bluuebunny/crossref_metadata_2025 huggingface dataset:
Three-dimensional extraction and reproduction of signals from an interfering sound field
Authors: Fukudome Kimitoshi
Published: October 2005

This paper describes a method for the three-dimensional reproduction of any constituent sound signals existing in the interfering sound field. The method is composed of three parts. (1) Estimation of incident directions and spectra of unknown sound sources using sphere-baffled microphones and the diffractive information of the sphere [K. Fukudome, J. Acoust. Soc. Jpn. 44, 272–281 (1988)]. (2) Extraction of sound source signals by adding the output signals of the extracting filters that are fed by the sphere-baffled microphones. Characteristics of the filters are determined according to the estimated results of the incident directions and the diffractive information. (3) Three-dimensional reproduction of any constituent sound signals using headphones. The right and left signals of the headphones are generated after convolving the head-related transfer characteristics with the constituent sound source signal. The validity of the present method has been verified by computer simulation. Finally, the problems to be considered at the implementation are described. [Work supported partly by The Sound Technology Promotion Foundation.]
The application of digital filters to the de-Dopplerization of acoustic signals
Authors: Glegg Stewart A L. , Mouches Jean Marc
Published: October 2005

When acoustic measurements are made of a moving vehicle by a stationary observer, the Doppler frequency shift has two detrimental effects on the interpretation of the measured spectra. First, the spectra are smeared by the change in Doppler factor during the vehicle pass-by, and, second, the motion of the source introduces a phase shift in the signals that causes errors in source location measurements. The measured signals can be corrected back to source time if a moving time-delay correction is applied, and this overcomes both the problems mentioned above. However, when the signals are sampled digitally at equal intervals in observer time, this time-delay correction requires an estimate to be made of the signal level between samples. This can be achieved by using a digital filter with time varying coefficients, which estimates the signal from at least two adjacent samples. Results will be given that show how this technique has been applied to source location measurements on moving highway vehicles.
Control of inharmonicity by the addition of a point mass
Authors: Suzuki Hideo
Published: October 2005

A method for controlling the inharmonicity of a stiff string by adding a lumped mass (or masses) was originally suggested by F. Miller, Jr. [F. Miller, Jr., J. Acoust. Soc. Am. 21, 318–322 (1949)]. In the present paper, the same idea is reviewed in a more precise manner. The effect of the stiffness of the string on its inharmonicity is discussed for different support conditions (simple-simple, simple-fixed, and fixed-fixed conditions). The simple-simple support condition gives the largest inharmonicity (deviation from a harmonic series), even though the differences are small in most cases. The theoretical and experimental results for the resonance frequencies of a string in the fixed-fixed condition show very good agreement with each other. For a middle C string, a mass of 0.2–0.4 g, when attached 13–9 mm from one end of the string, reduces the inharmonicity significantly.
Measuring piano hammers and modeling their interaction
Authors: Hall Donald E. , Askenfelt Anders
Published: October 2005

Comparison of measured piano string vibration spectra with predictions indicates that modeling the hammer with a simple mass and Hooke's law spring is not very satisfactory [Hall and Askenfelt, J. Acoust. Soc. Am. 83, 1627 (1988)]. Measurements have been made on additional hammer sets to see if values of the exponent p in a nonlinear force law F = Kξp can be connected either with hammer quality or with other aspects of piano design and history. The corresponding theory of hammer/string interaction requires solution of nonlinear equations of motion, and has previously been done only for a few isolated examples. Steps toward a flexible general computer program that can explore the dependence of string spectra upon striking parameters with nonlinear hammers will be discussed.
Fundamental theory and computer simulation concerning the decay characteristics of piano sound
Authors: Nakamura Isao
Published: October 2005

It is shown by theory and computer simulation that the decay characteristics of piano sound are dependent upon the degree of coupling between the strings. String and soundboard vibration are calculated using an equivalent circuit. Each string is comprised of resonance circuits that correspond to partials, and the behaviors of individual partials are described independently of each other by these circuits. When the most simplified version of only one partial is considered, the equivalent circuit is expressed by the two resonant circuits coupled with the soundboard impedance. The degree of coupling between the two strings is dependent on the ratio of two constants: the degree of mistuning and the ratio of soundboard impedance to string impedance. If the former is smaller than the latter, the two strings are closely coupled with each other and a double decay characteristic results. If the relation between the two constants is reversed, the coupling is loose and a beat type of decay characteristic results. The decay characteristics are changed by the above two constants. The results show that the degree of tuning and the characteristics of the strings and the soundboard determine the decay characteristics of piano sound.
Harpsichord modal behavior
Authors: Savage William R. , Marshall Kenneth , Kottick Edward L. , Hendrickson Thomas
Published: October 2005

