bluuebunny/crossref_metadata_2025 huggingface dataset:
Formant discriminating mechanism suggested by factor analysis of vowel spectra
Authors: Baker Wm. J. , Rozsypal A. J.
Published: October 2005

An objective basis for perceptual distinctions among vowels was developed by a principal factors factor analysis of the spectra of English vowels in the neutral h(V)d consonantal context, produced by eight speakers. This analysis resulted in a five-factor solution. Two factors were related to speaker distinctions. All the three vowel discriminating factor profiles show single pronounced peaks, mutually shifted in frequency. Frequency ranges of the first three vowel formants cover the lower frequency slopes of the three corresponding factor peaks. This configuration resembles, for each formant, a single tuned circuit frequency discriminator as used in demodulation of frequency-modulated signals.
Effects of voice-onset time duration, and number of adaptor repetitions on the scaling of stop consonant voicing
Authors: Ohde Ralph N. , Sharf Donald J.
Published: October 2005

This investigation examined the effects of duration and number of repetitions of the adapting stimulus on the voicing feature scaling of stimuli varying in VOT before and after adaptation. The adapting stimuli and trials were varied as follows: (1) the adapting stimulus was either 5, 25 or 55 ms VOT, and (2) the number of repetitions during each adaptation trial was 5, 32 or 95. Fifteen subjects used a six-point rating scale in responding to both preadaptation and adaptation identification stimuli. The major findings of this study were as follows: (1) The 55-ms adaptor was rated as p-like and the 5 and 25 ms adaptors were rated as b-like, (2) generally greater phonetic boundary shifts occurred after adaptation with the 55-ms adaptor than with the 5- or 25-ms adaptors and with the 5-ms adaptor than with the 25-ms adaptor, (3) significantly greater phonetic boundary shifts occurred with 32 and 95 repetitions than with 5 repetitions, and (4) significant changes in rating responses occurred for other than boundary stimuli. The results support the following conclusions: (1) Rating changes support a fatigue type model that assumes modification in the sensitivity to boundary and end point VOT stimuli, (2) effects of adaptor repetition support an auditory component of voicing analysis, and (3) adaptor percept influences direction of shift while the acoustic nature of the adaptor influences magnitude of shift.
Speech and nonspeech percepts from the same sound
Authors: Isenberg D. , Liberman A. M.
Published: October 2005

The minimal pair /ra-la/ can be cued by the direction of the third formant transition alone. When the portion of the acoustic signal common to the members of the pair (the base) is presented to one ear and the F3 transition differentiating the members of the pair is presented to the other ear, /ra/ or /lla/ is heard contingent upon the direction of the F3 transition. Interestingly, the speech sound is typically heard on the side to which the base is presented, and in addition a nonspeech sound (typically described as a frog croak or a cricket chirp) is also reported on the side which received the isolated F3 transition. Thus the isolated F3 transition is heard as a speech sound and a non speech sound simultaneously. This phenomenon is similar to one briefly described by Rand [J. Acoust. Soc. Am. 55, 678–680 (1974)] in a study of stop consonant transitions. When the F3 transition changes in intensity listeners report hearing a change in the loudness of the nonspeech percept. To elicit reports of changes in the loudness of the speech it is necessary to change the intensity of the base. Thus the intensity of the F3 transition cue—the only cue to the difference between /r/ and /l/—does not appear to affect the perceived loudness of the phonetic segments or of the syllable. Further studies were performed to investigate some of the limits of this phenomenon. [Work supported by NINCDS and NICHD.]
The relative pitch of 30 English monosyllabic words: The rank order in comparison with a proposed model
Authors: Asp C. W. , Berry J. S. , Bessel C. S.
Published: October 2005

Ten studies between 1829 and 1919 ordered the vowels /u, o, a, e, i/ from low to high pitch, respectively. More recent research is consistent with this rank order. Peterson and Asp reported a rank order for prevocalic consonants based on pitch differences. From these studies and clinical experience, a hypothetical pitch (“tonality”) model was constructed that divided consonants, vowels, and diphthongs into pitch categories. From this model, 30 monosyllabic words were selected, ten each from the low, middle, and high pitch categories. The phonemes within each word were from the same category, e.g., /rum/ for low pitch and /sis/ for high pitch. The 30 words were recorded in 435 pairs. Nineteen normal hearing listeners selected the word from each pair that was higher in pitch. Using a matrix, the 30 words were rank ordered from low to high pitch. The rank order agreed with the proposed model. Statistical analysis separated the words into groups that corresponded to the low-, middle-, and high-pitch categories. [Work supported in part by Biomedical Science Support Grant.]
Labeling and discrimination of monotone and changing F stimuli
Authors: Shockey L. , Lehiste I.
Published: October 2005

