bluuebunny/crossref_metadata_2025 huggingface dataset:
Frequency difference limens for short-duration tones
Authors: Moore B. C. J.
Published: October 2005

Models which attempt to account for our ability to discriminate the pitch of pure tones are discussed. It is concluded that models based on a place (spectral) analysis should be subject to a limitation of the type Δf⋅d ⩾ constant, where Δf is the frequency difference limen (DL) for a tone pulse of duration d. The value of this constant will depend on the ability of the system to resolve small intensity differences. If a resolution of 1 dB is assumed, the value of the constant is about 0.24. In principle, a mechanism based on the measurement of time intervals could do considerably better than this. Frequency DLs were measured over a wide range of frequencies and durations. It was found that at short durations the product of Δf and d was about one order of magnitude smaller than the minimum predicted from the place model, except for frequencies above 5 kHz. A “kink” in the obtained functions was also observed at about 5 kHz. It is concluded that the evidence is consistent with the operation of a time-measuring mechanism for frequencies below 5 kHz, and with a spectral or place mechanism for frequencies above this.
Brillouin scattering in aqueous electrolyte solutions
Authors: Maret Arthur R. , Yeager Ernest
Published: October 2005

Brillouin scattering studies have been carried out in a number of aqueous electrolyte solutions at various concentrations up to 4M. Two types of results have been obtained: (1) hypersonic velocity data from the frequency shift of the Brillouin components; and (2) hypersonic absorption coefficients from the line widths of the Brillouin components. The results compare favorably with those from ultrasonic velocity and absorption data at lower frequencies.
A theoretical model for the backscattering strength of a composite-roughness sea surface
Authors: Bachmann Wolfgang
Published: October 2005

The surface-backscattering strength is calculated according to Rayleigh theory for surface excursions much smaller then the sound wavelength. By taking account of self-shadowing and modulation of grazing angles, the validity of the result is extended to arbitrary roughness. The predictions of this new composite-roughness scattering model are compared with a small set of experimental data.
Resolving power and sensitivity to mismatch of optimum array processors
Authors: Cox Henry
Published: October 2005

Mismatch in a beamformer occurs when the knowledge of the signal directional properties is imprecise. The effects of mismatch on a conventional beamformer and two optimum beamformers are compared. One optimum beamformer is based on inversion of the noise cross-spectral matrix while the other is based on inversion of the signal-plus-noise cross-spectral matrix. When there is mismatch, the inclusion of the signal in the matrix inversion process can lead to dramatic reductions in the output signal-to-noise ratio when the output signal-to-noise ratio of a perfectly matched beamformer would be greater than unity. However, the corresponding effect on the total beamformer output is less dramatic since an increase in the noise response partially offsets the decrease in signal response. The question of suppressing mismatched signals is closely related to the question of resolving closely spaced sources. Exact conditions are presented for resolution of closely spaced sources by an optimum beamformer. These results are applied to a line array and compared with the resolution capability of a conventional beamformer. It is found, for example, that an output signal-to-noise ratio of about 47 dB is required to achieve a resolving power with an optimum processor which is ten times that given by the classical Rayleigh limit. Conditions are also presented for the resolution of two sources of unequal strength.
Correlation of echographic visualizability of tissue with biological composition and physiological state
Authors: Fields Scott , Dunn Floyd
Published: October 2005

It is argued that the elastic properties of soft tissues are largely responsible for their echographic visualizability and that these are determined, for the most part, by structural collagen-containing components.
Author's reply to “Comments on ’Effectiveness of continuous spatial averaging in a diffuse sound field‘ ” [M. S. Corrington, J. Acoust. Soc. Am. 54, 813–814 (L) (1973)]
Authors: Lubman David
Published: October 2005

It is acknowledged that there are errors in Eq. A10, together with Eqs. A4 and A5 of our paper [J. Acoust. Sco. Am. 53 650–659 (1973)]. The advantages of the Corrington method of solution, in certain circumstances, are discussed.
Some observations on negative potential in scala media during anoxia
Authors: Konishi T.
Published: October 2005

It has been widely accepted that the endocochlear potential (EP) is highly dependent on metabolism and changes its polarity almost immediately after onset of anoxia. The origin of the negative potential in the scala media at the early stage of anoxia has not been fully understood. In this study, changes in EP and Na+ and K+ concentration in the endolymph were measured in anesthetized and immobilized kanamycin-treated guinea pigs and in waltzing guinea pigs. EP and Na+ and K+ concentration in the endolymph were in the normal range in those animals, whereas the cochlear microphonic and whole-nerve action potential were severely suppressed. Under anoxic conditions, rise of Na+ and fall of K+ concentration in the endolymph did not show significant differences from those found in normal guinea pigs. However, the magnitude of the negative potential in the anoxic cochlea was substantially smaller than that found in normal animals. The mechanism underlying generation of the negative potential in scala media during anoxia is discussed. [Supported by NIH Grant.]
Interruption of recovery during intermittent stimulation produces greater loss in sensitivity
Authors: Price G. Richard
Published: October 2005