The modal behavior of the harpsichord in the acoustics laboratory at the University of Iowa was determined by percussing approximately 600 points on the external surfaces of the instrument with an instrumented hammer. Both the input from the hammer and the response of the harpsichord were recorded. The data were digitized and the response of each tapped point was converted to a numerical value. These data have been processed into a computer-generated, three-dimensional video tape showing the harpsichord's vibrating modes from 0–500 Hz. Of particular interest is the way in which all the external structural elements of the harpsichord can be shown to vibrate at some point. The results of this work agree closely with the published [E. L. Kottick, Galpin Soc. J. 48, 55–77 (1985)] and unpublished data developed by the authors of this paper.
Field measurements of ultrasonic diagnostic equipment
Authors: Ide Masao
Published: October 2005

Recent progress and popularization of ultrasonic diagnostic equipment are remarkable, and the standardization of this equipment, especially of its acoustic output which is related to bioeffects, has become important. IEC began by dealing with the standardization of the measuring methods of acoustic fields. Studies on methods of measuring acoustic fields of diagnostic equipment have been made by such researchers as R. C. Preston, G. R. Harris, P. A. Lewin, and M. Ide, and others [IEEE Trans. Ultrason. Ferroelec. Freq. Contr. 35 (1988)]. Since the field irradiated from the diagnostic equipment is pulsed ultrasound, many parameters have to be considered. The most useful measuring method, at present, is the planar scanning method using a small hydrophone. Using this method, the values of the main parameters can be obtained relatively easily. This paper mainly shows this type of measuring method in the laboratory. By scanning a small hydrophone in water in three dimensions, the acoustic pressure at each point is measured and stored in a computer, and then all the spatial and temporal parameters are computed. Examples of measured acoustic data from ultrasonic diagnostic equipment merchandised in Japan are shown.
Absorption and attenuation in biological media
Authors: Kremkau Frederick W.
Published: October 2005

Attenuation of ultrasound in tissues is due primarily to absorption at the macromolecular level and is dominated by proteins. Protein absorptions are large compared to those for the amino acids of which they are made. Conversion of structured native proteins to random chains reduces absorption significantly in linear proteins but causes only small changes in globular proteins. Phosphate buffer ions dramatically increase absorption in sugars and amino acids but make only small changes in polysaccharides and proteins. Protein absorption increases nonlinearly with concentration. Organized cellular components (plasma membrane and nucleus) have higher absorption than the cell as a whole. Current working hypotheses regarding biomolecular absorption include: (1) The underlying absorption as a function of molecular weight is similar to that for dextran, (2) secondary structure contributes to absorption, (3) solvation contributes to absorption through proton transfer and perturbation of bound-free water equilibria, (4) tertiary structure inhibits solvation absorption in globular proteins, and (5) macromolecular interactions in high concentration, aggregation, and cross-linking conditions contribute to absorption. [Work supported by NCI and NIGMS, DHHS.]
Analysis of echoes from a sphere that includes the directivity of a transmitter and a receiver
Authors: Ueda Mitsuhiro , Morimatsu Eishi
Published: October 2005

Echoes from a sphere are important in biomedical ultrasound in many respects. For example, a sphere is used as a test object for evaluating the resolving power of pulse echo systems, and it shows a characteristic echo pattern such as the cometlike pattern on B-mode images. The speckle pattern on B-mode images can also be modeled by suspension of the spherical particles. The analysis of echoes from a sphere, however, has been limited to the case of plane-wave illumination. Consequently, it has been rather difficult to apply the results of this analysis to medical ultrasound since focused beams have been used in this area. In this report, the planewave theory for echoes from a sphere is extended to the case where arbitrary transmitting and receiving ultrasonic transducers are used, and the effect of the directivity of transducers is analyzed both on the waveform of echoes and on the backscattering coefficient measurement of suspension.
Flow vector mapping through stream function from two-dimensional Doppler imaging
Authors: Ohtsuki Shigeo , Tanaka Motonao , Okujima Motoyoshi
Published: October 2005

The distribution of Doppler velocity in the heart or large vessels can be observed in real time by a sector-scan or linear-scan Doppler ultrasound system. This Doppler velocity means one velocity component detected by the Doppler effect of reflected ultrasound from a moving target. Coding the magnitude of the Doppler velocity to a color, the distribution is displayed in Doppler color imaging. On the other hand, the distribution of a flow vector component on an observing plane gives more information than the Doppler color imaging. In order to derive flow vectors on a plane from Doppler color imaging, a technique to calculate stream functions on a plane was developed. The contour lines of this function are stream lines. The calculation of a stream function consists of the integration of weighted Doppler information. The distribution of a two-dimensional flow vector component can be calculated from this stream function. This technique was applied to the blood flow in the heart.
Numerical models of perfused hyperthermia phantoms
Authors: Shrader Eric J. , Jeuck Phillip R. , Edmonds Peter D.
Published: October 2005

