bluuebunny/crossref_metadata_2025 huggingface dataset:
An Approach to the Computer Recognition of Single-Syllable English Words
Authors: Medress Mark
Published: October 2005

A method has been developed to abstract distinctive features information about vowels, stops, and fricatives from a filter bank representation of single-syllable, single-morpheme English words. The procedure is organized with respect to and exploits the phonological structure and rules for such words, and no reference to a lexicon is made. Because the program relies primarily on gross spectral properties, it can handle a variety of male speakers with no alterations required. Location, classification, and false alarm scores of better than 92%, 91%, and 7%, respectively, were obtained for repetitions of a list of 105 such words by seven speakers. Based on this experience, 24 words were carefully chosen and subjected to the recognition procedure. An average word recognition score of 85% was obtained for eight speakers. Recognition was significantly improved if lexical information was used in addition to acoustic and phonological information. [Work supported in part by grants from the National Institutes of Health and the Air Force Cambridge Research Laboratories.]
Reduction of Long-time Reverberation by a Center-Clipping Process
Authors: Mitchell O. M. M. , Berkley D. A.
Published: October 2005

A center clipping process has been found to be effective in removing the reverberant tails of speech produced in a room with long reverberation time without significantly reducing intelligibility. The instantaneous output of a center clipper is zero unless the absolute value of the input exceeds a threshold value and otherwise varies linearly with the input. The input speech is divided into several channels by a set of contiguous band filters each less than 1 oct wide, and the output of each filter is independently center clipped. Harmonic distortions introduced by the center clippers are then removed by an output filter bank identical to the input set of filters. A six-channel system using 23-oct filters (250–3500 Hz) was simulated on a CDC 3300-EA1 8800 hybrid computer to process input speech recorded in an auditorium. Clipping levels were set so that the output of each center clipper was zero approximately 50% of the time. An output tape showing reduction of reverberation will be played.
Optimum Linear Processing of Signals for Speech Transmission
Authors: Fujisaki Hiroya , Niwa Kunihiko
Published: October 2005

Efficient utilization of a communication system requires that statistical properties of the signal be matched to characteristics of the over-all system noise. This can be achieved, although approximately, by an appropriate linear processing of the speech signal prior to modulation at the transmitter, and restoring the signal by another processing at the receiver. Based on measurements of signal characteristics and various performance criteria, optimum processings at the transmitter have been theoretically derived for conventional modulation systems including AM, PCM, FM, and PM. These systems have then been simulated and evaluated by articulation tests. The results indicate that the processing which maximizes the average rate of information transmission yields the highest articulation score, proving the validity of the information transmission rate as theoretical criterion for system performance. The maximum improvement is equivalent to a gain in signal-to-noise level of about 9 dB in the case of FM. It has also been shown that the requirements on fidelity and intelligibility coincide in the determination of the optimum processing characteristic at the receiver.
Application of the Temporary-Threshold-Shift Paradigm for Assessing Sound Transmission in the Auditory System during Vowel Production
Authors: Karlovich R. S. , Luterman B. F.
Published: October 2005

Ten female subjects were exposed to a 110 dB SPL 1000-Hz fatigue stimulus for 3 min. During the exposure time, the subjects repeatedly produced a voiced vowel, a whispered vowel, or performed a nonvoiced articulatory gesture representing a vowel without whispering. Pre- and postexposure thresholds were tracked with a Békésy-type procedure for a stimulus 12 oct above the fatigue frequency. Temporary threshold shift (TTS) following voiced /a/ and /i/ vowel conditions was significantly less than that observed for corresponding whispered or nonvoiced conditions at each postexposure recovery time measured. The magnitude of these differences ranged from 9 dB at 10 sec to 3 dB at 3 min of recovery. The results strongly indicate that voiced-vowel production impedes transmission and hence reduces the energy delivered to the cochlea from a 1000-Hz fatigue stimulus as evidenced by the reduced TTS magnitudes associated with this condition. Two potential mechanisms are proposed to account for alteration in sound transmission during voiced-vowel production. One mechanism involves possible middle-ear muscle contraction during voiced-speech production, and the other concerns the effects of inertial bone conduction during such conditions. [Supported in part by the Wisconsin Alumni Research Foundation.]
Auditory Sensitization—Fact or Artifact
Authors: Melnick William
Published: October 2005

One of the effects of exposing the ear to sounds of moderate level appears to be an enhanced sensitivity of the auditory system following the exposure. This investigation was interested in whether the increased sensitivity merely reflects longer isolation from higher level ambient noises that is confounded with measurement of postexposure thresholds in the usual temporary-threshold-shift experiment. Fifteen males with normal hearing were tested monaurally under two sets of binary conditions: (1) immediate measurement vs a delay of 5 min; (2) exposure to 1000 Hz at 40-dB sensation level for 3 min vs 3 min of quiet. Pre- and postexposure thresholds were measured for 1000 and 1500 Hz. Threshold was tracked for 3 min postexposure at 1000 Hz, then was tracked for 1 min at 1500 Hz. Results showed no effect for the time-of-test-onset variable. Surprisingly, both the exposure and the quiet conditions showed a decrease in postexposure threshold for 1000 Hz, with a greater decrease for the quiet condition. Maximum decrease approached 2 dB. A decrease also was observed at 1500 Hz in the fourth minute postexposure.
Response Latencies in the YES-NO Signal Detection Situation
Authors: Emmerich David S.
Published: October 2005

