bluuebunny/crossref_metadata_2025 huggingface dataset:
Theory of nonlinear electroacoustics of dielectric, piezoelectric, and pyroelectric crystals
Authors: Nelson D. F.
Published: October 2005

Based on a fully electrodynamic Lagrangian theory of elastic dielectrics, a completely deductive derivation of the dynamical equations and constitutive relations for elastic, electric, and electroelastic phenomena is presented. The equations apply to crystals of arbitrary symmetry, structural complexity, and nonlinearity. A key technique of this work is the use of material frame electromagnetic fields. We show that the electrostriction tensor differs from the low-frequency limit of the elastooptic tensor by the Maxwell stress tensor of a polarizable medium. A general expression containing three new terms for the effective third-order elastic constant measured by mixing of two input acoustic waves is obtained. Our expression is more general in applying to all crystals and to acoustic waves of all types propagating collinearly or noncollinearly in any direction. We derive the coupled mode equations governing the Thompson-Quate experiment, that is, the parametric interaction of two counterpropagating bulk acoustic waves and a spatially uniform electric field oscillating at the second harmonic of the acoustic frequency. A general expression for the effective nonlinear piezoelectric tensor governing this process is derived.
Relationship between solid nonlinearity parameters and thermodynamic Grüneisen parameters
Authors: Nakamura Akira , Cantrell John H. , Breazeale M. A.
Published: October 2005

The propagation of an initially sinusoidal ultrasonic wave in a nonlinear solid can be described in terms of a nonlinearity parameter whose magnitude is a function of the orientation in the solid. The nonlinearity parameter, which can be evaluated from data on the magnitude of the fundamental and second harmonic of the initially sinusoidal wave, is only slightly dependent upon temperature in those samples studied. (We have measured to 3 °K in copper and germanium single crystals and in fused silica.) This fact, plus a number of other considerations, has led us to realize that there is an intimate relation between the acoustical nonlinearity parameter and the thermodynamic mode Grüneisen parameters. This relation is discussed, and the relationship is given between the nonlinearity parameters for the principal directions in a cubic crystal and the appropriate mode Grüneisen parameters. [Research supported in part by the Office of Naval Research.]
On the road to computational acoustics
Authors: Hogge H. D.
Published: October 2005

Acousticians have largely overlooked the methods of computational fluid dynamics (i.e., the direct numerical integration of the nonsteady, compressible continuity, momentum, and energy equations) because of the success of the linearized normal-mode approach and because the numerical viscosity inherent in traditional computational methods damp out acoustic disturbances at an unrealistic rate. The advantage of the computational approach is that it allows inclusion of physical phenomena excluded from the linearized normal-mode approach such as nonlinear convection, nonisentropic losses, and phase change effects. The recent development of SHASTA, a relatively nondiffusive computational method [J. P. Boris and D. L. Book, J. Comp. Phys. 11, 38–69 (1973)], has made possible the accurate solutions to acoustics problems. SHASTA is applied to a piston driven shock wave, an acoustic traveling wave, and an acoustic standing wave. The solutions of these problems by other standard numerical schemes are shown for comparison. It is found that only SHASTA is acceptable for all problems considered. As a practical example the computational approach is applied to the acoustic-wave/entropy-wave interaction associated with reflections from a choked flow wall. [Work supported by DARPA.]
Analysis of nonlinear self-scattering of a surface wave using energy conservation
Authors: Woodsum Harvey C.
Published: October 2005

Coupled differential equations for amplitudes of the harmonic components in a surface wave generated by nonlinear self-scattering are derived using the energy conservation equations due to Westervelt. [Proc. 3rd Int. Congr. Acoust., edited by L. Cremer (Elsevier, Amsterdam, 1960), p. 316] These equations are compared to the phenomenological model of Lean and Tseng [J. Appl. Phys. 41, 3912 (1970)], which has been successful in explaining experimental results. Similar equations are displayed for harmonic generation in a fluid. The equations for the fluid case are the same as for the surface wave case, except that the surface waves are dispersive, and Grüneisen's constant corresponds to the parameter of nonlinearity of the fluid.
Detection in diotic and dichotic conditions using repeatable samples of noise
Authors: Frank Alan S. , Gilkey Robert H. , Robinson Donald E.
Published: October 2005

Previous research using repeatable samples of noise, frozen noise, has concentrated on evaluating models of signal detection [A. Ahumuda, Jr., R. Marken, and A. Sandusky, J. Acoust. Soc. Am. 57, 385–390 (1975)] or estimating the “internal” noise of the auditory system [D. M. Green, Psychol. Rev. 71, 392–407 (1964)]. The present research investigates the detectability of a 500-Hz sinusoid, masked with different samples of frozen noise under both diotic and dichotic conditions. The results from several samples of noise and starting phase angles of the sinusoid will be presented. The data show differences in detection across different samples of noise as well as differences between diotic and dichotic conditions. The ability of various theories of binaural processing to account for the results will be discussed. [Work supported by NSF research Grant BNS-77-17308 and PHS Mathematical Psychology Training Grant MH-11219.]
Frequency selectivity under diotic and dichotic listening conditions
Authors: Yams Mark F. , Robinson D. E.
Published: October 2005

