bluuebunny/crossref_metadata_2025 huggingface dataset:
Airport noise and land use planning
Authors: Yilmaz Sevtap
Published: October 2005

In this study, the aircraft noise near airports at a range of 1-mile propagation has been investigated and the corresponding sound pressure level has been computed. Due to the meteorological conditions, the acoustic impedance of ground and various wave propagation phenomena, the attenuation of sound level propagated from source to receiver, has been calculated [G. J. Ruijgrok, J. Sound Vib. 72(4), 469–479 (1980)]. Also, during the temperature lapse and inversion, the boundaries of upwind zone and downwind zone (acoustic shadow and light zones) have been determined [F. M. Wiener and D. N. Keast, J. Acoust. Soc. Am. 31, 724–733 (1959)]. Accordingly, the residential areas have been established within the boundary of shadow zone in which sound is not absorbed, while the light zone has been used for certain purposes. Further, the heights of residential buildings in the shadow zone have been found according to the rays of sound waves which propagate respectively upward and downward during the temperature lapse and inversion; this will aid architects in design layout and in decision concerning the exterior construction of buildings. [Work supported by the Ministry of Public Works.]
A phonetic band?
Authors: Remez Robert E. , Rubin Philip E.
Published: October 2005

Explanations of phonetic perception describe the conversion of the acoustic signal to phonetic segments. By making the descriptions of phonetically relevant patterns abstract, these explanations have accomodated the great variability evident in the acoustic signals produced by different talkers and in different utterances by the same talker. In order to refine this characterization of perception, we sought to determine the precise limits of phonetically useful information over frequency variation. A frequency transposition test was performed with three sentences, employing the technique of sinusoidal replication of natural utterances. This technique emphasized time-varying acoustic information, and also controlled for timbre changes over frequency. The results show that a phonetic listening band may exist within the human range of audible frequencies. The relation of this finding to frequency coding, perception of musical pitch, and development of sensory prostheses will be discussed. [Supported by the National Institute of Child Health and Human Development.]
Effect of pressure on sound absorption in seawater and in aqueous solutions of MgSO4
Authors: Hsu C. C. , Fisher F. H.
Published: October 2005

Measurements have been made at 25 °C of the effect of pressure on sound absorption in Lyman and Fleming synthetic seawater from 30–300 kHz up to 307 atm. Similar measurements were made in 0.02M MgSO4 solution and in a 0.02M MgSO4-0.6M NaCl solution. The decrease with pressure of absorption per wavelenght (αλ) in seawater differs substantially from those reported by other investigators. The relaxation frequency, within experimental error, was found to be independent of pressure. For seawater, increasing pressure to 307 atm reduced absorption by 26.3%, very nearly the same (28.2%) as was observed previously for 0.5M MgSO4 solutions. The addition of 0.6M NaCl to the 0.02M MgSO4 solution decreased the absorption by a factor of 4 and the effect of pressure was found to be independent of the NaCl concentration. The Eigen and Tamm multistate dissociation parameters which best describe the pressure dependence of sound absorption and electrical conductance in a 0.02M MgSO4 solution predict a 21.4% decrease in absorption at 307 atm compared to the observed 19.6% decrease. However, these parameters yield an equilibrium constant which differs from conductance derived value by a factor of 3.
Does silence play a peripheral role in the perception of stops?
Authors: Pastore Richard E. , Szczesiul Rosemary , Rosenblum Lawrence , Schmuckler Mark
Published: October 2005

A number of recent studies have investigated the role played by silence in the perception of speech stimuli, demonstrating that the perception of a subsequent stop is dependent upon the duration of the silent interval. If silence is acting simply to separate the stop from relatively peripheral interference arising from the pre-silence components of the stimulus, then we would expect that a separation of the same stimulus components by differential lateralization would, in a similar fashion, improve the perception of the stop consonant; in effect, lateralization would be traded for silence. We investigated this prediction using an /s-day/ natural speech continua. By selectively inverting the stimulus tokens, we created binaurual stimuli which were interaurally phase shifted by either 0 or π rad. The temporal boundary for the perception of /stay/ was equivalent for all four combinations of the binaural /s/ and /day/ tokens. These results, and the results with other natural speech tokens, will be discussed in terms of their implications concerning the role of peripheral versus central interactions among stimulus components. [Work supported by NSF Grant BNS 8003704.]
Psychoacoustic properties of imagined speech stimuli
Authors: Farrar Christopher D.
Published: October 2005

