bluuebunny/crossref_metadata_2025 huggingface dataset:
Application of linear prediction coding and additive synthesis in computer music
Authors: Klapper Magdalena
Published: July 2002

The contemporary computer technique gives a composer quite new possibilities for sound creation. The interference into the sound structure and modification of it, through the mediation of computer, allows one to obtain a new aesthetic value. The paper presents involvement of two methods of signal processing: additive synthesis and linear prediction coding, to obtain an amazing musical effect. Both of the methods were used in two stages: analysis and resynthesis. The stage of analysis allows for the description of a sound by a certain group of information about its structure. Modification of these parameters allows the changes to be made to the sound structure. Employing the analysis methods to the sounds they are not dedicated to may also give an effect of far sound structure modification. It may be given as an example analyzing nonharmonic sounds with a heterodyne filter, or lpc analysis of fast changing sounds. In these cases the sound structure modification results from loss of data during analysis. A wide range of possibilities of sound creation, made accessible by the contemporary computer technique, is a difficult challenge for imagination and a unique aesthetic of a composer.
Directivity of the violin radiation
Authors: Otčenášek Zdeněk , Syrový Václav
Published: July 2002

The sound of a violin played by a person in an anechoic room was recorded simultaneously by 16 microphones lying in a circle with the violin in the middle (the angle among the microphone positions was 22.5 degrees). To create the sound radiation pattern for the individual frequencies, the levels of harmonics were specified in the spectrum obtained from the SPL recordings on the microphones in the given geometrical layout. Next, the violin was excited artificially, by the personal presence of the same player, in the same position as being played by a person. The manner of recording remains unchanged. The constant exciting independent of the frequency enabled the comparison of the directivities of the radiation for different frequencies. Consider the results obtained from both ways of violin exciting, a wide range of angles was searched in which the radiation is stable enough even in the sense of listening to tones recorded when the violin was played by a person.
Physico-acoustical properties of emission affected resonance spruce wood
Authors: Rajcan Emil , Urgela Stanislav
Published: July 2002

High-quality resonant spruce wood for violin top plate and wood from traditional Slovak resonant spruce locality with various degrees of emission damage were studied experimentally. The modal analysis was done on quarter-cut spruce plates of wedge shape typical for violin making. Acoustical resonance and holographic interferometry were used to measure the relevant physico-acoustical characteristics. These characteristics (density, Young’s modulus, and acoustical constant) were obtained by using the bending beam vibration mode (2,0). The obtained results showed that the sorting of resonant spruce wood for violin based on empirical experiences has a physico-acoustical reason, and actual industrial emissions have not significant influence on relevant physico-acoustical characteristics of investigated resonant spruce wood.
Modern methods for attenuation in ductwork and silencers
Authors: Mechel F. P.
Published: July 2002

Silencers in power plant ductwork should find a good compromise between the (generally rather high) needed sound transmission loss and the static flow pressure loss induced by them. The annual costs of the electric power consumption created by a silencer may exceed the costs of investment. Modern design possibilities will be displayed. All of them can be computed with the instrument of mode analysis. On the classical background, that attenuation is concentrated in broadband silencers with straight, homogeneous linings, the following improvements will be illustrated with examples. First, because a broadband silencer needs thick linings for low frequencies and small free duct width for high frequencies, both increasing the static pressure loss, subdivide the silencer in sections for different frequency subbands, with stepping lining thicknesses and duct widths. It will be indicated how such steps can be computed Second, insert the necks of Helmholtz resonators between lining sections designed for medium and high frequencies. The resonator volumes are behind the lining sections. The increase in silencer length is very small; the increase in thickness is small, because the main extension of the resonator volumes is in the duct direction. Such resonators may be designed for an efficient low-frequency additional attenuation. Third, duct elbows with their static pressure losses exist anyhow in most ductworks. They can be designed as very efficient high-frequency silencers, thus permitting to avoid the narrow lining spacings in the silencer. Criteria for good corner silencers will be illustrated. Fourth, most silencers in a ductwork begin and end with a conical duct transition. If they are rigid, their combined effects are small. If, however, they are suitably lined, they can produce remarkable additional attenuation.
How to accurately distinguish between industrial and traffic noise—New statistical methods for the acquisition and evaluation of data
Authors: Martinez Sergio C.
Published: July 2002

Statistical methods are used to analyze long-time data series and to distinguish between industrial and traffic noise. Different approaches must be observed for the interpretation of measured data and for the relative influence of noise sources: linear or energetical superposition, and addition or product of mathematical probabilities. Consistent results are available with these methods. Numerical simulations furthermore reduce possible errors. Data series at additional emmission points are used to obtain re- dundant parameters in the given environmental conditions. Numerical calculations are presented on the problem of linear or energetical superposition and the application of them on two real-life-scenarios. The progress in the application of these methods [S. C. Martinez, Fortschritte der Akustik DAGA 95, 77–85 (1995)] will be demonstrated. Special emphasis will be put on some possible errors in the application of the guideline VDI 3723 (application of statistical methods for the description of variating ambient noise levels).
Reverberation in waveguides with rough surfaces
Authors: Berman David H.
Published: July 2002