Results of several experiments on binaural hearing suggest that the auditory filter employed in MLD producing conditions is wider than that used under diotic or monaural listening conditions. The present experiment tests this hypothesis using Greenberg and Larkin's probe frequency technique. [J. Acoust. Soc. Am. 44, 1513–1523 (1968)]. Subjects were first trained in the detection of a 500-Hz tone and psychometric functions were obtained for NOSO and NOSπ masking conditions. In the next part of the experiment, and unknown to the subjects, probe frequencies both higher and lower in frequency than 500 Hz were substituted for the primary signal with a probability of 0.25. Performance at each probe frequency was then converted to an equivalent amount of attenuation in decibels, via the psychometric functions, and corrected for changes in the MLD with frequency. Resulting filter characteristics will be discussed in terms of the hypothesis that the filter for the binaural system is wider than that for the monaural system.
Effects of auditory fatigue an masked-speech intelligibility
Authors: Parker D. E. , Martens W. L. , Johnston P. A.
Published: October 2005

Speech intelligibility (percentage of words correctly reported in a shadowing task) was determined before and after fatigue produced by a 5-min exposure to an octave band of noise (1500–3000 Hz) set at 115 dB. The stimulus word lists and the masking noise in which the words were embedded were filtered to a 2825–3185-Hz band. Masking-noise levels were varied within and across experiments. Experiment 1 employed masking-noise levels of 0, 15, and 40 dB SL; experiment 2 masking levels were 15, 40, and 65 dB SL; masking levels in experiment 3 were 40, 65, and 90 dB SL. Speech stimulus intensity was adjusted to obtain approximately 50% intelligibility before fatigue at each masking-noise level. Following fatigue, intelligibility decreased when the speech was embedded in 0- and 15-dB SL masking noise, remained approximately constant when the speech was embedded in 40-dB SL masking noise and increased when the speech was embedded in 65- and 90-dB SL masking noise. The results obtained with the speech stimuli are incongruent with earlier results employing pure-tone stimuli [D. E. Parker, R. L. Tubbs, P. A. Johnson, and L. S. Johnston, J. Acoust. Soc. Am. 60, 881–885 (1976)] and fail to support a previously developed recruitment model. Rather, the speech observations support a model in which the ratio of the speech signal to the masking noise is improved following fatigue. [Work supported by USAF.]
Integration of duration and frequency in pure-tone discrimination: Effect of the temporal and spectral separation between the cues
Authors: Espinoza-Varas Blas
Published: October 2005

In previous work [J. Acoust. Soc. Am. 63, S30(A) (1978)] we examined how listeners use information from two acoustic dimensions to discriminate between tones. For cues consisting of a change in duration followed by a change in frequency, listeners could trade information between dimensions to maintain a constant performance level. It was proposed that the two dimensions were integrated. In this study we investigated how this tradeoff was influenced by the temporal and spectral separations between the tones in which the changes occurred. The procedure was a same-different discrimination between pairs of tones. Each pair consisted of two 80-ms tone bursts, separated by an interval of either 10, 60, of 120 ms. The frequency of the first tone was either equal to that of the second tone (1500 Hz), higher (1/4 octave), or lower (1/2 or 1 octave). In all cases the intensity was 75 dB SPL. For the “different” trials the comparison pair contained increments in the duration (Δt) of the first tone and/or in the frequency (Δf) of the second tone. The just noticeable Δt (d′ = 2.0) was measured as a function of Δf, and vice versa. Tradeoff was observed for all temporal and spectral separations between the tones in the pair. No major differences appeared among the various conditions. In all cases the discriminability of the complex was well predicted from the discriminability of the single dimensions.
A consideration on loudness discrimination of tonal signals with changing intensity
Authors: Suzuki Y. , Sone T. , Nimura T.
Published: October 2005

This study aims to investigate how sensitively people perceive the intensity change of a tone. The stimulus tone consisted of a logarithmically changing part and some steady parts. When the stimulus was made up of just a changing part, the detectable limen was about 2.4 dB for 50–100-ms duration. This value gradually decreased and it reached about 1 dB as the duration was elongated. These limens did not so strongly depend on the Icyels of stimuli as in the loudness discrimination of tone bursts. This seems to show that detection of intensity change is attributable to a mechanism different from loudness discrimination. When the stimulus had both preceding and trailing steady parts, limens were determined by the difference in intensity in both parts when their duration exceeded 50 ms. When only the preceding part was steady, the longer its duration was, the smaller the limen was. Conversely, too long trailing steady part (>100 ms) made the limen increase. We propose a model to explain some of these findings. This consisted of an LPF, a differentiator, a gate, an integrator, and a detector.
Discrimination of tonal patterns with several levels of stimulus uncertainty
Authors: Watson C. S. , Kelly W. J.
Published: October 2005

