bluuebunny/crossref_metadata_2025 huggingface dataset:
Suppression of human acoustic distortion product: Dual origin of 2f1−f2
Authors: Gaskill Sally A. , Brown Ann M.
Published: October 2005

In order to explore the origin of acoustic distortion product (DP) in the ear canal, the effect of the level and the frequency of the suppressor tone on the level of the DP was investigated. The results broadly support previous studies in showing that, with increasing suppressor level, the DP level shows an initial relatively stable plateau or gradual decline, followed by a rapid decline. The shape of the human iso-suppression contours derived from these ‘‘suppression-rate’’ curves was found to depend upon the size of the iso-suppression criterion. Low-level suppressor tones close in frequency to 2f1−f2 were effective at suppressing the DP by up to 3 dB in subjects where the distortion frequency fell at a region of strong stimulus frequency re-emission. It is probable that contributions from the DP frequency region, as well as components from the primary frequency region, make up the ear canal measured DP. With a 6-dB iso-suppression criterion, suppressor tones in the primary frequency region suppress DP most effectively.
Correlated individual differences in conditions used to measure psychophysical suppression and signal enhancement
Authors: Wright Beverly A.
Published: October 2005

Two-tone suppression and notched-noise signal enhancement (overshoot) were measured at 1000 Hz using standard procedures in 40 normal-hearing, naive adults, counterbalanced for sex, listening ear, and testing order. The threshold range across subjects for the same condition often exceeded 30 dB. In the suppression conditions, for some subjects the 20-ms forward-masked signal was as much as 15 dB easier to detect when the tonal masker (1000 Hz) and tonal suppressor (1150 Hz) were presented together, than when the masker was presented alone, indicating suppression. For other subjects, however, presenting the masker and suppressor together increased thresholds by as much as 13 dB, indicating additional masking. On average, adding the suppressor to the masker increased threshold by 0.6 dB. In the enhancement conditions, for every subject the 20-ms signal was harder to detect when it was presented at the onset of a 500-ms notched-noise masker (notch width: 1000 Hz) than when signal onset was delayed by 400 ms, and was hardest to detect when the signal and masker were gated on and off together for 20 ms. The average thresholds were 7 and 19 dB higher for the two conditions in which the signal was presented at masker onset than for the condition in which the signal was presented after a delay, indicating signal enhancement. The thresholds in all of the listening conditions appeared to be primarily determined by a single factor. However, it is not clear whether the best designation for this factor would be sharpness of frequency tuning or across-channel inhibitory strength.
Analytical and experimental comparisons between the frequency-modulated–frequency-shift measurement and the pulsed-wave–time-shift measurement Doppler systems
Authors: Wilhjelm Jens E. , Pedersen Peder C.
Published: October 2005

In previous publications, a new echo-ranging Doppler system based on transmission of repetitive coherent frequency-modulated (FM) sinusoids in two different implementations was presented. One of these implementations, the frequency-modulated–frequency-shift measurement (FM–fsm) Doppler system is, in this paper, compared with its pulsed-wave counterpart, the pulsed-wave–time-shift measurement (PW–tsm) Doppler system. When using transmitted PW and FM signals with a Gaussian envelope, the parallelism between the two systems can be stated explicitly and comparison can be made between the main performance indices for the two Doppler systems. The performance of the FM and PW Doppler systems is evaluated by means of numerical simulation and measurements of actual flow profiles. The results indicate that the two Doppler systems have very similar levels of performance.
Compliant layer requirements for eliminating the plane wave reflection from submerged inertial panels
Authors: Garrelick Joel M. , Carroll Gerard
Published: October 2005

It is shown that the plane acoustic wave reflection from a submerged panel may be nulled with a compliant coating of constant storage modulus and modest loss factor at all frequencies for which the panel and coating are ‘‘thin.’’
Relaxation dynamics of spontaneous otoacoustic emissions perturbed by external tones. III. Response to a single tone at multiple suppression levels
Authors: Murphy W. J. , Tubis A. , Talmadge C. L. , Long G. R. , Krieg E. F.
Published: October 2005

The relaxation dynamics of spontaneous otoacoustic emissions (SOAEs) interacting with an external tone have been successfully described using a van der Pol limit cycle oscillator model [Murphy et al., J. Acoust. Soc. Am. 97, 3702–3710 (1995) and Murphy et al., J. Acoust. Soc. Am. 97, 3711–3720 (1995)]. Data were collected for an SOAE interacting with a single-frequency ipsilateral suppressor. Transitions between different suppressed states were achieved by adding or removing signal at the suppressor frequency. The relaxation dynamics of the van der Pol model provided a good fit to the data.
A biomechanical laryngeal model of voice F0 and glottal width control
Authors: Farley Glenn R.
Published: October 2005

