can be satisfied by a ship-deployed, surface-suspended, hydrophone system if self-noise levels due to the motion of the hydrophone and suspension components in the water can be made low. Self-noise mechanisms including flow noise, Aeolian sound, hydrostatic-pressure changes, and acceleration noise are described. The results of experiments carried out at sea to determine the dominant noise mechanisms and to investigate procedures for reducing self-noise levels are presented. The final suspension system is discussed with particular reference to the requirements for streamlined acceleration-balanced hydrophones and vertical and horizontal decoupling from surface excitations. Measured noise spectra are compared with published averages for similar sea states and traffic conditions.
Method of Data Reduction in Studies of the Effect of Size on the Response of Transducers to Turbulent Boundary-Layer Pressure
Authors: Geib F. E.
Published: September 2005

The response of a flush-mounted transducer to the pressure field in a turbulent boundary layer is known to depend on the spatial characteristics of the transducer. Corcos [J. Acoust. Soc. Am. 35, 192–199 (1963)] examined this dependence and presented his results in terms of universal functions, each of which pertains to a class of similar transducers. The class is defined such that the spatial operator of one transducer of the class can be derived from another by a simple scaling of the typical spatial linear dimension of the transducer. It is shown that in an experimental study, if one wishes to express the dependence in a manner corresponding to that of Corcos, one must employ a pair of transducers. These transducers must belong to the same class and have different spatial sizes. The method of data reduction to achieve the desired form of presenting the results of the experiment is discussed. The nature of the assumptions and idealizations involved in validating the data-reduction procedure are made explicit.
Solution of the Wave Equation for Moving Boundaries
Authors: Brown Bernard , Libelo Louis F.
Published: September 2005

Boundary-value problems in which boundaries translate rigidly at constant velocity have been studied. The wave equation has been solved for small-amplitude harmonic disturbances within translating boundaries. The effect of motion on normal modes and wave patterns will be presented for the case of a moving rectangular parallelepiped.
Expansion of an Underwater Cylinderical Gas Bubble
Authors: Epstein David , Keller Joseph B.
Published: September 2005

The theory of the expansion of an underwater spherical gas bubble is rather well known; in this paper the theory for cylindrical gas bubbles is developed. It is customary to treat the motion of the fluid in the incompressible approximation, but this procedure leads to difficulties in the cylindrical case owing to the potential's becoming infinite as r approaches infinity. Therefore, the fluid is treated as compressible, and equations for the bubble radius and the radiated pressure as functions of time are derived. Then these equations are examined in the limit of large speed of sound. Approximate equations obtained by employing simplifying assumptions are analyzed, and the properties of the motion discussed.
On Acoustic Boundary Conditions at a Liquid-Solid Interface
Authors: Witting James
Published: September 2005

When sound waves in a liquid reflect from a solid boundary at almost glancing incidence, should one invoke a boundary condition of vanishing normal velocity or of vanishing pressure? Experiments in water-filled aluminum and mercury-filled stainless steel waveguides had agreed much better with the vanishing pressure condition. Viscosity or air pockets have been given as an excuse for the unexpected result. Now, for reasonably small angles of emergence e, only evanescent waves are excited in the solid, and the wave in the liquid is reflected without change of amplitude. The phase of the velocity potential changes, however. Vanishing pressure involves a 180° phase change; vanishing normal velocity, a 0° phase change. I have calculated this phase change assuming a nonviscous fluid and an isotropic solid for angles of emergence between 0° and that which generates a Rayleigh wave. For e→0°, the pressure vanishes at the boundary. Furthermore, the phase angle lies closer to 180° than to 0° for a large enough range of angles near e = 0° to explain quite naturally the results of the previously performed experiments. [Work supported in part by the National Aeronautics and Space Administration.]
Perception of Intonational and Accentual Signals Processed by a Sampling-Synthesizing Technique
Authors: Takefuta Yukio , Earle Floyd E.
Published: September 2005

