bluuebunny/crossref_metadata_2025 huggingface dataset:
Synthetic aperture imaging using sources with finite aperture: Deconvolution of the spatial impulse response
Authors: Lingvall Fredrik , Olofsson Tomas , Stepinski Tadeusz
Published: July 2003

A method for ultrasonic synthetic aperture imaging using finite-sized transducers is introduced that is based on a compact, linear, discrete model of the ultrasonic measurement system developed using matrix formalism. Using this model a time-domain algorithm for deconvolution of the transducer’s spatial impulse responses (SIRs) is developed that is based on a minimum mean square error (MMSE) criterion. The algorithm takes the form of a spatiotemporal filter that compensates for the SIRs associated with a finite-sized transducer at every point of the processed image. A major advantage of the proposed method is that it can be used for any transducer, provided that its associated SIRs are known. This is in contrast to the synthetic aperture focusing technique (SAFT), which treats the transducer as a point source. The performance of the method is evaluated with simulations and experiments, performed in water using a linear phased array. The results obtained using the proposed method are compared to those obtained with a classical time-domain SAFT algorithm. For a finite aperture source, it is clearly shown that the resolution obtained using the proposed method is superior to that obtained using the SAFT algorithm.
Whole-lung resonance in a bottlenose dolphin (Tursiops truncatus) and white whale (Delphinapterus leucas)
Authors: Finneran James J.
Published: July 2003

An acoustic backscatter technique was used to estimate in vivo whole-lung resonant frequencies in a bottlenose dolphin (Tursiops truncatus) and white whale (Delphinapterus leucas). Subjects were trained to submerge and position themselves near an underwater sound projector and a receiving hydrophone. Acoustic pressure measurements were made near the thorax while the subject was insonified with pure tones at frequencies from 16 to 100 Hz. Whole-lung resonant frequencies were estimated by comparing pressures measured near the subject’s thorax to those measured from the same location without the subject present. Experimentally measured resonant frequencies for the white whale and dolphin lungs were 30 and 36 Hz, respectively. These values were significantly higher than those predicted using a free-spherical air bubble model. Experimentally measured damping ratios and quality factors at resonance were 0.20 and 2.5, respectively, for the white whale, and 0.16 and 3.1, respectively, for the dolphin.
The design of bells with harmonic overtones
Authors: McLachlan Neil , Nigjeh Behzad Keramati , Hasell Anton
Published: July 2003

Musical bells have had limited application due to the presence of inharmonic partials in the lower part of their acoustic spectra. A series of bells has been designed that contains up to seven partial frequencies in the harmonic series beginning at the fundamental frequency. This was achieved by choosing geometries for finite-element analysis models in which as many purely circumferential bending modes as possible occurred at frequencies below any mode with an axial ring node. The bell models were then fine tuned using gradient projection method shape optimization and the resulting profiles were cast in silicon bronze. A range of bell geometries and timbres is analyzed using psycho-acoustic models and is discussed in relation to European carillon bells.
Mammalian spontaneous otoacoustic emissions are amplitude-stabilized cochlear standing waves
Authors: Shera Christopher A.
Published: July 2003

