bluuebunny/crossref_metadata_2025 huggingface dataset:
Auditory nerve fiber activity and the growth of temporary threshold shift (TTS) in the chinchilla
Authors: Markuszka J.
Published: October 2005

Chinchillas were exposed to a 95 dB SPL octave band of noise centered at 0.5 kHz for 8, 12, 24, 48 h, or 9 days, and the responses of primary neurons were studied. Beginning 3–4 h after the exposure, threshold at CF, spontaneous rates, and frequency tuning curves (FTC) of units were studied over a 12–16-h period. The pattern of thresholds revealed a TTS of 60–70 dB that first occurred in the low frequencies. After an 8-h exposure, units below 1 kHz reached asymptotic levels (ATS), and the level of TTS declined as a function of frequency; above 6 kHz units were normal. The major effect of exposure duration was a spread of TTS to higher frequencies. After 48 h, all units below 13 kHz reached ATS levels 60–75 dB above normal. There was no further increase in TTS after 9-day exposures. FTC's underwent systematic changes in shape such that after 2 days the results were similar to those of Salvi, et al. [J. Acoust. Soc. Am. 62, S35 (1977)]. FTC tails of high CF units were normal and 20–30 dB below the ATS levels of low-frequency units. This may account for the disparity in single unit and behavioral estimates of TTS values for exposures lasting less than 48 h [H. M. Carder and J. M. Miller, Trans. Am. Acad. Ophthalmol. Otolaryngol. 75, 1346–1354 (1971); W. W. Clark and B. A. Bohne, ARO paper (1977)]. [Supported by NS 03856.]
Recommendations for advancing speech recognition
Authors: Lea Wayne A. , Shoup June E.
Published: October 2005

Surveying the opinions of over 100 workers in speech recognition, we found widespread agreement about the gaps in current technology and the needs for future research and development. Improved phonetic segmentation and labeling is considered a top priority need in future studies. Also frequently mentioned are phonological rules (covering coarticulation effects, phone substitutions, deletions, and insertions, and word boundary effects); and prosodic aids to providing more efficient phonemic analysis and syntactic parsing. Work should continue on strongly constraining hypothesized word sequences with syntactic and task-dictated constraints, while permitting naturalness of expression. The practical needs for continuous speech are not well documented, and the wide gap between available isolated word recognizers and experimental limited speech understanding systems needs to be bridged by work on several systems of varied complexities and capabilities. Performance evaluation should be given special attention. We recommend the establishment of speech science centers with speech and linguistics expertise, powerful computer facilities, and a mechanism for visiting researchers to use such facilities to advance their work and refine the resident recognition systems. [Work supported under Office of Naval Research Contract Number N00014-77-C-0570.]
Performance of Harpy speech recognition system for telephone quality speech input
Authors: Yegnanarayana B. , Reddy D. Raj
Published: October 2005

Spectral distortions in speech affecting the performance of speech processing systems can be classified into multiplicative and additive types. We report in this paper the results of our study on the performance of Harpy continuous speech recognition system for multiplicative type of distortion with special reference to telephone quality speech. It has been observed that the recognition performance of Harpy system for digit task dropped from 99% to 93% for telephone speech [G. Goodman, B. Lowerre, D. R. Reddy, and D. Scelza, J. Acoust. Soc. Am. Vol. 60, S11(A)]. The effect of distortion on segmentation and labeling will be discussed for simulated and actual telephone data. Performance evaluation is obtained for A1 information retrieval task (1011 words) in terms of word and sentence accuracies for different types of distortion. The results will be discussed with reference to signal processing functions in the system and some normalization techniques are suggested to account for the distortion. Comparison of signal characteristics under telephone and quatization noise distortions is made to explain the recognition performance of the system for these distortions in the input speech.
Comparison of speech spectra for additive type spectral distortion
Authors: Yegnanarayana B. , Reddy D. Raj
Published: October 2005

Parameters representing smoothed spectral characteristics of short segments of speech are often used as features in speech processing systems. The main pattern recognition problem in speech is matching the test spectrum with the reference spectrum. In this paper we show that the matching methods usually adopted do not yield a true measure of the actual differences in the envelopes of spectra. This is particularly true for additive type of noise degradation in speech. Two types of such degradation namely, the quantization noise of waveform encoding and additive bandlimited white noise, are considered for illustration. We show that the parameters, linear prediction coefficients or cepstral coefficients, do not represent the true spectral envelope information of the distorted signal, which explains the discrepancy among various distance measures based on these parameters. We propose a more practical approach which involves transforming one spectrum relative to the other to bring both of them to the same level of dynamic range before any comparison is made between them. The main result of this study is that quantization distortion of ADPCM speech is not very significant even at low bit rates, whereas additive white noise is deleterious even for high signal to noise ratio. This result explains to some extent the good recognition capability of Harpy speech recognition system for ADPCM speech even for the lowest bit rate [B. Yegnanarayana and D. Raj Reddy, J. Acoust. Soc. Am. 62, S27 (A)].
On finding the optimum excitation for LPC speech synthesis
Authors: Atal Bishnu S. , David Nancy
Published: October 2005