In a previous experiment [Lehiste, Proc. 9th ICA p. 522 (1977)] it was demonstrated that fundamental frequency change on synthesized words influences the perception of the final consonant in a continuum ranging from beat to bead: the crossover from the perception of beat to that of bead occurs at a shorter /i/ duration for stimuli with changing fundamental frequency. The present study expands the previous one in two respects. One-step, two-step and three-step discrimination tests were run for the beat-bead continuum, synthesized with vowel durations ranging from 150 to 350 ms in 20-ms steps. Labeling tests were accompanied by production tests, in which subjects were asked to imitate the word they heard. While the labeling tests gave near-categorical results, both the discrimination tests and production tests indicate some ability to differentiate between stimuli falling in the same word category.
Discriminant analysis on an imitation dialect
Authors: Papcun George
Published: October 2005

A data set consisting of phoneticians' limitations of a set of target vowels was LPC spectrum analyzed and quantized into 19 one-third-octave bins. These data were treated by discriminant analysis. Depending on the input data set, classification accuracy varied from 88.5% to 99.5%. Vowel categories are found to be arranged in discriminant function space much as they would be expected to appear on a phonetic vowel chart or a F1/F1 space except at the “front-back” dimension is relatively foreshortened. The results are interpreted as an analytic form of the dispersion hypothesis, according to which the vowels of a given language are maximally dispersed in vowel space [cf. J. Liljencrantz and B. Lindblom, “A numerical simulation of vowel quality systems,” Language 48, 839–862 (1972)]. [Work supported by NSF.]
Eigenray interpolation for determining coherent pressure levels for broadband analysis
Authors: Weinberg H.
Published: October 2005

The difficulties that existing models have in predicting coherent pressure levels for broadband computations is discussed, and an alternative technique is suggested. It is assumed that the acoustic field has a raylike representation that can be determined at a few fundamental frequencies. An interpolation algorithm provides additional coherent levels. Unlike other methods, multipath interference does not degrade accuracy since the interpolation is performed on individual eigenrays, rather than on the total pressure. Two examples illustrate this procedure. First, the autocorrelation of a received signal is determined by taking the inverse Fourier transform of the power spectrum density. Then, a frequency-time display simulates the received signal of a passing noise source.
A numerical treatment of underwater wave propagation with arbitrary boundaries and boundary conditions
Authors: Lee D. , DiNapoli F. R. , Papadakis J. S.
Published: October 2005

Numerical ordinary-differential-equation (ODE) methods have been shown to have distinct advantages for solving parabolic wave equations. In this presentation, we describe how to apply ODE approaches to handle arbitrary boundaries and boundary conditions associated with parabolic wave equations. A procedure has been developed to handle general boundary conditions and to advance the solution in a nonrectangular region all by ODE methods and plane trigonometry. Application of ODE methods is aiming at range-dependent problems; since these methods are quite general purpose, the range-independent cases are automatically accommodated. The effectiveness of ODE applications will be shown by a problem with irregular bottom and Neumann bottom boundary condition.
The measurement and use of machinery vibratory source characteristics
Authors: Kern Fred R.
Published: October 2005

Norton's and Therenin's theorems as applied to simple mechanical systems will be reviewed. Measurement techniques to determine blocked force, free vibration, and coupled intermediate impedance system response will be presented. The relative merits of these characterizations will be compared on the basis of ease of measurement and utility of application of results to predicting installed machinery vibration. Specific examples of source characteristics for mechanical equipment such as diesel engines, pumps, and compressors, fans, and hydraulic equipment will be presented.
Noise generation by rain on lightweight roofs
Authors: Hirtle Parker W. , Vér István L. , Watters Bill G.
Published: October 2005

Raindrops impacting on the lightweight roofs of buildings, such as performing arts facilities and sports arenas, may generate noise of sufficient intensity to interfere with hearing conditions. This paper identifies the key physical parameters that control the noise generation process and presents an analytical model that permits the prediction of the intensity and spectral distribution of rain-induced noise.
The Coupling of Ground Vibration to a Building Structure
Authors: Grootenhuis P.
Published: October 2005

Good coupling of ground-borne vibration to a building structure is obtained with piled foundations and with a raft and casement construction. The magnitude of the response of the structure depends on how close the natural frequencies of its main components are to the predominant input frequency. With separate foundation blocks or footings dynamic coupling through the soil can arise between adjacent foundations in different modes. Results will be given of computation of the coupling between foundations resting on an infinite elastic half space and a comparison with some experiments will be made. The transmission of vibration through the buildings will depend upon the degree of damping. Some estimates will be given of the Q factors for modern buildings. Where the expected vibration levels in a building due to ground borne vibration are too high, a degree of decoupling can be achieved at the foundations by mounting the structure on rubber bearings and examples will be shown of actual installations.
Architectural acoustic measurement by impulsive excitation
Authors: Tachibana H. , Ishii K.
Published: October 2005

