bluuebunny/crossref_metadata_2025 huggingface dataset:
Model for F generation based on multispeaker observations
Authors: Choppy Christine , Liénard Jean-Sylvain
Published: October 2005

This work is completed for a text to speech device for French that is not reproducing someone's voice. The corpus is read by six speakers (three male, three female) and analyzed with a spectrograph. The F0 values are normalized so that the variation range is the same for all speakers, and the F0 curves are plotted versus the phonemes (not versus time as usually is): This provides us with comparable curves and insures that one deals only with F0. The F0 variations that are common to all speakers are considered as representative of a normative intonation. Our first step is to reproduce this normative intonation. The model is generating it automatically out of the written test and is not based on a standard syntactic analysis. Instead, the parsing of the test into word groups is using a list of frequent function words (which is an account for syntax) and word length considerations. The F0 contour is obtained out of a combination of a sentence contour and word group contours, and modification induced by punctuations. The generated contour has the same main variations as natural contours. Comparison tests indicate a preference of 67% to speech synthesis with intonation (as opposed to without intonation). The results are expected to fare better with a new device giving a better voice quality.
Prosodic features connected to punctuations: Some cases with commas
Authors: Choppy Christine
Published: October 2005

In text to speech systems for French, the usual rule for commas (Cr) is that they are preceded by a high continuation rise, a lengthening of the last vowel, and marked by a pause. This work is investigating different possible types of “prosodic commas.” Besides (Cr), there exists another type of commas (Cf) which are expressed by a F0 fall. (Cf) are found where phrases or part of phrases are either pre- or postponed (this configuration appears mostly in spoken language, e.g., “She is in her office, I think”), but they are not found at interpolated clauses. Instead interpolated clauses signal themselves by a lower F0 value on the average. To provide us with a feeling for how much these commas express the subjects writing competence, the following procedure is set up: In a first session, the text is presented without commas, subjects have to insert them where they think they are needed, and read the text. In a second session they read the standard punctuated text.
The Theory and Efficient Calculation of the Probability of Coincidence
Authors: Neal Edgar H.
Published: October 2005

A problem of interest in the calculation of system performance is the probability of coincidence. For example, the probability that two nonsynchronous scanning beamed systems look at each other. Or, that an agile frequency source and an agile frequency receiver occupy the same frequency band coincidentally. Or, various combinations of periodically or randomly occurring window functions in time, frequency, or space. To efficiently calculate the probability of coincidence for the multiple window function case requires —for the periodically occurring case—reliance on the structure of modulo algebra. Otherwise, the computer storage and/or arithmetic operations becomes prohibitively large for many realistic system parameters. This paper will describe the basic theory of the calculation of the probability of coincidence, the modulo algebra required to do the calculation efficiently, and will present specific results.
A time/speaker normalization technique for word verification
Authors: Heisey D. L. , Kau C. K. , Li K. P.
Published: October 2005

The present approach for solving the word verification problem involves matching an input template with a known word reference template, The multidimensional template consists of formant values over the duration of the utterance. Due to speaking rate and idiosyncratic variations of speakers, both temporal and spectral normalizations are required. The time normalization technique employs a piecewise linear time warping function to map the unnormalized utterance into the normalized reference space. Pivot points used to align the input data template with the reference template are located in the utterance using maximum a posteriori (MAP) estimation. Statistics required to define the multidimensional density function for the estimator are obtained by training on known pivot point locations. Frequency normalization is then performed where the input template is linearly scaled (with a gain and bias) to minimize the weighted mean-square error between the scaled input template and the reference template. This error in addition to the corresponding gain and bias terms are used in a metric for word verification.
Bilateral asymmetry in lateral suppression
Authors: McFadden D. , Pasanen E. G. , Kearney J. K.
Published: October 2005

A forward-masking paradigm was used to determine masking patterns for a tonal masker both with and without a second, suppressing tone. The masker was 2000 Hz at 50 dB SPL, the suppressor was 2270 Hz at 70 dB SPL, and the signal was one of eight frequencies in the range from 1825 to 2200 Hz. The duration of the masker and suppressor (when present) was 200 ms and that of the signal 20 ms. Both were gated with a rise-decay time of 10 ms, and the signal followed the masker immediately. The procedure was single-interval forced choice with feedback. Complete sets of data were collected for both the left and right ears in a counterbalanced series. The masking patterns determined for the masker alone and for the suppressor alone are reasonably similar across and within subjects. Those determined with masker plus suppressor, however, not only differ markedly across subjects, they are often quite different for the two ears of the same subject. This asymmetry will be discussed in relation to the various proposed explanations of lateral suppression. [Work supported by a Research Grant from NINCDS.]
Effect of orthotropy on the sound transmission through plywood panels
Authors: Ordubadi Afarin , Lyon Richard H.
Published: October 2005