An impedance formalism for treating scattering at multiple rough boundaries in waveguides with depth-varying sound speeds is developed. The formalism is used to give a compact result for the time history of multiply scattered reverberant sound. In general, the multiple scattering contribution to reverbation decays more slowly than the single scattering contribution. When attenuation mechanisms other than surface scattering are small, the reverberant field has a diffusive character, and this leads to algebraic (∼1/t) decay of the reverberation of originally pulsed signals. As a by-product of the formalism, conventional reverberation models are derived from “first principles.” The derivation provides the precise form of the scattering strength used in such models. In addition, it shows that the mean field should be used for propagation to and from the scattering surface. Finally, the computations present the possibility of treating the scattering by two rough boundaries simultaneously.
Backscattering enhancements due to retroreflection of ultrasonic leaky Rayleigh waves at corners of solid elastic cubes in water
Authors: Gipson Karen , Marston Philip L.
Published: July 2002

Backscattering enhancements can be important for interpreting acoustical images of scatterers. Experiments were performed on a solid stainless steel cube in water in order to investigate a significant mechanism for producing a large backscattered signal. Enhanced high-frequency backscattering is observed when the block is rotated such that the incident acoustic wave strikes one of its flat surfaces in the vicinity of the coupling angle for launching a leaky Rayleigh wave at the steel–water interface. This enhancement can be explained as a retroreflection of the leaky wave at a corner of the block, which results in a reversal of the incident wave vector and thus produces a large backscattered signal. The magnitude of the peak backscattered pressure is approximated using a convolution formulation [P. L. Marston, J. Acoust. Soc. Am. 97, 34–41 (1995)]. Observed backscattering enhancements support the validity of the convolution formulation to approximate the general magnitude of the peak pressure, as well as being consistent with the dephasing predicted by that theory as the block is rotated away from the coupling angle. Tilting the block while maintaining the coupling angle reduces the backscattered amplitude because of changes in the reflection coefficients at the corner as well as increased attenuation.
Insights into linear and nonlinear cochlear transduction: Application of a new system-identification procedure on transient-evoked otoacoustic emissions data
Authors: Krishnan Geetha , Chertoff Mark E.
Published: July 2002

Transient-evoked otoacoustic emissions (TEOAE) were used to characterize linear and nonlinear cochlear transduction using a new system-identification procedure. In this technique, a computational model of the system is first developed. From the measured stimulus and response records, spectral-density functions and multiple coherence functions are calculated. The coherence functions allow the characterization of linear/nonlinear processes as a function of frequency. Summations of linear and nonlinear coherences provide a goodness-of-fit of the chosen model. Finite impulse response pulses with a bandwidth of 1–8 kHz were used to evoke otoacoustic emissions. Eleven adults with normal hearing served as subjects. Third- and fifth-order polynomial models were used to model the data, and the results indicate that the fifth-order model is a better fit to the TEOAE data. The results of this study suggest that this system-identification procedure can be successfully applied to model cochlear transduction using a broadband stimulus. Most importantly, coherence functions provide useful insights into linear and nonlinear cochlear processes and have the potential to be developed as a clinical measure for monitoring changes in cochlear status.
Acoustic design of the Tsuyama Music Cultural Hall based on the preference theory
Authors: Ando Y. , Suzumura Y. , Yamamoto I.
Published: July 2002

The city of Tsuyama is located about 100 kilometers west of Kobe. The fundamental shape of the plan and cross sections of the hall were designed by applying the theory of subjective preference [Ando, Architectural Acoustics (AIP Press/Springer-Verlag, New York, 1998)]. Special attempts made in the acoustic design of this hall are: (1) a number of columns distributed in front of the walls at the audience level and on the stage: This may keep a small value of the IACC and a certain initial time delay gap between the direct sound and the first reflection, making an increase of the subjective preference for both listeners and musicians; (2) a stage enclosure with the canopy of several triangular reflectors at adjustable height for musicians, to control the preferred delay time of the reflections in performing a certain type of music; and (3) the shape of the rear wall of the stage, to control the IACC for listeners.
Analysis of the role of inhibition in shaping responses to sinusoidally amplitude-modulated signals in the auditory system
Authors: Pollak George D.
Published: July 2002

