bluuebunny/crossref_metadata_2025 huggingface dataset:
Temporal Effects on Negative Threshold Shift
Authors: Melnick William
Published: October 2005

A small but significant negative threshold shift (decrease in hearing threshold) has been reported following particular sound exposures and has been called sensitization. Similar negative threshold shifts of 2 to 3 dB have been observed in experiments of similar design but with a period of quiet substituted for the sound exposure. This negative shift following what amounts to an interruption in threshold tracking was the dependent variable in this series of three experiments concerned with temporal independent variables. The test tone was a continuous tone of 500 Hz. Threshold was tracked using a recording attenuator. As the interrupting quiet period was lengthened from 1 to 4 min the magnitude of the negative shift increased from 2 to 4 dB. In a 14-min tracking session during which seven tracking periods were interrupted by 1-min quiet intervals, the negative shift was maximum at the outset but still was increasing at the end of the session. Finally, 1 h of isolation in a sound-treated room prior to experimental tracking reduced but did not eliminate the negative-threshold shift.
Monaural Detection with Partially Correlated Contralateral Cues
Authors: Sorkin R. D.
Published: October 2005

Trained observers detected monaural signals in a two-interval-forced-choice detection task in which the level of the continuously present monaural noise was controlled by a PEST program. Observers were run in a variety of monaural-signal-contralateral-cue conditions. The signal was a 120-msec duration pulse train composed of pulses of 100 μsec in width delivered to a headphone. Signal conditions included cases where the interpulse period was fixed or random over trials (nominally around 2 msec) and where the pulse polarity was either fixed at +1, or was randomly +1 or −1. The different cue conditions included cases where no cue was present, cases where the cue was identical to the signal that might be present, and cases where certain of the cue pulses were different from the signal. The resulting psychometric functions are interpreted in terms of the cross-correlation and spectral relationship between the signals and cues employed.
Effect of Masker Duration on the Detectability of Pulsed Diotic and Dichotic Tonal Signals
Authors: Miller Theodore H. , Trahiotis Constantine T. , Dolan Terrence R.
Published: October 2005

It has been reported [D. McFadden, J. Acoust. Soc. Amer. 40, 1414–1419 (1966)] that the magnitude of the advantage exhibited by the binaural auditory system over the monaural auditory system when detecting a pulsed tonal signal in a background of pulsed masking noise is partially determined by the amount of time by which the onset of the pulsed masker precedes the onset of the pulsed signal. The present study was intended to determine whether the magnitude of the binaural advantage could also be increased by extending the duration of the masker beyond the offset of the signal. In this masking condition the onsets of the signal and masker occur simultaneously, but the offset of the signal precedes the offset of the masker. The results indicate that the magnitude of the binaural advantage increases slightly as the masker extends beyond the offset of the signal. The results are discussed in reference to current explanations of binaural detection phenomena.
Binaural Masking Level Difference from Signal-to-Noise Measurement on an Analog Ear
Authors: Stewart J. L.
Published: October 2005

Signals at several locations along the basilar membrane of an analog ear are half-wave rectified to represent the neural equivalent of mechanical motion. Two such systems with simple signal addition represent the binaural case of principal interest here. In one experiment, a tone is mixed with noise and applied alike to the two ears. In a second experiment the noise, but not the signal, is inverted before application to one ear. Comparison of signal-to-noise ratios in these two experiments represents the so-called S0Nπ case with an estimate for BMLD which peaks at 12 dB at 400 Hz, falls off gradually at lower frequencies, and falls off more rapidly at higher frequencies. The extension of concepts to the SπN0 case is also considered where, in this case, a form of cancellation rather than summation may apply. Practical and theoretical implications of the model-demonstrated theory are discussed.
Interaction of Amplitude and Frequency Discrimination
Authors: Feth L. L.
Published: October 2005

Neural excitation-pattern models of the peripheral auditory system lead to the implication that simultaneous amplitude and frequency discrimination should exhibit strong interactions in the presence of band-limited masking noise. Earlier experiments have used signals simultaneously modulated in amplitude and frequency, and the results showed the predicted interactions. However, spectra of these modulated signals suggest another interpretation of these data based on spectral similarity. We report the results of 2IFC discrimination experiments which used unmodulated tones and bandlimited noise maskers. The excitation-pattern models predict similar interactions for these stimuli; but such interactions were not found, and we, therefore, conclude that these models need to be revised. [This research was supported by the National Institutes of Health, Public Health Service, U. S. Department of Health, Education, and Welfare, and while the author was a National Science Foundation Postdoctoral Fellow in the Department of Psychology, University of California, San Diego.]
Instrumentation for the California Airport Noise Standards
Authors: Partridge Gordon R.
Published: October 2005

