bluuebunny/crossref_metadata_2025 huggingface dataset:
Acoustic source-level measurements for a variety of merchant ships
Authors: Scrimger Paul , Heitmeyer Richard M.
Published: October 2005

This report presents a set of 50 source spectra obtained from merchant ships of opportunity near Genova, Italy. The source spectra were calculated from radiated-noise spectra measured on a towed array together with a transmission-loss spectrum computed from a parabolic equation model. The aggregate source spectra are characterized in terms of a mean source spectrum and three source-level histograms computed for different frequency bands. It is shown that the mean spectrum is comparable in level and shape to a spectrum computed from a well-known empirical model and that the source-level histograms are approximately Gaussian with standard deviations of 5, 5.5, and 6.8 dB, respectively. Finally, a sub-sample of 36 spectra is drawn from the aggregate and separated into three identifiable ship classes, namely, passenger/ferries, cargo ships, and tankers. It is seen that the source spectra for the three different ship classes have comparable means and standard deviations. From this result it is conjectured that both the mean spectrum and the source-level histograms are not sensitive to ship class and can therefore be taken as representative of shipping in other regions with different ship class percentages.
Underwater sound generation by rainfall
Authors: Laville Frédéric , Abbott Grayson D. , Miller Matthew J.
Published: October 2005

This paper presents experimental findings on the mechanism of underwater sound generation by rainfall. Although using underwater sound to measure the rate of rainfall is a promising technique, conflicting models have been proposed for the spectral contributions of the two rainfall sound sources (raindrop impacts on the water surface and air bubble resonances) and correlating rainfall rate to spectral level has proven difficult. In order to resolve these problems, high-speed data acquisition and processing of underwater sounds recorded in a lake under real rain and artificial raindrop conditions were used. The two rainfall sound sources have been identified in the time domain and their respective contributions to the long-term spectrum have been determined: Bubble resonances were found responsible for the spectral peak around 13–15 kHz and raindrop impacts were found responsible for a broadband spectrum with a negative slope. The poor correlation reported in literature between the rainfall rate and the level at 13–15 kHz is now explained by the sensitivity of bubble generation to raindrop distribution and surface conditions. The better correlation obtained outside this frequency range is explained by the systematic occurrence of raindrop impacts.
Evidence of diffuse surface reflections in rooms
Authors: Hodgson Murray
Published: October 2005

In a recent publication [M. Hodgson, J. Acoust. Soc. Am. 84, 253–261 (1988)], predictions were compared with measurements made in an empty scale-model room and in various nominally empty factories and gymnasia. Consistent deviations between prediction and experiment were found. In order to explain these deviations, it was hypothesized that diffuse surface reflections were occurring in the test rooms. To test this hypothesis, further predictions have now been made using a ray-tracing model that takes diffuse surface reflection into account. It was found that excellent agreement between prediction and experiment is obtained as follows: (a) in the scale model, if all surfaces are 10%–40% diffusely reflecting; (b) in nominally empty factories, if the ceiling and walls are 60%–90% diffusely reflecting. This result represents strong evidence that typical untreated room surfaces are strongly diffusely reflecting.
Aperture synthesis for imaging systems using rectangular boundary arrays
Authors: Ji Chaunyi , Kassam Saleem A. , Hoctor Ralph T.
Published: October 2005

A planar antenna array, the elements of which are deployed on the boundary of some finite region in the aperture plane, is known as a boundary array. Such a deployment of array elements can produce a point spread function with a narrow mainlobe, but this type of deployment also results in high sidelobes. Because of the potential for high resolution from a small number of array elements inherent in these arays, it is of interest to investigate their use, both in the context of passive (receive-only) imaging of incoherent source distributions and in the context of active (transmit/receive) imaging of coherent source distributions. In recent years, a technique has been proposed for creating the effect of imaging with a filled circular or elliptical aperture using an aperture consisting only of the boundary of the same circular or elliptical region. This technique consists of the addition of multiple images that are formed with different point spread functions to synthesize an image with the desired point spread function, and it is applicable to both active and passive imaging. In this paper, a technique is presented for creating the effect of imaging with a rectangular aperture using only the boundary of that aperture. This technique is also applicable both for active and passive imaging. Also, simulated images are given of small numbers of point targets from the imaging systems discussed. These are used to motivate a discussion of some of the interesting features of the aperture synthesis scheme.
Quadratic detection of signals with drifting phase
Authors: Veeravalli Venugopal V. , Poor H. Vincent
Published: October 2005

The problem of detecting a sinusoid with drifting phase in the presence of additive white Gaussian noise is considered. A Lorentzian signal model is used, in which the signal to be detected is modeled as a sinusoid whose phase is drifting with Brownian motion. A class of quadratic detectors that trade off coherent and noncoherent averaging of the received waveform is studied. The deflection ratio is used as a performance criterion, and the optimum quadratic detector structure as parametrized by the phase bandwidth is derived. Then, the performance relative to the optimum of a class of suboptical detectors called m-order noncoherent detectors is considered. It is shown that the best detector in this class performs nearly as well as the optimum quadratic detector. This is in sharp contrast with standard envelope detection whose performance is shown to degrade severely in the presence of phase drift. Simulated detection probabilities that verify this performance disparity are also presented.
Speech deterioration in postlingually deafened adults
Authors: Lane Harlan , Webster Jane Wozniak
Published: October 2005

