bluuebunny/crossref_metadata_2025 huggingface dataset:
Sound attenuation of single and double wall barriers with openings
Authors: Russell G. A. , Lewis N. D.
Published: October 2005

Sound attenuation characteristics of single and double wall barriers with openings, such as might be found along a suburban roadway, were studied using acoustic scale model experiments. The model experiments were carried out using techniques and equipment developed by Lyon [Lyon and Cann, “Acoustical Modeling in Urban Environments,” Inter-Noise '74 Proc. 457 (1974)]. A scale factor of 64:1 was employed with measurements being made above both reflective and absorptive surfaces. Experimental results correlated closely with analytical predictions based on specular reflection and simple diffraction theory. The results indicate a significant deterioration in excess attenuation for both single and double wall barriers as openings are made in the barriers.
Scattering from coated edges
Authors: Eisler T. J. , New R.
Published: October 2005

An exact solution in the form of an eigenfunction expansion has been obtained for scattering of a plane wave from a rigid wedge whose edge is surrounded by a penetrable cylinder of absorbing material. The backscattered field from a coated half-plane has been calculated in both the near- and farfields for cylinders of various sizes and absorption. The effectiveness of the coating is found to be critically dependent on the impedance match at its boundary.
Creeping waves and lateral waves in acoustic scattering by large elastic cylinders
Authors: Frisk G. V. , Überall H.
Published: October 2005

The connection between creeping wave and flat surface wave theory is established by investigating the limit of acoustic scattering from a solid elastic cylinder imbedded in a fluid, as its radius tends to infinity. After applying the Watson-Sommerfeld transformation to the scattering solution, it is shown analytically that the asymptotic expressions for the residue terms (creeping waves) in the Whispering Gallery mode series combine to form separately both the longitudinal and the transverse lateral waves on a flat surface. The cylindrical Rayleigh wave tends individually towards the flat Rayleigh wave, while the Franz and Stoneley waves disappear.
Method of measuring the acoustic properties of ship wakes
Authors: Malme C. I.
Published: October 2005

Propulsor cavitation during high sea-state conditions produces a stationary “wake” which can interfere with the acoustic navigation information required by a station-keeping drillship. A system was developed to determine the influence of this wake by measuring the sound absorption and velocity changes caused by entrained gas bubbles near the ship. The system consists of a directional projector-hydrophone pair spaced 1 m apart and connected to an electronic feedback loop. The loop gain is adjusted to approximately compensate the transduction and spreading losses. Thus when the transducers are in normal sea water a short, injected tone-burst is recycled and decays in amplitude at a reference rate set by the loop gain. When bubbles are present in the acoustic path, the tone-burst decay rate increases. Absorption loss in decibels/m is determined by noting the calibrated gain increase required to return the decay rate to the reference value. Sound velocity is determined by counting the tone bursts recycled during a selected interval. Changes of 1%–2% in sound velocity and ± 0.2 dB/m in absorption loss are resolved over a frequency range of 25 to 200 kHz.
Listener discrimination of broad-band FM echoes from single elastic targets
Authors: Biercks K. J.
Published: October 2005

Use of human listeners for analysis of broad-band echo wave-forms is described. Linear frequency modulated (LFM) transmissions were used. Five signal bandwidths were investigated, the largest being 0.625 octave. Targets were solid and hollow metallic spheres and cylinders in water. Echoes were recorded at ultrasonic frequencies and translated to the audio frequency range by time expansion. Listeners identified two echo characteristics which were used for target differentiation: the rate of amplitude fluctuation during the echo, and the structure of the echo onset. The amplitude structure characteristic, useful only with the wider bandwidth signals, separates the targets into thin- and thick-walled categories. The onset structure clue identifies target shape. Echoes from spheres have a slower rise-time than those from cylinders and exhibit an amplitude “notch” shortly after onset. This characteristic was independent of signal bandwidth.
On-line, adaptive speaker-independent word recognition system based on phonetic recognition techniques
Authors: Lin W. C. , Ganesan K.
Published: October 2005