Recent work on hearing loss to intermittent stimulation [G. R. Price, J. Acoust. Soc. Am. 56 (1974)] has documented an early recovery process (ERP) in the cochlear microphonic (CM) and led to the hypothesis that if the ERP is interrupted by subsequent exposures, loss in CM sensitivity would be greater than the loss to an equivalent amount of continuous stimulation or to an equal amount of intermittent stimulation timed to avoid interrupting it. This hypothesis .was tested by exposing anesthetized cat ears to 5.0-kHz pure-tone stimulation at an intensity 30 dB above that producing a maximum in CM output at the round window. Stimuli were on for 1 min and off for an appropriate period until a total of 30 rain of stimulation had been delivered. In one experimental group (N = 10), the exposure schedule interrupted the ERP when it had 5 dB left to run and for a second group (N = 10), the inte rexposure interval was 9 min, a period allowing the ERP to run to completion. Interrupting the ERP did result in significantly more loss than continuous stimulation. Allowing the ERP to go to completion resulted in less loss, although the difference was not statistically significant. Data from studies using human ears indicate that similar processes may also be acting in the human ear as well.
Auditory responses from the saccule: further evidence for the mechanical origin of inhibition
Authors: Capranica R. R. , Moffat A. J. M.
Published: October 2005

Action potentials were recorded from single fibers from the saccule in the American toad (Bufo americanus). While some of the fibers respond only to substrate vibration, many others are sensitive to airborne sound. The response of a fiber to an excitatory tone can be totally inhibited by a second higher-frequency tone. Furthermore, each fiber can be excited by a pair of tones with a frequency difference approximately equal to the unit's best excitatory frequency. These saccular fiber responses are very similar to those of the inhibitable fibers from the amphibian papilla. Our previous studies of the amphibian papilla have demonstrated that inhibition and difference tone excitation are related phenomena having a mechanical rather than a neural origin. That these two organs, so structurally different, should exhibit such similar response properties indicates that inhibition and difference tone excitation do not occur in the organs themselves, but rather in the fluids of the inner ear or in the middle ear. [Supported by NIH Grant NS-09244.]
Speech coding by Kalman filtering and stochastic approximation
Authors: Arcese A. , Goldberg A. J.
Published: October 2005

In this paper, we consider compressing speech by Kalman filtering and by a class of stochastic estimation algorithms. Using the Kalman estimation procedure, we will derive the stochastic estimation algorithms which are recursive procedures for approximating either predictor or partial correlation coefficients. In these algorithms the analyzer generates an error signal that is quantized and transmitted. The predictor coefficients or partial correlation coefficients are updated at each sampling instant from the quantized error signal and need not be sent, because these data are also available at the receiver. The synthesizer recomputes the predictor coefficients or partial correlation coefficients and reconstructs the speech waveform. Audio tapes will demonstrate speech processed with these algorithms.
Low-frequency sound attenuation in Baffin Bay
Authors: Browning David G. , Mellen Robert H. , Ross John M. , Merklinger Harold M.
Published: October 2005

At low frequencies the attenuation in underwater sound channels is evidently dominated by forward scatter from large-scale inhomogeneities. Propagation-loss measurements were made in Baffin Bay during early September, when there is a relatively strong channel with axis near 100 m. The results show a frequency-independent scatter loss (30–300 Hz) of approximately 0.006 dB/km. Comparison is made with the values for the deep sound channel and for other shallow sound channels throughout the world.
Method for calculating the absorption of sound by the atmosphere
Authors: Sutherland L. C. , Piercy J. E. , Bass H. E. , Evans L. B.
Published: October 2005

The aim is a standard method of calculation with the credibility necessary to support major compromises between economic and environmental benefits desired by society. A two-relaxation model for the absorption of sound in air is fitted to a maximum of available laboratory and field measurements, as well as the detailed knowledge of basic mechanisms described in the previous paper. In this way, a set of seven fairly simple equations are generated whose form is determined by the physical nature of the mechanisms. These formulas may be used to compute the absorption of a pure tone over the following range of variables to an estimated accuracy of ±(10 + 1.5|t − 20|)%: frequency/pressure 102 to 105 Hz/atm, temperature (t) 0° to +35°C (32°−95°F), relative humidity 10%–100%, pressure 2 atm or less. A method of corrected the result to apply to bands of noise is proposed.
Biochemistry of neomycin ototoxicity: inhibition of calcium binding and turnover of membrane polyphosphoinositides
Authors: Schacht J.
Published: October 2005