The discrimination of word-length tonal patterns has previously been shown to be strongly dependent on the level of stimulus uncertainty of the psychophysical procedure employed [Watson, Kelly, and Wroton, J. Acoust. Soc. Am. 60 (1976)]. An experiment was conducted to determine the range of stimulus uncertainty effects and the time course of discrimination training, under various levels of stimulus uncertainty. Listeners were taught to detect differences between tonal patterns using a three-alternative, forced-choice adaptive psychophysical procedure. The differences to be detected were increments in frequency, level, or duration of one of four target tones whose frequency, duration, intensity, and temporal position were constant in the standard patterns. Asymptotic discrimination performance for the 45-ms target tones presented in the context of a temporal sequence of ten tones with a total duration of 450-ms, ranged from mean threshold values of ΔI, ΔT, ΔF of 13 dB, 40 ms, and 1100 Hz, respectively, under the highest levels of stimulus uncertainty, to levels only slightly greater than those measured for the target tones in isolation. [Work supported by a Biomedical Research Development Grant from NIH.]
Informational masking in word-length auditory patterns
Authors: Watson C. S. , Kelly W. J.
Published: October 2005

Extremely imprecise resolution of the components of word-length tonal sequences, under conditions of high stimulus uncertainty, has led us to reconsider an earlier conclusion about the effective level of the components of such patterns. Watson and Spiegel [J. Acoust. Soc. Am. 58, S84 (1975)] reported that temporal masking of individual components of the tonal patterns used in their experiments amounted to no more than 15–20 dB. The sensation levels of the 75 dB SPL components of their patterns were therefore estimated to be at least 40–50 dB, which provided no explanation for extremely poor temporal, spectral, and intensive resolution of those components. We now realize that an erroneous conclusion was made in that experiment, since resolution measured under high stimulus uncertainty was compared to detection thresholds measured in a minimal-uncertainty procedure. When component thresholds are measured under conditions of minimal, intermediate, and high stimulus uncertainty, the sensation levels of 75-dB SPL components range from 50 to −5 dB, values consistent with listener's resolution of the same components. [Work supported by a Biomedical Research Development Grant from NIH.]
Within-subject variability in confidence ratings as a function of signal-to-noise and experimental method
Authors: Corner Paul T.
Published: October 2005

It is generally agreed upon that the criterion of a subject varies during an experiment, and that this variability is proportional to the number of criteria a subject must maintain. To assess response variability, two experimental methods were employed, one having a fixed signal-to-noise ratio (SNR) and the other a variable SNR. The task involved detecting a 500-Hz centered band of noise presented in noise. Subjects responded on a size-point confidence rating scale indicating their decision regarding the presence or absence of the dichotomous band. Within the context of a larger study in which data from 48 experimental sessions were collected, four subjects each listened to 15 prerecorded experimental sessions twice. Correlations were obtained for each method at a variety of SNRs, each based on 24 or 30 pairs of responses. The average correlations for the fixed and variable methods were 0.91 and 0.27, respectively, and were related to the SNR of the signal. Implications for repeated measures designs and pooling of data are discussed.
Pitch of complex tones for normal and hearing-impaired listeners
Authors: Wightman Frederic , McAdams Stephen E.
Published: October 2005

Current pitch models emphasize the ability of the hearing mechanism to resolve spectral components, and either deemphasize or leave unspecified the importance of temporal processing. Our study attempts to assess the relative importance for pitch perception of spectral and temporal encoding. Three groups of listeners were tested in a simple two-interval pitch DL task: normal listeners, listeners with impaired spectral processing (as indicated by abnormally broad psychophysical tuning curves), and listeners with impaired spectral and temporal processing (as indicated by broad tuning curves and poor gap-detection performance.) The task was identification of the direction of change of the (missing) fundamental, from one interval to the next, of a four-component harmonic complex. Pure tone frequency DLs were also measured. All hearing-impaired listeners performed less well than normals, though listeners with both spectral and temporal deficiencies performed considerably poorer than those with spectral deficiencies alone. Further, pitch DLs were much larger than pure-tone DLs in listeners with temporal deficiencies. These results suggest that the role of temporal processing in pitch perception may be much greater than current theories indicate. [Work supported by NIH NS12045.]
Computational models of speech sound generation
Authors: Flanagan J. L.
Published: October 2005

A comprehensive characterization of the acoustics of speech sound generation is central to the techniques of analysis/synthesis telephony. The classical model of linearly separable excitation source and resonant acoustic system has, for many years, served vocoder technology and speech-synthesis efforts. Now, analytical understanding of vocal-system acoustics is more extensive, is better quantified, and can be more readily represented in digital form. Improvements may ensue, therefore, from adopting a more sophisticated—yet computationally tractable—model of the speech signal. We review here analytical formulations that aim to describe the dominant acoustic effects in speech production. In particular we include models to represent the self-oscillating properties of the vocal cords, the acoustic interaction between cords and tract, noise generation from turbulent air flow, sound radiation from the mouth, nose and yielding tract walls, and the dynamics of vocal-tract shape. Finally, we offer preliminary examples of how the composite system, formulated for computer simulation, can be utilized for speech analysis and for speech synthesis.
Synthesis by rule of consonant-vowel syllables
Authors: Klatt Dennis H.
Published: October 2005