Neurons in the inferior colliculus (IC) typically respond with phase-locked discharges to low rates of sinusoidal amplitude modulated (SAM) signals and fail to phase-lock to higher SAM rates. The hypothesis that these properties are shaped by the integration of phase-locked excitation and inhibition, as they are in lower nuclei, was tested. Responses were recorded from IC neurons evoked by SAM signals before and during the iontophoretic application of bicuculline, a competitive antagonist for GABAA receptors, strychnine, a competitive antagonist for glycine receptors, and the GABAB receptor blocker, phaclofen. The hypothesis that inhibition shapes responses to SAM signals in the IC was not confirmed. In more than 90% of the ICc neurons tested, the range of SAM rates to which they phase-locked was unchanged after blocking inhibition with bicuculline, strychnine, or phaclofen, applied either individually or in combination. These results illustrate that the same response property, phase-locking restricted to low SAM rates, is formed in more than one way in the auditory brainstem. In lower nuclei, the mechanism is coincidence of phase-locked excitation and inhibition, whereas in IC the same response feature is formed by a different but unknown mechanism. [Work supported by NICD.]
Sound focusing in reverberating rooms: The time-reversal approach
Authors: Yon Sylvain , Dorme Christian , Fink Mathias
Published: July 2002

Time reversal is already known as an efficient ultrasonic method to focus through inhomogeneous or multidiffusive media. The presence of pointlike reflectors in these media allows applications of this technique in fields like nondestructive testing, medical imaging, or underwater acoustics. An extension of this technique in audible range acoustics is presented. An array of 70 microphone/loudspeaker couples is used to refocus sound inside of a reverberating room. At a desired focal point, directivity patterns are measured and compared with those obtained by focusing with a cylindrical beamforming technique through the same antenna. Time reversal is shown to strongly improve the focal spot pattern (beamwidth and sidelobes level). These results are related to the ability of time reversal to compensate for reverberation and scattering induced in the studied room. The process acts as a spatio-temporally matched filter to the propagation transfer function of the desired focal point through the room. It makes this technique an auto-adaptive focusing system to a random 3-D cavity. Theory, experimental results, and future applications will be described.
Determination of structure of the NARMAX model for dynamic loudspeaker using the orthogonal algorithm
Authors: Pruchnicki Piotr , Dobrucki Andrzej
Published: July 2002

NARMAX is a general method for modeling of nonlinear systems. In the paper, an application of this method for modeling of nonlinearities in dynamic loudspeakers is presented. In most cases, a polynomial representation of the NARMAX model causes an ill-conditioned equation system which indicates that the use of orthogonalization is necessary to solve the system. The high number of coefficients is the main disadvantage of the NARMAX method, and it results in an erroneous model, and an algorithm for identification of coefficients that is noneffective. A modification of the orthogonal algorithm (classical Gram–Schmidt) is presented. It is based on the premise that the coefficient providing the highest error reduction ratio is chosen in every step of the algorithm. The creation of the model stops when the given accuracy is achieved. This way leads to the choice of only the most significant coefficients. The work and effectiveness of this modified algorithm for systems with given nonlinearities at different levels of accuracy of the model are presented in the paper.
Acoustics: A route to science literacy in the 21st Century
Authors: Rossing Thomas D.
Published: July 2002

Two trends have been apparent during the second half of the 20th Century: the disappearance of acoustics from physics curricula and the decline in the percentage of students who take physics, both in schools and universities. The relationship, if any, between these trends is well worth discussing. More intriguing, however, is the likelihood that acoustics, with its rather wide appeal to students and to the public, can serve as a vehicle to promote science literacy.
A complete period-doubling cascade on a cromorn
Authors: Gibiat Vincent , Castellengo Michele
Published: July 2002

The cromorn is a very simple double reed instrument of the Renaissance which shares with the recorder the valuable property that there is no contact between the musician’s lips and the double reed, allowing one to study it without any human intervention and so to control the parameters of the sound production. This has been studied by varying the blowing pressure from small values to the pressure needed to obtain the normal sound of the instrument and then by decreasing it. The sound has been recorded and analyzed in time frequency domain. A complete cascade of period-doubling bifurcations has been obtained following the increase of the pressure. Two chaotic regions have been found, one just after the third period doubling and another after a period tripling. For the lowest pressure in the decreasing process the coupling of the reed and the bore becomes too weak and the reed squeaks. The instrument is simple enough to follow the schematic given by Maganza et al. for cylindrical instruments and it will be shown that the decrease of the pressure is equivalent to the sharpening of the nonlinear function which relates pressure and flow that gives the possibility of period doublings.
Direct measurement of dynamic elastic constants of a disc
Authors: Nieves Francisco J. , Gascón Francisco , Bayón Ana
Published: July 2002