Finite difference numerical models based on bioheat transfer [H. H. Pennes, J. Appl. Physiol. 1, 93–122 (1948)] and incomplete countercurrent heat exchange [S. Weinbaum and L. M. Jiji, J. Biomech. Eng. 107, 121–139 (1985)] have been developed and tested. Predicted performance of hyperthermia phantoms heated by ultrasound applicators will be compared with actual performance to validate the models. The models will then be used as aids in designing ultrasound applicators to produce heat deposition patterns believed advantageous for hyperthermia treatment of experimental animals and patients. The 3D models admit adiabatic or isothermal boundaries and arrays of paired, straight tubules for counter current heat exchange. Results of parametric variations will be shown. Preliminary results obtained by using as inputs the design parameters for a proposed kidney phantom perfused at 110 ml/100 g (of tissue-equivalent gel) per minute have yielded satisfactory time constants on the order of 150 s. When the simulated ultrasound applicator emitted 56 acoustic watts at 1 MHz, applied as an 8-cm-diam, unfocused Gaussian beam, the predicted steady-state, spatial maximum temperature rise was 6 °C. [Work supported by PHS Grant No. R01 CA33749 awarded by the National Cancer Institute, DHHS.]
Ultrasonic imaging of the internal vibration of soft tissue under forced vibration
Authors: Yamakoshi Yoshiki , Sato Jun'ichi , Sato Takuso
Published: October 2005

To estimate the dynamic properties of soft tissues, an imaging system that can display both the amplitude and phase distributions of vibration in soft tissues under forced vibration was developed. In this method, low-frequency sinusoidal vibration with a frequency from 10 Hz to 1 kHz is applied from the surface of the sample, and the movement in it is measured using simultaneously transmitted probing ultrasonic waves. Then scatterers in the sample move sinusoidally under the forced vibration, and the reflected ultrasonic waves are subject to the frequency modulation by the Doppler effect. The amplitude of vibration is estimated from the frequency modulation index of the reflected waves. This is derived from the amplitudes of the spectrum components of the output of the quadrature detector of the received signal. On the other hand, the phase of vibration is estimated from the angle of the fundamental spectrum component. Basic experiments have been carried out using a 3.0-MHz ultrasonic wave system. The accuracy of the method is evaluated and two-dimensional color maps of the amplitude and phase are shown for several phantoms, as well as tissues, in vivo.
Spectral estimation of speech based on a composite cosine wave model
Authors: Saito Shuzo , Tamaribuchi Kazuhiro
Published: October 2005

An algorithm of an analysis for acoustic signals composed of cosine waves and its application for speech signals are reported. It is shown theoretically that when the number of the frequency components of an acoustic signals is m, only 3m discrete sequential data are needed for frequency analysis. It was found that a kind of analysis-by-synthesis technique on the signal waveforms is needed for the analysis of the amplitude-quantized data of the speech signals, setting the allowable error limit on the reconstructed signal amplitude, which is usually set at 0.5 LSB in this paper. This method of analysis is applied to the frequency analysis of several artificial acoustic signals as well as natural speech signals. These signals are sampled at 10 kHz and their amplitudes are quantized at 12 bits and then analyzed. Results show that this method can be used for amplitude quantized acoustic/speech signals under a procedure where the number of frequency components m is set first at a rather large value, e.g., 15, and then an analysis-by-synthesis technique is applied recursively to the reconstructed waveform using the reduced number of frequency components and/or an appropriate allowable error limit.
Detection of increments and decrements in modulation depth of SAM noise
Authors: Grantham D. Wesley , Bacon Sid P.
Published: October 2005

Difference limens (DLs) for the modulation depth of a sinusoidally amplitude-modulated (SAM) wideband noise were measured using a two-interval forced-choice adaptive procedure. For increment detection the reference modulation depth (mr) was 0.0, 0.25, 0.50, or 0.75. For decrement detection, mr was 0.45, 0.60, 0.75, or 0.90. Modulation frequency was 2, 4, 64, or 512 Hz. The DL is defined as 10 log|mthr2 − mr2|, where mthr is the modulation depth just discriminable from mr. For mr = 0.0, the DLs (or “absolute” modulation thresholds) varied from −25 to −10 dB as modulation frequency varied from 2–512 Hz. As mr increased, the increment DL increased monotonically for each frequency from its absolute detection level to about −6 dB at mr = 0.75. These data are in good agreement with those of Wakefleld and Viemeister [J. Acoust. Soc. Am. Suppl. 1 72, S90 (1982)]. The decrement DLs overlay and extended the increment DL functions, although the DLs for the highest mr (0.90) decreased slightly (−10 dB). In a separate manipulation, no effect on the modulation onset phase on modulation depth discrimination was found. [Work supported by NIH.]
Interference of modulation rate discrimination of pure tones
Authors: Yost William A. , Sheft Stanley , Opie Jane
Published: October 2005