Synthetic consonant-vowel syllables were generated by factorially crossing four values each on two acoustic continua (VOT and F2-F3 transition onset frequency) to produce 16 stimuli identifiable as /ba/, /pa/, /ta/, and /da/. In addition, four silent stimuli were generated. The complete set of 20 stimuli was presented to subjects in a paired comparison test for scalar judgments of similarity. Silent stimuli were presented paired both with sounded stimuli and with other silent stimuli. On the response sheet sounded stimuli were indicated by an underscore and silent stimuli by “ba,” “pc,” “da,” or “ta,” depending upon which syllable the subject was supposed to “hear.” Subjects were instructed to make comparisons involving silent stimuli by imagining themselves pronouncing the silent syllable. Responses from 13 subjects were analyzed using the INDSCAL individual differences scaling approach [J. Carroll, in Multidimensional Scaling: Theory and Applications in the Behavioral Sciences, edited by Shepard, Romney, and Nerlove (1972)]. The recovered two-dimensional perceptual space clearly showed the four silent stimuli located appropriately with respect to the sounded stimuli, indicating that subjects were capable of making linguistic judgments of sounded and silent stimuli alike. These results provide direct support for prototype models of speech perception.
Contextual effects and speaker quality in vowel perception
Authors: Fox Robert Allen
Published: October 2005

Studies utilizing an anchoring paradigm have suggested that vowel identification can be explained on the basis of the acoustic nature of the vowel, the immediate phonetic context, and in terms of contrast with an “auditory ground” or adaptation level [J. R. Sawusch and H. C. Nusbaum, Percept. Psychophys. 25, 292–302 (1979); H. J. Simon and M. Studdert-Kennedy, J. Acoust. Soc. Am. 64, 1338–1357 (1978)]. A study was designed to investigate the nature of this auditory ground, in particular, to determine whether the relevant acoustic information is represented in a veridical fashion or has undergone specific types of auditory processing (e.g., speaker normalization). A series of experiments utilizing the anchoring paradigm was conducted. Ss identified synethetic vowels from a [hɪd]-[hεd] continuum under two conditions: an equiprobable control and an anchoring condition (with one continuum endpoint token occurring at least three times more often than any other). Three different stimulus continuua were constructed to represent vowels from a low F0 male, a high F0 male, and a high F0 female. Ss were run under each of 18 different control/anchor combinations. Contrast effects were found such that phoneme boundaries shifted toward the anchoring category and differential shifts of the boundaries were found as a function of both F0 and relative F-pattern height. Neither feature detector fatigue nor response bias can account for such results, and they suggest that information in the postulated auditory background has already undergone formant normalization and its characteristics are consistent with Massaro's synthesized auditory memory.
A method for treating statistically dependent inputs to an OR-ing device
Authors: Struzinski William A.
Published: October 2005

In previous studies (Dow et al., 1970; Nuttall, 1975, 1978, 1982; Struzinski, 1982), OR-ing data reduction models were developed for implementation in signal processing systems. All models assumed that the N input channels were statistically independent. In signal processing systems, statistical independence may or may not be a good assumption. Therefore this paper describes a technique to handle statistically dependent inputs to an OR-ing device. The technique is generic, as it can be implemented in all OR-ing models. An illustration is presented by computing the OR-ing loss employing a previously published model. Finally, a brief comparison is made between the OR-ing loss for dependent inputs and the OR-ing loss for independent inputs by noting the difference in the magnitude of the loss. It is concluded that the OR-ing loss decreases with an increase in the correlation coefficient. [Work supported by the Naval Sea Systems Command.]
Active damping of resonant ultrasonic transducers for robotic sensor applications
Authors: Miller G. L. , Boie R. A.
Published: October 2005

Ultrasonic ranging in air over short distances offers advantages for robotic applications. Such sensing methods can employ transducers small enough to fit into a robot gripper, thereby providing terminal homing capability. Any such short range time-of-flight systems, however, have to solve the problem of receiver recovery after the transmit pulse. In the present case the problem has been solved by a controlled damping arrangement. Following each transmit pulse essentially all of both the electrical and mechanical energy is rapidly extracted from the transducer, leaving it ready to receive the echo. In this way ranging is readily achievable down to about 1 in. which is adequate for robot gripper applications.
Short-latency auditory-evoked response thresholds in infants and adults
Authors: Klein Alan J.
Published: October 2005

Auditory-evoked potential thresholds were measured on a group of normal-hearing adults and on a group of healthy infants without any history of middle-ear disease. Acoustic stimuli were computer generated tone pips with frequencies of 500 Hz, 4 kHz, and 10 kHz. For 500-Hz stimuli, presentation rate was 40/s and bioelectric activity was filtered at 60–2000 Hz in order to optimally record wave V of the BSR. Mean evoked potential thresholds in adults are 10–15 dB higher than behavioral thresholds to the identical stimuli. Preliminary results from infants less than 8 months old suggest evoked response thresholds are similar to adults at high frequencies. Large intersubject variability at 500 Hz confound threshold comparisons at this time.
Evaluation of a microprocessor-based, 40-Hz ERP system
Authors: Lynn James M. , Lesner Sharon A. , Daddario Cynthia C. , Wood Lynn
Published: October 2005

