bluuebunny/crossref_metadata_2025 huggingface dataset:
Model reference glottal inverse filter of high F voice
Authors: Milenkovic Paul , Bless Diane M.
Published: October 2005

Linear predictive coding analysis (LPC) has been advanced as a technique for constructing an inverse filter which recovers the glottal airflow wave from the acoustic speech wave [D. Y. Wong, J. D. Markel, and A. H. Gray, Jr., IEEE Trans. Acoust. Speech Signal Process. ASSP-27, 350–355 (1979); M. G. Berouti, D. G. Childers, and A. Paige, IEEE ICASSP, 33–36 (1977)]. Because of the problem of interaction of LPC analysis with the glottal source, the LPC analysis interval is restricted to the closed glottis interval. This requirement is a problem in analyzing breathy and falsetto voice in males and all voice types in females. A model reference technique has been developed which constructs the inverse filter from information contained in an entire pitch period. The technique computes the inverse filter by performing a joint optimization of a model of the glottal pulse together with the inverse filter coefficients. A striking comparison is obtained between the model reference technique and LPC applied to female voice and falsetto male voice. [Work supported by NIH.]
Defining and developing intelligibility in a text-to-speech system
Authors: Goudie Kathleen M.
Published: October 2005

A text-to-speech system has three linguistically related components that each affect the intelligibility of the speech it produces. These three components are: (1) the quality of the individual sounds and how they fit together (or how the algorithm fits them together), (2) the accuracy of the output of the ASCII to phonemic unit algorithms and/or rules, and (3) the quality of the prosodic algorithms, including work-stress and pausegroup implementations. Text-to-speech development, therefore, should involve the testing of each of these components for the purposes of maximizing the accuracy and acceptability of each component's output. This presentation will therefore discuss an implementation of testing algorithms for phonemic construction intelligibility, for ASCII-to-phone accuracy, and for prosodic naturalness for the development of a text-to-speech system for the T.I. Professional Computer.
Nonlinear bubble oscillations
Authors: Prosperetti A.
Published: October 2005

The propagation of strong pressure pulses in bubbly liquids, the transition to chaotic behavior in dynamical systems, and the development of new experimental methods of investigation motivate recent interest in large-amplitude forced bubble oscillations. The results of some numerical and analytical investigations of simple mechanical models of the phenomenon will be reviewed. Physical processes not accounted for in these studies, namely the compressibility of the liquid and the complex thermal processes taking place in the bubble, will then be discussed by means of suitable modification of the available theoretical formulations. Numerical results will be presented showing the importance of these effects in large-amplitude motion. An application to the damping of pressure pulses in bubbly liquids concludes the presentation. [Work supported in part by ONR and NSF.]
Subharmonic emissions of a cavitating bubble
Authors: Commander Kerry W. , Crum Lawrence A. , Holt Ray Glynn
Published: October 2005

The nonlinear oscillations of a gas bubble pulsating in a liquid are critically dependent on the damping of these oscillations. We have used an accurate numerical formulation of the problem to investigate the subharmonic emissions from forced oscillations of the bubble. A systematic approach to chaotic behavior will be shown through successive bifurcation of the subharmonic emissions as the driving pressure is increased. [Work supported in part by the NSF and the ONR.]
Optical scattering properties of bubbles of interest in acoustics and cavitation research
Authors: Marston Philip L. , Langley Dean S.
Published: October 2005

Light scattering gives the size of bubbles and could be used to measure their responses to steady and modulated sound. We have observed and modeled novel optical scattering phenomena of bubbles. Recent progress to be summarized includes: (a) Models and observations of the optical glory [P. L. Marston and D. S. Langley, Proc. SPIE 489 (1984).] We sum the transmitted-wave glory amplitudes and the axial reflections; the total scattering agrees with the Mie partial-wave series in the size range of interest. (b) High resolution measurements and calculations were made of near critical angle (φc) scattering [D. S. Langley and P. L. Marston, Appl. Opt. 23, 1044–1054 (1984)]. (c) A proposed method for measuring the radial pulsation amplitude Δa of a bubble of mean radius a uses the shift Δ I in the irradiance scattered to an angle somewhat below φ c ≃83°. The observed region should be close to the center of the N th coarse oscillation cycle below φc with a peak-to-peak irradiance variation (as a function of the scattering angle) of Ipp. The magnitude of the shift ΔI≃(Δa) π Ipp N/2a should greatly exceed the shift due to changes in geometric cross section and it could facilitate the detection of small Δa. [Work supported by ONR.]
Theory of flow-excited cavity resonance in an incompressible fluid
Authors: Elder S. A.
Published: October 2005