Modulation rate discrimination (Δfm) thresholds were obtained for sinusoidally amplitude modulated (SAM) pure tones. The carrier frequencies were 1000 and 4000 Hz and the modulation rates were 5, 10, 20, and 50 Hz. The Δfm thresholds of one carrier (the probe) were also obtained when the other carrier (the masker) was presented simultaneously. The masker was either SAM or was unmodulated. In all conditions the depth of SAM was 100%. This experiment was designed to determine how much the modulated masker interferes with the discrimination of modulation rate and how this interference changes as a function of the difference in modulation rate between the masker and probe. It has been shown previously [Yost and Sheft, J. Acoust. Soc. Am. Suppl. 1 83, S35 (1988)] that a modulated masker interfered significantly with a listener's ability to detect SAM of the probe when the masker and probe were modulated at the same rate. This modulated detection interference (MDI) was reduced when the masker was modulated at a different rate than the probe. These earlier results suggested that there might be auditory channels “tuned” to different rates of amplitude modulation. The present experiment was designed to further test the concept of AM tuned channels. [This research was supported by the NINCDS.]
Detection of changes in frequency- and time-transposed auditory patterns
Authors: Kidd Gary R. , Watson Charles S.
Published: October 2005

It is well known that listeners can recognize familiar melodies that have been transposed in pitch or played in a different tempo [e.g., J. C. Bartlett and W. J. Dowling, J. Exp. Psychol.: Hum. Percept. Perform. 6, 501–515 (1980); R. Francès, La perception de la musique (J. Vrin, Paris, 1958)]. However, the effect of transposition on the recognition of unfamiliar auditory patterns that do not have the pitch-material or stuctural constraints of musical patterns has not been investigated. Listeners' abilities to detect frequency changes in transpositions of randomly generated five-tone patterns were examined using a same-different adaptive-tracking procedure. On each trial a standard pattern was followed by a comparison pattern that was transposed in frequency, time, or both dimensions. The magnitude of a frequency difference introduced in a single tonal component was adjusted from trial to trial, based on the listener's performance. Frequency and time transpositions consisted of increases in frequency and duration, respectively, ranging from 12%–100%. Listeners were told to ignore the changes in absolute frequency and time and make their judgments on the basis of the pattern of intervals. Increases in both frequency and temporal transposition lead to decrements in performance. However, on the average, frequency transposition yields more severely degraded performance than temporal transposition in the range of conditions studied. [Work supported by AFOSR and NIH.]
Effect of attention on hearing ability
Authors: Suzuki Shin'ichi , Usagawa Tsuyoshi , Ebata Masanao
Published: October 2005

The temporal characteristics of attention in auditory signal processing are studied experimentally. The effect of information about the timing of a tone's occurrence on its threshold and DL (difference limen) was examined. When uninformed about the timing of tone occurrence, subjects must keep their attention throughout the experimental period. This concentration makes the subject so tired that he often fails to detect the tone. In the present study, the test tone was presented at fixed and random timings. The threshold was measured using a 1-kHz pure tone at seven levels, and the DL was measured using an overmodulated AM tone by the method of constant stimuli. The results show that the information on the timing of the tones lowers thresholds and reduces DLs. This phenomenon can be regarded as a result of an improvement in the S/N ratio because of the correlation between the input auditory signal and the knowledge of its timing.
Attentional filters and frequency uncertainty in the detection of one or more tones
Authors: Schlauch Robert S. , Hafter Ervin R.
Published: October 2005

A probe-signal method [G. Z. Greenberg and W. D. Larkin, J. Acoust. Soc. Am. 44, 1513–1523 (1968)] is used to study the listening bands for tonal signals in noise. In a two-interval, forced choice paradigm, cues mark the frequency of a randomly chosen target signal on about 70% of the trials while probe signals on the remaining trials have frequencies that differ from the cues by ratios ranging from 0.7–1.4. Based on pretesting, levels are set to produce 90% correct detections when the signal's frequency matches the cue; this is true whether or not the target is roved from trial to trial. Listening bands are studied by plotting all of the data relative to the target frequencies. In a test of a theory that postulates that bandwidth may vary in response to uncertainty [Johnson and Hafter, Percept. Psychophys. 28, 143–149 (1980)], two or more equally likely targets are cued on every trial, allowing for simultaneous measurement of multiple listening bands.
Speech masking. II. Simultaneous masking thresholds under “naturalistic” listening conditions
Authors: Spiegel Murray
Published: October 2005

This study analyzed the role of listening conditions in determining thresholds for probe tones masked by natural speech. These thresholds are of interest because they are a sensitive probe of the activity profile, or “internal” spectrum, of sounds such as speech in the auditory system. Most human performance tests are carried out under highly artificial listening conditions, which may not reflect how people listen to speech in common listening environments. In this study, a reference condition (similar to minimal uncertainty listening conditions used in many performance tests) was compared to a “naturalistic” listening condition and another, intermediate, condition. In the naturalistic listening condition, listeners did not know the frequency or the position of probe tones; additionally, they were required to attend to the semantic content of full sentences. In the reference condition, listeners knew the frequency and position of probe tones masked by single syllables. Average thresholds differed in the two conditions by 4 dB, and thresholds tended to be elevated more for higher-frequency probe tones. The results provide previously unknown information about the resolution of speech sounds in the auditory system during speech comprehension.
Techniques for transducing movements of points on articulatory structures
Authors: Perkell J.S. , Cohen M.H. , Garabieta L.
Published: October 2005

