bluuebunny/crossref_metadata_2025 huggingface dataset:
Loudness of high-intensity aircraft flyovers
Authors: Ward W. Dixon
Published: October 2005

Casual listening to a recorded aircraft flyover at various peak levels up to 111 dBA gave the impression that reducing the level by 10 dB did not seem to reduce the loudness by half. Therefore a group of eight listeners (lab assistants and students experienced in high-intensity noise exposure) were given pairs of flyovers, the first of which was always at 111-dBA peak and called “100” in loudness, and were asked to indicate the loudness of the second, whose peak level ranged from 61 to 106 dBA. Although the loudness-intensity relation was reasonably close to a power function for each individual, the slopes ranged from 6 to 35 dB per halving loudness, with a median of 16 dB. No listener assigned a loudness of “50” or less to a 101-dBA-peak flyover. Thus, if one wishes to halve the loudness of aircraft noise at the ends of runways, it will be necessary to reduce peak levels by considerably more than the 10 dB that represents an extrapolation from loudness judgments made at moderate levels. The same holds, of course, for “noisiness,” and probably also for “annoyingness.” [Research supported by the National Institute for Occupational Safety and Health, Public Health Service.]
Effect of imposing various skewed distributions of hearing loss on a normally distributed population
Authors: Johnson Daniel L.
Published: October 2005

One of the surprising facts, with respect to noise-induced hearing loss, is that many studies have shown that there is a greater change in the higher percentile levels (worse hearing) between the statistical distributions of noise- and non-noise-exposed populations. To provide better insight into this phenomenon, the theoretical results of taking a normally distributed group of a non-noise-exposed population and calculating (via computer methods) the resulting distributions after various noise exposures were assumed are presented. The key assumption was that noise exposure would not improve an individual's hearing. The resulting distributions show the greatest changes at the percentiles with the worse hearing, even though the computer model assumed that there was no relation between sensitivity to noise-induced hearing loss and initial hearing level. The application of this result to existing data on noise induced hearing loss is discussed.
Measurements of complicated wave fields in underwater acoustics
Authors: Cox Henry
Published: October 2005

The propagation of acoustic waves in the ocean frequently involves multiple refractions and reflections leading to multipath signal arrivals and complicated noise fields. This paper reviews a number of measurement and estimation techniques which have been used or are being considered for use in underwater acoustics. Special emphasis is placed on the assumptions underlying the various techniques, their interrelationships, and the limitations inherent in those techniques. Among the topics discussed are communication theoretic approaches, beamforming, maximum likelihood estimation, maximum entropy estimation, spatial harmonic models, frequency-wavenumber analysis, and principal component analysis.
Array design and processing techniques for study of atmospheric wave fields
Authors: Young J. M. , Bedard A. J.
Published: October 2005

We distinguish between atmospheric acoustic and gravity waves, describe their characteristics, and discuss optimum array configurations for studying them in the presence of turbulent pressure fluctuations. The separation of two or more waves present simultaneously—and the distinction between waves from point- and distributed sources in the presence of inhomogeneities and uniform variations in the medium—present problems that are only partially solved. We outline the application of both analog and digital array processing techniques to the determination of transient atmospheric pressure wave characteristics as a function of space and time. These techniques are applied to several examples of infrasonic signals and atmospheric gravity waves to illustrate the present state of processing. A review of recent improvements suggests to us that future work should be directed toward (1) optimum use of constraints imposed by wave dispersion, source mechanisms, etc., in order to arrive at conclusions more directly, and (2) optimum display of results to enhance recognition of qualitative and quantitative differences in the distribution of wave parameters corresponding to waves of different types.
Seismic signal and noise processing
Authors: Blandford Robert
Published: October 2005

The character of seismic signals and noise is known on the basis of theoretical solutions of the elastic wave equation and from observations made at seismic stations and arrays. The design of these arrays depends on the correlation structure of the signals and noise, and on whether the goal is to separate mixed signals or to detect signals in noise., Advanced signal-processing techniques have been shown to be of practical importance in the mixed signal problem. Among techniques which have been developed beyond beamforming are three-dimensional frequency-wavenumber spectra, nonlinear three-component waveform filtering, maximum-likelihood processors using either a priori or measured signal and noise characteristics, and eigenvalue-eigenvector decompositions of mixed signals. Theory and examples will be given for each of these techniques.
Quasiholographic techniques in the microwave region
Authors: Leith Emmett N.
Published: October 2005

Following the intransification of holographic activity in the 1960s, various researchers reported experiments that were direct microwave counterparts of the optical holography which preceded them. This work, which has recently become extensive, may be termed true microwave holography. As the title of our paper implies, we deal not with this rather restrictive field, but with a much broader one which embodies holographic-like techniques. There exists in the field of communication science a variety of techniques that resemble holography in various degrees, both in concept and in their mathematical formulation. Yet these processes have developed quite independently of holography and in no way depend upon principles originating in holography. We select four such techniques for our discussion: the synthetic aperture, the chirp radar, the rotating-target imaging system, and the beamforming technique. Each in its own way resembles holography, and viewing them as holographic processes offers, in some cases, new insights that lead to new implementations.
Sequential aspects of auditory patterns
Authors: Hirsh Ira J.
Published: October 2005