The peak in sound transmission through a single isotropic panel at the critical frequency causes a drop in transmission loss (TL) of that panel and of composite walls constructed with such a panel as face sheets. In investigations on a composite wall with plywood face sheets and styrofoam core, such a coincidence dip was not observed. The aim of this study is to investigate the disappearance of coincidence notch and the probable causes for such phenomena. The bending stiffness of a single ply of plywood changes gradually from the softest direction to the stiffest direction. Therefore, the bending wave phase velocity and sound transmissibility change with direction. It is hypothesized that for a group of sound waves randomly incident on the panel, this change in transmissibility eliminates or lowers the peak and, therefore, the coincidence notch smoothens. A plywood panel consists of plies laminated with their grains in different directions. These laminations of the plies may reduce orthotropy in thick plywood panels. A comparison of theoretically predicted and experimentally observed TL of plywood sheets supports the hypothesis that anisotropy is the reason for a lack of coincidence transmission for some plywood sheets.
Novel features of elastic plate response under the combined influence of fluid loading and compliant coating
Authors: Crighton D. G.
Published: October 2005

A thin elastic plate has a vacuum on one side, and the other is coated with a compliant coating and exposed to fluid loading. The plate is driven by concentrated excitation, and the acoustic and vibration fields are studied. Two novel features are apparent in the response; firstly, the “Lloyd's mirror” effect—according to which no energy-carrying acoustic field can propagate at grazing incidence over any surface of finite impedance—no longer operates in the presence of compliant coating. Secondly, the coupled system can, if the coating is sufficiently highly compliant, sustain a subsonic free surface wave of low phase speed, in addition to the subsonic wave with wave number close to kp, the vacuum-free wave number. The amplitude of the low-speed wave is small in the case of concentrated surface loading: however, the wave may be well matched in wave number to the pressure fluctuations of a turbulent boundary layer, in which case it may lead to large-amplitude surface response.
Experimental comparison of parameter interpolation in LPC vocoders
Authors: Tohkura Y.
Published: October 2005

LPC parameter interpolation is effective in smoothing synthesized speech roughness. To choose an appropriate interpolation parameter, LPC parameter interpolation properties were compared experimentally, using: (1) spectral distortion between interpolated spectrum and actual spectrum, and (2) spectral distance relationships among interpolated spectrum and two boundary spectra. Both k parameter and LAR (log area ratio), which are used as transmission parameters because of their good quantization properties, sometimes provide an undesired speech spectrum interpolation. These two parameters cannot interpolate formants correctly, in particular, when the formant bandwidth shows a big change between the two boundary spectra. On the other hand, α parameter provides better speech spectrum interpolation than the former two. Listening tests show that unnatural formant transition due to the k-parameter interpolation can be perceived as slurring and gliding, which are hardly perceived in the α-parameter interpolation.
Digital encoding of speech based on properties of the auditory system
Authors: Krasner Michael A.
Published: October 2005

This paper describes the development of a digital encoding system for speech. The system is designed to minimize the audibility of the encoding error by considering the limitations of the auditory system. By dynamically shaping the error spectrum as a function of the input signal, the error is inaudible. System parameters to ensure masking of the error by the signal are determined from experiments and from previous results taken from the literature. The input is filtered into 19 frequency channels each of approximately critical bandwidth. Each channel is independently quantized using block floating point PCM. Experiments show that the differential threshold of encoding degradation for 4 kHz bandwidth speech occurs at a rate of 34 kbps. [Work sponsored by Defense Advanced Research Projects Agency.]
A robust vocoder with pitch-adaptive spectral envelope estimation and an integraled maximum-likelihood pitch estimator
Authors: Paul Douglas B.
Published: October 2005

This paper describes a low bit-rate vocoder designed for improved speech reproduction quality and robustness. The vocoder includes a new algorithm, the Spectral Envelope Estimator, which forms the nucleus of the spectral analyzer. In addition to estimating the speech spectrum, the spectral analyzer also allows determination of a continuous estimate of the background noise spectrum which is used for noise suppression. A maximum-likelihood pitch estimator, which shares the signal processing of the spectral envelope estimator, has been integrated into the vocoder to yield accurate pitch estimates of noisy speech. This system yields high-quality speech reproduction at bit rates of 2.4 and 8.8 kbps. [Work sponsored by Defense Advanced Research Projects Agency.]
Experiments in channel equalization for telephone input to LPC systems
Authors: Davis S. B. , Markel J. D.
Published: October 2005

Successful narrow-band speech compression is based upon the use of a linear speech production model. With speech recorded in a quiet room or anechoic chamber, very good synthesis quality has been demonstrated. However, if speech is processed after being passed through a telephone channel, typically moderate to poor synthesis quality is achieved. An equalization method has been developed to compensate for the telephone channel. This method uses an inverse filter which is based on the long-term average spectral characteristics of the incoming speech divided by stored reference characteristics of the speaker. If the speaker is unknown, a population-based reference is used. The degree of the compensation may be assessed by using a distance measure between the original and the compensated speech. The results indicate that it is possible to use a speaker's long-term spectral characteristic to compensate for spectral distortion introduced by the telephone channel when neither the input or the channel is known.
Speech enhancement using a soft-decision maximum likelihood noise suppression filter
Authors: McAulay Robert J. , Malpass Marilyn L.
Published: October 2005