For measurements in architectural acoustics, impulsive excitation is often used. In this study, the theoretical relation between impulsive excitation and stationary random excitation in a linear acoustic system was considered initially, and the following relation was found: 〈y2(t)〉 = N  ∫ 0∞ h2(t)dt. In this equation, the left side is the square mean of the response to stationary random excitation, and the right side is the square integration of the response to impulsive excitation where (N is the noise power per unit frequency). This relation shows that the same result can be obtained either by stationary random excitation or by impulsive excitation in the measurement of intensity distribution in acoustic propagation systems. In order to confirm experimentally the fact mentioned above, measurements of SPL distribution in a room and the vibration propagation property of a building structure were made using these two methods of excitation.
Noise prediction in the city area
Authors: Kaku J. , Yamashita M.
Published: October 2005

The distribution of houses plays an important role to the noise propagation in the city area. The purpose of this study is to make clear the tendency of road traffic noise propagation into the city area, and it is carried out by means of scale model and field measurements. In model experiments, an incoherent line source (jet noise type) is used as road traffic noise. Houses are defined to be only one or two story houses and are distributed in random order. In this model of the residential area, the spacial distribution of sound pressure levels at the specified distance from the road are observed. From the experimental results, we can derive an empirical equation which gives the excess attenuation caused by houses as a correction term. This correction is given as a function of the density of houses, the source height and the distance from the road. Predicted sound level obtained by using the correction term agrees well with the result of the field measurements.
A unified statistical study of road traffic noise with many types of vehicles by use of an Erlang distribution model
Authors: Nishimura M. , Ohta M. , Yamaguchi S.
Published: October 2005

As is well known, on a traffic noise problem approached with a systematic background of research, the exponentially distributed and the equally spaced vehicles models are very often used. The two typical models correspond, respectively, to two idealized situations of road traffic flow, that is, a probabilistic mode with perfect freedom of flow and a deterministic mode of equally restricted vehicle interval. But it admits without doubt that the actual mode of road traffic flow is somewhere between these two idealized situations. This fact means that the generalized model of traffic flowing with an Erlang distribution should be used for the actual traffic flow, since an Erlang distribution model can contain the above two situations as specialized cases. In this paper, at first we consider the road traffic noise produced by the generalized traffic flow whose interval between two vehicles is distributed by an Erlang distribution and types of vehicles are alloted according to a multinomial distribution. After deriving the moment generation function of the road traffic noise theoretically, the correctness of the theoretical result is confirmed about several moments by use of the digital simulation technique.
A new trial for statistical estimation of road traffic noise in an arbitrary sound propagation environment by use of Stratonovich's viewpoint for random points system
Authors: Yamaguchi S. , Ohta M. , Hatakeyama K.
Published: October 2005

Up to now, many approaches for estimating the statistics of road traffic noise have been carried out by the introduction of several models, e.g., an equally spaced vehicles model, an exponentially distributed vehicle model, and an Erlang distribution type model in a simplified sound propagation environment such as a free sound field. However, in several studies based particularly on the latest Erlang distribution type model (i.e., gamma headway distribution model), only the first- and second-order moments of the sound intensity fluctuation which can be derived from the statistical information on the location of merely one and/or two vehicles (e.g., the distributions for the position of the vehicle and the distance between two arbitrary vehicles flowing in the same direction) are taken individually into consideration. On the other hand, the higher-order statistical properties of traffic noise are rather important in order to investigate the whole shape of the noise distribution form from which any noise evaluation index can be derived. Thus, in this paper, our main interest is devoted to considering quantitatively the relation between the multidimensional correlation properties of the sound intensity and the higher-order information on the vehicles flow by use of Stratonovich's stochastic theory for a random points system. Furthermore, the relation between our theoretical result and well-known previous studies is discussed with experimental confirmation for several lower-order moments.
Sound communication and auditory physiology in anurans
Authors: Capranica R. R.
Published: October 2005

Behavioral field studies involving playback of natural and synthetic calls indicate that anurans (frogs and toads) respond selectively to the calls of their own species. Furthermore they can localize a sound source with remarkable accuracy (despite the relatively small interaural distance between their eardrums). Electrophysiological studies reveal that their peripheral auditory system is tuned to the spectral and temporal features in their species-specific calls, and that their ear is directionally sensitive. Recordings from central auditory nuclei verify a hierarchical organization in encoding complex sounds of biological significance as well as the existence of binaural interaction subserving sound localization. These studies of anurans serve as a model for neural processing in the auditory systems of higher vertebrates. [Supported by NIH Grant NS-09244.]
Propagation characteristics of acoustic shock pulse in Fe-Cr-Al alloy
Authors: Yasumoto Y. , Takeuchi R.
Published: October 2005

High-damping materials have become to be interesting for noise control and vibration damping. Fe-Cr-Al alloy is called Silent alloy and shows high mechanical strength in addition to its high damping capacity. The acoustic shock pulse was used in order to obtain the fundamental data of the acoustic characteristics for longitudinal wave of this alloy. The amplitude dependence of attenuation and propagation velocity of acoustic shock pulse was measured at room temperature. The attenuation constant increases as the amplitude of shock pulse increases, and shows a maximum value at about 6.5 × 10−5. The propagation velocity of shock pulse slightly decreases with increasing the amplitude of shock pulse. To obtain the clear amplitude dependence of attenuation constant and propagation velocity, it is necessary to measure them in the higher amplitude range.
Binaural asymmetry for centering
Authors: Nabelek Anna K. , Mason David L.
Published: October 2005