In contrast to the well-established antibacterial mechanism of aminoglycoside antibiotics, the biochemical basis of their toxic effects towards mammalian cells remains unclear. The nature of these effects suggests that the primary site of action is at the membrane. Of the various membrane components the polyphosphoinositide lipids play a crucial role in the control of membrane permeability. To analyze these trace lipids, phospholipids of stria vascularis, spiral ligament, and ductus cochlearis were labeled with radioactive phosphate. After short incubation times, the most highly labeled lipids were phosphatidylinositol phosphate and phosphatidylinositol diphosphate. Treatment of guinea pigs with neomycin decreased labeling of these polyphosphoinositides, but not of other lipids. In vitro experiments on the labeling and hydrolysis of polyphosphoinositides demonstrate that this effect is caused by an inhibition of polyphosphoinositide turnover. Neomycin was also shown as a competitive inhibitor of calcium binding in vitro. Evidence suggests that neomycin directly binds to the polyphosphoinositides, displacing calcium and rendering these lipids inaccessible for their enzymes. The toxic effects result from this interference with calcium homeostasis at the membrane and inhibition of the turnover of vital membrane components. [Supported by HIH-NIEHS Contract NO1-ES-2-2110 and NIH Program Project Grant NS 05785.]
Fourier analysis of second-turn cochlear microphonic response in chinchilla
Authors: Drescher D. G. , Gibian G. L.
Published: October 2005

Chinchilla cochlear microphonic responses to tones for input levels from 20 to 110 dB SPL have been examined by a fast-Fourier-transform algorithm. Distortion of cochlear microphonic response in the second turn (CM2) in response to excitation at 225 Hz was found to grow rapidly as the level of stimulus was varied from 50 to 70 dB SPL. Like the input-output function for the complete CM2 response, the input-output functions for the harmonics showed saturation near 80 dB SPL. Changing body temperature between 29°C and 39°C had little effect on the pattern of the harmonics examined, but exposure to steady noise resulted in marked changes in output waveforms of CM2. Some possible mechanisms for noise-induced reduction of CM may be excluded by these differences between the effects of temperatures and acoustic stress. [Supported by NINDS Grant NS 03856.]
Model of neuro-mechanical sound filtering in the cochlea
Authors: Zwislocki J. J. , Sokolich W. G.
Published: October 2005

Little doubt seems to remain that tuning curves of acoustic-nerve fibers are narrower than corresponding distributions of mechanical amplitudes along the cochlea. Single-unit responses to trapezoidal waveforms produced in cochleas of kanamycin-treated and untreated Mongolian gerbils have brought us to the conclusion that inner and outer hair cells interact approximately in phase opposition. This finding leads to a simple model for neural sharpening of mechanical filter characteristics of the cochlea if we accept the anatomical evidence that corresponding inner and outer hair cells are separated along the cochlear spiral by the length of the spiral fibers and assume that normal inner hair cells are only slightly less sensitive than normal outer hair cells. The last assumption is controversial, but amplitude and phase predictions of the model are consistent with directly relevant neurophysiological data. The model seems to apply to units with low and medium CFs. It can be extended to higher CF units by assuming a rectification process that produces a polarity opposition between inner and outer hair cells. The extension is currently under investigation.
Site selection methodologies for community noise monitoring
Authors: Konheim A. G. , Bragdon C. R. , Ruth J. S.
Published: October 2005

The United States Environmental Protection Agency (EPA) has derived a site selection methodology for a pilot community noise-monitoring project. This site selection methodology was arrived at in the following step-wise fashion. First, three main sampling procedures were identified (random, deterministic, and stratified) and their advantages and disadvantages were stated in terms of the degree of preliminary data they require, their amenability to statistical analysis, and their ability to insure representation of certain major noise sources or land uses, which may be of interest [H. B. Safeer, J. Sound Vib. 26, 489–502 (1973)]. Secondly, the relative importance of these considerations was found to vary In conjunction with a survey's objectives and as a function of the total number of measurement sites, the size of the total sampling area, the length and/or number of measurements to be made at each site, the number of equipment systems and operators available, and the availability of funds. Lastly, the relationships of these project parameters to site-selection methodology were determined. A survey of previously conducted noise studies was performed to ascertain the extent to which these relationships were embodied in the site selection methodologies they employed.
Sound propagation in urban areas
Authors: Donavan P. R. , Lyon R. H.
Published: October 2005

An investigation of the propagation of sound from identifiable sources in urban areas has recently been concluded. As a result of this study, sound-level estimation procedures have been developed for typical configurations including: propagation through intersections, both single and multiple; propagation around corners, also both single and multiple; and propagation into side streets. These procedures apply to near-ground-level noise sources which are identifiable above background level and can be approximated by either a point or line source. Methods were developed for assessing the modification of the basic estimation procedures due to diffraction, ground interference effects, and sound-level nonuniformity across street sections near intersections. Additional modification of the basic estimation procedures accounting for building surface irregularities has been determined and can be used in sound-level prediction. Levels predicted by the estimation procedures have been compared with both field and model data. [Work supported by U. S. Department of Transportation and National Science Foundation.]
Octave adjustment by non-western musicians
Authors: Burns Edward M.
Published: October 2005

