bluuebunny/crossref_metadata_2025 huggingface dataset:
Formant extraction by local approximation of the speech spectrum
Authors: Fujii Kensaku , Ohga Juro
Published: October 2005

The present study proposes a new method of formant extraction that flattens the residual signal with reduced computational load. In this method, the formants are extracted in reducing order of the amplitude. Each formant is represented by an IIR filter of F(Z) = (1 − 2Qa cos θ0Z−1 + Qa2Z−2)/(1 − 2Qb cos θ0Z−1 + Qb2Z−2). This formula has three unknown quantities Qa, cos θ0, and Qb. They are estimated from quadratic equations on the speech spectrum. The coefficient cos θ0 corresponds to the formant frequency, which is obtained as the peak of the second-order curve defined by the maximum line spectrum A and the neighboring line spectra B and C. Here, Qb is related to the formant bandwidth which is given by the geometrical series average 1/(1 − Qb) = [1/(1 − eaT) + 1/(1 − ebT)]/2. Here, eaT and ebT are the damping terms in the impulse response of the second-order IIR filter. The relative magnitude of the frequency responses of this filter at the frequencies of the line spectra A and B is set to coincide with the relative amplitude of the line spectra A and B. The ebT is determined in a similar way by using the line spectra A and C. The coefficient Qa is the damping term of the frequency response which expresses the global shape of the spectrum. It can be estimated by using the line spectrum A and the maximum frequency component.
Acoustic scattering from surfaces generated by nonstationary, nonuniform wind fields
Authors: Novarini J. C. , Keiffer R. S. , Caruthers J. W. , Kerr G.
Published: October 2005

In an earlier paper [Novarini et al., J. Acoust. Soc. Am. Suppl. 1 83, S48 (1988)], the scattered acoustic fields resulting from fully developed (generated from the Pierson-Moskowitz spectrum) and fetch-limited (generated from the JONSWAP spectrum) 3-D wind-driven sea surfaces were compared. Within the context of the Helmholtz-Kirchhoff theory, it was demonstrated that there are significant acoustic differences between these two descriptions of the sea surface. In this meeting, the scattered acoustic field resulting from 3-D sea surfaces generated from nonstationary, nonuniform wind fields are considered. These sea surfaces are generated from a spectral representation which, given the wind field specified over an extended area, is provided by the Global/Regional Deep Water Wave Model (D. Ressio, “Program Wave,” Offshore & Costal Technologies Inc., Vicksburg, MS). This model has the ability to describe very realistic sea surfaces that include swell from distant storms and growing wind-driven seas. The scattered field resulting from such surfaces will be compared with results from sea surfaces generated by simple spectral descriptions which are functions of wind speed or wind speed and fetch. Preliminary results show strong differences in the coherence as well as the scattering strength over the entire frequency range under consideration (100–2000 Hz).
Scattering amplitudes for rough interface scattering
Authors: Berman David H.
Published: October 2005

A new exact expression for the scattering amplitudes in the case of fluid-solid interfaces is presented. This expression has the virtue of being manifestly reciprocal in the incoming and outgoing horizontal wavenumbers and is, therefore, a good starting point for making physically motivated approximations. The expression is a quadratic functional of various surface fields, but does not explicitly involve any Green's functions. In the limiting case of pressure release surfaces, the exact expression reduces to an expression derived by Dashen, and leads to a Kirchhoff-like approximation that agrees well with exact results over all scattering angles. Similar approximations for fluid-solid rough interfaces will be compared to exact calculations of the scattering amplitudes. [Work supported by NRI.]
The Valsan 727-200RE—A FAR part 36, stage 3, re-engining program
Authors: Marsh Alan H.
Published: October 2005

Boeing 727-200 airplanes powered by three Pratt & Whitney JT8D low-bypass ratio turbofan engines contribute much of the noise annoyance around most of the world's airports because of the high level of jet exhaust noise produced during departures and because of the high frequency of operations by the large number of 727-200 airplanes that are in service. Valsan Partners is managing a program to remove the two fuselage-mounted JT8D engines from 727-200 airplanes and replace them with newer higher-bypass-ratio JT8D-200 Series engines installed within new nacelles. A special internal-mixer nozzle is installed within a new acoustically treated tailpipe on the existing JT8D center engine. Re-engined 727-200RE airplanes produce significantly less noise during departures and also during landings. Cruise fuel consumption is also improved. Noise certification tests demonstrated compliance with the stage 3 noise level requirements of FAR part 36 as well as the chapter 3 requirements of ICAO Annex 16. The re-engining modification and test results will be described.
Finite element approaches to acoustic fields perturbed by a cylindrical tank
Authors: Hiramatsu Tsutomu , Komura Yoshiaki , Shintani Masanori
Published: October 2005