A FORTRAN synthesis-by-rule program has been developed for controlling a cascade/parallel formant synthesizer. The rules have been evaluated by generating consonant-vowel-/t/syllables involving all phonologically legal combinations of the consonants /PP, BB, TT, DD, KK, GG, CH, JH, FF, VV, TH, DH, SS, ZZ, SH, WW, YY, RR, LL, HH, MM, NN/ with the vowels /IY, IH, EY, EH, AE, AA, AH, AO, OW, UH, UW, ER, AY, OY, AW, YU/. Five phonetically sophisticated listeners transcribed a randomized list of 336 syllables. There was no prior familiarization with the synthesis. Ninety-five percent of the consonants and 98% of the vowels were correctly transcribed. Synthesis rules are based on a detailed acoustic analysis of the same syllables spoken by three male and three female talkers. The analysis indicates that some consonants require two or three allopones for accurate synthesis (one before the front vowels /YU, IY, IH, EY, EH, AE/, one before the back unrounded vowels /AA, AY, AW, AH/, and one before the rounded vowels /AO, OW, UH, UW, ER, OY/). Frication spectra and plosive burst spectra are largely invariant before each of these vowel groups. Formant transitions for a consonant can be described by a modified locus theory within each vowel group. [Supported in part by the National Institutes of Health.]
A two-state model of F0 control
Authors: Hanson R. J.
Published: October 2005

A two-state model of F0 dynamics in speech production has been developed which characterizes F0 control as a succession of alternating steady-state and transition control states. The control states have been defined in terms of pitch period distribution parameters obtained in an earlier study [R. J. Hanson and P. F. MacNeilage, J. Acoust. Soc. Am. 55, S31(P8) (1974)]. The model attempts to synthesize observations about sensorimotor integration in speech while capitalizing on differences in the perception and production of steady-state versus transition portions of the speech signal. A preliminary version of the model has been implemented in a FORTRAN program and used to recognize certain control state changes in a database gathered in two previous studies [R. J. Hanson, Proc. 9th Int. Congr. Acoust. 1, 519 (1977), and R. J. Hanson, 7th Int. Congr. Phon. Sci. (1975)]. The results produced by the model compare very favorably with those produced by hand segmentation, indicating the viability of defining control states with distributional parameters.
Ocean basin reverberation at very low frequencies
Authors: Blatstein I. M.
Published: October 2005

At low frequencies (<20 Hz), energy propagates very efficiently in the ocean. Acoustic energy can scatter off basin walls, islands, and sea mounts and propagate to a receiver as reverberation. For underwater explosions, large sources of acoustic energy, ocean basin reverberation can be especially significant. It may, for example, be useful in determining bottom and propagation parameters. Several models for the encounter of the acoustic energy with the basin wall have been suggested. These range from a simple scattering representation to more detailed multiple scattering or multiple reflection models. These models will be discussed, as well as reverberation data from several CHASE shots. Quantities focused on will be the reverberation level versus time and frequency dependence. The contribution of basin reverberation to low-frequency background noise will be discussed. [Work supported by DNA.]
Sources for very low-frequency underwater acoustics
Authors: Young A. M.
Published: October 2005

There has been an increasing demand for high-power, very low-frequency sound sources for use in various applications ranging from long-range propagation studies to the calibration of large low-frequency receiving arrays. Virtually all such sources have the undesirable characteristics of large size, low efficiency, relatively poor reliability, and high costs. A review of the basic problems associated with the radiation of very low-frequency acoustic energy in the ocean will be presented and will show that these undesirable characteristics are direct results of the physics of the radiation problem itself. The available transducers and transduction mechanisms will be discussed in terms of their relative advantages and disadvantages. Compromises between design parameters and performance requirements will be addressed.
Limitations of ocean noise measuring hydrophones at frequencies below 20 Hz
Authors: Hugus G. D.
Published: October 2005

Noise measuring hydrophones, used at sea, produced spurious outputs which severly degrade their measurement accuracy. When suspended from a ship or buoy, a hydrophone can experience large vertical periodic displacements due to surface waves, subsurface waves, or subsurface currents. A pressure sensitive hydrophone, when subjected to vertical periodic displacements, produces voltage outputs due to two significant nonacoustic effects: (1) sensitivity to the resultant periodic hydrostatic pressure amplitude and (2) sensitivity to the inertial acceleration of the sensor element. The first effect, predominates at frequencies below 20 Hz and will be addressed in this paper. Data is given for the acoustic pressure equivalent and hydrophone sensitivity to typical vertical displacement amplitudes. This effect is shown in data for a noise measuring hydrophone. Equations and curves are given showing the maximum permissible vertical displacement amplitude to produce a 20-dB signal-to-noise ratio for typical ambient sound pressure levels in the ocean. Methods and constraints for reducing this problem to an acceptable level are given.
Very low-frequency sound transmission in the Arctic seabed
Authors: Baggeroer A. B. , Koch S. , Shepard G. W. , Dyer I.
Published: October 2005