California law requires measurement of both hourly noise level (HNL) and total daily noise exposure (Community Noise Equivalent Level, or CNEL) in the vicinity of airports. The 24-h total is determined by a formula which adds greater weight to evening and nighttime noise because of the additional annoyance experienced during those hours. An instrument is described for determining these noise levels. The hourly noise computation is performed by sampling, squaring, digitizing, and storing in memory a cumulative sum-of-the-squares of voltages proportional to the sound-pressure level. The 24-h total is obtained by additional pre-scaling, chosen to give the required evening and night weightings. The contents of both the HNL and CNEL memory registers are converted to decibels for the final output. Recently developed MOS integrated circuits make possible battery operation with very low average power drain. A novel printing head consisting of resistors deposited on glass “burns” the answers into thermochromic paper once each hour, with an additional printout at midnight for the daily CNEL. Sample measurements obtained near Logan Airport, Boston, are presented.
Prediction of Aural Detectability in Varying Noise Backgrounds
Authors: Fidell S. , Pearsons K. S.
Published: October 2005

The applicability of the theory of signal detectability to prediction of aural detection of noise signatures in varying noise backgrounds was assessed in two experiments. Ten observers participated in a preliminary two-interval forced-choice task administered by a computer based adaptive procedure. The purpose of the first experiment was to assess the detectabilities of 12 synthetic signals in three different noise backgrounds. Agreement between data and predictions based on an energy-detection model working on peak-signal-to-noise ratios in third-octave bands proved excellent over a 37-dBA range. A second experiment employed 10 observers to determine whether detectabilities of more complex synthetic signals could also be predicted as accurately. Detectabilitles of complex signals containing broad-band noises, pure tones, and both amplitude and frequency modulation were calculated by taking the square root of the sum of the squares of the detectabilities of individual components. Agreement between data and predictions was once again excellent. [This research was supported under Contract No. F33615-71-C-1220 with the Flight Dynamics Laboratory of Wright-Patterson Air Force Base.]
Development of Realistic A-Weighted Auditory Risk Criteria for Aerospace Operations
Authors: Gasaway Donald C. , Sutherland Harrell C.
Published: October 2005

Attempts directed at identifying potentially hazardous noise exposures have been vigorous. Unfortunately, unanimity of opinions relative to establishment of such criteria has not been achieved. The authors have previously proposed adoption of CHABA Work Group 46 criterion for steady-state-type noises for assessing degrees of auditory risk associated with aerospace operations. The authors review the salient features of various damage risk criteria and primary and secondary compromises are discussed. The authors, using noise data representative of ground and airborne operations, evaluate and critique the different criteria relative to degree of stringency imposed by each. A simple criterion using A-weighted sound levels is proposed for broadband and narrow-band steady-state-type noises.
Community Noise Measurements in Los Angeles, Detroit, and Boston
Authors: Bishop D. E. , Simpson M. A.
Published: October 2005

Quantitative descriptors of the noise environment have been obtained for 20 urban areas in Los Angeles, Detroit, and Boston. Measurements involved the acquisition of a 10-min noise sample once each hour for a complete 24-h day, and the subsequent analysis of each noise sample to obtain a statistical distribution of the A-weighted noise levels and the number and frequency of occurrence of discrete noise intrusions. We present the time patterns of various community noise descriptors, such as the 50% level, the energy mean, the Noise Pollution Level (NPL), and the Traffic Noise Index (TNI), for several of the areas sampled. These noise measures show considerable spread in average daytime and nighttime levels. When classified in three groups according to gross differences in exposure to motor vehicle traffic, distinct differences in daytime and nighttime noise patterns and in sources of discrete noise intrusions are apparent. The several noise measures (including the 1, 10, 50, and 90% A-levels, TNI and NPL), generally show high correlation among each other, with the exception of the TNI when it is calculated on other than a 24-h basis.
Measurements in Architectural Acoustics—Historical Perspective and Physical Principles
Authors: Ingard Uno
Published: October 2005

A brief overview of some fundamental physical principles is presented as a basis for adequate understanding and evaluation of measurements in architectural acoustics.
Privacy for Apartment Dwellers
Authors: Schultz Theodore J.
Published: October 2005

At the present time there exist U. S. standards for measuring the transmission loss of partitions and of floor/ceiling structures, both in the laboratory and in the field. Such measurements are, by definition, restricted to evaluating the attenuation interposed in only one at a time of the several possible paths by which sound may find its way from one location in a building to another. The ultimate concern, however, is to provide privacy for neighbors in multifamily dwellings, no matter how many sound paths may be involved. Current efforts in the U. S. to devise standard procedures for evaluating privacy are discussed. These are related to existing European practice and to the attendant technical and practical problems posed by field measurements.
Measurements in Room Acoustics—Recent Progress
Authors: Damaske P.
Published: October 2005