Response latencies were recorded from observers given the task of detecting sinusoidal signals presented in a white noise background. On each trial, the probability that a signal would be presented was 0.5. The observers indicated their decisions as to whether or not a signal had been presented by pressing buttons. They were instructed to adopt different decision criteria (strict, medium, or lax) for making a YES response during different sessions of the experiment. For most criteria adopted, the mean latency of YES responses was less than the mean latency of NO responses. In addition, the mean latency of YES responses made after a signal had been presented was less than the mean latency of YES responses made after noise alone had been presented. Similarly, the mean latency of NO responses to noise alone was less than the mean latency of NO responses to signals. Latencies varied widely as a function of the criterion adopted. Receiver operating characteristics were constructed from the latency distributions, as well as from the YES and NO responses ignoring latency.
Real-Time Spectral Analysis of Aircraft Flyover Noise
Authors: Galloway W. J. , Horonjeff R. D. , Burns R. N.
Published: October 2005

The noise produced on the ground by a jet aircraft flying overhead is a signal whose amplitude varies by as much as 40 dB and whose duration may vary from 10 to 30 sec or more. Recent specifications for aircraft noise require a 12-oct band frequency analysis over the audible frequency range of this signal at each 0.5 sec interval of the time pattern. We describe a hybrid data analysis system used for this purpose which consists of a Hewlett-Packard model 8054-A Real-Time Spectrum Analyzer controlled by a DEC PDP-8 computer using a conversational language in the control program. The effect of analyzer time constants and various scanning rates on data repeatability and reliability are presented. These results are compared to those obtained with the more commonly used filter-graphic level recorder combination.
Preprocessing of Analog-Filter Outputs for Computer-Aided Data Acquisition
Authors: Maling G. C.
Published: October 2005

Experimental analog circuits for squaring, integrating, and taking the square root of an analog filter output have been evaluated with particular emphasis on the problems encountered with high crest-factor signals. The true root mean square value of signals having crest factors as high as 10:1 can be accurately determined, but higher (30:1) crest factors present problems because of d-c offset at the integrator input. The problem of crest factor variation with filter bandwidth can be handled by feedback from the computer used to acquire the data. The analog system is compared with a simulated digital system that samples the filter output at random time intervals, squares the amplitude, and accumulates the results. The digital approach appears to have several advantages provided that the number of samples taken is chosen to be large enough to obtain a statistically stable spectral estimate.
Interactions of Two Signals in a Bandpass Smooth Limiter
Authors: Lee J. S. , Hughen J. H.
Published: October 2005

The performance of a bandpass smooth limiter represented by the error function is analyzed when the input is a sum of two signals. Two input cases are considered: sine wave plus sine-wave interference and sine wave plus narrow-band Gaussian noise. Corresponding to various input ratios, output ratios of inband signals are obtained and displayed graphically as a function of a parameter q that specifies the degree of the limiter hardness. One of the graphic results is the behavior of the “captured limiter” as the limiter hardness is decreased. The error function characterization of the limiter is believed to be realistic since it is capable of describing a wide range of practical operating characteristics. In the limits, when q = 0 the smooth limiter becomes a linear amplifier while q = ∞ corresponds to an ideal hard limiter. The results corresponding to q = ∞ are shown to agree with those known for the ideal hard limiter. The significance of the various output ratios is discussed with the awareness that the limiter is often used in signal processing applications.
Calculation of Annoyance Level for Sounds Containing Multiple Pure Tones
Authors: Wells R. J.
Published: October 2005

At the last meeting, a new method for computing the annoyance level (ANL) of steady-state noise was presented. This method is basically similar to the calculation of perceived noise level, but different in certain details. For example, puretone corrections are based upon the concept of an effective critical bandwidth. It was noted that a modification in the procedure was necessary for cases where the pure tone occurs at a frequency differing from that of the peak of the broadband noise. Recently, in making comparison of ANL calculations with subjective judgments for spectra containing strong multiple pure tones spanning several octaves, it was observed that this secondary “off-peak” correction was not necessary, owing to the fact that maximum noisiness occurred in the MPT frequency range though the peak of the broad-band noise was elsewhere. A variation in the method of combining the annoyance of broad-band and tonal components of noise has been devised that eliminates the need for special consideration of “off-peak” cases. This procedure may be applied directly to MPT spectra without further modification.
Possible Modifications to Procedures for the Calculation of Perceived Noisiness
Authors: Kryter K. D.
Published: October 2005