Mammalian spontaneous otoacoustic emissions (SOAEs) have been suggested to arise by three different mechanisms. The local-oscillator model, dating back to the work of Thomas Gold, supposes that SOAEs arise through the local, autonomous oscillation of some cellular constituent of the organ of Corti (e.g., the “active process” underlying the cochlear amplifier). Two other models, by contrast, both suppose that SOAEs are a global collective phenomenon—cochlear standing waves created by multiple internal reflection—but differ on the nature of the proposed power source: Whereas the “passive” standing-wave model supposes that SOAEs are biological noise, passively amplified by cochlear standing-wave resonances acting as narrow-band nonlinear filters, the “active” standing-wave model supposes that standing-wave amplitudes are actively maintained by coherent wave amplification within the cochlea. Quantitative tests of key predictions that distinguish the local-oscillator and global standing-wave models are presented and shown to support the global standing-wave model. In addition to predicting the existence of multiple emissions with a characteristic minimum frequency spacing, the global standing-wave model accurately predicts the mean value of this spacing, its standard deviation, and its power-law dependence on SOAE frequency. Furthermore, the global standing-wave model accounts for the magnitude, sign, and frequency dependence of changes in SOAE frequency that result from modulations in middle-ear stiffness. Although some of these SOAE characteristics may be replicable through artful ad hoc adjustment of local-oscillator models, they all arise quite naturally in the standing-wave framework. Finally, the statistics of SOAE time waveforms demonstrate that SOAEs are coherent, amplitude-stabilized signals, as predicted by the active standing-wave model. Taken together, the results imply that SOAEs are amplitude-stabilized standing waves produced by the cochlea acting as a biological, hydromechanical analog of a laser oscillator. Contrary to recent claims, spontaneous emission of sound from the ear does not require the autonomous mechanical oscillation of its cellular constituents.
Distortion product otoacoustic emission suppression tuning curves in normal-hearing and hearing-impaired human ears
Authors: Gorga Michael P. , Neely Stephen T. , Dierking Darcia M. , Dorn Patricia A. , Hoover Brenda M. , Fitzpatrick Denis F.
Published: July 2003

Distortion product otoacoustic emission (DPOAE) suppression measurements were made in 20 subjects with normal hearing and 21 subjects with mild-to-moderate hearing loss. The probe consisted of two primary tones (f2,f1), with f2 held constant at 4 kHz and f2/f1=1.22. Primary levels (L1,L2) were set according to the equation L1=0.4L2+39 dB [Kummer et al., J. Acoust. Soc. Am. 103, 3431–3444 (1998)], with L2 ranging from 20 to 70 dB SPL (normal-hearing subjects) and 50–70 dB SPL (subjects with hearing loss). Responses elicited by the probe were suppressed by a third tone (f3), varying in frequency from 1 octave below to 12 octave above f2. Suppressor level (L3) varied from 5 to 85 dB SPL. Responses in the presence of the suppressor were subtracted from the unsuppressed condition in order to convert the data into decrements (amount of suppression). The slopes of the decrement versus L3 functions were less steep for lower frequency suppressors and more steep for higher frequency suppressors in impaired ears. Suppression tuning curves, constructed by selecting the L3 that resulted in 3 dB of suppression as a function of f3, resulted in tuning curves that were similar in appearance for normal and impaired ears. Although variable, Q10 and QERB were slightly larger in impaired ears regardless of whether the comparisons were made at equivalent SPL or equivalent sensation levels (SL). Larger tip-to-tail differences were observed in ears with normal hearing when compared at either the same SPL or the same SL, with a much larger effect at similar SL. These results are consistent with the view that subjects with normal hearing and mild-to-moderate hearing loss have similar tuning around a frequency for which the hearing loss exists, but reduced cochlear-amplifier gain.
Modeling vocal fold motion with a hydrodynamic semicontinuum model
Authors: LaMar M. Drew , Qi Yingyong , Xin Jack
Published: July 2003

Vocal fold (VF) motion is a fundamental process in voice production, and is also a challenging problem for numerical computation because the VF dynamics depend on nonlinear coupling of air flow with the response of elastic channels (VF), which undergo opening and closing, and induce internal flow separation. The traditional modeling approach makes use of quasisteady flow approximation or Bernoulli’s law which ignores air compressibility, and is known to be invalid during VF opening. A hydrodynamic semicontinuum system for VF motion is presented. The airflow is modeled by a modified quasi-one-dimensional Euler system with coupling to VF velocity. The VF is modeled by a lumped two mass system with a built-in geometric condition on flow separation. The modified Euler system contains the Bernoulli’s law as a special case, and is derivable from the two-dimensional compressible Navier–Stokes equations in the inviscid limit. The computational domain contains also solid walls next to VFs (flexible walls). It is shown numerically that several salient features of VFs are captured, especially transients such as the double peaks of the driving subglottal pressures at the opening and the closing stages of VF motion consistent with fully resolved two-dimensional direct simulations, and experimental data. The system is much simpler to compute than a VF model based on two-dimensional Navier–Stokes system.
An approximate Green’s function for a locally excited fluid-loaded thin elastic plate
Authors: DiPerna Daniel T. , Feit David
Published: July 2003