Nine Indian musicians were asked to adjust the second tone of a pair of pure tones, presented in unending succession, to an octave above the first tone. The results were essentially the same as found in experiments with Western musicians, that is, small intrasubject variability, large intersubject variability, and a small but statistically significant frequency-dependent “stretch” of the physical octave when adjusting the subjective octave. Although these results shed no light on the question of whether this stretch is due to some inherent property of the auditory mechanism or is learned from speech sounds, as suggested by Terhardt, they do seem to rule out the explanation that this stretch is learned from the “stretched” tuning of the piano to which Western musicians are universally exposed. This stretched tuning of the piano is due to certain physical characteristics of the piano strings and is not found in most instruments to which the Indian musicians are exposed.
Speaker-independent digit recognition system
Authors: Sambur M. R. , Rabiner L. R.
Published: October 2005

This paper describes an implementation of a speaker-independent digit recognition system. The digit classification scheme is based on segmenting the unknown word into three regions and then making categorical judgments as to which of six broad acoustic classes each segment falls into. The measurements made on the speech waveform include energy, zero crossings, and two-pole LPC analysis. A formal evaluation of the systems showed an error rate of 2.7% for a carefully controlled recording environment, and a 5.6% error rate for online recordings in a noisy computer room.
Comparison of two speech understanding systems
Authors: Lowerre B. T.
Published: October 2005

This paper presents the results of a comparison of two speech systems developed at Carnegie-Mellon University: Hear-Say-1 (1) and Dragon (2). Hear-Say-1, Which consists of cooperating but independent knowledge sources, utilizes a best first search technique of the syntatic grammer. The search terminates when the evaluation of a completed proposed sentence achieves a heuristic threshold. In this way only the most likely paths are searched, thus achieving a fast though not always accurate recognition. Dragon utilizes a combined network of grammar and phonetic spellings for a generative Markov process. This allows Dragon to search all possible paths in parallel in an amount of time that is linear with utterance length. Dragon obtains a higher accuracy of recognition but with a higher computational overhead. Both systems have been tested on the same five sets of data consisting of 102 utterances and 564 words. Hear-Say-1 correctly identifies about 60% of the words in about 5 to 30 times real time, depending on the number of wrong paths searched. Dragon correctly identifies about 85% of the words in an almost consistent 50 times real time. The types of errors produced by both systems are discussed. [(1) Written by D. R. Reddey, L. E. Erman, R. D. Fennell, and B. T. Lowerre, (2) Written by J. K. Baker.]
Automatic formant tracking
Authors: Kameny Iris , Brackenridge William A. , Gillmann Richard
Published: October 2005

A formant tracker has been implemented which assigns frequency, amplitude, and bandwidth to each of the first three formants for each 10-msec voiced segment of continuous speech. Its input parameters are fundamental frequency, rms, and up to five spectral peaks below 5000 Hz with their respective amplitudes and bandwidths. Peak information is obtained from linear prediction spectra. Techniques distinguishing this formant tracker from previously reported ones are as follows: (1) All spectrum computations are accomplished in the peak picking phase, before ferment tracking—this may yield spurious formants but few missing ferments. (2) Anchor points are located by selecting three consecutive 10-msec segments in which three formant frequencies do not differ from one segment to the next by more than a threshold amount. Tracks for the first three ferments are extended in both directions from these anchor points. (3) Frequency pattern matching aids decision making when more than one ferment is possible for a given slot; this is particularly useful for nasals and /l/ and /w/. Frequency pattern information is derived from speaker vowel-sonorant frequency tables. [This research was supported by the Advanced Research Projects Agency of the Department of Defense under Contract No. DAHC15-73-C-0080.]
Acoustic effects of style of speech
Authors: Ladefoged Peter , Kameny Iris , Brackenridge William A.
Published: October 2005

Recordings were made of ten speakers of California English producing the set of test words: “bee, bow, boy, bed, bad, bud” in seven different styles of speech: (1) free speech during an interview in which subjects were induced to say the test words without their having been previously spoken by the experimenter; (2) spontaneously spoken lists of words; (3) spontaneously produced sentences; (4) repetition of sentences spoken by the experimenter; (5) reading a continuous passage; (6) reading the test words in the sentence “The word is ___”; (7) reading the test words in lists. Each word was made to occur in phrase final, stressed position. The data were analyzed using computer programs that determine the frequency, amplitude, and intensity of the first four formants. The nucleus of each vowel in each word in each style was determined by an algorithm. The nucleus of the vowel differed in some styles of speech. Some valid generalizations about the speech of educated Southern Californians can be made on the basis of data from read speech. [This research was supported by the Advanced Research Projects Agency of the Department of Defense under Contract No. DAHC15-73-C-0080.]
Current technology of recording and reproduction of sound on film in the Soviet Union, and the direction of future developments
Authors: Belkin B. G. , Ioshin O. I.
Published: October 2005