The research reported in this paper deals with a new method of phonemic analysis of speech by statistical pattern recognition techniques and its application to the problem of Automatic Speech Recognition (ASR). An on-line, adaptive, trainable speaker-independent system is implemented using this approach. The details of the system follow: first, the beginning and end points of the speech utterance are detected. The utterance is then sent for automatic segmentation where it is segmented into the following classes: (1) voiced, (2) unvoiced, (3) transition, and (4) silence. An 11-dimensional feature vector consisting of 10 linear predictor coefficients and zero-crossing rate is extracted from these regions. For voiced and transition region, the feature extraction is done pitch synchronously and for unvoiced regions, a constant frame of 6.4 msec is used. A new phonetic unit called phoneme-pair is defined for the transition regions, while the unvoiced and voiced regions are represented using the phonemes of the IPA. Conditional probability densities for each of the phonemes and phoneme-pairs are estimated using non-parametric methods as a single polynomial in the 11-dimensional space. The classifier makes Bayes' minimum risk decision based on these probability densities. The recognition results of the ASR system are Training Set: 98.4%, Test Set: 96.0% (for speakers in the training set) and 91.0% (for speakers not in the training set). The present vocabulary of the system is 60 words and any new word can be added by entering its corresponding phonetic transcription. The adaptive and trainable characteristics of the system will also be demonstrated.
Analysis and correction of ground reflection effects in measured narrow-band sound spectra using cepstral techniques
Authors: Miles Jeffrey Hilton
Published: October 2005

Ground reflections generate undesirable effects on acoustic measurements such as those conducted outdoors for jet noise research, aircraft certification, and motor vehicle regulation. Typically, microphones have been placed a few feet above the ground leading to cancellations and reinforcements at low frequencies due to interference of the direct signal and from the ground. Theoretical treatment of ground reflections is difficult, and the problem is generally controlled by modifying experiments to include ground or high pole microphones which can be subject to operational problems. Analytical methods applicable to narrow-band sound spectra have been developed for signal processing and can be used to correct ground reflection effects. This paper shows how cepstral techniques developed in speech processing and used to find the pitch period of a voiced speech segment can be adapted to identifying the echo delay time and to correcting for ground reflection effects. Sample results are presented using both simulated and measured sound pressure level narrow-band spectra. The technique can readily be adapted to existing fast-Fourier-transform-type spectrum measurement instrumentation to provide field measurements of echo time delays.
A review of current procedures for normalizing aircraft flyover noise data to reference meteorological conditions
Authors: DeLoach Richard
Published: October 2005

Procedures which are currently used to normalize raw aircraft noise data to reference weather conditions are reviewed. These procedures sometimes result in calculated values of molecular absorption which differ significantly from measured values, especially at higher frequencies. An explanation is offered for this discrepancy, and the effect which it has on normalized sound levels and on EPNL calculations is examined. It is found that when current procedures are used to correct raw data for weather effects, EPNL values computed from data acquired under relatively cool or dry conditions can be larger than EPNL values computed from data acquired under relatively warm or humid conditions, even when both sets of data describe the same noise source and both sets are normalized to the same reference weather conditions.
Statistics of blast propagation in the atmosphere
Authors: Schomer P. D.
Published: October 2005

Results from the analysis of 10 000 recorded blast events made simultaneously in four directions at distances of 2, 5, 10, and 15 miles are given. These recordings resulted from seven hundred 5-lb charges detonated at a central location. Concurrently, meteorological data was gathered by a specially equipped FAA plane. The results are analyzed for “macro”-propagation statistics and spectral content. Use is made of the primary statistics to predict community response. Spectral content yields typical spectral distributions (such as the difference between linear and A-weighting or linear and C-weighting) as functions of level and distance. The results demonstrate that acoustics is probably a better indicator of weather conditions than are readily available meteorological conditions a predictor of general acoustical levels.
Passive signature analysis and range determination for airborne acoustic sources
Authors: Huffnagle N. P. , Brooks G. W.
Published: October 2005

An analysis technique has been developed for single point receivers of airborne acoustic sound that is capable of discriminating between various classes of acoustic sources and passively determining range. Discrimination is performed via comparison against preset source signature data. Although single sensors are used as the data collecting means, microprocessor techniques are used to sort and collect source information. Range measurement is based on first-principle observables available in the undecoded data. By using a multiplicity of data-reduction channels, the technique implemented leads to quite accurate bounding of the source's range and velocity (if any). Directional capability of the system is developed via a cluster of sensors having overlapping fields of regard. A prototype unit has been successfully demonstrated against simulated sources. Although there was no attempt to miniaturize the equipment, the main signal processing unit and associated decision logic units occupied roughly 6 in.3 Source sorting and classification was accomplished to within practical limits, and range determination accuracy was within 10% of that predicted.
Doppler effect of sources in subsonic oscillator motion
Authors: Hwang Y. F. , Tichy Jiri
Published: October 2005

The sound fields of a plane-wave source and a monopole source (both at frequency ω) in oscillatory motion (at frequency Ω) are presented. In either case, the source motion is represented in the wave equation by a Dirac delta function. Problems are solved by utilizing Fourier transform and contour integral in the plane-wave case and by utilizing Fourier-Hankel transforms in the monopole case. Solutions are represented by sidebands ω+mΩ, m=…,−2, −1,0,1,2,3,…, which means that the sound waves are modulated due to the oscillatory motion of the sources. In both cases, the amplitude of each sideband ω+mΩ can be represented by a constant factor times the Bessel function of the first kind of order m with an argument in proportion to the amplitude of the source oscillation. These sidebands represent an acoustic signal, the frequency of which changes continuously and periodically at a period of 2π/Ω. [Work supfsported by NSF at Pennsylvania State University.]
Acoustical analysis for voice disorders: An attempt towards clinical applications
Authors: Hiki S. , Imaizumi S. , Hirano M. , Matsushita H. , Kakita Y.
Published: October 2005