In the classic treatment of the line-driven, fluid-loaded, thin elastic plate, a branch cut integral typically needs to be evaluated. This branch cut arises due to a square root operator in the spectral form of the acoustic impedance. In a previous paper [J. Acoust. Soc. Am. 110, 3018 (2001)], DiPerna and Feit developed a methodology, complex layer analysis (CLA), to approximate this impedance. The resulting approximation was in the form of a rational function, although this was not explicitly stated. In this paper, a rational function approximation (RFA) to the acoustic impedance is derived. The advantage of the RFA as compared to the CLA approach is that a smaller number of terms are required. The accuracy of the RFA is examined both in the Fourier transform domain and the spatial domain. The RFA is then used to obtain a differential relationship between the pressure and velocity on the surface of the plate. Finally, using the RFA in conjunction with the equation of motion of the plate, an approximate expression for the Green’s function for a line-driven plate is obtained in terms of a sum of propagating and evanescent waves. Comparisons of these results with the numerical inversion of the exact integral show reasonable agreement.
Theory of the Greenspan viscometer
Authors: Gillis Keith A. , Mehl James B. , Moldover Michael R.
Published: July 2003

We present a detailed acoustic model of the Greenspan acoustic viscometer, a practical instrument for accurately measuring the viscosity η of gases. As conceived by Greenspan, the viscometer is a Helmholtz resonator composed of two chambers coupled by a duct of radius rd. In the lowest order, η=πfρ(rd/Q)2, where f and Q are the frequency and quality factor of the isolated Greenspan mode, and ρ is the gas density. In this level of approximation, the viscosity can be determined by measuring the duct radius and frequency response of the resonator. In the full acoustic model of the resonator, the duct is represented by a T-equivalent circuit, the chambers as lumped impedances, and the effects of the diverging fields at the duct ends by lumped end impedances with inertial and resistive components. The model accounts for contributions to 1/Q from thermal dissipation (primarily localized in the chambers) and from a capillary used for filling and evacuating the resonator. A robust, prototype instrument is being used for measuring the viscosity of reactive gases used in semiconductor processing. For well-characterized surrogate gases, the prototype viscometer generated values of η that were within ±0.8% of published reference values throughout the pressure range 0.2–3.2 MPa. Remarkably, we achieved this level of agreement by only slight adjustment of the numerically calculated inertial and resistive end effect parameters to improve the agreement with helium reference values. No other parameters were adjusted.
Perceived naturalness of spectrally distorted speech and music
Authors: Moore Brian C. J. , Tan Chin-Tuan
Published: July 2003

We determined how the perceived naturalness of music and speech (male and female talkers) signals was affected by various forms of linear filtering, some of which were intended to mimic the spectral “distortions” introduced by transducers such as microphones, loudspeakers, and earphones. The filters introduced spectral tilts and ripples of various types, variations in upper and lower cutoff frequency, and combinations of these. All of the differently filtered signals (168 conditions) were intermixed in random order within one block of trials. Levels were adjusted to give approximately equal loudness in all conditions. Listeners were required to judge the perceptual quality (naturalness) of the filtered signals on a scale from 1 to 10. For spectral ripples, perceived quality decreased with increasing ripple density up to 0.2 ripple/ERBN and with increasing ripple depth. Spectral tilts also degraded quality, and the effects were similar for positive and negative tilts. Ripples and/or tilts degraded quality more when they extended over a wide frequency range (87–6981 Hz) than when they extended over subranges. Low- and mid-frequency ranges were roughly equally important for music, but the mid-range was most important for speech. For music, the highest quality was obtained for the broadband signal (55–16 854 Hz). Increasing the lower cutoff frequency from 55 Hz resulted in a clear degradation of quality. There was also a distinct degradation as the upper cutoff frequency was decreased from 16 845 Hz. For speech, there was a marked degradation when the lower cutoff frequency was increased from 123 to 208 Hz and when the upper cutoff frequency was decreased from 10 869 Hz. Typical telephone bandwidth (313 to 3547 Hz) gave very poor quality.
Suppression tuning in noise-exposed rabbits
Authors: Howard MacKenzie A. , Stagner Barden B. , Foster Paul K. , Lonsbury-Martin Brenda L. , Martin Glen K.
Published: July 2003