Postlingually deafened adults reading the Rainbow Passage differed from hearing-control subjects in producing greater pitch variability and mean pitch on stressed and unstressed vowels, greater fluctuations in pitch within sentences, less correlation of intrinsic pitch with vowel height and slower temporal parameters. When reading the Phonetic Inventory Sentences, they revealed less differentiation of place of articulation in fricative and plosive consonants. The present findings, taken together with those of longitudinal and implant studies, are applied to constraining models of the role of self hearing in the elaboration of speech.
Method for the location of burst-onset spectra in the auditory-perceptual space: A study of place of articulation in voiceless stop consonants
Authors: Jongman Allard , Miller James D.
Published: October 2005

A method for distinguishing burst onsets of voiceless stop consonants in terms of place of articulation is described. Four speakers produced the voiceless stops in word-initial position in six vowel contexts. A metric was devised to extract the characteristic burst-friction components at burst onset. The burst-friction components, derived from the metric as sensory formants, were then transformed into log frequency ratios and plotted as points in an auditory-perceptual space (APS). In the APS, each place of articulation was seen to be associated with a distinct region, or target zone. The metric was then applied to a test set of words with voiceless stops preceding ten different vowel contexts as produced by eight new speakers. The present method of analyzing voiceless stops in English enabled us to distinguish place of articulation in these new stimuli with 70% accuracy.
Training Japanese listeners to identify English /r/ and /l/: A first report
Authors: Logan John S. , Lively Scott E. , Pisoni David B.
Published: October 2005

Native speakers of Japanese learning English generally have difficulty differentiating the phonemes /r/ and /l/, even after years of experience with English. Previous research that attempted to train Japanese listeners to distinguish this contrast using synthetic stimuli reported little success, especially when transfer to natural tokens containing /r/ and /l/ was tested. In the present study, a different training procedure that emphasized variability among stimulus tokens was used. Japanese subjects were trained in a minimal pair identification paradigm using multiple natural exemplars contrasting /r/ and /l/ from a variety of phonetic environments as stimuli. A pretest-posttest design containing natural tokens was used to assess the effects of training. Results from six subjects showed that the new procedure was more robust than earlier training techniques. Small but reliable differences in performance were obtained between pretest and posttest scores. The results demonstrate the importance of stimulus variability and task-related factors in training nonnative speakers to perceive novel phonetic contrasts that are not distinctive in their native language.
The effects of articulation on the acoustical structure of feline vocalizations
Authors: Shipley Carl , Carterette Edward C. , Buchwald J. S.
Published: October 2005

Feline isolation calls were analyzed, and a model was developed to relate the acoustical features of these calls to the physical processes used in their production. Fifty isolation calls were recorded from each of five cats for a total sample of 250 vocalizations. By combinations of Fourier transform, autocorrelation, and linear prediction methods, the fundamental frequency (glottal-pulse period) F0, the energy of F0, the frequency having maximum energy Fmax (not always F0), and the energy at this frequency were computed. Mean F0 ranged from 400–600 Hz for individual cats. For some cats F0 was consistent within calls, but for other cats sudden shifts in F0 occurred within calls. Here, Fmax was almost always a harmonic of F0 and generally ranged from 1–2 kHz. For individual cats, the energy ratio E = (energy of Fmax/energy of F0) varied from 1 to 60 and the grand average E over the time course of the call varied from about 12 to 38. The mean rms call intensity was an inverted-U function of time. Measured jaw opening was strongly correlated with acoustical features of call. A Bessel-horn model with time-varying flare gave a good account of acoustical parameters such as Fmax. The presence of formantlike resonances in cat vocalizations and the important role of jaw movements (vocal gestures) in the production of these calls suggest that cats may provide a useful model for some aspects of human vocal behavior.
Numerical solutions of Poisson's formula
Authors: Remillard Wilfred J.
Published: October 2005

This Letter describes a numerical method for solving Poisson's formula for an impulsive initial condition and uniform pressure throughout a specified region in space. Four examples are presented, three with known analytical solutions and one (a finite cylindrical source) with no known closed-form solution.
Loudspeaking telephone using echo cancellers and a voice-switching circuit
Authors: Furukawa Hiroki , Ibaraki Satoru , Naono Hiroyuki
Published: October 2005