Glottal sound waveforms were derived by an inverse filtering method from sustained vowels uttered by 217 patients suffering from 15 kinds of laryngeal diseases and by 20 normal speakers. Various parameters which indicate individual characteristics were extracted from autocorrelation function, correlograms of time series of fundamental period and maximum amplitude in each fundamental period, and spectral envelope of the glottal sound waveform. Then, dependency on each other among acoustical parameters was examined by principal component analysis. Meanwhile, medical parameters utilized in diagnosis were evaluated quantitatively, and the relation between those and the acoustical parameters was examined also by principal-component analysis. Combining those results, more than 10 acoustical parameters which seemed to be useful for diagnosis and to be rather independent of each other were selected. Using canonical analysis the axes which discriminate the kind of disease best were obtained based on those acoustical parameters of the patients of several diseases and normal speakers, and the degree of discrimination and the contribution of each acoustical parameters to each axis were examined.
Adaptive testing of speech discrimination in persons with sensorineural hearing loss
Authors: Becker M. , Hochberg I.
Published: October 2005

The purpose of this study was to evaluate the clinical utility of adaptive methodology in measuring speech discrimination performance of CNC words in 18 persons (24 ears) with varying degrees of sensorineural hearing loss. Ss were 18 to 82 years and included average hearing levels (500, 1000, and 2000 Hz) from 35 to 85 dB HTL (ANSI-1969). Speech reception thresholds (SRT), loudness discomfort levels (UCL), and most comfortable loudness (MCL) levels for spondees were obtained for all subjects. A simple-up-and-down procedure was used to determine the 50% response level for all target levels, whereby two test words, one 5 dB above and one 5 dB below the target level, were employed. The initial target level was designated approximately midway between the SRT and UCL. Three performance-intensity configurations were observed: a steeply rising curve that plateaued for two or more estimation points; a rising curve that plateaued for two estimation points and rolled over; and a rising curve that reached PB-max at only one estimation point and then rolled over. The latter function characterized 16 of 24 ears (66%). PB-max scores were determined by adaptive testing and at MCL for spondees. Twenty-two ears (91%) obtained higher PB-max scores with adaptive testing than at MCL (Mean difference of 20%; range of 17% to 33%). The further advantages of flexibility of test procedure, shorter testing time, and fewer test trails suggest that adaptive methodology appears most suitable for evaluating speech discrimination performance in hearing impaired individuals.
Speech-reading skills of deaf children
Authors: McGarr N. S. , Levitt H.
Published: October 2005

An experimental test of speech-reading was designed to assess the relationship between speech-reading skills and language comprehension in a group of 13-year-old deaf children. The test was constructed to satisfy the criteria of simple active-declarative sentences in which the subject, object, and action could be systematically varied. Four different sentence types were generated that varied in range of difficulty. The experimental speech-reading test was administered as part of a comprehensive battery of language, speech reception and speech production tests. Results suggest that speech-reading skills are more closely correlated with language tests such as the tests of syntactic comprehension. Speech-reading was less closely correlated with speech reception and speech production measurements in this population. [Research supported by New York State Education Department, GREED Project ED 73-001 DC.]
Intelligibility of speech with simulated single reflection for normal and hearing-impaired subjects
Authors: Nabelek Anna K. , Slawinska Elizabeth , Robinette Larry N.
Published: October 2005

Simulated reflection was produced by a computer as a delayed form of the speech material (Modified Rhyme Test). The delays were 0, 5, 10, 20, 40, 80, and 160 msec. The signals were fed separately into two loudspeakers, positioned in a sound-insulated booth 6 ft in front of a subject and forming a 60° angle. The intelligibility was tested with five to nine normal and hearing-impaired subjects in binaural and monaural conditions with masking noise (babble of eight voices) reducing intelligibility to about 75%. The intelligibility of normals did not change up to 20-msec delay and decreased for longer delays for both conditions. Scores were lower for monaural listening. Such an advantage for binaural listening was not observed for the impaired listeners. In both conditions, their intelligibility decreased gradually as the duration increased. The lack of plateau in the impaired subjects' results might indicate that any reflected energy in reverberant rooms is harmful for their intelligibility. Normals, however, might benefit from early reflections. [Supported by NIH NISND grant 1 R01 NS 10962-01.]
Phoneme errors on a nonsense syllable test
Authors: Resnick S. B. , Dubno J. R. , Hoffnung S. , Levitt H.
Published: October 2005