Computers are being used to refine reverberation theory. The shape of the enclosure and the actual distribution of sound-absorbing materials cannot be disregarded. The assumption of diffuseness, usually made to simplify calculations, must be dropped. For the measurement of energy decay, an impulse method, best realized by digital techniques, is suitably applied. It yields reproducible decay curves, especially for the early reverberation, which is important for the percepted “reverberancy.” A new omnidirectional sound-reproduction system was developed. Sound signals in the room are picked up by a dummy head with built-in microphones. The signals are then filtered in order to eliminate acoustical crosstalk, caused by conventional loudspeaker playback. Radiation of the filtered signals by only two loudspeakers yields a reproduction closely resembling natural transmission. The system is being applied to the direct acoustical comparison of European concert halls. With the dummy head, impulse responses of concert halls are recorded binaurally and stored in a digital computer. By convolution with these stored impulse responses, any desired signal can be made audible as if heard in any of the concert halls.
Measurements in Architectural Acoustics—Future Trends
Authors: Northwood T. D.
Published: October 2005

Measurements in architectural acoustics may be said to have begun with the work of W. C. Sabine, who used a collection of organ pipes, a stopwatch and his two ears to measure reverberation times and to develop the theory that bears his name. Since then several generations of instrumentation have been developed, each surpassing its predecessors in the amassing of information on the physical aspects of architectural acoustics. It is predicted (or at any rate, it is hoped) that the next decade or so will see an equivalent advance on the subjective side of acoustics. Then the physical data, which we are now so adept at measuring, will become relevant to the real-life problems of architectural acoustics. To this end the present status of “subjective” architectural acoustics will be reviewed and an attempt will be made to identify the lines of study most likely to bear fruit in the near future.
Electret Condenser Microphones for Measurement
Authors: Djuric S. V.
Published: October 2005

Condenser microphones using the electret principle have been developed with stable acoustic characteristics when exposed to high humidity and relatively high temperatures. At the same time they possess smooth frequency response characteristics and good sensitivity. Three different sizes have been made: a 1516-, 12-, and 14-in.-diam microphone. Two different types of 1516- and 12-in. microphones have been designed, one for flat perpendicular incidence free-field response and the other one for flat random incidence response. The 14-in. microphone is designed for flat perpendicular incidence free-field response. Acoustic characteristics of microphones are compared and the influence of environmental factors is discussed.
Analysis of a Piezoelectric Disk Microphone Sensing Element
Authors: Fritz T. C. , Thurston R. N.
Published: October 2005

The axially symmetric motion of a structure consisting of two thin piezoelectric ceramic disks bonded to opposite sides of a thin conducting spacer disk is analyzed. The optimum ratio of ceramic-disk-thickness to spacer-thickness is found, as well as the optimum electrode radius. The support is represented by two mechanical impedances: an impedance to translation and an impedance to tilting, both assumed uniformly distributed around the supporting circle. A good support should have very low impedance to tilting, but high impedance to translation.
Solid-State Hydroacoustic Image Converter
Authors: Knoll-man G. C.
Published: October 2005

A solid-state, linear-array piezoelectric acoustic image transducer is described which has been developed for realtime underwater viewing, especially in turbid and/or turbulent oceanic environments. Sensitivity and resolution of the acoustoelectric converter are on the order of 10−11 W/cm2 and 1 mm, respectively, at a nominal frequency of 2.5 MHz. Imaging range up to 10 m is possible in turbidity concentrations of suspended ocean sediment (average particle diameters of from 1 to 10μ) as large as several thousand parts per million. Expected ranges are at least five times larger in clear water. Fabrication and performance of the transducer array are discussed. Image-scanning mirror, internal and external electronic circuitry, and the image display, all associated with the hydroacoustic converter, are also described. [Supported by the Lockheed Independent Research and Development Programs.]
New Portable Sound Analyzing System
Authors: Kundert W. R. , Marteney E. R.
Published: October 2005

An on-site or field analysis of a noise is usually preferred to a measurement in a laboratory if it can provide a sufficient amount of data for the given purpose. It may also be desirable to record the noise being analyzed for reference and possible further analysis at a later time. Field equipment must be rugged, small, lightweight, and, above all, very convenient to set up and operate. With these thoughts in mind, a new portable sound analysis system has been developed. The system consists of three major components: a precision sound-level meter and analyzer, a cassette data recorder, and a power supply and charger. The paper will describe the data recorder and charger briefly with emphasis on the unusual features of the precision sound-level meter and analyzer. Chief among these is the inclusion in one package of both weighting networks and octave filters, an extendible microphone mast, an impulse detector in addition to the usual “fast” and “slow” detectors, a unique automatic-rangechanging feature that obviates the dual or coaxial range controls found in other sound analyzers, a detected (or dc) output with 40-dB dynamic range, small size, and weight under 6 lb.
Effect of Context on Vowel Recognition by Dispersion Analysis
Authors: Li K. P. , Hughes F. W. , House A. S.
Published: October 2005