The finite element method is applied to the acoustic field around a cylindrical tank in order to analyze the diffraction effects caused by the obstacle. A method used in this study is the so-called hybrid-type finite element method. The cylindrical tank situated in a finite domain perturbs the acoustic field radiated from a point source. The finite domain is divided into many finite elements, which are combined with the infinite elements so as to satisfy the continuity of the flux flow on the interelement boundary using an arbitrary interpolation function. The approach is compared with an experimental result which was obtained for the acoustic field around the water tank at a thermal power plant. From these studies, it can be seen that the hybrid technique using the combined finite and infinite elements is a useful method of analyzing and predicting the noise field around obstacles like water tanks.
A system for speech synthesis from Japanese orthographic text
Authors: Kawai Hisashi , Hirose Kcikichi , Fujisaki Hiroya
Published: October 2005

A system has been developed for speech synthesis from Japanese orthographic text of Japanese. The system consists of four processing stages. The linguistic processing stage utilizes natural language processing techniques for extracting lexical, syntactic, semantic, and discourse information from each paragraph of the input text. The phonetic processing stage utilizes this information to derive a string of segmental and prosedie symbols for the entire paragraph. The acoustic processing stage generates time-varying patterns of parameters from these symbols to control the final stage, which is a formant-type synthesizer. The Fujisaki-Ljungqvist model is adopted for the excitation of the voiced sounds [Proc. ICASSP 86, 1605–1608 (1986)], and its fundamental frequency is controlled by a model of F0 contour generation [H. Fujisaki and K. Hirose, J. Acoust. Soc. Jpn. (E) 5, 233–242 (1984)]. The segmental features, on the other hand, are synthesized by concatenating pole-zero frequency patterns prestored for each syllable. The validity of the system, especially of the prosodic feature synthesis, was confirmed by the naturalness of the accent and intonation of the synthesized speech. [Work supported by Grant-in-Aid for Scientific Research on Priority Areas from Ministry of Education, Science and Culture of Japan, No. 63608002.]
Text-to-speech system for English and Japanese
Authors: Matsui Kenji , Hara Noriyo , Kitano Masaaki , Javkin Hector , Hate Kazue , Wakita Hisashi
Published: October 2005

A real-time text-to-speech system for English and Japanese has been developed. This system consists of a language processing module, a phonetic acoustic processing module, and a synthesis module. Full general English and Japanese sentences can be converted to speech. The Japanese software and English software are independent except for the synthesis module. The features of this system are as follows. (1) The synthesis module is a phoneme-based cascade-parallel formant synthesizer with high observed intelligibility (73.5% for the 119 Japanese monosyllables). (2) This system has a 3000-morphene English dictionary and 40 000-word Japanese dictionary with a high-speed search algorithm. (3) A large speech database was collected for the development of Japanese prosody rules. (4) For the precise control of pitch contour, the Fujisaki model was adopted. (5) One of the two systems developed can stand alone; the other requires a personal computer with a high-speed DSP board. (6) In the development of this system, some powerful interactive tools have also been developed for varying speech parameters in real time.
Shipboard rest, sleep, and seasickness: Evidence of interaction
Authors: Bittner A. C. , Guignard J. C.
Published: October 2005

Sea-keeping trials of a United States Coast Guard cutter included a human factors engineering evaluation having particular regard to problems of seasickness in certain work stations. Some motion sickness incidence and related data, and the derivation of human factors engineering principles therefrom, were reported in 1985 [A. C. Bittner, Jr. and J. C. Guignard, Nav. Eng. J. 97 (4), 207–213; 97 (5), 107–111]. Further analysis [A. C. Bittner, Jr. and J. C. Guignard, in Trends in Ergonomics/Human Factors V, edited by F. Aghazadeh (Elsevier, New York, 1988), pp. 529–539] showed that, contrary to traditional assumptions, seasickness is characterized by at least two functionally independent factors, identified as F1 (symptomatic general motion illness) and F2 (retching-vomiting). It was shown that the validity of future sea trials requires: (a) multiple-score scaling of motion sickness (e.g., using F1 and F2); (b) control of subject crew activity and movements about the ship during periods of observation; and (c) avoidance of steaming patterns that induce extraneous carryover effects. The present paper draws attention to an incidental finding concerning the influence of brief sleep and rest (lying down in the bunk) upon motion sickness susceptibility and recovery in the shipboard environment. Analyses of questionnaire data including amount and subjective quality of sleep showed that these were affected by the passage of time in an adverse (nauseogenic) motion environment, as has been observed generally in naval operations. An apparent ameliorative effect of unscheduled rest periods and naps upon susceptibility to adverse motion effects was incidentally detected and is commented upon in this report. [Observations made while the authors were at the Naval Biodynamics Laboratory, New Orleans.]
Noise measurements using a portable instrument controller
Authors: Brittain Frank H. , Gmerek Mark M.
Published: October 2005

Sound-level meters are very useful and flexible for brief measurements in the hands of an experienced operator, but measurements for periods of an hour to several days are often not practical. While community noise monitors are useful for measurements of longer duration, they typically lack the flexibility for many specialized measurements. The desired flexibility is obtained by interfacing a sound-level meter to a portable computer. The two are connected through an RS-232 interface. The computer is programmed to control and collect data from the sound-level meter. The data are stored and then formatted and printed or transferred to a larger computer. The need to write data by hand is eliminated. Several examples are given to illustrate the technique.
Compensation for phase mismatch in sound intensity measurements
Authors: Ebbitt Gordon
Published: October 2005