In synthesizing speech from LPC parameters, the excitation for voiced speech is generally approximated by a sequence of pulses spaced one pitch period apart. This idealization assumes that the different harmonics in the excitation have equal amplitudes and zero phases. The object of this study was to determine the best modification—particularly with respect to the phrase—of this simple pulse excitation for optimizing the quality of resultant synthetic speech. We systematically modified the amplitudes and phases of the harmonics in the excitation signal obtained by inverse filtering of the speech waveform. Our results show that replacing the actual harmonic amplitudes with constants—even when the phases are correct—produces audible distortion in the synthetic speech. Furthermore, exact reproduction of the amplitude spectrum alone is not sufficient for producing natural-sounding speech—the relative phases of different harmonics must also be considered. Preliminary results suggest that actual phases can be replaced by frequency-dependent but time-invariant phases without introducing significant distortion.
Development of a real-time voice pitch extractor
Authors: Scott Brian L.
Published: October 2005

A real-time pitch extractor has been implemented in analog hardware. The algorithm is temporally based and involves the detection of the envelopes of eight bandpassed channels. The envelopes of the eight channels contain a common periodicity, the inverse of which is the fundamental frequency. Adding the envelopes yields a signal devoid of vocal tract resonances. This signal is passed through a one-shot yielding a single 15 V by 1-ms pulse every pitch period. Integrating the pulses provides a dc voltage bearing a linear relation to the fundamental frequency. The device has been tested with sets of synthesized vowels in male, female, and child voices. These results show the apparatus to be unaffected by speaker or vowel. A more stringent test with naturally recorded sentences from a male, female and child voice indicate problems with gain control and other limitations which will be discussed. Results indicate the feasibility of a low-cost real-time pitch extractor. [Supported by NS03856 to Central Institute for the Deaf.]
A physical method for measuring the quality of speech communication channels
Authors: Steeneken H. J. M. , Houtgast T.
Published: October 2005

A physical method has been developed for measuring the quality of speech transmission systems. The method quantifies the deterioration of the intensity fluctuations of running speech. The decrease of these fluctuations as a function of the fluctuation rate (Modulation Transfer Function, MTF) is expressed by a single index, the Speech Transmission Index (STI). A hardware device using an artificial test signal determines the STI on the basis of the MTF's measured for each of the octavebands from 125 Hz up to 8 kHz. The method has been applied successfully to distortions as bandpass limiting, interfering noise, peak clipping, digital modulation methods, automatic gain control and reverberation. For a set of 200 of such communication channels a high correlation was found between the physical index STI and the PB-word score as determined with talkers and listeners.
Predicting the effect of room acoustical treatment on a large industrial facility
Authors: Hanson William J.
Published: October 2005

Acoustical absorptive materials have been proposed by some to alleviate the problems of excessive noise in large industrial facilities. Under certain limited conditions, this is perhaps the optimum solution with respect to long-term cost, maintenance, convenience, and performance. However, many industrial facilities do not conform to the basic requirements necessary for successful treatment in the form of architectural modifications. Noise propagation was investigated in an existing untreated facility to determine the room's dissipating and absorptive characteristics. Reverberation or decay rate and loss due to distance measurements were made to establish the absorption data. Absorptive surfaces within the existing installation were located. and a test was initiated to evaluate the typical range of absorption of these components. Upon completion of this phase, a portable reverberation chamber was designed for in situ measurement of absorptive equipment and materials. The previously obtained information was employed in the design of a scale model to mimic the acoustical behavior of the full-scale environment. Once construction was completed, a variety of architectural and equipment layout modifications were made to determine their effect on noise propagation. This information made it possible to predict the maximum noise reduction of each modification, and at various locations.
Vibrotactile encoding of speech signals
Authors: Dolan Terrence R. , Finkenzeller P. , Keidel Wolf D.
Published: October 2005

Preliminary results of a research effort concerned with the sensory encoding of speech by humans through the vibrotactile system are described. The research involves the use of a hardware system which includes a frequency-transformation device [see Finkenzeller, P., Habilitation Thesis, Universität Erlangen, Erlangen, Germany (1973)] and a vibrotactile transducer which is used to present the transformed speech signal to the hand or arm. The transformation device operates on the speech signal essentially in real time and is capable of lowering the spectral content of the waveform while preserving part of the information available in the fine structure. In this paper, the transformation device and results of early speech-discrimination experiments are described. These initial experiments have primarily involved the psychophysical measurement (using an A-B-X paradigm) of intelligibility of words selected from the W-22 and Modified Rhyme Test lists. The results suggest that the utilization of appropriate spectral and intensive transformations of the speech signal lead to increased speech intelligibility when speech is presented via the vibrotactile system.
Perception of frequency-lowered vowels by naive listeners
Authors: Picheny M. A. , Hicks B. L. , Braida L. D. , Durlach N. I.
Published: October 2005