A simplified mathematical model of the larynx, based on biomechanical principles, is described. Components represented include cartilages (cricoid, thyroid, arytenoids, and corniculates), muscles (thyroarytenoid [TA], cricothyroid pars rectus [CTR], cricothyroid pars oblique [CTO], posterior cricoarytenoid [PCA], lateral cricoarytenoid [LCA], and transverse arytenoid [TrA]), ligaments (cricoarytenoid [CAL], anterior cricothyroid [ACTL], posterior cricothyroid [PCTL], and vocal ligaments [VL]), and subglottal pressure (PS). Model outputs included equilibrium positions of cartilages, the glottal width, and the estimated fundamental frequency (F0) of vocal fold vibration. Major findings were that TA, CTR, CTO, and TrA all had substantial effects on F0; that PCA caused glottal opening by rotating the arytenoids laterally; that LCA both ventrolaterally translated and medially rotated the arytenoids, producing minimal effects on glottal closure; and that TrA had major effects on glottal closure by dorsomedially translating and medially rotating the arytenoids. The effects of LCA and PCA were generally diminished as activation of other muscles was increased. Muscle activation plots (MAPs) were used to study the effects of independent parametric variation of several muscles on F0 and glottal width. Both of these parameters were found to be under simultaneous control by TA, CTR, CTO, and TrA. LCA and PCA also had some influence on F0 and glottal width contours, but this appeared to be of limited functional significance, since changes in F0 tended to be offset by changes in glottal closure. Finally, the functional significance of rotation and subduction of the cricothyroid joint was examined. It was found that the combination of subduction with rotation provided greatest control and range of F0 as muscle activation was varied systematically. Strengths and limitations of the current model are discussed, future developments are suggested, and implications of model results as constraints for neural modeling efforts are described.
Motion of a piano string: Longitudinal vibrations and the role of the bridge
Authors: Giordano N. , Korty A. J.
Published: October 2005

Using strain gauges as a tool, new results have been obtained for the amplitude of the vibrations of the bridge and soundboard of a piano. These results should prove valuable for the construction of theoretical models of the role of the soundboard in sound production. In addition, they indicate that longitudinal string vibrations, i.e., the stretching and compressing of the string, make a very important contribution to the initial sound signal. The amplitude of this longitudinal motion is a nonlinear function of the transverse vibration amplitude. The mechanism by which the longitudinal vibrations couple to the bridge motion has also been considered.
Analysis of nonlinear effects in a piezoelectric resonator
Authors: Pérez R. , Albareda A.
Published: October 2005

A theoretical model that explains the impedance variation due to nonlinear elasticity of piezoelectric materials is described. A modified equivalent circuit is used in order to explain the behavior of a piezoelectric resonator near its series resonance, driven by a high-amplitude signal. A simple relation between the motional impedance Z1 and the motional branch current I1 has been obtained, defining nonlinear impedance coefficients. This model is able to explain and predict: (1) the amplitude-frequency effect, (2) the decreasing of the mechanical quality factor, and (3) the appearance of hysteretic phenomena. An experimental method is proposed to measure the above coefficients, and is used to verify the theoretical model on disk ceramic resonators. Experimental measurements have been carried out to compare the nonlinearity of different ceramic materials.
Low-frequency acoustic wave generation in a resonant bubble layer
Authors: Druzhinin Oleg A. , Ostrovsky Lev A. , Prosperetti Andrea
Published: October 2005

The nonlinear response of a bubble layer subject to harmonic and biharmonic excitation at frequencies smaller than the individual bubble resonance frequency is considered. The nonlinear resonance properties of the layer and generation of difference-frequency signal are studied analytically and numerically. It is found that, for bubble volume fractions β≊10−3 and pump amplitudes of the order of 10−1 atm, the power of the low-frequency signal (including waves radiated in both directions) may reach 10% of the total power of the incident biharmonic wave. The efficiency is restricted by the rapid formation of shocks already at relatively low driving amplitudes, which may not occur in a more complete model accounting for the inertia of the bubble pulsations.
A numerical model for sound propagation through a turbulent atmosphere near the ground
Authors: Chevret P. , Blanc-Benon Ph. , Juvé D.
Published: October 2005

In this paper a series of numerical simulations of the effect of turbulence on the propagation of acoustic waves in the atmosphere are presented. First the technique of representing the turbulence as a set of realizations of a random field generated by a limited number of Fourier modes is described. Through each individual realization, the acoustic waves are propagated in a wide-angle parabolic approximation to obtain the sound-pressure level. Ensemble averaging is then performed to compute the statistical properties of the acoustic field: mean sound-pressure level, intensity fluctuations, and amplitude distributions. The method is applied first to a nonrefractive atmosphere, both in the presence of a rigid boundary and of an impedance ground, and then to an upward refractive atmosphere with an impedance ground. The model, which contains no adjustable parameters, is tested using the experimental data of Parkin and Scholes, Daigle, and Wiener and Keast. Good agreement between numerical simulations and experiments is obtained.
Airflow resistivity of models of fibrous acoustic materials
Authors: Tarnow Viggo
Published: October 2005