Psychophysical, basilar-membrane (BM), and single nerve-fiber tuning curves, as well as suppression of distortion-product otoacoustic emissions (DPOAEs), all give rise to frequency tuning patterns with stereotypical features. Similarities and differences between the behaviors of these tuning functions, both in normal conditions and following various cochlear insults, have been documented. While neural tuning curves (NTCs) and BM tuning curves behave similarly both before and after cochlear insults known to disrupt frequency selectivity, DPOAE suppression tuning curves (STCs) do not necessarily mirror these responses following either administration of ototoxins [Martin et al., J. Acoust. Soc. Am. 104, 972–983 (1998)] or exposure to temporarily damaging noise [Howard et al., J. Acoust. Soc. Am. 111, 285–296 (2002)]. However, changes in STC parameters may be predictive of other changes in cochlear function such as cochlear immaturity in neonatal humans [Abdala, Hear. Res. 121, 125–138 (1998)]. To determine the effects of noise-induced permanent auditory dysfunction on STC parameters, rabbits were exposed to high-level noise that led to permanent reductions in DPOAE level, and comparisons between pre- and postexposure DPOAE levels and STCs were made. Statistical comparisons of pre- and postexposure STC values at CF revealed consistent basal shifts in the frequency region of greatest cochlear damage, whereas thresholds, Q10 dB, and tip-to-tail gain values were not reliably altered. Additionally, a large percentage of high-frequency lobes associated with third tone interference phenomena, that were exhibited in some data sets, were dramatically reduced following noise exposure. Thus, previously described areas of DPOAE interference above f2 may also be studied using this type of experimental manipulation [Martin et al., Hear. Res. 136, 105–123 (1999); Mills, J. Acoust. Soc. Am. 107, 2586–2602 (2002)].
The effect of speech presentation level on measurement of auditory acclimatization to amplified speech
Authors: Munro Kevin J. , Lutman Mark E.
Published: July 2003

A systematic improvement in auditory performance over time, following a change in the acoustic information available to the listener (that cannot be attributed to task, procedural or training effects) is known as auditory acclimatization. However, there is conflicting evidence concerning the existence of auditory acclimatization; some studies show an improvement in performance over time while other studies show no change. In an attempt to resolve this conflict, speech recognition abilities of 16 subjects with bilateral sensorineural hearing impairments were measured over a 12-week period following provision of a monaural hearing instrument for the first time. The not-fitted ear was used as the control. Three presentation levels were used representing quiet, normal, and raised speech. The results confirm the presence of acclimatization. In addition, the results show that acclimatization is evident at the higher presentation levels but not at the lowest.
An account of monaural phase sensitivity
Authors: Carlyon Robert P. , Shamma Shihab
Published: July 2003

Listeners can detect phase differences between the envelopes of sounds occupying remote frequency regions, and between the fine structures of partials that interact within a single auditory filter. They are insensitive to phase differences between partials that differ sufficiently in frequency to preclude within-channel interactions. A new model is proposed that can account for all three of these findings, and which, unlike currently popular approaches, does not discard across-channel timing information. Sensitivity is predicted quantitatively by analyzing the output of a cochlear model using a spectro-temporal decomposition inspired by responses of neurons in the auditory cortex, and by computing a distance metric between the responses to two stimuli to be discriminated. Discriminations successfully modeled include phase differences between pairs of bandpass filtered harmonic complexes, and between pairs of sinusoidally amplitude modulated tones, discrimination between amplitude and frequency modulation, and discrimination of transient signals differing only in their phase spectra (“Huffman sequences”).
Delayed models for simplified musical instruments
Authors: Barjau Ana , Gibiat Vincent
Published: July 2003