In two preceding papers [F. J. Nieves, F. Gascón, and A. Bayón, J. Acoust. Soc. Am. 104, 176–180 (1998); J. Acoust. Soc. Am. (to be published)], the possibility of measuring the two dynamic elastic constants by excitation of axisymmetric vibration of an isotropic cylinder, whose length L equals its diameter D, and by means of the detection of the two lowest natural frequencies, one symmetric and another antisymmetric, has been demonstrated. The variational method and the Ritz procedure with polynomies are followed to calculate numerically the two lowest nondimensional frequencies for a disc with L/D=0.1 and the result is antisymmetric. The quotient between both nondimensional frequencies is a function only of the Poisson ratio and it is single-valuated. Then the measurement of these frequencies allows the calculation of that elastic constant. The quotient dependence on the Poisson ratio is strong and almost linear, which implies a much better sensibility than it was for L/D=1. Moreover, one of the calculated frequencies, the disc diameter, and the material density permit the calculation of the shear modulus. The theory is checked with a laboratory experiment on a stainless steel disc of 5 cm in diameter.
Inline process monitoring of thermosets by ultrasonic measurements in a compression mould
Authors: Rath M. , Hinrichsen G. , Döring J. , Stark W.
Published: July 2002

Moulding compounds are heterogeneous materials and consist of granules or pellets, usually mixed with a special resin-hardener combination, reinforcing agents, accelerators, lubricants, pigments, and other additives ready for manufacturing. Thermosetting moulding compounds are defined as raw materials manufacturing plastic parts under the influence of pressure and heat, mainly by compression, transfer, and injection moulding. Current research addresses ultrasonic quality sensor utilization for inline cure monitoring, to control and analyze compression moulding. A ‘‘smart’’ process control would require the measurement of elastic moduli and viscosity during cure. In this research, an ultrasonic impulse-transmission method is used to directly monitor changes in the mechanical properties of moulding compounds throughout the cure cycle. Measurements were performed during the processing using transducers which were designed to operate at elevated temperatures and pressures in a compression mould. Measurements of ultrasound velocity and change of attenuation coefficient of longitudinal and shear waves were made throughout the process cycle. Use was made of the ultrasonic measurements to monitor the influence of various parameters on the manufacturing process, such as filler concentration, humidity content, artificial weathering at elevated temperatures, and hardener concentration. [Work supported by Deutsche Forschungsgemeinschaft (DFG).]
Dichotic perception of AM and FM signals
Authors: Ozimek E. , Konieczny J. , Wicher A.
Published: July 2002

The study deals with the binaurally perceived modulation depth (mp) of amplitude-modulated signals (Exp. I) and binaurally perceived frequency deviation (Δfp) of frequency-modulated signals (Exp. II), for their diotic and dichotic perception. Various combinations of modulation depth (m) and frequency deviation (Δf) at the left and right ear were used. It was found that when the interaural difference in modulation depth (Δm=ml−mr) or interaural difference in frequency deviation (ΔΔf=Δfl−Δfr) were equal to zero (i.e., diotic perception), the binaurally perceived modulation depth or binaurally perceived frequency deviation were the same as those in monaural perception. For sufficiently small Δm or ΔΔf, the perceived modulation depth, or frequency deviation, approximated the mean of the magnitudes presented to the left and the right ear. However, for large interaural differences in Δm or ΔΔf, the binaurally perceived quantities were lower than their arithmetic means, and showed nonlinear trends. Comparison of the perceived modulation depths determined in the experiment and those calculated under the assumption of the linear summation of AM signals from the left and the right ear are also discussed.
Virtual auditory localization in noise
Authors: Ericson Mark A. , Nelson W. Todd , Bolia Robert S.
Published: July 2002

Interfering sounds have different interaural cross correlations (IACCs) depending on the location and environment in which they are produced. IACCs, in addition to signal bandwidth and signal-to-noise ratio, influence localization acuity. Most localization experiments use stimuli presented in the free-field without maskers or with one directional masker [Good and Gilkey, J. Acoust. Soc. Am. 99, 1108–1117 (1996)]. Hirsh [J. Acoust. Soc. Am. 22, 196–200 (1950)] and Mershon and Lin [Ergonomics 30, 1161–1173 (1987)] have measured auditory localization in reverberant environments. However, in these studies the signals and masker(s) were presented in the reverberant environment. No studies have addressed the effects of diffuse noise on auditory localization over headphones. The current study examines the effect of band passing a directional noise signal presented over headphones masked by noises of different interaural cross correlations. The signal-to-noise ratio is varied from +10 to −10 dB, and the signal is low-pass filtered at 1.6, 4, 8, and 15 kHz. Findings from this study have implications for models of binaural hearing and the design of directional audio displays.
Investigating the precedence effect with a virtual auditory environment
Authors: Djelani Thomas
Published: July 2002