One way of enhancing speech in acoustic noise environments is to process the data using a noise suppression filter which performs a spectral decomposition of a frame of noisy speech and attenuates a particular spectral line depending upon how much the measured speech plus noise power exceeds an estimate of the background noise. In this paper a two-state model (speech absent or speech present) is used in the derivation of a maximum likelihood estimator for the speech power. The result is a suppression curve that is similar to existing algorithms but which is weighted by the a posteriori probability that the current measurement corresponds to the speech state. A class of curves is obtained by varying the value of a suppression factor which can be chosen to trade off noise suppression against speech distortion. The algorithm has been implemented in real time in the time domain, exploiting the structure of the channel vocoder to perform the spectral decomposition. Extensive testing has shown that the noise can be made imperceptible by proper choice of the suppression factor. A tape recording will be played demonstrating the noise suppression properties of the filter and demonstrating how the algorithm can be used in tandem with a narrow-band vocoder, such as LPC, to enhance noisy speech. [Work sponsored by Department of the Air Force.]
Processing of speech by the auditory nervous system (an overview)
Authors: Kiang N. Y. S.
Published: October 2005

Single-unit and gross potential responses to speech can now be recorded at virtually any level of the auditory nervous system desired. It may be useful at this time to review what is known, what is foreseeable in the immediate future, and what appears to be feasible with the present state of technology. This presentation will focus on conceptual and methodological issues. The time seems appropriate for speech scientists in general, and certainly phoneticians in particular, to participate in the analysis of how the brain processes complex acoustic signals such as speech. [Supported in part by NIH grants.]
Noise attenuation in power plant control valves—past, present, and future
Authors: Hammit D. E.
Published: October 2005

Noise measurement and control in power plant valves have seen significant advances in the past several years. Aerodynamic noise in control valves is now predictable within a fairly wide range of variables in fluids, while hydrodynamic noise (principally from cavitation) prediction is still somewhat limited in application. A great variety of proprietary and nonproprietary means of noise control for valving now exists. The evolution of noise controls and devices has been from all segments of the using, supplying, and consulting industries and significant further advances are possible. This is particularly true in the realm of high recovery valves, emergency vent valves, and cavitation prediction and control devices. Valve suppliers are detecting a higher level of sophistication among their users with a pronounced tendency to specify only that degree of noise control necessary for a particular application. Examination of several case histories indicates the extremely high cost penalties inherent in overspecifying acceptable levels of control valve noise.
Gas turbine noise control
Authors:   Louis A. Challis and Associates Pty. Ltd.
Published: October 2005

The use of gas turbine powered generators and pumping stations are likely to increase over the next two decades. Alternative fuel systems utilizing fluidized coal beds are likely in the near future, and direct combustion of pulverized coal is also a possibility. The primary problem of generally unacceptable noise levels from gas turbine powered equipment affects both community noise and hearing conservation alike. The noise criteria of such plant remain a significant design factor. The paper looks at the technical and historical aspects associated with the noise generation process and examines past, present, and possible future approaches to the problem of silencing gas turbine units; adequately specifying the acoustical criteria and ratings; evaluates the techniques by which these criteria should be measured; and correlates these with the typical results achieved in the field.
Gas turbine power plant noise source identification using contours
Authors: Cummins J. R.
Published: October 2005

In order to apply engineering controls to a noise source it must first be identified. This can be a problem in the power plant environment where a source may dominate in one band and not in others. The problem is complicated by the directivity of the various sources such as fans, ducts, and machinery. A technique of generating sound level contours using relatively large numbers of measurement locations has been developed and aids in source identification. Using a high speed computer to display the sound levels in each frequency band allows the engineer to pinpoint the dominant sources and their frequency. The method is suitable for octave, third-octave or narrower bands; several examples are given. Extensions including automation using computer controlled graphics are also discussed.
Sound propagation in coupled interior industrial spaces
Authors: Timmerman N. S.
Published: October 2005

A generalized approach to predicting the sound-pressure level in interior industrial spaces, such as power plant turbine buildings, is presented. Complex spaces are subdivided into coupled rectilinear subspaces, which are analyzed in terms of direct or diffracted and reverberant fields. Details of the analysis, such as choice of equations for effective source-power level, barrier insertion loss, and room constant, are investigated in detail, and recommended choices presented for a few specific cases. Limiting cases of the generalized analysis are noted and results compared with standard theory and other authors' experimental results.
The single element unidirectional microphone
Authors: Bauer B. B.
Published: October 2005