Underlying considerations for analysis of sequential properties of sound patterns are outlined. First, certain sequential properties may provide the basis for stating gestaltlike principles of organization in acoustic patterns. Second, listeners' performance may be used to adduce properties of the auditory system beyond those of peripheral processing, and yet short of those characteristic of memory. Third, examples of sequential aspects in the messages of music and of speech can serve both to illustrate application of perceptual principles, and to permit possible separation between effects typical of human auditory systems generally and those that depend on learning. [Supported by a grant from NINDS.]
Noise in space
Authors: Rader W. P. , Baratono John , Bandgren Harry , Erwin Robert
Published: October 2005

The Skylab program presented an excellent opportunity to investigate the effects of noise on man confined in limited space for long periods of time. This paper summarizes the results of a four-year study to achieve a habitable noise environment for the Skylab astronauts. Noise control measures are described and noise measurements obtained during the Skylab missions are presented, as well as the astronauts' reactions to and evaluations of the noise environment.
Considerations in reducing the Noise Pollution Level of roadway traffic
Authors: George Ballard W.
Published: October 2005

Barriers are measures commonly employed to mitigate roadway noise. Typically, the requirement is to reduce the L10 or other statistical level. When design objectives also include reduction of the Noise Pollution Level, LNP, which depends partly on the noise fluctuation, additional complications are introduced. Specifically, for combined automobile and heavy truck traffic, the attenuation of LNP tends to exhibit a dip and may become negative at certain barrier height values. This effect involves changes in the relative degree of masking of truck noise by automobile noise. Calculations were made and design curves plotted using procedures of the Department of Transportation to assess the variation of barrier performance with barrier height and with truck-automobile mix ratio. Barrier performance is noted to become most stable when it is located close to the receiver. For level terrain, negative barrier attenuations of up to 5 dBA may occur. These maximum negative values occur in conjunction with truck-automobile mix ratios from 1%–5% and effective source to barrier distances of approximately 50 ft or less.
Sound scattering from an elastic cylinder covered with a viscoelastic coating
Authors: Gaunaurd G. C.
Published: October 2005

An infinite hollow elastic cylinder is coated with an ideally bonded layer of viscoelastic material. The structure separates an outer fluid (water) from an inner fluid (air). A plane sound wave is incident on the coated cylinder and scattered by it. The “classical” normal-mode solution of this scattering problem is obtained. The solution for the pressure field in the water is slowly convergent. Use is made of the Sommerfeld-Watson Method to transform that solution into a fastly convergent one by means of contour integrations and contour deformations in the complex plane v of the order of the resulting Bessel functions. The geometrically reflected part of the solution can be handled by the saddle-point method and rest, or “creeping”-wave part, by the Residue Theorem. Expressions for the differential scattering cross section and the sonar cross section are then found which are fastly convergent at the high frequencies of interest. This provides a way to determine the effectiveness of the coating material in reducing the target strength of the bare cylinder.
On the classification of underwater acoustic signals. II. Experimental applications involving fish
Authors: Deuser L. M. , Middleton D. , Plemons T. D. , Vaughan J. K.
Published: October 2005

The general technique described in Part I is applied to several experimental classification problems. An experiment conducted at 116 kHz in the ARL Test Tank Facility is described, in which it was possible to obtain echoes from a known number of scatterers random in location and acoustic cross section. The main scatterers of interest here are one variety of small fish, all of about the same length (10 cm). Other origins of backscattering, which form the signals for competing categories, are seaweed (aquarium grass), surface reverberation (mechanically driven), and an artifact (glass bottle). Using these data, we formulate three classification problems. The three problems are (generally) the categorization of received signals which result from scattering by (a) fish versus surface, seaweed, and/or artifacts; (b) fish and surface versus surface; and (c) low versus medium versus a high density of fish. Classifiers are designed and tested with and without estimates of the environmental state. The usefulness of the approach is illustrated by the improvement in performance which results from monitoring the environmental state. [Work supported by the Naval Sea Systems Command.]
Correlation between zooplankton population and the volume-scattering coefficient
Authors: Anderson W. B.
Published: October 2005