An examination is made of three sets of data pertaining to paired-comparison judgment tests of the perceived noisiness of, primarily, aircraft noise: (1) 17 studies for which perceived noise levels (PNLs) calculated from peak of maximum sound-pressure levels of 143 noises are available; (2) a field study for which effective perceived noise level (EPNL is equal to PNL integrated over the duration of a noise) as well as maximum and peak PNL of 18 different aircraft noises are available; and (3) a study for which PNL and EPNL of impulsive noise (sonic booms) and the noise of subsonic aircraft are available. From these data, it is concluded that: (1) the perceived noisiness of aircraft sounds can best be predicted by combining 13- or 1-oct-band sound-pressure levels below 355 Hz in certain ways (to account for the critical bandwidth of the ear) prior to the calculation of PNL or by the use of a sound-level meter with a frequency weighting herein designated as D2; (2) time-integrated measures of sound are significantly better predictors of judged perceived noisiness than are so-called max or peak sound measures; and (3) corrections should probably be applied to EPNL of (a) nonimpulsive sounds of different onset duration and of (b) impulsive sounds whose level increases more than 40 dB in 0.5 sec. [Work supported by NASA.]
Loudness and Impulsive Noise
Authors: George D. L. , Eibner J. A.
Published: October 2005

A commonly accepted criterion for noise control is a family of curves known as the NC or N curves. This criterion was based upon a study by Beranek relating speech interference levels and loudness levels to subjective limits of acceptibility within various environments. Impulsive noise is sufficiently different from ordinary background noise so that these relationships no longer apply. The difficulty of using present-day limits for other than typical background noise is that, if the spectral shape of the noise does not follow the NC contour, there is no way of relating measurable parameters to subjective annoyance factors. Previous studies have shown that loudness limits based upon the relative loudness of impulsive noise as subjectively compared to random noise leave something to be desired. The purpose of this paper is to present certain assumptions and related subjective data showing the correlation between measured noise levels and subjectively evaluated “noisiness” for both continuous and some types of impulsive noises. Various noise-measuring techniques were evaluated in an effort to find out which method would provide consistent results regardless of the type of noise applied. The results presented in this paper can be used as a basis for future work on establishing noise limits for office, data processing, and even factory-type impulsive noise environments. Characteristic responses of present acoustical instrumentation are discussed and compared to the subjective noise evaluations.
Further Investigation into the Relationship between the Doppler Shift and Subjective Noise Levels
Authors: Vogel J. W. , Blakney D. F.
Published: October 2005

In a previous paper [J. Acoust. Soc. Amer. 46, 99 (A) (1969)], the authors discussed the spectrum changes that result when the noise source is in motion relative to an observer and the effects of these changes on perceived noise level (PNL). In the present work, the analysis is extended with consideration of two sources, pure jet and turbofan aircraft, to include the influences of wind, angle of attack of the aircraft, and atmospheric absorption coefficients. The results are presented in terms of PNL, and the significance of the various parameters is discussed.
Cessna Model 150 Airplanes and Damage-Risk Criteria
Authors: Flugrath James M.
Published: October 2005

Noise inside ten 1969 Cessna model 150 airplanes was measured by C-weighted and A-weighted sound levels and octave-band levels 31.5–8000 Hz, during runup, climb, and level flight. The variability between the cockpit noise levels of the airplanes was quite small although a slight correlation was noted between noise level and the total number of hours the airplane had been flown. Damage-risk criteria were not exceeded unless pilot exposure was over 30 h/week. To corroborate these results, the hearing level of five instructors was checked, and no acoustic trauma attributable to flying the Cessna 150 airplane was found.
Noise Hazard Meter
Authors: Botsford James H. , Laks Barry R.
Published: October 2005

A small electronic instrument has been developed to appraise noise exposures in terms of the hazard to human hearing. An analog computer integrates the A-weighted sound level of the noise over the exposure period according to the laws describing growth and recovery of temporary threshold shift [see J. H. Botsford, “Theory of TTS,” J. Acoust. Soc. Amer. 44, 352 (A) (1968)]. The result is a meter reading that indicates the average temporary loss of hearing that the exposure would produce in a group of normal ears. Since this average temporary hearing loss is roughly proportional to the permanent hearing loss that would result from many repetitions of the same exposure, the meter is valuable for appraising noise hazard. All noise exposures that indicate little or no temporary loss are probably safe. The new instrument can be worn by a roving workman and permits evaluation of many exposures to irregular noise that are unassessable by present methods.
Attenuated and Nonattenuated Conditions of Speech-Interference Encountered in Aerospace Vehicles
Authors: Gasaway D. C.
Published: October 2005

Aeromedical personnel are confronted with the task of assessing degrees of speech interference that are routinely encountered in a variety of aerospace vehicles. Six indices of speech interference are reported for both protected and unprotected exposures that are found in 249 fixed- and rotary-wing aircraft. The influence of noise spectra on the assessment of degrees of speech interference as determined by four octave-band averaging and two frequency-weighting methods is identified and described. Noise attenuation provided by headset devices commonly used by Air Force air-crew members is shown for different groups of noise spectra, and equivalent indices of speech interference are identified and demonstrated for different phases of airborne operation.
Underwater Explosion Signals Measured at Long Ranges
Authors: Hershey J. B.
Published: October 2005