A decision tree to place sounds into 10 vowel categories has been constructed in a manner described earlier [Li, House, and Hughes, J. Acoust. Soc. Amer. 44, 390(A) (1968)]. Training data consisted of spectral samples from the central portions of the stressed vowels in 325 utterances spoken by two talkers. The process independently arrived at the same decision tree for both talkers. To test the effects of context on vowel separation using such a tree together with its separating hyperplanes, three additional sets of utterances were processed without further training: (1) new repetitions of portions of the hoə'CVC list, (2) a list of English CVC words, and (3) connected speech containing the CVC words from (2). The errors (7%) made on material essentially identical to the training data can be interpreted as inherent in this method; the errors (11%) made on the CVC words can be attributed to consonantal environment; errors made in continuous speech are mainly due to stress reduction. [This research was supported in part by the Air Force Cambridge Research Laboratories.]
Aural Identification of Children's Voices
Authors: Cort S. , Murr T.
Published: October 2005

Recent studies have emphasized the importance of aural identification of speakers; this study sought to assess children's ability for aural-identification purposes. Specifically, the purpose of this study was to determine if children can make aural identification of their peers and, secondly, to determine if three repetitions of the sample increased identification scores. Fourth-grade children were divided into two groups of 10 subjects, 5 male and 5 female in each group. Children recorded a paragraph, sentence, and sustained vowel which were randomized and later played back to each group. Group 1 heard the samples once; group 2 heard them three times. The children were instructed to write the name of the speaker. The results indicate that children can identify peers from a group of 10. Identification increased for both groups as the repertoire of the sample increased from vowel to sentence to paragraph; however, three repetitions of the same sample-type did not increase identification significantly. The results suggest that repertoire rather than actual duration of the sample provided cues to the children.
Mimic Resistance of Speaker Verification Using Phoneme Spectra
Authors: Rekieta T. W. , Hair G. D.
Published: October 2005

The susceptibility to mimicry of the speaker-verification technique discussed in the paper “Automatic Speaker Verification Using Phoneme Spectra” was evaluated with the assistance of a professional performer specializing in impersonations. Subjects to be mimicked were six speakers selected from the previously recorded population. After becoming familiar with each speaker, the impersonator was permitted to mimic the speaker immediately after his utterance of each selected word. The subjects' utterances and the impersonator's mimic attempts were then processed for speaker verification on a single-phoneme basis and using combinations of several phonemes. Spectral analysis of individual phoneme segments revealed that some increase in similarity was accomplished by the mimic for certain speakers and phonemes. However, when the verification procedure was applied using features from five phonemes, the impersonator was unsuccessful in all mimic attempts.
Test of an Automatic Speaker Verification Method with Intensively Trained Professional Mimics
Authors: Lummis R. C. , Rosenberg A. E.
Published: October 2005

Several professional mimics selected by audition from a large group, were given intensive training on utterances of the eight “customers” in Doddington's 40speaker population [J. Acoust. Soc. Amer. 49, 139 (A) (1971)]. The training method uses an interactive DDP-516 computer to provide the mimic with immediate playback of his practice utterances as well as immediate A-B comparison between his own and customer utterances. Recordings of the best utterances from the best four mimics were processed by the computer verification system described previously [Doddington, op cit., and R. C. Lummis, J. Acoust. Soc. Amer. 50, 106(A) (1971)]. The system uses five features for verification: pitch, level, first, second, and third formant frequencies. If the acceptance-rejection criterion that yields equal-error performance in Doddington's speaker population is used for the mimics, 27% of the best utterances by the best mimics would be accepted. This acceptance rate compares to 1.2% for nonmimicking impostors. Extensions to the verification procedure are being studied that will make successful mimicry more difficult.
Listener Performance in a Speaker-Verification Task with Deliberate Impostors
Authors: Rosenberg A. E.
Published: October 2005

In a previous experiment [Rosenberg, J. Acoust. Soc. Amer. 50, 106(A) (1971)] listener performance was evaluated in a speaker verification task in which 32 “casual” impostors were pitted against eight “true” speakers. A “casual” impostor is one who makes no attempt to mimic the “true” speaker but simply repeats the same test utterance in his own natural voice. In the present experiment four professional mimics were engaged to deliberately imitate each of the eight “true” speakers. After intensive training their recorded utterances were used in an experiment in which 10 listeners participated. Each stimulus presentation was a paired comparison consisting of a challenge and a reference utterance. The reference utterance was one of the “true” speaker utterances while the challenge was, with equal likelihood, a mimic utterance of the reference speaker, a natural utterance of one of the mimics, or another utterance from the reference speaker. The results of the present experiment are compared with the results of the previous experiment cited and the performance of an automatic system for speaker verification [Lummis, J. Acoust. Soc. Amer. 50, 106(A) (1971)].
Latest Developments in Voice Identification
Authors: Tosi O. , Oyer H. , Nash E.
Published: October 2005