The effect of frequency-lowering on the perception of vowels (in /b/-V-/t/ contexts) by naive listeners was studied using an identification paradigm. The stimuli consisted of 16 vowels and diphthongs processed by a variety of techniques (including time dilation, frequency warping, and linear prediction) to achieve pitch-variant and pitch-invariant lowering combined with both uniform (proportional) and nonuniform (warped) compression of the frequency axis. Lowpass filtering conditions were also included for comparison purposes. Perceptual confusions were analyzed in terms of articulatory features. In general warped frequency lowering (which primarily compressed the higher frequencies) resulted in a differential degradation of feature perception, while proportional lowering degraded the features more uniformly. Lowering fundamental frequency by a factor of two had more severe effects on the perception of vowels spoken by males than by females. Also, lowering both fundamental and formant frequencies of female vowels failed to make them as resistant to lowpass filtering as male vowels. [Work supported by NIH.]
The effect of frequency lowering on consonant discrimination
Authors: Reed C. M. , Hicks B. L. , Braida L. D. , Durlach N. I.
Published: October 2005

We have investigated the basic discriminability of consonants that were processed using pitch-invariant frequency lowering with either uniform (linear) or nonuniform (warped) compression of the frequency axis. The bandwidth of the unprocessed signal was compressed by factors of two and four to achieve bandwidths of 2500 and 1250 Hz, respectively. For each of the lowering factors, five warping characteristics were studied in addition to lowpass filtering to the equivalent bandwidth. Stimuli were CV syllables made up of 24 consonants and three vowels, recorded by two male and two female speakers. Discrimination tests were performed using an AB-BA procedure. As expected, performance for the 2500-Hz bandwidth was superior to that for the 1250-Hz bandwidth. Overall performance with the best lowering-warping combination was roughly comparable to that with lowpass filtering for the same bandwidth; however, the patterns of performance over articulatory features were significantly different. In general, lowering-warping performance was superior to filtering performance for contrasts of fricative sounds but inferior for contrasts of nasals and semi-vowels.[Work supported by NIH.]
Intelligibility of one- and two-syllable words in speech-modulated noise
Authors: Streeter Lynn A. , Landauer Thomas K. , Alvarez Aldofo
Published: October 2005

A classical finding in the auditory word recognition literature is that two-syllable words presented in white noise are identified more accurately than one-syllable words [see Licklider and Miller in Handbook of Experimental Psychology, edited by S. S. Stevens (1951)]. An explanation commonly given for this phenomenon is that two-syllable words have greater redundancy. However, the effect may have been due to inadequate methods for matching the S/N ratio of the two word sets. We randomly selected one- and two-syllable words of low, medium, and high frequency of occurrence in English. We used speech-modulated noise in which the amplitude of the noise was correlated with the amplitude of the speech signal. Words were presented at a 0 dB S/N ratio to 24 listeners for identification. There was no significant difference in proportion correct between one- and two-syllable words (0.42 and 0.40, respectively). However, there was a reliable frequency effect; more frequent words were more often identified correctly.
Application of viscoelastic materials for structural damping and isolation
Authors: Halvorsen W. G. , Nashif A.
Published: October 2005

Structural damping and isolation are two of the most useful approaches for modifying structural dynamics for noise and vibration control. Most such damping and isolation systems employ viscoelastic materials. This presentation will review measurements of the modulus and loss factor characteristics of viscoelastic materials and will discuss the variation in those properties with temperature and frequency. The presentation will then discuss the prediction of the performance of structural damping and isolation systems based on the viscoelastic material properties. Examples will be presented illustrating various computations and plotting techniques that have been found useful for prediction of system performance.
Transverse vibrations of circular plates having nonuniform edge constraints
Authors: Leissa A. W. , Laura P. A. A. , Gutierrez R. H.
Published: October 2005

The free vibrations of circular plates having flexible edge supports have been studied by several researchers for the restricted case when the supports are represented by springs having constant stiffness. In the present paper a general method is presented for dealing with supports having translational and rotational flexibilities which vary in an arbitrary manner around the boundary. It is shown that the varying stiffnesses can be represented as accurately as desired by expanding them into trigonometric series in the polar angle. The exact solution in polar coordinates of the differential equation of motion for the plate is then substituted into the elastic boundary conditions. The resulting infinite characteristic determinant is solved by successive truncation. As an example the case of a plate having a simply supported edge (infinite translational stiffness) with rotational stiffness varying according to L0 + L1 cos θ (L0 and L1 being constants) is considered. Numerical results are obtained by the method described above and also by using the Ritz method with functions which approximate both the differential equation and the boundary conditions.
Natural frequencies of membranes having mixed boundary conditions
Authors: VanSickel J. R. , Torvik P. J.
Published: October 2005