Pressure-time characteristics of underwater-explosion signals have proven to be broadly useful in scientific study of the ocean floor and of the ocean itself. The general nature of the modifications to explosion signals by transmission via different paths through the ocean and the ocean floor will be discussed.
Measurements of Source Characteristics of the Shock Wave from Small TNT Charges in the Deep Ocean
Authors: Maples L. C. , Thorp W. H.
Published: October 2005

Shock-wave spectra for 3-lb charges of TNT at 60, 500, and 1000 ft received simultaneously on a ship-suspended hydrophone at 4200 ft and one bottom moored at 14 000 ft have shown excellent consistency for the 13-oct bands centered between 80 and 1000 Hz. The analysis of shock-wave characteristics alone was made possible through a technique employing the Adage Ambi-log System by digitizing the entire wave train at a 30-kHz rate, retaining only the desired portion while replacing the deletions with appropriate noise samples. Edited wave trains are then desampled onto analogue tape for analysis with the 12-oct band filters. Much of the observed variation in shot-to-shot detail was found to stem from the relatively low-amplitude negative pressure phase between shock wave and first bubble pulse. The mean spectrum from 75 charges show outstanding agreement between the two measurement systems but is displaced about 4 dB from a Weston-derived spectrum and the measurements of Stockhausen. These deviations are not contained by the experimental variance, and, at this stage of the continuing investigations, no explanation has been found.
Signal Processing for the Detection of Acoustic Transients
Authors: Bartram James F.
Published: October 2005

There is a variety of possible ways of classifying sounds. One approach, which is explored in this paper, is to use the first-order probability distribution. Specifically, a mathematical model is constructed, consisting of a family of first-order probability distributions with a parameter called the “distribution index,” with which to select a particular member of the family. It is then assumed that the set of real acoustic sounds can be described by this model and characterized by the distribution index. A signal processor is derived, based on the distribution-index concept. Finally, this signal processor is tested with real input sounds and is found to respond characteristically to ambient noise, sinusoidal signals, and sharp transients, as is desired.
Noise Generated by a Descending Hydrophone
Authors: Dahlke H. E. , Kidani R. J.
Published: October 2005

Experimental and theoretical studies have been conducted to investigate the major sources, i.e., the boundary-layer flow noise and the acceleration noise, of a descending, nonacceleration compensated hydrophone. The tests, using a 2-in. diam, 36-in. long cylindrical body with a hemispherical nose section, were performed in the velocity range of 3–7 ft/sec, and analyzed in the frequency range from 30 to 500 Hz. In the velocity range investigated, the Reynolds number at the location of the acoustical element of the hydrophone lies below its critical value, and the flow is considered as laminar. The measured flow noise is due to radiation from the turbulent boundary layer. At a sinking rate of 5 ft/sec, the results indicate a flow-noise spectrum level at 100 Hz of −20 dB re(1μbar)2/Hz, with an increase of 11 dB for double speed. These data are in agreement with results obtained by others. Surface roughness of the body has a negligible effect on flow noise at velocities less than 5 ft/sec. At higher velocities, the effect of roughness on the noise is greatly increased. The drop tests show clearly that the acceleration sensitivity of the hydrophone presents a serious problem in the low-frequency range. Any slight instability of the descending hydrophone raises the noise level more than 10 dB.
Automatic Estimation of Formant Frequencies for Voiced Speech
Authors: Schafer R. W. , Rabiner L. R.
Published: October 2005

A system has been developed for automatically estimating the lowest three formants and the pitch period for voiced speech. The cepstrum is used to obtain an estimate of the pitch period and the spectral envelope. The algorithm for estimating the formants from the spectral envelope incorporates constraints on formant-frequency ranges and relative spectral levels. When two formants are close together and are found to be unresolved in the initial spectral envelope, a new spectral analysis algorithm is used to resolve the two formants. The formant and pitch-period data are used to control a digital formant synthesizer. Examples of the synthetic speech produced from the estimated formant and pitch data are presented.
Articulatory Timing in Speech Production; Articulatory Distinctive Features
Authors: Coker C. H.
Published: October 2005

Controversy over “distinctive features” of phonemes—either conceptual, perceptual, or articulatory—has continued for a number of years, mainly for lack of definitive experiments. A feature hypothesis based on articulatory constraints is testable, however, through speech synthesis. In a synthesis-by-rule system based on an articulatory model (Coker; Boston, 1967; Kyoto, 1968), several of the model coordinates found useful for simplicity of synthesis rules correspond to distinctive features in current literature. Target values for most model coordinates are, indeed, discrete valued, either binary or ternary. Additional information is required for these discrete coordinates, however. Since articulator transition times are slow compared to phoneme durations, it is necessary, for even marginally acceptable speech, to stagger transition times of coordinates according to a scale of priorities for phonemes. This important/unimportant (“marked/unmarked”) distinction seems to be more multivalued and dialect dependent than do the target values. Moreover, there are asymmetries in the priority list, with respect to the beginning and end of the phonemes. The feature set implied by the present synthesizer rules is discussed. The results of this set of rules and the consequences of alternatives are demonstrated.
On Vowel Duration and Pitch Prominence
Authors: Coker C. H. , Umeda N.
Published: October 2005