An influence of binaural hearing aids on centering phantom images from stereophonic loudspeakers was investigated. When the subject is a symmetrical receiver the location of the phantom image depends on the level difference ΔL between the two signals. A phantom image appears on the midline from both loudspeakers when ΔL is 0 dB and the listener is a symmetrical receiver. The symmetry can be altered by peripheral means such as earplugs, monaural aids, or binaural unbalanced hearing aids, or by asymmetrical hearing loss. The asymmetry of the subject, called “binaural asymmetry,” was measured for 10 normal-hearing and 14 hearing-impaired subjects in unaided and aided conditions. In aided conditions binaural hearing aids with identical gains (balanced) or with 10-dB disparity between the gains (unbalanced) were used. It has been found that the normal range of binaural asymmetry in the unaided condition is ±2 dB and it is slightly greater for the aided balanced condition. Unbalanced aids introduced average ±3-dB additional asymmetry. The variability among subjects was greater for hearing-impaired than for normal-hearing subjects and was greater for aided than for unaided condition. It has been postulated that difficulties in adaptation to hearing aids and errors in localization may result from additional asymmetry introduced by hearing aids. (Work supported by NIH.)
Estimates of internal noise
Authors: Gilkey R. H. , Frank A. S. , Robinson D. E.
Published: October 2005

The concept of internal noise appears frequently in the literature of psychoacoustics. It has been used to explain the differences between the performance of human subjects and various ideal detectors and also to explain the variability in human performance. Internal noise has also appeared as a free parameter in many models of auditory processing. Watson [Ph.D. thesis, Indiana U. (1962)] estimated the ratio of internal to external noise by comparing the decisions of multiple subjects on a trial by trial basis. Swets et al. [J. Acoust. Soc. Am. 31, 514–521 (1959)] and Green [Psychol. Rev. 71, 392–407 (1964)] were able to estimate the ratio of internal to external noise for individual subjects by using reproducible noise as a stimulus. The present paper evaluates the performance of multiple subjects in a single interval detection task using computer generated reproducible noise as a masking stimulus. Multiple-subject and individual-subject methods of estimating internal noise are compared. General problems with estimates of internal noise are discussed. [Research supported by NSF research grant BNS-77-17308.]
Intermodulation products fh + f1 and 2fh + f1: Masking and growth with variable low-frequency primary
Authors: Erdreich John , Erdreich Linda S.
Published: October 2005

Masked threshold for a phase incremented pulsed tone at fh + f1 or at 2fh + f1 was measured in the presence of phase-synchronized maskers fh and f1. The frequency of f1 was either 250, 500, or 1000 Hz and the ratio fh/f1 was fixed at 1.2. The amplitude of the menaural phase effect varied with the level of the primaries, being smaller at the high and low extremes of the range of primary amplitudes. Growth of masking is frequency dependent. Combination tone estimates were calculated based on the assumption that the menaural phase effect is the result of interaction between an aural combination tone and the maskee. Analysis of the growth rates does not permit rejection of a frequency-independent nonlinear mechanism for their generation.
Phase independence in low-frequency pulse detection
Authors: Ahumada A. , Hersh S.
Published: October 2005

The detectability of 10-Hz pulse trains was measured for four pulse waveforms having identical Fourier series amplitude specta. One pulse was a simulated helicopter rotor pulse, the second was a polarity reversal of the rotor pulse, and the other two were random phase waveforms selected for low peak factor. The amplitude spectral envelope of the pulses had a broad maximum at 100 Hz. The detectibility was measured in a two interval forced choice task with 1-s signals in continuous white noise. The results for two experienced and two naive subjects showed no detectibility differences between the pulses either across subjects or within single subjects, even though the sounds are clearly discriminable at suprathreshold levels and the waveform of the random phase pulses is similar to that of the masking noise. If the observed maximum detectibility difference between the pulses had attained 2 dB, it would have been significant at the 0.01 level. [Work supported by a grant to the Stanford-NASA Ames Joint Institute for Aeronautics and Acoustics from the Army Air Mobility Research and Development Laboratory, Ames Directorate.]
Loudness of fluctuating sound
Authors: Kado H. , Hattori S.
Published: October 2005

The stationary sound level was measured as the point of subjective equality (PSE) to randomly fluctuating sound by loudness discrimination test. The fluctuation was specified with two parameters, fluctuating width and velocity, which were obtained from the distribution of envelope of sound. The stimuli presentation was carried out by a kind of stochastic approximation in which the level of stationary sound was varied sequentially by observing the response, thereby PSE was roughly estimated. Furthermore, final PSE was presumed in a maximum likelihood estimation by numerical technique. The result is that the sound is louder as the fluctuating width is larger and velocity is faster. To explain the result, a nonlinear temporal integration model of hearing system is proposed. The distribution of model's output has several frequently occurring levels. The average of those levels is closely related with PSE from the experiment.
From discrete linguistic units to continuous speech characteristics—a functional formulation of articulatory and phonatory dynamics
Authors: Fujisaki Hiroya
Published: October 2005