A new loudspeaking telephone is proposed that makes duplex communication possible using an adaptive voice-switching circuit and two adaptive echo cancellers. An acoustic echo canceller with a delay capacity of 20 ms cancels the acoustic echo between a loudspeaker and a microphone. An electric echo canceller with a delay capacity of 8 ms cancels the electric echo caused by the impedance mismatch between the line and the balancing network. The characteristics of the acoustic echo path are subject to change by movement of the talker. To cope with such changes, two adaptive filters were installed in an acoustic echo canceller. At the start of talk, the voice-switching circuit inserts a large loss to suppress howling, because the echo cancellers do not replicate the impulse responses of the acoustic and electric echo paths at the beginning of the process stage. When the echo cancellers are replicating the impulse responses with the talker's voice, the inserted loss decreases to 3 dB. Duplex communication can be done in this way. An experimental circuit with three digital signal processors showed that duplex communication becomes possible after repeating voice transmission and reception a couple of times.
An estimation of the characteristics of an ultrasonic transducer by means of waveform analysis of its transient response
Authors: Okuyama Daitaro , Mori Yoshizo
Published: October 2005

A transient current is generated when a steplike voltage is applied to a piezoceramic transducer. It is shown that the frequency spectrum and the damped capacity obtained by an FFT analysis of the waveform have a close similarity to the frequency characteristics of the admittance and the damped capacity as measured by an ordinary method. To generate the steplike voltage with a 0.01-μs rise time, a mercury relay and a power source, whose internal impedance is less than 1 ω, are used. The large capacity (about 500 μF) is connected parallel to the output terminal of the dc power source to obtain such a low impedance. The damped capacity is calculated from the data length, the value of the applied dc voltage, and the dc component of the frequency spectrum. The resonant characteristics of the transducer can be obtained from the results of FFT analysis on the transient current waveform. Results for various types of transducers, such as piezoceramic disks, cylinders, and honeycombs, etc., obtained by the present method show good agreement with those obtained under the ordinary techniques.
Discrepancies between published piano scores and composers' recorded performance—Data obtained by a computerized FFT transcribing device
Authors: Katoh Mizue
Published: October 2005

The differences between published piano scores and recordings of the same music performed by its composers are presented. In order to demonstrate objectively such discrepancies, recorded performances of Debussy, Rachmaninoff, and Gershwin were transcribed by a personal computer (NEC-9801) using the fast Fourier transform. In order to verify the findings, the data obtained from Debussy's “La plus que lente” was reproduced on an electronic piano via a computer. The most outstanding differences ascertained were as follows. (1) Notes not written were played. (2) Notes written were omitted. (3) Notes to be played simultaneously were played separately. (4) Tempo and rhythm differed from those indicated on the score. As well as an analysis of the performances of the composers themselves, the FFT transcribing technique can be applied to the analysis of jazz and other music forms of irregular tempo, such as various types of non-Western music. In addition, the results of the FFT analysis can contribute to the fields of musical education, musicology, and so on.
Optical phonons and electron-phonon interactions in high-Tc materials as probed by Raman scattering
Authors: Klein M. V. , Cooper S. L. , Slakey F. , Pazol B. G. , Rice J. P. , Ginsberg D. M.
Published: October 2005

Raman scattering has played two roles in high Tc superconductivity studies: characterization and fundamental physical properties. After one has separated intrinsic spectra from those due to impurity phases, the physical properties of the intrinsic superconducting material may then be studied. In single crystals of YBa2Cu3O7−δ, the most interesting Raman-active mode is one at 330 cm−1 due to O atoms in the “planes.” This mode has nearly pure B1g tetragonal symmetry, even in the orthorhombic phase, and it shows strong softening below Tc. In detail, its Raman spectra reveal a Fano antiresonance with a continuum [Cooper et al., Phys. Rev. B 37, 5920–5923 (1988)] of the same symmetry. An Ag mode at 118 cm−1 assigned primarily to Ba vibrations shows an even stronger antiresonance, but little softening. The continuum responsible for its antiresonance has Ag symmetry and a somewhat different shape from that of the B1g continuum, but both Raman continua extend all the way to zero Raman frequency shift, implying linear low-frequency response. For T≪Tc, the linear response is still present with five times reduced intensity, implying some form of gaplessness to the superconducting state. Efforts to rule out spurious surface effects will be described, and comparisons with results on materials such as Bi2Sr2Ca1Cu2O8 + δ will be made. [Work supported by NSF under DMR 8715103 and DMR 8612860.]
Mechanoelectrical transduction by vertebrate hair cells
Authors: Hudspeth A. J.
Published: October 2005

The initial cellular stage in the cat's response to sound is the transduction of mechanical energy into electrical signals of hair cells of the internal ear. This presentation will summarize in vitro experiments on the biophysical basis of transduction and will present a model for the process. Mechanoelectrical transduction rests upon the activity of transduction channels, relatively nonspecific, cation-permeant ion channels. These mechanically sensitive channels, which number about 100 per cell, appear to lie near the distal tip of the hair bundle. Excitatory stimuli cause transduction channels to open with a latency of only a few microseconds; transduction is therefore unlikely to require a second messenger. The dependence of the channels' rate of response upon the stimulus amplitude suggests that stimulation affects the rate constants for channel opening and closing. Mechanical measurements of changes in the bundle's stiffness during channel gating support a model for transduction in which excitatory stimulation acts through an elastic linkage in the hair bundle to open each channel's gate.
Ultrasonic studies of the roles of the electron-phonon interaction in high-Tc superconductors
Authors: Mase Shoichi , Horie Yuuji
Published: October 2005