In heavy incompressible fluids, such as water, Helmholtz resonance is not expected to occur. However, tests with a towed free-flooding cavity have shown that an incompressible breathing-mode resonance does occur at certain speeds which produces tones analogous to those caused by Helmholtz resonance in air. The effect seems to be dependent on the presence of wall-vibration modes capable of producing a modulation in the cavity cross section. Following a procedure developed for air-cavity resonances, a feedback theory is derived in terms of lumped mechanical circuit parameters. It is shown that Strouhal number for resonance is fixed by quantization of stability waves in the separated shear layer. System amplitude is believed to be limited by nonlinear orifice impedance analogous to that which is known to take place in air cavities. [Work supported by Naval Sea Systems Command General Hydromechanics Research Program, administered by the David W. Taylor Naval Ship R & D Center, Bethesda, MD.]
Turbulent boundary layer pressure fluctuations in a glycerine flow
Authors: Daniels Mark A. , Lauchle Gerald C.
Published: October 2005

Measurement of turbulent boundary layer (TBL) wall pressure fluctuations are oftentimes contaminated by facility generated noise. A technique originally proposed by Horne and Hansen (Naval Research Lab) is considered to aid in the extraction of a signal due to the TBL wall pressure fluctuations from extraneous background noise. This technique involves utilizing an array of three, axisymmetric, flush-mounted hydrophones. Difference signals are generated between two pairs of hydrophones. Then the cross spectrum of these difference signals is calculated, yielding the power spectrum of the TBL. The facility in which the abovementioned procedures are being conducted uses glycerine as the working fluid. Because the viscosity of glycerine greatly exceeds that of water—for a fixed temperature—a study of the electroacoustic characteristics of the hydrophones in both water and glycerine was conducted. This study involved a pulse-tube-type calibrator for the hydrophones and various comparison techniques between water and glycerine. The results show a negligible difference in hydrophone sensitivities between the two media. [Work supported by NAVSEA and ARL/PSU E and F.]
Measurements of low-wavenumber spectra of wall turbulence in a bouyant body facility
Authors: Long D. F. , Wurtz M. , Lambert R.
Published: October 2005

In the measurement of wall pressure fluctuations beneath a turbulent boundary layer, acoustic contamination is a primary concern, particularly at low wavenumbers where the true energy level is quite low. In view of this, an experiment is being designed to measure the wall pressure on the surface of a bouyant underwater body. This eliminates the need for a pumping device which is the major source of noise contamination in conventional facilities. The experiment is conducted at Reδ* = 15 000 to be comparable to an ongoing water tunnel experiment. The experimental test duration is ∼ 1 s. Preliminary power spectral density measurements agree well with the water tunnel data as long as the individual spectra from each run are ensemble averaged. Further experiments will employ an array of finite sized transducers appropriately spaced to study the low wavenumber region. The transducers will be simultaneously sampled and subsequently processed by computer. Using some new techniques in spectral estimation it is hoped that this will result in a uniformly valid expression for the wavenumber frequency spectrum, particularly at low wavenumber. [Work supported by ONR, SRO program in Hydroacoustics.]
The National Voluntary Laboratory Accreditation Program for acoustical laboratories
Authors: Thomas Douglas B.
Published: October 2005

The National Voluntary Laboratory Accreditation Program (NVLAP) administered by the National Bureau of Standards (NBS) established a program in September 1982 to accredit laboratories that provide acoustical testing services. The Acoustics Laboratory Accreditation Program (Acoustics LAP) covers 50 national and international test methods for which laboratories can apply for accreditation. The test methods cover procedures for measuring the properties of acoustical materials, sound power and sound pressure level measurements, and field measurements of noise produced by products. Applicant laboratories are evaluated through on-site visits by experienced assessors and by participation in proficiency testing programs administered by NVLAP. Proficiency testing programs are currently in operation for test methods ASTM C423 and ASTM E90. Seven laboratories are currently accredited for various test methods and additional laboratories are in the process of being evaluated for accreditation.
Laboratory accreditation: A manufacturer's experiences
Authors: Lang M. A.
Published: October 2005

The experiences of a manufacturers laboratory which has recently completed the NVLAP accreditation process for acoustical testing will be discussed in this paper. The accreditation process involves a significant amount of effort on the part of the laboratory undergoing accreditation and the motivation for undergoing accreditation, the costs, benefits, and problem areas which have surfaced will be discussed for both the initial laboratory accreditation and the subsequent proficiency testing.
The acoustics NVLAP program: A day or two in the life of an assessor
Authors: Sullivan Joseph W.
Published: October 2005

Since the National Voluntary Laboratory Assessment Program (NVLAP) for acoustics has been in effect for less than two years and the assessors in the program have been exposed to only a few laboratories, it may be presumptuous to make any general statements about the program. Rather than attempt any such generalizations, the intent of the paper will be to explain the role of the assessor and then to give a synopsis of what it's like to inspect a laboratory, beginning with establishing a rapport with understandably apprehensive laboratory personnel; coping with the constant press of time throughout the inspection to check records, equipment, and observe test procedures; handling the voluminous paperwork; and finally ending with the good news/bad news briefing, all compressed within one or two days.
Mössbauer measurements of basilar membrane tuning curves in the chinchilla
Authors: Robles L. , Ruggero M. A. , Rich N. C.
Published: October 2005

