bluuebunny/crossref_metadata_2025 huggingface dataset:
Statistical representation of word-initial obstruents: Adult data
Authors: Forrest Karen , Weismer Gary , Milenkovic Paul
Published: October 2005

The utility of a statistical representation of word-initial obstruents (Weismer et al., previous paper) was assessed for the speech of normal, young adults. Ten subjects repeated monosyllabic words, where the initial consonant was a voiced or voiceless stop or fricative, in a carrier sentence. Each place of obstruent articulation was paired with at least two vowels and each target word was repeated six times. The target words were digitized and the first four moments (mean, variance, skewness, and kurtosis) were calculated for successive 10-ms spectral slices, beginning with the onset of the obstruent and continuing through the third cycle of the vowel. Moments were calculated from linear and bark transformed spectra. Three of the moments (mean, skewness, and kurtosis) provide a good description of the shape and center of the spectral slices. These moments were plotted in three-dimensional space to determine whether a unique, three-dimensional pattern could be defined for each place of articulation. Discussion will center on the use of these moments, as they define a three-dimensional space, to quantify acoustic events associated with obstruent production. [Work supported by NIH Awards NS13274 and NS20976.]
Graphics editor for speech synthesis experiments
Authors: Helck Andrew , Coville Ian , Umeda Noriko
Published: October 2005

A graphics editor of acoustic parameters for a speech synthesizer [an assembly language version of the software synthesizer by D. Klatt, J. Acoust. Soc. Am. Suppl. 1 68, S18 (1980)] has been written. The program provides an easy selection of various editing functions on the screen. It gives users visual patterns of time-varying acoustic parameters and the time relationship among parameters (such as that among fricative, hiss, and voice amplitude for stop consonants). Variable parameters are grouped into pages according to function, and simple key presses on the terminal allow users to select pages, parameters, and editing functions for the parameter. Editing functions are: moving the cursor, inserting points for linear interpolation, changing parameter values on curves, setting a mark for playing the designated portion, etc. In addition, there are commands to assign a set of parameter values at a specified point of time from a table of phoneme definitions. The editor has been used to synthesize short sentences and phrases of about 1s and has proven extremely useful. [Work supported by NSA.]
Implementing prosodic phrasing for an experimental text-to-speech system
Authors: Bachenko Joan , Fitzpatrick Eileen , Lacy John
Published: October 2005

While text-to-speech systems tend to perform well on word pronunciation, they fall short when it comes to providing good prosody for complete sentences. An experimental text-to-speech system that uses a natural language parser and prosody rules to determine prosodic phrasing for English input to text-to-speech synthesis will be described. Building on information from the syntax tree, the prosody rules specify the location and relative strength of prosodic phrase boundaries; these specifications are then used to dictate modulations in pitch and timing for the Olive-Liberman synthesizer [J. P. Olive and M. Y. Liberman, J. Acoust. Soc. Am. Suppl. 1 78, S6 (1985)]. Two important assumptions motivate the prosody rules. First, constituency below the level of the sentence is the crucial determinant for boundary location; i.e., boundary location is influeneed by noun, prepositional, and adjective phrases, but not by clauses or verb phrases. Second, the relative strength of boundaries is determined by balancing prosodic phrases around a verb (or predicate) according to length, adjacency to a syntactic head, and the subject-predicate relation. The ability to predict phrasing reliably allows the system to use an expanded pitch range and insert pauses appropriately.
Nasal vowel study: Formant structure, perceptual evaluation, and neural representation in a model of the peripheral auditory system
Authors: Cheng Yan Ming , Guérin Bernard
Published: October 2005

In this study, the focus is on the search for a model that is articulatorily possible and perceptually plausible. A high-quality nasal sound synthesis structure in the acoustic domain, with a parallel formant synthesizer, that includes nasal articulation effects and defines independent acoustic parameters which are the basis of perceptual studies, has been obtained. A pole-zero-pole structure combined with a bank of formants is proposed as a high-quality nasal vowel synthesizer. A set of perceptual tests is conducted to test nasality variation with nasal synthesis eigenparameters. The results of tests are used to establish a new model describing [+ nasal]. Based on this model, the nature of the French nasal vowel and its distributional properties are discussed. Finally, nasality variation is tested at the auditory-nerve level with the help of a model of the peripheral auditory system. The neural firing pattern and the synchronization index pattern are discussed in terms of the nasality model established.
Children's perception of synthetic speech produced by rule
Authors: Greene Beth G.
Published: October 2005

Several experiments examined the perception of synthetic speech by school-age children ranging in age from 5- to 11-years old. These tasks all used synthetic speech produced by commercially available text-to-speech systems. In tasks involving free and ordered recall of lists of words, 8-year-old children recalled naturally produced items better than synthetically produced items even when the items were equated for familiarity and intelligibility. Other tasks included children's responses to spoken commands and sentence recall using meaningful and semantically anomalous sentences. As in our previous studies, performance was better for natural speech than synthetic speech, paralleling the general findings obtained with adults in our earlier studies. Synthetic speech, even when the speech is highly intelligible, may interact with task requirements to produce significant decrements in performance when children engage in complex cognitive tasks. [Supported by NINCDS.]
Some relationships between preference and intelligibility among synthetic voices
Authors: Logan John S.
Published: October 2005