A number of approximate methods have been given for determining the natural frequencies of membranes of arbitrary shape and uniform boundary conditions. In this paper, a method is presented whereby natural frequencies can be determined for membranes of arbitrary shapes with a portion of the boundary clamped (no displacement) and a portion free (vanishing slope). The method is developed through expanding the displacements in a series of functions, each of which satisfies the governing differential equation, but not the mixed boundary condition. A variational principle is used to find those expansion coefficients which give the best approximation to the desired boundary condition. The required variational principle is developed from Hamilton's principle through the same relaxation of the class of admissible functions used to develop Reissner's principle from the Theorem of Minimum Potential Energy. Results for a rectangular membrane, clamped on three edges, with the fourth edge partially clamped and partially free, are given, and are compared with results obtained for the same geometry through a collocation procedure and through a finite element analysis.
Investigating structural response with a dual-channel spectrum analyzer
Authors: Sramek John
Published: October 2005

Structural design problems that surface in operating equipment historically have been difficult to identify and costly to correct. Recent advances in dual-channel signal processing equipment, however, can greatly simplify analysis procedures. Important structural response characteristics such as impulse response and system transfer functions can now be easily identified experimentally. In addition, the current generation of dual-channel analyzers promises to be an invaluable aid to the designer for detecting problems prior to production.
The discrepancy between the measured sound field of an oscillating solid and that predicted from the Kirchhoff formulation
Authors: Pierce A. D.
Published: October 2005

Considerable discussion was evoked at the Miami meeting by T. F. Brooks' report [J. Acoust. Soc. Am. 62, S33(A)(1977) and NASA TP-1048 (Dec. 1977)] that, for carefully performed experiments on sound radiation from a rotationally oscillating (peak angular displacement α) airfoil shaped solid, the Kirchhoff integral over surface pressures and accelerations consistently overpredict SPL's at moderate distances by 2–5 dB. The present paper gives quantitative development of explanation proposed at that time by J. E. Brooks and independently by the author. SPL data at 2.2 m radius from the airfoil (chord length 2b = 0.46 m) at f = 301 Hz (α = 82.1 and 259.3 μrad) and at f = 447 Hz (α = 30.8 and 97.6 μrad), respectively, suggest pattern can be regarded as a dipole-quadrupole field superposition. Surface pressure amplitudes divided by f2α are nearly independent of f and α, so, even though kb is not extremely small, multipole concepts and matched asymptotic expansion formulation are applicable. Dipole moment vector [proportional to ∫pndS] would be identically zero were airfoil an elliptic cylinder, but is merely “small” for the actual experimental airfoil. Although accuracy of p measured at any one surface point is probably within 10%, p changes sign from leading to trailing edge and |∫pndS1| over top or bottom surface is less than 15% of corresponding ∫|p|dS1. The expected value of the experimentally derived [∫pndS]2 exceeds its error-free value by N times the mean-squared error in individually measured p's (N = 9). Consequently, the expected value of the square of the magnitude of the dipole vector moment calculated from surface measurements of p will be about 2–3 times its error-free value, which is consistent with the 2–5 dB discrepancy reported by T. F. Brooks.
A further study into the Gauss Laguerre formulation of the nearfield of a circular plane piston transducer
Authors: Cavanagh Eduardo , Cook Bill D.
Published: October 2005

The advantages of interpretation using Gauss Laguerre/Hermite formulation was adopted to acoustics from optics by Cook and Arnoult [J. Acoust. Soc. Am. 59, 9–11 (1976)]. Recently Cavanagh and Cook [J. Acoust. Soc. Am. 62, S22(A) (1977)] have shown this formulation has possibilities in acousto-optic projection analysis of ultrasound waves. The question remains of the number of modes required to adequately describe the field from plane piston transducers. This paper describes the ongoing research into that problem for a circular transducer.
Acoustic scattering cross section predicted from an accurate model of the spectrum of temperature fluctuations
Authors: Hill R. J.
Published: October 2005

Recent theoretical and experimental advances have produced an accurate model of the power spectrum of temperature fluctuations in turbulent air. This model temperature spectrum and a semiempirical energy spectrum are used to predict the shape of the acoustic scattering cross section as a function of wave number and scattering angle. Experiments have shown that the temperature spectrum in air has a “bump” at high wave numbers, and the model temperature spectrum shows that this feature is expected theoretically. A corresponding bump is predicted to appear in the acoustic scattering cross section. The bump in the cross section is reduced if velocity fluctuations dominate the scattering, because the energy spectrum has no bump. For typical energy dissipation rates in the atmospheric boundary layer, a 2-kHz acoustic sounder senses scattering attributable to the bump in the cross section. The amount of scattering attributable to the bump depends on the acoustic wave number and the microscale and can be as much as 50% greater than the scattering predicted by the usual theoretical interpretation.
Sonic “Banshee” tonal bursts from underwater flames
Authors: Fitzgerald J. W. , Carey W. M. , Doolittle R. D.
Published: October 2005