Basilar membrane velocity was measured in the basal turn of the chinchilla cochlea using the Mössbauer technique. The threshold of the compound action potential (AP) recorded at the round window in response to pure tone bursts was used to monitor the physiological state of the cochlea. Isovelocity curves at 0.05 to 0.1 mm/s are sharply tuned, with characteristic frequencies (CF) between 8 and 9 kHz. Intensity functions are linear for frequencies below 7 kHz, but display a marked nonlinearity for frequencies around and higher than CF. Isovelocity curves at 0.1 mm/s for the best experiments—those with the least elevation of AP thresholds at CF—reach their minimum at about 14 dB SPL, have a Q10 of about 5, a high-frequency amplitude slope of about 300 dB/oct, and tip-to-tail ratios of about 44 dB at one octave below CF and 64 dB at 1 kHz. Thus, in the chinchilla, mechanical tuning curves are as sharply tuned as frequency-threshold curves of auditory nerve fibers with corresponding CF. [Work supported by NINCDS and NSF.]
Tone-on-tone masking in auditory nerve fibers with medium spontaneous rates
Authors: Sinex Donal G. , Havey D. C.
Published: October 2005

Auditory nerve fibers can be categorized by their spontaneous discharge rates (SRs). Although the significance of these categories is not known, it has been suggested that the low and medium SR fibers may contribute to frequency analysis at signal levels at which the more sensitive high SR fibers are saturated. Data obtained from chinchilla auditory nerve fibers in response to two-tone stimuli are consistent with that possibility. The stimuli were presented in a tone-on-tone masking paradigm analogous to that used by Long and Miller [Hear. Res. 4, 279–285 (1981)] for psychophysical measurements in chinchillas, and average discharge rate data from fibers with medium and high SRs were analyzed separately. Good agreement between the “masked thresholds” of medium SR fibers and the behavioral masked thresholds was observed. Masked thresholds of medium SR fibers were sometimes lower than the masked thresholds of high SR fibers, even though their quiet thresholds tend to be higher. This difference was greatest (10–15 dB) for masker-probe tone pairs with the smallest frequency separations. High SR fibers were often driven to saturation by the masker, so that no masked threshold could be estimated; however, probe tones nearly always produced rate changes in medium SR fibers. [Supported by NIH.]
A model for post-spike recovery effects in PST histograms
Authors: Bosch Walter R.
Published: October 2005

Gaumond, Kim, and Molnar [J. Acoust. Soc. Am. 74, 1392–1398 (1983)] developed a Markov chain “wheel model” as a tool for estimating the stimulus-dependent and recovery-dependent factors of the spike discharge probability for auditory nerve spike trains. Using this model, we have calculated the expected PST histograms for various recovery functions (RF) and excitation functions (EF) in order to examine the distortion in the shape of the PST histogram resulting from post-spike recovery effects. For constant EF the Markov chain state occupancy distribution varies with value of the EF. Using an RF similar to that found by Gaumond, it is found that step changes in the EF lead to overshoot/undershoot in the PST histogram which decays as the state occupancy distribution approaches its new equilibrium. A plot of equilibrium firing rate versus EF value yields a compressive nonlinear function. For a slowly varying EF, the state occupancy distribution is never far from equilibrium, and the PST histogram is well-approximated by applying this nonlinear function to the EF waveform. [Work supported by NIH grants GM07564 and NS07498.]
Lateral inhibition and the processing of speech evoked activity in the auditory nerve
Authors: Shamma Shihab
Published: October 2005

A relatively simple and biologically realistic model of a uniform lateral inhibitory (LIN) network is shown capable of extracting from the complex spatio-temporal firing patterns of the cat's auditory nerve the formants and low-order harmonics of synthetic voiced speech stimuli. The model provides a realistic mechanism to utilize the temporal aspects of the firing and thus supports the hypothesis that the neural coding of complex sounds in terms of average rates can be supplemented by the information coded in the synchronous firing. At low levels of intensity the LIN sharpens the average rate profiles. At moderate and high levels the LIN uses the cues available in the distribution of phases of the synchronous activity which exhibit rapid relative phase shifts at specific CF locations (corresponding to the frequencies of the formant and low-order harmonies in the stimulus). These temporal phase shifts manifest themselves at the input of the LIN as steep and localized spatial discontinuities in the instantaneous pattern of activity across the fiber array. The LIN enhances its output from these spatially steep input regions while suppressing its output from spatially smooth input regions (where little phase shifts occur). In this manner the LIN recreates from the response patterns a representation of the stimulus spectrum using the temporal cues as spatial markers of the stimulus components rather than as absolute measures of their frequencies. Similar results are obtained with various lateral inhibitory topologies, e.g., recurrent versus nonrecurrent, single versus double layer, and linear versus nonlinear.
Further measurements, and modeling, of interactions among spontaneous oto-acoustic emissions
Authors: Burns Edward M. , Tubis Arnold , Jones Kenneth , Strickland Elizabeth A.
Published: October 2005