Reverberation backscatter measurements have been conducted in Puget Sound and the Strait of Georgia. The acoustic measurements were made in the frequency spectrum common to commercial fishing sonar systems. Using a downward-looking transducer, the acquired reverberation backscatter data are used to calculate the volume-scattering strength coefficient of the water column, a coefficient analogous to the target strength of a unit volume of water along the acoustic axis of the transducer array. Concurrent with the acoustic measurements, vertical biological trawls are conducted using a 1-m2 plankton net to collect the zooplankton samples from the water column directly below the measurement transducer. The biological samples are preserved for analysis of settled volume per depth interval and separation by taxa. Correlation is apparent between zooplankton layers and acoustic backscatter. A typical dense layer of zooplankton, nominally 100 ft thick, can result in a 10-dB increase in the volume-scattering-strength coefficient over that depth interval. Adjacent regions in the water column having a lesser volume-scattering coefficient are seen to have a lesser zooplankton population.
Deep-scattering-layer observations in the Indian Ocean
Authors: Vent R. J. , Batzler W. E.
Published: October 2005

Acoustic volume-scattering measurements using 3.5 and 12 kHz pulses and broad-band explosive signals were made from the USNS CHAS H. DAVIS during an oceanographic cruise in the Indian Ocean in May–July 1969. Seven acoustic stations were occupied, three in the Java Trench area and four in the Bay of Bengal. Downward-looking directional transducers were used in the pulse-type measurements. Strong scattering layers (maximum scattering strength, −55 dB to −70 dB re 1 yd) were observed during daytime measurements at 12 kHz. Layer depths at 12 kHz ranged from 325 to 550 yd and layer thickness (at 6 dB down) varied from 80 to 300 yd. Layers were less well defined at 3.5 kHz and maximum scattering strengths in the layer are less than at 12 kHz by from 10 to 30 dB. In general, layers are thicker and extend to greater depths than at 12 kHz. Column strengths obtained from pulse measurements (column lengths 700–900 yd) range from −35 to −50 dB at 12 kHz and from −45 to −65 dB at 3.5 kHz. With two exceptions, column strengths at both 3.5 and 12 kHz are higher at night than for daylight observations. The exceptions, both at 12 kHz, have day column strengths which exceed the night values by 4 and 12 dB, respectively. Results obtained using explosive signals (1.8 lb TNT) and omnidirectional hydrophones cover 13-octave bands from 1 to 6.4 kHz. Overall values of column-strength range from about −75 dB at 1 kHz to −40 dB at 5 or 6 kHz. There is good agreement between the pulse results at 3.5 kHz and those found at 3.2 kHz by the explosive procedure. Our results, from both the pulse and explosive procedures, show fair to good agreement with the extensive column-strength results obtained by Hall [M. Hall, J. Acoust. Soc. Am. 54, 473–477 (1973)] in some of these same Indian Ocean areas.
Volume-scattering strengths at 60 kHz in the Bering and Chukchi Seas
Authors: Feldman H. R.
Published: October 2005

Volume-scattering strength measurements were made with a 60 kHz, downward looking, pulsed echo-sounder in March–April 1973, in the Bering Sea, and in August 1973 and September 1974 in the Chukchi Sea. The measurements in the Bering Sea were all made from helicopters during the day beneath heavy ice cover. Those in the Chuki Sea were made under varying ice conditions, from helicopter in August 1973 and from a ship in September 1974. In August, daylight is almost perpetual in the Chukchi Sea; in September, there is substantial darkness, and measurements were made day and night. The average volume-scattering strengths observed were about −100 dB (re 1 m−1) in the Bering Sea, −81 dB in the Chukchi Sea in August 1973, and −73 dB in the Chukchi Sea in September 1974. The maximum strength of any layer observed was −87 dB in the Bering, −73 dB in the Chukchi in August 1973, and −66 dB in the Chukchi in September 1974. Correlations with ice cover, water temperature, and daylight were noted.
Theory for hydroacoustic multiple target scattering assessment with experimental verification
Authors: Stevenson E. A. , Shepard W. Steve
Published: October 2005

A new formulation for hydroacoustic scattering is presented. An average target density is defined assuming that for each aggregation of N scatterers, an ensemble of configurations is possible. Each configuration gives rise to an unique reflected signal which is input to a linear time invariant bandlimited system and the output or observable signal is processed to determine N. The problem is formulated as a nonstationary stochastic process with the autocorrelation function of the input process being derived and shown to consist of stationary and nonstationary parts. Then the energy spectrum of the output process is determined using a two-dimensional Fourier transform and linear system theory. The developed theory is compared with experimental data obtained in an elaborate test conducted by NMFS Fisheries Engineering Laboratory at Bay Saint Louis, Mississippi. Calculated target densities are found to lie within 89% confidence limits based on Tchebycheff's inequality.
Intracellular and extracellular dc potentials in the cochlea of the alligator lizard
Authors: Weiss T. F. , Altmann D. W. , Mulroy M. J. M. J.
Published: October 2005

