bluuebunny/crossref_metadata_2025 huggingface dataset:
The musician as programmer
Authors: Abbott Curtis
Published: October 2005

Computers offer the possibility to musicians of creating sounds, instruments, performance styles, and interpretive behaviors with relatively few a priori constraints. We discuss the definition of these terms in the perspective of computer music, and the fact that realization of these utopian possibilities requires the musician to act in some sense as a programmer. The main body of the paper is concerned with issues in the design of programming languages for musicians. As the organization of time is one of the most salient and uniquely musical aspects of music, we will be especially concerned with the problem of time in such languages. A programming language for musicians developed by the author at IRCAM will be described.
Synthesis of bowed strings
Authors: Smith Julius O.
Published: October 2005

A model for the bowed string is derived which is inexpensive to implement in digital hardware. Since it requires a pulse-train input, it is ideal for use in conjunction with automatic signal modeling such as linear prediction and system identification methods. Due to the physical nature of the model, it is typically easy to add subtle refinements based on advances in the understanding of bowed strings. [Work supported by a Hertz Foundation graduate fellowship.]
The synthesizer hardware of a commercial digital musical instrument
Authors: Alonso Sydney A.
Published: October 2005

Synclavier® II is a musical instrument which employs a digital sound synthesizer controlled by a minicomputer. A variety of inputs (keyboard, push button switches, knob, foot pedals, etc.) allow real-time musical performance. The digital synthesizer hardware consists of one or more units of eight-channel FM oscillators. Each such synthesizer unit is a time-division multiplexed, pipelined special purpose computer containing a sample rate generator, phase angle incrementer, frequency modulator, phase adder, 32 wave memorys, modulation index interpolator, amplitude envelope interpolator, volume register, and digital-to-analog converters. The most unique features of the design are: the method of sample rate generation which produces more than 8000 discrete frequencies per octave; and the method of digital to analog conversion which achieves the equivalent performance of a 24-bit converter in musical synthesis application.
The systems concepts digital synthesizer at CCRMA
Authors: Samson Peter R.
Published: October 2005

The Systems Concepts Digital Synthesizer is a large (2500 integrated circuits), powerful (20 million multiplications per second) machine designed to synthesize complex musical pieces in real time. To accomplish this, it has several computational elements which operate simultaneously. The requirement of real-time synthesis necessitates special-purpose structures in the architecture of the Synthesizer, as well as general-purpose ones. Since the Synthesizer was put in service at the Stanford University Center for Computer Research in Music and Acoustics, it has received intensive use by many composers, employing a wide variety of musical idioms. Illustrative excerpts are presented.
High quality music from digital signal processors
Authors: Burr D. J.
Published: October 2005

Low-cost, high-quality sound synthesis is now possible in real time with digital integrated signal processors. In particular, a program for music synthesis using the Bell Laboratories Digital Signal Processor (A3990 DSP) is described. The program is based on frequency modulation (FM) synthesis and is capable of producing a single FM voice at 20-kHz bandwidth or several such voices at reduced bandwidth. The advantages and limitations of available digital signal processor chips in relation to music synthesis are given. In summary, suggestions are presented to improve future versions of such devices.
The control channels of instrumental playing in computer music
Authors: Luciani A. , Cadoz C. , Florens J. L.
Published: October 2005

We are concerned here especially with the instrument/instrumentist relation in the playing time, more generally in musical instrumental experiments. Three channels at least support this relation: the gestual channel, the acoustical, and the visual ones. We shall first propose a comparative analysis of them as “intentional” ways or control ways. Schematically, we can say that gestual and visual channels are devoted to control purposes, whereas the acoustic channel bears expressive intentionality. In traditional practice all these functions are implicitly assimilated and coordinated during primitive learning. In the context of real-time digital synthesis of sounds, we have to consider explicitly these different aspects. We shall propose a preliminary study of the specificity and complimentarity of these channels. Then we shall present the problems set by specific transducers to be built and connected to real-time synthesizers. Concerning gestual control we shall introduce a typology of instrumental gesture taking into account its bilateral aspect (as transmitter and as receiver). We shall describe corresponding “gestual transducers with mechanical feedback.” Concerning visual control we shall present a real-time modelization of musical instruments corresponding to their pertinent visual elements during playing. All these studies are correlated with the CORDIS system developed by ACROE research group in Grenoble (C. Cadoz et al.).
Development of speech research software on the MIT lisp machine
Authors: Shipman David W.
Published: October 2005