In an effort to construct a test which permits detailed analysis of the phonemic errors made by naive hearing-impaired individuals under conditions of hearing aid use, we have developed a closed response nonsense syllable test consisting of CV, VC, and CVC syllables organized into modules. The modules differ with respect to the class of consonants represented, the consonant position within the syllable, and the vowel context (/i/, /a/, or /u/). Responses to syllables within a given module are limited to syllables within the same module. The response foils (which correspond to all the syllables within a module) include the most frequent perceptual confusions reported for normal-hearing and hearing-impaired individuals. The resulte of tests administered to hearing-impaired individuals under difficult listening conditions indicate a pattern of errors consistent with those reported by previous investigators, including (1) greater errors on final than on initial consonants; (2) greater errors on unvoiced than on voiced consonants; and (3) a dependence of consonant identification on vowel context. [Research supported by NINCDS.]
Acoustical study of certain parameters of voice quality and their influence on voice classification in male singers
Authors: Cleveland T. F.
Published: October 2005

An investigation to determine the significance of formant frequencies, pitch, and source spectrum slope on voice classification is reported. Eight professional singers sang five common vowels on four common pitches, and, in a forced-choice test, vocal pedagogues classified the vocalizations as tenor, baritone, or bass. Source spectrum slopes of −12 dB per octave approximated the vocalizations of the three voice classes, while formant frequencies and pitch accounted for most of the difference in jury evaluation results. To evaluate more accurately the significance of formant frequencies, pitch, and source slope on voice classification, vowel synthesis was employed. The independent variables, representative voice class formant frequency combinations, four different pitches, and source spectrum slope were systematically altered as dependent variables, vibrato, and vowels were stabilized. The original jury classified the synthesis in a forced-choice test, and the dependence of the jury evaluation upon formant frequencies and pitch was evidenced in percentage and analysis of variance tests. Formant frequency percentage differences between basses and tensors were similar to those found between males and females. The results suggest that voice classification in male singers is dependent on at least one significant property: the formant frequencies of the singer.
How can vowel systems differ?
Authors: Papçun George
Published: October 2005

On what parameters may vowel systems differ? To seek a partial answer to this question, formant data from two five vowel languages, Spanish and Japanese, are compared. Three mode factor analysis is demonstrated as a method of normalizing the data and extracting parameteric differences between languages and among speakers within languages.
Modifying fundamental frequency envelopes
Authors: Melby A. K. , Millett R. , Strong W. J. , Lytle E. G.
Published: October 2005

An experiment was designed to investigate the hypothesis that changes in fundamental frequency alone are sufficient to cue several readings of a sentence. A sentence was recorded and digitized in four versions and analyzed by a predictor coefficient method. First, the sentence was synthesized with its natural F0 envelope. Then it was synthesized with its F0 envelope being replaced by artificial ones patterned after the envelope of one of the three stressed versions. In each case, the synthesized output was clearly perceived to be the stressed version whose F0 envelope had been borrowed, although other prosodic features such as duration, pause, and amplitude were unaltered. This experiment seemed to support the importance placed on F0 by a recent article by Olive [J.P. Olive, J. Acoust. Soc. Am. 57, 476–482 (1975)].
Ear structure and sensitivity compared in tropical and desert-dwelling heteromyid rodents
Authors: Webster D. B. , Webster M.
Published: October 2005

Spiny pocket mice, rodents of tropical jungles, are closely related to desert species we have studied (kangaroo rats, kangaroo mice, and pocket mice). All middle ear features are considerably smaller in the tropical heteromyids than in their desert-dwelling relatives. The middle ear cavity is smaller both absolutely and relative to animal size. However, the area ratio (stapes footplate to tympanic membrane) is the same, and the lever arm ratio (incus to malleus) is significantly greater in the tropical species. The theoretical transmission efficiency of the middle ear is 78%–86% in tropical heteromyids and 94%–100% in desert heteromyids. In the organ of Corti, desert heteromyids have specialized border cells not found in tropical heteroymids; but the zona pectinata of the basilar membrane is found in all heteromyids, including tropical species. As might be predicted from this morphology, cochlear microphonic data demonstrate, among the heteromyids, a greater overall sensitivity in desert species, and a higher-frequency range in tropical species. [Work partially supported by NIH grants NS-05800 and NS-11459.]
Behaviorally determined auditory sensitivity of the mongolian gerbil
Authors: Ryan Allen
Published: October 2005