We have previously reported on the existence of various interactions among spontaneous oto-acoustic emissions (SOAEs) including: (1) intermodulation distortion products, (2) mutual amplitude suppression, and (3) linked-noncontiguous SOAEs. In this paper, we report on the relationships of these phenomena to behavioral threshold finestructure, stimulus frequency emission (SFE) finestructure, and frequency-locking tuning curves. We will also discuss possible explanations for these phenomena with particular emphasis on nonlinear generalizations of the class of models proposed by Kemp [Hear. Res. 2, 533–548 (1980)]. [Work supported by NINCDS and NSF.]
Acoustic emissions of single-tone harmonies in the gerbil ear canal
Authors: Schmiedt A.
Published: October 2005

The most commonly measured and strongest emissions in the ear canal are the cubic difference tones (CDT), like the 2f1-f2 component produced by two primary tones at frequencies of f1 and f2. These emissions are generated by nonlinearities within the cochlea and are absent in a dead or damaged cochlea. Another way to monitor cochlear nonlinearities is to measure the harmonics resulting from a single primary tone. Here, we monitored the second and third harmonics of a tone swept from high to low frequencies. Like the CDT, single-tone distortion can be 10 to 30 dB higher in the ear canal as compared to a metal cavity. Further, both the CDT and the single-tone distortion can briefly increase above their normal values after asphyxia of the animal, before declining to the noise floor of the system. However, unlike the CDT, single-tone distortion is very small for primary frequencies above about 3 kHz. Also, maskers and short-term fatiguing tones associated with the primary frequency do not necessarily reduce the harmonics generated by that primary. Instead, the masker or fatiguer appears to act at the frequencies of the harmonics. These differences in the behaviors of single-tone harmonics and CDTs may be due to their different excitation patterns along the cochlear duct. [Work supported by NSF.]
Computation intensive propagation modeling using the PESOGEN system
Authors: Tappert F. D.
Published: October 2005

The development of a special-purpose microcomputer system (called PESOGEN = parabolic equation solution generator) that performs high-speed PE model computations using the split-step Fourier algorithm was reported previously [L. Nghiem-Phu, S. C. Daubin, and F. Tappert, J. Acoust. Soc. Am. Suppl. 1 75, S26 (1984)]. Applications of PESOGEN to three-variable modeling are reported here. (1) Full-wave computations of pulse propagation, signal waveforms, and cross-correlation functions require p(r,z,t), a function of three variables. PESOGEN computes p(r,z,ω) in the frequency domain for many values of ω sequentially, and then obtains p(r,z,t) in the time domain by numerical Fourier transforms [L. Nghiem-Phu and F. Tappert, J. Acoust. Soc. Am. Suppl. 1 68, S51 (1980)]. Examples of pulse transmission to long range in deep and shallow range-dependent oceans are presented, with emphasis on numerical accuracy (tested by time-domain reciprocity) and PESOGEN throughout times (typically less than 1 h). (2) Full-wave computations of responses of directional receivers to line-radiation sources require p(x,y,z), a function of three variables. Using specified farfield beam patterns, PESOGEN computes p(x,φ,z] for many values of the azimuthal angle φ sequentially, and then obtains TL′ = TL − ASG, where ASG is the array signal gain [F. Tappert and L. Nghiem-Phy, J. Acoust. Soc. Am. Suppl. 1 75, S63 (1984)]. Examples of directional receiver response to long range sources in three dimensions are presented, with emphasis on full-wave effects in the beam deviation loss and PESOGEN throughput times. Conceptual designs of large special-purpose computer systems for three-variable propagation modeling using multiple PESOGEN units in parallel with single-instruction multiple-data architectures are also presented.
Evaluation of propagation for two underwater acoustic ducts
Authors: Pedersen M. A. , Gordon D. F. , Edwards D.
Published: October 2005

Perturbations in mode eigenvalues, associated with interactions between two ducts, were first investigated by the authors in 1969. The investigation of this interaction is extended to the frequency dependence of the coupling between ducts. A numerical measure of this coupling has been developed and used to determine optimum frequencies of propagation between a source located within one duct and a receiver located within the other duct. These optima occur near frequencies where the phase velocity difference between two adjacent modes forms a minimum. The coupling is strongly frequency dependent. For an example from the North Atlantic, one optimum frequency was found at 53.23 Hz. Propagation loss versus range plots show about 10 dB less loss at 53.23 Hz as compared to 64 Hz, where the mode phase velocities are well separated. Contour plots of propagation loss in the range-receiver depth plane have been generated for a source in the upper duct. At 53.23 Hz these plots exhibit a range periodic transfer of energy back and forth between ducts, with this range period determined by the frequency and the phase velocity difference between the critical adjacent modes. [Work supported by ONR code 4250A.]
Sound propagation in range-degendent double ducts
Authors: Bucker Homer P. , Schey Philip W.
Published: October 2005

In a companion paper, M. A. Pedersen et al. (Abstract Q4) discuss sound propagation in range-independent double ducts. Here we examine the effects of range dependence on the interchange of energy between adjacent acoustic ducts associated with a pair of resonant normal modes. The basic interchange mechanism is reviewed and the interchange distance (i.e., distance required to transfer acoustic energy from one duct to the other) is related to profile parameters. Calculations for a partial field (two modes) and a complete field (all modes) will be shown. Comparisons are then made between the exact solution (normal modes) and the approximate solution (parabolic equation) which will be used for range-dependent calculations. Propagation loss curves and sound intensity contours will be given for two realistic cases of range dependence. It is shown that moderate range dependence does not destroy the fundamental resonance transfer of energy between ducts, but it does cause significant variation of the interchange range. [Work supported by ONR code 425AC.]
Nonlinear effects in long range underwater propagation
Authors: Cotaras Frederick D. , Morfey Christopher L. , Blackstock David T.
Published: October 2005