Study of the acoustic-phonetic properties of speech is limited by the considerable overhead involved in the recording, transcription, measurement, and statistical analysis of speech tokens. Conventional computer speech facilities ease this burden by providing storage, signal processing, and display functions. Spire, for Speech and Phonetics Interactive Research Environment, extends this basic functionality by providing a highly human-engineered user interface. The goal is to dramatically decrease the time required to perform phonetic studies by increasing the ease and flexibility of user and program interaction. Based on the MIT Lisp Machine, the system utilizes a high-resolution bit-raster-scan graphics display (100 pixels per linear inch) and hand-held “mouse” pointer as its primary means of user interface. Spire's current capabilities include coordinated computation and display of spectrograms, spectral slices, LPC parameters, pitch, formants, transcriptions, and other data. Cursors on the various display windows are synchronized and multiple user-formatted layouts are available. Future capabilities include statistical displays, phonetic retrieval from large databases, and semi-automatic transcription aids. A typical interaction with the system will be demonstrated. [Sponsored in part by U.S. Air Force Contract F30602-78-C-0120.]
Simulated nerve fiber activity in response to one- or two-formant speech-like sound
Authors: Dolmazon J. M. , Boulogne M.
Published: October 2005

Representations of speech-like sounds in discharge patterns of auditory nerve fibers were studied by mean of a model. At a previous meeting we presented the model of the peripheral auditory system developed in our laboratory. The model was shown to reproduce the excitation patterns of single auditory nerve fibers in response to one-tone, two-tone, trapezoidal, or impulse stimuli. We present new results obtained in response to one-and two-formant stimuli. Time patterns of simulated fibers are studied with Fourier analysis to elicit the frequency components involved in each response. Special care is taken to fundamental frequency representation. The main conclusion is that fundamental frequency can be present in all units regardless of the tuning properties cf the fibers. All the results obtained are compared with available experimental data. [Work supported by A.D.I. convention No 80/379.]
Time-varying features in stops produced at different speaking rates
Authors: Kewley-Port Diane , Luce Paul A.
Published: October 2005

Recent work has demonstrated that place of articulation in initial voiced stops can be accurately identified from visual displays of linear prediction smoothed spectra updated at 5-ms intervals [Kewley-Port, Res. Speech Percept., Tech. Rep. No. 3, Indiana University (1980)]. The present research extends this analysis in several directions. The visual displays were modified to incorporate auditory filtering and other frequency characteristics of the ear. Our earlier time-varying features were redefined to describe syllable initial stops in fluent speech. The present experiment examined syllables beginning with voiced and voiceless stops paired with five vowels. Two males and two females read the syllables in a carrier phrase at fast, normal and slow tempos. Employing a predefined set of time-varying features, judges first attempted to locate the burst and then to identify place of articulation in the visual displays. The results will be discussed in terms of possible strategies for speech recognition in fluent running speech using acoustic processing techniques based on the psychophysical properties of the auditory system. [Supported by NINCDS Grant NS-12179.]
Can a human speaker reproduce Fant's nomograms?
Authors: Ladefoged Peter , Bladon Anthony
Published: October 2005

The authors attempted to simulate the articulations modeled by G. Fant [The Acoustic Theory of Speech Production (Mouton, The Hague, 1962), p. 82] in his description of the relation between articulatory movements and formant frequencies. They each produced sets of vowels with an articulatory constriction which was intended to have a constant size at varying distances from the glottis. Additional vowels were also made with constant tongue positions but varying lip positions. The first four formants of these vowels were plotted in a way comparable with the data in Fant's nomograms. There were three kinds of discrepancies: (1) It appears that human speakers can produce vowels only over a range that is less than half that represented in Fant's nomograms. (2) When the constriction is moved progressively further away from the glottis the frequency of F2 does not decrease as it does in the most forward part of Fant's nomograms. (3) Variations in lip rounding affect front and back vowels differently with no gradual change from one class to the other.
Anticipatory lip rounding
Authors: Gelfer Carole E. , Harris Katherine S. , Bell-Berti Fredericka
Published: October 2005

Two accounts of coarticulation, (1) that the anticipation of segmental gestures is determined primarily according to the compatibility of the feature specifications for preceding and anticipated phones, and (2) that the extent of anticipatory gestures is delimited according to relatively invariant temporal specifications, yield very different predictions regarding articulatory organization. We tested these predictions via simultaneous EMG, movement and acoustic recordings for two speakers who produced /iCnV2/ utterances, where V2 was either /i/ or /u/ and Cn was /t/, /s/, /st/ and various combinations thereof. For the first speaker, onset of /u/-related EMG activity for orbicularis oris occurred systematically relative to the acoustic onset of that vowel, thus providing support for a model whereby anticipatory gestures are temporally organized. For the second speaker, however, lip rounding activity was not related in any direct way to either the acoustic onset of /u/ or to consonant string length and/or feature composition. Thus, no simple physiological account adequately explains the results obtained for this speaker. Analysis of movement and acoustic data are currently underway. [Work supported by NINCDS and BRS grants.]
Lingual coarticulation in VCV sequences
Authors: Parush Avraham , Ostry David J.
Published: October 2005