The present state of motion picture sound technology in the USSR is considered. Briefly described are the technical parameters and the designing peculiarities of the sound-equipment basic elements, and the main features of the modern technology of sound recording-reproduction in cinematography. It is shown that the available characteristics of modern equipment approach their physically possible limits, and that further considerable improvement of quality and widening of the creative possibilities in sound technology can be achieved only by transition from the analog method to the digital one. The general principles of sound recording-reproduction technology in cinematography are formulated, the main quantitative correlations are determined and the advantages and the shortcomings of the digital method, as well as the technical and economic difficulties encountered upon its introduction are considered. In conclusion, it is suggested that the digital method should be developed jointly by Canadian, American, and Soviet scientists and designers.
Radial catenoidal horn in loudspeaking telephony
Authors: Lee H. K.
Published: October 2005

One of the adverse factors going against designing the loudspeaking telephone set powered solely via the telephone line has been the critical shortage of power to drive the loudspeaker adequately. A folded radial catenoidal horn of moderate size has been designed as a means of boosting the efficiency of the small, therefore inherently inefficient, loudspeaker used in the desk-top model loudspeaking telephone set. Based on the classical plane wave horn theory, the catenoidal horn had been selected as most suitable for this purpose. With the aid of a computer, the horn parameters had been optimized to the particular driver loudspeaker parameters. The folding and the bell portions had been rigorously calculated and build based on these parameters optimized through the iteration method. Test results showed that the horn boosted the radiation efficiency of the loudspeaker 12 ± 5 dB over the frequency range 300–1500 Hz, where most of the speech energy is stored.
Leaky infrasonic guided waves in the atmosphere
Authors: Kapper Christopher Y.
Published: October 2005

Prior theoretical formulations and computational techniques for the prediction of pressure waveforms generated by large explosions in the atmosphere have considered only fully ducted modes. In the present paper, a technique for including weakly leaking guided modes in concert with fully ducted modes is developed. Modification of previous theory includes the extension of the boundary condition at the upper halfspace to include a complex horizontal wavenumber. The major alterations to the computer program Infrasonic Waveforms (as described in report by Pierce and Posey, 1970) incurred consist of the computation of the imaginary part of the newly incorporated complex wavenumber, extension of the normal-mode dispersion function to lower frequencies, and a second-order correction factor to the phase velocity. [Work sponsored by Air Force Cambridge Research Laboratories.]
Calculating of two-cycle-engine exhaust pressure oscillations in a muffler
Authors: Mutyala Brahmaji , Soedel Werner
Published: October 2005

In designing a muffler, the acoustic engineer must consider two objectives: (1) the specified insertion loss and (2) the estimates of change in engine performance brought about by an exhaust muffler on the performance of a small two-cycle gasoline engine by predicting the exhaust muffler pressure oscillations in a muffler. Such engines are commonly used for garden tools, power boats, and motocycles. The exhaust muffler consists basically of two elements: an expansion chamber or cavity and a tailpipe open to the atmosphere. A mathematical model is presented which includes the simulation of two-cycle-engine and exhaust muffler pressure oscillations. Two methods are given for coupling the nonlinear mass flow through the exhaust ports with an acoustic wave equation model employed for the exhaust pressure simulation. With the first method, the Helmholtz resonator approximation is used to model the wave equation in the exhaust system and is solved simultaneously with the engine thermodynamic equations using the Runge-Kutta method. With the second method, a steady-state solution for the acoustic impedance at the exhaust ports is coupled with the nonlinear flow through the exhaust ports to obtain a solution. The calculation of the pressure oscillations in a muffler will help in estimating the change in engine performance brought by the muffler. Comparisons between theory and experiment are given.
Algorithm for determining the endpoints of isolated utterances
Authors: Rabiner L. R. , Sambur M. R.
Published: October 2005

An important problem in speech processing is to detect the presence of speech in a background of noise. This problem is often referred to as the endpoint location problem. By accurately detecting the beginning and end of an utterance, the amount of processing of the speech data can be kept to a minimum. The algorithm proposed for locating the endpoints of an utterance is based on two measures of the signal, namely, zero crossing rate and energy. The algorithm is inherently capable of performing correctly in any reasonable acoustic environment where the signal-to-noise ratio is on the order of 30 Db or better. The algorithm has been tested over a variety of recording conditions and for a large number of speakers and has been found to perform reasonably well across all tested conditions.
Modeling linguistic stress patterns in connected speech
Authors: Cheung J. Y. , Minifie F. D.
Published: October 2005