The system described in the preceding paper by Zittel et al. was used to measure sound propogation in the Arctic seabed by exploding TNT charges on a line up to 80 km from the multichannel array. At these distances and very low frequencies, energy propogates in seismic paths that are associated with strata interfaces within the seabed. The array was used to measure the phase and group velocities of these paths. The path losses versus frequency were also analyzed in an attempt to estimate the transmission losses in the strata. This information plus seismic reflection data was then used to analyze the structure of the Arctic seabed near the camp. [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.]
Down-slope enhancement models for low-frequency ocean noise and propagation
Authors: Northrop J. , Wagstaff R. A.
Published: October 2005

A simple model has been derived for down-slope enhancement (DSE) of underwater sound (the so-called "Megaphone effect"). The model predicts DSE for shallow source/deep receiver travel paths across a sloping bottom as the difference between semispherical spreading (15 logR) + αR and cylindrical spreading (10 logR) + αR plus a constant K. In this formulation the frequency dependent attenuation term (αR) is canceled out by the subtraction operation leaving the simple formula: DSE = (5 logR) − K. By comparing measured values of DSE with those computed using this formula, the value of the constant K was found to be about 5 dB when the range R is expressed in nautical miles. DSE for sound crossing the sloping sides of seamounts was found to be about 5 dB higher than that for continental slopes. These models have been used to compute the effective low-frequency source levels for shipping lanes crossing the continental slopes in areas where the horizontal directionality of ambient noise has been measured. [Work supported by LRAPP.]
Surface decoupling effects in ambient noise modeling
Authors: Wagstaff R. A. , Bannister R. W. , White D.
Published: October 2005

Image interference effects near the surface are known to affect the coupling of surface energy from shallow sound sources into propagation channels. Since sound sources which contribute most to ambient noise in the ocean are located within the upper 20 m of the water column, "surface decoupling" is an important factor in noise modeling. The presentation deals with the sensitivity of the surface decoupling effect to frequency, range, sound depth, bottom loss, bottom slope, and water depth and concludes that significant regional changes can result. The implication of geographic changes in ambient noise are modelled by comparing two different noise regimes—one with predominantly close shipping in a bottom limited environment, the other having more distant shipping and a sound speed excess. Changes in received omnidirectional spectra are interpreted in terms at surface decoupling effects. [Work supported by Office of Naval Research (LRAPP program).]
Long-range explosive sound propagation in the adjacent sea of Japan
Authors: Otsubo H. , Ishiwata T. , Ohta K.
Published: October 2005

Since several years ago we have been working over the ocean acoustical research which ranges from the shallow to the very deep waters. In this report, a newly developed ocean-acoustic data acquisition system is described first. Several sound propagation losses versus range plots with various source and receiver depths of interest are introduced and discussed with the theoretical models; normal mode, modified ray tracings, and the FFP [F. R. DiNapoli, NUSC Rep. No. 4103 (1971)]. It is exemplified that, in any cases, the validity of the loss prediction models are controlled by bottom loss information. We confirmed the information can be conveniently supported by Hamilton's sea floor geoacoustic modelling [E. L. Hamilton, Ocean Acoust. Modeling, Pt. 4—Sea Bottom, SACLANTCEN (1975)] and Morris' linear gradient bottom sediment model [J. Acoust. Soc. Am. 48, 1198–1202 (1970)].
Glass fiber products for industrial noise control
Authors: Harris David A.
Published: October 2005

A wide range of glass fiber products may be employed to reduce excessive industrial noise levels. Descriptions of these products, specific examples of their use as ceiling and sidewall absorbers, within sound attenuation barriers, air-duct attenuation, pipe quieting, etc., and their acoustical values will be presented.
Experimental investigation of quiet portable air compressor noise escalation resulting from degradation of noise control features
Authors: Gray L. , Broe J. , Mehta A. , Mirabelli R.
Published: October 2005

A series of tests have been conducted to experimentally show the overall portable air compressor (PAC) noise escalation resulting from the degradation of design features intended to control air and structure-borne PAC noise. These experiments were designed to determine the potential noise escalation associated with the degradation of each noise abatement feature individually and in combination with the other noise abatement features. Using information generated in the course of the investigation, a time-dependent correlation has been developed showing the expected overall machine noise escalations associated with typical usage. The potential maximum and minimum noise escalations attained from varying degrees of machine maintenance were similarly determined. Test results have shown that an increase in noise as much as 18.5 dBA can occur when the prime noise suppression design features of the PAC are degraded. A tape recording will be played to give a subjective appreciation of the level and character of noise escalation from the degradation of various sound control features. [Work supported by EPA.]
One-dimensional analysis of expansion-chamber-type muffler with flow
Authors: Okda Josuke
Published: October 2005