The English vowels and consonants processed by a sampling-synthesizing technique were reported to be intelligible. The purpose of this study was to investigate the effect of processing speech by spectral sampling on the perception of intonational and accentual signals by human listeners. The frequency spectra of 40 sentences and a set of 170 two-syllable words recorded by six native American speakers were processed by sampling at 22 points, approximately 13 oct apart, along the frequency scale of 55 to 9600 Hz. A group of 20 listeners detected the intonational and accentual signals from the sample speech before and after the processing. Their responses were classified into four conjunctions of HIT, CORRECT REJECTION, FALSE ALARM, and MISS to investigate the effect of spectral sampling on these linguistic signals. A standard index of signal detectability was also calculated to quantify the efficiency of intonational and accentual signals with and without processing by this technique. [This research was supported by a contract between The Ohio State University Research Foundation and the Office of Naval Research.]
Speaker Variability in Intelligibility Testing
Authors: Preusse John W.
Published: September 2005

Frequently, when recorded test lists are used in intelligibility testing of communication equipment, only a very few speakers are used to provide the source material, in contrast with the relatively large number of listeners employed. Our studies have shown that such a small number of speakers as two or three is not sufficient to provide a fair evaluation in all cases. Particularly, in evaluating “speaker-sensitive” equipment, some extraordinary differences among speakers have been found. Thus, it is easy to see how such equipment might be unfairly biased by the use of a small number of speakers, all of whom happened to be not well suited to it. One cannot, how, use large numbers of speakers speaking all test lists since the amount of source material would be prohibitive. A solution is to rotate a large number of speakers within each test list so that each list provides a sampling of all the the speakers used.
Evoked Potential (EP) Correlates of a Click Stimulus Intensity Difference Limen (DL)
Authors: Saunders James C.
Published: September 2005

An intensity DL was determined in cats by using a modified Békésy tracking procedure. Clicks were continuously presented the subject, one every 400 msec. A discrimination trial occurred when the intensity of alternate clicks was increased. The difference click was designated C1 while the standard was C2. On successive trials the acoustic difference between C1 and C2 was increased or decreased with a correct or incorrect behavioral discrimination. With the standard stimulus arbitrarily set at −25 dB re 1 dyn/cm2, the DL in four subjects averaged 4.5 dB with a standard deviation of 0.7 dB. All subjects had electrodes implanted in the cochlear nucleus (CN) and auditory cortex (AC). The EP's produced by C1 and C2 were computer averaged for every trial, and the microvolt difference between EP1 and EP2 was correlated with the acoustic difference and the behavioral response. The EP's at the CN reflected the acoustic differences regardless of whether a correct or incorrect behavioral discrimination occurred. The EP's at the cortex, however, did not always reflect the acoustic input, and differences were observed for correct and incorrect discriminations.
Reduced-Memory Receiver Design for Adaptive Detection of Transient Signals
Authors: Baxa E. G. , Nolte L. W.
Published: September 2005

Detection of one of M transient signals that recurs randomly in the present of noise is investigated. The signal is described as one of 2n fixed n-bit binary components with a time-recurrence structure that results in a signal ensemble of time sporadically recurrent binary components. The optimum likelihood-ratio detector can be realized as a sequential detector for this sporadic Poisson signal process. Even this sequential realization requires a rapidly expanding memory as the component ensemble size increases. This imposes a limitation on the practicality of implementation of the receiver design. Reduced-memory receiver designs are considered which provide near optimum detection performance. A memory scheme derived directly from the optimum detector design has been incorporated in a detector which exhibits very nearly optimum performance. This reduced-memory detector has been evaluated at various signal-to-noise ratios using the Receiver-Operating-Characteristic (ROC) curve. The reduction in memory afforded by the suboptimum detector appears to be quite significant, while variation from the optimum detector performance is relatively insignificant. [Work supported by the Office of Naval Research (Acoustic Programs).]
Shadowing of Randomly Rough Surfaces
Authors: Hawker K. E. , Welton P. J.
Published: September 2005

Measurements have been made of the reduction in illuminated area of a randomly rough surface due to shadowing in the geometrical optics limit. These measurements were made using extremely high-contrast photographs of a randomly rough model surface having known statistics. The illumination was quasicollimated incoherent light. The resulting photographs were measured by two different methods in order to determine the ratio of total area to shadowed area. Data are given for several surfaces having different mean-square slopes. The resulting shadowing curves, given as a function of grazing angle, are compared with a theory developed by Wagner. The agreement was found to be generally good, and the theory for area reduction due to shadowing appears to be sufficiently accurate to warrant its incorporation in the theory of scattering from randomly rough surfaces.
Attenuation of Acoustic Waves by Submerged Viscoelastic Spherical Shells
Authors: Workman George , Hayek Sabih I.
Published: September 2005