The single element unidirectional microphone exhibiting the Limaçon directional pattern m + (1 − m) cos θ was an accidental discovery during an effort to develop a bidirectional piezoelectric microphone, affirming the research director's lament that the best inventions are those leading into unexpected directions. Refined through equivalent circuit analysis and embodied in many configurations this principle over the years has been employed in many important and interesting microphone applications in Public Address, Broadcasting, Recording, and others. The theory and some of the most important configurations and applications of this principle are described.
Reciprocity and the calibration of microphones; its history and new problems
Authors: Cook Richard K.
Published: October 2005

A reciprocal theorem is the basis of the reciprocity method for the absolute calibration of microphones and other electroacoustical instruments. Such theorems are commonly found in the mechanics and acoustics of gases, liquids. and solids, in electroacoustical systems, in electric circuits, and in the propagation of electromagnetic waves. The earlier invention of the condenser microphone facilitated the invention of the reciprocity method in 1940. It was then developed and eventually standardized as a very accurate and convenient method for measuring the response of a microphone in both amplitude and phase. The first applications were to the sound pressure calibration of microphones used at audio frequencies in air. Earlier calibration methods were based essentially on standard sources such as the thermophone, pistonphone, and electrostatic actuator, and on the use of a Rayleigh disk. Systematic differences between these led to a search for a more accurate method, and discovery of the reciprocity method. During the war years (1940–45) it was applied to the calibration of underwater hydrophones, and shortly thereafter to the calibration of accelerometers. The extension to freefield calibrations led to problems still present today, and for which we offer a solution.
The quantitative significance of images reconstructed by acoustical holography
Authors: Strasberg M. , Maidanik G.
Published: October 2005

Acoustical holography is being considered for quantitative studies of acoustic source distributions. We examine the quantitative significance of images reconstructed using the holographic technique described by R. L. Cohen in Pennsylvania State University Applied Research Laboratory Tech. Memo. 76-05, Jan 1976, for sources distributed over a plane surface. The image is formed by measuring the magnitude and phase of the sound pressure due to the sources on a parallel hologram plane and then calculating values of a Helmholtz integral of these pressures for field points located on the original source plane. Values of this integral form the reconstructed image. If the source and hologram planes are many wavelengths apart, this image represents a pressure distribution over the source plane which is nearly identical to the radiating, low wavenumber portion of the original pressure distribution on the source plane. The original pressure distribution may contain nonradiating high wavenumber components which are attenuated in the reconstructed image. To provide insight for understanding reconstructed images, the radiating portion of the pressure distribution is exhibited for point sources and for a finite circular membrane vibrating in an infinite rigid baffle. The radiating portions of these distributions differ considerably from the total pressure distribution for these sources.
Simultaneous measurements of acoustic pressure and particle velocity by an optical holographic method
Authors: Clark J. A.
Published: October 2005

An extension of an optical, holographic interferometery method for the measurement of instantaneous acoustic pressure previously reported [J. A. Clark, Holographic Visualization of Acoustic Fields, J. Sound Vib. 56, (1978)] is described. Optical interference fringe patterns are photographed which determine the acoustic condensation or relative density variations associated with a propagating sound wave. If the usual linear equation of state for a liquid is assumed, these same fringe patterns determine the acoustic pressured distribution, but with a different scale factor. Since optical data is available over an extended region of space for a set of different times, it is possible to also determine other independent measures of the acoustic field, simultaneously with the pressure measurements, by spatially or temporally differentiating or integrating the data. A procedure for numerically obtaining the components of particle acceleration by spatial differentiation (assuming the equation of motion) and for then obtaining the components of particle velocity by temporal integration is demonstrated. A second procedure for obtaining pressure and velocity measurements without assuming a linear equation of state is also discussed. This latter approach could be useful in experimental investigations of some nonlinear effects associated with waterborne sound and shock waves. [Research supported by ONR.]
The acoustic properties of coal
Authors: Venne H. Joseph
Published: October 2005

A survey of recent literature reveals sizable discrepancies in quoted values for the wave speed in coal; quoted values range from 1.2 × 103 m/s to 2.3 × 103 m/s. Negligible data exist for the values of the attenuation in coal. Present paper discusses design and execution of experiments (still in progress) to determine these properties. A coal slab is immersed in water between matched transducers. Both emit pulses, but their relative phases and amplitudes are externally controlled. The amplitude and phase of the right transducer is adjusted until the received pulse at the left transducer is a null. The received pulse consists of sound from the right transducer transmitted through the slab, plus sound from the left transducer reflected from the slab. The technique is a modification of that developed by McSkimin [J. Acoust. Soc. Am. 23, 429–434 (1951)]. The gain factor G times exp (iφ), where φ is relative phases of the two incident pulses, is shown to be a transcendental function of the complex wavenumber. Simultaneous graphical solution of the real and imaginary parts of this equation yields the sound speed and the attenuation. Results obtained using this technique are discussed. [Work supported by NASA.]
Measurement of the coupling force between foam and fluid
Authors: Jameson P. W.
Published: October 2005