Galambos et al. [Proc. Natl. Acad. Sci USA 78, 2643 (1981)] described a procedure for recording the auditory middle-latency response potentials which they named the 40-Hz event-related potential (40-Hz ERP). Galambos and Makeig [J. Acoust. Soc. Am. Suppl. 1 72, S54 (1982)] recently have suggested that since the 40-Hz ERP resembles a sine wave, cross-correlation techniques can be used to identify the presence of a response and should, therefore, yield accurate threshold estimations. Axionics Instruments has developed a microprocessor-based evoked potential system, called AUDIT, that estimates 40-Hz ERP with a similar procedure. Results of an investigation of the predicative accuracy of AUDIT will be reported. Estimates of the hearing sensitivity of 25 adult hearing-impaired subjects for 500-, 1000-, and 2000-Hz signals based on AUDIT data will be compared to behavioral pure-tone audiograms.
Human auditory cortex: A preliminary report on studies using positron emission tomography (PET)
Authors: Lauter Judith L. , Raichle Marcus E. , Herscovitch Peter
Published: October 2005

In recent years information has been gathered regarding the organization of auditory pathways in nonhuman subjects. However, because of the invasive nature of current techniques for studying the central nervous system, there is little direct measurement of central auditory processes in humans. Positron emission tomography (PET), used to monitor changes in cerebral blood flow and metabolism, offers a relatively noninvasive technique that may allow us to observe brain responses in humans to auditory and other sensory stimulation. We have studied a number of young normal volunteers, combining the PET technique with controlled auditory stimulation. Examination of PET images of cerebral blood flow before, during, and after stimulation (including monaural and binaural presentation of a variety of tones and noises) indicates that discrete, highly localized changes in cerebral blood flow can be evoked with such stimulation. A number of acoustical variables seem to influence these changes, including sound spectrum, rate and level of presentation, and ear of input.
Time course of suppression of spontaneous otoacoustic emissions
Authors: Ruggero Mario A. , Rich Nola C.
Published: October 2005

Few data are available on the time course of suppression, by externally introduced sounds, of spontaneous otoacoustic emissions (SOAEs), and none have been presented for humans. By means of extensive averaging in the time domain and digital filtering, we have obtained such data for the (synchronized) SOAE produced by the first author's right ear (7529 Hz, 10–16 dB SPL). Upon presentation of a tone pip, there is a latency (1.2 ms at high suppressor levels, longer at lower levels) before any change can be detected in SOAE level. Most of the suppression then develops over an interval lasting 1.5–2.0 ms, but this relatively fast onset is followed by a roughly exponential slope with a time constant longer than 4 ms. At low suppressor levels, the latency to the beginning of recovery from suppression may be similar to the latency to suppression onset; for higher suppressor levels, the latency may be several tens of ms and recovery may not be complete before 100 ms. Given the uncertainty that still exists on travel times in the human basilar membrane, our data cannot rule out the possibility that the latency to onset of suppression matches the travel time in the cochlea to and from the site of origin of the SOAE. [Work supported by NIH Grant NS-12125.]
Vibration analysis of machine tools
Authors: Brown David L.
Published: October 2005

This talk reviews the state-of-the-art vibration testing methods as applied to machine tools. Both current testing methods for determining forced and self-excited machine tools chatter conditions and several new multi-input, multi-output testing methods which are currently used at the University of Cincinnati will be reviewed.
Analytical and experimental evaluation of a new nonlinear viscoelastic active decoupler
Authors: Karbo Donald J. , Shtarkman Emile
Published: October 2005

A simple, one-degree-of-freedom active vibration isolation system employing a new, nonlinear, viscoelastic-hydraulic-pneumatic decoupler has been studied. Results of the computer simulation and laboratory measurements were in good agreement. In the region of the mass-compliance resonance, transmitted vibration was reduced more than 20 dB compared to the performance of the same system operated in the passive mode. Well below and above resonance, isolation in excess of 30 dB was achieved. Both swept single frequency and random noise inputs were used.
The temporal organization of speech and the process of spoken word recognition
Authors: Nooteboom S. G.
Published: October 2005

Timing patterns in speech show a number of regularities which have been established many times for many different languages, such as the shorter average duration and greater durational variability of unstressed as compared to stressed syllables, the decrease of syllable duration with increasing word length, anticipatory shortening of syllables in words and phrases, prepausal lengthening, and the decrease of word duration with increasing contextual redundancy. An attempt will be made to account for such regularities starting from the general assumption that speed and accuracy of articulatory movement in coherent phrases are continuously adapted to what the listener needs at each moment in time. Against this background important aspects of timing patterns in speech can be meaningfully related to properties of the process of spoken word recognition. Words are often recognized before they have been heard completely. The moment in time that a word is recognized depends on its lexical redundancy, frequency of occurrence, contextual redundancy, and speech quality. After initial recognition of informative words and at the end of phrases listeners may need extra time for confirmation and interpretation. Together, these aspects of the recognition process seem to determine when a speaker can economize on speed and/or accuracy of articulatory movement and when he cannot.
Isochrony in speech
Authors: Port Robert
Published: October 2005