Experience with cineradiography and the x-ray microbeam has shown that midsagittal-plane data on the movements of multiple articulators inside and outside the vocal tract can be extremely useful in studying articulatory-to-acoustic relationships and strategies for speech motor control. This paper describes two recently developed alternating magnetic field systems that produce point-movement data much like the x-ray microbeam. One system uses two external magnetic-field transmitters and small biaxial. receivers that are attached to the articulators; the other uses three magnetic-field transmitters and single-axis receivers. Construction, extensive testing, and simulation of both designs have shown that each system can produce useful data and each has a specific set of advantages and disadvantages. These results are described and the systems are compared with the new x-ray microbeam system at the University of Wisconsin. Examples of data are presented to illustrate their role in the study of speech production, in relation to overall requirements for articulatory data. [Work supported by NIH.]
Structural acoustics in the presence of mean flow
Authors: Crighton David G.
Published: October 2005

Effects of mean flow on the response and radiation of a fluid-loaded elastic structure have often been ignored as being equivalent to Doppler effects on frequency and amplitude, negligible at low Mach number. Mean flow effects are, however, generally much more significant than this, and can profoundly change the whole flow of energy into and through the structure-fluid complex. This paper will illustrate these effects in the context of the response of an elastic plate to single-frequency localized drive in the presence of uniform inviscid flow. For an infinite plate, the response typically consists of exponentially amplifying instability waves down-stream of the drive and two different neutral waves upstream; the drive point admittance generally has negative real part, implying energy flow into the structure and the drive, from the fluid. Approximations for various different parameter regimes will be given to predict these features. Extensions to semi-infinite and finite structures will be outlined together with interpretations in terms of negative-energy waves, and a discussion of viscous boundary layer effects. [Work supported by ONR.]
Effects of random heterogeneity on wave fields
Authors: McCoy John J.
Published: October 2005

A theory of wave propagation in random media must actually refer to a suite of predictive models of the effects of a random heterogeneity. This is because a single general formalism, suitable for all stochastic experiments, would be too complicated for realistic scenarios to have any computational capability. Thus predictive models expressed in partial descriptors of a propagating radiation field, intended for specific experiments are acceptable. In this talk, a number of thought experiments for the propagation of waves in random media will be presented. The point of this is to highlight a richness in the observations one might expect to obtain as a result of random heterogeneity, and to provide a discussion of the possible physical bases for these observations. Appropriate descriptors of the radiation field for each of the thought experiments will be considered, as will be the derivation of prediction models expressed in these measures. A number of specific issues to be addressed are: the possibility of a spatial localization as a result of random heterogeneity; the physical interpretation to be given the completely coherent field; the utility of effective modulus theories; a statistical interpretation for mixture theories; energy flow and coherence formulations; etc.
Vibration and acoustic radiation from inhomogeneous structures: Effect of motion of forcing function
Authors: Pierucci Mauro
Published: October 2005

Homogeneous structures are known to radiate only for supersonic structural wavenumbers. The presence of inhomogeneities such as boundaries, discrete stiffeners, and variable stiffness has been shown to introduce additional modes that will radiate if the structural wavenumbers are supersonic. The radiation pattern for an infinite plate with evenly spaced stiffness gives rise to a stop-passband spectrum. The plate with a continuously varying stiffness also gives rise to similar conditions. All of the previous analyses have always considered stationary forcing functions. In this paper, the analyses for the fixed forcing function will be reviewed and then they will be generalized to the case of the forcing function moving along the plate at speeds up to and greater than the wave speeds in the solid. The results presented will show that for certain values of the frequency, rib spacings, and forcing function velocity, traveling wave modes may change direction, decaying modes may become traveling modes, and modes may change from subsonic to supersonic and vice versa depending upon the relative motion of the forcing function. The effect of the motion may thus enhance or reduce the acoustic radiation.
Zeros of a transfer function in a structural vibrating system
Authors: Tohyama Mikio , Lyon Richard H.
Published: October 2005

This paper analyzes a transfer function (TF) of a structural vibrating system in the complex-frequency domain based on wave theory. There are three cases of zeros in a TF. One is the usual case where a zero is located on the pole line connecting two adjacent poles. Another is the double-zero case, that is, there are two zeros on the pole line between two adjacent poles. These zeros mentioned above are minimum-phase zeros. The third case, however, can contain a nonminimum-phase zero. A pair of minimum- and nonminimum-phase zeros is located symmetrically with respect to each other at equal distances from the pole line. One member of this pair of zeros can be a nonminimum-phase zero under the condition that the damping is small. Such zeros have been numerically confirmed using the TF of three-dimensional space vibration. The truncation effects of an impulse response on the zeros are also mentioned.
A comparison of alternative procedures for digital spectral analysis of speech
Authors: Kania Alex , Jamieson Donald G. , Ramji Ketan , Nearey Terrance M.
Published: October 2005