Formulation of the process of conversion from discrete linguistic units to the continuous acoustic characteristics of speech and vice versa is crucial to both speech synthesis by rule and automatic speech recognition. This paper will present a review of recent works on this problem, with an emphasis on the functional modeling of the processes whereby the segmental and suprasegmental features of speech are realized through the dynamic control of the articulatory and the phonatory systems. The trajectories of acoustic or articulatory parameters of speech are to be interpreted as consequences of the control of the articulatory system by a sequence of discrete phonemic commands, while the trajectory of the voice fundamental frequency is to be regarded as a consequence of the control of the phonatory system by a sequence of discrete prosodic commands. Speech analysis and synthesis based on these models are shown to lead to a better understanding of the dynamic processes of speech production. [Work supported by Ministry of Education Grant-in-Aid for Scientific Research No. 239005.]
Modeling of the tongue using a quadratic curve
Authors: Hashimoto K. , Tanimoto M.
Published: October 2005

An analytic procedure in which a quadratic curve (QC) is used as a reference was applied to the description of the shape of the tongue during speech. By free fitting the QC to each frame of the tongue profile, the loci of the vertices and the centers of curvature were traced. The distributions of the eccentricities and the curvature radii at the vertices are almost Gaussian and their means are 0.85 and 2.5 mm, respectively, while the respective standard deviations are 0.08 and 0.5 mm. Aspects of the deviations from the average QC seen typically around the tongue tip at the articulation of the alveolars are characterized by using the principal components of the deviations. Finally, multiple regression analyses are applied to determine the relations of the QC parameters to the results of the acoustical analyses and also their relations to other models.
Some low-frequency effects in propagation loss models
Authors: Bartberger Charles L.
Published: October 2005

This paper is concerned chiefly with ray and normal mode models. The deterioration in the performance of ray models at low frequencies is illustrated with examples of shadow zones, sound channels, and caustics. As the frequency decreases, the acoustic properties of the bottom become increasingly important, and in constructing wave models for very low-frequency applications it is necessary to take the properties of the bottom realistically into account. A normal mode model is used to show how the maximum depth to which the bottom properties must be known increases with decreasing frequency, and to illustrate the effects of shear waves. The paper concludes with a discussion of the validity of neglecting the coupling of compression and shear waves in homogeneous sediments.
A review of DREA's very low-frequency ambient noise program
Authors: Fraser I. A. , Merklinger H. M. , Stockhausen J. H.
Published: October 2005

For the past 13 years, Defence Research Establishment Atlantic has been actively engaged in measuring properties of the low-frequency ambient noise field in the North Atlantic Ocean and Canadian coastal waters. Concurrently, there has been a strong emphasis on developing surface-suspended arrays that can reliably and inexpensively extend such measurements down to the infrasonic range. Special attention has been paid to hydrophone design and shrouding, and to minimizing the effects of sea-surface motion and current shear. The success of this program has led to the construction of small superdirective research arrays, and more recently, quiet large-aperture vertical and horizontal line arrays. Near-surface noise directionality measurements made in midocean with these arrays complement reported results from bottom-moored research arrays. The measurements of vertical directionality, in particular, show distinct trends, with arriving low-frequency noise becoming more concentrated towards the horizontal in the northern ocean, and more diffuse with decreasing frequency. Azimuthal anisotropies are less pronounced, with some bias towards busy shipping lanes and fishing grounds.
Very low-frequency reverberation in the Arctic Ocean
Authors: Zittel J. , Shepard G. W. , Dyer I. , Baggeroer A. B.
Published: October 2005

During March and April of 1978 a camp was established on the Arctic ice over the Canada Abyssal Plain, including a horizontal receiving array, a multichannel data handling and storage system, and a digitally operated data analysis system. With the use of TNT charges of up to 880 pounds, acoustic reverberation in the Arctic Ocean was measured. Emphasis in data analysis was placed on the frequency range centered on 10 Hz. Reverberant (scattering) returns were obtained for elapsed times up to and greater than about 30 min (ranges greater than about 1300 km). The receiving array was about six wavelengths long, and helped to identify various topographic features in the Arctic Ocean. Measurement results and theoretical models will be presented and discussed. [These efforts were undertaken jointly with The Woods Hole Oceanographic Institution, the Lamont-Doherty Geological Observatory, and the Bedford Institute of Oceanography. Work supported by the Arctic Program Office of ONR.]
Regional dependence of very low-frequency sound attenuation in the deep sound channel: Correlation with internal wave measurements
Authors: Fecher Michael J. , Browning David G. , Mellen Robert H.
Published: October 2005