The coupling impedance between porous foam and liquid resulting from their relative motion has been measured in an apparatus which excites torsional waves in an annulus of liquid-saturated foam. The apparatus consists of a foam slab sample cemented around a thin metal cylinder which in turn is mounted on a torsional rod of variable length. The resonant frequency and damping rate of torsional oscillations of the system when the cylinder is in air and when it is immersed in fluid are used to determine the shear loading on the cylinder from the foam for the two cases. For the low viscosity fluids tested, the coupling impedance determined from the experimental results was largely viscous with a small springlike component. The fact that the impedance was springlike rather than inertial suggests that bending of the small foam fibers may have been an important effect. [Work supported by NUSC.]
Interactive effects of electrocutaneous and acoustic stimulation of the middle-ear muscle reflex
Authors: Ruth R. A. , Tucci D. L. , Schoeny Z. G.
Published: October 2005

The present study is aimed at further defining the potentiating interaction between electrocutaneous and acoustic stimulation of the middle-ear muscle response. A previous study by Ruth et al. (1978) investigated the temporal properties of this facilitory process by varying the location in time of a subreflex threshold electrocutaneous stimulus with respect to an acoustic. As predicted the greatest amount of potentiation was found to occur when the two signals were simultaneous and overlapping. This present study was designed to evaluate the effect of varying levels of subreflex threshold electrocutaneous stimulation on the magnitude and persistance of the facilitated response. Results indicate that amplitude of the acoustic reflex response increases as the level of the electrocutaneous stimulation is increased from threshold of sensation up to threshold of discomfort. Evidence of acoustic reflex sensitization was observed as long as ten seconds following an electrocutaneous stimulation. Possible neural substrates for these interactions will be discussed.
Utilization of the RKU equations to design constrained layer damping treatments
Authors: Knighton Donna L. , Jones David I. G.
Published: October 2005

Constrained layer damping treatments, consisting of alternate layers of viscoelastic and elastic materials, are being widely used in industry for vibration and noise control. Many efforts have been made over the past two decades to apply diverse mathematical techniques to the analysis of the dynamic response of various structures with such treatments added, either on the surface, or incorporated into the fabrication of the structure. It remains true, however, that the now classical equations of Ross, Kerwin, and Ungar (RKU) are the simplest, most readily used, and often the most reliable approach toward making such predictions, even in spite of the fact that the equations apply only to beams or plates with pinned edges. While these equations are strictly limited to this type of boundary conditions other classical boundary conditions have commonly been accounted for simply by defining an appropriate modal semiwavelength in terms of the eigenvalue and the actual length. This seems to work quite well in most cases, but breaks down quite badly for the fundamental mode of a cantilever beam. This is particularly unfortunate, since cantilever beams are becoming very widely used for measuring the damping properties of viscoelastic materials. The aim of this paper will be to (i) discuss the derivation of equivalent modal wavelengths for nonpinned boundary conditions; (ii) describe the reduction of data from tests on cantilever and clamped-clamped beams with multiple layer damping treatments added in order to determine equivalent free layer properties; (iii) use the equivalent free-layer approach and the RKU approach to predict modal damping of some particular structures, making use of given complex modulus properties of the materials selected for the viscoelastic layers, and compare the two analyses with measured results; and (iv) discuss a computer program to optimize the geometry of constrained layer treatments for beams of various cross sections and for plates.
Testing linguistic stress rules with listeners' perceptions
Authors: Lea Wayne A.
Published: October 2005

Stress patterns provide information about the wording, phrasal divisions, syntactic categories, and grammatical relations in English sentences. This study attempts to experimentally verify alternative stress rules published by linguists like Chomsky, Halle, Bresnan, Lakoff, and Bolinger. Consistency and rule testing utility were studied for stress perceptions assigned with either three levels (stressed, unstressed, or reduced), a continuum of alternative values, or pairwise comparisons of relative stress levels. Preliminary results from averaging five listeners' three-level assignments indicated that: some expected nuclear (i.e., rising) stress patterns were perceived as falling stress patterns; reduced stress levels occur in subordinate phrases and on repetitive words in coordinate constructions; and sequences of prenominal adjectives have falling stress patterns like compound constructions. These and other results are examined with the other perception techniques, which more directly verify relative stress levels. [Work supported in part by NSF and ARPA.]
Psychoacoustical correlates of phonetic features in Spanish sounds
Authors: Guirao M.
Published: October 2005