A new way of calculating the airflow resistivity of randomly placed parallel cylinders is presented. The calculation is based on Voronoi polygons, and the resistivity is given by the mean spacing between the fibers, their diameters, and the physical properties of air. New explicit formulas for the resistivity are given, which are valid for the cylinder (fiber) concentrations found in acoustic materials. A one-dimensional model consisting of parallel plates with random spacing between the plates is first discussed. Then a two-dimensional model consisting of parallel cylinders randomly spaced is treated for flow parallel and perpendicular to the cylinders. The resistivity formulas are exact for plates and approximate for cylinders.
Distortion product otoacoustic emission (2 f1−f2) amplitude as a function of f2/f1 frequency ratio and primary tone level separation in human adults and neonates
Authors: Abdala Carolina
Published: October 2005

Distortion product otoacoustic emission (DPOAE) (2 f1−f2) amplitude is dependent upon both the frequency ratio and level separation of the eliciting primary tones. In adults it has been established that, on average, a f2/f1 ratio of 1.22 is optimal for evoking the most robust DPOAE; DPOAE amplitude is systematically reduced when f2/f1 ratio is either increased or decreased, thus forming a bandpass function (Harris et al., 1989). The frequency ratio function (DPOAE amplitude×f2/f1 ratio) is thought to reflect the filtering properties of the cochlea (Allen and Fahey, 1993; Brown et al., 1993). Primary tone level separation also influences DPOAE amplitude, with a 10- to 15-dB level difference between the primary tones (L1>L2) typically generating largest amplitude in adults. Equivalent studies have not been conducted in neonates. The present study evoked the 2 f1−f2 DPOAE in adults, term and premature neonates to define the optimal f2/f1 ratio and L1−L2 level separation and to investigate the filtering properties of the developing cochlea. Two f2 frequencies were investigated: 1500 and 6000 Hz. F2 was held constant while f1 was varied to produce 13 frequency ratios. Primary tone level separation varied from 15 to 0- in 5-dB intervals. ANOVA were conducted on the resulting f2/f1 frequency ratio and level separation data. Results showed that the mean optimal frequency ratio for DPOAE generation is comparable in adults and neonates. Also, either a 15- or 10-dB level separation (L1>L2) produced the largest DPOAE amplitude for adults and term neonates whereas DPOAEs from premature neonates appeared to be relatively insensitive to primary tone level separation. The f2/f1 frequency ratio functions were similar in shape, slope, and bandwidth for adults and neonates, suggesting adult-like cochlear filtering prior to term birth. This finding is in agreement with previous work from our laboratory reporting adult-like DPOAE suppression tuning curves in term-born neonates [Abdala et al., Hear. Res. (1996)].
Detection and discrimination of gliding tones as a function of frequency transition and center frequency
Authors: Madden John P. , Fire Kevin M.
Published: October 2005

Two experiments were performed to investigate subjects’ ability to detect and discriminate 50-ms up-glides in frequency in several different conditions. In the first experiment, the subjects indicated which of two signals increased more in pitch. The comparison, or standard signal, was a sinusoid which increased in frequency by either 0, 250, or 500 Hz. Center frequencies were 0.5, 1, 2, 4, and 6 kHz. Discrimination thresholds were obtained in both nonroved and roved conditions. In the roved condition, the actual center frequencies of the signals were varied randomly over a range equal to 0.1 times the nominal center frequency. The second experiment was the same as the first, except that the standard signals were swept over a frequency range equal to 0.5, 1, and 2 times the equivalent rectangular bandwidth (ERB) of the auditory filter at the nominal center frequency. Discrimination thresholds expressed as ΔHz/ERB varied little as a function of center frequency as long as the frequency transition of the standard was a constant proportion of ERB. In addition, discrimination thresholds did not vary significantly as a function of the frequency extent of the standard when the extent was one ERB or less, but doubled when the extent was two ERBs. The relatively small amount of variation in threshold across center frequency and the pattern of variation across different standard transitions supports a place mechanism of frequency coding for these signals based on the detection of changes in the excitation pattern.
Masking produced by broadband noise presented in virtual auditory space
Authors: Carlile Simon , Wardman Daniel
Published: October 2005

An attempt has been made to relate the masking effects studied under dichotic listening conditions to masking seen in the free field. Rather than use a free-field masking paradigm combined with monaural and binaural listening conditions, broadband maskers presented in virtual auditory space (VAS) have been used. Two virtual locations were tested: One was the right interaural axis (+90° from the anterior midline) and the other was 40° right of the anterior midline. Narrow-band (critical bandwidth) dichotic and diotic maskers were also derived from the VAS masker by bandpass filtering around the test frequency. This procedure preserved the interaural differences within the critical band about the test frequency but removed information outside the critical band. Using a diotic target tone of 0.6 kHz with a narrow-band masker centered on 0.6 kHz there was an increase in signal detection in the dichotic conditions when compared to that attributable to either ear alone. Furthermore, there was no further advantage in signal detection at this target frequency when a broadband VAS masker was used. This suggests that for low-frequency targets, the binaural differences within the critical band about the target frequency are sufficient for effective unmasking. In contrast, for a target frequency of 4 kHz, a dichotic narrow-band masker resulted in a reduction in detection compared to that attributable to either ear. However, detection improved to the level attributable to the far ear when a broadband VAS masker was used. This suggests that information outside the critical band is involved in the unmasking of high-frequency targets.
Relative spectral change and formant transitions as cues to labial and alveolar place of articulation
Authors: Dorman Michael F. , Loizou Philipos C.
Published: October 2005