Pulsed-echo ultrasound was used to study coarticulation in tongue dorsum movements during the production of VCV sequences. The stimuli were velar stop consonants /k/ and /g/ combined with posterior vowels /a/, /o/, and /u/. Vertical displacement, maximum velocity, and duration were obtained for all V1C and CV2 transitions. Right-left (effect of CV2 on V1C) and left-right (effect of V1C on CV2) coarticulation Was examined by stepwise multiple regression. Highly significant effects were obtained with /a/. Total duration of V1C was negatively related both to displacement from linguo-palatal contact to the point of maximum velocity and to total displacement of CV2 (RL effects). Total displacement and maximum velocity of CV2 were negatively related to the displacement to the point of maximum velocity and total displacement of V1C (LR effects). The LR effects of V1C on CV2 were larger than the corresponding RL effects. Similar patterns were obtained for both intervocalic consonants. The findings are discussed in terms of the contributions of mechano-inertial and motor timing effects to coarticulation.
An acoustic and perceptual study of R-L versus L-R coarticulation
Authors: Mount Kay H.
Published: October 2005

Coarticulation has been observed in physiological, acoustic, and perceptual investigations. However, conflicting evidence exists at each level of observation concerning the relative importance of right to left (R-L) versus left to right (L-R) coarticulation. Differing speech samples and research methods may account for conflicting data. To test this hypothesis, effects of R-L and L-R coarticulation were examined in methodologically related acoustic and perceptual factorial experiments. Eighty-four 60-ms vowel segments devoid of transitions were extracted from three isolated vowels, 27 ′V, CV,27 ,V′CV, and 9 CVC utterances and were examined for change in F2 of contextual [i,ɑ,u] and perceptual recovery of adjacent [p, t, k]. Listeners were 18 phonetically naive college students. No simple perceptual or acoustic advantage of one coarticulatory direction over the other was found. Instead, coarticulatory direction interacted with stress to affect consonant recovery and change in F2. This finding led to speculation that listeners may have responded to cues common to stress and consonant environment including amplitude and fundamental frequency rather than to frequency information provided by F2.
FreqUency analyses of syllable initial and final liquids spoken by American English talkers
Authors: Syrdal Ann K.
Published: October 2005

First, second, and third formant frequencies measured during the relatively steady-state portion of the liquid were converted to the Bark scale, an auditory scale of perceived pitch in units of critical bands. Analyses were performed on previously reported acoustic data derived from spectrographic measurements of liquids produced by female and male adult and young children talkers [I. Lehiste, U. of Michigan Comm. Sciences Lab. Report No. 9 (1962), and R. M. Dalston, J. Acoust. Soc. Am. 57, 462–469 (1975)], and on formant frequencies of adult talkers estimated by linear prediction techniques. The differences between Bark-transformed F3 and F2 and between F2 and F1 were calculated. F3−F2 differences greater than 3.5 Bark were consistently found for /l/, while those less than 3.5 Bark were found for /r/. Syllable initial and final allophones of /l/ were differentiated by F2−F1 differences greater than and less than 3.5 Bark, respectively. These results confirm the importance of a critical formant frequency distance of about 3.5 Bark found in perceptual experiments by the author with synthetic liquids, and by L. A. Chistovich and V. V. Lublinskaya for synthetic Russian vowels [Hearing Res. 1, 185–194 (1979)]. [Work supported by NIH and performed in part at the Research Lab. of Electronics, M.I.T.]
Dynamic analysis of Japanese vowels using articulatory model
Authors: Ishizaki Shun
Published: October 2005

The dynamic analysis starts from area functions obtained directly by analyzing speech waveforms, where the adaptive inverse filtering method [T. Nakajima et al., SCS Stockholm, 1–3 Aug. (1974)] is adopted. Applying the articulatory model [S. Ishizaki, Bull. Electrotech. Lab. 42, Nos. 11 & 12 (1978)] to the area functions, articulatory parameters such as the tongue position, the lip opening area and the vocal tract length are extracted. In continuous Japanese vowels' analysis, the parameters at the quasistationary part of the vowels are obtained. The parameters representing the articulatory organ movements are enhanced linearly to estimate target positions of the movements [S. Ishizaki, 6th IJCAI 1, 422–424 (1979)]. The enhanced articulatory positions are shown to move on toward the positions for each vowel which are obtained from isolated vowels. Such dynamic enhancement is examined in the formant domain and the result is shown to be corresponding with the articulatory one.
Frequency analyses of American English vowels
Authors: Syrdal Ann K.
Published: October 2005