Due to the precedence effect humans are capable of localizing sound in reverberant environments. Despite the importance of this effect its mechanism is not completely understood yet. Especially the question of how the precedence effect is controlled by higher levels of the auditory system is still open. A new approach for further investigations of the precedence effect is the use of a real-time virtual auditory-environment system (SCAT-Lab), which allows one to create and auralize virtual rooms. The subjects listen to the presented scenarios via headphones using individual head-related transfer functions (HRTFs). The SCAT-Lab uses a head tracker to measure the listeners head position and orientation and modifies the HRTFs accordingly. The system offers a high degree of flexibility in creating realistic and complex scenarios for psychoacoustic experiments. For example, it can be used to set up the Clifton effect experiment [J. Acoust. Soc. Am. 82, 1834 (1987)] just by moving one wall in a virtual room. The evaluation of the SCAT-Lab system for precedence effect experiments has been performed to test the possibilities and limits given by this new tool for further psychoacoustic experiments.
Changes in perceived velocity following adaptation to simulated auditory motion
Authors: Uematsu Hisashi , Kashino Makio
Published: July 2002

To investigate whether adaptation to auditory motion can cause changes in perceived velocity, measurements of perceived velocity were conducted for apparently moving stimuli produced by varying only interaural time differences (ITDs). Following adaptation to a 10-s 500-Hz tone, a 500-Hz test tone was presented, followed by a 250-Hz standard tone. The velocity (ITD change rate) of the adapter tone was chosen from the range of 300 to 2400 μs/s, and that of the standard tone was fixed at 300 μs/s. Subjects judged which of the test and standard tones was perceived as moving faster. The constant method was used to obtain the velocity of the test tone judged to be equal to that of the standard tone. It was found that the perceived velocity of the test tone following adaptation was slower than the physical value by up to 70 μs/s. Adaptation did not affect the perceived direction of the test tone in the present velocity range of the adapting and test tones. These results indicate that adaptation to auditory motion cannot only produce countermotion of a physically stationary sound, as has been reported, but also cause changes in perceived velocity of a moving sound. Theoretical implications will be discussed.
Classification of the acoustic environments using a psychoacoustic model
Authors: Ostendorf Melanie , Hohmann Volker
Published: July 2002

Complex digital hearing aid algorithms employ different signal processing stages, e.g., noise reduction and loudness control. Most of these stages are suitable only for special acoustic environments or need different parameter sets for different environments to assure optimum rehabilitation success. Future ‘‘intelligent’’ digital hearing aids should therefore incorporate a model of the acoustic environment to adapt the signal processing to the acoustic environment. In particular, the presence of a speech signal and of different types of ambient noise should be detectable. In this study, the perception model introduced by Dau et al. [J. Acoust. Soc. Am. 102, 2829–2919 (1996)], which was originally developed to model different psychoacoustical tasks, e.g., forward masking and modulation detection is used as a preprocessor for this type of classification algorithms. This approach seems promising, as the model includes a modulation frequency analysis which was shown to be a good estimator of speech activity [Ostendorf, Fortschritte der Akustik DAGA 98, Zürich]. First results on the classification accuracy using this model are presented.
On the relationship between auditory evoked potentials and psychophysical loudness
Authors: Wegner Oliver , Dau Torsten , Kollmeier Birger
Published: July 2002

This study examines the question of whether the neurophysiological events reflected by the auditory evoked potentials are similar to the events underlying loudness perception. Recent studies showed that, at brainstem level, loudness does not seem to be directly related to the potential amplitude (wave V). This is the case for the loudness growth function for tone pulses [J. K. Nousak and D. R. Stapells, ‘‘Loudness and the ABR/MBR in noise-masked normal-hearing subjects,’’ ARO meeting, p. 39 (1998)] as well as for loudness summation obtained with optimized chirp stimuli [Wegner et al., ‘‘Untersuchungen zu Lautheitskorrelaten in akustisch evozierten Potentialen,’’ DAGA 98, Zürich (in press)]. In both conditions, the results obtained with early evoked potentials suggest a higher compression as that normally derived from psychophysical data and as reflected in common loudness models. This study presents brainstem and cortical potentials evoked by bandlimited stimuli of equal loudness. The role of the synchronization, stimulation rate, and temporal integration at both processing levels is discussed.
Localized modes of long floating structures
Authors: Azalinov Dmitri
Published: July 2002

The paper is devoted to the problem of fluid–structure interaction. Nowadays the problems of construction and maintenance of buoyant air stations arouse large interest. As a rule, a buoyant air station is a long plate of defined width and it locates near a shore at a small depth. It means that for the construction of a mathematical model one can use the equations of the thin plate oscillations on shallow water. It is shown in what frequency ranges the propagation of bending-gravitation waves is possible. Apart from cutoff frequencies a real spectrum of eigenfrequencies and conditions of its existence have been determined. The presence of such frequences makes possible phenomena of wave localization in area of the plate and its resonance. The localization of a standing wave type occurs in one of the directions, at the same time the wave propagation occurs in the perpendicular direction. The eigenmode is symmetrical and has maximum in the middle of the plate. The parameters of a plate–liquid system have been determined, when these effects can be possible. These problems have been solved with a combination of analytical and computer approaches.
Active cancellation of rotor/stator interaction tones in turbofan engines
Authors: Walker Bruce E. , Hersh Alan S. , Heidelberg Lawrence J. , Spencer Michael E.
Published: July 2002