The suggestion that speech in a number of languages is constrained by rules that provide for equal spacing of certain phonetic events has been around for a long time. In particular, English (and certain other Germanic languages) has been claimed to have equally spaced stressed syllables and Japanese has been claimed to have mora units that are of equal duration. Experimental tests of these hypotheses have not fully supported the claims as formulated. The key question is whether these isochronous units should be viewed as essentially perceptual phenomena that do not exist in the speech signal but rather in listener's perceptual constructs or whether the concepts can be redefined so as to be discoverable in the acoustics. I shall argue that for the case of Japanese, at least, the mora is a genuine timing unit in the acoustic signal (once it is properly defined) and thus that is must also be a timing unit of the speech production process. Some possible formal models for the temporal effects observed in Japanese will be discussed. [Funded in part by NIH.]
On the cyclical basis of speech production
Authors: Kelso J. A. Scott , Bateson Eric V.
Published: October 2005

Though certain languages are hypothesized to possess an inherently rhythmical structure, the basis of any temporal regularity in speech production remains elusive. Efforts to identify rhythmical properties in the speech signal have been plagued by the influence of segmental variation on the measurement process. Here we remove such variation as much as possible but do not annihilate the spoken act's stress and prosodic structure. Subjects produced sentences from the rainbow passage in a reiterant speech mode in which normal syllables were replaced by /ba/ or /ma/. Infrared LEDs were placed on the jaw and lips and monitored using a modified SELSPOT optical tracking system. When pauses marking syntactic boundaries were removed, a high degree of rhythmicity was observed revealed by a periodicity in peak-to-peak jaw displacement of 5–6 Hz and small coefficients of variation (SD of jaw period/mean period). Though the cycling was not perfectly isochronous, it nevertheless exhibited orbital stability on the phase portrait (jaw displacement versus velocity). Such stability regimes are characteristic of nonlinear, limit cycle clocking modes which seem to underlie a variety of motor activities, and may constitute the fundamental “grid” within which segmental events must cohere. [Work supported by NINCDS, BRSG, and ONR.]
Normal mode identification for impedance boundary conditions
Authors: Lindberg Jo B. , Koch Robert A. , Vidmar Paul J.
Published: October 2005

When using the plane wave reflection coefficient (impedance condition) to represent bottom interaction for a normal mode calculation, counting the number of eigenfunction zeroes to identify normal modes is impractical. A method for identifying these modes by knowing the characteristics of the reflection coefficient is presented [R. A. Koch, P. J. Vidmar, and J. B. Lindberg, J. Acoust. Soc. Am. (to be published)]. In the absence of attenuation, the mode number is the number of eigenfunction zeros for the water column plus a contribution given by the number of times that, as a function of horizontal wavenumber, the reflection coefficient circles the origin in the complex plane. [This work was sponsored by the Office of Naval Research.]
SOFAR propagation conditions in the Subarctic Northeast Pacific Ocean
Authors: Chow R. K. , Powell J. W. , Browning D. G.
Published: October 2005

An analysis of two extensive sound speed profile surveys (winter and summer) of the subarctic region (above 40° N) in the Northeast Pacific Ocean shows that the depth of the deep sound channel axis is symmetric about the Alaskan Gyre (centered at 53° N, 153° W). The axis depth ranges from 100 m or less in the Gyre to 400 m or more along the coast of North America. A model for axis depth contours is developed which has some interesting results. For example, a north-south track along 140° W has little change of axis depth with latitude, while an east-west track along 53° N would have a significant change in axis depth. An estimate of the yearly variability of the model is made from long-term oceanographic data obtained at weather station “Papa” (50° N, 145° W). Sound propagation loss is computed for selected tracks using a range-dependent acoustic prediction model and these results are compared to experimental data.
Acoustic effects of a Northeast Pacific Ocean warm-core eddy
Authors: Browning D. G. , Chow R. K. , Powell J. W.
Published: October 2005

A persistent large (200-km-diam) warm core eddy has been reported [S. Tabata, J. Phys. Oceanogr. 12, 1260–1282)] in the Northeast Pacific Ocean off Sitka, Alaska (centered at 57° N Lat., 139° W. Long.). Not associated with a strong boundary current, this eddy differs from a previously observed South Pacific warm-core eddy [P. Scully-Power et al., J. Acoust. Soc. Am. 63, 1381–1388(1978)]. It appears to be formed from an influx of warm water at 100–500 m which moves northward along the North American Coast. At these high latitudes the surrounding water mass is relatively cold with a deep sound channel axis at 200 m and a secondary channel axis at 100 m. It was found, as in other warm core eddys, that there is a broadening of the secondary sound channel. For this eddy there is also a significant change in the depth of the deep sound channel axis (400 m at center of eddy, 200 m outside the eddy). An analysis was conducted using a range dependent propagation modeling program for a sound speed cross-section of the eddy. Results which show the relative effect on sound propagation are given for various source and receiver depths. Computations are also made of the frequency dependence of both the deep and secondary sound channels and their interdependence as suggested by Hall [M. Hall, J. Acoust. Soc. Am. 66, 1102–1107 (1979)].
Circumferential wave propagation in a fluid filled borehole: Experiment
Authors: Plona T. J. , Orton M. , Kleinberg R. L. , Chow E. Y.
Published: October 2005