The phenomenon of strong vowel lengthening when followed by voiced consonants does not occur uniformly in American English but only in rather special circumstances. Vowel lengthening is always accompanied by a rather prominent change in fundamental frequency during the vowel and the following consonant. The “pitch” change can be either a drop associated with stress, a continuation rise, or a final drop. The amount of lengthening varies with the complexity of the pitch inflection. The most striking elongations, for example, occur in the combination of falling stress and a continuation rise: the so-called fall-rise intonation. Not all such pitch patterns, however, cause vowel lengthening. The terminal pitch change is, in general, spread from the last primary or secondary-stressed syllable to the end of the breath group or phrase. Lengthening occurs only if the entire change is forced to fall within one syllable. When this occurs, vowel lengthening is greatest when followed by voiced consonants, but some lengthening occurs with unvoiced consonants, and in final vowels.
All-Digital Speech Compression Device
Authors: Houghton R. , Phyfe D. L.
Published: October 2005

An all-digital speech compression device was developed that is superior in quality, performance, size, and weight to 2400 bits per second speech compression devices. The device is compatible with conventional present-day equipments and uses commercially available multifunction integrated circuits and MOSFET circuits to implement the unit. The unit performs a digital mathematical transformation that closely resembles the channel vocoder transformation (but significant differences in the transformation enhance quality) and gives clues to an improved mathematical model of speech generation and perception. The transformation performed is classical in form and permit wide-band, 30-channel “quality applications,” and possesses powerful formant-detection potential to permit very narrow-band applications.
Multidimensional Analysis of Similarity Judgments of Multicomponent Tones
Authors: Copperman N.
Published: October 2005

The relationship between relative amplitude differences of the components of the four-component tones and their perceived difference was studied by multidimensional scaling. All possible pairs of 12 stimuli were presented to three subjects, who were instructed to rate each as either “rather similar” or “rather different.” The components were sine waves of constant frequency, and the sum of their intensities (voltages) was constant. A nonmetric multidimensional analysis of the data showed that the relationship between distance to criterion (d′) and nometrically scaled distance between observations was linear; thus, it was possible to scale the distance between observations in d′ units. In a second experiment, the signal power was successively doubled in four conditions. The average distance between observations did not increase systematically with signal power, and the discriminal dispersion could not be attributed to noise in the “input channels.” The average angle between response dimensions was 87° indicating that a response to one component was minimally affected by energy differences in the others. Subjects were most sensitive to the highest component but showed marked individual differences in their sensitivity to the others.
Comparison of Sorting and Pairwise Similarity Judgment Techniques for Scaling Auditory Stimuli
Authors: Bricker Peter D. , Pruzansky Sandra
Published: October 2005

Twenty subjects used a sorting apparatus to form hierarchically related clusters, based on similarity, of 24 three-parameter frequency-modulated tones (the HC task; duration: 15 min). Another partially overlapping group made judgments of similarity of all 552 ordered pairs of the same tones (the PC task; duration: 60 min). Both sets of data were analyzed by a new multidimensional scaling method (due to J. D. Carroll and J-J. V. Chang) that accommodates individual differences in weights assigned to the unique dimensions of a group space. The structure based on PC data recovered the generating parameters (modulation frequency, deviation extent, and modulation waveform) and revealed some interaction but little scaling or intersubject difference. Stimuli were less uniformly distributed in the HC structure, and subjects differed considerably, both in the scaling of modulation frequency and in the assignment of weights to dimensions. HC and PC data from the same individual are not equivalent; each provides its own type of information about perceived similarity. However, averaging HC data over subjects and scaling by the Shepard-Kruskal technique produces a structure that can be rotated into close correspondence with the PC structure.
Two-Click Thresholds in Presence of Band-Stop Noise
Authors: Hall J. L. , Lummis R. C.
Published: October 2005

The mechanism of auditory transduction was studied by measuring the auditory threshold of a pair of clicks in the presence of band-rejected masking noise. This threshold depends on the interval between the two clicks. As the interval is changed, the threshold intensity alternately increases and decreases, with maxima at (2n+1)/2f sec (n=0, 1, 2, ⋯) for masking noise, with a spectral gap centered at f Hz. Experimental determinations of this fluctuation are in quantitative agreement with predictions of a simple energy-detection model. In the model, band-stop noise produces a band-pass filtering of the click stimuli, and the click pair is detected if its filtered energy exceeds some threshold value. Small discrepancies between experimental data and predictions of the model are observed when the two clicks are of unequal intensities.
Sensory Interaction in Signal Detection
Authors: Fidell Sanford
Published: October 2005