The role of the electron-phonon interaction in high-Tc superconductors MBa2Cu3O7 (M = Y, Nd, and others), (La1−xSrx)2CuO4, and BaPb1−xBixO3 were studied by acoustic means. At scores of MHz, only BaPb1−xBixO3 showed an appreciable electronic contribution to the sound attenuation, suggesting the largeness of the electron-phonon interaction. For higher Tc samples, no evidence of the electronic contribution was found, as can be predicted by the absence of the isotope effect in YBa2Cu3O7. Structural instability was, however, investigated for some samples, and the energies of optical modes were estimated. Based on these experimental results and others, the phonon dispersion curves and the density of state curves D(ε) were calculated in agreement with inelastic neutron scattering data on D(ε). The calculated results show that the optical modes due to Cu-O and O–O bonds in YBa2Cu3O7 provide large contributions to the electron-phonon interaction in view of the energy scale and symmetry properties of wavefunctions. There will be some discussion on what factors counterbalance these advantages. Using single crystals of YBa2Cu3O7 and NdBa2Cu3O7, measurements are being carried out to derive more accurate acoustic information.
Coarticulation effects in Japanese velar stop consonants: Observations with dynamic velography
Authors: Suzuki Noriko , Michi Ken-ichi
Published: October 2005

The positioning of the tongue body is strongly influenced by coarticulation effects. The tongue body contact position for /k/ and /g/ varies especially widely, moving in the direction demanded by the surrounding phonemes. Coarticulation effects of post-consonantal vowels on the place of tongue body contact for Japanese velar stop consonants were observed by dynamic velography. The dynamic velography system, developed by the authors is a 36-electrode electropalatographic technique for observing lingual contact with the soft palate. The resulting velograms can be compared with spectrograms made from simultaneous recordings of speech. Three Japanese male subjects produced VCV utterances with stress on the second syllable. The velogram patterns for [aka], [ako], [aga], [ago] were similar, with contact located at the posterior part of the soft palate. Contact for [ake], [aki], [age], [agi] was located more anteriorly. Thus, as suggested by previous findings, the tongue contact position for Japanese /k/ and /g/ is strongly influenced by the vowels following it.
Acoustic measurements of induced slips of the tongue in children and adults
Authors: Smith Bruce L.
Published: October 2005

Although many studies of slips of the tongue have been conducted with adults, very few investigations have considered children's slips of the tongue. Furthermore, because spontaneous slips of the tongue typically are not captured with tape recordings, very few acoustic studies of slips of the tongue have been conducted with any subjects. The present study utilized an elicitation technique to induce slips of the tongue in a group of 5-year-old children and a group of adults. Subjects repeated short tongue-twister phrases (e.g., Swiss wristwatch shop), as well as control phrases that were easier to produce (e.g., Swiss chocolate store). The types of errors that subjects made (substitutions, dysfluencies, etc.) were computed, and acoustic measurements of certain segments were made. One observation made from the acoustic data was that segment durations produced by both the children and the adults were 30%–4.0% longer for the tongue twisters versus the control phrases, even when productions were correct in both conditions. Additional findings and their implications concerning speech production will be presented.
Two-stage adult acquisition of intonational contours
Authors: Keller Eric
Published: October 2005

The precise process of the acquisition of new speech motor patterns in adults has not been examined extensively. In particular, it is unclear how and in which order control over an unfamiliar motor pattern is acquired. The acquisition by two French-language speakers of unfamiliar (Chineselike) intonational contours overlaid on the second syllable of /aCa/ nonsense stimuli was studied. Stimuli were recorded from a native speaker of Beijing Chinese. Thirteen hundred F0 measurements of responses obtained in nine training sessions, spread over 3 weeks, were evaluated for three types of learning: (1) accuracy (coefficients of variation over multiple attempts), (2) proximity to target frequency (Hz), and (3) contour fidelity (similarity between target and response contours, Hz). Results showed two acquisitional phases: A rapid, major improvement on measures (1) and (2), evident over the rest of the training period. There was a significant positive linear correlation between measures (1) and (2). These data support notions of both rapid learning and slow fixation and are thus in support of the concept of stored speech motor control patterns. This contradicts current motor theories that advocate new trajectory calculation for every movement, even in the case of well-established motor patterns. [Work supported by NSERC, Canada.]
What do mimics do when they imitate a voice?
Authors: Papcun George
Published: October 2005