The perceptual salience of relative spectral change [Lahiri et al., J. Acoust. Soc. Am. 76, 391–404 (1984)] and formant transitions as cues to labial and alveolar/dental place of articulation was assessed in a conflicting cue paradigm. The prototype stimuli were produced by two English speakers. The stimuli with conflicting cues to place of articulation were created by altering the spectra of the signals so that the change in spectral energy from signal onset to voicing onset specified one place of articulation while the formant transitions specified the other place of articulation. Listeners’ identification of these stimuli was determined principally by the information from formant transitions. This outcome provides no support for the view that the relative spectral change is a significant perceptual cue to stop consonant place of articulation.
The representation of Japanese moraic nasals
Authors: Otake Takashi , Yoneyama Kiyoko , Cutler Anne , van der Lugt Arie
Published: October 2005

Nasal consonants in syllabic coda position in Japanese assimilate to the place of articulation of a following consonant. The resulting forms may be perceived as different realizations of a single underlying unit, and indeed the kana orthographies represent them with a single character. In the present study, Japanese listeners’ response time to detect nasal consonants was measured. Nasals in coda position, i.e., moraic nasals, were detected faster and more accurately than nonmoraic nasals, as reported in previous studies. The place of articulation with which moraic nasals were realized affected neither response time nor accuracy. Non-native subjects who knew no Japanese, given the same materials with the same instructions, simply failed to respond to moraic nasals which were realized bilabially. When the nasals were cross-spliced across place of articulation contexts the Japanese listeners still showed no significant place of articulation effects, although responses were faster and more accurate to unspliced than to cross-spliced nasals. When asked to detect the phoneme following the (cross-spliced) moraic nasal, Japanese listeners showed effects of mismatch between nasal and context, but non-native listeners did not. Together, these results suggest that Japanese listeners are capable of very rapid abstraction from phonetic realization to a unitary representation of moraic nasals; but they can also use the phonetic realization of a moraic nasal effectively to obtain anticipatory information about following phonemes.
Evaluation of various sets of acoustic cues for the perception of prevocalic stop consonants. II. Modeling and evaluation
Authors: Smits Roel , Bosch Louis ten , Collier René
Published: October 2005

The purpose of the study presented in this paper and the accompanying paper [Smits et al., J. Acoust. Soc. Am. 100, 3852–3864 (1996)] is to evaluate whether detailed or gross time-frequency structures are more relevant for the perception of prevocalic stop consonants. To this end, first a perception experiment was carried out with ‘‘burst-spliced’’ stop-vowel utterances. This experiment is described in the accompanying paper. The present paper describes the second part of the investigation, i.e., the simulation of the behavior of the listeners in the perception experiment. First, a number of detailed and gross cues are measured on the stimuli. Next, these cues are mapped onto the observed perceptual data using a formal model of human classification behavior. The results show that in all cases the detailed cues, such as formant transitions, give a better account of the perceptual data than the gross cues, such as the global spectral tilt and its initial change. The best-performing models are interpreted in terms of the acoustic boundaries which are associated with the perceived linguistic contrasts. These boundaries are highly interpretable linear functions of five or six acoustic cues, which give a quantitative description of the often-discussed ‘‘trade-off’’ relation between the various cues for perception of place of articulation in stop consonants.
Consistency among speech parameter vectors: Application to predicting speech intelligibility
Authors: Power Matthew H. , Braida Louis D.
Published: October 2005

Previous researchers interested in physical assessment of speech intelligibility have largely based their predictions on preservation of spectral shape. A new approach is presented in which intelligibility is predicted to be preserved only if a transformation modifies relevant speech parameters in a consistent manner. In particular, the parameters from each short-time interval are described by one of a finite number of symbols formed by quantizing the output of an auditory model, and preservation of intelligibility is modeled as the extent to which a one-to-one correspondence exists between the symbols of the input to the transformation, and those of the output. In this paper, a consistency-measurement system is designed and applied to prediction of intelligibility of linearly filtered speech and speech degraded by additive noise. Results were obtained for two parameter sets: one consisting of band-energy values, and the other based on the ensemble interval histogram (EIH) model. Predictions within a class of transformation varied monotonically with the amount of degradation. Across classes of transformation, the predicted effect of additive-noise transformations was more severe than typical perceptual effects. With respect to the goal of achieving predictions that varied monotonically with human speech-perception scores, performance was slightly better with the EIH parameter set.
Patterns of note onset asynchronies in expressive piano performance
Authors: Repp Bruno H.
Published: October 2005