The relationship between preference and intelligibility among synthetic voices was examined. Preferences among the speech produced by three text-to-speech systems were assessed using an A/B paired comparison procedure, and the intelligibility of the synthetic voices was measured using a transcription task. The text-to-speech systems tested were DEC talk, Prose 2000, and Infovox. Phoneme specific sentences served as stimuli and enabled a preliminary examination of the relationship between intelligibility and preference as a function of specific phoneme categories. Results indicated a close relationship between mean intelligibility and preference for a voice. In some cases, preference was found to vary with the intelligibility of individual classes of phonemes, accounting for some of the variability associated with preference decisions. These results will be discussed in terms of previous research involving preference and intelligibility. [Work supported by NIH and USAF.]
Mechanism of endocochlear potential generation by stria vascularis
Authors: Salt Alec N. , Melichar Ivo , Thalmann Ruediger
Published: October 2005

It is widely assumed that the endocochlear potential (EP) is generated by electrogenic ion transport by the marginal cells of stria vascularis (SV). It has recently been reported [I. Melichar and J. F. Syka, Hear. Res. 25, 35–43 (1986)] that a potential up to 10 mV higher than EP was recorded as microelectrodes were advanced through SV. Dye marking studies demonstrated that this potential is associated with the marginal cell region. The present authors have studied this potential further using small-tipped (0.5-2.0 μ o.d.), double-barreled potassium selective electrodes. As the electrodes were advanced through SV, a region of high K content with near zero potential was observed, corresponding to the basal cells. Immediately following, a region of positive potential (higher than the EP) and low K content was observed. In most cases endolymph was then reached, but sometimes two distinguishable regions of positive potential and high K were observed, corresponding to marginal cells and endolymph, respectively. The observation of a positively polarized low K region is not consistent with the role of the marginal cells as the generator of EP. On the basis of the observation that K is close to equilibrium between the basal cells and the extracellular space of stria, it is proposed that EP is generated by passive diffusion of K across this boundary. [This work supported by NIH.]
Cytochalasin D suppresses the compound action potential and summating potential of the guinea pig cochlea
Authors: Barron S. E. , Bobbin R. P. , Guth P. S. , Norris C. H.
Published: October 2005

Although actin has been localized in hair cells, evidence for its role in intact cochlear preparations is lacking. The present study was undertaken to test the hypothesis that actin is involved in cochlear function (e.g., the active process) by observing the effects of a potent filamentous muscle actin inhibitor, cytochalasin D, on evoked cochlear potentials. Perilymphatic spaces of guinea pig cochlea were perfused with Ringer solutions containing either cytochalasin D (10−8 to 10−5 M) or DMSO (0.00005% to 0.05%) at a rate of 2.5 μ1/min for 10 min [Bledsoe et al., Hear. Res. 4, 109 (1981)]. Immediately after each period of perfusion, the compound action potential (CAP), cochlear microphonics (CM), and summating potential (SP) evoked by 10-kHz tone bursts of varying intensity were recorded from an electrode in the basal turn. Cytochalasin D suppressed CAP and SP in a dose-dependent fashion but did not affect CM. DMSO had no effect. These results provide pharmacological evidence that actin is involved in cochlear function—possibly the active mechanical process of the organ of Corti. [Supported by NSF, NINCDS, the Southern Hearing and Speech Foundation, Kresge Foundation, and the LA Lions Eye Foundation.]
Anatomical effects of intense tone stimulation in the goldfish ear: Dependence on sound-pressure level and frequency
Authors: Cox Mardi , Rogers Peter H. , Popper Arthur N. , Saidel William M.
Published: October 2005

The final results of an experimental investigation to study the extent of frequency regionalization in the ear of goldfish are presented. Goldfish about 6 in. in body length were subjected to intense tones at 250 and 500 Hz, and four different sound-pressure levels. They were placed in a waveguide and constrained as close as possible to a pressure antinode so that the primary response of the inner ear was due to the induced motion of the swimbladder and Weberian ossicles. Both saccular and lagenar maculae were examined under a scanning electron microscope to determine the location and extent of hair cell damage as a function of frequency and sound-pressure level. The results are not inconsistent with the gross frequency regionalization in the saccular macula of codfish determined by P. S. Enger [Hearing and Sound Communication in Fishes (Springer, New York, 1981), pp. 243–255]. In addition, the results indicate a possible breakdown of the Weberian apparatus at extremely high sound-pressure levels where the primary site of damage switches from the saccule to lagena. This is consistent with the behavior of the system based on its viscoelastic properties as postulated by R. McN. Alexander [J. Exp. Biol. 38, 747–757 (1961)]. Work supported in part by ONR and NIH.]
The effects of hypoxia and temperature change on tuning and rate suppression in goldfish auditory nerve fibers
Authors: Fay R. , Ream T.
Published: October 2005