Sound intensity measurements place stringent requirements on the phase matching of the measuring apparatus (including the transducers). In the past this has nearly always been accomplished by relying on the manufacturer to provide phase matched instrumentation and transducers. These days more and more processing power is being built into analyzers. This has opened the door to the possibility of compensating for the phase mismatch between the two measurement channels. The procedure consists of first measuring the phase mismatch between the channels and then correcting for it as a postprocessing operation. In practice, this correction can be included as part of the calculations required to compute the intensity from the two input signals. While there are clear benefits to this technique (matched transducers are not required), there are a number of practical problems. Chief among these is getting an accurate measurement of the phase difference between the channels. The benefits and difficulties associated with this technique will be discussed and measurements made with a phase-matched system will be compared with those made using phase compensation.
An application of the Wigner distribution to loudspeaker evaluation
Authors: Nakamura Kazuhiro , Miyagawa Takeshi
Published: October 2005

The problem in the sound quality evaluation of loudspeakers using their frequency characteristics is that the sound quality difference perceived by human ears is not distinctly represented in the frequency characteristics. To solve this problem, an evaluation method using the Wigner distribution that is derived from the impulse responses of loudspeakers is employed. The effectiveness of this evaluation method in comparison with conventional frequency characteristics was examined using two loudspeakers with clearly different sound qualities. While there was no noticeable relationship between the frequency characteristics of the loudspeakers and listener's subjective impressions, the Wigner distributions showed a recognizable correlation with listener's subjective impressions. Thus it can be concluded that the Wigner distribution can represent the sound qualities of loudspeakers more effectively than conventional frequency characteristics.
Simulated focusing/beamforming with a phased array of nonlinear sources
Authors: McDonald B. Edward , Kuperman W. A.
Published: October 2005

Numerical simulations have been carried out on ocean acoustic scales for a simple array of three nonlinear sources. The nonlinear progressive wave equation (NPE) model [B. E. McDonald and W. A. Kuperman, J. Acoust. Soc. Am. 81, 1406–1417 (1987)] has been used to calculate the evolution of initially nonintersecting underwater shocks of Mach number 0.05. Three impulsive sources equally spaced on a vertical line are fired with top and bottom elements together, and the middle one phase delayed. The resulting wave produces a weak “focus” at the intersection of the three expanding spherical waves. Linear and nonlinear calculations are compared. In the linear calculation, the individual waves retain their identity past the focal range. In the nonlinear calculation, however, Mach stems form as individual shocks merge. A directed beam is formed, with the three shocks joined irreversibly into a single shock. Nonlinear processes are seen to increase peak pressures in the farfield as compared to a linear solution.
Static pressure differential measurements in a thermoacoustic engine
Authors: Hofler T.
Published: October 2005

High-amplitude standing waves interact with the solid boundaries of a resonator in the region of the acoustic boundary layer. This interaction causes thermoacoustic heat transport as well as a partial rectification of the oscillatory velocity into a steady flow, known as acoustic streaming. The traditional analysis of streaming assumes that the bore of the resonator duct is much larger than the boundary layer and that there is no net flow through the closed vessel, which requires that a counter flow at the center opposes the streaming flow at the walls. However, the “stack” of a thermoacoustic engine has channels that are only two to four boundary layers wide. Any streaming counter flow in these tiny channels will generate substantial viscous shear that must be sustained by a static pressure gradient. The results of measurements of static pressure differences across the “stack” will be presented.
Vibrotactile perception thresholds in hands occupationally exposed to vibration: Interpretation of sensorineural loss
Authors: Piercy J. E. , Brammer A. J. , Nakamura H. , Nohara S. , Auger P. L. , Haines A. T.
Published: October 2005

The tactile performance of the hand is now known to be critically dependent on neural activity in one population of slow-adapting (SAI) and two populations of fast-adapting (FAI, FAII) mechanoreceptors. A technique has recently been developed to establish the sensitivity of these receptor populations at the fingertip from vibrational perception thresholds determined psychophysically with sinusoidal stimulation [A. J. Brammer et al., J. Hand Surg. 12A, 870–875 (1987)]. A comparison of data from 98 vibration-exposed and 34 normal hands, all screened to exclude confounding factors, has revealed two patterns of abnormal threshold elevation. The first, involving similarly elevated thresholds at all frequencies (2–200 Hz), is indicative of sensorineural losses of similar magnitude in each type of receptor-nerve system, and so is suggestive of peripheral nerve degeneration. The second pattern involves elevated thresholds at frequencies mediated only by one or sometimes two receptor types (commonly SAI and/or FAII). A mechanoreceptor-specific mechanism is unlikely to occur within a nerve trunk, and so this frequency-dependent pattern is suggestive of selective damage to the nerve endings.
Evaluation of a digital hearing aid and fitting system
Authors: O'Connell Michael P. , Skinner Margaret W. , Engebretson A. Maynard , Miller James D. , Pascoe David P.
Published: October 2005