Circumferential wave propagation measurements were made in an 8-in-diam, water-filled, cylindrical borehole in a limestone block. The bulk compressional and shear speeds of the limestone were faster than the compressional speed in water. Transmission measurements were made as a function of circumferential angle from a source to a receiver both of which were located at the borehole wall. The broadband measurements were centered at 80 kHz. From the time waveforms, arrivals due to circumferentially propagating compressional, pseudo-Rayleigh and Stonely waves can be identified as well as direct and multiply reflected fluid arrivals. A direct measurement of the velocity dispersion in the pseudo-Rayleigh arrival was made. The measured velocity increased with frequency and approached the speed of the pseudo-Rayleigh wave of the planar interface case. The time waveform arrival structure and the velocity dispersion of the arrivals are shown to be in good agreement with the predictions of the theory described in the previous paper.
Structural response of a rectangular panel under heavy fluid loading
Authors: Crighton D. G. , Innes D.
Published: October 2005

Studies reported to the last two meetings of the Society have given a method for calculation of the motion of finite plates under heavy fluid loading and driven by concentrated forcing. The cases considered previously—strips of finite width and circular plates—are typified by the fact that the surface waves excited by the forcing are normally incident upon the plate edges. Moreover, it was shown that on reflection the incident wave suffers a change in phase, but not in amplitude. The phase shift, calculated from a series of semi-infinite problems, has been tabulated for eight different cases of common interest. This rigorous asymptotic method is here extended to surfaces—for example, the rectangular plane—involving oblique incidence on the edges. The reflection coefficient, as a function of incidence angle, is obtained from the solution of canonical semi-infinite problems. The resonance frequencies and mode shapes for a freely suspended rectangular membrane with a fixed or free edge are predicted; futher extensions to finite plates are promised. [Work supported by ONR and DTNSRDC.]
Accuracy of various fluid loading approximations for computing far field acoustic pressure from an infinite orthotropic cylinder
Authors: Haberman Robert C.
Published: October 2005

Numerical methods for predicting acoustic radiation from ring stiffened cylindrical shells generally consist of large finite element or finite difference models of the shells coupled with finite element or integral representations of the fluid. The time requited to develop the fluid model and perform the computations can be reduced by any of the following fluid loading approximations: (1) incompressible fluid, (2) purely resistive fluid represented by the specific acoustic impedance ρc, (3) combination of incompressible and resistive fluid to form a doubly asymptotic approximation, and (4) no fluid. The accuracy of these approximations is estimated by using them and a compressible fluid in a closed form solution for the farfield pressure from a point driven infinitely long orthotropic cylinder. Numerical comparisons as a function of ka and observation angle reveal that, of the four approximations, the ρc fluid gives, in general, the most accurate solution. Furthermore, the absolute accuracy of the solution is shown to be surprisingly good.
Voice source modeling
Authors: Fant Gunnar
Published: October 2005

This is a review of recent research within our department directed toward an extended source-filter theory of voice production, inverse filtering, and other experimental analysis technologies, parameterization, and studies of voice source dynamics. The relative merits of various definitions of source and filter functions are discussed. An extension of the Fant 1979 source model is presented and an automatic parameter analysis procedure is described. Source parameter changes with increasing voice intensity and frequency and articulatory interaction are discussed. The interactive components are generally not preserved in synthesis, but appears to be kept at high F0.
Vowel formants in the soprano voice
Authors: Oncley Paul , Irwin Carol Oncley , Krueger Barbara , Clark Eileen
Published: October 2005

The formants which determine vowel quality lie largely in the range of 250–1850 Hz. Since normal speech, male or female, rarely has fundamentals higher than 250 Hz, it is generally assumed in speech studies that the formants modify the harmonics of the laryngeal tone. The same assumption is valid for the lower singing voices with occasional exceptions. The fundamental pitch of the soprano singer however often goes as high as 1320 Hz, and sometimes in the coloratura voice, considerably higher. Very little attention has been given by voice scientists to the interaction of formants and laryngeal tone in this region. The present study has been addressed to the question of whether harmonics of the formants modify the fundamental of the glottal tone in this range (sometimes called the “whistle register”). The presentation will include videotapes of one of the authors (EC), singing tones between 1000 and 1500 Hz, using vowel positions of most of the usual singing vowels. Each vowel had formants whose harmonics reinforce the fundamental pitch, but there is virtually no audible difference in the vowel character of the resulting sounds.
Ohm's acoustical law for repeated random signals
Authors: Pollack Irwin
Published: October 2005

Following Guttman and Julesz, we employ repeated random signals as a tool for the study of low-frequency auditory periodicity. We have examined the periodicity perception for the weighted mixture of two repeated random components. Three hypotheses are suggestive for the dominant periodicity of the mixture: separability, in which the periodicities of the separate components each can be discerned (this is Ohm's acoustical law for repeated random signals); strict periodicity, in which the dominant periodicity is identified with the lowest common multiple of the component periodicities; and beat frequency, in which the dominant periodicity is identified with the frequency difference in periodicity of the components. The vast majority of periodicity matchings tend to support the separability hypothesis or Ohm's acoustical law for repeated random signals. The proportion of periodicity matchings assigned to the separate components is extremely sensitive to the relative weightings of the components. Moreover, periodicity matchings are sensitive to the specific random patterns employed.
Incorporating intensity effects into a temporal model of frequency discrimination
Authors: Wakefield Gregory H. , Nelson David A.
Published: October 2005