An ongoing project to investigate the importance of nonlinear effects in long range underwater propagation is described. The project consists of three tasks: (1) to generalize weak shock theory to include effects of relaxation in the ocean, (2) to determine the effects of the stratified medium on propagation of finite-amplitude waves, and (3) to investigate the nonlinear effects close to a caustic. Results, in the form of time waveforms and their power spectral densities, are presented for task 1, task 2, and simplified merger of tasks 1 and 2. In particular the propagation of an explosion pulse along a single ray tube is computed. Emphasis is placed on the differences between the results obtained using nonlinear theory and linear theory. The role of the reduced range variable x̄ = s0 ln(Gs/so), wherein s is the path length, s0 is the reference range, and G is a number indicating the effects of the stratification on nonlinearity, is discussed. [Work supported by ONR.]
The uniform WKB approach extension to multiple channels and bottom propagation
Authors: Henrick R. F. , Brannan J. R. , Forney G. P.
Published: October 2005

The treatment of more complex acoustic signals that have significant bandwidth of limited time duration requires significantly more effort to model ocean propagation. Consideration of such signals requires accurate inclusion of ocean properties such as diffraction and bottom interaction that may show significant variation across the signal bandwidth. Inclusion of such full wave effects typically implies significant computational requirements. A joint analytical/numerical approach using uniform asymptotics was previously demonstrated [R. F. Henrick, J. R. Brannan, D. B. Warner, and G. P. Forney, J. Acoust. Soc. Am. 74, 1464—1473 (1983)] to yield significant reduction in computer run time compared to brute force numerical techniques. This initial application was, however, limited to single channels and did not accurately include bottom interacting energy. Recent extensions include both multiple channel leakage and a realistic geoacoustic bottom. Examples are presented illustrating both the necessity of including the bandwidth of the signal and the accuracy of the approach. It is shown that judicious use of analytical techniques and analysis can be used in conjunction with numerical techniques to reduce the computational load by one to two orders of magnitude.
Underwater acoustic signal simulating in range-dependent environments using Maslov asymptotic theory
Authors: Kendall Elizabeth A.
Published: October 2005

Asymptotic ray theory can be used to simulate acoustic waves in range-dependent oceans, but singularities often occur at points of great interest. Transform methods, such as the WKBJ solution used by M.G. Brown [Ph.D. thesis, University of California, San Diego, CA (1982)], describe waves at the singular points, but they are restricted to laterally homogeneous media. Maslov asymptotic theory combines asymptotic ray theory and transform methods to provide a uniform result applicable in range-dependent environments, and the technique was used by C. H. Chapman and R. Drummond [Bull. Seismol. Soc. Am. 72 (6), S217–S317 (December 1982)] to compute synthetic body-wave seismograms. This paper describes a. computer simulation of underwater acoustic signals in range-dependent oceans that is based upon Maslov asymptotic theory. The theory and computer implementation will be summarized, and examples will be discussed. Preliminary results on range-dependent signal inversions for underwater acoustic tornography will also be presented.
A computer program for synthesizing English intonation
Authors: Pierrehumbert Janet B. , Anderson Mark
Published: October 2005

This paper will describe a computer program for synthesizing English fundamental frequency contours (F0 contours) which attempts comprehensive coverage of the different English melodies. The system's phonological representation of F0 contours is based on Pierrehumbert's work. [“The Phonology and Phonetics of English Intonation,” Ph.D. dissertation, MIT (1980)]. Experiments on production motivate our treatment of pitch range and its interaction with choice of melody [Liberman and Pierrehumbert, “Intonational Invariance Under Changes in Pitch Range and Length,” Language Sound Structure (MIT Press, Cambridge, MA, 1984)]. A set of rules has been developed for aligning crucial features of the F0 contour with the speech segments. The program was tested both with LPC-encoded natural speech and with synthetic speech. A demonstration tape will be played at the meeting.
Synthesis of male and female voices: Control of voice quality based on source characteristics
Authors: Price P.J.
Published: October 2005

Voice quality can serve to distinguish individuals, social classes, and linguistic groups. It can mark the difference between male and female voices and between “normal” and pathological or synthetic voices. The goal of this study is a better understanding of these differences in order to synthesize more natural and more varied voices, in particular, female voices. Inverse filtering was used to retrieve the glottal waveform from isolated monosyllables produced by four women and four men. The resulting waveforms were examined at onset, at their midpoints, and at offset. Analysis included temporal and spectral measures. Significant differences were found between the male and female groups. The results of the analysis were used to develop a set of parameters to characterize the glottal cycle and its changes within an utterance. A tape will be played demonstrating the effects of varying these parameters.
Spanning the two cultures 1550–1750: Scientists and humanists seeking a quantitative account of contemporary musical scales
Authors: Lindley Mark
Published: October 2005