Rotor/stator-generated aerodynamic noise in turbofan engines, HVAC axial fans, and other turbomachinery is often dominated by blade passage harmonics produced by dipole sound sources. Radiation toward the inlet and exhaust depends upon the effective orientation of the dipole sources relative to the propagation angles of the excited modes. Absorption of this tonal noise using passive treatment is often compromised because of a lack of knowledge of the relative phases and amplitudes of the propagating modes. A modal-based active cancellation system has been developed for global suppression of rotor/stator interaction tones. Axially spaced annuli of controlled wall-mounted actuators are located immediately upstream and downstream of the hubs and tips of stator vanes. Wall-mounted error microphone arrays are located in the inlet nacelle and exhaust duct. Actuator drive vectors are determined in an adaptive-quadrature control algorithm that seeks to minimize total radiated noise. The concept was initially developed and tested on a 24-in. axial fan facility and then extended and demonstrated on the 48-in. Active Noise Control Fan facility at NASA Lewis Research Center. Simultaneous suppression of modes (4,0) and (4,1) of 8–17 dB toward the inlet and 3–11 dB toward the exhaust was achieved at the second blade-passage harmonic.
Adjustments in kinematic parameters of bilabial articulatory gestures during repeated oral readings of the same passage
Authors: Max Ludo
Published: July 2002

Repetitive practice of nonspeech motor acts typically results in behavioral, kinematic, and kinetic changes that reflect underlying neuromotor adjustments (motor learning). The effects of repetitive performance on the organization of speech movements, however, are less well understood. Yet, knowledge of such effects could (1) contribute to important theoretical notions in speech motor control, (2) reveal methodological concerns with currently used research paradigms, and (3) provide additional theoretical and clinical insights into speech disorders that are positively affected by repetitive practice. The present study therefore investigated the effects of repeated oral readings of a single passage on kinematic parameters of bilabial articulatory gestures (only closing gestures are discussed here). Seven adults, all normal speakers, performed five readings of a 200-word passage containing five instances of each of three target words. Movements of the lips and jaw were transduced with a strain gauge system whereas articulation rate was determined acoustically. All subjects increased articulation rate from reading 1 to reading 5. Kinematic adjustments were more consistently found (across subjects and target words) for a derived signal representing lip aperture than for movements of the individual articulators. Preliminary analyses suggest that some adjustments may be mediated by phonetic context and/or syllable stress.
Fully immersive audio environments—musical, aesthetical, and computational considerations
Authors: Malham David G.
Published: July 2002

Before the development of the telephone and the phonograph, all audio experiences were essentially three-dimensional. Composers and musicians had to accept the limitations of human performers and the acoustics in which they played. The development in the nineteenth century of the technological means necessary to provide performances of music which were remote in either time or space from the original was accommpanied by an almost complete removal of this three-dimensional aspect. Engineers and musicians have been searching ever since for ways of restoring the balance. Progress in this endeavor has been rapid in the last two decades, so much so that composers, musicians, and sonic artists now have a wide range of techniques at their disposal for manipulating the spatial elements in their compositions. This paper looks at various aspects of the technologies involved and the implications their availability has for composers and sonic artists working in the medium. This is presented within the context of a historical perspective. Special consideration is given to technologies which are intended to allow presentation to audiences, rather than, for instance, the more typical virtual reality systems which is concerned with smaller numbers of individuals.
Multichannel sub-band inverse filtering for room dereverberation
Authors: Wang Hong
Published: July 2002

Room dereverberation is difficult because the impulse responses are long, nonminimum phase and change significantly with time and source/receiver positions. The problem was approached in two steps: (1) Explore the possibility of inverse filtering when a training signal is available; (2) blindly estimate impulse responses from reverberant signals and then perform inverse filtering. It is found that by dividing the signals into a large number of sub-bands, the impulse responses of many of the sub-bands become invertible. The invertibility of each sub-band depends on the zero distribution of the full band impulse response. When multiple microphones are used, full band dereverberation is achieved by selecting the best microphone in each sub-band and reconstructing the full band signal from dereverberated sub-band signals. An almost perfectly recovered reverberant speech of RT=0.85 s, when the training signal is used, will be demonstrated. For the second step, reverberant signals from multiple sensors are divided into sub-bands. In each sub-band, the ratio of impulse responses from two sensors is estimated by the least-mean-square estimation. Pade approximation is used to estimate each individual impulse response. The sub-band signals are then inverse-filtered and combined to form a full band dereverberated signal.
Practical aspects of analyzing cyclostationary noise emitted by real acoustic sources
Authors: Vokurka Karel
Published: July 2002