The effects of duration and of frequency on frequency discrimination are consistent with a neural-based model that used the temporal information present in the phase-locked response of VIIIth nerve units to estimate the frequency of a pure-tone stimulus [J. L. Goldstein and P. Srulovicz, “Auditory-nerve spike intervals as an adequate basis for aural frequency measurement” in Psychophysics and Physiology of Hearing, edited by E. F. Evans and J. P. Wilson (Academic, London, 1977), pp. 337–346]. We have extended this model to include the effects of intensity on the frequency difference limen by incorporating (1) the functional dependence of the synchronization index on intensity [D. H. Johnson, doctoral dissertation, MIT, 1974] and (2) an appropriate dependence of average discharge rate on intensity. When fit to the psychophysical data, the extended model accounts for much of the effect of intensity on the frequency DL above 10 dB SL. At levels lower than this, the model does not exhibit the same rapid increase in the DL observed in the human data. We will discuss this discrepancy as well as our attempts to generalize the model to account for the effects of hearing impairment on frequency discrimination. [Work supported by NINCDS Grant NS12125.]
Auditory representation in the cerebellum of the CF-FM bat, Pteronotus parnellii parnellii
Authors: Sun Xinde , Kamada Tsutomu , Jen Philip H.-S.
Published: October 2005

Auditory representation in the cerebellum of the mustache bat, Pteronotus parnellii parnellii, was studied with microelectrode mapping technique. This study reveals that a large area of the bat's cerebellum contains units responding to acoustic stimuli. A study of the frequency representation and frequency tuning of the recorded units show that auditory specialization for processing of species-specific echolocation sounds also exists in its cerebellum. There is a large number of units with best frequencies between 60 and 64 kHz. These units have extremely sharp threshold curves with a Q10-dB value as high as 394. These auditory units are organized into a fractured pattern of small patches within which units are tonotopically organized. The units also appear to be columnarly assembled according to their best frequencies. Although different in sensory modality, such a finding is highly comparable to the fractured somatotopy in the tactile areas of cerebellar hemispheres [Shambes et al. Brain Behav. Evol. 15, 94–140 (1975)] [Work supported by NSF 80-07348 and USPH 1-K04-NS00433 to P. Jen.]
Frequency resolution, auditory disability, and handicap
Authors: Lutman Mark E.
Published: October 2005

Nine hundred subjects have been examined in our clinics using a battery of tests including pure-tone audiometry, frequency resolution (PTC), an audio-visual speech disability test, and disability/handicap questionnaires. Subjects were sampled from the electoral register, stratified according to self-reported hearing loss and tinnitus, and sub-sampled disproportionately to increase the incidence of impairment in the subject group. Results indicate a high correlation between frequency resolution and pure-tone sensitivity. The derived auditory component of the audiovisual disability test was correlated with age and pure-tone sensitivity but there was negligible further correlation with frequency resolution after partialling. Relationships between questionnaire measures of disability/handicap and auditory impairment has also been examined. These data provide population statistics for the various test measures and a powerful means of identifying their interrelationships.
Tone-on-tone masking in the chinchilla auditory nerve
Authors: Sinex D. G. , Havey D. C. , Boettcher F. A.
Published: October 2005

The psychophysical masking of one tone by another has been attributed to the pattern of mechanical events produced in the cochlea by the two tones. To evaluate that possibility, we compared the patterns of synchronized discharges of chinchilla auditory nerve fibers elicited by tone pairs to tone-on-tone masking patterns previously obtained behaviorally in the same species. The contributions of a 1000-Hz masker (F1), various probe tones (f2), and the distortion products f2−f1 and 2f1−f2 to the synchronized responses were determined from period histograms. A “masked threshold” for each fiber was derived from plots of synchronized rate versus the level of f2. The pattern of fiber masked thresholds versus f2 for the most sensitive fibers resembled the masking pattern obtained behaviorally. Thresholds for cochlear distortion products also resembled the behavioral thresholds, lending support to the suggestion that notches in the masking pattern indicate detection of aural distortion products. The neural data do not reproduce the behavioral data in all respects. However, the data suggest that the general form of the masking pattern is determined peripherally and that psychophysical masking and physiological two-tone suppression are closely related. [Supported by NIH.]
Response properties of cochlear nucleus units specialized for encoding amplitude modulation
Authors: Frisina Robert D. , Smith Robert L. , Chamberlain Steven C.
Published: October 2005