Intracellular resting potentials (RP) and extracellular endolymphatic potential (EP) were recorded in the cochlea of the alligator lizard (Gerrhonotus multicarinatus) by means of micropipets. Ninety percent of the RPs recorded during penetrations of the region of the basilar papilla lie in the range −25 to −125 mV with a median of −70 mV. Supporting cells and hyaline epithelial cells appear to have RPs whose median value is more negative than the median RP of hair cells. RPs of cells in the distal portion of the cochlear nerve are generally more positive than −35 mV. The dc potential difference between endolymph in scala media and the perilymph (EP) has a median value of +14 mV. The RPs and EP are compared with results obtained in other hair cell organs. [Supported by NIH grants.]
Evaluating nerve function of the inner ear
Authors: Mitchell Curt
Published: October 2005

The N1 potential of the cochlear nerve has long been used to evaluate cochlear function. It has not been clear what population of nerve fibers contributes to this potential. Several lines of evidence suggest that N1 potentials evoked by tone pulses activate a very limited population of nerve fibers and thus are frequency specific. (1) The latencies correspond to cochlear travel time and neural delays. (2) Fatiguing tones produce frequency specific depressions of N1. (3) Tones mask these N1 potentials in a frequency specific manner. The effects of rise-time and tone-pulse duration on these observations will also be discussed.
Recordings of the slow potential change (COCP) evoked by crossed olivocochlear bundle (COCB) stimulation in the scala tympani, organ of Corti, and scala media and its response to anoxia
Authors: Bobbin R. P.
Published: October 2005

The COCP recorded in the scala media of guinea pigs via the spiral ligament was negative in polarity and ranged up to 4 mV. Upon terminating artificial respiration, the COCP fell within 2 min to 0.25 mV, where it remained for 4 min before declining to zero. The COCP recovered rapidly when artificial respiration was reinstated within 3 min of termination. COCP recorded in the scala tympani and organ of Corti was positive in polarity and ranged up to 0.2 mV. Recordings were also made during intracellular recordings in the organ of Corti. Niagara Sky Blue dye-containing microelectrodes were used to mark the recording sites. [Supported by National Institutes of Health, USPHS Grant Nos. NH-11647-01 and NS-10463-01.]
Interactive experiments with a Digital Pattern Playback
Authors: Nye Patrick W. , Cooper Franklin S. , Mermelstein Paul
Published: October 2005

Facilities which make spectrograms immediately available for visual comparison, easy modification of spectral data, and resynthesis of speech have proved to be particularly useful tools in speech research. This paper reports an experiment in which such an interactive research tool—a Digital Pattern Playback (DPP)—is being used to evaluate a spectrum-matching and dictionary-search technique for speech recognition. The DPP is a computer-supported analysis-synthesis facility which, in the present experiment, displays spectrograms of “unknown” sentences so that an analyst can list the important acoustic features of marked segments of the unknown sentence. Interrogation of a feature-based dictionary then recovers all items with features which match the unknown segment. If necessary, additional features may be assigned to narrow the search. The reference spectrograms retrieved from the dictionary are compared, one at a time, with the spectrogram of the unknown sentence and the best match is selected for each unknown segment. Instrumentation, strategies, and factors governing the efficiency of the algorithms for feature-selection and spectrogram-matching will be discussed. [Work supported by NSF and ARPA, Department of Defense.]
Acoustic-phonetic experiment facility for the study of continuous speech
Authors: Schwartz Richard M.
Published: October 2005

While gathering acoustic data for the acoustic-phonetic analysis of speech, it is necessary to consider many different sounds in varying phonetic environments to assure that the results are statistically significant. In order to reduce the amount of time required to test hypotheses, a facility has been developed which provides an interactive environment for performing a wide variety of acoustic-phonetic experiments on a large data base of continuous speech. Using this facility, one can formulate an experiment, run it on selected portions (or all) of the data base, display or tabulate the results in a meaningful way, and then run another experiment (or a variant thereof) based on the results. Due to the ease of interactions, formulating or revising an experiment, running it, and displaying the results normally takes less than 5 min. This facility has been used in combination with a data base of 69 hand-labeled sentences to develop algorithms for acoustic-phonetic segmentation and labeling in a speech understanding system. Several examples of its use and the results obtained will be presented.
Approach to automatic segmentation in the acoustic phonetic transformation
Authors: Kasuya H. , Wakita H.
Published: October 2005

As a first step in the study of automatic acoustic-phonetic transformation of speech sounds by an arbitrary speaker, the segmentation of connected speech into vowel-like and consonantal segments has been investigated. Area functions obtained by the linear prediction method have been found to provide useful parameters for this purpose. An advantage of utilizing area functions is that the segmentation scheme can be easily combined with interspeaker normalization [H. Wakita, J. Acoust. Soc. Am. 57, S3(A) (1975)] to handle an arbitrary speaker. Two criteria were used for determining [+ consonantal] features: (1) the minimum of the Euclidean distances [MED] between the input area function and the area functions of all ten American vowels registered as references, and (2) the ratio of the back cavity volume to the total volume of the vocal tract [RBC]. The [+ consonantal] feature is assigned to those segments for which either the MED or the RBC exceeds a corresponding threshold value. Experiments with seven short sentences spoken by two adult males and an adult female resulted in quite satisfactory segmentation. Although [l] segments were sometimes labeled as [− consonantal], the nasal segments surrounded by high vowels were correctly detected. Further attempt to subdivide the [+ consonantal] segments into individual consonant segments will be discussed.
On the similarity of noisy phonetic strings produced by different words
Authors: Baker James K.
Published: October 2005