Imitations by both professional and amateur mimics were studied to determine what similarities are achieved between the imitated voice and the imitation thereof. A wide variety of characteristics was approximated, including the following: mean F0, F1, and F2; frequency contours of F0, F1, and F2; degree of nasalization (but not frequency of nasal formants); speech rate and dynamics, including timing, attack, and release characteristics. Contours of F0, F1, and F2 were often matched accurately, even when their absolute frequencies differed considerably from those of the original. Specific images of words and phrases were used, as well as general phonetic characteristics. Imitators tended to concentrate on imitating unusual characteristics of a voice, rather than attempting to imitate all characteristics equally. This observation may be formalized as a model according to which the importance of a parameter is nonlinearly related to the extent to which it diverges from its mean population value. Professional mimics exaggerated the distinctive characteristics of voices and thus may be considered caricaturists rather than mimics, per se.
In vivo and in vitro studies of intracochlear electrical potential gradients
Authors: Brownell W. E.
Published: October 2005

Currents of similar magnitude to those measured in vivo evoke rapid length changes of nearly 1 μm in isolated outer hair cells (OHCs) maintained in vitro. The ability of OHCs to follow electrical stimulation at greater than 8 kHz makes OHC electromotility a strong candidate for the active mechanical process assumed to contribute to the exquisite tuning and sensitivity of the intact cochlear partition. An attempt is made to identify the biophysical mechanism responsible for these rapid movements by measuring the length changes evoked by potential gradients produced with two tight seal whole cell electrodes. During these experiments, a second, considerably slower, type of electrically induced OHC shape change has been observed. The mechanism responsible for this slow, reversible, electrically induced volume change appears unlike that responsible for the fast electromotility. [Work supported by NIH (NS23567) and ONR (441k704).]
Backscattering and forward loss from near-surface bubbles
Authors: McConnell Steven O.
Published: October 2005

Measurements have been made of high-frequency (15–45 kHz) vertical incidence backscatter from near-surface bubbles generated by wave breaking. Interspersed with these backscatter measurements were measurements of the forward loss at low grazing angles owing to absorption by this near-surface bubble layer. In agreement with backscatter measurements from the same experiment reported earlier as well as those reported by others [see articles in Noise in the Sea, edited by B. Kerman (Reidel, New York, 1988)], the bubble volume scattering cross section decreases rapidly away from the surface usually displaying an exponential behavior with depth. Both the backscatter and forward loss data were taken using a rapid sequence of short pulses having different frequencies (100 ms between pulses). Thus far, the results show that the depth-integrated volume scattering strength averaged over a series of pings (typically 50–100 pings) increases linearly with frequency over the measured frequency interval. On a ping-to-ping basis, however, the fluctuations in this depth-integrated quantity show surprisingly little correlation between frequencies and even longer term temporal trends that appear at one frequency are not necessarily reproduced at other frequencies. [Work supported by NORDA.]
Environmental dependence of high-frequency volume scattering
Authors: Holliday D. V. , Pieper R. E.
Published: October 2005

Acoustic volume scattering strengths were measured in the upper 100 m at 21 frequencies between 100 KHz and 10 MHz. The scattering strength and its variability are examined with frequency and depth in the context of the local physical and biological environments. The data for this study were collected in the coastal transition zone off northern California, at the front that separates the Gulf Stream from the continental slope water near Cape Hatteras, in the Gulf Stream and in the southern California bight. [Work supported by ONR and NSF.]
Correlation estimates of high-frequency ocean bottom backscatter
Authors: Stanic S. , Briggs K. , Fleischer P. , Ray R. I. , Sawyer W. B.
Published: October 2005

Two environmental acoustic scattering experiments have been conducted in shallow water off the coast of Florida. The ocean bottom ranged from a smooth homogeneous medium sand to one where the surface roughness was entirely due to a coarse surficial shell layer. The acoustic transmitting system used a pair of narrow-beam parametric sources operating at secondary frequencies from 20–180 kHz. The receiving system consisted of a 16-hydrophone two-dimensional spatial array with broadband capabilities to 250 kHz. Ping-to-ping correlation estimates are presented as a function of frequency, grazing angle, azimuthal angle, and environmental conditions. Cross-correlation estimates as a function of hydrophone separation, frequency, and environmental conditions are also presented.
The sound of cracking sea ice
Authors: Farmer David M. , Xie Yunbo
Published: October 2005

When ice cracks, it radiates energy both into the surrounding ice sheet and directly into the water. A goal in Arctic acoustics is the recovery of information about the cracking process, and thus the ice rheology, from the detailed structure of the acoustic emission. Recent observations, using an array of broadband hydrophones in the Canadian Arctic, have provided new insights on the processes of crack formation. Although it is possible to detect contributions leaking from the ice waveguide, including Crary waves, which can be scattered by discontinuities in the ice, most of the detected acoustic signal travels directly from the source. The resulting signal is interpreted with a moving source model analogous to that used in earthquake mechanics, modulated by fine structure in the cracking process. The fine scale and baseband components of the signal are related to vertical and horizontal scales of the developing crack.
High-frequency acoustic scattering from under-ice microscale surface roughness
Authors: Bishop Garner C.
Published: October 2005