The characteristics of the expansion-chamber-type muffler with mean flow was analyzed by means of a one-dimensional model. Under the condition of continuity of mass flow and acoustic power at the boundary, the reflection coefficient of sound pressure was determined, and the modified four-terminal constants of a pipe and the transfer matrix connecting two pipes were deduced. The reflection coefficient is given by −[(1 − M)/(1 + M)][(1 − z)/(1 + z)] and the relation between sound pressure and volume velocity at the end of the pipe is given by an assumed impedance (M + z) Z0, where Z0 is the characteristic impedance of the pipe in absence of mean flow, M is Mach number, and z Z0 is the load impedance. When there is mean flow, sound pressures of both side of a boundary are not equal to each other generally. For given input sound pressure or volume velocity, the radiation power from the expansion chamber type muffler can be calculated by means of the complete modified four-terminal constants of the whole muffler.
Behavior and psychophysics of hearing in birds
Authors: Dooling R.
Published: October 2005

Psychophysical investigations of hearing in a number of avian species over the last decade have added significantly to our knowledge of hearing capabilities characteristic of this vertebrate group. Behavioral measures of absolute auditory sensitivity in a wide variety of bird species show a region of maximum sensitivity between 1 and 5 kHz with a rapid decrease in sensitivity at higher frequencies. On the basis of this general measure, birds fall between two other major vertebrate groups: reptiles and mammals. Discrimination and masking data from birds include measures of frequency, intensity, and duration difference limens; critical ratios, critical bands, and psychophysical tuning curves. There are also data on temporal summation, temporal resolving power, and temporary threshold shift from noise exposure. Taken together these data suggest that, in the region of 1–5 kHz, birds show a level of hearing sensitivity similar in most respects to that found for the most sensitive members of the class Mammalia with avian performance clearly inferior above and below this range of frequencies. Possible exceptions to this general picture must include the echolocating oilbird and growing evidence that pigeons are sensitive to infrasound at moderate intensity levels. The relation among critical ratio, critical band, and intensity difference limen measures in the parakeet is similar to that described for the human, but the pattern of masking as a function of frequency is dramatically different from that observed in mammals. Examples of a correspondence between hearing sensitivity and vocalizations can be demonstrated in a number of species.
Identification of a /sa/-/sta/ continuum by normally developing and language-delayed children
Authors: Tallal Paula , Stark Rachel
Published: October 2005

Previous studies have demonstrated that language-delayed children as a group are impaired in processing nonverbal and verbal auditory stimuli that change rapidly over time, but in processing similar stimuli that change more slowly [P. Tallal and M. Piercy, Neuropsychol. 13, 69–74, (1975)]. In the present study, 36 language-delayed children and 36 controls were presented with tokens from a /sa/—/sta/ continuum. The syllables were synthesized to differ only in the duration of silence between the offset of frication and the onset of vowel formants which ranged from 0 ms in /sa/ to 100 ms in /sta/. A nonverbal target identification procedure was used in obtaining responses. The language-delayed children were unimpaired on the two end points of the continuum. However, they were either inconsistent in their responses to the entire continuum or required a significantly longer silent period than the normals to identify a syllable as /sta/, as evidenced by a shift in the crossover point on their phoneme boundary functions. Mechanisms which might account for the temporal processing difficulties of language-delayed children and their relation to the language disorder will be discussed. [Work supported by NINCDS.]
Some characteristics of oral air flow for fricative consonants produced by hearing-impaired speakers
Authors: Whitehead R. L. , Barefoot S. M. , Lieberth A. K.
Published: October 2005

The average volume velocities of oral air flow during production of fricative consonants by ten normal-hearing, ten moderately to severely hearing-impaired and ten profoundly-deaf adult males were investigated. Air flow was measured during production of CV and VCV syllables consisting of the vowel /a/ and the consonants /z/, /s/, /v/, and /f/. All hearing-impaired and deaf subjects had congenital hearing losses and were able to intelligibly produce the syllables. The results indicated that for all three groups of subjects, in both syllable environments, the voiceless fricatives were characterized by greater oral air flow than their voiced cognates. In addition, in both the CV and VCV contexts, the average individual flow rates associated with each of the four fricatives were similar for the normal-hearing and hearing-impaired groups. With respect to the deaf subjects, however, the average individual flow rates for each of the four fricatives were significantly greater in the CV context and significantly less in the VCV context, when compared with the normal-hearing and hearing-impaired subjects. [Work supported by HEW.]
Voiced stop-consonant discrimination in adult aphasics: The effect of formant transition duration
Authors: Kitselman Kurt
Published: October 2005