A 2-yr voice-identification experiment was completed at Michigan State University in December, 1971. A total of 34 996 experimental trials of identification were performed by 29 trained examiners, involving a variety of conditions. Those experimental conditions correlated with forensic models (open trials fixed and random contexts, noncontemporary spectrograms) yielded an error of approximately 6% of false identifications and 12% of false eliminations. Examiners were forced to reach a positive decision in each experimental trial. They judged approximately 60% of their wrong answers and 20% of their right answers as “uncertain.” This suggests that if the examiners had been able to express no opinion when in doubt, only 74% out of the total number of tasks would have had a positive answer with approximately 2% errors of false identifications and 5% errors of false eliminations. The experience obtained in both laboratory and field work, during more than 3 yr, convinced the authors that voiceidentification technique could yield a negligible error providing that (1) the examiner is properly trained in phonetics, spectrography, and speech sciences, in addition to completing a supervised 2-yr apprenticeship, and (2) he is willing to make no positive decisions when he is not absolutely certain.
Computer-Generated Spectrograms and Cepstrograms
Authors: Wood M. L.
Published: October 2005

Computer-generated spectrograms have been produced using the FFT algorithm, a PDP-9 computer, a cathode ray tube display, and Polaroid film. The results compare to spectrograms produced by conventional analog techniques. It was found that achieving such a quality requires appropriate choices for filter bandwidths, frequency shaping, aspect ratios, contrast, and film exposure. One second of speech sampled at 10 kHz can be processed in about 2 min. Thus the facility is practical to use as a flexible, on-line, interactive aid in computer-based speech research. Cepstrograms (three-dimensional time, pseudotime or frequency, and intensity displays of the short-time cepstrum) have also been produced; they give a visual indication of pitch period behavior during voicing. Examples of the results obtained will be presented with conventional results for comparison purposes.
Animal Sonar: Measurements and Meaning
Authors: Diercks K. J. , Trochta R. T.
Published: October 2005

Problems in detection, recording, and analysis of animal sonar signals are described. Specific examples from echolocation studies are discussed. The bionics contributions of animal sonar research are examined, and suggestions for future bionically oriented research are proffered.
Some Aspects of Fluctuations on Detection Performance
Authors: Kirby W. D. , Spooner R. L.
Published: October 2005

In the past few years, increased interest has been evident in the area of signal and noise fluctuations in the ocean mediums. Both measurement and theoretical efforts have been carried out to develop more comprehensive knowledge of the nature and ramifications of these effects. In this context, the purpose of this paper is to indicate some aspects of the effect of fluctuations on receiver design, performance, and performance modeling. In particular, the effect of rapid-noise fluctuation on receiver design and performance is considered and the role of hard clippers in rapidly fluctuating noise environments is reconfirmed. The asymptotic performance of this type of receiver is also determined. In addition, the incorporation of signal fluctuation in performance modeling is also considered in the paper. The effect of such fluctuations on recognition differential and the probability of detection are illustrated for both optimum and energy receivers. These results are incorporated in a discussion of the sensitivity of performance modeling results.
Programmable DIMUS Processor
Authors: Nickles John C.
Published: October 2005

A 48-element, 256-beam, 25-kHz sample-rate programmable DIMUS processor is currently being constructed to work with a variety of arrays on the research platform FLIP. The design approach eliminates the need for a discretionary hard-wired steering matrix, while the system clock rate remains essentially the same as for a serial beamformer of conventional design. The processor is easily reprogrammed for different array configurations or for changes in the set of beam steering directions. Although the technique has been developed specifically for a single bit DIMUS beamformer (using clipped hydrophone outputs) it is equally applicable to multiple level Digital Multibeam Steering Systems.
Series Solution of Digital Beamformer Problems
Authors: Wang H. S. C.
Published: October 2005

Due to the nonlinear characteristics of the input quantizers, the statistical average-power output of a digital beamformer corresponding to Gaussian random input signal and/or noise can best be expressed as an infinite series of a “quantizer function” Qm(K,q̃), where m is the term order and K, q̃ are quantizer parameters. To ensure valid calculations of important performance criteria as array gain, directivity patterns, etc., the convergence of such series is rigorously proved and numerically demonstrated. Quantizer functions of orders up to 150 corresponding to 1–4 bits of quantization per input channel have been calculated by means of “reduced Hermite polynomials.” With such a formulation, values of on-target array gain are computed as a function of the quantizer step size, which clearly exhibits a maximum representing optimum design condition for each specific number of quantization bits. This optimum array gain increases with the number of bits per channel, but approaches the ideal linear array gain for the same number of receiving elements when 4 bits is reached. For arbitrary signal incident angle θ, the output power of an array of N elements spaced d/λ wavelengths apart assumes the form (q̃2/2π)Σm−0∞[Qm2(K,q̃)Θm(N,d/λ,θ)]. Directivity patterns of line and circular arrays corresponding to narrowband and broadband signals of various bandwidths are calculated with relative ease and presented. The special case of 1 bit/channel DIMUS beamformer is discussed in considerable detail and compared with known closed-form solutions.
Sampling Statistics for Higher-Order Cylindrical Modes
Authors: Waterhouse Richard V. , Jones Kenneth E.
Published: October 2005