Goldfish auditory (saccular) nerve fibers are modestly tuned with best frequencies (BF) between 200 and 1500 Hz. Many fibers tuned below 400 Hz also exhibit rate suppression for signals well above BF. To investigate the origins of this frequency selectivity and suppression, the responsiveness of saccular fibers was measured during and after transient hypoxia, and during changes in body temperature of about ± 5°C. Respirator water temperature controlled body temperature (monitored in the brain). In hypoxia studies, the respirator water was turned off for about 20 min. Low temperature and hypoxia had similar effects: a reduction in spontaneous activity, sensitivity, and responsiveness, and greater adaptation and suppression. Although temperature reduction and hypoxia caused a reduction in the most excitatory frequency at any given intensity (suggesting a lowering of the fiber's BF), the present authors could not clearly distinguish between these effects and those caused simply by reducing sound intensity. Thus, tuning is not strongly dependent on temperature or oxygen supply, and the mechanisms of suppression in the saccular nerve are likely different from those operating in the cochlear nerve of most other vertebrates. [Work supported by a Program Project Grant from NIH and NINCDS.]
Otoacoustic emissions evoked by bone-conduction stimuli
Authors: Yuan-yuan Lu , Qin Zhang
Published: October 2005

Bone-conduction stimuli evoked otoacoustic emissions could be clearly recorded in normal-hearing subjects. The sound probe was a 1/8-in. microphone mounted in a rubbery earplug inserted in the ear canal. A bone-conduct vibrator (RADIOEAR B-71, which was formerly used in an audiometer) as a stimulator, was applied to the surface of arcus of zygmaticus. Stimuli were generated by delivering 0.1-ms rectangular electrical pulses to the vibrator. With the exception of stimuli, the recording procedure was the same as Kemp's [J. Acoust. Soc. Am. 64, 1386–1391 (1978)]. By comparison in the same ear, both wave patterns and frequency spectra of the response evoked by bone conduction were quite similar to those evoked by click sounds at the stimulus levels approximate to the subjective thresholds. Because the frequency components above 800 Hz of skull vibrations go directly to the inner ear rather than being conducted through the middle ear bones, the finding of the present study provides new evidence to support the hypothesis that otoacoustic emissions originate from the cochlea. The new method of applying the stimulus also has the advantage of simplifying the structure of the detective probe and improving its frequency response.
Some measures of intelligibility of natural and coded speech for native and non-native speakers of English
Authors: Ozawa Kazunori , Logan John S.
Published: October 2005

Segmental intelligibility of both natural and coded speech was measured using the modified rhyme test (MRT). Pitch predictive multipulse excited speech coding (MPC) [K. Ozawa and T. Araseki, Proc. ICASSP86 33.3, 1689–1692 (1986)] was used to produce high-quality coded speech at 8 kb/s. As a reference, μ-law PCM was also used. To investigate not only perceptual differences between natural and coded speech but also the differences due to linguistic knowledge and experience with English, both native and non-native speakers of English were used as subjects. All of the stimuli were bandlimited through a low-pass filter with 4.5-kHz cutoff frequency and sampled at 10-kHz sampling frequency. Closed format MRT scores were obtained to measure the intelligibility for three conditions, natural speech, μ-law PCM and 8 kb/s MPC. Detailed analysis of the results of the intelligibility test will be presented. Differences between the intelligibility scores of native and non-native subjects and the relationship between these scores and their linguistic experience will also be discussed.
Speech enhancement and the instantaneous phase signal
Authors: Walsh S. J. , Clarkson P. M.
Published: October 2005

An approach to speech enhancement based on a representation of the speech signal by an envelope and instantaneous phase model is proposed. Envelope modification for the removal of room reverberation and noise from signals has already been studied and found to produce signal-to-noise ratio but not intelligibility improvements. The envelope alone does not account sufficiently for the detailed structure of speech. A noisy instantaneous phase has been found to have a perceptually degrading effect even with a perfectly restored envelope. Thus it is proposed to enhance speech by also replacing the measured instantaneous phase signal (in a number of contiguous frequency bands) by a modeled version. To determine the characteristics of the instantaneous phase function both synthetic and recorded speech have been studied. A simple model for the instantaneous phase of speech is proposed and an analytical solution is developed relating the digital model of speech production to an envelope and instantaneous phase model. Enhancement of a noisy instantaneous phase function can be achieved by using a compensating frequency shift, low-pass filtering and linear prediction.
Adaptive speech intelligibility testing for communication systems
Authors: Hogan D. L. , Gumas C. C. , Moore C. R. , Oshika B. T. , Hamlet S. L. , Pisoni D. B.
Published: October 2005