Underwater oxyacetylene flames radiate sound energy throughout a wide frequency spectrum. For a flowrate of ∼70 ml/s through a ∼0.04 in. ID nozzle, flame-on, long-term average spectra show peak levels of ∼170 dB re 1 μPa/Hz at 1 in. at ∼250 Hz. Corresponding values for the flame-out condition (i.e., an oxyacetylene gas jet, not burning) are about 10 dB lower at the same peak frequency. Thereafter the spectrum levels decrease to ∼95 dB and ∼85 dB at ∼10 kHz, respectively, for the flame-on and flame-out conditions. Above 10 kHz the spectrum levels of flame-out flow continue to decrease, probably at about −6 dB/oct. Typical flame-on spectrum levels, however, are still ∼90 dB at 80 kHz, with broad peaks: at ∼26 kHz of ∼105 dB; at ∼54 kHz of ∼98 dB; and at ∼75 kHz of ∼93 dB. Tape recordings made at 60 ips and played back at 334 ips show that the flame radiates sound in intermittent tonal bursts of 1–5-ms duration, sounding like the wails of “Banshees”. The average peak-to-peak source levels are typically ∼170 dB re 1 μPa at 1 in, corresponding to a thermoacoustic efficiency of ∼10−6 for the stoichiometric oxygen-acetylene reaction. The tonal bursts appear to be chemical-reaction driven monopoles and imply that under-liquid flame combustion processes are intermittent. rather than continuous. These “Banshees” could serve as a basis for an acoustic leak-detection system for liquid-sodium steam generators.
Effect of nonlinear propagation distortion on random signals
Authors: Webster Don A. , Blackstock David T.
Published: October 2005

Two aspects of nonlinear propagation distortion of random signals are investigated. First, we ask whether the spectral distortion of a noise can be determined by the initial power spectrum and amplitude probability density of the noise. A nonlinear acoustical model is used to make predictions of noise distortion which are compared with experimental measurements. It is shown that in general the initial spectrum and amplitude density are not sufficient information on which to base predictions of propagation distortion. Second, a theoretical analysis of the effect of propagation distortion on amplitude density of a sample noise signal is given. It is found that under certain general conditions the amplitude density is unaffected by the distortion. [Work supported by ONR, AFOSR, NASA, and NOAA.]
Stochastic direct and inverse radiation problem on two ring random sources
Authors: Maestrello L.
Published: October 2005

A model consisting of two ring sources is developed for the study of the farfield behavior: correlation, coherence, and phase on a round jet. The sources on the rings are random and monopole with Gaussian profiles. They couple within the rings as well as between rings such that only adjacant sources correlate. Numerical results show that the sources are compact, in a broadband sense, when the farfield correlation peaks at zero time delay; otherwise, they are noncompact with respect to the acoustic wavelength. For noncompact cases the phases have a sawtooth type of distribution joined with linear ones with increasing frequency. This is the resultant interference effect. The behavior is well documented by the systematic variation of the distance between rings, diameter and the number of sources. The inverse radiation approach is in general ill-posed in Hadmark sense; however, the asymptotic behavior of the phase function, as the frequency tends to zero and infinity, permits one to obtain explicitly the location of the rings with respect to the virtual origin, within few percent error. The model is useful in jet noise for establishing compactness and source distribution. It is also useful toward the inverse radiation approach.
Subjective preference tests of simulated sound fields for optimum design of concert halls
Authors: Ando Y.
Published: October 2005

Results of preference tests of sound fields with a single echo showed that [Y. Ando, J. Acoust. Soc. Am. 62, 1436–1441 (1977); Y. Ando et al., Acustica 37, 111–117 (1977)]: (1) the preferred time delay gap between the direct sound and the first echo can be determined by the coherence of autocorrelation function of sound source and the amplitude of echo, (2) the preference score of sound fields increases by decreasing the degree of interaural crosscorrelation, so that the preferred echo direction is found in the range centered on 55° in the azimuth, and (3) the frequency-independent reflecting transfer function of walls is more preferable than the other, i.e., the spectrum of the first echo should not be differed from that of the direct sound. For the purpose of optimal acoustic design of concert halls, preference tests of simulated sound fields with early discrete echoes and the successive reverberation were conducted. The results confirm that the above findings can be applied to such sound fields also.
Frequency discrimination for tones of different levels but constant E/N
Authors: Dye Raymond H.
Published: October 2005

Wever's theory of pitch perception asserts that both place and periodicity mechanisms are used by the auditory nervous system, with periodicity predominant at low frequencies and place predominant at high. One way of gaining insight into the coexistence of the two mechanisms is to find an experimental parameter that, when varied, produces differential effects for low frequencies and high. To this end, signal level was varied and frequency discriminability was measured for tones of 500, 1000, and 4000 Hz. In order to examine the effects of level independent of E/N0, the level of a noise masker was chosen so that the signal-to-noise ratio remained constant. Raising level was found to improve frequency resolution in the 500and 1000-Hz conditions, while performance became worse in the 4000-Hz condition. The high frequency data can be explained in terms of Henning's modified energy detector model, with the added assumption that filter width increases with level. The low-frequency data are accounted for by a level-dependent increase in the number of active channels in the counting model of Luce and Green.
Compensation for occupational PTS for several railroad engineers and firemen
Authors: Campanella A. J.
Published: October 2005