This paper presents an analysis of the asynchronies among nominally simultaneous notes in ten graduate student pianists’ performances of three compositions (Schumann’s ‘‘Träumerei,’’ Debussy’s ‘‘La fille aux cheveux de lin,’’ and Chopin’s Prelude in D-flat major), each repeated twice and recorded in MIDI format on a Yamaha Disklavier. Note onset times were sensed from hammer motion shortly before hammer-string contact. A pervasive tendency was found for the highest-pitched notes (usually the principal melody) to lead lower-pitched notes, especially those played with the same hand. Inner notes of within-hand chords tended to lag behind outer notes. Strong correlations between average MIDI velocity difference and average lead time were found within each hand, as well as between hands for some of the pianists. Other pianists had a tendency to lead with the left hand, independent of MIDI velocity. These individual differences in between-hand coordination were stable across the three compositions and did not reflect handedness. The results suggest that within-hand asynchronies and melody leads are largely a consequence of dynamic differentiation of voices (i.e., an artifact of hammer travel time), whereas left-hand leads are an individual characteristic and, in part, a deliberate expressive strategy exhibited by some pianists.
A lumped parameter model for the acoustic power output from a vibrating structure
Authors: Fahnline John B. , Koopmann Gary H.
Published: October 2005

Previous applications of lumped parameter models to acoustic radiation problems assume that the characteristic dimension of the vibrating structure is small in comparison to the acoustic wavelength. In this paper, the frequency range of the lumped parameter model is extended by dividing the surface of the structure into elements and characterizing the amplitude of the radiation from each element by its volume velocity. The model is derived by truncating all but the lowest-order (monopole) terms of a multipole expansion for the acoustic power output. The multipole expansion differs from those derived previously because it is based on elemental quantities rather than global quantities. By comparing the full multipole expansion for the power output to the lumped parameter model, the error in the lumped parameter model as a function of the acoustic and structural wavelengths (k and K) and the size of the largest surface element (L) is determined. This approach is general and provides a means of determining bounds on the accuracy of any lumped parameter model based on elemental quantities. For example, the analysis predicts that when the overall volume velocity of a vibrating structure is nonzero, the maximum possible error in the lumped parameter model is equal to C(kL)(KL), where C is a constant. Likewise, when the overall volume velocity of a vibrating structure is zero, the model predicts that the maximum possible error in the lumped parameter model is equal to C′(KL)(L/R12), where C′ is another constant, and R12 is the largest distance between any two points on the structure. The results of the analysis show that it is desirable to formulate acoustic models in terms of elemental volume velocities, because the power output predicted by any such model converges absolutely to the correct solution as the element mesh is refined.
An electromagnetic liquid shock wave generator for the production of a pulsed water jet
Authors: Mortimer Bruce J. P. , Skews Beric W.
Published: October 2005

In this study the feasibility of using a novel adaptation of an electromagnetic source (EMAS) for the production of a pulsed water jet is investigated. This device consists of a high-voltage, 4 to 8-μF, 20-kV capacitor which is discharged through a flat pancake coil. An insulated metal disk is placed in close proximity with the coil and is mounted in a water-filled vessel. Lorentz forces due to the eddy currents induced in the disk accelerate the disk away from the coil. The disk acceleration produces a pressure shock wave which is focused by a convergent reflector onto a nozzle. The reflection of this shock wave from the water–air interface produces a discrete water jet. A theoretical analysis of the electrical characteristics of the transducer is presented. A finite-element package was used to study different convergent reflector shapes. Experimental results from a prototype generator are given, including pressure measurements from a needle hydrophone and shadowgraph photographs of the water jet.
Absorption of sound by noise in one dimension
Authors: Larraza Andrés , Denardo Bruce , Atchley Anthony
Published: October 2005

Experimental results are presented for the excess attenuation of monofrequency waves in the presence of high-intensity noise in one dimension. The theory by Rudenko and Chirkin predicts a Gaussian attenuation of a weak signal in the presence of shockless noise, thus showing that in one dimension translational invariance has been broken. The theory is modified to incorporate wall losses providing, only then, excellent agreement with experiment. The agreement is shown as the frequency, noise level, and distance from the source are varied. In addition, the spectral intensity of the high-frequency tail of fully developed shockless noise is observed to be an f−3 power law in the frequency f, in accord with theory. This power law is a consequence of the far off equilibrium nature of the system.
Inversion scattering in finite space domain
Authors: Novotný Tomáš , Matsumoto Eiji , Shibata Toshinobu
Published: October 2005

The general integral representation of a scattered acoustic field with arbitrary boundary conditions is introduced. Assuming weakness of the scatterer, two linearized methods for the recovery of an object’s shape and position are derived and briefly discussed. The methods are applied to one- and two-dimensional problems in order to compare them with the standard use of the first Born approximation (1BA). By presenting the numerical examples, this paper discusses the influence of the power spectrum of the incident pulse on exactness of the recovered object, the influence of ‘‘weakness’’ of scatterer, and comparison of proposed methods to 1BA.
Use of a correlational method to estimate a listener’s weighting function for speech
Authors: Doherty Karen A. , Turner Christopher W.
Published: October 2005