The most promising mechanism for the very-low-frequency sound attenuation observed in the deep sound channel is diffusive scattering by internal waves. Mellen et al. [J. Acoust. Soc. Am. 60, 1053–1035 (1976)] have obtained estimates for the extra attenuation using the Garrett-Munk internal wave model and find consistency with the lower experimental values reported. Kibblewhite et al. [J. Acoust. Soc. Am. 63, 1169–77 (1978)] have shown a definite regional dependence in the Pacific. In this paper we compare regional oceanographic measurements with the Garrett-Munk internal wave model and also correlate local acoustic measurements with estimates of the extra attenuation.
Identification of frication by normally hearing and hearing-impaired children
Authors: Simon Helen J. , Whaley Patricia
Published: October 2005

The purpose of the present investigation was to examine the identification functions of a /so/-/to/ continuum by normally hearing children and children with conductive and sensorineural hearing losses. Two six-step continua were generated on the OVE III speech synthesizer at Haskins Laboratories. The continua differed only in the duration of friction. The friction duration of the first continuum ranged from 175 to 25 ms in 25-ms steps while the friction duration of the second continuum ranged from 500 to 50 ms in 50-ms steps. Normally hearing children listened to the short duration tape while the hearing impaired children listened to both continua. Results will be discussed relative to speech perception and the implication for articulation development.
A test to determine the auditory speech perception ability of hearing-impaired children
Authors: Jones K. O. , Whitehead R. L.
Published: October 2005

Differences of auditory speech perception ability between 12 normal-hearing children (average age 9 years, 3 months), and 12 children (average age 9 years, 7 months) with severe hearing loss (better-ear pure-tone average between 85 and 92 dB ANSI) were determined by a test of auditory speech perception. The test consisted of 64 pairs of recorded consonant-vowel speech signals. The recorded speech signals were carefully analyzed to insure control over intensity, frequency, and duration differences. Then the test-speech signals purposely were varied in terms of intensity, frequency and duration. Subjects were required to determine if the two speech signals in each pair were the same or different based upon one of the three variables. Results indicated that the children with normal hearing performed significantly higher than the hearing-impaired children. Implications are discussed.
Intelligibility for the zero-crossing rate compression of speech
Authors: Miyaguchi H. , Ozawa S. , Ono H. , Saito S.
Published: October 2005

A patient with hearing loss mainly in the high-frequency range can not clearly comprehend speech. Therefore, if the frequency band is compressed to a lower range, it is possible that the speech perception of such a patient can be improved. To test this possibility, the instrument described below was constructed and experimentally tested. First of all, a speech sample was divided into seven frequency bands, and the zero-crossing waves and envelopes were determined. The zero-crossing waves were converted to lower frequencies by flip-flop circuits, and they were amplitude-modulated by their respective envelopes. Frequency-compressed speech was obtained by the addition of the seven signals thus processed. Successful training of a person with normal hearing ability was conducted using this frequency-compressed speech. It was concluded that a high intelligibility can be thus obtained. We expect to employ this instrument in the treatment of patients with hearing impairments.
A tactually cued vocoder using a microprocessor
Authors: Ifukube T. , Tanaka K. , Nitadori Y. , Yoshimoto C.
Published: October 2005

In order to improve the recognition of consonants by our tactual vocoder [T. Ifukube, 9th ICA, I-72 (1977)], a tactually cued vocoder was designed for the deaf. In this apparatus, the acoustic cues of Japanese monosyllables were extracted automatically in real time using a microprocessor. These cues were displayed on a small vibrator matrix with 16 × 6 piezoelectric devices so as to be perceivable by the vibrotactile sense of a fingertip. Experimental results showed that unvoiced consonants, except plosives, could be detected at about 90% correct rate. Sixty-eight Japanese monosyllables were separated into three groups (fricatives, plosives, and others) and were detected at a rate of about 80% correct. The possibilities for the practical use of our cued vocoder were discussed based on the results of tactile speech recognition by the deaf.
Polymer electret transducers: Recent Developments
Authors: West J. E.
Published: October 2005

The first thin-film polymer microphone, reported in 1962 [G. M. Sessler and J. E. West, J. Acoust. Soc. Am. 34, 1787 (1962); 40, 1433 (1966)], stimulated a variety of new transducers ranging in frequency from 10−3 to considerably more than 108 Hz. The success of these highly-sensitive modern electret transducers stems from the favorable properties of certain polymers which permit stable long-term charge storage under most environmental conditions. More recently, the piezoelectric properties of polymers have been applied to acoustic transducers. A review of electret transducers with special emphasis on new developments will be given. Included are, for example, microphones for precision sound measuring systems, transducers for telephony, underwater sound, and headsets.
Piezoelectric polymer transducer and its application to acoustics
Authors: Chubachi N.
Published: October 2005