The results of the evaluation of a body-worn, programmable digital hearing aid in short-term field studies of 3 days to 3 weeks on six hearing-impaired subjects will be presented. The hearing aid was programmed to implement a four-channel instantaneous compression hearing aid model. Frequency selective channel gains and maximum power output settings of the hearing aid were controlled by a computer-based, hearing-aid evaluation system. This system integrated real-ear measurements and hearing aid control capabilities to achieve several different real-ear frequency/gain characteristics for each subject. The desired real-ear gains were calculated based on a prescriptive fitting method applied to hearing threshold and most comfortable listening levels. The performance of the device with the user-preferred real-ear gain curve was evaluated using speech intelligibility testing and HAPI questionnaires completed after using the digital hearing aid. The speech intelligibility testing consisted of a computer automated test using the CUNY nonsense syllables for seven different speech levels under three noise conditions (21 total conditions). All subjects were current hearing aid users and completed identical intelligibility tests and HAPI questionnaires with their own hearing aids. [Work supported by the VA, the 3M Corporation, and NASA.]
Lateralization predictions for high-frequency binaural stimuli
Authors: Stern Richard M. , Shear Glenn D. , Zeppenfeld Torsten
Published: October 2005

The position-variable model [R. M. Stern, Jr. and H. S. Colburn, J. Acoust. Soc. Am. 64, 127–140 (1978); G. D. Shear and R. M. Stern, J. Acoust. Soc. Am. Suppl. 1 81, S27 (1987)] is extended to describe the subjective lateral position of amplitude-modulated tones and bandpass noise, as well as other complex stimuli that are presented within spectral regions at which the binaural system appears to be unable to make use of cycle-by-cycle interaural temporal differences. Predictions of the model are based on the centroid of the cross correlation of the hypothetical auditory-nerve response to the stimuli, which is either calculated using analytical techniques or simulated numerically. The model of auditory-nerve activity, which is typically used to describe the response to stimuli at lower frequencies, also extracts envelopes of higher frequency stimuli, as discussed previously by Colburn and Esquissaud. This information appears to be useful in predicting the lateral position of such high-frequency stimuli. Preliminary results indicate that the model is able to describe most of the ways in which the laterality of high-frequency binaural stimuli with low-frequency envelopes depends on modulation frequency, carrier frequency, and other stimulus parameters. The model also predicts the relative salience of interaural temporal cues at different frequencies. [Work supported by NSF.]
A new approach to English text-to-phoneme conversion using Delta, version 2
Authors: McCormick Susan , Hertz Susan R.
Published: October 2005

The Delta System, version 2 is being used to develop a new set of text-to-speech rules for English. This paper focuses on the text-to-phoneme portion of these rules. These text-to-phoneme rules are based on an earlier rule set developed by Hertz [J. Acoust. Soc. Am. Suppl. 1 69, S83 (1981)], but are much more elegant and complete, taking advantage of Delta's flexible facilities for building multilevel utterance representations and matching patterns against them. The rules convert an input text into an utterance representation containing the lexical, morphological, and phonological structure of the utterance. Sophisticated algorithms are used for identifying prefixes (productive and nonproductive ones), suffixes, and roots (including the roots of compound words), for predicting lexical stress, for assigning grammatical categories, and for predicting pronunciations. Portions of these algorithms will be described, with a focus on how the Delta programming language allows them to be expressed succinctly and clearly. The rules will be available for demonstration.
The effects of perceptual learning on capacity demands for recognizing synthetic speech
Authors: Lee Lisa , Nusbaum Howard C.
Published: October 2005

Previous studies have demonstrated that recognition of synthetic speech improves with moderate amounts of training. Three experiments were performed to determine whether the effects of training are due to changes in the attentional capacity required to process the speech. Attentional capacity available for recognizing speech was manipulated using a recall task with different length lists of visually presented numbers. In two experiments, before and after training, the primary task required subjects either to recall lists of words or to perform speeded word recognition for synthetic speech. Whereas training improved performance with synthetic speech, decreasing available capacity impaired performance. In a third experiment, the intelligibility of the speech was manipulated by using speech from different text-to-speech systems. The results of these experiments suggest that neither training nor intelligibility affects the amount of attentional capacity required to recognize synthetic speech. Rather, as perceptual learning occurs, listeners begin to allocate their attentional resources more efficiently to the processing of synthetic speech. [Work supported by AFOSR and BRSG.]
Comparing a magnitude estimation technique and a pair comparisons technique for use in assessing quality of text-to-speech synthesis systems
Authors: Pavlovic Chaslav , Sorin Christel , Roumiguiere Jean Pierre , Lucas Jean Pierre
Published: October 2005

This study was designed to compare a magnitude estimation procedure and pair comparisons procedure for use in scaling quality of text-to-speech synthesis systems. To this goal, the psychophysical scale values that result from the magnitude estimation procedure were compared to the values obtained from a pair comparisons procedure. Four different synthesis systems and three different prosody rules in all possible combinations were scaled using a magnitude estimation technique. To obtain the pair comparisons scale values, the results reported by Lucas et al. [CNET report NT/LAA/TCS/375, Tom I, Lannion, France (1988)] for the same systems were used. They were reanalyzed for this purpose. The data indicate a good general agreement between the results of the two experiments. Thus the two methods seem to validate each other. The relative efficiency of the magnitude estimation technique is greater than that of the pair comparisons technique.
Time-evolving characteristics of midfrequency ambient noise
Authors: Hodgkiss W. S. , Fisher F. H.
Published: October 2005