Most musical instruments contain, at their very basis, a continuous vibrating element (string or air column) which can be treated as a one-dimensional system. Its oscillation is obtained either through an initial condition or by means of a continuous energy input through a nonlinear device. In both cases and as a first approach, the excitation can be localized at one single point, and the continuous system can be considered as a linear one. The coupling between these two elements is often represented through a convolution integral. This convolution will be rewritten here in a way that different phenomena taking place in the continuous element (internal losses, radiation at the ends…) are separated. Different choices in the formulation of these processes and some mathematical manipulation will lead to either algebraic iterative or delayed differential equations. These equations are valid for any form of energy input. Once this energy input is defined, they can be used to simulate the behavior of different instruments in a more efficient way than that of traditional convolution. Moreover, these equations allow an analytical analysis of possible regimes using the tools of nonlinear dynamical systems (NLDS). The case of woodwinds will be emphasized throughout the paper, while that of strings will be presented briefly for the sake of completeness.
Reciprocity and the active control of distributed acoustic sources (L)
Authors: Elliott Stephen J. , Rees Lewis
Published: July 2003

The velocity distributions of a pair of arbitrary acoustic sources are approximated by arrays of elementary sources, and a reciprocal relationship is established between the generalized force on one source and the total velocity of the other source. When the two distributed sources are individually constrained to vibrate with a single phase and are used for active sound control, it is shown that the attenuation in sound power from the pair of sources is the same regardless of whether one source velocity is fixed and the velocity of the other source is adjusted to minimize the power output, or vice versa.
Production benefits of childhood overhearing
Authors: Knightly Leah M. , Jun Sun-Ah , Oh Janet S. , Au Terry Kit-fong
Published: July 2003

The current study assessed whether overhearing Spanish during childhood helps later Spanish pronunciation in adulthood. Our preliminary report based on a subset of the data [Au et al., Psychol. Sci. 13, 238–243 (2002)] revealed that adults who overheard Spanish during childhood had better Spanish pronunciation, but not better morphosyntax, than adult learners of Spanish who had no childhood experience with Spanish. We now present data from the full sample with additional morphosyntax and pronunciation assessments, as well as measures to help rule out possible confounding prosodic factors such as speech rate, phrasing, and stress placement. Three groups of undergraduates were compared: 15 Spanish–English bilinguals (native Spanish speakers), 15 late learners of Spanish who overheard Spanish during childhood (childhood overhearers), 15 late learners of Spanish who had no regular experience with Spanish until middle or high school (typical late L2 learners). Results confirmed a pronunciation advantage for the childhood overhearers over the typical late L2 learners on all measures: phonetic analyses (VOT and degree of lenition), accent ratings (phoneme and story production), but no benefit in morphosyntax. Importantly, the pronunciation advantage did not seem attributable to prosodic factors. These findings illustrate the specificity of overhearers’ advantage to phonological production.
Acoustic communication in two freshwater gobies: Ambient noise and short-range propagation in shallow streams
Authors: Lugli M. , Fine M. L.
Published: July 2003

Noise is an important theoretical constraint on the evolution of signal form and sensory performance. In order to determine environmental constraints on the communication of two freshwater gobies Padogobius martensii and Gobius nigricans, numerous noise spectra were measured from quiet areas and ones adjacent to waterfalls and rapids in two shallow stony streams. Propagation of goby sounds and waterfall noise was also measured. A quiet window around 100 Hz is present in many noise spectra from noisy locations. The window lies between two noise sources, a low-frequency one attributed to turbulence, and a high-frequency one (200–500 Hz) attributed to bubble noise from water breaking the surface. Ambient noise from a waterfall (frequencies below 1 kHz) attenuates as much as 30 dB between 1 and 2 m, after which values are variable without further attenuation (i.e., buried in the noise floor). Similarly, courtship sounds of P. martensii attenuate as much as 30 dB between 5 and 50 cm. Since gobies are known to court in noisy as well as quiet locations in these streams, their acoustic communication system (sounds and auditory system) must be able to cope with short-range propagation dictated by shallow depths and ambient noise in noisy locations.
Informational masking: Counteracting the effects of stimulus uncertainty by decreasing target-masker similarity
Authors: Durlach Nathaniel I. , Mason Christine R. , Shinn-Cunningham Barbara G. , Arbogast Tanya L. , Colburn H. Steven , Kidd Gerald
Published: July 2003