The purpose of this study was to determine if it is feasible to use the correlational method (Lutfi, 1995; Richards and Zhu, 1994) to estimate how listeners use or weight the information contained within various frequency bands of speech. Three naturally spoken vowel–consonant–vowel (VCV) syllables (/aba/, /aga/, and /ada/) were presented monaurally to listeners. Each of the VCV waveforms were filtered into three separate frequency bands (i.e., low, mid, and high). Each band was then independently and randomly degraded at various signal-to-noise (S/N) levels (−7, −5, −3, −1, or +1). On each trial, listeners were asked to identify the VCV that was presented to them. For each trial, the S/N level of each frequency band, the stimulus that was presented, and the listener’s responses were all recorded and stored in a file. From this trial-by-trial data, a point biserial correlation was computed between the listener’s response (correct or incorrect identification) and the degradation within each frequency band. The stronger the correlation, the greater influence that given frequency band had on the listener’s performance on the task. From these relations it was shown that it is possible to obtain a listener’s weighting function for speech. Results showed that although most listeners weighted the mid-frequency band the greatest, several of the listeners used different weighting strategies to perform the task. Several methodological issues are discussed in regard to improving the future application of the correlational method to speech.
The role of individualized headphone calibration for the generation of high fidelity virtual auditory space
Authors: Pralong Danièle , Carlile Simon
Published: October 2005

The auditory system localizes a sound source on the basis of binaural cues as well as monaural cues provided by the outer ear. These cues are described by the head-related transfer functions (HRTFs). The individual variation in the frequency and amplitude of perceptually significant features in the human HRTFs raises the issue of whether a given listener can obtain adequate localization cues from stimuli based on another listener’s HRTFs. This question can be addressed in ‘‘virtual auditory space,’’ using headphone stimuli which are convolved with HRTFs and the inverse of the headphone transfer function (HpTF). Using an in-ear recording system we have measured HpTFs (Sennheiser 250 Linear) for both ears of ten subjects and found that they also display significant individual differences in the 4–10 kHz region. Therefore, the proper reconstruction of both individualized and nonindividualized HRTFs requires that the listener’s HpTF be deconvolved from the stimulus. We have verified this acoustically using the in-ear recording system for both individualized and nonindividualized HRTFs. Finally, we show that the use of nonindividualized HpTFs creates considerable distortions in the 4–10 kHz range, which could partly account for the increased incidence of mislocalizations reported in previous virtual auditory space localization experiments.
A developmental study of vowel perception from brief synthetic consonant–vowel syllables
Authors: Ohde Ralph N. , Haley Katarina L. , McMahon Christine W.
Published: October 2005

The purpose of this study was to assess the perceptual role of brief synthetic consonant–vowel syllables as cues for vowel perception in children and adults. Nine types of consonant–vowel syllables comprised of the stops [b d g] followed by the vowels [i 𝕒 u] were synthesized. Stimuli were generated with durations of 10, 30, or 46 ms, and with or without formant transition motion. Eight children at each of five age levels (5, 6, 7, 9, and 11 years) and a control group of eight adults were trained to identify each vowel in a three-alternative forced-choice (3AFC) paradigm. The results showed that children and adults extracted vowel information at a generally high level from stimuli as brief as 10 ms. For many stimuli, there was little or no difference between the performance of children and adults. However, developmental effects were observed. First, the accuracy of vowel perception was more influenced by the consonant context for children than for adults. Whereas perception was similar across age levels for stimuli in the alveolar context, the youngest children perceived vowels in the labial and velar contexts at significantly lower levels than adults. Second, children were more affected by variations in stimulus duration than were adults. This finding was particularly prominent for the syllable [g𝕒], where the dependency on duration decreased with age in a nearly linear fashion. These findings are discussed in relation to current hypotheses of vowel perception in adults, and hypotheses of speech perception development.
Bioeffects of positive and negative acoustic pressures in vivo
Authors: Bailey M. R. , Dalecki D. , Child S. Z. , Raeman C. H. , Penney D. P. , Blackstock D. T. , Carstensen E. L.
Published: October 2005