An automated adaptive interactive speech intelligibility test was implemented on an IBM-PC. Subjects hear digitized short sentences such as “grass is green” or “escalators have doors” over headphones and must key in a T or F response for each sentence. Quality of speech stimuli is controlled by different iterations of a degrading transfer function before playback. The level of degradation presented to the listener in successive trials is dependent on tbe just-previous pattern of accuracy in subject response. The test run is programmed to converge to that level of stimulus degradation at which the subject responds at a selected % correct criterion. The test is repeated for multiple % correct criteria, to permit not only assessment of a communication system under good intelligibility conditions but also measurement of sensitivity to reduced intelligibility. The approach permits flexible manipulation of parameters such as speech stimulus material, type of stimulus degradation, response time measurement, etc. The test is under computer control from initial instructions to the subjects to final summary of responses. The automated adaptive procedure has been evaluated for 100 normal hearing subjects. [Work supported by ESD, AF Systems Command.]
Absorption of ultrasound in denatured proteins
Authors: Kremkau Frederick W.
Published: October 2005

Absorption data have been obtained on nine globular proteins in aqueous solution denatured with 6M guanidine hydrochloride at neutral pH. The hypothesis that there is an underlying absorption behavior with respect to molecular weight similar to that for dextran [Kremkau and Cowgill, J. Acoust. Soc. Am. 76, 1330–1335 (1984)] was tested. If the scatter in previous data for native proteins is due to structure, denaturing would draw the points toward the underlying value. However, the scatter exists in the denatured protein absorption values as well. Various solvent contributions to absorption of the denatured proteins may cause the scatter. A second hypothesis was tested: secondary structure contributes to absorption in globular proteins but tertiary structure reduces this contribution because of solvent shielding [Kremkau and Cowgill, J. Acoust. Soc. Am. 77, 1217–1221 (1985)]. This hypothesis predicts that solvent buffer ions would increase absorption to a greater extent in denatured form (with many more side chains exposed to the solvent) than in native form. Such behavior was seen in some, but not all of the proteins. [Work supported by NCI, DHHS.]
Ultrasound absorption and attenuation loss in liver in the finite amplitude range
Authors: Fry F. J. , Dines K. , Etchison M.
Published: October 2005

The ultrasound absorption coefficient in fresh excised rat liver has been obtained at 1-MHz fundamental frequency in the intensity range (SPTP) from low intensity (50 W cm−2) to high intensity (800 W cm−2). At atmospheric pressure the pressure absorption coefficient starts at 0.035 (50 W cm−2) and rises to 0.090 (800 W cm−2). Theoretical calculations over this same intensity range show agreement in the shape of the absorption coefficient curve as a function of intensity up to approximately 450 W cm−2. Above this intensity the experimentally derived values flatten off and slightly decline with increasing intensity. The theory shows no such characteristic. The absorption coefficient was also measured in the same intensity range at a pressure of 350 lb in.−2. (This pressure provides a bias for 193 W cm−2 at 1 MHz.) There is a small (10%) and statistically significant reduction in the absorption coefficient in the intensity range from 200–500 W cm−2 but below and above this range there is no statistically significant difference in the absorption values from those obtained at atmospheric pressure. Attenuation losses show the same characteristic charges with intensity as does the absorption coefficient but the total attenuation loss is approximately twice that due to absorption alone. [Work supported by NIH grant 5R01 CA 41073-05.]
Ultrasonic absorption in mammalian ovaries
Authors: Carnes K. I. , Dunn F.
Published: October 2005

The human ovary, as well as that of other mammalian species, has the opportunity to receive ultrasonic exposure during diagnostic and therapeutic procedures. However, the ultrasonic absorption behavior seems never to have been investigated. Such information is considered essential for both safety and treatment planning. The ovary is a dynamic organ exhibiting interspecies variation as well as variation among the intraovarian structures, viz., the follicles, corpora lutea, etc. The in vitro ultrasonic absorption coefficient was determined, by the transient thermoelectric method, for ovaries of bovine, canine, feline, ovine, and murine species at 1 MHz. Similarities in values exist for intraovarian structures among the species measured. For example, of the stromal tissues, the outer cortex layer exhibits absorption coefficients approximately 50% greater than that of the inner medulla. The follicle absorption is the lowest and the corpora lutea values vary with the stage of development. These results are consistent with the macromolecular composition of these structures. [Work supported by the NIH.]
Frequency spectrum shifts of ultrasound echoes as a function of depth in a female breast
Authors: Kelly-Fry Elizabeth , Morris Steven T. , Sanghvi Narendra T. , Madsen Ernest L.
Published: October 2005

Knowledge of the changes in waveform characteristics as an ultrasound wave penetrates multiple layers of soft tissues within a breast can be important for both detection and diagnosis of breast pathologies by ultrasound mammography techniques. Using a spectrum analyzer system in combination with an automatic, bi-plane scanning, B-mode breast imaging unit, techniques were developed for recording the center frequency and bandwidth characteristics of signals received from the subcutaneous fat, mid-breast, retromammary, and pectoralis muscle regions of the breast of a post-menopausal subject. The same techniques were applied to a breast phantom that has internal architectural features and acoustic parameters (density, speed, attenuation coefficient) that are comparable to those of a normal breast. A single focus, 7.5-MHz transducer was used for most of the recorded data. Lower frequency transducers were also used. Spectral analysis data accompanied by relevant breast and phantom images will be presented. [Work supported by Indianapolis Center for Advanced Research and Labsonics, Inc.]
Intense focused ultrasound for producing complex volume lesions in brain
Authors: Fry Francis , Sanghvi N. T. , Morris Rich , Griffith S. , Hastings J.
Published: October 2005