Many early scientists as well as humanists investigated those aspects of music which had traditionally been dealt with by music theory in the medieval quadrivium. Their agenda included a reckoning of tempered scales and to a lesser extent vocal intonation. The main challenge in the 1500's was to overcome certain old precepts, e.g.: that the major third (the ditonus) naturally had the monochord ratio (9:8)2 = (81:64); that the comma was some kind of “ultimate” interval; that “superparticular” intervals (n:n − 1) could not be divided into equal parts (because music theory must reckon with rational “multitudes” and not “magnitudes,” music being “sonorous number”). In the 1600's, regular temperament and various equal divisions of the octave containing nicely tempered consonances (19, 31, 43,…, as well as 12) were sorted out, often with logarithms; but then keyboard musicians took up irregular temperament, and for some time the sophisticated French-speaking scientists were so off the track that less urbane German organists outstripped them in developing an adequate account of the contemporary scale.
Relation of consonance theory to perception of chords
Authors: Roberts Linda A.
Published: October 2005

Theories of consonance make interesting predictions for chord perception. Some of these theories will be described, with emphasis on Helmholtz' (1863) views. According to Helmholtz, perceived consonance of chords depends on both beating harmonics and on combination tones. Helmholtz made some distinct predictions of how we perceive musical chords and some recent studies that support his theory are described. Results of one study [L. A. Roberts and M. L. Shaw, in press (1984)] demonstrated that same-different and similarity judgments of musical chords depend on the degree of consonance rather than relative interval sizes. These results were supported by findings based on direct consonance judgments of these chords [L. A. Roberts, J. Acoust. Soc. Am. Suppl. 1 73, S85 (1983)]. In another study, it was shown that consonance theory has bearing on perceptions of unfamiliar chords as well [L. A. Roberts and M. V. Mathews. J. Acoust. Soc. Am. 75, 952–959 (1984)]. Results of the latter study also demonstrated that listeners' intonation preferences are not simply based on some ideal intonation. Together, these results indicate that consonance theory has relevance for both traditional and new scale systems.
Voice timbre in solo and choir singing: Is there a difference
Authors: Rossing Thomas D. , Sundberg Johan
Published: October 2005

Professional and amateur bass/baritone singers with experience both as soloists and choir singers heard their own voice mixed with a piano accompaniment while they performed a specially composed solo selection. Then the piano accompaniment was replaced by a recording of a choir made in the bass section during rehearsal and the subjects were asked to sing the bass part. In each case the subject's voices were recorded, and comparisons were made between similar vowels sung at the same pitch and sound pressure level. The output spectra are noticeably greater in the region of the singer's formant [J. Sundberg, J. Acoust. Soc. Am. 55, 838 (1977)] in the solo singing mode. However, differences in the source spectra, obtained by inverse filtering of the formants, are not consistent among the singer subjects.
The frequency-domain grating
Authors: Hartmann W. M.
Published: October 2005

It is shown that a complex tone can be analyzed into its harmonics by adding to it a series of N − 1 similar complex tones with uniformly spaced fundamental frequencies. The harmonics appear in a time ordered sequence, and this is the basis of the computer music effect employed by composer Jean-Claude Risset. The mathematical derivation involves the summing of a geometric series to form a slowly varying amplitude factor. The effect is directly analogous to the analysis of white light into its constituent colors by a diffraction grating with N slits. The temporal ordering of the sequence does not depend upon the relative phases of the harmonics of the complex tone, nor is it altered if any harmonic(s) of the summed complex tones are successively attenuated by a common factor. The order can be changed by a progressive delay of any harmonic in the summed complex tones. Calculations are made to investigate the stability of the effect against nonuniform frequency spacing and modulation to explain why the effect does not normally appear in instrumental choruses.
Is real-ear attenuation at threshold a function of hearing level
Authors: Berger E. H.
Published: October 2005

In the course of measuring the real-ear attenuation at threshold (REAT) of experimenter-inserted E-A-R™ foam earplugs on 100 subjects, it was noted that attenuation was significantly correlated with hearing level (for normal listeners, HTL<20 dB) at test frequencies from 2–8 kHz. The relationship was most robust at 6 and 8 kHz. For hearing levels greater than 20 dB, attenuation appeared independent of hearing level. A hypothesis was developed to explain the relationship for the normal listeners, based upon the fact that the high-frequency attenuation of the plug was nearly bone-conduction-limited. The hypothesis suggested that the attenuation of a foam earplug modified to give substantially lower protection would not exhibit the same relationship. Data for such a plug were collected for 50 subjects, and indeed demonstrated no correlation between attenuation and hearing level. Implications of the experiment with regard to (a) hearing level requirements for hearing protector attenuation test subjects, (b) utilization of hearing impaired listeners to measure REAT at suprathreshold (with respect to normal listeners) sound levels, and (c) constancy of hearing protector attenuation as a function of sound level, are discussed.
Development of a prototype wave-vector-frequency spectral analysis system
Authors: Strawderman W. A.
Published: October 2005