Piezoelectric polymer has several unique properties compared with conventional piezoelectric materials, namely (1) low acoustic impedance, (2) high flexibility, (3) a relatively low dielectric constant, (4) high breakdown voltage, (5) resistance mechanical and thermal shock. A thin sheet of wide area can be easily produced, and it can be properly given local piezoactivity. At the present stage, polarized stretched films of polyvinylidene fluoride (PVDF) are considered to be most promising in practical use. Various transducers with PVDF films have been developed in the field of acoustics from audio engineering to ultrasonics, such as microphones, headphones, loudspeakers, electromechanical transformers, ultrasonic transducers, and so on. Principles and constructions of these transducers especially for ultrasonics are described including several new constructions which have been recently proposed. Applications of these transducers to electronic devices as well as ultrasonic imaging systems are also described.
Improvement of piezoelectric polymer tweeters
Authors: Yamamoto T. , Tamura M. , Maekawa K.
Published: October 2005

Piezoelectric polymer is one of the most promising materials to expand the designing technique of electroacoustic transducers for audio frequency. At first, pulsating cylinder-type tweeters with piezoelectric polymer films have been commercialized; omnidirectional patterns have been realized up to 20 kHz in the horizontal plane. This report presents a further improvement of the sensitivity of the cylinder-type tweeter by the use of disk-shaped horns.
A model for the environmental noise contribution of refuse vehicles
Authors: Mintz Fred , Goldstein Jeffrey
Published: October 2005

As part of a study to determine the need for a Federal noise regulation for refuse vehicles, a computer model was developed which incorporated the major operational, situational, and environmental parameters associated with the noise contribution of refuse vehicles in urban and suburban areas. The primary components of the model consider the trash collection frequency, the noise emission time-history characteristics of different refuse vehicles, and the sound propagation features of the affected neighborhoods. These factors are defined for five basic dwelling-type categories treated: suburban single-family detached, suburban duplex, urban row apartments, dense urban apartment, and very dense urban apartments. The features of the various neighborhood types are outlined and the resulting environmental noise contributions are discussed briefly.
Resonant tube for measurement of sound absorption in gases at low frequency/pressure ratios
Authors: Zuckerwar Allan J. , Griffin William A.
Published: October 2005

A resonant tube has been constructed to measure sound absorption in gases at room temperature over a range of frequency/pressure ratios from 0.1 to 1000 Hz/atm. Essential specifications of the tube are the following: material, type 304 s/s; length = 18.29 m, corresponding to a fundamental frequency of 9.38 Hz in air at 20 °C; bore diameter = 0.1524 m; wall thickness = 0.0254 m; maximum operating pressure = 100 atm. The exciting element is located outside the tube in order to minimize background losses, and excites the gas into axial resonance through one of the end plates. A high-sensitivity quartz microphone is flushmounted in the other end plate. Sound absorption is measured by the method of free decay. A calibration study reveals that the acoustic losses in argon are with few isolated exceptions in excellent agreement with the wall losses predicted according to the Kirchhoff formula. Results of preliminary measurements of the vibrational relaxation peak of nitrogen, and its shift to a higher f/P ratio in the presence of a small addition of carbon dioxide, are reported. [Work supported by NASA Grant NSG-1324.]
Scanning acoustic microscope with concave transducers
Authors: Chubachi N. , Sannomiya T. , Iyama T.
Published: October 2005

Recently, a scanning acoustic microscope has been developed by Quate and Lemons which utilizes acoustic lenses for focusing ultrasonic waves. In this paper, an alternative scanning acoustic microscope is described which utilizes concave film transducers for focusing ultrasonic waves. A pair of concave transducers are arranged confocally to compose the transmission-type scanning acoustic microscope. The transducer is a type of composite-resonating mode transducer constructed by depositing a ZnO piezoelectric film on the curved surface of the gold hemispherical shell. The transducers employed here operate in a VHF range and their focal lengths are about 1.8 mm. Specimens are placed at the focal plane in a water cell and scanned mechanically through the focused beam in a raster pattern. Transmitted acoustic power detected with the output concave transducer is translated into an intensity modulation of the synchronized luster on a CRT display. A spatial resolution of better than 10 μm is attained at 115 MHz. Several images of biological specimens observed with this microscope will be demonstrated.
Stress-acoustic relations for stress analysis of slightly orthotropic materials
Authors: Okada K. , Torikai Y.
Published: October 2005

For the stress analysis in slightly orthotropic materials with ultrasonic technique, useful stress-acoustic relations are theoretically introduced and their validity is verified experimentally. In the development, two assumptions are made: (1) Change in ultrasonic velocities is given by a linear combination of stress components; (2) the stress-acoustic relations have the same symmetry as that in the elastic stress-strain relations. Theoretical results show that, in contrast to an isotropic material, three stress-acoustic coefficients are required to characterize Δv/v = φ in stressed state, where Δv/v is a fractional difference in shear wave velocities and φ is an angle between a principal direction of particle motion of shear waves and a symmetry axis of unstressed material. In the experiment, rolled aluminum alloy of 5 mm thickness was used, in which Δv/v = 3.45 × 10−3 for unstressed state. The three stress-acoustic coefficients were determined in an uniaxial tensile test preliminarily made on three specimens cut out at different angles with respect to the rolling direction. The values of Δv/v and φ were measured in other specimens with several orientations as a function of applied stress. The results show good agreement with the curves theoretically predicted with the three coefficients.
Impulse wave diffraction by rigid wedges and plates
Authors: Bremhorst J. , Medwin H.
Published: October 2005