Analyses were made of acoustic data derived from spectrographic measurements of ten American English vowels spoken by 15 children, 33 men, and 28 women talkers [G. E. Peterson and H. L. Barney, J. Acoust. Soc. Am. 24, 175–184 (1952)]. Frequencies of F0, F1, F2, and F3 were converted to the Bark scale, an auditory scale of perceived pitch in units of critical bands. A critical distance between formants of about 3.5 Bark has been reported to be important in the perception of synthetic Russian vowel quality [L. A. Chistovich, R. L. Sheikin, and V. V. Lublinskaya, Frontiers Speech Commun. Res., 143–157 (1979)]. Three distance measures were calculated from the Bark-transformed Peterson and Barney data: F1−F0, F2−F1, and F3−F2. Each of the three distance measures was binarily classified according to whether or not frequency differences exceeded 3.5 Bark. These binary classifications related to the linguistic dimensions of vowel height, compactness, and place of articulation. The procedure also served as a normalization technique. [Work supported by NIH and performed in part at the Research Lab. of Electronics, MIT.]
Aspirin can potentiate noise-induced temporary threshold shift
Authors: McFadden Dennis , Plattsmier H. S.
Published: October 2005

Temporary threshold shift (TTS) was measured in the same subjects under various schedules and durations of aspirin usage. The exposure stimulus was a 2500-Hz tone of 10-min duration; its intensity was determined for each subject individually in an initial series of sessions conducted without aspirin. TTS was monitored only at 3550 Hz. The highest aspirin dose was 4 grams per day for 2.75 days—a schedule not uncommon among sufferers from arthritis; the lowest dose was 0.49 grams taken 2 h before the exposure. In some condition(s) for all subjects the aspirin itself decreased hearing sensitivity; the typical pattern was of greater loss for the higher frequencies than for the middle frequencies, and no loss below 1000 Hz. Whenever there was aspirin-induced hearing loss, the loss induced by the exposure added to it, producing a total loss greater than that in the no-drug condition; occasionally the total loss was greater even in the absence of a measureable aspirin-induced loss. The potentiation was often in excess of 10 dB. Most recovery curves appeared to be similar in slope for drug and no-drug conditions. Chronic users of aspirin should be cautioned against excessive exposure to intense sounds. [Work supported by NINCDS and Merck, Sharp and Dohme.]
Binaural implications in help for certain tinnitus sufferers
Authors: Weiler Ernest M. , Grabowski Richard
Published: October 2005

A statistically significant proportion (37 out of 55) of clients were helped to obtain relief from tinnitus by a program of masking and residual inhibition. Although no single variable of history or audiometric tests separated those helped from those not helped, a multiple discriminant analysis did yield significant results. The properly weighted combination of right versus left ear losses accounted for a majority of the successful treatments. The authors interpret this as indicating binaural effects in certain tinnitus cases, which agrees with several nearly simultaneous observations reported by others.
Measuring the intensity-response function in individuals with sensori-neural hearing loss
Authors: Yund E. William , Simon Helen J. , Efron Robert
Published: October 2005

A model was developed to explain the ear dominance and intensity independence found in studies of the perceived pitch of simultaneous dichotic pure tone stimuli [E. W. Yund and R. Efron, J. Acoust. Soc. Am. 62, 607–617 (1977)]. One stage of this model corresponds to the intensity-response (I-R) transduction mechanism occurring in the cochlea. In previous work with normal-hearing subjects, I-R functions measured in one set of dichotic experiments were used to predict the results of another set of experiments [R. Efron, E. W. Yund, and P. L. Divenyi, J. Acoust. Soc. Am. 66, 75–86 (1979)]. The purpose of the current study is to extend the measurements of these frequency-specific I-R functions to the ear with sensorineural hearing loss. The slopes of the measured I-R functions for the hearing-loss subjects change more rapidly with intensity than do the I-R slopes of normal-hearing subjects. The I-R functions of hearing-loss subjects also tend to have their steepest slopes at higher intensities (80 to 100 dB SPL) while the maximum slopes for normal-hearing subjects occur below 80 dB SPL. The possible significance of these results to the design and fitting of hearing aids will be discussed. [Work supported by VA RER&D.]
Performance on an auditory backward recognition-masking task by elderly and young subjects
Authors: Newman Craig W. , Spitzer Jaclyn B.
Published: October 2005

The rate of processing pure-tone stimuli was measured in an elderly (75–85 years) sample and young (20–29 years) control group through application of an auditory backward recognition-masking task. Trials consisted of one of two equi-probable target stimuli (TS) followed by a masking stimulus (MS) after a variable silent inter-tone interval (ITI). Subjects were required to identify the 770 Hz TS as low and the 870 Hz TS as high. Ninety-six pitch identifications were made at each ITI. For both groups, the mean percent correct recognition scores improved systematically with increases in ITIs, reaching an asymptote followed by an apparent plateau. The mean recognition scores at each ITI were lower for the elderly subjects than for the controls; however, equivalence between groups was observed at prolonged ITIs. The elderly group reached asymptote at a longer ITI (360 ms) than the: controls (240 ms), a statistically significant difference. The findings were interpreted as suggestive of a slowing in auditory information processing in the elderly sample in comparison to the young controls.
The perceptual reality of tone chroma in early infancy
Authors: Demany Laurent , Armand Françoise
Published: October 2005