Thirteen mongolian gerbils (Meriones unguiculatis) were trained in a shock avoidance task based on that developed by Miller [J. Acoust. Soc. Am. 48, 513–523 (1970)] for the chinchilla. The gerbil proved to be readily trainable and behaviorally stable. Thresholds were obtained by a method of limits, and in the case of one animal, by generating psychometric functions, from 0.1 to 60.0 kHz. For pure tones between 1.0 and 16.0 kHz, the threshold of the mongolian gerbil is about 4.6 dB SPL. Between 1.0 and 0.1 kHz, sensitivity declines at a rate of 12 dB/octave. Between 16.0 and 40.0 kHz, threshold increases at a rate of 13 dB/octave, and above 40.0 kHz at a rate of 66 dB/octave. Thresholds derived from psychometric functions were slightly lower than those obtained by the method of limits. [Work supported by grants from the NINDS. I thank Carol Sanes for her contribution to this research.]
Perception of biologically meaningful sounds by dogs
Authors: Heffner H.
Published: October 2005

Dogs were trained to classify sounds into two categories on the basis of the nature of the sound source. The categories were dog sounds and nondog sounds—specifically, barks, whines, whimpers, growls, and pants versus various mechanical sounds, and mammalian, avian, and amphibian vocalizations. Once the dogs learned to classify 32 different sounds, they were given tests of their ability to classify 96 novel sounds into the same two categories. The results indicate that that the animals did indeed classify the sounds on the basis of their source (i.e., dog versus nondog) rather than on differences in the physical properties of the sounds themselves (e.g., frequency, intensity, timbre). This result shows that dogs, like humans, find the classification of sounds quite easy, providing the sounds are biologically relevant to them. [Supported by grant HD 02528 to the Bureau of Child Research, University of Kansas.]
Noise and vibration assessment for the Denver transit study
Authors: Hanson C. E. , Wittig L. E. , Graf P. J.
Published: October 2005

Noise and vibration considerations entered the planning process for Denver's proposed automated Rapid Transit System at four stages—alternative analysis, environmental overview, environmental assessment of system and environmental assessment of first segment. Such considerations include comparison of noise sources for various transit concepts, establishment of noise and vibration criteria, noise impact predictions, and assessment and noise abatement measures and costs. Establishment of noise criteria was a result of the need for being responsive to three requirements: provide an environmental noise impact assessment, provide an assessment of the potential for annoyance, and provide noise design goals for the new system. The environmental noise assessment was performed by applying the recently Introduced Fractional Impact Method.
Air pressure transients in the BART transbay tube
Authors: Saurenman Hugh J.
Published: October 2005

Air pressure transients caused by rapid transit system operations in subway tunnels have long been recognized as a potential source of both patron discomfort and equipment failure. A study of the pressure transients inside BART trains during normal revenue service through the Transbay Tube was performed. The static air pressure was sensed by means of a high sensitivity static pressure transducer, with amplified and frequency-modulated signal being recorded on a magnetic tape recorder. For analysis, the pressure data were reproduced on a strip chart recorder. Measurements were taken simultaneously in the leading and trailing cars of the complete time history of static pressure on a number of trips between the Oakland West and Montgomery Street Stations. The analysis indicates that the pressure transients during a high-speed passage of the Transbay Tube do considerably exceed the criteria that have been established for passenger comfort. The criteria are exceeded at high-speed entrances and exits of the Tube portal, and high-speed passbys of the vent and blast shafts. Pressure transients were measured with magnitudes up to 11.1 in. H2O and rise rates as high as 5.5 in. H2O. Presented along with the sample strip chart recordings of the measured pressure transients will be a summary of the preliminary recommendations for the control of the magnitude and rise rates of the pressure transients.
Experimental demonstration of the absorption of sound by sound in water
Authors: Konrad William L. , Moffett Mark B. , Carlton Lynn F.
Published: October 2005

Intense low-frequency sound can severely attenuate sound of higher frequencies through nonlinear interaction. On the other hand, intense high-frequency sound does not appreciably alter a lower-frequency sound field. An experiment is described wherein a 65-kHz underwater sound beam was made to propagate collinearly with a 240-kHz beam, demonstrating the absorption of sound by sound. [Work supported by NAVSEA.]
Parametric receiving array and the scattering of sound by sound
Authors: Truchard J. J.
Published: October 2005