By the mid-seventies, the utility of wave-vector-frequency spectral analysis as a diagnostic tool for structural acoustics had been theoretically and experimentally demonstrated. In anticipation of a continuing need for this diagnostic capability, we decided to develop a wave-vector-frequency spectral analysis system suitable for general application in spatially planar fields. This paper outlines the development of this prototype system, from theory to hardware. Emphasis is focused on the practical aspects of the development and the consequences of design decisions. Data from a simple, but practical, application of the system are presented and discussed.
Transient response of fluid-loaded planar vibrators with coupled modal coefficients
Authors: Obara Robert J. , Stepanishen Peter R.
Published: October 2005

An approach is presented to evaluate the transient pressure field and vibratory response of a finite fluid-loaded rectangular vibrator when excited by a wideband force and mounted in a planar baffle. The approach utilizes an in-vacuo eigenfunction expansion with time dependent coefficients to describe the velocity of the vibrator in the fluid. In contrast to the in-vacuo problem, the time dependent modal coefficients are coupled for the in-fluid problem. The coefficients are shown to be the solution of a coupled set of convolution integral equations which can be solved by marching forward in time. Near- and farfield pressures are expressed as a sum of convolution integrals involving the mode dependent velocity coefficients and spatially dependent impulse responses. Numerical results for several problems will be presented to illustrate the effects of fluid loading and modal coupling on the response of several vibrators. [Work supported by ONR.]
Continuous density hidden Markov models for speaker-independent recognition of isolated digits
Authors: Juang B. H. , Levinson S. E. , Rabiner L. R. , Sondhi M. M.
Published: October 2005

In a recent report [L. R. Rabiner, S. E. Levinson, and M. M. Sondhi, J. Acoust. Soc. Am. Suppl. 1 74, S17 (1983)], it was shown that the quantization error inherent in discrete-symbol hidden Markov models introduces a small but consistent degradation into speaker-independent isolated word recognition. Our present study demonstrates that this degradation can be eliminated by a number of techniques based on the theory of hidden Markov models with continuous density functions [L. R. Liporace, IEEE Trans. Inform. Theory IT-28, 729–34 (1982)]. By using these methods we have achieved 98% speaker independent recognition accuracy on isolated English digits. This rate is statistically indistinguishable from the best results obtained with the LPC/DTW system [L. R. Rabiner, S. E. Levinson, A. E. Rosenberg, and J. G. Wilpon, IEEE Trans. Acoust. Speech Signal Process. ASSP-27, 134=141 (1979)].
Continuous phonetic speech recognition
Authors: Makhoul John , Schwartz Richard , Chow Yen-Lu , Kimball Owen , Roucos Salim , Krasner Michael
Published: October 2005

We report on research to develop an automatic phonetic recognition system for continuous speech. The system is based on a hidden Markov model (HMM) representation of phonemes in context. That is, the model parameters depend on the left and fight phonetic contexts for each phoneme. The HMM structure is the same for all phonemes, but the model parameter values are different for different phonemes and different contexts. Automatic training procedures are used to adjust the model parameters, using a given set of training speech data. A major focus of our work is to maximize the robustness of the phonetic models given a limited set of training data. An initial system is now operational. Results of phonetic recognition accuracy in continuous speech will be presented. [Work supported by ARPA and monitored by ONR.]
Testing Bolinger's hypothesis of pre-accentual lengthening
Authors: Lea Wayne A.
Published: October 2005

Algorithms were developed for acoustic prosodic analysis (Lea and Clermont, Proc. 1984 ICASSP, paper 42.7), including voicing decisions, syllabic nuclei locations, phrase boundary detections, and stressed syllable locations. Other algorithms permit interactively entering listener decisions of syllabic durations, perceived stress levels, identifications of syllables (or other segments) that test specific hypotheses, and linguistic predictions. Using such algorithms and standard signal processing and statistics software, researchers can automatically and efficiently test prosodic hypotheses. Selected for initial application of such capabilities was Dwight Bolinger's unpublished hypothesis that any full syllable is longer (and of higher Fo) immediately before another full syllable than before a reduced syllable. Minimal-pair contrasts in accent sequences were embedded in 50 sentences spoken at fast and slow rates by each of two talkers, and 45 all-sonorant sentences spoken twice by one speaker. Syllabic durations were derived from listener markings and from automatic syllabification, then compared, and statistically analyzed, for verification of the hypothesis. Fundamental frequency values at centers of nuclei were also studied.
Interactions in a multichannel cochlear prosthesis
Authors: White M. W.
Published: October 2005