The U.S. Department of Labor, Office of Workman's Compensation has a procedure, based on NIOSH findings, for locomotive engineers and firemen (among others) to obtain compensation for hearing PTS related to their working environment. Five such recent cases of successful compensation claims are reviewed for their audiogram, work history, percent hearing loss, and monetary compensation contents. Individuals claiming compensation were typically in their late 50's, had performed decades of railroad service, and had sustained sufficient PTS to render them unfit for unlimited railroad service. Environmental noise measurements were made in situ on some locomotives from April 1975 to July 1977 [A. J. Campanella, J. Acoust. Soc. Am. 60, S23(A)(1976)]. In retrospect, it was observed that the railroads, being exempt from OSHA procedures, are still not offering hearing protection to engineers and firemen. This is ostensibly because of the need for unprotected ear communication capability to assure safety in all their operations.
Keyword extraction based on syllabic information
Authors: Li K. P. , Kau C. K.
Published: October 2005

The syllabic structure of a spoken word in continuous speech can be extracted by a pattern of gross phonetic classes. A computer program has been written to transform continuous speech into these segmental patterns. In keyword extraction, portions of the speech utterance that display similar syllabic characteristics to the keyword reference pattern are identified. One approach to keyword extraction is to start with the phonemic transcription of the word and then utilize known phonological rules to accommodate syntactical environments and speaker variations. Another approach which yields comparable results is based on a Markov model incorporating time duration information and segment transitions to determine statistically whether the input segmental pattern is a candidate keyword. This approach which has been tested on continuous speech from a variety of different speakers and different keywords has application in a word recognition or word spotting system.
Paired comparison judgments in low-fidelity circuitry
Authors: Yonovitz A. , Ferrell Dianne R. , Harris Bernard
Published: October 2005

The effects of electroacoustic distortions on speech transmission through hearing aids has met with little agreement as to which characteristics are most detrimental to the understanding of speech. The purpose of this study was to examine the capacity of listeners to yield proximity (dissimilarity) matrices and preferences of hearing aids through the method of paired comparisons. Evaluative data consisted of pairwise judgements of 12 hearing aids. Fifty normal hearing adults representing five different acoustic competencies were used as listeners. These five competencies included (1) musicians; (2) audiologists; (3) engineers; (4) audio enthusiasts; and (5) noncategorical. Their task was to judge the dissimilarity between 78 pairs of hearing aid transduced spech and music on a nine-point scale and to mark their hearing aid preference. Characteristics for each of the 12 hearing aids were determined for the following parameters: overall frequency response, fundamental frequency response, second, third and total harmonic distortion, phase distortion, transient distortion, intermodulation distortion, and internal noise. This perceptual data was analyzed using procedures of multidimensional scaling. INDSCAL was used to determine those electroacoustic characteristics germane to the decision process of judging quality in low fidelity circuitry. Correlation analysis determined the regression structure of the derived electroacoustic distortions on preference. Data will be discussed examining the individual and group differences including the precision and variance by which subjects judge low-fidelity circuitry.
Montreal French vowels durations
Authors: Yaeger Malcah , Bourdeau Marc
Published: October 2005

Vowel duration in Montreal French (MF) is found to be directly related to the nature of the following consonant, the length of the word, the position of the vowel relative to the tonal syllable in the word, and the vowel's position relative to the tonic syllable of the breath group. Results of a factor analysis on data from seven speakers shows that most of the influences on duration are constant across class level and age, and are, in fact, quite similar to results reported for English materials [Klatt, J. Phonet. 3, 129–140 (1975)].
Sound transmission-loss characteristics of three-layer composite wall constructions
Authors: Moore James A.
Published: October 2005

The sound transmission-loss (TL) characteristics of three-layer composite wall constructions are considered. Analytical models predicting TL were developed for constructions with homogeneous isotropic cores or homogeneous orthotropic cores. Experimental results for composite panels with isotropic and orthotropic cores were compared with analytical predictions with acceptable agreement. A new design process for layered composite panels with high TL was developed. TL values exceeding mass law by as much as 10 dB are achievable. A composite panel was fabricated and provided a verification of the design process. Greater-than-mass law TL was obtained. [Work supported by NSF.]
Exterior-Wall Noise Rating (EWNR)—A single-number metric for evaluating outdoor-indoor noise reduction
Authors: Mange Gary
Published: October 2005