A pulsating sphere is located inside a thick concentric viscoelastic spherical shell, which is surrounded by an infinite acoustic medium. The space between the sphere and the shell is filled with the same acoustic medium. A shaped beam of axisymmetric spherical harmonic wave, emitted from the sphere, is transmitted through the shell to the outside medium. The shell is analyzed by use of elastic wave potentials. The emitted acoustic waves are transmitted to the external medium by coupling to the shell through continuity and boundary conditions at the inner and outer surfaces of the shell. The investigation shows that the radiated farfield pressure can be attenuated by as much as 10% through the use of stiff polymers.
Effect of Dispersion and Scattering on the Startup of Brass Instrument Tones
Authors: Benade A. H.
Published: September 2005

A clean attack is favored if the round-trip time Tr for the initial sound from the lips sent down the bore and back is an integral multiple of the playing frequency period Pn. Consider a Bessel horn of length L, radius r, with r(y)=B/yα. For a pulse with spectral center ωn, the group velocity vg=c[1−(c/ωn)2α(α+1)]12, whence Tr=(2/c) [L2−ln2]12, where ln2=(c/ωn)2α(α+1). The oscillation period of the nth mode of this horn is approximately Pn=(4/c) (L−ln)/(2n−1) implying that a Bessel horn does not give a clean attack. A mouthpiece, mouthpipe, cylindrical tube, and Bessel bell can be designed to give good attack plus correct tuning. Short constrictions in a bore give early partial reflections which also damage tone onset. A 10% diam reduction 2 cm long (as in some horn valves) gives a 7% reflected amplitude at 330 Hz. Valves in combination can give considerably larger reflections. Missed notes and burbles are less common in instruments with slight constrictions.
Scaling Parameters for Some Normal Incidence Absorption Measurements. Apparatus for Measuring Grazing Incidence Absorption
Authors: Colson Ernest
Published: September 2005

Absorption coefficients of various samples constructed from acoustically resistive materials, air spaces, and perforated plates were measured with a standing-wave tube apparatus. Scaling the test apparatus. Scaling the test data with respect to the “Helmholtz resonance parameters” and/or the acoustic wavelength is shown to be a useful empirical technique for the design of absorbent structures. For broad-band absorbers “Helmholtz resonance absorbers” should usually be placed behind “tube resonance absorbers.” The wave-amplitude dependence of absorption coefficients is noted and its scaling discussed. A rectangular tube was constructed consisting of two rigid walls a fixed distance apart; adjustable cross walls serve as sample holders. Exploratory measurements with rigid and absorptive terminations will be discussed.
Relaxation Spectroscopy of Liquids
Authors: Litovitz Theodore
Published: September 2005

A discussion of ultrasonic, orientational, and dielectric relaxation in liquids showing how these different transport phenomena are related.
Acoustical Vestibular Stimulation in the Guinea Pig: Head and Eye Movements
Authors: Reschke M. F. , Parner D. E.
Published: September 2005

Postural reflexes and eye movements associated with slow time-varying pressures in the external auditory meatus were recorded using guinea pigs as subjects. These responses have been traditionally associated with stimulation of the vestibular apparatus. Latencies to pressure onset and release, durations, and amplitudes were observed for both head and eye movements as a function of stimulus intensities, durations, and onset rates. Head and eye movements elicited by pressure stimulation were found to be significantly correlated. The results from control studies employing streptomycin intoxication, centrifuge removal of the otoconia from the saccular and utricular maculae, and destruction of the cochlae with broad-band noise are interpreted as supporting the hypothesis of acoustical stimulation of vestibular receptors.
Hearing in Prosimians
Authors: Heffner Henry E. , Thurston Eric , Masterton Bruce
Published: September 2005

As part of a study of the auditory capacities of mammals, behavioral audiograms were determined for three species of prosimians: African bushbaby (Galago senegalensis), potto (Perodicticus potto), and slow loris (Nycticebus coucang). The method of conditioned suppression was used to assess thresholds for frequencies ranging from 125 Hz to 64 kHz. Comparisons of the three audiograms with those of more primitive mammals on one hand, and anthropoids on the other, reveal that prosimians retain several primitive hearing characteristics while in some other characteristics, such as high-frequency hearing, they more closely resemble Anthropoids.
Signal-Gain Comparison of Three Schemes for Filtering Signals that Have Propagated through a Random Multipath Medium
Authors: Mangano D. T.
Published: September 2005