The ultrasound dosage for producing single site focal lesions in the experimental animal brain has been well documented. Complex sheet lesions in white matter have been generated from a multiplicity of individual lesions at 1-MHz frequency without undesirable side effects in animal brains. At 4-MHz frequency, lesions in both white and gray matter involving a multiplicity of individual lesions have been generated without side effects in the animal brain. At 1-MHz frequency, the production of complex lesion shapes from a multiplicity of individual small lesions in both white and gray matter required establishing the lesion site intensity that would not initiate off primary site undesirable side effects. Intensities below 300 W cm−2 at the primary lesion site do not lead to such effects. A good working hypothesis for production of side effects involves microbubble formation at the primary lesion site. These microbubbles can be carried through the vascular circulation providing sites for cavitation formation outside the primary lesion site. It is now possible to prescribe primary lesion site intensities in brain in the frequency range from 1–4 MHz that negates the possibility of microbubble formation. [Work supported by Labsonics, Inc., Mooresville, IN.]
A novel approach of generating various frequency bands using a wideband PVDF ultrasound transducer and its application for tissue characterization
Authors: Sanghvi N. T. , Morris S. T. , Wendt P. W. , Kelly-Fry E.
Published: October 2005

The technological advances in electronics in recent years have made it possible to manufacture polyvinylidene fluoride (PVDF) ultrasound transducers with sensitivity that could be routinely used in clinical instruments for ultrasound imaging and diagnosis of tissue. Last year, we have taken a noval approach and developed electronics to excite PVDF transducers with a sharp spike of voltage with controlled energy-frequency spectrum to generate and transmit a selected frequency band with its center frequency totally controlled by the excitation pulse and selected filter. With this approach four different center frequencies with approximate band width of 45%–50% is generated. This paper will describe the analytical model and compare theoretical data with the experimental results. As a result of this development a prototype device is fabricated to study various breast tissue types and differentiate them based on their scattering characteristics as a function of different frequency bands. [“Bi-Plane Scanning with a High Frequency, Multi-Frequency Ultrasound Instrument for Improved Differentiation of Malignant and Benign Breast Tumors,” E. Kelly-Fry et al. submitted for publication in Radiology. J.]
Acoustically enhanced thermal diffusion
Authors: Thompson Charles
Published: October 2005

The role of acoustic vibration in the enhancement of the thermal diffusion process near a solid boundary is discussed. Our purpose is to discuss the degree with which flow unsteadiness can be exploited for the purposes of transport augmentation. There are three factors that govern augmentation: the behavior of the Stokes layer; unsteady flow instabilities; and the geometry of the surface boundary. The interelationship of these three factors on the augmentation process will be discussed and preliminary results of our analysis presented.
Piano string excitation: Theory and reality
Authors: Hall Donald E.
Published: October 2005

An idealized model of piano string excitation has been shown soluble by a combination of analytic and computer techniques. It is also useful to step aside from the mass of details this model generates, and discuss general physical arguments about what overall envelopes should limit the string vibration spectra produced. This also provides a framework for more careful consideration of the limits of validity of the theory. It is argued that the most important weakness is the non-Hookean nature of the hammer compliance, followed by string stiffness effects; possible ways of including these effects will be considered. Further laboratory measurements of string motion will be presented, especially bearing upon the question of whether striking the string a fraction 1/N of its length from one end does or does not cause mode number N to be missing from the spectrum. Measurements of radiated sound that would help clarify the role of the soundboard will also be considered.
New method of measuring input admittance of stringed instruments
Authors: Weinreich Gabriel , Yoder Levon
Published: October 2005

When a stringed instrument, such as a violin, is subjected to an applied time-harmonic force and the resulting velocity of the bridge is measured, the quantity usually referred to as the “input admittance” in fact functions as an output admittance. Its value can be inferred by loading the bridge with a known external impedance and comparing the velocities with and without this loading. In our experiment, the external loading impedance is a known mass, and the applied force results from an irradiating acoustic field. By taking data with two different field configurations, we can independently determine the four elements of the two-dimensional input admittance matrix referred to the two possible polarization directions of the string vibration. We utilize the digital data acquisition and processing system that has been previously developed for radiativity measurements, in which an appropriate broadband stimulating signal allows a large spectrum of data to be obtained in a few seconds. Some results will be shown. [Work supported by NSF.]
Boundary conditions and violin vibrations
Authors: Marshall Kenneth D.
Published: October 2005