A brief review is made of previous single-number rating development with emphasis on the sound transmission class (STC). A single-number metric, called Exterior Wall Noise Rating (EWNR), has been developed in a similar, but independent, manner such that the A-weighted interior noise level, due to highway noise sources, can be estimated directly from the value of EWNR and the A-weighted exterior noise level. The development is defined first in terms of the basic theory for noise reduction from outdoors to indoors in a single frequency band. The overall effective noise reduction is a summation of these values over many frequency bands. The EWNR single-number rating replaces this complex summation and provides a greatly simplified calculation procedure. In addition, the EWNR can be used for direct comparisons of the noise insulation properties of exterior building elements. The EWNR metric is applicable to other exterior noise sources if the proper adjustment factors are included. Results are shown from recently conducted field tests which indicate good accuracy for predicting interior noise levels due to aircraft noise sources.
Evaluation of sound-insulating material for use underwater
Authors: Nicholls B. H. , House J. R. C.
Published: October 2005

The performance of underwater sound-insulating material varies both with frequency and hydrostatic pressure, and experimental evaluations were necessary to check theoretical predictions. A novel technique was devised for this purpose involving a back-to-back, dual-piston source with known radiating characteristics. Material evaluation consisted of the application to one piston face, provision being made to monitor changes in the radiation characteristics over a frequency range 600–10 000 Hz and at hydrostatic pressures to 2.76 MN m−2. The technique, designated SARC (Submersible Acoustic Radiation Comparator), was designed to operate within a suitable pressure vessel or, alternatively, in free-field deep-water conditions. Results obtained compared favorably with predicted material responses. The advantage of SARC over the more traditional evaluation methods, such as waveguide techniques, concern the increase in the scale of evaluation, and ease of data acquisition due to a real-time evaluation capability.
The free waves on an infinite thin fluid-loaded elastic plate
Authors: Crighton D. G.
Published: October 2005

The familiar problem of the response of a fluid-loaded thin elastic plate to external forcing is considered again, with the aim of determining those “free waves” of the coupled system which can actually be excited. General properties of the quintic equation whose roots yield the free wave numbers can be used to establish the character of these roots, and a generalized “coincidence condition” can be given at which the nature of the roots changes. It is argued, however, and demonstrated, that no significance can be attached to this condition, or to any of the roots except that single one characterizing an undamped subsonic surface wave. The contributions associated with all other waves may be altered at will by change of branch cuts in the complex wave-number plane, and thus these contributions do not represent physically meaningful free modes.
Intensity effects in optical fiber acoustic sensors
Authors: Carome E. F. , Satyshur M. P.
Published: October 2005

Studies are continuing on the use of long, low-loss optical fibers as direct acoustic sensors [J. A. Bucaro, E. F. Carome, and M. R. Layton, J. Acoust. Soc. Am. 62, S72 (1977)]. Currently we are employing a single-path coiled-fiber system and examining in detail acoustically induced variations in the transmitted light intensity. We have detected effects that we attribute to changes in the intensity of the light propagated in a particular fiber mode. Details of the experimental technique employed and quantitative data on these effects will be presented. [Supported in part by Office of Naval Research.]
Shock formation and cavitation as limitations on acoustic source level
Authors: Horneck F. R. , Young J. W.
Published: October 2005

The nonlinear phenomena of shock formation and cavitation impose different limitations on acoustic projector source level. It is found, however, that both effects can be conveniently parametrized in terms of the so-called scaled source level L0* of Mellen and Moffett [J. Acoust. Soc. Am. 61, 325 (1977)]. The well-known cavitation data of Esche [Akust. Beih. (Acustica) 4, AB208 (1952)] are reformulated as functions of L0*. and directivity index DI for a piston radiator. The shock wave formation distance is calculated in terms of these parameters and the Rayleigh distance R0. A linear model based on weak shock theory is used to combine existing plane and spherical wave models. The comparison between maximum achieveable L0* due to shock formation and cavitation is shown as a function of frequency with DI and water depth as parameter.
An adaptive-transform baseband coder
Authors: Makhoul John , Berouti Michael
Published: October 2005

In a baseband coder, the baseband may be coded in a number of different ways. In this paper, we describe coding of the baseband LPC residual using adaptive transform coding (ATC). At the receiver, high-frequency regeneration is accomplished by our recently developed method of spectral duplication. Care is taken to ensure that the harmonic structure is not interrupted as a result of the spectral duplication process. The results will be compared to time-domain coding of the baseband residual.
Channel vocoder quality improvements
Authors: Gold Bernard
Published: October 2005

This talk describes a set of experiments designed to improve the quality of a channel vocoder. Initial experiments indicated that a spectrally flattened synthesizer with 29 equally spaced channels covering a band of 3600 Hz gave good spectral fidelity; however, the essentially zero phase of the excitation caused the usual “buzziness.” Several methods will be described whereby quality is improved by manipulating the phase of the excitation. These include both fixed frequency dependent phase differences plus dynamic phase differences generated by a linear predictive vocoder. [Work sponsored by Defense Advanced Research Projects Agency.]
Experimental evaluation of voice conferencing techniques
Authors: Forgie J. W.
Published: October 2005