In a speech recognition system with an acoustic processor which attempts to automatically estimate a phonetic transcription, it is necessary to know the similarity of the probability distributions of phonetic strings when different words are spoken and input to the acoustic processor. Let aεA,a=a1a2a3…an represent an arbitrary phonetic string. Define the similarity between the words W1 and W2 by S =  ∑ a Pr(a|W1) Pr(a|W2). The number of terms in the sum defining S grows exponentially with the length of the words W1 and W2. However, if the nodes of the phonological graphs for W1 and W2 are properly ordered, S can be calculated inductively by a generalization of the computations used in modeling a probabilistic function of a Markov process. The number of computations is approximately the product of the number of arcs in W1 times the number of arcs in W2.
Speaker recognition using orthogonal linear prediction
Authors: Sambur M. R.
Published: October 2005

The effectiveness of a set of speaker recognition features is usually characterized in terms of the ratio of the interspeaker variability of the feature to its intraspeaker variability (F-ratio). A recent experiment in speech synthesis [M.R. Sambur, “An Efficient LPC Vocoder,” Bell Syst. Tech. J. (to be published)] has shown that by an appropriate eigenvector analysis, a set of orthogonal parameters can be obtained that is essentially constant across an utterance for a given speaker (i.e., zero intraspeaker variability). If the same eigenvector analysis is applied to the same utterance spoken by another speaker, the resulting values of the orthogonal parameters are, however, different. These orthogonal parameters were therefore examined for their ability to differentiate different speakers. They were formally tested in a speaker recognition experiment involving 21 speakers. The speech data consisted of six repetitions of the same sentence spoken by each speaker on six separate occasions. The identification and verification accuracy of the orthogonal parameters exceeded 99%.
Selection of features and speech segments for speaker verification
Authors: Lin W. C. , Pillay S. K.
Published: October 2005

A speaker verification system based on the results of two thorough and systematic feature ordering techniques is described. First the Information Theoretic approach is used to reduce the redundancies among the features that are originally present in the feature pool. This selection procedure measures the amount of reduction in the uncertainty on deciding on the identity of the speaker when more features are added to the set of already picked features. Once the redundant features are removed, the Between-to-Within multifeature variance ratio feature ordering algorithm is applied. It is a linear transformation technique which transforms the pattern vectors from measurement space into a new vector space. In the new space, the optimum combination of interspeaker separability and intraspeaker variability is achieved with a few number of features. Based on the results of the feature ordering algorithms, the effectiveness of each segment is determined and a set of minimum number of features is chosen. Preliminary results show that just ten pitch periods in the sounds /el/ or /mi/ alone are sufficient to identify three speakers all the time.
Evaluation of selected acoustic parameters for use in speaker identification
Authors: Doherty E. T.
Published: October 2005

The effectiveness of certain acoustic and temporal properties of the speech signal long-term power spectra (LTS), speaking fundamental frequency (SFF), and speaking time (ST) in the determination of a speaker's identity from his voice alone were tested alone and in various combination. Further, the effect of distortions—limited passband, stress, or disguise—were evaluated. Various analytical procedures, Euclidean distance, cross-correlation or discriminant analysis, are used. Two groups, 50 college-age males who read “normally” and 25 males, aged 25–45, who read normally and while subjected to stress and while attempting voice disguise were selected. Acoustic/temporal analyses were performed on the speakers' utterances to extract the LTS, SFF, and ST vectors. Filtering was simulated for LTS. Results indicated that (1) the LTS vector is extremely effective for identifying speech produced normally, (2) SFF and ST were far less effective, (3) combining vectors usually improved correct identification levels, (4) under stress or attempting a disguise, no single vector or combination adequately differentiated talkers, and (5) a discriminant analysis is a more better method of determining identity than is cross correlations or Euclidean distance.
Spectrographic and aural examination of professionally mimicked voices
Authors: Hall M. , Tosi O.
Published: October 2005