Researchers now enjoy an increasingly wide selection of alternative procedures with which to characterize the time × frequency × amplitude variation in a speech waveform. Spectral analyses using a filter-bank technique, the digital Fourier transform (DFT), and a variety of linear predictive coding (LPC) procedures are now widely available. Approaches using an autoregressive (AR), moving average (MA), auto-regressive moving average (ARMA), or maximum entropy (ME) procedure are already widely available, or are becoming available. The present paper reviews the assumptions of, and the interrelations between, alternative methods, and outlines the relative advantages of the various approaches in providing information about the speech waveform. Finally, the human factors aspects of alternative spectral displays are discussed.
Analysis of Kohonen's self-organizing feature map for vector quantization of speech
Authors: Li K.-P. , Naylor J.
Published: October 2005

This work presents an analysis of a vector quantization (VQ) process based on Kohonen's self-organizing feature map algorithm [T. Kohonen, Self-Organization and Associative Memory (Springer, New York, 1988), 2nd ed.]. This process is an unsupervised learning technique that converges to a set of ordered nodes that represent cluster centers. These ordered nodes, the VQ codebook elements, are closely related to the sample density function of the training data. The VQ codebooks generated with this algorithm are compared to those generated with a previously developed speech VQ algorithm based on K-means clustering. Performance criteria include the average distortion between input vectors and their coded values and the maximization of entropy, a measure of the distribution of codebook element usage. The Kohonen algorithm provides superior codebooks without requiring more processing. Also, because the codebooks generated by this algorithm are ordered and reflect the statistical characteristics of the training data, the codebooks are useful tools for speech analysis, as well as for speech categorization.
Characteristic behavior of long-term speech vector sums with application to speaker identification
Authors: Thomas Timothy R. , Zakrzewski Wojciech
Published: October 2005

It is known that the parameters derived from linear predictive coding of speech can provide a basis for speaker identification, and that long-term averages improve the reliability of such recognition schemes. The present study obtained long-term descriptors of speech by converting 20-ms frames into unit direction vectors in 14-dimensional cepstral coefficient space. Various numbers of these vectors were then summed, either sequentially or randomly, and the statistical characteristics of the lengths and directions were explored, using both analytical and computational tools. Interspeaker and intersession comparisons were then made, as well as comparisons to sums obtained from randomly generated vectors and from frames selected on the basis of total energy. Several interesting findings were revealed, including the observation that while the average angle between frames and the frame-to-frame changes in direction remained remarkably consistent across speakers and sessions, the direction of the long-term sums was reliably different between speakers, but not sessions. Furthermore, a proper selection on the basis of total energy in the frame can enhance interspeaker differences.
Evaluation of cepstral lifters for articulatory codebook search
Authors: Meyer Peter
Published: October 2005

Of great importance for the success of the articulatory approach to speech coding [Schroeter et al., IEEE-ICASSP, 308–311 (1987)] is the use of a good distance measure between a given speech signal and the entries in a stored codebook of impulse responses and corresponding vocal tract shapes (articulatory codebook). One promising distance measure is the weighted cepstral distance. Since the impulse responses in the articulatory codebook do not include glottal characteristics, optimal weighting functions (cepstral lifters) are derived to reduce the influence of a varying glottal source on the cepstral distance measure. This is done by examining the ensemble of cepstral coefficients of speech produced by an articulatory speech synthesizer that also includes a vocal-cord model. The obtained cepstral lifters are optimal for the given ensemble of cepstral coefficients and for given constraints on the weighting function. They are different for cepstral coefficients derived from the power spectrum (FFT cepstra) and those derived from LPC coefficients (LPC cepstra). The performances of the obtained cepstral lifters are compared in an articulatory codebook search.
Horizontal refraction of normal modes due to variation in bathymetry and subbottom sedimental structure
Authors: Ohta Kazuhiko
Published: October 2005

The horizontal refraction of normal modes propagating over a sloping bottom with range-dependent sedimental structure is studied within the context of adiabatic approximation. Numerical examples show that the effect of the rangewise variation of the subbottom sedimental structure on the horizontal refraction of modes is even more significant than that of the bathymetric variation, depending on the frequency and the mode considered. To investigate the relation between the bottom interaction and the horizontal refraction of modes, the range evolution of the local eigenvalue along the slope, which is proportional to the horizontal refraction index, is examined. Its derivative with respect to range has been derived from the depth equation and can be expressed in an analytical form using depth gradients, mode-depth functions, and the differences in both vertical wavenumbers and densities at each interface. This result may explain the behavior of the horizontal refraction of modes.
A hydrodynamic siren as an underwater sound source
Authors: Yoshikawa Shigeru , Koyano Kiyoshi
Published: October 2005

This paper describes the acoustic characteristics of a hydrodynamic siren in which a perforated disk (rotor) rotates in front of a similar disk (stator) fixed with very slight clearance and compressed water passes through holes in the rotor and stator. The stream is interrupted with the frequency given by the product of the rotation rate times the hole number. The rotor and stator have 8 and 20 holes on their inner and outer circles, respectively. The sound due to the greatest common divisor of 8 and 20 (i.e., 4) is also generated by the superposition of two waveforms derived from 8 and 20 holes. The siren thus produces thickly populated line spectrums that extend from 0.1–5 kHz. The motor and pump are controlled by inverters to stably adjust the rotation rate of the rotor and the pressure of the water, respectively. The maximum sound pressure level integrated over the above spectrum range reaches about 190 dB re: 1 Pa at 1 m when the water pressure is 9 kg/cm2.This siren has been used as a jammer or broadband mixed sound projector for sea experiments on adaptive beamformers.
The development of cues to the perception of the [m]-[n] distinction in CV syllables
Authors: Ohde Ralph N.
Published: October 2005