The requirement to support voice conferences in which some participants have only narrow-band speech communication capability prevents the use of the conventional signal summation (bridging) technique. In order to evaluate the relative acceptability of alternative signal selection techniques, a conferencing simulation facility has been constructed, and a series of experiments has been designed and carried out. The effects of conference size, communication delays, and voice encoding techniques have been investigated. Conference control based on speech activity detection has been compared with control using push-to-talk switches or tone keys. Both centralized and distributed control algorithms have been simulated. Test scenarios have been developed that provide both task performance measurements and subjective judgments of conferencing system adequacy. Results indicate that simple broadcast protocols are generally preferred over more complex schemes and that signal selection techniques can perform comparably to signal summation. [This work was sponsored by the Defense Communications Agency for the Department of Defense.]
Acoustic correlates of some phonetic categories
Authors: Stevens K. N.
Published: October 2005

A review is given of the properties of speech sounds that identify different phonetic categories. It is hypothesized that the auditory system gives a distinctive response to sounds with these special properties and hence facilitates classification of the sounds in terms of phonetic categories. The properties that will be discussed include: presence of a rapid spectrum change; presence of an abrupt increase in amplitude; a narrow midfrequency spectral prominence as opposed to a diffuse spread of spectral energy; spectral energy concentrated at high frequencies as opposed to low frequencies; periodic as opposed to nonperiodic sound; spectral characteristics of vowels; presence of low-frequency sound preceding an onset; two onsets that are simultaneous as opposed to sequential. Evidence for these acoustic properties is presented both from acoustic analysis and from experiments on the perception of speech and speechlike sounds. [Supported by NIH grants.]
Representation of speech-like sounds in the discharge patterns of auditory-nerve fibers
Authors: Delgutte B.
Published: October 2005

It is now possible to study how acoustic characteristics important for speech discrimination are represented in the discharge patterns of single auditory-nerve fibers. Responses to simple tonal stimuli and speechlike sounds (such as one or two formant synthetic stimuli) have been studied in order to elucidate the coding of certain vowel characteristics. At stimulus levels typical of speech, information about formant frequency is carried predominantly by fibers with characteristic frequencies close to the formant frequency, whereas information about fundamental frequency is present in fibers over a wide range of characteristic frequencies. Specific examples will be given for the representation of detailed spectral patterns for vowels and fricative consonants, of rapid changes in intensity and spectrum for stop and nasal consonants, and of speech signals in the presence of background noise. [Supported in part by NIH grants.]
Effects of nonlinearities on speech encoding in the auditory nerve
Authors: Sachs M. B. , Young E. D.
Published: October 2005

In this paper we will consider some effects of auditory-nerve nonlinearities on the representation of complex stimuli such as vowels. At low sound levels, profiles of discharge rate versus characteristic frequency in populations of auditory-nerve fibers show well-defined peaks at frequencies corresponding to the formants of a vowel stimulus. At levels above about 60 dB SPL, these peaks are not seen, primarily because of effects of rate saturation. In addition, effects related to two-tone suppression act strongly on units with characteristic frequencies in the vicinity of the second and third formants, suppressing their responses at high levels and contributing to the loss of distinct peaks in this region. Units with spontaneous rates less than about 1/s show the effects of suppression more dramatically than do higher spontaneous units; despite the suppression, however, this population retains formant peaks in its rate profiles up to levels at least 20 dB higher than does the higher spontaneous rate population. The temporal patterns of response of auditory-nerve fibers contain considerable information about the spectrum of a steady state vowel. In general, phaselocking to the formant frequencies is stronger than locking to other harmonics of the stimulus. As sound level increases, synchrony to the formant frequencies saturates. However, responses to nonformant harmonics are suppressed by responses to the formant harmonics. This synchrony suppression allows the phaselocking to the formant frequencies to remain dominant even at high levels. The exception to this behavior is locking to harmonics and intermodulation products of the first two formant frequencies (2F1, 3F1, F2−F1, etc.) which increases rapidly in amplitude at higher sound levels. These distortion products are, to some extent, a consequence of the rectification inherent in hair cell/nerve fiber transduction and may also reflect the presence of propagating combination tones. [Supported by grants from the National Institute of Neurological and Communicative Disorders and Stroke.]
On the applicability of nonlinear system theory to the modeling of responses of the peripheral auditory system to complex stimuli
Authors: Johnson D. H.
Published: October 2005

The responses of single auditory-nerve fibers to simple stimuli (e.g., single tones, clicks) have been studied extensively. The modeling of these response patterns has certainly enlarged our knowledge of how such simple stimuli are processed by the peripheral auditory system. In contrast, such stimuli do not occur often in the natural environment. Current research is turning to the analysis of the responses to complex stimuli, a major example of which are the responses to speech. Nonlinear system theory, in particular, the contributions due to Wiener, yields a systematic method of describing any system—linear or nonlinear. This report will review critically Wiener theory with respect to the special issues of concern in the peripheral auditory system. Attention will focus on whether Wiener theory may yield extra information on models of the peripheral auditory system and whether studies employing this method may be fruitful.