The under-ice surface, characteristic of the interior Arctic, may be partitioned into three more or less distinct roughness regimes. Large-scale under-ice surface roughness is produced by discrete large-scale relief features, e.g., ice keels, and may be parametrized by ice depth that varies from several meters to several tens of meters. Small-scale surface roughness is produced by ice blocks and/or ice rubble, and may be parametrized by the maximum linear dimension of the ice blocks and/or rubble that varies from several tens of centimeters to several meters. Microscale surface roughness is present on the various surfaces of the ice blocks and/or rubble and is produced on the upper ice surface by snow and/or erosion, on vertical surfaces by the fracture of ice, and at the sea water-ice boundary by ice dendrites formed when saline ice freezes, and may be parametrized by an rms roughness parameter that varies from several millimeters to several centimeters. The micro-scale surface roughness produced by ice dendrites has been modeled by an array of close-packed, infinite and parallel semicylinders whose cross section can be varied from circular to elliptical. A Fredholm integral equation of the second kind has been developed and solved numerically to calculate the pressure field scattered from the microscale surface roughness. The effects of various boundary conditions and geometric parameters on the scatter of a high-frequency (>2 kHz) plane wave have been determined. [Research supported by the Office of Naval Technology and Naval Underwater Systems Center IR/IED.]
Articulatory dynamics in long and short vowel pairs
Authors: Harris Katherine S. , Alfonso Peter , Baer Thomas
Published: October 2005

In English, several vowels, such as /i/ and /ı/ may be said to form pairs. The members have been described as “tense” and “lax” because, while they are believed to be grossly similar in vocal tract topography, the lax member of the pair is shorter in acoustic duration and represents a less extreme tract deformation. Temporal and configurational aspects of “tense” and “lax” pairs were investigated by an x-ray microbeam technique using the facilities at the University of Tokyo. Tongue, lip, and jaw trajectories were examined for a single speaker, producing multiple utterances of the form /əpVp/. Preliminary analysis showed that the duration of tongue movement trajectories reflected acoustic duration differences, but configurational differences were more complex. [This research was supported, in part, by grants from NINCDS and the Voice Foundation. The collaboration of the staff of the Institute of Logopedics and Phoniatrics, University of Tokyo, is gratefully acknowledged.]
Speech production models—Constraints and control strategies
Authors: Fant Gunnar , Lin Qiguang , Badin Pierre
Published: October 2005

Our modeling includes five areas of speech production. (1) Extension of the three-parameter vocal tract area function model of Fant (1960) to include bandwidths and to allow for variations of length and degree of asymmetry of the tongue section, and covariation of these parameters as well as overall VT length and configurations in the laryngeal region. (2) Algorithms for inverse transform, i.e., deriving VT parameters from formant frequencies. (3) Control strategies for synthesis of combinations of vocalic sounds from VT area function parameters. (4) A study of essential cavity structures of fricatives and stops with a discussion of source functions, in part based on data from Fant (1960). (5) Discussion of how source-filter interaction may be implemented and how it affects control strategies in formant synthesis based on underlying vocal tract simulations. [This work has been supported by grants from the Swedish Board of Technical Development and the Swedish Telecom.]
Automatic optimization of articulatory text-to-speech
Authors: Coker C. H. , Parthasarathy S.
Published: October 2005

Internal parameters of a text synthesizer are manipulated by an optimization strategy to obtain best fits to natural speech. Working units of the optimization are vectors of parameters for an articulatory model (usually one vector per phoneme) and vectors of time and speed of transition for each parameter. Criteria for optimization are evaluated over a time window spanning a target segment and its adjacent transitions. Problems with excess degrees of freedom (“ventriloquist effects”) are minimized by the long evaluation window and by designating only certain critical parameters as eligible for adjustment for each phoneme. The most serious caveat is that certain sounds inherently produce greater error than others (e.g., fricatives because of variability in spectra of noise), and thus can overshadow subtle effects in adjacent sounds. Error criteria must be carefully chosen and weighted across time to focus attention on perceptually relevant points. Results confirm that synthesis with phoneme-like units can produce speech comparable to reasonably good frame-by-frame speech coders. A single prototype “interpolation function” is sufficient to generate parameter transitions, but length and time of transition must vary significantly with context, especially for the slow parameters that dominate vowels and semivowels.
Applying the program NEWPAR to extract dynamic parameters from movement trajectories
Authors: Smith Caroline L. , Browman Catherine P. , McGowan Richard S.
Published: October 2005

In a model of articulatory movement, based on a dynamical systems approach, parameter values derived from actual articulatory data are being used. One way of obtaining these values relies on the program NEWPAR, which has been developed to analyze articulatory movement in order to extract dynamic parameters such as frequency that will serve as coefficients in the model equation [McGowan et el., J. Acoust. Soc. Am. Suppl. 1 83, S113 (1988)]. Alternative hypotheses for the application of NEWPAR have been tested, including starting the half-cycle windows at different points in the trajectory and fixing either the initial displacement and velocity or the initial and final displacement as a means of reducing the number of parameters to be fitted. Tests have been made on simulated data, consisting of sinusoidal curves with damping varied from completely undamped to critically damped. Patterns in the results of these tests and some preliminary analyses of actual lower lip movement trajectories will be shown. [Work supported by NSF and NIH.]
Aerodynamics in two-dimensional vocal tract models
Authors: DeLucia James
Published: October 2005