The impulse wave solution of point source radiation which is diffracted by an infinite rigid corner or wedge, [M. A. Biot and I. Tolstoy, J. Acoust. Soc. Am. 29, 381–391 (1957)], describes a closed form time-domain solution to an old problem that is generally studied in the frequency domain. The theory is tested by an experiment using models of wedges and plates in an anechoic chamber, and the source regions of diffraction are identified and compared with theoretical predictions. In addition, numerical Fourier transformation of the dominant part of the solution provides the transmission loss as a function of frequency without further assumptions or approximations. [Work supported by the Office of Naval Research.]
Prevention and recognition of athletic injuries
Authors: Allman Fred L.
Published: October 2005

Participation in athletics is a privilege involving both responsibilities and rights. The athlete's responsibilities are to play fair, keep in training, and to conduct himself with credit to his sport and his school. In turn, he has a right to optimal protection against injury and this may be assured through good conditioning and technical instruction, proper regulations and conditions of play, and adequate health supervision. With the number of people involved in all types of recreation in sports, and the high injury rate even in noncontact sports, there is an ever increasing. need to make these activities as safe as possible. There are a number of factors that must be considered if sports and recreational activities are to be kept safe and if proper treatment is to be administered. What is sports medicine? Sports medicine is the study of problems and the application of solution to the problems as they relate to the physiological, psychological, and pathological nature of the athlete. It involves sports, but is not limited to organized athletics; it involves medicine, but is not limited to injuries. Sports medicine concerns itself with the health implications of the organism and physical activity; thus, sports medicine spans the gamut from the highly trained athlete who is endowed with sufficient skill and a physique to participate in modern competitive athletics with great intensity and diversity, to the neophyte “little leaguer,” who often lacks skill and physique and who might play in the vacant lot without supervision.
Acoustic multipath measurements in the ocean
Authors: Wallman L. H.
Published: October 2005

The acoustic multipath structure is of considerable interest to those concerned with the performance of oceanic acoustic systems. This is because the performance of these systems is directly affected by the nature of the multipath structure. The multipath structure was measured using a cyclic linearly swept frequency modulated (LFM) signal. The path structure in each convergence zone was determined accurately and continuously to ranges exceeding the third convergence zone at 91 nautical miles. The path structure at each convergence zone normally consists of four caustics with their associated shadow zones. Each caustic also has associated with it two distinct propagation paths. Using the LFM signal all of these features are measured and plotted as relative travel time versus range. These plots show dramatically the relationship between the different paths in addition to providing a qualitative measure of their relative strength and therefore their importance to the overall transmission loss. Utilizing these plots and the transmission loss the acoustic propagation modeler can determine readily how accurate his model is.
Some results of an optical simulation of acoustic propagation in the ocean
Authors: Fain G. , Estes L. E.
Published: October 2005

We are generating analog models of sound propagation in a variable sound-speed ocean by using a previously described system [L. E. Estes and G. Fain, J. Acoust. Soc. Am. 62, S17(A) (1977)] that propagates a CW laser beam through a differentially heated fluid. The differential heating is effected at the correct wavelength scale by imaging a negative with the appropriate gray scale onto the fluid. The energy source is a high power pulse laser. The negative typically requires 500 000 resolution elements. It is exposed using a computer directed electromechanical deflection and acousto-optical modulation system to control a low power CW laser. Essentially any pattern can be placed on the film. During operation a photographic measure of sound intensity at all ranges and depths is made. A simultaneous measurement of the velocity distribution is made using a Mach-Zehnder interferometer. Results presented include range independent profiles that are compared to analytic solutions and range dependent profiles which are compared to a full diffraction theory numerical model. [Work supported by the Office of Naval Research.]
Sea mount reconnaissance sonar study
Authors: Heimann Jack H.
Published: October 2005

The merits of utilizing current direct path, preformed beam sonars for the detection and localization of submerged sea mounts have been assessed. Both hull mounted and variable depth (VDS) configurations of the DE1160 series of shipboard sonars were examined. The application being considered is one of sea mount mapping to reduce the hazards of submarine operations in areas of known or suspected sea mounts. The sonars have source levels, frequency, waveform, and processing suitable for the application and can provide sweep widths up to 36 km. The VDS configurations provide better ray path coverage, self noise, and roll stability. The predicted detection and localization capability is limited by ray path refraction. Ray path computation and display is required to optimize selected array depth and elevation angle. The ray path capability is also employed for mapping echo arrival time and angle into sea mount range and depth. Intrabeam arrival angle measurement is required because of ray path spreading. An added requirement is a display format capable of presenting cross range versus sea mount depth. [Funded by NORDA.]