The distribution functions are considered for the random sampling in space of a three-dimensional acoustic mode. Expressions are obtained for the probability density and cumulative functions for a single mode excited in an ideal elastic fluid contained in a right-circular cylindrical enclosure. In earlier work, axisymmetric modes of lower order were treated; here the work is extended to some modes of higher order, and in addition the rms values of the modal function are sampled, as well as the mean-square values. The rms modal function |J0(⋅)| contains cusps on the abscissa, and these cause the distribution functions to differ from those pertaining to the square of the modal function. In both cases, however, the density functions have a number of poles which increases with the order of the mode. The values of the functions were calculated by digital computer.
Use of Cepstrum to Detect a Finite-Length Target in a Low-Signal-to-Reverberation Ratio Environment
Authors: Ellis G. E. , Shooter J. A.
Published: October 2005

In sonar problems, targets and boundaries such as an air-water interface are frequently modeled as point scatters. Therefore, sound backscattered from a target has the same character as the sound backscattered from a boundary (reverberation) and the sonar is said to be reverberation-limited. If the target is a rigid body of finite length and the medium is homogenous then the target will backscatter consistent returns to the projector/receiver. Therefore, if one has absolute phase control, then it would be possible to average over repeated pings to achieve a processing gain, thus enhancing the presence of a target in reverberation. A condition on the reverberation is that it must be random, which would be true of the sound backscattered from the air-water interface. The condition of absolute phase control is near impossible except under laboratory conditions. It is shown here that an alternate procedure for detection is to compute cepstrums of the reverberation plus target and to average the cepstrums over repeated pings. That is, the cepstrum is independent of absolute phase and the processing gain is proportional to the number of averages. A model experiment was performed in a freshwater lake. Time-varying cepstrums of short segments of return data were computed for 200 pings. The detection results are shown in the form of 3 ROC curves for averages over 1 ping, 3 pings, and 10 pings. It is clear that the averaging improves the probability of detection. [This work was supported by the U. S. Naval Ship Systems Command.]
Scattering of Sound Waves at a Periodic, Pressure-Release Surface
Authors: Holford R. L.
Published: October 2005

A rigorous solution is presented for the scattering of plane sound waves at a periodic, pressure-release surface of arbitrary but smooth profile. With the help of a Green's function, the problem is first reduced to an integral equation of the second kind. This, in turn, is reduced to an infinite system of algebraic equations whose solvability follows from a result of Urusovskii's in a related problem [Sov. Phys. Acoust. 10, 287–293 (1965)]. It is noted that the solution differs from an earlier one due to Uretsky [J. Acoust. Soc. Amer. 35, 1293–1294 (L) (1963); Ann. Phys. (N. Y.) 33, 400–427 (1965)], which is shown not to be rigorous. Some preliminary numerical results will be presented, and it is expected that these will ultimately provide a suitable test of the method of physical optics which has been so pervasive in the work on rough-surface scattering. [Work supported by the Office of Naval Research under contract.]
Sound Waves in the Atmosphere at Infrasonic Frequencies
Authors: Cook Richard K.
Published: October 2005

Various geophysical processes generate sound waves in the atmosphere. Some typical sources are auroral discharges in the upper atmosphere, tornadoes and severe storms, surface waves on the oceans, volcanic explosions, earthquakes, and atmospheric oscillations arising from unstable wind flow at the tropopause. Man-made sources include powerful explosions and the shock waves from vehicles moving at supersonic speeds, at altitudes below about 125 km. The components of sound-wave energy at infrasonic frequencies (oscillation periods >1.0 sec) are propagated for large distances (thousands of kilometers) over the earth's surface with very little loss of energy from absorption by viscosity and heat conduction. But the propagation depends strongly on (a) the horizontally stratified temperature structure of the atmosphere, (b) the influence of gravity at oscillation periods greater than the atmospheric resonance period ≈300 sec and (c) the nonuniform distribution of atmospheric winds. The microphones and electroacoustical apparatus at an infrasonics observation station, e.g., the one at Washington, D. C., measure (1) the amplitude and waveform of incident sound pressure, (2) the direction of local propagation of the wave, (3) the horizontal trace velocity, and (4) the distribution of sound wave energy at various oscillation frequencies. Researches on propagation require observational data from a network of stations separated geographically by large distances, coupled with theoretical analysis of sound propagation, to arrive at useful results on the acoustics of the atmosphere.
Digital Processing of Microbarograph Array Data
Authors: Smart E. , Flinn E. A.
Published: October 2005

This is a review of recent developments in techniques for the detection and analysis of infrasonic signals recorded by arrays of microbarographs. We describe a high-speed algorithm for the calculation of frequency-wavenumber spectra, and discuss two data processing methods the new algorithm makes possible to carry out in real time: first, signal detection by searching for maxima of power in frequency-wavenumber space, together with estimation of the significance of such maxima using a Fisher statistic [Alexandria Laboratories, SDL Rep. 263]. This estimation procedure is compared to the classical broadband N-4 correlator detection algorithm, and shown to be superior in several ways. Second, we discuss a signal spectral estimation algorithm in which the signal power spectrum, phase velocity, azimuth of arrival, and Fisher statistic are calculated and displayed as functions of frequency; the estimates are constructed by tracing power maxima through frequency-wavenumber space, frequency by frequency. Time-domain waveform estimates using this “best-beam” technique are subject to considerably less error and distortion than the conventional procedure, which uses a constant phase velocity. Examples of these algorithms are shown using data from the large aperture microbarograph array in Montana.
Ear as an Analyzer of Musical Tones
Authors: Gibson H. Lee
Published: October 2005