A numerically tractable formulation of the Navier-Stokes equations, suitable for studying fluid flow in the vocal tract during speech production and for doing more natural speech synthesis, is described. These qualities are obtained by restricting consideration to two-dimensional vocal tract geometries and by introducing a velocity stream function A (ψ,θ) and a velocity potential Ω (ψ,θ). Here, (ψ,θ,φ) are curvilinear coordinates with φ being ignorable. Taking the divergence and curl of the momentum equation isolates the compressible wave motion in the velocity potential and the background “quasi-steady-state” flow in the stream function. This analytic separation of the two flow components allows for dramatic improvements in the accuracy and efficiency of any numerical simulation as compared to those obtained by directly integrating the primative equations. Full steady-state flow is represented compactly by a “momentum density stream function” Φ (ψ,θ). For invisid flow, a second-order elliptic PDE for Φ is obtained with the fluid density given self-consistently by Bernoulli's law. With viscosity, a higher-order equation for Φ results. Details of the formulation, geometry, and boundary conditions are presented.
A study on the subjective effects of an active-control system for sound fields in an auditorium
Authors: Kato Yoichiro , Ando Yoichi
Published: October 2005

This paper presents subjective effects of an active-digital-control system for sound fields in an auditorium. A pilot study using various kinds of music was made to optimize a multipurpose hall in actual concerts. A wide range of music from baroque to contemporary, such as chorus, flute sonata, string quartet, wind quintet, and music for string ensemble was performed on the stage of a hall with 915 seats. In order to control early reflections and reverberation time as well as the IACC, signals were picked up by several microphones close to the sound sources. These signals were then fed to eight loudspeakers located at the ceiling above the seats after passing through digital-delay machines and digital reverberators. Questionnaires asking subjective preference, loudness, clearness, reverberance, diffuseness, and other related items such as experiences of attending concerts, performing music, etc., were distributed to about 200 listeners. The data for both conditions, on and off, of the system were analyzed. The results indicate that subjective preference for sound fields may be improved by a carefully adjusted active-control system for a wide range of seating areas.
Acoustical properties of ancient Chinese musical qing instrument
Authors: Tong Chen
Published: October 2005

The qings were important percussion musical instruments in ancient China. A qing is a stone plate with a special form. The vibration of a qing is studied both theoretically and experimentally in this paper. The vibrational modes of a qing have been calculated by using a finite element method in accordance with the experimental results. As compared with a rectangular plate, the frequency relations between the overtones are varied with the changes of shape of the qing. From the analysis, it is possible to speculate upon the idea of design of the qing in ancient time. In order to confirm the idea, the sound of a qing has been simulated by computer and judged subjectively. An empirical formula for the calculation of the pitch frequency of a qing sound is given. Some examples show the musical scale of ancient qing groups.
Three-dimensional reconstruction of real-time ultrasonic scans
Authors: Watkin Kenneth L.
Published: October 2005

Real-time ultrasound scan images are gathered at unequally spaced, noncoplanar intervals. Three-dimensional reconstruction of ultrasonic scan images requires information on the location and angle of the ultrasound transducer simultaneously with the ultrasound image. Using a pulsed inductive coil motion monitoring system and a real-time ultrasound scanner, the purpose of the present investigation was to compare measurements of a fetuslike phantom with the three-dimensional reconstructed image of this phantom. The output of the motion sensor and the ultrasound monitoring system were recorded simultaneously, digitized and then used to determine transducer position/angle and the edges of the fetuslike phantom immersed in water. Geometric modeling techniques including finite element mesh generation were used to reconstruct the phantom. Area and volume comparisons of the phantom and reconstructed image were conducted. Although cross-sectional area comparisons revealed similar measurements, accuracy of volume estimations were dependent on the number of slices used for the calculation. Results will be discussed in light of current geometric modeling methods and the reconstruction of ultrasonic scans.
Speech quality assessment in low-bit-rate speech coding taking into consideration quality variation among speakers
Authors: Nagabuchi Hiromi
Published: October 2005

A method for assessing the quality of low-bit-rate speech coding is proposed. This method considers the speaker dependency of coded speech quality, which has not been done before. First, it is shown that coded speech quality varies with speakers more in low-bit-rate coding than in PCM. Next, the power of the linear prediction error of the signal of speech weighted by speech power, Gopt, is introduced as a measure for estimating quality variation among speakers. It is shown that Gopt, can be used as an effective index for selecting speakers in coded speech quality evaluation. Finally, a speech quality assessment method is proposed in which (1) test speakers are selected to cover uniformly the whole range of the Gopt, distribution; and (2) the assessment results are expressed in terms of not only the average but also the standard deviation of measured values for each speaker weighted according to the Gopt distribution characteristics. This method can assess the performance of low-bit-rate speech coding more accurately due to the increased reproducibility of assesed values when applied to different speakers.
Discrimination of recycled word-length sequences
Authors: Bashford James A. , Warrren Richard M.
Published: October 2005