In the framework of a habituation-recovery paradigm, we presented 40 infants, aged 70–110 days, two cyclically repeating melodic sequences of pure tones. The habituation sequence, A B C A B C …, was the same for all infants; its component tones, A, B, and C, had a level of 80 dB SPL and respective frequencies of 736.7, 487.4, and 428.1 Hz. B and C were replaced by X and Y in the test sequence, A X Y A X Y …, which had three different versions, heard by three separate groups of infants. For each group, X and Y were 85 dB SPL and X Y was an exact musical transposition of B C at lower frequencies. The interval between B and X (and C and Y) was equal to 1003 cents for group 1 (N = 12), 1200 cents—i.e., one octave—for group 2 (N = 16), and 1389 cents for group 3 (N = 12). Significant reactions to the sequence change were observed in groups 1 and 3, but not in group 2. Young infants would therefore seem to perceive as strongly similar two pure tones one octave apart. This tends to support nativist explanations of the perception of tone chroma.
Optical fiber path differential wideband interferometer
Authors: Garg Avinash O. , Claus Richard O.
Published: October 2005

The performance of a wideband differential interferometric system which uses single mode optical fiber to transmit coherent light from input optics to a surface which supports ultrasonic waves is described. Polarized light from a 2.0-mW helium-neon laser source is divided and coupled to two flexible bundled single mode optical fibers which transmit the light to a small remote detection head. The light at the output ends of the fibers is collimated and focused by a variable focus lens system to points on the surface of a specimen to be inspected. Elastic waves on the specimen surface differentially modulate the relative phases of the two optical beams due to the periodic change in particle displacement at the surface. Upon reflection, the beams are superimposed, filtered, and detected to produce an output signal directly proportional to instantaneous displacement. The acoustic sensitivity of the interferometer may be adjusted by changing the separation distance between the spots on the surface; a 3-dB bandwidth of two frequency decades while maintaining a 20 dB SNR is possible using frequency domain signal processing techniques. Applications which specifically take advantage of the flexibility allowed by the remote interferometer sensing head are described. [Work partially supported by NSF and NASA.]
Measurement of a Tonpilz transducer in terms of a lumped-parameter equivalent circuit model
Authors: Schafer Mark E. , Hughes W. Jack
Published: October 2005

An in-air measurement procedure has been developed for determining element values in the lumped-parameter equivalent circuit model for a Tonpilz-type transducer. The technique first finds the electrical immitance of the transducer at the resonant, antiresonant, and half-power frequencies, to determine the shunt electrical elements and the motional elements. The electromechanical transformation factor is derived from measurements of the radiating face velocity and the input voltage and current of the transducer at resonance. Once the transformation factor is found, calculated acoustical radiation impedance may be accurately incorporated into the circuit. The utility of the technique lies in the ability to predict the response of the transducer to any acoustic load. Thus, the effects of variations in the acoustic load caused by mutual interaction effects in arrays can be predicted. Results will be shown which indicate the accuracy of the model, and which demonstrate the variations in the transformation factor with frequency. [Work supported by NAVSEA.]
Geoacoustic models for propagation modeling in shallow water
Authors: Chapman D. M. F. , Ellis Dale D.
Published: October 2005

Acoustic propagation in shallow water is viewed as a guided-wave phenomenon, with the sea surface and seabed forming the boundaries. At subkilohertz frequencies, the acoustic properties of the seabed to a depth of several wavelengths can have a strong effect on propagation. The computer modeling of propagation requires estimates of such parameters as sound speed, density, attenuation, and layer thicknesses, which collectively are called the geoacoustic model of the seabed. Direct measurement of these quantities is difficult, and methods must be devised to infer these values from other experiments, often employing acoustic techniques. At DREA, we have adopted the approach of independently determining as many geoacoustic parameters as possible, and adjusting less precisely known parameters within reasonable limits to effect an agreement between theory and experiment. To this end, we have used sub-bottom reflection profiles to determine sediment types and layer thicknesses, large and small scale seismic refraction experiments to estimate sound speeds, and processing of sub-bottom vertical reflection data to estimate volume attenuation. Techniques used by other researchers will be reviewed. Examples of geoacoustic models and comparisons with experiment will be presented for shallow water sites on the Scotian Shelf and the southwestern approaches to the English Channel.
Sonic gas purity monitor for helium recovery/re-liquification systems
Authors: Garrett S. L. , Swift G. W. , Packard R. E.
Published: October 2005

One common problem with helium recovery systems is the inadvertent introduction of air into the recovery lines. We have constructed a device which determines the purity of the recovered gas by measuring sound speed. This is accomplished by tracking the frequency of the fundamental plane wave resonance of the helium-air mixture in a cylindrical resonator. An electronic feedback circuit maintains the resonance and a “tachometer chip” converts the frequency to a voltage permitting a continuous, direct analog display of the percent of air contamination. Because the temperature of the gas recovered from the cryogenic liquid can vary substantially, an electrical resistance thermometer, mounted on the resonator, automatically compensates for changes in gas temperature. The total cost of all necessary electronics including integrated circuits, chassis, power supply, connectors, meter, alarm, etc., is under $100. Air contamination as small as 0.1% is easily detectable. [Work supported by N.S.F. DMR 79-25098 and a fellowship (S.L.G.) from the Miller Institute for Basic Research in Science.]
Some often encountered but less familiar aspects of sound level measurements
Authors: Wong G. S. K.
Published: October 2005