Five observers detected a sinusoid in noise in a two-interval forced-choice experiment. The signal occurred on an earphone, on an oscilloscope, or on both devices simultaneously. Detection performance was studied as related to (1) mode of occurrence of the signal(s); (2) the external noise correlation in the auditory and visual channels; and (3) the observers' a priori knowledge of the mode of occurrence of the signal. The observed improvement in sensitivity (measured in d′ units) as a function of bimodal signal presentation closely followed the predictions of a statistical summation model, and was much lower than predicted by linear and probabilistic addition models. Under conditions of independence of noise in the auditory and visual channels, some improvements in sensitivity were almost as great as 3 dB. The improvement in sensitivity afforded by a priori knowledge of the mode of occurrence of the signal was less for bimodal signals than for unimodal signals.
Rotational Relaxation in Polar Diatomic Gases
Authors: Evans L. B. , Winter T. G.
Published: October 2005

Ultrasonic-absorption and velocity-dispersion measurements were made on HCl and HBr from room temperature to 1073°K over the frequency range of 1 MHz/atm–100 MHz/atm. The data were analyzed by both single relaxation theory and Gordon's multiple relaxation theory, and the results were compared with the predictions of Zeleznik's theory for rotational relaxation in polar gases. It was found that (1) is was necessary to include the frequency dependence of the specific heats in the calculation of the classical absorption; (2) the apparent relaxation frequency in HCl shifted from 300 to 30 MHz/atm as the temperature was increased from 298° to 793°K and in HBr from 100 to 25 MHz/atm as the temperature was increased from 298° to 773°K; (3) the temperature dependence is in excellent agreement with the prediction of Zeleznik's theory and the dipole moment and moment of inertia dependence agrees qualitatively; (4) Gordon's theory fits the data as well as single relaxation theory in each case, much better in several cases, and yields collision numbers that are about half the single relaxation-theory collision numbers.
Measurement of Absorption and Velocity of Ultrasound in Liquid Alkali Metals
Authors: Kim M. G. , Kemp K. A. , Letcher S. V.
Published: October 2005

Ultrasonic absorption and velocity measurements have been made in pure liquid alkali metals at temperatures from their melting points to 250°C. Measured absorptions have been found to exceed the computed classical values by 10%–25%. Sound velocities decrease linearly with increasing temperature. The distilled liquid metals were transferred to the measurement apparatus under vacuum (better than 10−4 Torr) and measurements were made under a high-purity argon cover gas. No significant reaction with the cover gas was observed. No bonding agent was needed between the transducer and the clean liquid surface. [This work was supported by the Office of Naval Research.]
Ultrasonic Relaxation in Alcohol Water Mixtures
Authors: Madigosky W. M.
Published: October 2005

Additional measurements of the excess ultrasonic absorption have been made in binary mixtures of 1-propanol with water and tetriary butyl alcohol with water. Contrary to earlier expectation the relaxation spectra is much broader than that previously reported. Measurements of α/f2 did not approach a limiting low-frequency value even for frequencies as low as 1.0 MHz. The theory of Romanov and Solovyev, which assumes that the excess absorption is due to concentration fluctuations, still seems to predict the observed broad frequency dependence. Further low-frequency absorption measurements are being undertaken using the resonant-cavity technique and preliminary results are discussed.
Ultrasonic Absorption in Aqueous Solutions of Globular Proteins
Authors: O'Brien William D. , Dunn Floyd
Published: October 2005

The ultrasonic absorption and velocity were measured in aqueous solutions of bovine hemoglobin, a globular protein, which undergoes marked configurational change with pH, at 10°C over the frequency range 0.4 MHz to 50 MHz and over the pH range 1.5–13.5. As the pH is varied outside the range 5<pH<9, the excess absorption increases sharply, this effect being more pronounced at lower frequencies. Similarities in results between bovine hemoglobin and another globular protein, bovine serum albumin [L. W. Kessler and F. Dunn, J. Phys. Chem., to be published], are discussed, and a possible cause is proposed that accounts for these similarities. [This work is supported by the Institute of General Medical Sciences, National Institute of Health.]
The Barrier As a Sound Isolation Device
Authors: Hyde Jerald R.
Published: October 2005

Scale-model techniques using an omnidirectional ultrasonic noise source and receiver are employed in order to increase the effectiveness of barriers in reducing noise levels between adjacent spaces. In the general case of an infinitely long barrier, results are compared with several theoretical formulas for the noise reduction of the barrier. Experimental results are also given for the noise reduction of a barrier as a function of its aspect ratio (length versus height), its surface absorption, and ground absorption. Correction factors applicable to general theoretical expressions are given, which allow the estimation of noise reduction for the cases presented.
Increasing Aircraft Noise Reduction by Practical Modifications of Typical Home Construction
Authors: Pecora Salvatore D.
Published: October 2005