Earlier studies in this laboratory have dealt with the ability to distinguish between permuted orders using recycling or locked sequences of three of four items. More complex “Watson” sequences, presented as single bursts, typically of ten items, have been used to examine discrimination involving “word-length” patterns. The present study combines elements of both techniques, investigating the ability to discriminate minimal changes of order within recycled word-length sequences of 40 ms items: sinusoids (experiment 1); vowels (experiment 2); and frozen noise segments (experiment 3). Listeners made ABX judgments for pairs of ten-item sequences differing only in the ordering of two contiguous items. Pattern pools of 48 sequence pairs were employed in each experiment. This hybrid technique of recycling word-length sequences permits ready discrimination of permuted order: For example, naive listeners presented with tone sequences performed at levels well above chance at the outset of training and approached 96% accuracy in less than 100 trials. Special aspects of results obtained with sequences of vowels and frozen noise will be discussed. [Work supported by AFOSR.]
Informational processing of complex sound
Authors: Lutfi Robert A.
Published: October 2005

Normal-hearing listeners discriminated among complex, nonspeech sounds in which the difference to be discriminated was randomly varied from trial to trial. All other physical parameters of the sounds were fixed within a block of trials. On each trial, the listener heard two sounds (e.g., tone complexes). The values of the variable parameter (e.g., tone frequencies) were drawn from two normal distributions differing only in mean. The listener's task was to identify the sound having the higher mean value of the variable parameter. Discrimination performance was found to be largely independent of the particular physical dimensions along which the sounds varied. Rather, performance appeared to depend primarily on information content of the sounds. Information content was defined in terms of a stimulus equivocation factor that was derived from the data. Based on this model, transmitted information was estimated to be between 1.0–3.0 bits. [Work supported by AFOSR.]
Discrimination of missing fundamental frequency and the influence of competing pitch and timbre cues
Authors: Singh Punita G.
Published: October 2005

The DL for frequency discrimination of complex tones has generally been shown to be smaller than for pure tones of equivalent F0. A notable exception is the case where “residue” tones are used [A. Faulkner, J. Acoust. Soc. Am. 78, 1993–2004 (1985)]. It has been argued that the decline in performance is due to differences in timbre that make pitch discrimination difficult. The present experiment atttempts to determine such perceptual correlates of changes in fundamental frequency for complex tones presented in two-tone sequences. All tones comprised four equal-amplitude harmonics m, (m + 1), (m + 2), and (m + 3), with m = 1, 2, 3, 4, or 5. The first tone in a sequence had F0 = 200 Hz, while the second had F0 = (200 + 2n) Hz, where n = 0, 1, 2, 4, 8, 16, or 32. The spectral locus of tones in a pair was either the same (m,m) or different [(2,m) or (m,2)], where m and 2 denote the harmonic number of the lowest component. Listeners were required to indicate if the tones in a pair were: (1) the same, (2) different in pitch, (3) same in pitch, but different in “something else,” or (4) different both in pitch and “something else.” Results indicate that listeners use labels 1 and 2 for sequences in which tones had the same locus, with relative use of labels systematically varying across differing magnitudes of F0 change. When spectral locus changes, however, a considerable variation of responses is seen, dependent on overall spectral proximity as well as amount of change in F0. The order of locus change across tones, whether higher or lower in frequency, is another factor that influenced label choice. [Work supported by AFOSR.]
Consonance of tones of changing frequency
Authors: Ohgushi Kengo , Morishita Shuji , Hatoh Tadasu , Kurozumi Koh'ichi
Published: October 2005

The tonal consonance of two sinusoids with a temporal change in frequency was measured in psychological experiments. The stimulus consisted of a pair of tone bursts presented simultaneously. The tone bursts were 1s in duration with a continuous change of frequency and had a fixed frequency ratio. The initial frequency of the lower tone from which the frequency change started was 400 Hz, and the rate of the frequency change was 1 oct/s. The ratios of the instantaneous frequencies were from 1.0-2.0. Twenty-one types of stimuli were synthesized by the computer. In the experiment, two stimuli were presented in succession to the subjects who were instructed to judge which of the two stimuli was more consonant. The results revealed the following. (1) The tonal consonance was at its minimum when the frequency ratio was around 1.05. (2) The tonal consonance increased with the increase in the frequency ratio up to around 1.2. (3) The tonal consonance kept nearly the same value and the simple frequency ratios did not result in singular points on the consonance curve when the frequency ratio was over about 1.2.
A new pitch paradox
Authors: Deutsch Diana
Published: October 2005

A pattern of tones is described and explored that has the following paradoxical property: When transposed by a half-octave, the identical pitches may be heard, but as reversed in time. This perceived temporal reversal may even be accomplished when the pattern is transposed by changing the speed of playback on a tape recorder. The behavior of this phenomenon under detailed parametric manipulation is described, and its implications for theories of pitch perception are discussed.