Measurements have been made of the ambient noise field between 25 and 300 Hz with vertical arrays at 32° N (124° W, 136° W, and 150° W). Due to substantial differences in weather at the stations, these measurements provide an opportunity to obseve the effect of weather on the vertical distribution of ambient noise and how this distribution evolves with time. Array time series 3–4 h in length at each station have been analyzed. The time-evolving characteristics of ambient noise vertical directionality at each station will be compared. Specifically, the fluctuations of beam power for beams pointing near the horizontal will be compared with those pointing both upward and downward. [Work supported by ONR.]
Scattering inversion using the ramp response
Authors: Cook Bill D.
Published: October 2005

The objective of inversion is to have a result that is interpretable. Impulse and frequency techniques suggest that all could be known if one had the privilege of using all frequencies and could observe the scattering from a multitude of directions. Realistically, one has a limited bandwidth of frequencies and often only views from a few directions. At the sacrifice of inversion detail, one can use the ramp response that requires a much smaller bandwidth. Shape, size, and volume information can still be extracted from a few views. Experimental results at audible and ultrasonic frequencies demonstrate the viability of this technique.
Experimental validation of a model for predicting the impedance of multilayer systems by the transfer matrix method
Authors: Woodcock R. , Hodgson M.
Published: October 2005

As has been demonstrated by various authors, transmission line theory is a powerful technique for studying the sound transmission of multilayer systems with infinite lateral extent. This model has been extended to the study of the surface impedance of such systems. It is shown how the transfer matricies are derived from fundamental acoustic equations. These equations contain information on the conventions adopted concerning the assumed time dependance and orientation axis. It is shown that it is convenient to account for this during the juxtaposition of the quadrapoles in cascade, this modeling the superposition of the various layers. An experimental validation of this model has been undertaken on two types of mineral wool, placed in different multilayer configurations in an impedance tube, using the Chung and Blaser transfer function technique. The measured impedances are in excellent agreement with those predicted using the Delany and Bazley impedance model. Finally, it is shown that the model is only valid if each layer of the system is homogeneous.
Forced-choice discrimination of complex tones
Authors: Cohen Annabel J.
Published: October 2005

The study is part of a research program examining the effects of small integer frequency ratio relations on memory for unfamiliar (microtonal) sets of tones. Previous work indicated the benefit of both successive and simultaneous small integer context in an absolute judgment task. In order to determine whether such benefits were cognitive as opposed to sensory in origin, a quasifixed-standard two-alternative forced-choice discrimination task was conducted that had lower demands on memory than the absolute judgment task but could potentially lead to the same context effects. Discrimination of nine complex tones of a microtonal scale ranging from 545 to 636 Hz (tones separated and discriminanda differing by one-third semitone) was tested under simultaneous and successive context conditions. Preliminary data indicate that musically trained listeners, compared to untrained, showed a small but significant benefit from a dichotic simultaneous bass tone having the ratio 2:3 with one central tone of the nine-tone scale but not from a dichotic tone having a Unison relation with the central tone. Successive (melodic) context effects were less apparent in this discrimination study than in the absolute judgment experiments.
Effects of phase changes in low-numbered harmonics on formant frequency matches
Authors: McKeown J. Denis , Darwin Christopher J.
Published: October 2005

Phase changes to single harmonics (of a 125-Hz fundamental) in the first formant (F1) region can change the phoneme boundary between /I/ and /e/ along an F1 continuum in an identification task [C. J. Darwin and R. B. Gardner, J. Acoust. Soc. Am. 79, 838–845 (1986)]. Physiological correlates of this effect have been noted in the ALSR response of guinea-pig auditory nerve fibers [A. R. Palmer et al., in The Psychophysics of Speech Perception, edited by M. E. H. Schouten (Nijhoff, Dordrecht, 1987)] and in an auditory model with synchrony suppression. The present work demonstrates similar phase effects using a matching paradigm. Subjects matched sounds along a continuum that differed in F1 frequency and could have additional formants. The target sounds (fundamental = 125 Hz; F1 = 440 Hz) were taken from the matching continuum but had the phase of a single harmonic changed. Systematic and consistent differences in the matched F1 value were found as a function of phase. Although the starting phase of the components influenced the size of the matched difference in F1, the difference was still present for randomized starting phases. The effect required at least three frequency components and varied with fundamental frequency.
Underwater sound from whitecaps at sea
Authors: Hollett Reginald D.
Published: October 2005