A previously reported study by Gibson (1970) found a less categorical limitation in the analysis of harmonically complex tones than that stated by Plomp and Mimpen in 1968. Meanwhile Duifhuis (1970) has indicated a general capability to hear partials as high as the fifteenth to the twentieth when all are at a normalized level of 30 dB. Our further investigation of peripheral capabilities establishes that, in binaural listening to a two-way loudspeaker in a music room, experienced musicians can consistently duplicate the frequency of any intermittent partial as high as the eleventh for 31.25-Hz tones, the twelfth for 44-Hz tones, the fifteenth for 62.5-Hz tones, the fourteenth for 125-Hz tones, the eleventh for 250-Hz tones, the eighth for 500-Hz tones, the fifth for 1000-Hz tones, and the third for 2000-Hz tones. In binaural listening via earphones to a similar complex tone having the same phase in both ears, the capability may be reduced 10% to 25%. [Research sponsored by North Texas State University.]
Impedance Calculations for Oboe and Oboe-like Structures
Authors: Plitnik George , Strong William J.
Published: October 2005

A scheme has been developed for calculating the input impedance at the reed end of woodwind structures. The scheme is based on representing the bore and fingerholes of an instrument with lossy cylindrical sections. The log of the impedance is calculated and plotted at 10-Hz intervals over the range from 0 to 3000 Hz for each specified fingering of the instrument. Impedance curves have been calculated for the following oboe-like bores: (1) simple truncated cone, (2) smooth bores of one and five conical joints, (3) bores with discontinuities at the reed staple, (4) oboe with fingerholes and no losses, and (5) oboe with fingerholes and losses. Dr. Arthur Benade provided experimentally measured impedance curves for the oboe used as the basis of the calculations for case (5). The experimental and numerical results are compared and discussed.
Cochlear Blood Flow
Authors: Lawrence M. , Nuttal F.
Published: October 2005

A movie illustrating blood flow through the vas spirale as discussed in paper L4 is shown.
Sound Wave Demonstration Apparatus
Authors: Bassett Irvin G.
Published: October 2005

The apparatus described in this paper consists of four loudspeakers, driven by the same oscillator, but each having independently variable output levels. They can be arranged in two configurations: (1) Each loudspeaker on one side of a square box, facing into the center of the box. (2) The speakers are placed side by side presenting a linear array of four sources. Several demonstrations are possible in each configuration. A microphone, attached to a sound-level meter provides a quantitative monitoring device. In configuration (1), the speakers are set independently at the same reference level. Turning on combinations of two, three, and four at a time produces increases of 6, 10, and 12 dB, respectively, in the total level within the box. From this the student can gain a feeling for the decibel scale. By varying the frequency and the phase of the signals, it is possible to demonstrate interference phenomena and relationships between frequency and wavelength. Configuration (2) can be used to demonstrate multiple source interference patterns. The apparatus is useful in both classroom and laboratory situations.
Computer Movie of a Vocal-Cord Model
Authors: Shipley K. , Ishizaka K. , Flanagan J. L.
Published: October 2005

A two-mass model of the vocal cords [K. Ishizaka and J. L. Flanagan, this meeting] is programmed to display the glottal area and cord displacement during phonation. Area and displacement values are calculated at a sampling rate of 5000 sec−1, comparable to high-speed photography of the human vocal cords. Computer film is generated to display top and side views of the motion of the model. Effects of variation in subglottal pressure, cord tension, and phonation-neutral area are demonstrated by the film.
Baffles, Transistor Radios, and the Ear
Authors: Weber Louis R.
Published: October 2005

With a direct radiator loudspeaker, one has essentially an acoustic doublet, the energy from the front and back, differing in phase by 180°. It can easily be shown that the response spectrum depends on the length of the baffle. Because most transistor radios are small, the physical dimensions of the baffle must be small also. How does one account for the apparent presence of these lower frequencies in these small radios? By using three or four audio-oscillators, it can be easily demonstrated that these lower frequencies are supplied by the asymmetry of the ear and are not actually in the transistor radio radiating system itself. A similar situation in optics will be demonstrated. Thus, the ear has been created almost with a prior knowledge that man would need help in imperfect radiating systems so that an acceptable reproduction of a symphony orchestra would be possible.
Effects of Earmold Modifications as Measured in a 2-cm3 Coupler and upon the Psychophysical Experience of Loudness
Authors: Jacobson Edward J. , Krug Richard F. , Teter Darrel L.
Published: October 2005