Three filtering schemes for processing random, multipath signals are compared. The criterion of comparison is the average signal gain (a ratio of the time average power of the processor output to that of a typical hydrophone output). A fixed, onmidirectional source emits a signal that propagates through a multipath medium and arrives at an array of N receiving elements. The medium is characterized by a set of attenuation and delay random variables associated with each of the multiple paths. The received signals are then processed by either delay filtering, matched filtering, or frequency-inverse filtering. It is found, for all three filters, that the least upper bound on the average signal gain when the medium is deterministic is N times greater than the gain when the medium is random. Furthermore, this signal gain for frequency-inverse filtering is at least as large as that for delay filtering; for matched filtering, it can assume a range of values depending on the length of the source pulse and the time of integration. [Work supported by Electronic Systems Command, U. S. Navy.]
Instrumentation for Acoustic Modeling
Authors: Watters B. G.
Published: September 2005

Acoustic models are usually built so that one can make the same kind of measurements one would make if the full-sized space or structure were available. Thus, much of the instrumentation needed for model work is the same as that used to evaluate completed projects. Other instruments must be high-frequency analogs of conventional equipment in order to accommodate the model's frequency and time scale factor. This paper will deal with instrumentation and techniques that have proved useful in acoustic models with scale factors ranging from about 1:3 to 1:24. Suitable high-frequency microphones, spark-discharge sound source, cathode-ray-tube “graphic level recorders,” and similar equipment will be discussed.
Effect of Finite Size on Superfluidity; Third Sound
Authors: Rudnick I. , Fraser J. C. , Kagiwada R.
Published: September 2005

When the quantum liquid, He4, is bounded in size and when one or more of its dimensions becomes comparable with a characteristic healing length, the superfluid state is altered. In particular, the superfluid onset temperature is lowered and the superfluid fraction at a given temperature is reduced. Third sound is a surface wave that propagates in liquid-helium films where only the superfluid component moves. Third-sound measurements have been made in films several atomic layers thick from which reduced onset temperatures and superfluid fractions have been determined. These results will be discussed in the framework of current theories.
Nonlinearities in Cochlear Hydrodynamics
Authors: Tonndorf Juergen
Published: September 2005

Cochlear traveling waves show properties of capillary waves progressing along interfaces between two nonmixing fluids. Such wave motion is basically nonlinear. Mechanical cochlear models display at least two types of nonlinearities that may explain a number of auditory and electrophysiological phenomena. (1) The displacement of the cochlear fluids and of the partition is asymmetrical, larger amplitudes being directed away from the input toward scala tympani. The asymmetry is also present in the shearing displacement within the cochlear duct. These phenomena were first observed with the aid of beating signals and tone pips. The asymmetrical displacement of the partition lets the model detect the signal envelope due to its lowpass filter action. The degree of asymmetry is independent of signal amplitude. There is good indication that the asymmetrical displacement is caused hydrodynamically, specifically, that it is a boundary-layer effect. The latter appears to involve vorticity, a phenomenon that, by definition, introduces nonlinear factors. (2) A second source of hydrodynamically induced nonlinearity is constituted by Békésy's eddies. These rectified events appear at higher signal levels. They introduce an amplitude-dependent nonlinearity, i.e., harmonic distortion. [This work was supported by NIH grants.]
Nonlinear Cochlear Microphonic Potentials
Authors: Eldredge Donald H. , Engebretson A. Maynard
Published: September 2005

The principal features of the linear relations among CM's recorded with differential electrodes at various stations along the basilar membrane of the guinea-pig cochlea agree well with the relations described by yon Békésy for the traveling wave. However, there are about as many nonlinear as linear relations for these CM's. The CM recorded from upper turns is a nonlinear function of input sound pressure at all input levels for some frequencies, and for high input levels at all frequencies. Interference is a nonlinear phenomenon in which the presence of one tone decreases the amplitude of the CM response to a second tone. Using simultaneous measurements from multiple pairs of differential electrodes, we have found that interference between two tones may be confined to a single turn in the cochlea. Several observations restrict the place of nonlinear interference to the hair cell or to local mechanical phenomena as opposed to over-all hydromechanical patterns or to Békésy's eddies. [Work supported by National Institute of Neurological Diseases and Blindness and Division of Research Facilities and Resources.]