The contribution of the nasal murmur and vocalic formant transitions to the perception of the [m]-[n] distinction by adult listeners was investigated for speakers of different ages. Three children, ages 3, 5, and 7, and an adult female and male produced CV syllables consisting of either [m] or [n] and followed by [i,æ,u,ɑ]. Three productions of each syllable were modified according to several waveform editing techniques. Preliminary results of listening tests indicate that the murmur and vocalic transitions provide cues to place of articulation, with the latter property more prominent in perception than the former. The simultaneous presence of murmur and vocalic transition cues improved perception of place of articulation for some syllables, particularly for children's speech. The results will be discussed relative to the role of variability in production and integrated cues to the perception of place of articulation of nasal consonants in speech development. [Work supported in part by Biomedical Research Support Grant No. RR-05424.]
Listener experience and perception of voice quality
Authors: Kreiman Jody , Gerratt Bruce R.
Published: October 2005

This study examines the role of listener experience with populations of voices in the perception of vocal quality. Prototype models suggest that the strategy that naive listeners use to evaluate abnormal vocal qualities will differ significantly from that of listeners experienced with dysphonic populations. Listener groups should not differ in perceptual strategy for normal voices, where their experience is equal. The nature of differences in strategy is also examined: as listeners gain experience with populations of voices, do they hear them in more complicated ways, resulting in higher dimensional multidimensional scaling solutions and/or lower r2 values for these solutions? Alternatively, does experience increase the efficiency with which listeners utilize a relatively constant set of perceptual parameters? Six naive and six experienced listeners judged the similarity of 17 dysphonic and 17 normal voices. Separate MDS solutions were found for each listener group for each voice set, and regressions compared the solutions. The relative complexity and efficiency of individual perceptual strategies are also discussed.
Identification of stops in consonant sequences extracted from continuous speech
Authors: Lamel Lori F.
Published: October 2005

At the 114th meeting of the Acoustical Society of America, experiments on the listener's ability to identify singleton stop consonants in syllable-initial and noninitial position, and stops in noninitial homorganic nasal-stop clusters, were reported. The stimuli, consisting of portions of speech extracted from continuous sentences, were drawn from a corpus of about 3600 sentences spoken by over 450 talkers. This paper further investigates the effect of intervening consonants on the listener's decision. Ten listeners identified stops in clusters with semivowels. With the exception of /dr/ and /tr/ (which were confused primarily with the /ǰ,č/), the listener's performance (96.1% correct) was comparable to that of singleton stops (97.1%). Another group of listeners identified stops preceded by /s/ or /z/. The listener's performance (overall 88.3%) was dependent upon the identity of the fricative and whether or not the stop was in a cluster with the preceding /s/. Results will be compared with singleton stop identification tasks. [Work supported by DARPA under Contract No. N00014-82-K-0727, monitored through the Office of Naval Research.]
Fundamental frequency provides voicing information even with unambiguous VOTs
Authors: Whalen D. H. , Abramson Arthur S. , Lisker Leigh , Mody Maria
Published: October 2005

Earlier work [A. S. Abramson and L. Lisker, in Phonetic Linguistics (1985)] demonstrated that falling fundamental frequency (F0) after a syllable-initial stop was a cue to voicelessness, and that flat or rising F0 was a cue for voiced stops, but only when the voice onset time (VOT) was ambiguous. The present study replicated that finding with seven VOT values and five onset F0 values. In the first condition, subjects identified the stop as “b” or “p.” Results were nearly identical to the previous experiment. A second condition included not just the stop decision, but a reaction time as well. Here, inappropriate F0 slowed response time even for unambiguous VOTs. A final condition was, like the first, identification without time pressure. Here, it was found that subjects were distinguishing all five levels of F0 onset so that, the lower the onset was, the more “b” responses were obtained in the ambiguous region. Thus F0 contributes to the voicing distinction, even when the categorization is not changed. Also, F0 cues a “voiced” response incrementally as it starts below the F0 of the remainder of the syllable. [Work supported by NIH Grant No. HD-01994.]
On the role of the fundamental frequency in vowel perception
Authors: Hirahara Tatsuya
Published: October 2005