This paper discusses the performance of rms detectors in sound level measuring instruments and provides estimates on the effects of employing a type 1 sound level meter with a crest factor (CF) capability of 3 to measure levels with signal CF = 5 or more. The discussion includes errors due to tolerances of the A-weighting, and a method involving 1/3-octave measurements for the computation of noise exposure and its related descriptors. The proposed method enables the user to modify earlier results to obtain new answers compatible with the latest or future research without the necessity of repeating the acoustical measurements.
Acoustics of teleconferencing: Problems and solutions
Authors: Botros Radamis
Published: October 2005

Acoustics of telephony remained, for decades, the occupation of generations of scientists and engineers, including Georg yon Békésy, who won the Nobel Prize for his life-long study of the mechanism of hearing. In the 50's and 60's the introduction of loudspeaking telephony represented a new challenge: Room acoustics became part of the telecommunications chain and phenomena like the “rain-barrel” effect or signal coloration started drawing the attention and efforts of researchers. In the 70's the new concept of teleconferencing started penetrating business life. Both corporate decentralization within one country and the geographical distribution of “Multi-nationals” emphasized the need for holding meetings among many individuals or groups, using the world-wide telephone network as medium, in all transmission modes: terrestrial, submarine, and satellite links. New dimensions were now adding to the complexity of the problem: The need for keeping large teleconferences within the standards for telephone transmission and reducing the effects of disparity of quality among interconnected lines and terminals. This paper deals with the acoustic problems inherent in a teleconferencing terminal, reviews solutions adopted thus far, and describes results of recent research work as well as predictions for the near future.
Effects of linguistic context on the durations of syntactic categories
Authors: Charles-Luce Jan , Walker Laurie Ann
Published: October 2005

Miller and Isard [J. Verbal Learning and Verbal Behavior (1963)] showed that when subjects are required to shadow grammatical, anomalous, and ungrammatical sentences, percent correct shadowing increased as a function of the presence of syntactic and semantic constraints in the materials. These perceptual results suggest that production of speech may also be differentially affected by the degree to which syntactic and semantic information is used by the talker. In the present study, subjects read grammatical, anomalous, and ungrammatical sentences presented in isolation. Measurements of word length were then made for six different syntactic categories. Results show that for all categories, except nouns, words in grammatical sentences were shorter than the same words in either anomalous or ungrammatical sentences and that words in anomalous sentences were shorter than the same words in ungrammatical sentences. Nouns remained constant in duration across all three sentence contexts. Results from passages of connected speech will also be presented. [Work supported by grants from NIMH and NINCDS.]
Sentence intonation in British and American English
Authors: Lieberman Philip , Landahl Karen , Ryalls John H.
Published: October 2005

Recent theories for sentence intonation have proposed quantitative metrics for the F0 contours of sentences. Declination theories have claimed that (1) the “top line” of F0 peaks falls throughout a sentence, (2) the top line peaks at some specified time and then falls. A test of these theories is described using as a corpus the F0 contours that were published in the Lebendige Sprache monograph series between 1937 and 1940. These experimental studies derived F0 contours for speakers of British (RP) and American English reading stories, delivering political speeches, in theatrical performances, and in conversations. Neither claim (1) nor (2) is characteristic of most sentences in the corpus. Although F0 declinations occurred for some sentences for some speakers, other sentences were best characterized by relatively level F0 contours marked by F0 peaks corresponding to phonetic prominence and abrupt, sentence-terminal F0 falls. In contrast to the claims of declination theories, F0 contours that gradually rose throughout a declarative sentence occurred frequently. Certain F0 phenomenona appear to be “discourse” effects that introduce “new topics.”
Another look at reiterant speech in prosody research
Authors: Larkey Leah S.
Published: October 2005

Reiterant speech or nonsense syllable mimicry [Liberman and Streeter, J. Acoust. Soc. Am. 63, 231–233 (1978)] is useful for studying prosodic phenomena if it can be shown to preserve the suprasegmental duration and intonation differences relevant to perception, while neutralizing the intrinsic segmental duration differences normally found in speech. Syllable durations for normal and reiterant ambiguous sentences were measured to seek evidence of the duration differences which can enable listeners to resolve surface structure ambiguities in normal speech. These duration patterns were found in both normal and reiterant sentences produced by “good” speakers—speakers who neutralized intrinsic duration differences and whose sentences were independently rated by skilled listeners as good imitations of normal speech. Further, listeners were able to choose the correct interpretation when the ambiguous sentences were in reiterant form as well as they did when the sentences were spoken normally. [Supported by NIH and the University of Minnesota.]
Comparison between perturbative and exact treatment of bottom attenuation for shallow-water, low-frequency conditions
Authors: White DeWayne , Chin-Bing Stanley A.
Published: October 2005