Observations are presented of the underwater sound from whitecaps at sea. A vertical array of hydrophones was used to form an end-fire beam towards the sea surface, receiving the sound from a patch of the surface above the array. At the same time, the surface patch was visually monitored using a video-recording facility. The time series of the sound from the patch was spectrally analyzed at intervals of a fraction of a second, over the band from 187.5 to 1500 Hz. The resulting time variation of the sound spectrum shows distinct bursts of sound. The strong bursts coincide with visual detections of whitecaps above the array. The burst of sound from a whitecap lasts for several seconds and is seen to be associated with the breaking phase. The sound spectrum during the breaking phase has notable low-frequency content, the spectrum appearing almost flat from 187.5 Hz up to about 1000 Hz.
Pressure-flow relationships in the larynx
Authors: Smith Marshall E. , Berke Gerald S.
Published: October 2005

Measurements of air pressure and flow were made using an in vivo canine model of the larynx. Subglottic and supraglottic pressures at varying flow rates were taken during phonation, induced by laryngeal nerve stimulation. Results shows that, during constant vocal fold stiffness, subglottic pressure did not rise significantly with increased airflow. Increasing flow rate was associated with an increase in the open quotient as measured glottographically. Data from this experiment were compared with the theoretical two-mass model of the larynx. Model parameters were adjusted to approximate the canine vocal folds and the in vivo experimental pressure-flow relationships were simulated. The larynx in the in uivo canine model exhibits flow-dependent decreasing impedance during phonation in that increasing flow rate is not accompanied by increases in subglottal and transglottal pressure. [Work supported by VA Technical Merit Review Grant.]
Alterations in lower lip and jaw kinematics of stutterers and nonstutterers as a function of speech rate
Authors: De Nil Luc F. , Abbs James H.
Published: October 2005

Previous articulatory kinematic studies of stutterers and normally fluent speakers during perceptually fluent speech have revealed subtle, but possibly significant, between-group differences. Such observations lend support to the hypothesis that stutterers and nonstutterers somehow differ in their basic motor control ability for speech. Comparing the influence of speech rate on stutterers' and nonstutterers' articulatory kinematics would provide a further test for this hypothesis. To this end, seven adult stutterers and seven normal speakers produced repetition sequences of the syllable /bae/ at six different rates. Multiple kinematic measures of the lower lip and jaw movements during oral closure were applied. Peak velocity and displacement were observed to differ to subtle ways between the two groups, with greater differences apparent at faster speaking rates. Stutterers also differed from normal speakers in timing of peak velocities at various speech rates. The results provide additional information on the possible role of motor control processes in stuttering. [Work supported by NIH Grants NS-13274 and NS-16373.]
Phonetic product estimates of infant phonetic development
Authors: Bauer Harold R. , Robb Michael P.
Published: October 2005

The phonetic development of six children was estimated across the first 2 years of life. Using a phonetic product measurement scheme (Bauer, 1988), the children's monthly vocalization samples were analyzed for phonetic contrast. The number of observed phones in five different consonantlike (place of articulation) and three vowellike (tongue advancement) categories was multiplied to yield a phonetic product for each vocalization. The results were summarized and plotted for the six children's monthly vocalization samples with microcomputer software. The results indicated a general increase in the phonetic product of vocalizations as a function of chronological age. Individual trends for each child are discussed in relationship to the general function in our search for general measures spanning the prelinguistic-to-linguistic period.
Vibration modes and tuning of xylophone bars
Authors: Bork Ingolf
Published: October 2005

The characteristic quality of xylophone sound depends both on the inharmonic structure of its spectrum of partials and on the order in which the individual partials decay. Subjective tests have shown that a systematic tuning of the first three partials improves the quality. A procedure for the practical realization of such a tuning is described. With the help of modal analysis, it can be shown that when torsional vibration modes as well as bending modes contribute to sound radiation, the tuning becomes more complicated.
Pitch distribution in bell chimes of the Chinese Bronze Age
Authors: von Falkenhausen Lothar
Published: October 2005

Pitch measurements on a number of archaeologically discovered graduated sets of two-pitch bells [cf. M. Chengyuan, Chin. Music 3(4), 81–86 (1980); 4(1), 18–20 (1981); and 4(2), 31–36 (1981); S. Shen, Sci. Am., 104–110 (April 1987); T. D. Rossing et el., J. Acoust. Soc. Am. 81, 369–373 (1988)] from the Shang (ca. 1500–1050 BC) and Zhou (ca. 1050–256 BC) dynasties were compared, and the patterns of pitch distribution from octave to octave were scrutinized. Over time, an increasing mastery of the technology of manufacturing two-pitch bells can be observed, especially regarding the ability of manipulating the two fundamentals on each bell in such a way that both could be used in music making. From Western Zhou (ca. 1050–771 BC) onward, the pitch distributions of each chime show a basic division into a lower and a higher register, the former at first manifesting a greater density of pitches than the latter. In the higher register, the Western Zhou four-tone (mi-sol/la-do) pattern was later gradually enriched with additional pitches. The ever denser distribution of pitches seems to reflect the development of bell chimes from a musical instrument of merely accompanatory nature to one on which melodies could be played in full, sometimes in more than one tonality. Caution must be used, however, in generalizing from these phenomena on the development of tonal scales in China.
Measuring the characteristic properties of mallets
Authors: Bork Ingolf
Published: October 2005