A three-phase modification program was conducted on two houses in the vicinity of Los Angeles International Airport. The purpose was to develop and test construction modifications that would improve the aircraft noise reduction provided by typical houses. The modifications were chosen on the basis of existing knowledge, recent laboratory tests, and practicality for home construction. An increase of 10–20 dB of A-level noise reduction over existing construction was the design goal for modifications. The paper discusses the construction modifications and the conclusions drawn from the experimental program. [The program was supported by the Department of Housing and Urban Development.]
Reflective Surfaces for the Hearing of Speech and Music in an Anechoic Chamber
Authors: Walker B. E. , Delsasso L. P. , Knudsen V. O.
Published: October 2005

Speech articulation tests in our anechoic chamber, with and without the benefit of three reflective surfaces (to be described) for reinforcing the speech, reveal that the percentage of typical speech words heard correctly is 93.3% with the reflectors and only 69.7% without them. The directional characteristics of speech around the head of the speaker were consistent with those for tones of different frequencies; thus, the sound-pressure level of a 13 oct band of speech, centered at 4000 Hz, dropped 14 dB when the speaker turned 180° away from the recording microphone. The results obtained in the anechoic room, without the reflectors, are consistent with those we obtained 40 years ago at a noise-free location on a dry lake bed. The results are demonstrated with data and tape recordings, which reveal qualitatively and quantitatively the beneficial effects of suitable reflectors. The effects of early reflections on the quality of music are also demonstrated by tape recordings. These effects may be as important as is optimal reverberation time.
Acoustic Properties of Metals and Magnetostrictive Materials
Authors: Whymark R. R. , Lester William W.
Published: October 2005

The process of inducing plastic flow in metals under the combined action of static and large vibratory stresses using a high-powered horn driver is discussed. In particular, where the metal is ferromagnetic, one has the situation that many properties of the material, including magnetic properties, change under the combined stress action. This necessarily affects the end uses of the material as, for example, in magnetostrictive transducers. Experiments are described in which the properties of nickel are investigated, for example, and the plastic flow threshold curve under combined stress action is determined. It is found that vibratory stress is much more effective in producing flow than static stress. In addition, it is found that, subsequent to the occurrence of plastic flow, the mechanical Q of a sample is altered but partially recovers over a period of hours at room temperature, and the material is hardened against further flow. The separation of the total energy loss into macroeddy, elastic, and microeddy loss is discussed. The alteration of the magnetic properties of plastically flowed samples is quite marked at points near the flow maxima in the samples containing standing intense acoustic waves, and the change in internal friction provides a possible means for predicting impending fatigue in metals without destruction of the materials. [This work was supported by the Office of Naval Research.]
Acoustic Waves Transmitted through Elastic Cylinders
Authors: Brill D. , Überall H.
Published: October 2005

Besides the circumferential waves [R. D. Doolittle, H. Überall, and P. Uginčius, J. Acoust. Soc. Amer. 43, 1–14 (1968)], transmitted waves play an important role in the diffraction of sound waves from an elastic cylinder. Extending our results for liquid cylinders as reported previously [D. Brill and H. Überall, J. Acoust. Soc. Amer. 46, 116–117 (1969)], we have derived a generalized vander Pol-Bremmer expansion of the scattering amplitude from an elastic cylinder that mathematically describes the following: (a) an incident compressional wave gets refracted into the cylinder, thereby splitting into a compressional and a shear wave; (b) each of these either re-emerges from the cylinder as a compressional wave; or (c) gets reflected back into the cylinder, thereby each splitting again into a compressional and a shear wave; (d) this process repeats ad infinitum. [Supported by Project Themis in Under-Ocean Acoustics at Catholic University.]
Diffraction of a Sound Pulse by a Hollow Cylinder in Water
Authors: Neubauer Werner G. , Dragonette Louis R.
Published: October 2005

Motion pictures will be shown for a hollow cylinder similar to those shown for a solid cylinder at the 78th Meeting. Schlieren visualization will be shown of a 5-μ sec sound pulse at 6.6 MHz in a time sequence as it is emitted from a source, strikes a cylinder, and produces the various radiated constituents of the diffracted field—i.e., those caused by the creeping wave—and the transmitted refracted longitudinal and shear waves and their reflections in the cylindrical shell. The sequence of production of the constituent waves in the diffracted field at fixed times are also shown as the acoustic pulse is swept across the cylinder, normal to its axis. Shells have inner radius to outer radius ratios of from 0.6 to 0.993 and size parameters (ka, where k = 2π/λ and a is the outer radius) from 443 to 1419.
Transient Radiation from Pistons in an Infinite Planar Baffle
Authors: Stepanishen P. R.
Published: October 2005

An approach is presented to compute the near- and farfield transient radiation resulting from a specified velocity motion of a piston in a rigid infinite baffle. The approach, which is based on a Green's function development, utilizes a transformation of coordinates to simplify the evaluation of the resultant surface integral. A simple expression is developed for an impulse response function, which is the time-dependent velocity potential at a spatial point resulting from an impulsive velocity of a piston of any shape. The time-dependent velocity potential and pressure for any piston velocity motion may then be computed by a convolution of the piston velocity with the impulse response. Several examples illustrating the usefulness of the approach are then presented. The familiar steady-state nearfield on-axis pressure fluctuations resulting from the sinusoidal velocity of a circular piston is derived from the impulse-response data. The farfield time-dependent radiation from a rectangular piston is also discussed for both continuous- and pulsed-velocity conditions. For a pulsed velocity of time duration T, it is shown that the pressure at several of the field points can consist of two separate pulses of the same time duration.
Time-Domain Method of Computing Mutual Radiation Impedance between Pistons in a Rigid Infinite Planar Baffle
Authors: Stepanishen P. R.
Published: October 2005