Multichannel electrical stimulation of the auditory nerve can generate complex interactions between the individual electrode channels. One type of interchannel interaction can occur when two or more channels are stimulated simultaneously. This type of interaction is described as an “electric field interaction” because it may be the result of electric field summation and cancellation within the cochlea. Loudness and threshold measures in human subjects, implanted with scala tympani electrode arrays, are significantly altered by reversing the stimulus polarity of one of two simultaneously stimulated channels. The nearer the two channels, the greater the change. In some subjects, significant interactions occur even for very widely spaced electrode channels. In other subjects, the interactions occur only over a restricted region [M. W. White, J. Gardi, and M. Metzenith, Arch. Otolaryngol. (in press)]. The differences between the subjects may be related to the quantity and distribution of surviving nerve. Preliminary data from four subjects suggest a relationship between the spatial “spread” of channel interaction and other psychophysical measures. [Work supported by NIH and Hearing Research, Inc.]
Preliminary evaluation of a wearable 16-channel electrotactile sensory aid for the deaf
Authors: Saunders Frank A. , Franklin Barbra
Published: October 2005

Initial assessments will be presented describing the performance characteristics, effectiveness of speech feature transmission, and user acceptance of a wearable, 16-channel electrotactile aid. Normal-hearing and deaf subjects, wearing the device, received a battery of subtests evaluating the discriminability of male-female voices, number of syllables, syllable stress, F1 movement, F2 movement, voicing, manner, and place of articulation. Factors affecting the design of experiments to test the efficacy of the device will be discussed, along with strategies for developing a comprehensive training curriculum for auditory awareness, speech reception in conjunction with lipreading, and speech production. [Work supported by Technology for the Handicapped, Inc., Tacticon Corporation, and SBIR/NIH.]
Dynamic responses of auditory neurons
Authors: Smith R. L. , Westerman L. A. , Frisina R. D.
Published: October 2005

Changes in sound intensity generally produce dynamic or transient changes in the responses of auditory neurons in addition to sustained or steady-state changes. These dynamic components appear to enhance the response to changes in intensity and potentially play an important role in intensity discrimination. The distinction between dynamic and steady-state responsiveness appears to be minimal in responses of hair cells, significant in responses of auditory nerve fibers, and still greater in some units of the cochlear nucleus. Results will be reviewed comparing responses in the three regions to step, ramp, and sinusoidally modulated changes in intensity. Underlying mechanisms will be discussed that may be responsible for producing the observed increases in both the magnitude and operating range of the dynamic response components.
Fiber optic sensors
Authors: Lagakos N. , Cole J. H. , Jarzynski J. , Giallorenzi T. G. , Bucaro J. A.
Published: October 2005

Over the past several years, many groups have been carrying out research in the area of fiber optic sensor technology, and many types of fiber sensors have been developed. Two of the most promising fiber optic sensor types will be discussed in this talk: phase-modulated single mode interferometric sensors and intensity modulated multimode fiber sensors. Phase-modulated sensors, which utilize an all fiber Mach-Zehnder interferometer, have high sensitivity and geometric versatility. Since the phase sensitivity of a fiber depends primarily on its coating, different fiber coatings can be utilized to detect various physical perturbations, such as acoustic, magnetic, electric, and thermal. The intensity modulated sensor is based on microbending losses which depend on the amplitude of the bend of a multimode fiber. The microbend sensor has moderate sensitivity, simple optical and electronic construction, and utilizes available multimode technology. If the fiber deformer mechanical design is tailored to a given external perturbation, various fields, such as acoustic, pressure, acceleration, and magnetic can be detected. In this talk, the transduction mechanism of each sensor type will be discussed and mechanical designs for detecting various fields will be presented.
Measurements of ultra-high coupling in magnetoelastic transduction materials
Authors: Hathaway K. B.
Published: October 2005

Extremely high magnetomechanical coupling factors (k > 0.92) are exhibited by a new class of rapidly solidified amorphous magnetic alloys. These very high couplings produce an enormous increase in sensitivity over conventional sensor materials and make possible a wide variety of new applications. New techniques have been developed to determine coupling factors close to unity with a great deal of precision. Measurements were performed using standard resonance techniques for bar shaped samples, but with two significant improvements in data analysis: (1) Multiple resonances are readily observed in these materials. Inclusion of a mechanical loss Q factor in the basic resonance function for the bar allows a good fit to the data over the entire multiresonance spectrum. Such a fit yields significantly more accurate coupling factors than can be obtained by the standard methods. (2) The problem of appropriate assumptions of magnetic boundary conditions, which is unique to metallic magnetoelastic materials and is due to the presence of eddy currents, is satisfactorily resolved by making use of the higher resonance data.
Modeling the azimuthal dependence of bistatic surface scattering
Authors: Kinney Wayne A.
Published: October 2005

A time-domain version of the facet-ensemble method [W. A. Kinney and C. S. Clay (to be published)] is used to explain some of the results obtained by Zornig [J. G. Zornig, J. Acoust. Soc. Am. 63, 758–767 (1978)] in an experiment on the azimuthal dependence of bistatic surface scattering. The experiment involved a 1.3-MHz pulse in water scattered off a wind-generated surface at a grazing angle of 30°. Although the surface wavelets were mostly long crested within the ensonified region, their direction of propagation varied and they were slightly crescent shaped. The complexity of the features required that the facets in the facet-ensemble method be appropriately oriented in azimuth for each receiver location to simulate the scattering process. The agreement between predicted and experimental data is very good. [Work supported by NORDA.]