A measurement procedure is described that makes it possible to characterize different types of mallets for percussion instruments. From spectral analysis of the impulsive force generated by the mallet head, a frequency range is obtained over which each mallet optimally excites vibrations. The dependence of this optimum range on the strength of the blow constitutes a quality criterion that is also suitable for characterizing piano hammers.
Time-domain analysis of auditory-nerve fiber firing rates
Authors: Secker-Walker Hugh E. , Searle Campbell L.
Published: October 2005

A time-domain analysis of firing-rate data from over 200 fibers from the auditory nerve of cat has been used to estimate the formants of synthetic speech syllables. Distinct groups of fibers are apparent in the neurograms of the firing-rate responses. The intervals between peaks in the firing-rates of the fibers in each group are very similar, and reflect the period of the formant that dominates the group's response. Analysis of these intervals confirms that they correspond directly to the formant periods. It is concluded that the cochlear filters have much shorter impulse responses than the formants to which they respond. The time-domain analysis tracks the changes of lower frequency formants with more precision than previous analyses of the same neural data [M. I. Miller and M. B. Sachs, J. Acoust. Soc. Am. 74, 502–517 (1983); S. A. Shamma, J. Acoust. Soc. Am. 78, 1622–1632 (1985)]. The direct representation of the formant period in the time domain is contrasted with the diffuse spectral representation of the formant, the dependence of spectral peaks on nonformant parameters, and the distortion of the spectrum by physiological nonlinearities.
Frequency and duration discrimination by patients with central auditory pathway lesions
Authors: Thompson Mary Ellen , Abel Sharon M.
Published: October 2005

The performance of three groups of patients with central auditory pathology including subjects with lesions localized to the right temporal lobe, left temporal lobe, and eighth nerve was compared using three psychophysical tasks. These included detection, frequency discrimination, and duration discrimination. Speech perception was measured using the Four Alternative Auditory Feature Test. The patients were compared to a group of age-matched normal control subjects and hospitalized control subjects. Except for patients with acoustic neuromas, all subjects had normal hearing acuity. Performance deficits relative to normal were dependent on site of brain lesion, verified by CT and/or MRI scans. The relationship between the processing of frequency and duration and speech perception will be discussed. [Work supported by the Medical Research Council and National Sciences and Engineering Research Council.]
Auditory-nerve representation of communication signals in background noise
Authors: Schwartz Joshua J. , Ferragamo Michael , Simmons Andrea Megela
Published: October 2005

Anuran auditory-nerve fibers phase-lock to the periodicity of complex acoustic signals, even though many of these fibers do not synchronize to pure tones at their best frequencies (BFs). In this experiment, the effects of background masking noise on peripheral coding of both spectral and temporal structure of the male bullfrog's species-specific advertisement call was examined. Synthetic advertisement calls with 21 components (periodicity of 100 Hz) were digitally constructed and presented via a closed-field system against different levels of noise. Plots of average discharge rate versus BF show peaks corresponding to the periodicity and low-frequency spectral energy in the signal when presented at noise levels near threshold. These profiles change substantially in the presence of high levels of background noise. Plots of average localized synchronized rate (ALSR) against BF are much less influenced by noise. The data suggest that temporal coding of this natural call is more resistant to noise masking than is a representation based solely on discharge rate. [Work supported by NIH.]
Alternate-state otoacoustic emissions as a basis for episodic intermittent tinnitus
Authors: Burns Edward M.
Published: October 2005

The correlational evidence is so overwhelming that the vast majority of occurrences of severe, continuously present, tinnitus are not associated with spontaneous otoacoustic emissions (SOAEs). Similarly, subjects with SOAEs are rarely subjectively aware of their presence, a fact that is not incompatible with the literature on the adaptation of low-level sinusoidal stimuli. However, OAE researchers have often speculated that unstable SOAEs could be the basis of intermittent tinnitus. This paper reports on measurements made on one subject during an episode of intermittent tinnitus (reported by the subject to be audible in normal circumstances, loud, and extremely annoying), which demonstrate fairly conclusively that the tinnitus in this case is associated with alternate-state SOAEs.
Spontaneous otoacoustic emissions in Rana catesbiana, the American bullfrog
Authors: Gennosa T. J. , Voigt H. F. , Burkard R.
Published: October 2005

Temperature-dependent spontaneous otoacoustic emissions (SOAEs) are reported for the first time in Rana catesbiana, the American bullfrog. A low-noise acoustic measurement system was developed for recording SOAEs in a readily available species of frog. SOAEs were detected in one of five anesthetized frogs during late autumn. The frequency of the emission was temperature dependent, increasing from 1612–1762 Hz, with a change in body temperature from 22.1°–25.3°C. The SOAE level ranged from −3 to +4 dB SPL and was uncorrelated with temperature over the range studied. SOAEs have been reported in Rana esculenta [A. R. Palmer and J. P. Wilson, J. Physiol. 324, 66P (1981)] and Rana temporaria [Wilson et al., in Auditory Frequency Selectivity, edited by B. C. J. Moore and R. D. Patterson (Plenum, New York, 1986), pp. 39–46]. The emissions in these other species are also temperature dependent [P. van Dijk and H. P. Wit, J. Acoust. Soc. Am. 82, 2147–2150 (1987)]. Experiments are recommended to explore the possibility that the presence and incidence of SOAEs are seasonally dependent in this species. [Work supported in part by NIH.]
Comodulation masking release for three types of modulator as a function of modulation rate
Authors: Carlyon Robert P. , Buus Søren , Florentine Mary
Published: October 2005