A comparison between treating bottom attenuation perturbatively and exactly in a normal-mode expansion has been made for a representative shallow-water, low-frequency problem. The exact treatment introduces bottom attenuation through complex sound speed and thus complex depth functions, whereas the perturbative approach uses real valued depth functions and introduces mode attenuation only in the range function. Cases examined indicate transmission loss versus range calculations resulting from the two approaches disagree significantly near cutoff. Comparison with FFP calculations indicates the exact (complex depth function) solution is correct. When more than one mode is present, the other modes dominate the mode nearest cutoff; and although this mode is in error, it does not seriously affect the results. For those cases where the perturbative and exact solutions significantly differ, the error in the perturbative approach is mostly due to an incorrect normalizing factor; mode attenuation differences also account for part of the error. Results are discussed and a correction to the mode normalization factor derived from the perturbative solution is suggested.
Photoacoustic generations and applications
Authors: Tam Andrew C.
Published: October 2005

A wide variety of acoustic pulses (different magnitudes and temporal/frequency profiles) can be produced in a sample by optical absorption of various light beams, i.e., photoacoustic (PA) pulse generation. Various generation mechanisms are possible: thermal expansion, gas evolution, chemical changes, dielectric breakdown, electrostriction, etc. PA pulse generation has found several important applications, including ultrasensitive spectroscopic detections of linear and nonlinear optical absorptions, noncontact ultrasonic measurements in hostile environments, PA imaging techniques, PA operation of mechanical devices, and so on. This paper presents some basic understanding of a PA generation process, and examples of important recent applications.
Nearfield of an arbitrary piston in an infinite rigid baffle
Authors: Lubell Alan H.
Published: October 2005

Numerical calculations of the nearfield of several types of under-water radiators made in 1965 for NAA Columbus led to this author's rediscovery of the edge integral for pistons in an infinite rigid baffle. Interest in this very efficient numerical method applicable to any shape of plane piston was rekindled in conjunction with an ongoing investigation of the impedance of a plane symmetric axisymmetric stepped piston radiator (LL model 98 underwater loudspeaker). A program was devised for the TRS-80 Mod III computer and Used to duplicate a recently published absolute pressure field map for the circular piston [New et al., J. Acoust. Soc. Am. 70, 1518–1526 (1981)]. The agreement between the maps will be discussed as well as the cost of computation to a specified accuracy. Real and imaginary components of pressure on the piston face will be given for several values of ka. The value of the edge integral method for understanding certain aspects of diffraction theory more fundamentally than is possible with the Helmholtz integral theorem will be touched on.
Vibration and acoustic response of a damped circular plate in an infinite baffle
Authors: Latcha Michael , Akay Adnan
Published: October 2005

Transient sound radiation from an impactively excited circular plate clamped at the outer edge in an infinite baffle is obtained by numerical integration of Rayleigh's integral. Calculated sound pressure wave-forms are compared with the experimental results to demonstrate the sound radiation mechanism due to “rapid deformation” of the plate. These results show that on the axis of symmetry perpendicular to the plate, the sound pressure wave reproduces the plate velocity, as opposed to the classic case of a transiently excited rigid piston in an infinite baffle which reproduces the piston acceleration.
Waveform shape near Mach cone arrival for the pressure perturbation created by a modulated laser beam moving over a water surface at supersonic speed
Authors: Pierce Allan D.
Published: October 2005

Previous work by Lyamshev, Kolomenskii, Bozhkov, and others on acousto-optical sound generation is extended to a detailed study of the pressure perturbation following Mach cone arrival. The solution of the Westerfelt-Larson equation for sound generated by laser beam heating is presented in integral form using the Green's function for a supersonically moving point impulsive source. The laser beam points downward into the water, has a Gaussian beam shape (radius a), and moves rectilinearly with supersonic speed V; heat generation per unit time varies locally with z as e−μz, where μ is the optical absorption coefficient; time dependence of laser beam intensity is as 1 + m cos ωt where modulation index m is less than 1. The formal result requires a three-fold integration, which is approximated using a local coordinate system with origin at farfield observation point, one axis pointing normal to the Mach cone. Approximate integration proceeds with the assumption that time interval following first arrival is short compared with r/(V2 − c2)1/2, where r is cylindrical distance. Given that a ≪ 1/μ, a ≪ c/ω, and r ≫ 1/μ, the transient pulse initially has a period of the order of 2a/c, and subsequently evolves into a super-position of two signals of gradually decreasing frequency. The two frequencies are explained in terms of Doppler shift theory and ray travel times. Amplitude of the first peak decreases as 1/r1/2 and undulates with distance parallel to the beam track, the periodicity being 2πV/ω.
Parametric study of noise and vibration signals
Authors: Nicolas Jean , Lemire Gilles
Published: October 2005