This work attempts to describe a correspondence between acoustical signals, psychoacoustical dimension, and phonetic features. The speech material consisted of the 22 Spanish phonemes, as they are pronounced in South American countries. Seventeen consonants (followed by /a/) and the five Spanish vowels were first analyzed acoustically to determine their spectral characteristics and then grouped into three classes: periodic sounds, bands of noise, and bursts of noise. The same array of phonemes served to perform a perceptual test with the participation of 32 subjects. The subjects were instructed to assign numerical values to the sounds according to their apparent levels of intensity (low-soft), volume (large-small), density (compact-diffuse) and tonal quality (acute-grave). The phoneme /p/ in /pa/, called 10, was used as a standard. Participants also divided the sample into continuant and interrupted sounds. Preliminary results suggest that attributes of auditory sensation appear to be relevant in the differentiation of phonetic units.
Developmental differences in labeling VOT continua with varied fundamental frequency
Authors: Bernstein Lynne E.
Published: October 2005

Adults have been shown to use fundamental frequency in classifying the voicedness of phonemes [M. Haggard et el., J. Acoust. Soc. Am. 47, 613–617 (1970); D. Massaro and M. Cohen, J. Acoust. Soc. Am. 60, 704–717 (1976)]. The F0 cue has an additive effect for adults. As F0 increases, more stimuli are labeled as unvoiced. This study compares the relative effects of voice onset time and F0 cues in two types of labeling tasks performed by adults and children. It is hypothesized that the use of secondary cues such as F0 lags the VOT cue developmentally. VOT continua were synthesized from /gate/ to /Kate/ using values from +5 to +70 ms and three F0's (105, 135, and 165 Hz). In condition I adults and children (ages 3 and 7 years) are respectively instructed to write or say the word they hear. Condition II is a 2 AFC labeling task. Condition I allow insight into how subjects make choices in condition II. Adult's data indicate a significant effect for F0 in both conditions. In condition I, adults make place errors on unvoiced ends of continua. Data from children are being collected and a comparison will be made between their performance and adult's for both conditions.
Grinning from ear-to-ear: The perception of smiled speech
Authors: Tartter V. C.
Published: October 2005

The perceptual and acoustic effects of pronouncing speech with lips spread in a smile were assessed. Smiled and straight-faced tokens of 25 nonsense syllables and four meaningful sentences were collected from three male and three female English speakers. Three groups of 12 subjects distinguished the smiled token from its straight-faced counterpart in forced-choice identification tasks: Group 1 was instructed to choose the smiled sound, Group 2 the happier, and Group 3 the säddder. Groups 1 and 2 selected the smiled utterance significantly more often than chance for all six speakers. Group 3 chose the straight-faced utterance as consistently, for four speakers, but for one speaker reliably picked the smiled one. The results suggested that particular cue combinations may be identified as smiling specifically, whereas others may contribute to perception of emotionally in general. Acoustic analyses showed that smiling raised the second and third formant frequencies for all speakers, and fundamental frequency, amplitude, and/or duration for some. Duration and frequency both increased substantially during smiling only for the speaker whose smiled utterances were also judged as sad.
Auditory memory and vowel anchoring in CVC contexts
Authors: Sawusch J. R. , Nusbaum H. C. , Schwab E. C.
Published: October 2005

Previous vowel anchoring experiments have shown some evidence of multiple processes in the contrast effects found for vowels. In the present experiments vowels were embedded in CVC context ([CiC] to [CIC]) and presented to subjects in ABX format as well as standard identification in both equiprobable control and anchoring conditions. For the [CIC] end of our continua, the degree of change in category boundaries due to anchoring was found to vary with the degree to which the stimuli were perceived categories (from the ABX data). However, for the [CiC] end of our continua, the degree of anchoring was nearly constant and independent of the consonantal context and the degree of categoricalness in the ABX data. These data offer further support for our interpretation of vowel anchoring and adaptation results in terms of two distinct perceptual processes with one of these tied changes in auditory memory. [Work supported by NSF and NIMH.]
Influence of task demands on the categorical versus continuous perception of vowels
Authors: Cowan N. , Morse P. A.
Published: October 2005

Various task constraints were imposed on subjects making same/different, AX discriminations within and between phonetic categories. The experiments employed four 50-ms, synthetic vowels from an/i/-/I/continuum and five interstimulus delays (0-2000 ms). In the first experiment, stimuli were presented with all trial types randomized together or with between- and within-category shifts in separate trialblocks. Between-category performance was superior with separate blocks, whereas within-category performance did not differ in the two procedures. A second experiment involved AX presentations to subjects who were or were not initially informed of the intended category labels. Surprisingly, category knowledge improved within-category discriminations. In a third experiment, subjects were required to identify both vowels in an AX pair. Between-category discrimination was higher, and within-category discrimination lower, with AX identification as compared to a same/different response. The results of these experiments suggest that a stable phonetic and an unstable auditory short-term memory both contribute to within- and between-category discrimination, with some labeling occurring on a comparative basis. Furthermore, task demands alter the relative contributions of auditory and phonetic memory.
Comparison of “adaptation” effects on consonantal and vocalic features
Authors: Godfrey John J.
Published: October 2005