To investigate the dependence of comodulation masking release (CMR) on the type and frequency of modulator, thresholds were measured for a 4-kHz tone masked by modulated and unmodulated noises. The maskers were a 400-Hz wideband of noise centered on 4 kHz (NOCUE), the same noise with a 2700-Hz low-pass noise added (LPCUE), and a wideband noise with a passband between 3 and 6 kHz (WBCUE). Each modulated masker was presented with several modulation frequencies, fes and three types of modulators: a square wave, a sinewave, and a low-pass noise. Thresholds were lower in the presence than in the absence of modulation for all modulator types, except when fe was high. The modulated-unmodulated difference (MUD) decreased monotonically with increasing fe in all conditions. For any given fe, MUDs were greater for square-wave and low-pass-noise modulation than for sinusoidal modulation. The CMR, defined as the difference between the MUD obtained in the NOCUE condition and that obtained in each cue condition, did not vary systematically with fe over the range studied. CMR was smaller for sinusoidal modulation than for either square-wave or low-pass-noise modulation. Although there was some variability across listeners, low-pass-noise modulators that produced an fe of between 32 and 100 Hz resulted in substantial (5–9 dB) CMRs for all three listeners. [Work supported by the Piggott-Wernher Fellowship and NIH.]
Effects of homophasic and antiphasic stimulation on the binaurally evoked auditory brain-stem responses
Authors: Rawool Vishakha W. , Ballachanda Bopanna B.
Published: October 2005

This study investigated the effects of homophasic and antiphasic stimulation on the binaurally evoked auditory brain-stem responses (ABRs). Ten normal subjects were presented with two in-phase (binaural condensation and binaural rarefaction) and two out-of-phase (left ear condensation, right ear rarefaction and right ear condensation, left ear rarefaction) stimulus configurations. The stimuli were 100-μs clicks delivered at 47 dB SL (repetition rate 5.6/s). The ABRs were picked up between high forehead (active) and left earlobe (reference), the nasion served as ground. The earlier of the latencies obtained with homophasic stimulation was compared with the earlier of the latencies obtained with antiphasic stimulation. A subject-by-treatment analyses on the peaks (P) and troughs (N) of each component revealed significantly earlier latencies for the antiphasic condition for components IIN (p < 0.05), IIIP (p < 0.025), VIP (p < 0.01) and VIN (p < 0.025). Similar trends were apparent in components IVP (p < 0.1), IVN (p < 0.2), and VP (p < 0.1). No differences were observed for components IP, IN, IIP, and VN.
Excitation of finite and infinite length cylindrical shells by internal sound fields
Authors: Ward William C.
Published: October 2005

Cylindrical shells, such as industrial piping system components, are efficiently excited by broadband internal noise at discrete frequencies below the ring frequency. Two types of excitation are recognized: finite length pipe resonances that are visible in short pipes, and coincident excitation that has been studied for long or anechoically terminated pipes. Both mechanisms occur to varying degrees in pipes of any length, and both require that the acoustic and structural wavenumbers be closely (or exactly) matched. Because ducts possess an infinite number of potential coincidence frequencies, coincidence transmission (i.e., precisely matched wavenumbers) is dominant in pipes. In the limit for long shells, both mechanisms are damping controlled and approach the same levels. Experimental data for short and intermediate shells show that coincidence transmission (“infinite shell”) can be the sole cause of vibration over wide frequency bands where the density of resonant modes is low. A simple theory is used to predict the frequency and amplitude of response peaks caused by both mechanisms. [Work supported by NSF.]
Resonance characteristics of connected subsystems
Authors: Igusa Takeru , Achenbach Jan D. , Min Kyung-Won
Published: October 2005

Resonance characteristics of a collection of linear subsystems interconnected by waveguides is examined. The properties of the subsystems are determined separately using modal or mobility analysis, and the topology of the waveguide network is cast in matrix form using a wave propagation approach. It is found that harmonic responses are described by multivalued analytic functions defined in the complex plane (or a single-valued function on a Riemann surface). The excitation frequency is represented by the unit circle and resonances are poles located outside of the circle. The analytic form of the response functions provide insight into the relationships between the modes of each interconnected subsystem as well as the relationship between modal vibration and wave propagation effects. These relations are complicated by the fact that the interconnecting waveguides by themselves introduce new modes of vibration. The complexity of the problem is reduced by restricting the analysis to a window of complex frequencies, whose width is determined by the degree of interaction between modes of the subsystems and wave guides. The interaction is measured by parameters generalized from an earlier study of simpler subsystems [T. Igusa and A. Der Kiureghian, J. Eng. Mech. 101, 20–41 (1985)]. [Work supported by ONR.]