Five members of the IAVI-examined spectrographically and aurally pairs of samples corresponding to (1) mimicked voice-real voice, (2) mimicked voice-mimicked voice, and (3) real voice-real voice, to decide whether or not the two components of the pair belonged to the same or different persons. There were two types of recordings; one type obtained in a quiet environment, the other within an ambient noise environment. Results suggest that there is no significant spectrographic intraspeaker variability within the mimicked voices produced by the particular professional mimic employed in this experiment. However, the examiners found significant interspeaker variation between the real voice of each mimicked person and its mimicked voice by the mimic, variation that allowed them to produce right discriminations. In all cases, average fundamental frequencies differed. The recordings including noise yielded less significant results than the ones obtained in quiet. In addition, these pairs of voices were presented to nontrained listeners aurally only, in a free field, requesting them to decide whether or not the two voices of each pair were the same or different. They produced right answers in approximately 75% of the tests.
Some applications of atmospheric acoustics
Authors: Brown Edmund H.
Published: October 2005

Acoustic methods for atmospheric research have become powerful and increasingly useful tools. The physical properties available from—and limitations of—remote sensing of the atmosphere by echosondes (acoustic sounders) are briefly discussed. Such instruments exploit the amplitude and phase fluctuations of sound waves produced by random processes in the atmosphere to the measurement of the statistical properties of the physical mechanisms involved. A particular application describes how acoustic measurements of the turbulence-strength parameters CT and Cv, and the horizontal wind gradient Uz by a combined monostatic-bistatic Doppler echosonde provides the critical parameters u*, T*, and L* required in surface-layer similarity theory. The important advantages on extension of such acoustic measurements to the entire boundary layer are explored.
Range limitation on radar-acoustic sounding systems (RASS) due to atmospheric refractive turbulence
Authors: Clifford S. F. , Wang T-i
Published: October 2005

We have calculated the effects of acoustic refractivity fluctuations on the operation of a radar-acoustic sounding system (RASS). Using the Born approximation, we determine the electromagnetic field scattered from a spherical acoustic pulse whose sphericity is perturbed by transmission through the turbulent medium between the source and scattering volume. The resultant system gain reduction factor changes the dependence on range R of the total received power from a R−2 dependence for a homogeneous atmosphere to R−18/5 in strong turbulence. This new R dependence can occur at distances of the order of a few hundred meters for sufficiently strong turbulence.
Relation between steady and normally distributed sound levels
Authors: Embleton T. F. W. , Piercy J. E.
Published: October 2005

It is well established both theoretically and experimentally that sound pressure levels fluctuate and have an approximately normal distribution after waves have traveled through a medium containing turbulence, as occurs in most sound propagation outdoors. The related distribution of sound pressure is therefore skewed so that its most commonly occurring value (corresponding to the level Lf, the peak of the distribution of fluctuating sound levels) is less than the mean value (corresponding to the steady sound level Ls that would be measured in an ideal nonturbulent medium under otherwise similar conditions). A simple calculation shows that the difference Ls − Lf equals 0.12 σ2 dB, where σ is the standard deviation of fluctuations of sound pressure level. When the fluctuations are saturated, i.e., σ≃ 6 dB as occurs typically for middle frequencies at a distance of a few hundred feet on a summer day, this difference has a maximum value of approximately 4 dB. The design and interpretation of experiments on propagation outdoors should be considered critically to ensure that this mathematical artifact is not responsible for part of the reduction in sound levels variously ascribed to “terrain effects,” “attenuation by turbulence,” etc.
Mathematical comparison of the effects of multipath interference on estimates yielded by. additive and multiplicative processors
Authors: O'Connor J. C. P. , Nitsche J. R.
Published: October 2005

This analysis considers the use of additive and multiplicative processing techniques to estimate a signal level in an ocean environment that is conducive to an interfering arrival, such as a surface reflected arrival. Representing the auto- and cross-correlation functions as decaying exponentials having different effective correlation times facilitates the evaluation of the integral expressions for the variance. By varying these effective correlation times and the measurement geometry (to change the delay time of the interfering arrival) the sensitivity of the estimate on the input characteristics can be observed. The parameters that are found to govern the system performance are input signal-to-noise ratio, the degree of input noise correlation, and the amount of signal decorrelation. The results show that the estimates yielded by the system depend heavily on input signal-to-noise ratio with the multiplicative processor generally outperforming its additive counterpart for most cases at low input signal-to-noise ratios. [Work sponsored by Office of Naval Research.]
Measurements of the excess attenuation in air-to-ground sound propagation
Authors: DeLoach Richard
Published: October 2005

The attenuation of sound propagating from a fixed source at the top of a 152-m tower to the ground has been measured with simultaneous measurements of meteorological data made at various points on the tower. An attenuation in excess of losses due to spherical spreading, classical and molecular absorption, and refraction is reported. The excess attenuation displays a frequency dependence which is mild compared to the frequency dependence of molecular absorption losses and a correlation is observed between the magnitude of the excess attenuation and both wind shear and Richardson's number.
On the theory of thermodynamic wave cycles
Authors: Peter Abraham C.
Published: October 2005