The purpose of this paper is to present a model of linguistic stress patterns in connected speech. By categorizing the syllables roughly into three levels—unstressed, stressed, and prominently stressed—the magnitude of stress for the unstressed and stressed syllables can be accurately predicted, according to its relative position in the phrase group, knowing only the positions and the magnitude of a few prominently stressed syllables. The model shows that the perceived stress associated with each syllable is highly dependent on three factors: (1) the time lag between the present syllable and the last prominently stressed syllable (the recency effect), (2) the lead time between the same syllable and the forthcoming prominently stressed syllable (the anticipatory effect), and (3) the alternate stress rule. Data are collected from various speakers reading several chosen passages at different rates. The magnitudes of linguistic stress for these recordings are obtained both from perceptual judgments of trained listeners and from computer analysis. Results show that (1) stressed syllables occur at a regular rymthm, and (2) their magnitudes can be closely predicted by the model. Some implications of the present model and comparisons with other models are discussed.
Characterization of fundamental-frequency contours of speech
Authors: Maeda S.
Published: October 2005

Fundamental frequency (F0) contours for 60 isolated American English sentences and a text, read by three native speakers, have been analyzed and these contours have been schematized subjectively by ideal patterns. The basic elements (attributes) of the schematic patterns are a baseline (BL) which represents the gradual fall of the F0 contour along the sentence, a piece wise-linear trapezoidal pattern with rising (R) and lowering (L), and a peak (P) which often occurs at the onset of R. The trapezoidal pattern demarcates a sentence into phonetic groups (PGs) of words. Additional attributes are a rising (R1) which can occur at an intermediate content word in PG, and continuation rise. Intraspeaker differences for these schematic patterns were negligible for the sentences investigated. The organization of the patterns seems to be constrained by two primary factors: the structure of the sentence and a principle of physiological economy in the control of F0 during speech. [Research supported by an NIH grant.]
Computation of acoustical impedance of Helmholtz resonator in aircraft engine environments
Authors: Tseo G. G.
Published: October 2005

The nonlinear impedance of a Helmholtz resonator is an important parameter when predicting noise reduction in the environment of high sound-pressure level and grazing flow in fan ducts of aircraft engines. In recent experiments [G. D. Garrison, A. C. Schnell, C. D. Baldwin, and P. R. Russell, “Suppression of Combustion Oscillations with Mechanical Damping Devices,” Pratt & Whitney Aircraft Rep. PWA-FR-3299 (1969)], the impedance was found to be affected by the exciting sound spectrum distribution. An integration method was presented by Rice [E. J. Rice, “A Model for the Acoustic Impedance of a Perforated Plate Liner with Multiple Frequency Excitation,” NASA TMX-67950 (1971)[ to compute the impedance under such an environment. However, the integration requires substantial computation time. Since the impedance is determined by solving the multiple-frequency velocity components of the nonlinear resonator, a direct method to compute the velocity components of the resonator is desirable. A simple iteration method to determine the impedance is presented by the author. In this method, the lengthy integration is not used; instead, an algebraic equation converted from the nonlinear differential equation has to be solved. The physical modeling includes both the effects of multiple-frequency excitation and grazing flow, using similar relationships to those presented in Rice's paper. The impedance computation of this approach requires much less computer time and the results are in good agreement with those from the integration method.
Temporal integration in auditory signal detection
Authors: Marken R. , Ahumada A.
Published: October 2005

The outputs of computer-simulated energy detectors were correlated with the responses of observers in a detection task to determine the duration and shape of the temporal integration process. The 500-Hz 100-msec signal was masked by wideband Gaussian noise. Detectors followed by rectangular and exponential integrators were equally successful in accounting for 70% for the predictable variance in observers' responses. The time constants of the best correlating detector integrators averaged 300 msec for the rectangular and 150 msec for the exponential shaped functions, with single tuned filters of 20- or 40-Hz bandwidth. Energy variations in successive time intervals of 25-msec duration were correlated with responses of observers and energy detectors. A regression analysis which recovered the shape of the energy detector integration functions showed that the observers give significantly negative weight to energy prior to the signal when it is present, but positive weight to such energy when the signal is absent.
Diversity of middle-ear transformer mechanisms in kangaroo rats, kangaroo mice, and pocket mice
Authors: Webster Douglas B.
Published: October 2005

The tympano-ossicular system has been anatomically analyzed in 26 species of heteromyid rodents. Based on anatomical measurements, 24 of the 26 species should have a transmission efficiency of 94%–100% from tympanic membrane to cochlear, at the resonant frequency. The areal ratio of stapes footplate to 2/3 tympanic membrane is remarkably constant among the species, varying only from 0.04 to 0.07: in Dipodomys and Microdipodops this small ratio is due to the very large tympanic membrane; in Perognathus and Liomys it is due to the extremely small stapes footplate. The lever ratio of incus to malleus varies from 0.28 to 0.33 in Dipodomys and Microdipodops, from 0.37 to 0.46 in Perognathus, and from 0.55 to 0.60 in Liomys. In addition, the middle-ear volumes and the morphology of tympanic membrane, ossicles, ligaments, and muscles all combine to minimize both mass and stiffness. All these data suggest middle-ear mechanisms which are very efficient over a broad frequency range. We have cochlear microphonic data consistent with this conclusion. [Supported by NIH Grant 11459.]
Nonlinear scaling for saturation-limited parametric arrays
Authors: Fenlon F. H.
Published: October 2005