In this paper the vibrational behavior of a violin is examined under a variety of boundary conditions; held by the player, freely suspended, pinned supports, simple supports, clamps applied to the neck and corpus, and supported by sandbags. It was found that three different players imposed very similar constraints on the instrument, with the principal effect being a significant increase in the modal damping as compared to the free vibrational response of the violin. The effect of the player on the modal frequencies was negligible. It was also found that resting the instrument on a piece of foam rubber or the use of sandbags under the neck and the tail of the instrument provided a good first approximation to the influence of the player. The other boundary conditions that were investigated caused significant deviations from the normal response of the instrument, thus indicating that these mounting techniques should not be used to measure the vibrational behavior of a violin or other stringed instrument.
Iranian scale of music can be considered as a universal scale
Authors: Barkeshli M.
Published: October 2005

Iranians divide the major tone in two: Limma (L) and a comma (C). Safi-yud-din (the Iranian savant virtuoso from the 13th century) was the pioneer of a school that propagated the systematic theory, taking L as unity for intervals, and divided the tone 9/8 into L L C. We see this division in the tablature of tunbur Khorasan, a typical instrument used in Iran and neighboring countries (description by Farabi, the Iranian philosopher and musicologist from the 10th century). Our researches on measuring the intervals of contemporary Iranian scale [“La Gamme de la Musique Iranienne,” Télécommunication, Paris 5, 5 (1950)] showed that there are also other divisions L C L and C L L in Iranian music. With this arrangement, the octave is divided into 27 intervals or 28 degrees. This scale can be considered as a universal scale because it contains all elements of harmonic and melodic scales. In fact the interval L + C, the characteristic of oriental music, is equal to 16/15 with the difference 1.0011292, being less than 1/50 of a semitone, a difference nondiscernable by the normal ear [V. O. Knudsen, Architectural Acoustics]. Also with the same difference 10/9 = 2L, 5/4 = 4L + C, and so on. [Work supported by ASI and PSI.]
An exploration of the Chinese pure-tone system
Authors: Pan Hai-si , Rao Yu-An
Published: October 2005

After 300 years of effort in many countries aimed at resolving the inharmonicity of the 12-interval equal-tempered scale, a pure scale of 53 tones was proposed in 1816–1890, but it can not be applied usefully owing to its very great complexity. A survey of ancient Chinese musical literature and instruments and deductions from “Jing-Fang's 60 Notes” of the Han Dynasty (209 BC-220 AD) and Emperor Liang-Wu's “Four Pass” scale (502–519 AD), led us to propose a 23-tone scale. Its error curve corresponds precisely with that of the 53-tone pure scale. The principle of Schisma is used for solving problems such as (a) that of the inharmonicity between Gong (do) and Jiao (mi), i.e., the major third; and (b) that the major and minor chords coexisted in an octave. We have derived a pure-temperament scale of 23 tones that can be modulated for every tonic and that is consistent with the modern pure scale of 53 tones. We show that this 23 unequal-interval pure scale is consistent with the intervals which were obtained by measurement from a stone flute made in the Qing Dynasty (1644–1908 AD).
A comparison of the musical scales of an ancient Chinese Bronze bell ensemble and the modern bamboo flute
Authors: Rao Yu-An , Carterette Edward C. , Wu Yu-Kui
Published: October 2005

We compare over a two-octave range the tonal, interval, and scale relationships of an ancient Chinese bell ensemble of the Zhou Dynasty and a modern bamboo flute. By using subjective pitches, a tonal system with simple harmonic partials (the flute) could be compared with a tonal system having complex inharmonic partials (the bell). We argue that the underlying musical scales of the bell ensemble and the flute are closely related and lie between an unequal-interval pure system scale and the just-intonation scale. There is some evidence that at least an interval of 60 or 66 cents figures in the flute scale, that both flute and bell scales include a 90-cent interval and a true (1200) cent octave, and that neither scale includes a 100-cent interval. Since there is a true octave the scale cannot be a cycle of fifths, and since there is no equal-temperament half-tone (100 cents), the scales cannot be of equal temperament. We conclude that the flute and Zhou bell scales are very similar if not the same. Apparently strong cultural traditions and human perceptual constancies united to sustain a common flexible musical scale during twenty-four hundred years.
Broadband acoustic source localization
Authors: Tolstoy A. , Porter M. B.
Published: October 2005

This paper examines aspects of matched field processing for locating a broadband source in deep water. Concentration is on nearby sources (within 5 km of a vertical receiving array), on frequencies from 100 to 300 Hz, and on deterministic, simulation “data” only. The noise field has been modeled as Gaussian, uncorrelated, white noise while the acoustic field has been modeled by the fast field program, with subsequent processing by conventional and maximum likelihood (ML) methods to generate range-depth ambiguity surfaces (AMSs). The effects of mismatch in the description of sea surface roughness have been examined and it has been found that it can lead to significant errors in peak location on an AMS, particularly at the higher frequencies. However, incoherent frequency averaging of AMSs can, in the presence of relatively small mismatch, overcome the effect of such mismatch, even though almost all the component frequencies show errors in their predictions. Moreover, consideration of range averaging plus frequency averaging can, under such conditions, improve those cases not resolved by frequency averaging alone. It is hypothesized that frequency and/or range averaging will help only when the mismatch is sufficiently small that the AMSs have at least local maxima near the true source location.
Source localization in almost-stratified waveguides
Authors: Shang E. C.
Published: October 2005