A time-domain method of computing the mutual radiation impedance coefficients between pistons in a rigid planar baffle is presented. The method is based on the use of a Green's function approach to solve for the time-dependent interaction force on a piston resulting from the velocity of an adjacent piston. The approach may be used to study the interaction forces resulting from both sinusoidal and nonsinusoidal velocity pulses. Since the time-dependent problem is treated initially, the method may also be used to study the transient interaction force phenomena occurring in arrays of pistons with specified spatial and temporal velocity distributions. Several examples are presented to illustrate the application of the approach to compute the impedance coefficients. The results are shown to be in close agreement to those of previous investigators.
Passive Acoustic Beamforming with Spherical Liquid Lenses
Authors: Fisher B. , Brigham G. A.
Published: October 2005

The acoustic field within a spherical liquid lens due to plane-wave excitation is numerically computed by evaluation of the Helmholtz integral equation and compared to the exact boundary value solution over a lens diameter to wavelength range of 2 to 25. It is shown that rear hemisphere reradiation dominantly accounts for the computed differences, nearly exclusively affecting only the front-to-back spatial discrimination ratio. Lens focal radius as estimated by (1) minimum residual of the aperture phase and (2) ray (energy) considerations is compared and shown to be in excellent agreement with the computed field. Sidelobe and pressure gain degradation due to geometric aberration is evaluated and shown along with experimental data. Normalized beam patterns are presented showing this effect to be a negligibly weak function of refractive index. Finally, the lens passive sensitivity is developed and shown to be superior to an array of the same directive index.
Rapid Numerical Procedure for Solving Acoustic Radiation Problems
Authors: Martenson A. J. , Liu H. K.
Published: October 2005

A method for solving a very general class of radiation problems is given which is a variation of the integral formulation used by previous workers. A Fourier series is used to represent a continuous distribution of acoustic sources over the surface of the boundary. The values of the coefficients of the series then become the unknowns of the problem. This approach results in reduced computer times and therefore permits the solution of problems where the boundary dimensions are many wavelengths. Another advantage is that it offers a convenient way to avoid the problems at characteristic frequencies that have been encountered in the past. Comparisons with published solutions of classic acoustics problems show that the technique is accurate.
Test of the Capabilities of CHIEF
Authors: Van Buren Arnie Lee
Published: October 2005

The computer program CHIEF was designed by Schenck [J. Acoust. Soc. Amer. 44, 41–58 (1968)] to obtain approximate solutions to exterior steady-state acoustic radiation problems for arbitrary shape surfaces whose normal velocity is specified. To test its capabilities as a research tool, CHIEF was used to obtain the acoustic radiation impedance of caps and rings on oblate spheroids. The radiation impedance was also obtained using a harmonic expansion in oblate spheroidal functions. Improved FORTRAN computer programs were developed to evaluate the necessary spheroidal functions accurately. The accuracy of the CHIEF results are discussed in terms of acoustic size, the choice of surface subdivision schemes, and execution time on the CDC 3800 computer at NRL. New input subroutines to CHIEF, which provide numerous geometrical options, are also discussed.
Specification of the Minimum Number of Glottal Pulses Necessary for Reliable Identification of Selected Speech Sounds
Authors: Moore Thomas J. , Mundie J. Ryland
Published: October 2005

A panel of listeners was required to identify a series of nine vowels. In the series each vowel was represented from one-half a glottal pulse up to five glottal pulses. The order of presentation was randomized across vowels and number of glottal pulses. The minimum number of glottal pulses necessary for reliable identification of each of the nine vowels was determined, along with a specification of what perceptual confusions occurred.
Types of Mechanisms for Hair-Cell Stimulation
Authors: Wever E. G.
Published: October 2005

The final step in the mechanical action of sounds on the ear is the production of relative motion between the ciliary tufts of the hair cells and the bodies of these cells. This relative motion is achieved in three principal ways, depending upon three types of accessory structures in contact with the ciliary tufts. In Type I (tectorial or restraint system) each ciliary tuft is connected with a membrane that is firmly anchored to an immobile part of the cochlear wall, and thus is restrained from moving when the cell body moves. In Type II (sallet or inertial system), each tuft is attached to the undersurface of a body that provides inertia and thereby limits the ciliary motion. In Type III (combined system), each tuft is attached to a massive body that provides inertia and also is anchored to the cochlear wall. All three types of stimulating systems are found in lizard ears in great variety. Other reptiles, and also birds and mammals, employ one or another of these systems. Illustrations will be given of these types of stimulating arrangements, with discussion of their special characteristics and the relative performances of the different ears in sound reception.