The scattering of interaction frequency components from two sound waves interacting at a nonzero angle has been the subject of controversy for the last two decades. In this paper, the subject of scattering of sound by sound is related to the parametric receiving array in order that we might better understand the problem. First, the solution for a high-frequency plane wave interacting with a low-frequency plane wave over a half space is discussed. The high-frequency sound is assumed to originate at a planar boundary and propagate away from it. The interaction-frequency components at some distance from this planar boundary are found. This solution for two plane waves is in turn related to a solution for parametric receiving array with a low-frequency plane signal wave and a high frequency, narrow beam pump wave. The two solutions are found to have very similar properties. Some experiments as well as solutions for various configurations of the parametric receiving array are discussed in light of the problem of scattering of sound by sound. Also, some of the foibles associated with such measurements of two interacting waves are discussed using the theoretical results for the parametric receiving array. In this work, interaction components are found to exist even when the two sound waves are not propagating in the same direction. However, since both sound waves are present throughout the volume of interest, no comment can be made about the existence of interaction components outside of the interaction region.
Experimental determination of the Raman-Nath parameter using near- and far-optical fields of an ultrasonically produced optical phase grating
Authors: Haran M. E.
Published: October 2005

A technique has recently been developed [B.D. Cook, FDA Contr. 223-74-6039] to determine the Raman-Nath parameter of an ultrasonically produced phase grating. This technique requires measurements of the irradiance in the near-optical field of the grating. A 1.23-cm diameter, 2-MHz ceramic transducer was used to produce a progressing optical phase grating. At each of the values of voltage applied to the transducer, the Raman-Nath parameter was determined using both the near optical-field technique and the traditional farfield or Raman-Nath technique. The values obtained from each technique were compared over a range of 0.75 to 5. 0. Differences between the techniques were found to be less than 5%. Also discussed are applications of the nearfield technique to the measurement of acoustic intensity output from medical devices, an activity related to the enforcement of the Radiation Control for Health and Safety Act of 1968.
Deaf leading the dumb: A directive interview for helping the naive person identify impairments in speech transmission systems
Authors: Duncanson James P. , Labder Diane Z.
Published: October 2005

Evaluating telephone connections from the point of view of the user is a longstanding and difficult problem. Determining the perceptibility, prevalence, and bothersomeness of any particular impairment requires first that the impairment be identifiable. What is needed is a way of helping an interviewer (who is “deaf,” in that he cannot hear what the user hears) ask the user (who is “dumb,” in that he does not adequately describe what he hears) about the connection in a way that will lead him to discriminate between impairments. A directive interview was developed for this purpose. The user is asked a series of branching questions, with the last question in each branch being a set of descriptions from which he chooses that which matches what he hears. In a laboratory study, this method yielded a median of 83% correct identifications compared with 57% correct for a conventional method in which the user simply described what he heard and this description was categorized.
Quality improvement of bandlimited speech by adding higher-order harmonic components produced by nonlinear circuit
Authors: Kitayama S.
Published: October 2005

In this paper, the system to improve the quality of the bandlimited speech (0.3–3.4 kHz) through a telephone channel by adding the higher order harmonic components (3.4–7.0 kHz) produced by the simple nonlinear circuit is described. The restoration of the bandlimited speech is made based on the fact that most of the voiced sounds have periodicity. At the receive side, the higher-order harmonic components are produced from the transmitted bandlimited speech by the nonlinear circuit and are added to the bandlimited speech with shaping in tho frequency domain. This nonlinear circuit operates in the way that the periodicity of speech is preserved and the output level is directly proportional to the input level. This system mainly improves the naturalness of the bandlimited speech. The subjective evaluation test using the samples of the speech announced in Japanese proves that improvement of scale value 0.5 by the method of paired comparison with five quality categories is obtained. This system also will be applied to the quality improvement of bandlimited music.
Larynx height in Kwa
Authors: Lindau Mona
Published: October 2005

The vertical positions of the larynx in various vowels in Akan, Igbo, and Ijo were measured from radiographic data. These languages have a vowel harmony such that one set of vowels [i, e, u, o] is produced with an expanded pharynx, and another set [ι, ɛ, o, ɔ] with a constricted pharynx. The major function of vertical larynx displacement is to contribute to the variation in pharyngal size for categorizing vowels into harmonizing sets. The larger pharynx is produced as a combined effect of advancing the tongue root and lowering the larynx, and the smaller pharynx is produced by retracting the root and raising the larynx. The vertical position of the larynx is reasonably well correlated with the position of the tongue root. But within each harmony set there is no consistent relationship between larynx height and vowel height or between larynx height and backness. Sometimes phonologically high vowels have a higher larynx than nonhigh vowels, sometimes not. The only possible exception is the vowel [a]. In those languages where [a] occurs, there is a strong tendency for it to have the highest larynx position of all the vowels.
Mandibular movement in the sagittal plane and its consequences for biomechanical models of speech and nonspeech tasks
Authors: Ellenberg Gerald R.
Published: October 2005