Extraction of information about modulations in sound amplitude may be a specialized function of the cochlear nucleus. CHOPPERS encoding amplitude modulation (AM) over an intensity range 50 dB greater than auditory nerve fibers are found in the gerbil posteroventral cochlear nucleus (PVCN) [R.D. Frisina, R. L. Smith, and S. C. Chamberlain, Soc. Neurosci. Abstr. 8, 347 (1982)]. Although other CHOPPERS show large responses to AM over only a limited intensity range, further quantitative analyses did not support our initial classification of CHOPPERS into two groups. We performed analyses on responses to 150 Hz AM at many intensities and found that the magnitudes of the fundamental components formed a continuum rather than two discrete groups. Also, spontaneous activity rates, tuning curves, and the shapes of rate-intensity functions for pure and wide-band noise are similar for both kinds of CHOPPERS. We suggest that CHOPPERS in the PVCN represent a single class of units specialized for encoding AM with, perhaps, each unit having different AM tuning properties. [Supported by NSF and NIH.]
Acoustic streaming in a waveguide with slowly varying height
Authors: Thompson Charles
Published: October 2005

An analysis of acoustic streaming in a two-dimensional waveguide having slowly varying height is presented. Special attention is paid to waveguides with cross sections that are small compared to the acoustic and/or wall wavelength. It is shown that the dynamic behavior of the enclosed fluid can be parametrized by the values of three small parameters ε, 1/S, and 1/R, where ε is the wall slope, S is the oscillatory Strouhal number, and R is the oscillatory Reynolds number. An analytical solution describing the streaming flow is given in terms of a regular perturbation sequence in ε. Outside the boundary layer it is shown that the time averaged slip velocity is the sum of two term. The first terms is proportional to the product of the incident and reflected wave amplitudes. The second is proportional to the difference between incident and reflected acoustic intensities of the wave. It is shown that the streaming solution remains valid until εR/S2 = O(1). The results for a duct with sinusoidally varying height in the incompressible limit and waveguides with a Gaussian variation in height are given.
Reed regeneration apparatus with controllable up- and downstream impedances
Authors: Hoekje P. L. , Benade A. H.
Published: October 2005

Linear reed regeneration theory gives the oscillation condition as A⋅[(Zu + Zd)/Zr]⩾1, where A, Zr are the transconductance and impedance of the reed, and Zu and Zd are the input impedances of the air columns looking upstream and downstream from the reed. Extension to nonlinear regimes retains this relation between Zu, Zd and Zr. We have constructed a reed apparatus such that reeds of varying materials and thicknesses can be reproducibly set over continuous ranges of position and angle of the reed in the window in which it operates. Being end-for-end reversible and adjustable, the reed flow-controller can be operated in all four configurations (± E, ± I) of the Bouasse-Melde nomenclature, e.g., clarinet-type inward striking (+ E), and trumpet-type outward striking (+ I). Means are included to provide adjustment of peaks and dips of Zu, analogous to the accustomed control of the downstream air column Zd, while permitting DC flow at pressures above atmospheric. Provision is made for mircophones and probes placed symmetrically with respect to the reed. [Work supported by NSF.]
Modes of vibration of Indian drums
Authors: Rossing Thomas D. , Sykes W. Arnold
Published: October 2005

The tabla of North India and the mrdanga of South India have been studied by a succession of Indian scientists including Nobel laureate C. V. Raman. The harmonic partials of these drums are recognized as resulting from carefully loading the membranes with a paste of starch, iron oxide, charcoal, or other materials. Five harmonics result from nine modes of vibration, several of which have the same frequency. In order to study this effect, we recorded the sound spectra of a mrdanga at 32 stages during the application of a patch. The transformation of the unloaded head's vibrational modes into the harmonically related modes of the tuned drum is discussed. We also describe sound spectra of different strokes applied at different orientations of the drum.
Consonance and dissonance judgments of musical chords
Authors: Roberts L. A.
Published: October 2005

The terms consonance and dissonance refer to subjective impressions of musical sounds. No studies have examined perception of consonance with stimuli other than two-note chords. Naive and musically trained subjects rated the degree of consonance and dissonance (both smoothness and pleasantness judgments) of major, minor, diminished, and augmented three- and four-note chords. These chords were presented in root position, first inversion, and second inversion and varied in their harmonic structures and intonations. Naive subjects were strongly influenced by harmonic structures of chords. Neither group of subjects made judgments contingent on the intonation system used. Both naive and musically trained subjects rated major chords as most consonant, followed by minor, diminished, and augmented chords. These patterns occurred in all conditions and were most pronounced for musically trained subjects. When all of the harmonics were present, root-position chords were rated as most consonant and second inversions least consonant. These findings are discussed in terms of theories of consonance and dissonance.
Analysis of speech sounds transduced by different types and positions of sensors
Authors: Viswanathan Vishu R. , Stevens Kenneth N. , Karnofsky Kenneth F. , Alakel Michael N.
Published: October 2005