Previous work has indicated that target-masker similarity, as well as stimulus uncertainty, influences the amount of informational masking that occurs in detection, discrimination, and recognition tasks. In each of five experiments reported in this paper, the detection threshold for a tonal target in random multitone maskers presented simultaneously with the target tone was measured for two conditions using the same set of five listeners. In one condition, the target was constructed to be “similar” (S) to the masker; in the other condition, it was constructed to be “dissimilar” (D) to the masker. The specific masker varied across experiments, but was constant for the two conditions. Target-masker similarity varied in dimensions such as duration, perceived location, direction of frequency glide, and spectro-temporal coherence. Group-mean results show large decreases in the amount of masking for the D condition relative to the S condition. In addition, individual differences (a hallmark of informational masking) are found to be much greater in the S condition than in the D condition. Furthermore, listener vulnerability to informational masking is found to be consistent to at least a moderate degree across experiments.
Precedence-effect thresholds for a population of untrained listeners as a function of stimulus intensity and interclick interval
Authors: Saberi Kourosh , Antonio Joseph V.
Published: July 2003

Data are reported from 127 untrained individuals under lag- and single-click conditions in a precedence-effect task. In experiment I, each subject completed ten runs in a two-interval forced-choice design under a lag-click condition and three runs under a single-click condition. The cue to be discriminated was an interaural time difference (ITD). Stimuli were 125-μs rectangular pulses and the interclick interval (ICI) was 2 ms. Subjects were randomly assigned to three groups of approximately 30. Each group was tested at one stimulus intensity (43, 58, or 73 dB). Mean threshold within each group was greater than 500 μs for lag-click ITD conditions, although substantial intersubject variability and a clear effect of stimulus intensity on lag-click ITD thresholds were observed, with lower thresholds for higher intensities. In experiment II, the ICI was varied from 0.3 to 10 ms, and thresholds were obtained from groups of approximately 20 untrained subjects. Data were also collected from three highly experienced observers as a function of ICI. The best naı̈ve subject produced mean thresholds near, but not as low as those obtained from experienced subjects. Analysis of adaptive-track patterns revealed abrupt irregularities in threshold tracking, consistent with either losing the cue or listening to the wrong cue in an ambiguous stimulus.
Applications of laser-based ultrasonics to the characterization of the internal structure of teeth
Authors: Blodgett D. W.
Published: July 2003

Dental health care and research workers require a means of imaging the structures within teeth in vivo. For example, there is a need to image the margins of a restoration for the detection of poor bonding or voids between the restorative material and the dentin. In addition, a high-resolution imaging modality is needed to detect tooth decay in its early stages. If decay can be detected early enough, the process can be monitored and interventional procedures, such as fluoride washes and controlled diet, can be initiated to help remineralize the tooth. Currently employed x-ray imaging is limited in its ability to visualize interfaces and incapable of detecting decay at a stage early enough to avoid invasive cavity preparation followed by a restoration. To this end, nondestructive and noncontact in vitro measurements on “as-is” extracted sections of human incisors and molars using laser-based ultrasonics are presented. Broadband ultrasonic waves are excited in the extracted sections by using a pulsed carbon-dioxide (CO2) laser operating in a region of high optical absorption in the dental hard tissues. Optical interferometric detection of the ultrasonic wave surface displacements is accomplished with a path-stabilized Michelson-type interferometer. Laser ultrasonics is found effective in characterizing the anisotropic and inhomogeneous nature of dentin. In addition, time-of-flight analysis of the measured bulk transmission waveforms allows for detection of dentino-enamel and carious dentin-dentin junctions. These results are compared to those obtained for specially prepared tooth phantoms that mimic the mechanical properties of dental hard tissues.
The shock formation distance in a bounded sound beam of finite amplitude
Authors: Tao Chao , Ma Jian , Zhu Zhemin , Du Gonghuan , Ping Zihong
Published: July 2003