In water, the inertial collapse of a bubble is more violent after expansion by a negative acoustic pressure pulse than when directly compressed by a positive pulse of equal amplitude and duration. In tissues, gas bodies may be limited in their ability to expand and, therefore, the relatively strong effectiveness of negative pressure excursions may be tempered. To determine the relative effectiveness of positive and negative pressure pulses in vivo, the mortality rate of Drosophila larvae was determined as a function of exposure to microsecond length, nearly unipolar, positive and negative pressure pulses. Air-filled tracheae in the larvae serve as biological models of small, constrained bubbles. Death from exposure to ultrasound has previously been correlated with the presence of air in the respiratory system. The degree of hemorrhage in murine lung was also compared using positive and negative pulses. The high sensitivity of lung to exposure to ultrasound also depends on its gas content. The mammalian lung is much more complex than the respiratory system of insect larvae and, at the present time, it is not clear that acoustic cavitation is the physical mechanism for hemorrhage. A spark from an electrohydraulic lithotripter was used to produce a spherically diverging positive pulse. An isolated negative pulse was generated by reflection of the lithotripter pulse from a pressure release interface. Pulse amplitudes ranging from 1 to 5 MPa were obtained by changing the proximity of the source to the biological target. For both biological effects, the positive pulse was found to be at least as damaging as the negative pulse at comparable temporal peak pressure levels. These observations may be relevant to an evaluation of the mechanical index (MI) as an exposure parameter for tissues including lung since MI currently is defined in terms of the magnitude of the negative pressure in the ultrasound field.
An exploration of listener strategies in the lexical segmentation of hypokinetic dysarthric speech
Authors: Liss Julie M. , von Berger Stephanie , Caviness John , Adler Charles , Edwards Brian
Published: October 2005

This investigation examined listener transcriptions of phrases produced by speakers with mild to severe hypokinetic dysarthria to examine individual strategies for the identification of word boundaries in connected speech. It was hypothesized that the most efficient listeners would use syllabic strength information to guide their parsing of the continuous acoustic stream. Six-hundred transcribed phrases (10 listeners ×60 phrases) were coded independently by two judges to identify (1) correct word transcriptions, (2) evidence of accurate lexical parsing, regardless of exact word identification, and (3) the proportions of accurate parsing of strong and weak syllable word onsets. Linear regression analysis of the group data revealed that correct segmentation of strong syllable word onsets predicted listener performance on segmenting weak syllable word onsets [R=0.805, F(1,29)=51.440, p<0.001]. Despite a wide range of listener performance on ``words correct,'' individual strategies for perceptual segmentation were evident only in the transcriptions for the most severe speaker. In this case, the poorest listeners exhibited disproportionate difficulty with the segmentation weak syllable word onsets. Results suggest that a listener's ability to use syllabic strength information in lexical parsing determines, in part, their ability to recognize word onsets. [Work supported by NIDCD, NIH.]
Noninvasive temperature estimation in tissue via ultrasound echo-shifts. Part I. Analytical model
Authors: Maass-Moreno Roberto , Damianou Christakis A.
Published: October 2005

Temperature changes in tissue, caused by high-intensity focused ultrasound, cause time shifts in the echoes that traverse the heated tissue. These time shifts are caused by thermally induced changes in the distribution of the velocity of sound and by thermal expansion within the tissue. Our analytical model relates these shifts to changes in temperature distribution. It is proposed that these relationships can be used as a method for the noninvasive estimation of temperature within the tissue. The model shows that the echo shifts depend mostly on changes in the mean velocity along the acoustical path of the echoes and that no explicit information about the shape of the velocity distribution is required. The effects of the tissue thermal expansion are small in comparison, but may be significant under certain conditions. The theory, as well as numerical simulations, also predicts that the time shifts have an approximately linear behavior as a function of temperature. This suggests that an empirical linear delay–temperature relationship can be determined for temperature prediction. It is also shown that, alternatively, the distribution of temperature in the tissue can be estimated from the distribution of echo delays along the acoustical path. In the proposed system, low-level pulse echoes are sampled during brief periods when the high-intensity ultrasonic irradiation is off, and thus linear acoustic behavior is assumed. The possibility of nonlinear aftereffects and other disturbances limiting this approach is discussed.
Noninvasive temperature estimation in tissue via ultrasound echo-shifts. Part II. In vitro study
Authors: Maass-Moreno Roberto , Damianou Christakis A. , Sanghvi Narendra T.
Published: October 2005

Time shifts in echo signals returning from a heated volume of tissue correlate well with the temperature changes. In this study the relationship between these time shifts (or delays) and the tissue temperature was investigated in excised muscle tissue (turkey breast) as a possible dosimetric method. Heat was induced by the repeated activation of a sharply focused high-intensity ultrasound beam. Pulse echoes were sent and received with a confocal diagnostic transducer during the brief periods when the high-intensity ultrasonic beam was inactive. The change in transit time between echoes collected at different temperatures was estimated using cross-correlation techniques. With spatial-peak temporal-peak intensities (ISPTP) of less than 950 W/cm2, the delay versus temperature relationship was fit to a linear equation with highly reproducible coefficients. The results confirmed that for spatial-peak temperature increases of ∼10 °C, temperature-dependent changes in velocity were the single most important factor determining the observed delay, and a linear approximation could produce accurate temperature estimations. Nonlinear phenomena that occurred during the high-intensity irradiation had no significant effect on the measured delay. At ISPTP of 1115–2698 W/cm2, the delay–temperature relationship showed a similar monotonically decreasing pattern, but as the temperature peaked its slope gradually increased. This may reflect the curvilinear nature of the velocity–temperature relationship, but it may also be related to irreversible tissue modifications and to the use of the spatial-peak temperature to experimentally characterize the temperature changes. Overall, the results were consistent with theoretical predictions and encourage further experimental work to validate other aspects of the technique.
Comparing frequency-domain with time-domain solutions for a locally active nonlinear model of the cochlea
Authors: Kanis Luc J. , de Boer Egbert
Published: October 2005