In this paper, we present the results of our analysis of different classes of speech sounds, which we recorded from different types of sensors located at each of a number of positions in the vicinity of the talker. The sensor types we used include pressure microphone, first-order pressure-gradient microphone, second-order pressure-gradient microphone, and accelerometer (for measuring vibrations on the surfaces of head and neck). For measuring the pressure gradients, we constructed an array of three closely spaced electret microphones; in addition, we used a prototype first-order gradient microphone and a prototype second-order gradient microphone. A specially chosen set of speech utterances was recorded from three male and two female subjects. We considered both short-term and long-term amplitude and spectral characteristics of speech in our comparative investigation of the various sensor signals. Differences in the output of various sensors were especially noticeable for nasals and voiced obstruents for which the source of sound radiation is significantly away from the lips. [Work sponsored by Air Force Systems Command, Rome Air Development Center.]
Experiment with a color spectrogram
Authors: Kuhn G. M.
Published: October 2005

We define a color spoctrogram which retains all the information of the standard spectrogram, and which encodes information about the shape of each spectrum into the chromaticity with which that spectrum is illuminated. The chromaticity for each spectrum is derived by interpreting each spectrum as an energy distribution in the visible light frequencies. Examples are given for speech.
Techniques for the analysis of phonation types in natural languages
Authors: Ladefoged Peter , Maddieson Ian
Published: October 2005

A variety of techniques have been applied to the analysis of contrastive phonation types by the UCLA Phonetics group. The languages examined have included Burmese, Mon, !Xoo, Korean, Hindi, Bengali, and Marathi. To ensure that we are describing stable properties of the language, rather than individual idiosyncracies, several speakers of each language have been studied. In some cases (e.g., Burmese) the speakers have come to our laboratory, so a full range of experimental techniques can be used. In other cases we have collected data in a desert (e.g., !Xoo) or a monastery (e.g., Mon), so that we were limited to what was portable or socially acceptable. Laboratory techniques include recovery of the glottal volume/velocity waveform through inverse filtering, electro-glottography, and video-taped fiberoptic studies of vocal cord movements. Field techniques include using an airflow mask and pressure transducers, as well as making audio recordings for later spectrographic or other acoustic analysis. The costs and benefits of all these procedures in their application to natural language data will be discussed. [Research funded by NIH.]
A method to automatically constitute phonetic dictionaries for speech comprehension systems
Authors: Neel F. , Eskenazi M. , Mariani J.
Published: October 2005

The purpose is to match portions or segments of the continuous speech signal with phonetic symbols (phones, diphones, syllables …). The method consists of using a dynamic comparison algorithm with local optimum which finds the best path in the matrix of recognition scores. These scores are assigned to each phonetic symbol at each stable instant detected in the signal. The method deals with segmentation errors (missing or extra segments). The phonetic transcription of the phase that has been pronounced takes into account the phonological variations inherent in the language itself and due to the differences between various speakers (elisions, “liaisons”… may be used). First results from a database comprising 50 phrases (each less than 2 s long) pronounced by two different male speakers are given. The ultimate goal is to permit the adaptation of existing phonetic dictionaries to new speakers by requiring only a few chosen sentences to be pronounced.
Introduction to session on absorbing and damping materials
Authors: Holmer C. I.
Published: October 2005

This paper provides an introductory framework to a session dealing with absorbing and damping materials. The opening segment describes analogs and differences between the two material concepts. Each in its respective sphere represents the principal engineering means of adding energy dissipation to control reverberation in a system. Some historical perspectives are noted, including conceptual simplicity and practical difficulties of use. Next is a brief review of major physical principles which influence performance. The analog of the role of treatment thickness on absorption and extensional damping performance is explored. The relation between normal and random incidence absorption and extensional damping performance is explored. The relation between normal and random incidence absorption coefficient is discussed. The recent work of Bliss [J. Acoust. Soc. Am. 71, 533–545 (1982)] on introducing an alternative boundary condition is explored for its relevance to this problem. An analog to coupled-room acoustics problems exist in connected structures with applied damping. Maidanik's [J. Sound Vib. 77, 245–250 (1981)] recent work in this area will be discussed for its relevance to the performance of damping treatments. The third portion of the paper reviews recent properties measurements activities including ASTM C-384, C-423, and E-756 as well as new measurement procedures. Closing remarks will deal with present and future directions of materials development.
Acoustical properties of open-cell foams
Authors: Lambert R. F. , Tesar J. S.
Published: October 2005

This work presents a comprehensive study of acoustical propagation and surface admittance characteristics in open-cell, highly porous, elastic polyurethane foams. Studies of the acoustical structure of such foams have been reported [J. Acoust. Soc. Am. 72, 879–887 (1982)]. The foams are available commercially in graded pore sizes for which information about the static flow resistance, thermal time constants, dynamic structure factors, and the speed of sound is known. The theory is a synthesis of earlier work by Zwikker and Kosten and by Biot. Some ideas developed for analysis of sound in granular structure are carried over to fibrous structure. The agreement between prediction and measurements of attenuation, progressive phase constant, and normal incidence surface admittance is excellent over a frequency range extending from about 15 to 6000 Hz. The implications of fibrous structure on the surface admittance of open-cell foams is quite evident from the results. Some results on the admittance of samples of finite depth backed by a rigid, impervious surface also will be presented. The data are readily extended to predictions of surface absorption coefficients. [Work supported by NSF and NASA.]