A number of recent studies attribute selective adaptation effects in whole or in part to response contrast rather than to desensitization of “feature detectors” [J. S. Bryant, J. Exp. Psychol. 4, 610–620 (1978); R. L. Diehl, J. L. Elman, and S. B. McCusker, J. Exp. Psychol. 4, 599–609 (1978); H. J. Simon and M. Studdert-Kennedy, J. Acoust. Soc. Am. 64, 1338–1357 (1978)]. The hypothesized detectors were generally assumed to be available only for certain consonantal features; such phenomena as the susceptibility of vowel identity judgements to contextual effects were thus considered to be unrelated. Selective adaptation experiments were performed using acoustic continua spanning the ranges from steady-state [ε] to [εl] and from [εl] to [εd]. Endpoint adaptors in both experiments produced significant shifts in the predicted directions. The results fit well with a response-contrast interpretation, and argue against a strict perceptual dichotomy based on consonant/vowel classification alone.
Anchoring and selective adaptation may affect different processing mechanisms
Authors: Nusbaum H. C.
Published: October 2005

Fifteen synthetic CV syllables varying perceptually from [di] through [ti] to [si] were used in an identification task. In the control condition each stimulus occurred equally often. In the anchoring conditions, one of the phonetic exemplars occurred more often than the other stimuli. For the [di] and [ti] anchor groups no consistent changes in identification were found. The [si] anchor group showed a significant shift of the [ti]-[si] category boundary toward the [si] endpoint. The [di]-[ti] boundary showed no systematic change after anchoring. A qualitatively different pattern of results were obtained using a similar set of stimuli in a selective adaptation paradigm [W. A. Ainsworth, Percept. Psychophys. 21, 365–370 (1977)]. Both [di] and [ti] adaptors caused shifts of both category boundaries in the direction of the adaptor while [si] had no systematic effects. It is possible that selective adaptation may affect channels of analysis sensitive to relative timing information. The effects of anchoring may be attributed to the differential availability of auditory memory for stop consonants and fricatives. [Work supported by NSF and NIMH.]
Frequency compression effects on consonant intelligibility in the normal ear
Authors: Yonovitz A. , Thompson C. L. , Harper D.
Published: October 2005

The intelligibility of frequency-compressed speech was measured for forty-eight normal-hearing subjects. The speech stimuli consisted of twenty-one initial CVs (with /a/) recorded by a male speaker. Six conditions of frequency compression were generated: 0%, 10%, 20%, 30%, 40%, and 50%, with a General Electric TCS 3-5195A time compression system set to modify only the frequency of the stimuli. At each condition eight subjects wrote their response to 30 random presentations of each of the 21 CVs. Confusion matricies were constructed from these data. The data were analyzed in terms of consonant confusions and distinctive feature transmission. The functions relating feature transmission and frequency compression indicated only minor degradation up to 20% compression; however, at greater than 20% compression a dramatic decrease in information transmission occurred with differing patterns of feature transmission at each compression condition.
Development and connectivity of human auditory cortex
Authors: Daly D. M. , Daly D. D. , Wada J. A. , Drane J. W.
Published: October 2005

Using sets of synthetic, sparse acoustic stimuli (SAS) we have examined two patients who underwent unilateral surgical excision of all auditory cortex. Patient 1, an 8-year-old girl, had right hemispherectomy at age 5 years. Patient 2, a 30-year-old woman, had suffered extensive left hemisphere damage at birth and had left hemispherectomy at age 29 years; preoperative carotid amytal studies showed right hemisphere dominance for speech. Patients classified sets of SAS presented monaurally; responses were arranged in multidimensional contingency tables and analyzed for departure from homogeneity. For sets with falling F2 (|γε|-|yε|). each patient classified appropriately SAS presented to ear opposite remaining hemisphere (p = 0.0001); however, each classified randomly SAS presented to ear opposite excision (p = 0.9997). Repeated testing demonstrated no learning effects. These findings permit following conclusions for these SAS: (1) Neither subcortical mechanisms nor ipsilateral input to remaining auditory cortex are sufficient to classify these SAS appropriately. (2) If “plasticity” operated in patient 2, it was not evident in altered auditory connectivity.
Threshold shifts in chinchillas exposed to low- and high-frequency noise in combination
Authors: Burdick Charles K. , Mozo Ben T. , Hargett C. E.
Published: October 2005