Mandibular movement was measured utilizing a tooth splint, a mandibular extension and a pair of two-dimensional movement transducers. The movement paths were reconstructed in the sagittal plane using computer graphics and the regularity of the movement was calculated for several speech and nonspeech tasks. The instantaneous center of rotation of the mandible was calculated and will be compared with other published results. The range of movement of the condyle was calculated and the implications for a biomechanical model of the temporomandibular joint will be discussed. Hypothetical changes in the length and direction of the mandibular musculature were calculated and are presented in the context of a biomechanical model of the jaw. [Research supported by an NINDS grant.]
Force transformation model for the bridge of acoustic lute-type instruments
Authors: Houtsma Adrianus J. M. , Boland Robert P. , Adler Nicholas
Published: October 2005

Instruments of the lute type are characterized by a rather flat bridge on which the strings terminate and which is acoustically coupled to the sound chamber. The model assumes that the string acts as a simple elastic device and that its force on the bridge is at all times given by its instant elongation and angle at the bridge. The force transformation is nonlinear because (1) for any simple mode the string elongation is not proportional to the deflection amplitude and (2) the variable force component on the bridge in the direction of the string is full-wave rectified. For a particular string excitation, both orthagonal force functions were computed and harmonically analyzed, and weighted with empirical force-to-sound transfer functions obtained from two different guitars. Predicted outputs were compared with short-time Fourier transforms of open string notes played on the same two instruments with similar excitations. The model predicts the well-known tone difference resulting from a free stroke and a rest stroke on the guitar. [Work supported in part by a grant from Norlin Music, Inc.]
Analysis and synthesis of oboe tones by the linear prediction method
Authors: Dudley J. D. , Strong W. J.
Published: October 2005

The linear prediction method has recently been used with considerable success in the analysis and synthesis of speech sounds. The purpose of this paper is to describe its application to musical tones in general, and to the analysis and synthesis of oboe tones in particular. Preliminary indications are that five predictor coefficients are adequate for the synthesis of good oboe tones, and in some ways are better than 12. Adaptation of the analysis-synthesis algorithms and determination of the optimum number of predictor coefficients for various pitches and tone dynamics is discussed. Tones synthesized with various numbers of predictors coefficients are demonstrated.
Computer modeling of human crash kinematics
Authors: Karnes R. N. , Tocher J. L. , Twigg D. W.
Published: October 2005

The idea of designing a computer program for the simulation of the gross motion of human crash victims has received much attention during the past decade. The existence of reliable programs would drastically reduce the need for extensive full-scale crash testing in order to determine the effects of vehicle design perturbations on human trauma. A variety of reasons exist which so far have precluded the use of crash victim simulation programs in lieu of full-scale tests. There currently exist no crash victim simulation programs which can accurately reproduce an experimental test without considerable “tuning.” The choice of program input parameters and internal physical models is very much a matter of physical intuition. Once reasonable correlation of computer results with full-scale test results is attained, the model is assumed to be validated. However, there is an increasing body of evidence which suggests that full-scale tests utilizing anthropomorphic dummies lack sufficient repeatability so that the use of a single test to validate a computer model is questionable at best. A survey of the current state of crash victim simulation is presented, with emphasis on areas in which improvements are needed.
Modes of motion of a finite element brain model
Authors: Ward Carley
Published: October 2005

A structural dynamic model of the human brain has been developed using finite element techniques. In the modeling process, soft tissue and internal fluids are discretized into solid 8-node elements. One dimensional nodal constraints and membrane elements represent skull and internal folds of the dura, respectively. The geometric shapes of the cerebri, cerebellum, and brain stem are carefully duplicated. As a consequence of this emphasis on geometric detail, more than 1000 degrees of freedom are utilized. Linear elasticity and small deformations are assumed. The dynamic characteristics of the brain model are investigated. Symmetric and unsysmmetric natural frequencies and normal modes are presented, and their effects on brain response and injury are discussed. The rotational character of the lower modes clearly demonstrates the importance of head rotation in brain injury research. A comparison of modes computed with and without the internal folds of dura (the falx and tentorium), reveals that these membranes appreciably influence the response of the brain.
Noise control in new plant design
Authors: Thumann A.
Published: October 2005

A noise control program for new plant design has been implemented by an engineering-construction firm. The results of the program indicates that when designers and specifiers of equipment are aware of noise objectives, noise control measures are implemented more economically. In the design of a 100-million-dollar industrial complex, corrective measures were taken during the feasibility stage. During the procurement phase, competitive equipment was evaluated based on its effect on the acoustical environment. In several instances, the specifier of the equipment worked closely with the equipment manufacturer. This relationship, in the selection of fans, led to the use of slower speed fans with different blade types. The emphasis placed on motor noise paved the way for one manufacturer to offer a “quiet line motor.” By changing the motor insulation and redesigning the motor cooling fan, a significant noise reduction was achieved. A comprehensive computer program system was developed to help the design engineer evaluate several alternatives. The results of the program and its effect on plant costs will be presented.
Acoustically induced safety valve vibration
Authors: Brittain F. H.
Published: October 2005