Previous papers [e.g., Kanis and de Boer, J. Acoust. Soc. Am. 94, 3199–3206 (1993); 96, 2156–2165 (1994)] have described and applied an approximation method, called the quasilinear method, to solve nonlinear locally active cochlea models in the frequency domain. This method consists of treating higher-order distortion components as perturbations. This letter compares results obtained with the quasilinear method with results computed in the time domain. The time-domain method described by Diependaal et al. [J. Acoust. Soc. Am. 82, 1655–1666 (1987)] was used. The comparison for single-tone as well as for two-tone stimulation was performed. In most cases the match between the two solutions is excellent.
Sex differences in simultaneous masking with random-frequency maskers
Authors: Neff Donna L. , Kessler Christina J. , Dethlefs Theresa M.
Published: October 2005

Previously published data for 49 normal-hearing, young adult listeners in a simultaneous-masking task with high stimulus uncertainty were reexamined with regard to possible differences in performance for male versus female listeners. The task was the detection of a 1000-Hz, sinusoidal signal in the presence of multitonal simultaneous maskers comprised of sinusoids whose frequencies were drawn at random from a wide range with each stimulus presentation. The analysis focused on the ten-component masking condition, which included all listeners and showed large individual differences in performance. The average quiet threshold for the signal was near 6 dB SPL for both the 27 males and 22 females, but average masked threshold was 7–8 dB higher for females. More females than males were represented in the highest quartile or upper half of the performance range (‘‘high threshold’’), with more males than females in the lowest quartile or half (‘‘low threshold’’). Particularly for tasks exhibiting large individual differences, more attention should be paid to the possibility of sex differences in performance.
Relation between the phonation threshold pressure and the prephonatory glottal width in a rectangular glottis
Authors: Lucero Jorge C.
Published: October 2005

Recent experimental measurements have shown a nonlinear relation between the phonation threshold pressure and the prephonatory glottal width, with a minimum for the threshold pressure which would indicate the existence of an optimal glottal width for ease of phonation [I. R. Titze et al., J. Acoust. Soc. Am. 97, 3080–3084 (1995)]. This relation is studied analytically using a simplified vocal fold model which includes an explicit term for the air pressure losses due to glottal viscous resistance. It is shown that the observed nonlinearity may be a consequence of the viscous pressure losses, which cause an increase of the threshold pressure at small values of the prephonatory glottal width.
Monitoring of the transport of the ocean current by an acoustic transceiver array
Authors: Takeuchi Tomoyoshi , Taira Keisuke
Published: October 2005

Current velocity can be determined by measuring the difference of the sound wave traveling time between two points. The multipaths inverted echo sounder (MIES) [Takeuchi et al., J. Acoust. Soc. Jpn. 49, 543–550 (1993)] was developed in order to apply this method to the measurement of the volume transport of the Kuroshio. Measurement of mean current velocity can be made with a difference of reciprocal travel times of the sound wave along two sides of a triangle in a vertical plane, which is constructed by the acoustic paths with a base side of the mooring distance when two multipath inverted echosounders were deployed 10 km apart. The measurement of volume transport was attempted by applying the above method, where three multipath inverted echosounders were deployed so as to construct a regular triangle on the sea bottom. In this paper, the result of the measurement of the volume transport of the Kuroshio over Izu Ridge from 20 March to 25 April 1995 is presented.
Ocean current effects on a low-frequency acoustic field and the feasibility of their use to monitor ocean dynamics
Authors: Godin Oleg A.
Published: October 2005

Ray-travel-time nonreciprocity has been used in most tomography experiments to determine current velocity in the ocean by acoustic means at distances large compared to ocean depth. Although very successful in deep water, this approach is not applicable in the coastal ocean where ray arrivals are not separable and/or identifiable due to multiple bottom interactions. In this paper, other parameters of the acoustic field, including normal mode travel time, ray and mode horizontal refraction angles, and full field phase, are treated as possible data for the current velocity field. Some qualitative differences between acoustic fields in moving and motionless fluid are indicated and their importance for acoustic monitoring of ocean currents is emphasized. Existing mathematical models of low-frequency underwater sound propagation in a moving ocean are discussed. It is demonstrated that nonreciprocity of various acoustic field variables possess quite a different sensitivity to the flow velocity field and robustness with respect to unavoidable uncertainties in our knowledge of system and environmental parameters. Full-field inversion, based on an appropriate acoustic field variable, is concluded to be a promising technique for current velocity remote sensing in the coastal ocean environments. [Work supported by NSERC and RBRF.]