Previous studies [Burdick, et al., J. Acoust. Soc. Am. 64, (1978)] found that chinchillas exposed to octave bands of noise centered at 63 and 1000 Hz incur maximum hearing loss at the frequencies of 1400 and 2000 Hz. Since these bands of noise produce maximum hearing loss at the same frequencies, the question arose concerning the possibility of synergistic effects when exposed to both bands of noise simultaneously. Three groups of chinchillas were exposed to these noise bands for nine days. One group was exposed for four and one-half days to the low-frequency noise followed by four and one-half days of exposure to the high-frequency noise; another group was exposed to the same noise bands with the order of presentation reversed. The third group of animals was exposed to both noise bands simultaneously. The intensity level of each noise band was 94 dBA. No interactive effects were produced either by the successive exposures or by the simultaneous exposure.
Growth of TTS as a function of the duration of impact noises
Authors: Hétu Raymond , Trémbolières Claude
Published: October 2005

The effect of the duration of impact noises was investigated to assess the importance of this variable in establishing hearing damage risk criteria. Fifteen normal listeners were exposed to repeated series of 60 impacts presented once every 2 s at increasing peak levels from 107 dB SPL up to an individual critical level (CL). The CL was defined as the peak level producing a TTS3 of 15 dB at one frequency. Six different impact noises were studied. In each case, their average spectrum showed a maximum amount of energy in the third-octave band centered at 3.15 kHz. The e duration (the time for the pressure envelope to decay to 8.7 dB of its initial peak level) ranged from 20 to 210 ms. Results indicated that amounts of TTS3 at 4, 6, or 8 kHz are not a linear function of the impact energy but rather an exponential one. Moreover, data analysis confirmed, for each impact duration, that the TTS grows exponentially with the peak-pressure level of the noise. It also showed that the rise time may not be a highly relevant variable for predicting TTS from impact noise.
Effects of auditory fatigue on 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 5-min exposure to an octave band of noise (1500–3000 Hz) at 115 dB. The stimulus word lists and the masking noise in which the words were embedded were filtered (2825- to 3185-Hz band). Masking-noise levels were varied between 0- and 90-dB SL Speech stimulus intensity was adjusted to obtain approximately 50% intelligibility before fatigue at each masking-noise level. Alternative hypotheses to account for the experimental observations were developed. A recruitment hypotheses suggested that masked-speech intelligibility would decrease following fatigue, whereas a signal-to-noise ratio hypothesis predicted post fatigue improvement of intelligibility. The results of several experiments failed to provide clear support for either hypothesis. [Work supported by USAF.]
Effects of low-frequency noise on performance and annoyance
Authors: Key K. F.
Published: October 2005

This study examined the effects of low-frequency noise on complex psychomotor task performance and annoyance. It was hypothesized that performance would be decremented in the noise conditions versus the control condition. It was also hypothesized that females would perform worse and would rate the noise more annoying than the males. High-frequency noise was predicted to be more annoying and decremental to performance than low frequency. Thirty females and 30 males served as subjects for both parts of the experiment. The first part consisted of performance on a complex psychomotor task, under one of the experimental conditions: (1) low-frequency noise. (2) high-frequency noise, or (3) ambient noise. Subjects were given approximately 50 min practice for the task. Two days later the subjects again performed the task for 50 min under the randomly assigned experimental condition. The second part consisted of annoyance ratings of noises of various frequencies obtained through the method of magnitude estimation. Results indicated a decrement in females' performance under the high-frequency noise condition. The predicted relationship for annoyance ratings was found. Results are discussed in terms of an arousal-performance model.
A quantitative model for deriving an age correction factor in noise-induced hearing loss
Authors: Corso John F.
Published: October 2005

This paper considers the factors of presbycusis and noise exposure in relation to permanent hearing loss and provides a quantitative model from which an age correction factor in occupational hearing loss may be derived. While extant physiological and audiometric data provide a justifiable rationale for a correction factor [J. F. Corso, in Henderson et el., Effects of noise on hearing (Raven, New York 1976), pp. 497–524], the conventional additivity hypothesis is deemed inappropriate. From selected empirical functions a quantitative model is derived for assessing the age component of occupational hearing loss. The model is based on the concept of a variable ratio for partitioning the effects of presbycusis and noise exposure. Consideration is given to the application of the model in medicolegal cases of occupational hearing loss in which the degree of hearing impairment must be assessed.
Predicting hearing loss from the acoustic reflex
Authors: Rizzo Stephen , Greenberg Herbert J.
Published: October 2005

This study explored an alternative procedure for predicting hearing level from the acoustic reflex threshold. The acoustic reflex threshold (ART) was measured for pure-tone and noise-band signals on 158 subjects. A series of regression equations were developed for predicting the pure-tone average (PTA) from high-pass noise and 500 Hz reflex thresholds. A positive linear relationship (R = 0.83) existed between the PTA and the high-pass noise and 500 Hz reflex thresholds. The Rz value (0.69) revealed that this regression equation can account for 69% of the subject variability. The experimental design of this study allowed the development of a series of regression equations to predict hearing sensitivity from the acoustic reflex. The equations developed allow the prediction of the three and four-frequency PTA. Also, a high frequency decrease in hearing sensitivity at 4000 Hz can be determined.