An exact asymptotic farfield solution of the plane-wave Burgers' equation is derived for the case of a dual-frequency parametric source. The influence of spherical spreading losses is then included by matching the solution at low primary wave amplitudes with solutions obtained by the method of successive approximation. From the matched solution, the referred pressure of the difference-frequency component at 1 m from the source (i. e. , the equivalent difference-frequency source level) is derived as a universal function of scaled primary-wave amplitudes, frequencies, source dimensions, and physical constants of the medium. This function, which defines the saturation limit of a parametric array more completely than previous approximate models, permits the maximum conversion efficiency of the source to be evaluated for a particular set of input parameters. The amplitude dependence of the directivity function and directivity index for spreading loss controlled arrays can also be determined. Comparison of the characteristic with data published in the literature shows good agreement over quite a wide range of scaled primary-wave source parameters.
Aperture factors for parametric source beam patterns
Authors: Moffett Mark B. , Mellen Robert H.
Published: October 2005

It is well known that the beam patterns of parametric acoustic sources whose difference frequency is generated largely within the nearfield of the primary beam consist of a multiplicative aperture factor which can be important at large angles when ka ≳ 1, where k is the difference frequency wave-number and a is the aperture radius. Not so well known, however, is the fact that the same aperture factor arises in the case of a spherically spreading, conical primary beam of finite initial aperture. The importance of the aperture size in determining the off-axis behavior of parametric sources is discussed.
Radiation of difference frequency generated by nonlinear interaction of a silicone rubber cylinder
Authors: Ryder J. D. , Rogers P. H. , Jarzynski J.
Published: October 2005

We investigate both theoretically and experimentally a parametric source where the major part of the difference frequency generation occurs within a solid cylinder of silicone rubber. The primary waves (∼1.5 Mhz) are confined to a narrow region around the axis of the cylinder and, owing to the relatively high absorption at these frequencies, are largely attenuated before reaching the end of the cylinder. The silicone rubber cylinder serves two purposes: (1) owing to the low sound speed (∼1000 m/sec) and high parameter of nonlinearity (B/A ∼8) of the silicon rubber, the virtual sources are more than three times as strong as they would be in water; (2) again, owing to its low sound speed, the silicon rubber cylinder acts as a slow waveguide antenna, so that the beamwidth is considerably narrower than that which would be produced by a conventional parametric array of the same effective length. The system is analyzed by numerically solving the system of equations obtained by coupling the surface Helmholtz integral equation for the interior of the silicon cylinder (which includes the virtual sources) with the integral equation for the region exterior to the cylinder. Theory and experiment compare favorably.
Session U. Speech Communication: Automatic Speech Recognition, Speech Understanding, and Speech Science
Authors: Walker Donald E.
Published: October 2005

Have the goals of automatic speech recognition and speech understanding resulted in distinctive development in speech science? Are new developments in speech science required that will be uniquely responsive to these goals? Can work on automatic speech recognition and speech understanding contribute to speech science? These and related questions will be discussed by the panel.
Temporal acuity as a function of frequency difference
Authors: Wier C. C.
Published: October 2005

A sinusoidal waveform of fixed intensity and variable duration (T) was generated digitally. The waveform was composed of two segments each T/2 in duration. One segment was always a 1000-Hz sinusoid; the frequency of the other segment was variable, ranging between 1032 and 4000 Hz. Since some combinations of the frequency and duration of segments yield fractional cyclces of a sinusoid that may result in distinctive transients, the starting phase of each waveform was randomized from trial to trial. Two such waveforms were presented monaurally in a 2AFC paradigm. The two segments in each waveform were presented first in one order and then in the reverse order. Temporal acuity was estimated for each value of ΔF tested by varying the total duration (T) of the waveform. Results are compared with previous estimates of auditory temporal acuity. [This research was supported by the National Institutes of Health.]
Combination-tone measurement and monaural phase perception
Authors: Zurek P. M. , Leshowitz B.
Published: October 2005

A previously reported [Jenkins et al., J. Acoust. Soc. Am. 55, 390 (1974)] forced-choice procedure for the estimation of the phase and amplitude of aural distortion products was applied to the measurement of the combination tones 2f1 − f2, 3f1 − 2f2, and f2 − f1. This method assumes linear superposition of a combination tone with an external cancellation tone. With this procedure, a minimum in the function relating percent correct discrimination and the phase of the cancellation tone should be obtained when the cancellation tone is added in quadrature with the combination tone. The estimate of the phase of the combination tone is thus independent of the level of the cancellation tone. This expectation was realized with the difference tone, regardless of level, and with the higher-order combination tones at low levels. With high-level (70 dB SPL) primaries and an equal-level cancellation tone, a minimum did not always occur; i.e., essentially perfect performance was achieved. If it did occur, it was typically not at the same phase measured with lower-level cancellation tones. The hypothesis that the discrimination with high-level cancellation tones is due to the interaction of several combination tones was tested. A low-pass noise was added at an intensity sufficient to mask any distortion components which may interact. The discrimination was unaffected by the addition of the noise. These results demonstrate a monaural phase effect which cannot be attributed to combination-tone interactions.
Experiments with a wearable visual-indicating aid to speech reception
Authors: Gengel Roy W. , Pickett J. M. , Quinn Regan , Upton Hubert W.
Published: October 2005