Safety valve manufacturers generally recommend installing safety valves a minimum of eight to ten pipe diameters down stream of a fitting. Space limitations often require their installation much closer to a fitting. When so installed, acoustically induced vibration often leads to problems with noise, vibration, valve chattering, or valve integrity. Based upon measurements of safety valves exhibiting acoustically induced vibration, a procedure for installing safety valves was developed. This procedure involved alternating the lengths of inlet nozzles and specific details for their attachment to the steam line. Measurements of a successful installation of this procedure at a fossil-fueled power plant are presented and compared with the measurements made before the revised installation.
Application of a computer model for predicting remote noise levels
Authors: Judd S. H. , Dryden S. L.
Published: October 2005

The prediction of noise levels at selected remote locations is an integral part of estimating the environmental impact of new stationary sources or of noise reduction for existing facilities. A three-dimensional computer model, with provisions for temperature and humidity variation and for solid barriers and semitransparent arrays such as process plants or forests, predicts the composite noise level produced at remote locations by many sources. The output also shows the individual octave band contributions of the more important sources, ranked by their A-weighted level at each observer location. These features enable the model to be used for a variety of purposes, including Environmental Impact Statements, site selection and layout of process plants, determination of required noise limits on new equipment, estimation of the effect of major noise barriers, determination of the feasibility of meeting noise ordinances, and optimization of cost-effectiveness of noise control measures.
Echolocation for perception of the environment
Authors: Simmons J. A.
Published: October 2005

Microchiropteran bats, many cetaceans, and several species of birds have independently evolved active sonar for acoustic imaging of their surroundings. Echologcating birds emit acoustic clicks at 1 to 10 kHz for orientation. Echolocation enables birds to detect wires as small as 1.0 mm in diameter, providing a perceptual acuity appropriate for obstacle avoidance and navigation. Echolocating bats comprise about 20% of all mammalian species and exhibit sophisticated adaptations of hearing which led to early success and diversity. Two classes of bat orientation signals are used for different purposes: FM sounds for target description and localization, and CF sounds for improving target detection and determining target velocity from echo Doppler shifts. These signals occur in several patterns (FM-CF, CF-FM, FM) that relate to acoustic environments. Using broad bandwidths and high frequencies (20 to 150 kHz) to exploit the superb ultrasonic hearing of mammals, bats can resolve wires as small as 0.1 mm and can perceive fine target features. Bat echolocation appears primarily adapted to acoustic constraints influencing target detection (long ranges or small targets). Cetaceans use different kinds of orientation signals; short clicks of broad bandwidth and high frequencies (10–150 kHz) being best known. These clicks, at least, seem adapted for fine resolution rather than long-range detection in the highly reverberant underwater environment. Difficulties in observing behavior and conducting experiments limit available data on cetaceans. These data indicate that fine target resolution is possible.
Design considerations for digital systems for underwater acoustical measurements
Authors: Baker D. D. , Kodosky J. L. , Nowlin W. C. , Shorey J. L. , Truchard J. J.
Published: October 2005

This paper compares several digital measurement approaches which have been used for underwater acoustical measurements and experiments. The advantages of the digital measurements systems include (1) flexibility in data reduction, storage, and presentation; (2) flexibility in interaction with sophisticated signal processing and display systems; and (3) the capability for close comparison between theoretical and experimental results because experimental parameters can be used as inputs to a theoretical calculation. In this paper, three digital systems are considered: (1) a calenlator-based system, (2) a mini-computer based system with a minimum of support hardware, and (3) a larger mini-computer system, which includes a disk. The calculator-based system uses primarily existing measurement instruments with digital outputs; in other words, most of the signal processing is done with conventional analog instruments. The smaller computer systems use fixed routines that are linked to a BASIC program. This system allows for operator interaction by using the BASIC interpretative langauge. Subroutines written in FORTRAN and machine language can be included to enhance the speed and performance of the measurement system, but such subroutines are not easily modified in the field. The computer/disk system provides the most flexibility in the FORTRAN as well as BASIC programs can be written on-line by the operator. The application of these three systems to specific acoustical measurements is described.
Water-flow-induced cavity resonances
Authors: DeMetz F. C. , Farabee T. M.
Published: October 2005

The mechanisms of flow-excited resonances in simple underwater cavity systems were investigated. The resonance frequencies of the internal fluctuating pressure field were determined as a function of turbulent boundary-layer thickness, flow velocity, cavity wall stiffness, and opening dimension. Expressions are derived for the peak forcing function amplitude and Stouhal number as a function of boundary-layer thickness and opening dimension. The results are compared with measurements on similar cavity systems in air.