In previous presentations [K. Vokurka, J. Acoust. Soc. Am. 101, 3045(A) (1997); 103, 2819(A) (1998)] it was shown that time-frequency statistical characteristics of noise and vibration measured on reciprocating machinery yield information which can be conveniently used, e.g., in quality control or for diagnostic purposes. A basic assumption in developing suitable signal processing algorithms was that the number of samples in each period of cyclostationarity of the analyzed noise or vibration was exactly the same. However, this assumption is only seldom fulfilled by real acoustic noise or vibration. Hence, a suitable preprocessing technique must be applied to the measured signal. This technique may be either signal resampling, or external control of a sampling interval using multiple tachopulses, frequency multipliers, etc. In the paper, advantages and disadvantages of these techniques are discussed from the point of view of different types of measured acoustic noise or vibration signals, their possible phase modulations, and of the computed time-frequency statistical characteristics. [Work supported by the Grant Agency of the Czech Republic and by the European Commission.]
Algorithms for modeling nonlinear geometric distortion of vinyl records
Authors: Henke Robert , Wilk Eva
Published: July 2002

Classical vinyl records have a characteristic sound that is distinct from an ideal digital system and is often described as warm, mellow, or ‘‘analog.’’ Responsible for this sound is the sum of linear and nonlinear distortions which occur during recording and playback of the record. The distortions are due to mechanical and electrical nonlinearities. One main source of nonlinear distortion lies in the discrepancy between the cutter’s shape and that of the playback stylus: The cutter is a sharp tool while the playback stylus has a ball-like shape to distribute the tracking force over a wider area. This leads to a slightly different movement of the playback stylus in comparison to the recorded groove, yielding so-called ‘‘geometric distortions.’’ The mechanical, i.e., geometrical nature of this phenomenon, makes it possible to find accurate mathematical descriptions. This approach was used to develop a model for the real-time simulation of the typical sound of a vinyl record. The developed tool allows the simulation of the ‘‘vinyl sound’’ and can basically be used to create—and control—a wide range of nonlinear distortion which can vary from a slight enhancement of a given signal to new effects of sound design.
Ultrasonic reflection tomography: Signal processing and resolution
Authors: Lasaygues P. , Lefebvre J. P. , Mensah S.
Published: July 2002

Medical imaging is, with nondestructive testing of materials, the main application of ultrasonic reflection tomography (URT). This method results from an inverse born approximation (IBA). URT allows perturbations (theoretically small) of a reference medium to be visualized. One is then concerned with a Fourier synthesis problem and the first step is to correctly cover the frequency space of the object by an adequate spatial and frequency scanning. In practice, the frequency bandwidth of the used ultrasonic pulses is limited by the transducers, leading to a poor coverage of the object’s spectrum and to a relatively poor resolution of images. Low and high frequencies are restored by a Papoulis deconvolution method that presents advantage not to modify the spectral content of the signal within its frequency band. The resolution enhancement procedure was first tested on academic targets (wire), and on test targets (square aluminum and triangular PVC rods), showing the possibility to obtain high-resolution images with low-frequency transducers. Finally, we compare the resolution given by a 250-kHz transducer to the one obtained, with a classical technique, by a 2,5 MHz transducer. The overall dimension of test objects is in that case less than the wavelength.
Sensitivity analysis of the acoustic field in shallow water to environmental parameters
Authors: Guo Lianghao , Gong Zaixiao
Published: July 2002

Environmental parameters, such as sound-speed profile and bottom structure, have very important effects on the acoustic propagation in shallow water. In this paper, some detailed simulations are presented for a typical shallow water with a thermocline. The simulated properties include the coherent and incoherent transmission loss, averaged energy loss, group velocity, phase of the acoustic field, and the multipath arrival structure. The concerned environmental parameters include surface speed, thermocline depth and thickness, water depth, bottom structure, and its geoacoustic parameters. The results show that the coherent transmission loss and the phase of the acoustic field are sensitive to the environmental parameters, while the other properties are relatively stable. Theoretical explanations of these results are also given. Both the sensibility and the relative stability can be utilized. Through reasonable cost function design, the former can be used to invert the geoacoustic parameters of the bottom, while the latter can be used to localize distant targets. Finally, the geoacoustic parameter inversion for a complicated bottom is given by using the real data from a vertical line array. With the obtained bottom parameters, experimental transmission loss is well explained.
Characterization of the underwater channel for acoustic communications
Authors: Loubet Gerard , Faure Bernard
Published: July 2002