This study explored the relationship between physical measurements of intensity and psychophysical measurements of loudness using four standard-type earmolds. Two of the earmolds were unaltered, while a third contained a small vent, and the fourth a large vent. The physical measurements were derived using a 2-cm3 a coupler; psychophysical data were derived with an alternate binaural loudness balance technique using seven normal-hearing subjects. The test signals employed to collect the psychophysical data were pure tones of 300, 500, and 2300 Hz, each of which was presented at 45, 55, and 65 dB SPL. Significant changes in loudness did not occur due to vent size although frequency response changes were apparent for the vented and nonvented earmolds. The data which were obtained suggest that acoustic measurements derived in a laboratory with a 2-cm3 a coupler should not be indiscriminantly applied as an index to the way in which such physical changes in intensity are auditorily perceived.
Loudness Enhancement and Summation in Pairs of Short Sound Bursts
Authors: Zwislocki J. J. , Ketkar I.
Published: October 2005

When a pair of short temporally spaced sound bursts are presented monaurally, and the listeners are requested to match the loudness of a third comparison burst to the total loudness of the pair, a fundamentally different result is obtained than when they are instructed to match the loudness of the comparison burst to that of the second burst in the pair. The most striking difference has been obtained thus far with the sound frequencies of the first two bursts at 1 and 4 kHz, respectively, and the frequency of the comparison tone at their geometric mean when the listeners judged the loudness of the burst pair and at 4 kHz when they judged the loudness of the second burst. In the first situation, the loudness level of the pair appeared increased by almost 10 dB over the loudness level of a single burst and was independent of the time interval between the bursts in the pair. In the second situation, the loudness was increased by only 3 dB for very short time intervals, and the increment decreased to zero within 80 msec. Since, at durations greater than 80 msec the tone bursts are heard as distinctly separate entities, we conclude that listeners consciously summate the loudness of the bursts when asked to judge the total loudness of the pair.
Duty Cycle Effects on the Loudness of Repeated Tones
Authors: Hattler K. W. , Stokinger T. E.
Published: October 2005

The alternate binaural loudness balance technique was employed to investigate the alleged influence of duty cycle (the percentage of time the sound is on) upon perceived magnitude of 1000-Hz tones. The reference tone was fixed in the right ear at 50 dB SPL. In the comparison ear, the signal intensity was adjusted via the Békésy audiometer to derive equal interaural loudness. The perceived loudness of repeated signals increased as the signal duty cycle was increased. A low-duty cycle comparison tone was consistently judged to be softer than an equally intense reference signal of higher duty cycle. Furthermore, a high-duty cycle comparison signal was consistently judged louder than a low-duty cycle reference signal at equal intensities. The greatest interaural loudness disparity was observed for the experimental conditions in which the interaural duty cycle difference was 80%. During a substantial number of trials this direct relationship between duty cycle and loudness grew in magnitude for the extent of the 4-min balancing period. Duty-cycle-loudness effects were enhanced when a 400-msec silent interval separated alternating tones. These results help explain the occurrence of the type-V Békésy pattern in clinical audiology as well as results on some loudness-adaptation studies.
Some Parameters of Vocal Effort
Authors: Wright H. N. , Colton R. H.
Published: October 2005

Vocal effort (autophonic level) for the vowel /a/ in the modal register was measured by magnitude estimation and production as a function of the SPL developed as well as changes in fundamental frequency. Maximum, minimum, and most comfortable effort levels were determined. No standard was used. Magnitude of production was numerically assigned from data obtained from each of five male subjects. The results show that vocal effort is a power function of both SPL and fundamental frequency. As has been previously found in investigations of loudness over a wide range of values where a standard was not assigned, most of the present power functions were concave downward. It has been previously implied [Wright, “Is There A Standard Loudness?” J. Acoust. Soc. Amer. 37, 1175(A) (1965)] that the implicit standard is a most comfortable level and that this level occurs in the vicinity of the slope change in the power function. Such was the case in this investigation. The most comfortable effort level almost invariably occurred where the obtained power functions changed their slopes, both in terms of SPL and fundamental frequency.
Loudness Enhancement Following Contralateral Stimulation
Authors: Galambos R. , Bauer J. , Picton T. , Squires K. , Squires N.
Published: October 2005

The apparent loudness of a tone pip can be increased by a prior toneburst presented to the contralateral ear. A listener receives two tone pips S1 and S2 separated by 1.5 sec, and he adjusts the intensity of S2 to equal S1 in loudness. When a toneburst to the contralateral ear precedes S1 by an interval ΔT, the loudness of S1 is enhanced by as much as 14 dB. This enhancement (1) increases as the frequency of the contralateral toneburst approaches S1 from above or below; (2) decreases to zero as ΔT is increased up to 1 sec; (3) increases as the contralateral stimulus duration is increased up to 200 msecs; and (4) is monotonically related to contralateral intensity greater than 70 dB SPL. In a particular case with S1 at 8 kHz, changing the frequency of the contralateral toneburst from 8 to 8.2 kHz cut the loudness enhancement to one-half. The possible physiological basis and psychological significance of this phenomenon will be discussed.