This paper investigates the shock formation distance in a bounded sound beam of finite amplitude by solving the Khokhlov-Zabolotskaya-Kuznetsov (KZK) equation using frequency-domain numerical method. Simulation results reveal that, besides the nonlinearity and absorption, the diffraction is another important factor that affects the shock formation of a bounded sound beam. More detailed discussions of the shock formation in a bounded sound beam, such as the waveform of sound pressure and the spatial distribution of shock formation, are also presented and compared for different parameters.
The effects of a high-frequency suppressor on tuning curves and derived basilar-membrane response functions
Authors: Yasin Ifat , Plack Christopher J.
Published: July 2003

Forward-masked psychophysical tuning curves were obtained using a fixed, low-level signal at a frequency of 4 kHz, and masker frequencies of 2.0, 2.5, 3.0, 3.5, 3.75, 4.0, 4.25, 4.5, 4.75, 5.0, and 5.5 kHz, at masker–signal gaps of 20, 30, 40, 60, 80, and 100 ms. An adaptive two-interval, two alternative forced-choice (2I-2AFC) procedure was used to obtain the masker level at threshold. This procedure was repeated with the addition of a 4.75-kHz suppressor at 50 or 60 dB SPL, gated with the masker. Tuning curves were broader, and estimates of compression and gain from derived input/output functions were decreased in the presence of a suppressor as compared to the no-suppressor condition. The results are consistent with physiological results, which show that suppression leads to a broadening of tuning curves and a partial linearization of the midlevel portion of the basilar-membrane input/output function.
Acoustic characteristics of an electrodynamic planar digital loudspeaker
Authors: Furihata Kenji , Hayama Atsushi , Asano David K. , Yanagisawa Takesaburo
Published: July 2003

In this paper, an electrodynamic planar loudspeaker driven by a digital signal is experimentally discussed. The digital loudspeaker consists of 22 voice coils, 11 permanent magnets, a diaphragm with streamlined sections molded in plastic, and a suspension made of handmade Japanese paper between the diaphragm and the frame. First, the acoustic responses are affected by the arrangement of the voice coils, so an asymmetric arrangement is studied. This asymmetric arrangement is designed to obtain as flat a frequency response to an analog signal as possible. This arrangement is compared with a symmetric one and results show that the flatness of the frequency response around 1 kHz and 4 kHz is improved and that the sound reproduction band is from 40 Hz to 10 kHz. Second, to evaluate the acoustic responses to a digital signal, the digital loudspeaker is driven with a pulse code modulation signal. Results show that the digital loudspeaker can reproduce pure sound with a total harmonic distortion of less than 5% from 40 Hz to 10 kHz, exceeding this value only in a narrow frequency band near 4 kHz. This digital loudspeaker was demonstrated to have good linearity over its dynamic range of 84 dB.
Improving robustness of active noise control in ducts
Authors: Yuan Jing
Published: July 2003

A robust active noise controller (ANC) is proposed here for finite ducts. While the H∞ control theory provides theoretical ground and numerical algorithms to design robust controllers, it is important for an engineer to design and formulate a robust controller so that the objective is more achievable and the H∞ constraints less restrictive without sacrificing robustness. A new robust ANC is designed this way with an extra actuator to improve achievable performance and introduce more degrees of freedom to controller parameters. The new strategy relaxes H∞ constraints without sacrificing robustness and enables the ANC to tolerate a wide variety of errors and uncertainties including truncation errors between a finite model and an infinite field. Theoretical analysis, numerical examples, and experimental results are presented to demonstrate the improved performance of the proposed ANC when subject to a certain level of uncertainties in a duct.
Examination of a linear model in an informational masking study
Authors: Tang Zhongzhou , Richards Virginia M.
Published: July 2003