Twelve adult aphasics and 12 nonaphasic controls were tested for ability to discriminate (AX method) place differences between synthetic stop-consonants as a function of formant transition duration. Two 11-item arrays were generated, one with 40-ms transitions and the second with 25-ms transitions, and each array ranging in equal acoustic steps from [ba] through [da] to [ga]. Syllables were paired within these arrays at a two-step difference level. The Iwo sets of stimulus pairs were intermixed for discrimination testing. Discrimination functions were computed by pooling response data across subjects within each subject group and stimulus set. Rank correlation coefficients (tau) were computed to compare discrimination functions within subject groups and to compare stimulus matched discrimination functions between subject groups. The within-group comparisons revealed a greater association between the two functions within the control group than within the aphasic group. The between-group comparisons revealed a greater association between the two functions obtained with 40-ms transition stimuli. Formant transition duration significantly affected phonetic category boundries only in the aphasic group.
Voice onset and offset RTs in spastic dysphonia
Authors: Izdebski K. , Shipp T. , Dedo H. H.
Published: October 2005

Recent evidence suggests peripheral and/or central nervous system involvement in the etiology of spastic dysphonia, a communicative disorder characterized by difficulties in voice control, specifically voice initiation, termination, and maintenance. Temporal measures of vocal RTs to auditory stimulus, for voice initiation and termination, were obtained in ten adult male and female patients suspected of having spastic dysphonia. These auditory-vocal RTs were obtained for voice within and above the voice frequency used in normal speech production, since spastic dysphonic patients often exhibit differences in control of voice within and above the speech frequency range. The temporal measures obtained are used to explore the nature of RT variability as indications of their peripheral or central locus. The derived data are also compared with available RT values for other voice pathologies which presumably affect control and timing of laryngeal behavior (stuttering and dysarthrias). [Work supported by NIH.]
Time and intensity cues to prosodic features of sentences
Authors: Farrar C. L.
Published: October 2005

Profoundly deaf persons hear only time and intensity variations in amplified speech and must rely on lipreading to perceive speech segments [N. Erber, J. Acoust. Soc. Am. 51, 1224–1227 (1972)]. These persons may perceive prosodic features of sentences by hearing. It is not known whether linguistically significant variations in duration and intensity of speech segments are perceived by reference to the inherent duration of those segments and thus, are dependent upon intelligibility. Subjects responded to three sentence tests wherein the sentences differed in either phrase-structure, major constituent boundary (MCB), or contrastive stress. Responses were by forced choice to the prosodic features; subjects were not required to identify words. Stimuli were presented under audition-alone, vision-alone, and audiovisual conditions. The acoustic signal was noise, amplitude modulated by the sentence materials. The audiovisual condition resulted in highest mean scores. Identifying phrase-structure, audition-alone scores were near chance levels. Vision-alone and audiovisual scores were close in value, suggesting that time and intensity cues supplemented visual cues to a small extent. Identifying MCB location, vision-alone scores were near chance levels. Audition-alone scores were equivalent to audiovisual scores, suggesting that time and intensity variations provided primary cues. Identifying contrastive stress location, score data suggested that the acoustic signal allowed detection of contrastive stress and the exact word was located with visual cues. These results indicate that an unintelligible speech signal may supplement lipreading by providing certain prosodic information, increasing perceptual strategies available to profoundly deaf persons.
An investigation of the transmission of suprasegmental features of speech using the multipoint electrotactile speech aid (MESA)
Authors: Bourgeois Michelle S. , Sparks David W. , Kuhl Patricia K.
Published: October 2005

Previous investigations of the speech transmission capabilities of the MESA have shown that trained subjects can correctly perceive certain segmental features of speech when they are spoken in isolation. The present study was conducted in order to evaluate the perception of one of the components of sentential rhythm—syllable number. Sentence stimuli of one, three, five, and seven syllables in length were videotaped. Five normal-hearing adults were required to “tap out” the rhythm of the sentences on a Morse code key under three receptive conditions: (1) visual alone—lipreading, (2) tactile alone, and (3) tactile plus visual. When scored for the absolute number of correct syllables, the results demonstrate significant differences for the main effects of modality and for the length of utterance. Further analysis will examine the degree and direction of error for each modality and for length of utterance. A replication of this experiment is underway in which a limited frequency range on the MESA is delivered to the subject. Implications for the design of sensory aids which optimize the reception of suprasegmental features of speech will be discussed.
Speech training systems using the lateral shape of the vocal tract and an F1−F2 diagram for hearing-impaired children
Authors: Shigenaga M. , Sekiguchi Y.
Published: October 2005