In our previous papers [E. C. Shang, C. S. Clay, and Y. Y. Wang, J. Acoust. Soc. Am. 78, 172 (1985); J. Acoust. Soc. Am. 77, 1413 (1985)], source localization techniques based on mode filtering have been developed for stratified waveguides. In this paper, some problems of source localization in almost-stratified waveguides are discussed. Specifically, the Prony method is proposed for source bearing in range-dependent waveguides. It is found that the Prony method only requires a “local almost-stratified” condition, which means that within the data sampling aperture length the field can be treated as adiabatic modes. For most of the practical interesting cases, this condition can be satisfied quite easy.
Extraction of average under-ice reflection amplitudes and phases with matched field processing
Authors: Livingston E. , Diachok O.
Published: October 2005

Average low-frequency under-ice reflection amplitudes and phases in the central Arctic were extracted from long-range (259 km) signals from fixed cw sources detected on a long (1 km) vertical array (the FRAM IV experiment) using conventional [H. Bucker, J. Acoust. Soc. Am. 59, 368 (1976)] and maximum likelihood [R. Fizell, J. Acoust. Soc. Am. (to be published)] matched field processing methods. Theoretical computation of amplitudes and phases for all assumed ranges and depths were based on the Porter-Reiss [J. Acoust. Soc. Am. 77, 1760–1767 (1985)] normal mode code. Under-ice reflection amplitudes and phases were incorporated into the propagation code and varied iteratively to achieve maximum signal gain and minimum range and depth errors. The resultant best data-fitting amplitudes and phases will be compared with expectations based on under-ice scattering theories and laboratory-scale model experiments.
Ventriloquism and spurious sound sources in underwater acoustics
Authors: Roebuck Ian
Published: October 2005

With the increased use of active noise control has come wider awareness that the source distributions generating prescribed sound fields are not unique. In particular, the possibility of precisely reproducing the field due to a time-varying monopole by a multipole source (of infinite order) located elsewhere, has been established. In this paper, the fundamental physical limitations of carrying out such an “underwater ventriloquism act” in practice—developing a “constructive” algorithm for the various multipole coefficients and criteria for truncating the process are examined. It is shown that this is closely related to earlier ideas on the effective size of “point” sources [I. Roebuck, J. Acoust. Soc. Am. Suppl. 1 69, S87 (1981)] and further that acoustic efficiency limits the extent to which the transmitting multipole elements can be compacted without destructive mutual interaction. The manner in which the greater effective size of the “spurious” multipole source limits the potential for deception in the presence of varying boundaries is also analyzed.
Passive synthetic arrays
Authors: Carey W. M.
Published: October 2005

Passive-synthetic apertures [W. Carey and N. Yen, J. Acoust. Soc. Am. Suppl. 1 75, S62 (1984)] were formed with experimental towed hydrophone data in a sound channel that supported RR and RSR transmission. These apertures were formed with lengths up to 95λ with coherent temporal processing gains approaching 0.75 of theoretical. These results and those of previous investigators [R. Fitzgerald, J. Acoust. Soc. Am. 60, 752–753 (1976); R. Williams, J. Acoust. Soc. Am. 60, 60–73 (1976)] indicate that synthetic apertures can be formed by the coherent summation of the phase-corrected summation of either hydrophone or subaperture beams over successive time samples when the synthetic aperture length is less than the spatial coherence length and the processing time is less than the temporal coherence length. The evaluation of synthetic apertures requires comparisons with conventional and other high-resolution techniques. Comparisons between conventional array processing and high-resolution techniques [maximum entropy (ME) and maximum likelihood (ML) methods] are performed by use of the analytical expressions developed by A. T. Parsons (AUWE, TN 700/83) for the determination of the array, integration, and the net processing gains. Analytical comparisons between conventional and synthetic aperture arrays formed with either the same number of hydrophones or with the same effective length but a different number of hydrophones, show that, when the spatial processing gain exceeds the loss in integration gain, then the use of synthetic apertures is advantageous. [Work performed while at NORDA.]
The generation of propeller modulation sounds for sonar trainers
Authors: McArthur Gary , Schmid Charles , Warnecke Mike , Atlas Les
Published: October 2005

Sounds of propeller modulation were synthesized using linear predictive coding (LPC) techniques resident in digital signal processing hardware for the purpose of training submarine sonar operators. The combination of LPC techniques and digital sound generation produces realistic propeller modulation sounds which can be conveniently varied by the instructor to characterize the type, speed, and depth of surface and underwater vehicles. The synthesized sounds were generated with digital LPC (all-pole) filters similar to those commonly used in speech synthesizers. The coefficients to drive the LPC filters were calculated from spectral models which were formulated from the cavitation index, KTIP. The cavitation index was calculated based on the instructor's commands of the vehicle type, speed, and depth. Three different LPC filter designs were implemented to evaluate the computational efficiency and sound realism of each approach. The first design was a standard 12th-order LPC filter which was not capable of changing the spectrum over short time periods. The second design was a low order LPC filter with coefficients which changed quickly with time. The last design was a baseband LPC filter modulated by a time-varying sinusoidal carrier. Various propeller modulation sounds will be played to demonstrate the quality and flexibility of these synthesizer techniques for training sonar operators. [Work performed in support of NUWES, Keyport, WA.]
Low-frequency ambient noise near Bermuda: Filling in the notch
Authors: Von Winkle William A. , Browning David G. , Christian Raymond J. , Cobb A. Donn
Published: October 2005