Vowel identification tests were carried out using 200 synthesized vowel-like stimuli to examine the role of the fundamental frequency F0 in vowel perception. These stimuli were synthetic versions of the five Japanese vowels, /i/, /e/, /a/, /o/, and /u/, of which the F0 and/or the formant frequencies Fi (i = 1,2,3,4) were modified: ten F0 values were formed by adding n/3 Bark (n = 0,1,…,9) to the original F0. Four formant frequency sets were formed by adding m Bark (m = 0,1,2,3) to the original formant frequencies for each vowel. The results are the following: (1) perceived vowel height articulation shifts upward when the F0 shifts upward, while all formant frequencies remain the same: (2) this shift in vowel height is more distinct amid mid and low vowels than for high vowels; and (3) vowel height does not change when the F0 as well as all formant frequencies are shifted upward the same amount along the Bark scale. Further results, along with the hypothesis that a high F0 is regarded as the first formant in middle and low vowel perception, will be discussed.
Frication duration and amplitude rise time as cues to the voiceless fricative/affricate distinction
Authors: Walsh Margaret A. , Kluender Keith R. , Diehl Randy L.
Published: October 2005

This paper describes the perceptual role of frication duration and amplitude rise time in signaling the fricative/affricate distinction in initial position. Sets of edited natural tokens of /∫a/ and /t∫a/ were created in which fricative duration and rise time varied orthogonally. In one experiment, frication duration varied from 100 to 210 ms, and rise time was fixed at either 30 or 80 ms. Frication duration proved to be a robust cue for the fricative/affricate distinction, with longer durations yielding more /∫/ responses. Moreover, the longer value of rise time shifted the /∫/-/t∫/ labeling boundary toward shorter frication durations; thus rise time had an enhancing effect on the perception of the fricative-duration cue. In a second experiment, rise time varied from 20 to 110 ms, while frication duration was fixed at either 140 or 160 ms. Although the fricative-duration parameter had a reliable effect on the percentage of fricative responses, variation in rise time alone had very little effect. These results are analogous to those of Walsh and Diehl [J. Acoust. Soc. Am. Suppl. 1 82, S80 (1987)], who found that transition duration plays a far more significant role than rise time in signaling the stop/glide distinction. [Work supported by NINCDS.]
Delayed pitch fall in Japanese: Perceptual experiment
Authors: Hata Kazue , Hasegawa Yoko
Published: October 2005

Hasegawa and Hata [J. Acoust. Soc. Am. Suppl. 1 83, S29 (1988)] investigated the delayed pitch fall phenomenon in Tokyo dialect Japanese in production data and examined the relationships of perceived accent to the peak location and the steepness of the falling contour. Pitch fall, the only acoustic correlate of the accent in Japanese, sometimes occurred on the syllable following the accented syllable. A delayed pitch fall tends to be steeper the later it occurs. The present paper examines delayed pitch fall from a perceptual point of view. Three-syllable synthetic stimuli /ma ma ma/ were prepared with the F0 peak in different locations and different falling slopes. These stimuli were presented to native Japanese subjects in order to determine whether perception and production are correlated in this phenomenon, i.e., whether a change in the location of the peak and the degree of F0 fall in the second /ma/ causes the accent to be perceived on the first syllable. Implications for speech recognition will be discussed.
Cross-syllabic-position failures of adaptation are not due to acoustic-phonetic cancellation
Authors: Samuel Arthur
Published: October 2005

Why does adaptation with a syllable-initial consonant fail to affect perception of the same consonant in syllable-final position, and vice-versa? One account of this well-replicated result invokes a cancellation explanation [Pisoni and Tash, Percept. Psychophys. 18, 401–408 (1975)]: With the place of articulation stimuli used, the pattern of formant transitions switches with syllablic position, allowing putative phonetic level effects to be opposed by putative acoustic level effects. Three experiments to be reported tested the cancellation hypothesis by preempting the possibility of acoustic countereffects. In experiment 1, the test syllables and adaptors were /r/-/l/ CVs and VCs that do not produce canceling formant patterns across syllabic position. In experiment 2, /b/-/d/ continua were used in a paired-contrast procedure, believed to be sensitive to phonetic, but not acoustic, identity. In experiment 3, cross-ear adaptation, also believed to tap phonetic rather than acoustic processes, was used. All three experiments refuted the cancellation hypothesis. Instead, it appears that the perceptual process treats syllable-initial consonants and syllable-final ones as inherently different. [Work supported by AFOSR.]
Discrimination of level versus nonlevel pitch contours
Authors: Allen George D.
Published: October 2005

Phonetic transcription of intonation often requires a decision as to whether the pitch contour is rising, falling, or level. Although automatic procedures can measure voice fundamental frequency (F0) contours accurately, it is probably more appropriate to categorize them on the basis of what one's ears can discriminate. The fact that glottal periodicity includes jitter adds to the uncertainty of this discrimination. Pulse trains with level and rise-fall F0 contours were presented for discrimination by listeners in a 2I-FC paradigm. Within each pair, the F0 of the nonlevel began low, rose linearly for half the total duration, and then fell linearly to the end. Mean F0 was thus equal for the two items. Preliminary results using one duration (600 ms), three mean F0 levels (100, 125, and 150 Hz), three ΔF0 values (1%, 2%, and 4%), and four jitter amounts (0%, 0.5%, 1%, and 2%) indicate that discrimination increases with increasing ΔF0 and decreases with increasing jitter. Both trends were quite linear across the range of parameters employed in this preliminary study. The PEST procedures are now being used to map these relationships more precisely.