Three speech training systems for hearing-impaired children were designed and constructed using a minicomputer and a microprocessor. The first system displays the lateral shape of the vocal tract for each vowel estimated from the speech sound. In this system, three storages are prepared. One of them can be used to store a reference shape which may be estimated from a teacher's voice or a prepared standard shape. A child can articulate seeing the reference shape and rearticulate comparing his own first articulation with the reference shape. The second system displays successively the lateral shapes for articulation of any phoneme sequences containing consonants. The lateral shapes for Japanese phoneme sequences are synthesized by a speech synthesizer using an articulatory model. The third system displays the first two formant frequencies extracted from speech as a spot on the F1−F2 plane, where the regions of vowels are also shown. Preliminary use of these devices has shown that they complement each other to form an effective system of speech training.
Performance prediction models for piezoelectric polymer hydrophones
Authors: Ricketts Donald
Published: October 2005

Mathematical models of several types of piezoelectric polymer hydrophones have been developed which facilitate an evaluation of their low-frequency electroacoustic sensitivity as well as maximum operating pressure (Pmax). The flexural rectangular plate and disk and also cylindrical hydrophone elements have been investigated for various boundary conditions, where in each case the polymer film is attached to a substrate which establishes the geometric configuration of the hydrophone. The theoretical results for flexural elements together with practical design considerations indicate that matching of the elastic moduli of the polymer and the selected substrate material yields better hydrophone sensitivity. But to maximize the product of sensitivity and Pmax without pressure compensation, the selection of the substrate material should be based upon maximizing the ratio of substrate mechanical strength to elastic modulus. Experimental verification of the theoretical models has been obtained for both the bender-types and cylindrical piezoelectric polymer hydrophones. Numerical results are presented in the form of polymer hydrophone design charts that illustrate their performance limits.
The high power-to-volume capability of polyvinylide fluoride (PVF2)
Authors: Linvill John G. , Terwilliger Curt , Rezvani Behruz , Dameron David
Published: October 2005

High fields (∼30 V/μm) applicable to PVF2 transducers without depolarization provide potentially higher mechanical power output per unit volume than from PZT transducers which depolarize at 1/100 of this field. For narrowband operation, the (mechanical power)/volume is proportional to d312, the stiffness, and the stiffness/damping ratio. Comparison of these quantities for PVF2 and PZT reveals that for equal volumes about five times as much power should be obtained from PVF2. A simple measurement transducer was made and tested. It consists of a sheet of 9-μm poled PVF2 about 4 cm long and 1 cm wide. Its upper conductor surface was etched, producing electrically separate terminals of 2 × 1 cm. The transducer is clamped at its short ends and placed under tension. A signal applied between one of the upper terminals and the lower ground causes the unenergized half to be stretched incrementally. Alternating signals on the transmitter half transmit mechanical power to the receiver half. The open-circuit voltage appearing on the receiver permits evaluation of its deflection and calculation of the power transmitted. This “tug-of-war” resonator, operating at 15 kHz, exhibited the transmitting half transmitting 160 mW/mm3 of power to the receiving half, in agreement with prediction.
Acoustic bandlimiting filters for digital conversion of signals
Authors: Flanagan J. L.
Published: October 2005

Analog-to-digital conversion is required for digital processing of signals in a number of areas of acoustics. Examples include voice communications, underwater sound, acoustic imaging, and room acoustics. For each such signal, analog bandlimitation (prior to sampling and quantizing) is necessary to prevent aliasing distortion. Typically, the bandlimitation is accomplished by electrical filtering. As the cost of integrated digital logic diminishes, the expense of electrical filtering becomes a significant part of the total conversion cost. Similar factors apply in the desampling operation, or digital-to-analog conversion. We report here a design technique for acoustic filters which can perform the requisite band limitation. The filters can be made an integral part of the transducer assembly, both for transmitter and receiver. We present theoretical designs for fourth- and sixth-order low-pass filters that are appropriate for voice digitization, and we evaluate these designs by computer analysis. Next, we show physical implementations of the filters, and assemble them with two types of electret microphones. Finally, in an anechoic environment, we make acoustic measurements on these devices and demonstrate the validity of the theoretical design.
Polymeric ultrasonic probe
Authors: DeReggi A. S. , Roth S. , Kenney J. , Edelman S. , Harris G.
Published: October 2005

A small piezoelectric polymer hydrophone has been made and tested. It can be used over a broad range of frequencies to probe the temporal and spatial variations of the sound field produced by transducers such as those used in medical ultrasonics. The hydrophone is bilaminar. Each layer consists of a poled central area having a 2-mm diameter defined by evaporated aluminum electrodes on each side of a 25-μm-thick sheet of polyvinylidene fluoride about 7 cm in diameter. The combined sheets are held taut in a metal hoop. Well-separated leads of evaporated aluminum to the outside of the hoop were applied after poling so that the polymer under the leads is inert. The response does not vary by more than a few decibels from 0.5 to 10 MHz. At about 1 MHz, the change in the received signal when the hydrophone is inserted between a source and receiver is less than 0.2 dB. The broadband response of the hydrophone makes it useful for determining the output pattern of diagnostic transducers which typically use pulses. Its transparency to ultrasound minimizes formation of standing waves between it and therapeutic transducers which are typically continuous wave. [Work has been sponsored by the Bureau of Radiological Health.]