Cyclic thermodynamic processes governing the successive dissipation and formation of acoustic waves in perfect gases are studied under adiabatic flow conditions. The cyclic heat input into the fluid is due to the frictional conversion of the kinetic energy of the wave into heat. During the implementation of this dissipative heat addition, the pressure and temperature of the fluid increase and its density decreases. The elevated pressure at the end of the dissipative process causes and expansion, in which the pressure, temperature, and the density of the working medium all decrease, until the ambient pressure of the surroundings is reached. The fluid returns to its initial state by means of an isobaric cooling process, thus completing the wave cycle. The expansion process, in which the ambient pressure is reached determines the thermodynamic nature of the wave cycle; an isentropic expansion results in an isentropic wave cycle, similarly for an isothermal expansion, etc. Expressions for the mechanical work performed in various wave cycles are obtained and their efficiencies are discussed and related to the physics of dissipative wave motion.
Some acoustic features of hoarseness
Authors: Murry Thomas
Published: October 2005

Previous investigations of hoarseness have relied on either visual information from spectrograms or perceptual judgments of the voice to identify “hoarseness.” In the present study, voices which were previously rated for severity of hoarseness were examined to determine the basis for the hoarseness judgments. Twenty-nine voices which previously received severity ratings of hoarseness from 0–7 were acoustically analyzed to determine the fundamental frequency and spectral characteristics. The voices were divided into two categories: those which were highly confused according to the severity ratings and those which had general agreement as to the severity of hoarseness. The results indicate that spectrographic analysis alone is not sufficient to identify hoarseness. In addition, speaking fundamental frequency of severely hoarse voices was not significantly lower than the speaking fundamental frequency of the mildly hoarse or nonhoarse voices.
Computer simulation of pathologic vocal-cord vibration
Authors: Ishizaka K. , Isshiki N.
Published: October 2005

A dynamic model of the vocal cords, namely, a bilaterally symmetric two-mass model, is extended to a bilaterally asymmetric two-mass model Asymmetric behavior of the computer model is investigated for various conditions of bilateral imbalance of the cord parameters. The asymmetric behavior is found to be a strong function of the subglottal pressure and the glottal rest area, as well as the imbalance conditions. Also, vocal-cord vibration in the larynx of dog is studied by high-speed motion pictures. The vibratory patterns for tension imbalance are observed and found to agree closely with results from the computer simulation. Specifically, for a given condition of imbalance, the physiological preparation and the computer simulation display the same differences in phase and amplitude of cord vibration. The asymmetric vocal-cord model is finally incorporated into the dynamic vocal-tract synthesizer to produce speech with dysphonia. [This research was supported by the National Institute of Health.]
Intelligibility of speech produced with an electronic artificial larynx
Authors: Yeni-Komshian G. H. , Weiss M. S. , Heinz J. M.
Published: October 2005

Five normal-speaking adult males were trained to use an electrolarynx. A speech pathologist trained the talkers ever a six-week period. At the end of the training period each talker read two lists from the Modified Rhyme Test (MRT): a closed-response intelligibility test. Eight listeners provided responses to the MRT recordings. The overall intelligibility for the five talkers averaged 90%. Word-final consonants were more intelligible than word-initial consonants. Liquids, glides, and affricates were more intelligible than stops, fricatives, and nasals. The recorded materials were also transcribed phonetically by seven listeners. This technique revealed more errors than were obtained in the closed-response format. Overall intelligibility was 76%, with word-final consonants being more intelligible than word-initial consonants. Regardless of the method by which intelligibility was assessed, voicing errors constituted the major source of perceptual confusions. Voiceless consonants were often perceived as their counterparts. In contrast, the data revealed few place and manner errors. [Research supported by NINCDS and a grant from the Samuel W. Traylor Foundation.]
Low-frequency formant speech code for the severely/profoundly hearing impaired
Authors: Reeder K. , Strong W. J. , Palmer E. P.
Published: October 2005

A formant speech code was tested as a possible speech reception aid for the severely to profoundly hearing impaired. Three formant frequencies were extracted from speech and used to control three sine wave oscillators whose combined output was presented aurally to subjects. The formant frequencies were divided by four to place them in the residual hearing range of typical sensory-neural impaired ears and a 100-Hz bias was added to each to place it in a frequency range such as to minimize problems of acoustical coupling between headphones and the ears. The code was first tested using ten normally hearing subjects who learned from 20 to 50 of the most common English words. They also took a diagnostic rhyme test which provided a more refined test of discriminability. Six sensory-neural subjects were given the diagnostic rhyme test to check the ability of the impaired ear to discriminate the code. The tests showed positive results for both the normally hearing and the hearing-impaired subjects with the vocabulary used.
Speech perception via the tactile mode: A progress report
Authors: Saunders F. A. , Simpson C. A. , Hill W. A.
Published: October 2005