In recent years, acoustic underwater (AUW) communications have known a huge increase in research and development due to the growing needs of wireless transmission systems. While mobile radio channels fit well within the model of Rayleigh fading, this model is often used for the design of communication systems. However, many experimentations carried out by our laboratory (LIS ex CEPHAG) with the French Navy or within European Community MAST projects (LORACOM and PARACOM) have shown that this model do not fit for AUW transmissions. The AUW channel is well modeled, using ray tracing, by great deterministic propagation paths (macro-multipath) associated with random fluctuations (micro-multipaths). Experimental results will be presented. The slow variations of the deterministic part may be due to internal waves, the little, fast variations of the received energy by appearance or disappearance of micro-paths. A random channel modeling developed for LORACOM project (Contract No. MAS2-CTG4-0080) agrees with this behavior. Different AUW medium characterizations depending on the channel geometry (shallow or deep water communications), on the frequencies (between 1 and 10 kHz), and on the range (1 to 100 km) will be discussed. Spatial coherence results will also be presented.
Identification of the radiating sources in underwater acoustics
Authors: Salah Bourennane , Ammar Bendjama
Published: July 2002

In the context of the narrow-band or wideband array processing problem, in this paper a robust algorithm is developed to improve the accuracy of the estimation of the direction of arrival of the sources. It is well known that when the noise cross-spectral matrix is unknown, these estimates may be grossly inaccurate. The band noise spectral matrix, the classical noise model, is used. By means of a linear operator and an iterative algorithm for estimating the spatial correlation length, a new estimator for the direction of arrival of the sources is developed. The proposed algorithm is based upon a particular partition of the received signal vector which leads one to obtain an approximation of the noise subspace matrix without eigendecomposition. Then using both a propagation operator and noneigenvector algorithm to estimate the projection matrix, a new robust algorithm is developed for the source characterization problem in the presence of noise with an unknown cross-spectral matrix. It will be shown that the performance of bearing estimation algorithms improves substantially when this robust algorithm is used. Experimental data results are presented for the band noise spectral matrix.
A system for measuring the directional parameters of room acoustics
Authors: Okubo Hiroyuki , Otani Masamichi , Ikezawa Ryo , Komiyama Setsu
Published: July 2002

A new system for measuring the directional parameters of room acoustics has been developed. The system consists of five omni-directional microphones, audio interfaces, DSPs (digital signal processors), and control software. This software synthesizes the directional pattern of the microphones, and calculates the directional parameters automatically. The lateral component (LC), front- to-back ratio (FBR), and left-to-right ratio (LRR) of directional room impulse responses are easy to measure with this system. The system was used to take measurements in two halls. The first hall was a concert hall, with the measurements of impulse responses taken at 72 points. The second was a multipurpose hall, the shape of which could be altered three ways (for concerts, conventions, and exhibitions), and the measurements were taken at 12 points for each shape. The total of 108 data points from the four room shapes represents the range of room types. The directional parameters are sensitive to room shape, wall materials, and the positions of seats. These data may also be useful for studies on the correspondence between the objective description and subjective evaluation of the acoustical quality of rooms.
On the accuracy of the assessment of room acoustical parameters using MLS technique and ray-tracing simulation
Authors: de Avelar Gomes Marcio Henrique , Yousri Gerges Samir Nagi
Published: July 2002

Advances in psychoacoustics through the years have made it possible to evaluate the acoustical quality of a room, based on several numerical parameters that have been developed. Even though there is not total agreement about which parameters are truly important, some of them are accepted by most of the acousticians. Most of these parameters can be determined from the impulse responses of a room. This work presents a comparison between results simulated using a commercial ray-tracing computer program, and measured with one of the most modern techniques of measuring impulse responses, which uses a maximum length sequence (MLS) as the driving signal. Limitations concerning the use of the measuring technique, and the ray-tracing program, as well, are analyzed and discussed.
Ultrasonic-induced phase transitions of micrometer-size droplets
Authors: Kripfgans Oliver D. , Fowlkes J. Brian , Carson Paul L.
Published: July 2002

Droplets for various applications were generated by mixing albumin (bovine) with saline and a low boiling point liquid (dodecaflouropentane). The resulting emulsion contains small droplets with a lower size range on the order of a few micrometers. The type of mixing process was found to determine the range of droplet sizes. The albumin, as a surface active agent, is assumed to build a shell around the droplets and prevents coalescence, since mixtures without albumin did not produce stable droplets. Increases in temperature of the resting host fluid showed that droplets could be superheated well above their natural boiling point. It is assumed that this property is caused by an increase in pressure within the droplet due to surface tension. The superheated state could be overcome by the application of ultrasonic pulses and/or hydrodynamical action (flow). The presence of ultrasound contrast agents allowed ultrasonic-induced phase transitions at the natural boiling point. Droplets producing bubbles for transpulmonary applications have been studied previously. The interest here was primarily in larger bubbles for therapeutic applications and to serve as point beacons for aberration correction. [Work supported in part by PHS Grant No. R01 HL54201 from the National Heart, Lung and Blood Institute.]