Acoustic and vibration signals have been studied in various ways and conditions. Assuming that the information provided by a vibration accelerometer and a pressure microphone have some correspondence, we have investigated how far one can extend or not this relationship. The results of our experiments have shown that one must clearly separate the amplitude related parameter (power spectrum) from the functions yielding both amplitude and phase contents (coherence, acoustic intensity, etc.). The most significant parameter happens to be the acoustic velocity, obtained in the nearfield and compared to the structural velocity. The acoustic velocity has been obtained by an approximation of the pressure gradient given by a two-microphone probe. The procedure of calibration of the velocity has been found far more critical than the one for the acoustic intensity. The acoustic velocity found experimentally for a point source has been compared to that predicted by the exact theory; results show very good agreement both for the near and farfield. The revealing aspect of this study points out the fact that acoustic velocity is a complex function (intensity is only part of it). Therefore coherence function might be evaluated between the vibration and the acoustic velocity, opening the field for more knowledge about the influence of the reactive field and the radiation efficiency effects.
Aerodynamic mechanisms in jet-edge-resonator oscillation
Authors: Elder Samuel A.
Published: October 2005

Undulatory motions of a separated shear layer in a rectangular slot are found to give rise to three forces that are believed responsible for jet-edge and jet-edge-resonator oscillation at low Mach number. The three are (1) a transverse force on the downstream edge which gives rise to the transverse dipole feedback radiation postulated by Powell, (2) a longitudinal force on the downstream wall, which results in longitudinal dipole radiation believed responsible for oscillation in shallow cavities, and (3) a longitudinal force in the body of the fluid which is associated with pipetone drive. By direct integration of the Lighthill tensor over the entire region of fluid motion between the edges, the magnitude of each force is estimated. In every case the results are directly proportional to shear layer width, and cannot therefore be derived from theories using the vortex sheet approximation.
Stylus instruments and reference specimens
Authors: Vorburger T. V. , Teague E. C. , Scire F. E.
Published: October 2005

Stylus techniques play an important role in the research and metrology of engineering surfaces. This review will discuss the strengths and limitations of stylus instruments from several points of view, including horizontal and vertical resolution, range, accuracy, and speed. Some important properties of roughness calibration specimens will be described as well as the design, specifications, and potential uses of a new series of sinusoidal precision roughness specimens. Comparisons will be drawn between stylus methods and certain optical techniques, and the results of experiments to relate stylus measurements to optical scattering measurements will be described.
Frequency-domain investigations of cochlear stability in the presence of negative damping
Authors: Koshigoe Shozo , Tubis Arnold
Published: October 2005

For a linear-time-invariant cochlear response function, the requirements of causality and stability imply the absence of poles of spectral response function g(ω) for Im ω > 0 (assuming e−iωt as the time dependence). The existence of such poles invalidates conventional Hilbert transform (dispersion) relations. The testing of these dispersion relations provides a frequency-domain check on system stability for cochlear models which contain active elements such as negative resistance. The stability of two-dimensional cochlear models with active elements is checked using these dispersion relations. A simple circuit analog of the cochlea which contains active elements is used to illustrate the relationship between the pole locations of g(ω) and the strength and placement of active elements. [This work was supported by NSF.]
Unified computational model of stimulated and spontaneous acoustic emissions in the external auditory meatus
Authors: Koshigoe Shozo , Tubis Arnold
Published: October 2005

We consider two-dimensional cochlear models which may contain active elements and nonlinearities, and which give retrograde waves on the basilar membrane because of assumed discontinuities in cochlear parameters. Using a conventional middle-ear model and applying frequency and time-domain methods, we calculate the properties of stimulated [D. T. Kemp, J. Acoust. Soc. Am. 64, 1386–1391 (1978)] and spontaneous [P. M. Zurek, J. Acoust. Soc. Am. 69, 514–523 (1981)] acoustic emissions, and compare the results with experimental data. We assume that spontaneous emission is associated with the cochlear response to thermal noise. [This work was supported by NSF.]
A recursive incremental model for noninvasive cochlear measurements
Authors: Gomez P. , Rodellar V.
Published: October 2005

In recent years a number of models have been developed to correlate the cochlear mechanics to measurements obtained by several researchers. This paper proposes a model which, rather than proceed in this manner, describes the cochlear mechanics as an energy interchange process between the stapes and the cochlea. The present approach tries to focus this problem through the application of an incremental model successfully used in the description of speech analysis-synthesis problems. The model describes the cochlea as a chain of incremental sections; hence, by dividing the model into simple sections, simple solutions of the general differential equations can be found for each section and thus for the whole. By means of the boundary conditions between two adjacent sections, the energy flow can be related with the transmission parameters of the sections, establishing a close relationship between the wave propagation characteristics in a given section and those in the stapes. This leads to the possibility of the measurement of cochlear parameters by noninvasive methods.