Research is in progress to determine the extent to which hearing subjects can learn to process speech presented only in the tactile mode. This approach extends the theoretical ccncepts of sensory substitution to the area of profound deafness. A 20-channel tactile vocoder was used to transform audio speech stimuli into tactile patterns on a belt worn on the abdomen. The tactile transform is analogous to a frequency-to-spatial transform with increased resolution in the F2 region. A two-choice discrimination task, with feedback, was used to train and test hearing subjects on the tactile discrimination of monosyllabic words having minimal phonetic differences. After responding to each stimulus, subjects received simultaneous tactile and auditory feedback. Results from this study and from subsequent research will be presented.
Speech reception threshold testing materials
Authors: Campbell R. A.
Published: October 2005

This study was carried out to increase the number of spondaic words available for speech reception threshold testing. Although recent trends in clinical speech reception testing has been toward use of small lists of spondaic words, with which the testee has been familiarized, it was felt that the general utility of such testing would be improved by having available a larger number of words which could be used for extensive testing without having to repeat. Particular attention was paid to the intelligibility of these additional words as compared to CID Auditory Tests W-1 and W-2. Thirty words from an experimental list of 250 spondaic words were administered to a total of 84 clinical patients at their previously established speech reception threshold. From the data, a total of 150 new spondaic words, arranged in five 30 word lists, were obtained which appear to be more homogeneous in intelligibility than the CID lists.
Pitch performance in singing without auditory feedback
Authors: Dixon Ward W. , Burns Edward M.
Published: October 2005

Subjects, both trained and amateur musicians, were asked to sing an ascending diatonic scale (from “do” to “do” upwards) and then a descending one, in the presence of various levels of noise fed to both ears via earphones. Vocal output was recorded and later analyzed. The condition of greatest interest was when the sidetone was completely masked; in this case, of course, errors increased. Departures of vocal fundamental frequencies from the target tone (assuming equal temperament based on the frequency of the first “do”) were studied in order to determine to what extent a given individual was using absolute versus relative kinesthetic feedback in the absence of auditory cues.
Effect of number and peak level of clicks on annoyance
Authors: Pierce Linda , Pavel Michael
Published: October 2005

Impulse noises (clicks) occurring within speech presented via telephone handsets were evaluated with respect to annoyance. A single type of transient was evaluated; it was of 18-msec duration, and its peak level was varied from 95 to 125 dB SPL. Between one and four of these clicks were presented on a given trial. For trials with multiple clicks, all the clicks were presented within 250 msec of each other. After each trial, subjects evaluated the clicks for degree of annoyance by rating them on a seven-point annoyance scale. To maintain the subjects' attention, they also performed a subsidiary task with respect to the content of the speech sample. The results indicated that annoyance increased as a function of the number of clicks. There was a nonlinear increase in annoyance with peak level.
Relation between noise aversiveness and activity Interference
Authors: Lerner Neil D.
Published: October 2005

An experiment that concurrently measured task interference by noise and human aversion for noise was conducted. Subjects performed a rapidly paced paired-associate learning task while pressing a telegraph key to earn points on a counter. When temporary escape from intense sounds was made contingent upon key pressing, significant differences in rate of escape responding were obtained for five acoustic conditions: white noise at A-weighted levels of 90 and 104 dB, a 1000-Hz tone at 90 and 104 dB, and silence. Intense noise impaired paired-associate performance across the first five days of the experiment; but for the subsequent five days, it actually facilitated performance. The disappearance of task interference was not accompanied by decreased sensitivity to aversive aspects of noise; rather, aversiveness increased somewhat from early to later sessions. Also rate of escape responding relative to quiet and paired-associate performance relative to quiet were positively correlated across days. Thus recovery of performance in noise is not simply due to an adaptation process that reduces the effective intensity of acoustic stimulation. The data suggest interpretations concerning noise effects, coping, and annoyance reactions.
Effects of intravehicular noise on electromyographic activity associated with the braking reflex
Authors: Bhereur P. , Mergler-Racine D.
Published: October 2005

Previous results showed braking reaction times (RTs) increased with higher intensity intravehicular noise (IN). Experiments were performed to relate electromyographic (EMG) activity to these RT changes. IN, recorded in a moving car, was reproduced at 77 and 85 dB, through a calibrated sound system, to subjects in a simulated vehicle. During 3-h exposures, subjects were required to maintain a fixed speed on the speedometer and brake when a red light lit up at random intervals. Surface electrodes recorded EMG activity from right gastrocnemius and rectus femoris muscles. Digital counters provided the braking reflex RTs. For the rectus, at 77 dB, following a latency of 0.1–0.3 sec, there was a burst of activity which tapered off. With time, the amplitude of the response diminished and its duration increased. At 85 dB, this latter pattern. generally associated with fatigue, appeared earlier in time than at 77 dB, with significantly longer duration. For the gastrocnemius triphasic response, there was a progressive decrease in activity of the first and second components which occurred earlier at 85 dB than 77 dB. These findings suggest that higher intensity IN induces increasing motor fatigue.