Preliminary tests are reported to evaluate an electronic aid to speech reception that displays light patterns associated with certain speech features that are difficult to lipread. The light sources are segments of an alphanumeric LED module, arranged in a small projection system on the eyeglasses of the user. The light patterns are reflected into the user's eye so that he can view them projected on the face of the talker. The analyzing system employs four filters followed by a logic network that compares filter outputs and times burst-like sounds to turn the display light segments on and off. The sound features that may be displayed are “front” vowel and “back” vowel, high-frequency friction, low-frequency friction, plosion, and low-frequency murmur like that of nasal consonants. Hearing-impaired subjects appeared to be able to learn rapidly to use some of the light cues to improve lipreading in a phoneme-reception test but they experienced difficulty with sentences. Further testing will include extended everyday use. [Research supported by U. S. Social and Rehabilitation Service.]
Turbulence-induced spectral broadening of discrete frequency noise
Authors: Semrau W. R.
Published: October 2005

Spectral broadening of discrete-frequency compressor noise was observed during a series of noise measurements for small compressors. The phenomenon appeared to be similar to “haystacking,” which results from turbulence in the inlet flow. Measurements made in a long flow-measuring inlet duct indicated that near the compressor the blade-pass noise was discrete frequency in character, while at some distance upstream it was spectrally broadened, having a bandwidth (at −6 dB) up to 15% of the tone frequency. Thus, the broadening occurs in propagation rather than generation. The origin of the spectral broadening was found to be random frequency modulation of the compressor tone resulting from propagation through orifice-generated turbulence. Analysis and discussion of the broadening process is presented.
Intensity, spectrum, and directivity of turbulent boundary-layer noise
Authors: Tam Christopher K. W.
Published: October 2005

The problem of noise radiated from a turbulent boundary layer is studied. It is found that the farfield noise intensity, spectrum, and directivity can be determined completely if the nearfield pressure cross-correlation function is known. The role of supersonic and subsonic pressure components associated with the turbulent boundary layer in relation to noise radiation is examined. By using an empirical model of the wall pressure cross-correlation function it is found that the directivity pattern of turbulent boundary-layer noise differs considerably from that of a free dipole. One principal reason for this difference is the fact that the noise sources of a turbulent boundary layer are constantly in motion. The effect of mean flow outside the boundary layer on the radiated noise intensity and spectrum is also investigated. Numerical results indicate that this effect is very important for low-frequency noise components and for high subsonic flow Mach number.
Spinning mode sound propagation in ducts with acoustic treatment
Authors: Rice E. J.
Published: October 2005

Recent acoustic data have shown large noise attenuations for acoustically treated aircraft engine inlets without splitter rings. These encouraging data have stimulated a more detailed theoretical study of the acoustic propagation of spinning modes in acoustically treated open circular ducts. In addition, the suppressor with splitter rings was modeled by using the rectangular approximation to the annular duct. The theoretical models were used to optimize the suppressor designs for several spinning lobe numbers from zero to 50. Some interesting results of the analysis are that for circular ducts the maximum possible attenuation and the optimum wall impedance are strong functions of the lobe number. For annular ducts, the attenuation and optimum wall impedance are insensitive to the spinning lobe number for well-cut-on modes. The above results help explain why suppressors with splitter rings have been quite effective in spite of the lack of detailed information on the noise-source modal structure. Conversely, effective use of wall treatment only will require expanded knowledge of the noise-source structure. Approximate solutions are presented to help in understanding the more exact theoretical results.
Unified analysis of ducted turbomachinery noise
Authors: Zorumski W. E. , Lester H. C.
Published: October 2005

Hanson [J. Acoust. Soc. Am. (1973)] has modeled turbomachinery noise in a free field as a sequence of pulsed-dipole acoustic sources with randomly modulated amplitudes and arrival times. This approach produced a farfield noise spectra with a broad-band background in addition to the usual discrete noise component. A systematic method for calculating the internal and external acoustical fields of aircraft turbofan engines has been developed by Zorumski (NASA TR R-419). This work shows that the duct acoustic problem can be reduced to a linear matrix equation WA = Q where A and Q are vectors of modal amplitudes, and W is a square transmission/reflection matrix. In this paper the methodologies of Hanson and Zorumski are combined to yield a unified analysis of ducted turbomachinery noise. The essential features of Hanson's analysis are unchanged by the addition of duct acoustic effects. It is shown that the farfield, broad-band, and discrete noise spectral components can be expressed in terms of modal cross-spectral density matrices and directivity vectors which are derivable from the duct analysis.