Ambient noise models which assume no significant wind-generated noise mechanism in the 5- to 100-Hz frequency range predict a notch in noise levels, between shipping and microseismic noise, at approximately 10 Hz. Data are presented for the frequency range 3–200 Hz that show that this notch fills in at increasing wind speeds and is entirely eliminated above 35 kn for a North Atlantic Ocean location near Bermuda. These results are consistent with the low-frequency wind-generated noise mechanism proposed by Isakovich and Kur'yanov [Sov. Phys. Acoust. 16, 49 (1970)]. [Work supported by NUSC.]
Species and size-class discrimination of selected fish using broadband sonar
Authors: Skvorc Paul A.
Published: October 2005

A sonar system was designed and built that could transmit and receive a broadband signal. This transmission takes the form of a chirp, ranging in frequency from 50–150 kHz. Four species of fish represented by ten size classes were used in a laboratory environment as targets. An acoustic signature was developed for each of the species and each of the size classes within each species. Pattern analysis techniques were then employed to test unknowns against these signatures. The system as a whole was capable of 95% accuracy in both species and size class discrimination. (Those fish misclassified to specie were considered to be misclassified to size irrespective of the relative sizes of the signature and unknown.) Size-class accuracy is approximately ± 10% of the fork length of the fish. Hydroacoustic and pattern recognition techniques are presented.
Nonuniformly poled piezoelectric transducers
Authors: Hazony Dov
Published: October 2005

A transducer model similar to Mason's model [W. P. Mason, Electromechanical Transducers and Wave Filters (Van Nostrand, New York, 1948)] is presented and discussed. Many examples will be shown including the case when the mechanical response of the transducer to an electrical impulse is also an impulse. The presentation is based on a forthcoming paper [D. Hazony, J. Acoust. Soc. Am. (to be published)]. Corroborative experimental results will be given. [Work supported by Gould, Inc., and Tecsonics, Inc.]
Electroacoustic characteristics of a 1–3 composite transducer
Authors: Thompson William
Published: October 2005

A 1–3 composite transducer was constructed by dicing a solid square block of PZT-4 to create a large number of thin parallel pillars; the cuts were then backfilled with epoxy. The device was calibrated as both source and receiver and its response was compared with that of the same size solid block. Over the frequency interval 10 to 300 kHz, the transmitting voltage response of the 1–3 composite is comparable in level to that of the solid block, except in the neighborhood of the length resonance of the solid block; moreover, it is a smoother curve. The open circuit receiving response of the composite is some 10 to 20 dB lower over this frequency interval that is attributed to the relatively large capacitance of the cables that were used. Directivity patterns of the 1–3 composite are better than those of the solid block and the availability of multiple outputs makes it possible to implement both amplitude shading for pattern control, and phasing or time delays for beamtilting; both of these features were demonstrated. [Work supported by ONR.]
Bandwidth limitations of the mechanical impedance of single degree of freedom electromechanical vibrators
Authors: Johnson Michael P.
Published: October 2005

All mechanically resonant transducer devices exhibit a minimum in their mechanical impedance at their fundamental resonance. This minimum in impedance can result in deterioration of electroacoustic performance with possibly disastrous results if left unchecked. The standard technique used to prevent this is attributed to Carson [J. Acoust. Soc. Am. 34, 1191–1196 (1962)] and essentially provides an electrically induced antiresonance near the mechanical resonance frequency. This technique results in two minima, one residing on each side of the previous minimum. The relative locations of these minima is dependent only on effective electromechanical coupling coefficient and the ratio of the electrical antiresonance to mechanical resonance frequencies. The bandwidth between these minima appears to have a practical limit of approximately an octave.
Electroacoustic mathematical model of a capacitive solid-state microphone structure
Authors: Baumhauer John C.
Published: October 2005

A semiconductor microphone structure is described in terms of the optimization of its micromachining parameters. Microscaling of dimensions is subject to a minimum signal-to-electrical noise ratio requirement, with an objective of small silicon diaphragm area. Side constraints include electrostatic stability, acceptable frequency response, and low-condenser bias voltage. Relations on electrostatic biasing and acoustic sensitivity are presented. A lumped parameter model of the silicon diaphragm was justified for the equivalent circuit. An electrical noise model is described. Scaling to a submicron air film thickness was required to meet the above criteria. A novel (patented) acoustic hole array through the backplate overcomes the resulting (excessive) air squeeze-film damping (i.e., resistance). The array of 2000 holes, nevertheless, consumes a small fraction of the capacitive area. Analyses for mechanical vibration pickup sensitivity and perforated backplate mechanical stability are briefly discussed.