When multitone maskers are used in a two-interval, forced choice experiment, the amount of masking is larger when the masker is randomly chosen on each presentation interval compared to on each trial (the same masker in the two listening intervals). These conditions are referred to as having within- versus between-trial randomization. If it is assumed that an observer’s ultimate detection decision depends on a single decision variable (DV), it is probable that the DV’s variance will be substantially larger in the within-trial randomization condition compared to the between-trial randomization condition. The goal of the current experiment is to evaluate the degree to which this stimulus-based change in DV variance can account for the difference in thresholds in the within- versus between-trial randomization conditions. Thresholds are measured for the detection of a tone added to a six-component masker in between- and within-trial randomization conditions. The slopes of the psychometric functions provide an estimate of the variance in the DV for the between- and within-trial randomization conditions. Additionally, a channel model is fitted to the psychophysical results in the within-trial randomization condition. The resulting model is then used to predict the value of the DV for each trial, and ultimately to estimate the proportion of the total variance in the within-trial randomization condition that is attributable to changes in maskers across intervals. The variance of the DV in the between-trial randomization condition accounted for approximately 65% of the total variance in the DV in the within-trial randomization condition. Stimulus-based interval-by-interval masker randomization accounted for approximately 20% of the total variance of the within-trial randomization DV. The remaining 15% of the DV variance in the within-trial randomization condition remained unaccounted for. This result is fairly stable whether the maskers are drawn from a small versus large pool of potential maskers.
Transient elastography in anisotropic medium: Application to the measurement of slow and fast shear wave speeds in muscles
Authors: Gennisson Jean-Luc , Catheline Stefan , Chaffaı̈ Sana , Fink Mathias
Published: July 2003

From the measurement of a low frequency (50–150 Hz) shear wave speed, transient elastography evaluates the Young’s modulus in isotropic soft tissues. In this paper, it is shown that a rod source can generate a low frequency polarized shear strain waves. Consequently this technique allows to study anisotropic medium such as muscle. The evidence of the polarization of low frequency shear strain waves is supported by both numeric simulations and experiments. The numeric simulations are based on theoretical Green’s functions in isotropic and anisotropic media (hexagonal system). The experiments in vitro led on beef muscle proves the pertinent of this simple anisotropic pattern. Results in vivo on man biceps shows the existence of slow and fast shear waves as predicted by theory.
Effects of simulated cochlear-implant processing on speech reception in fluctuating maskers
Authors: Qin Michael K. , Oxenham Andrew J.
Published: July 2003

This study investigated the effects of simulated cochlear-implant processing on speech reception in a variety of complex masking situations. Speech recognition was measured as a function of target-to-masker ratio, processing condition (4, 8, 24 channels, and unprocessed) and masker type (speech-shaped noise, amplitude-modulated speech-shaped noise, single male talker, and single female talker). The results showed that simulated implant processing was more detrimental to speech reception in fluctuating interference than in steady-state noise. Performance in the 24-channel processing condition was substantially poorer than in the unprocessed condition, despite the comparable representation of the spectral envelope. The detrimental effects of simulated implant processing in fluctuating maskers, even with large numbers of channels, may be due to the reduction in the pitch cues used in sound source segregation, which are normally carried by the peripherally resolved low-frequency harmonics and the temporal fine structure. The results suggest that using steady-state noise to test speech intelligibility may underestimate the difficulties experienced by cochlear-implant users in fluctuating acoustic backgrounds.
Basilar-membrane responses to multicomponent (Schroeder-phase) signals: Understanding intensity effects
Authors: Summers Van , de Boer Egbert , Nuttall Alfred L.
Published: July 2003

Harmonic complexes comprised of the same spectral components in either positive-Schroeder (+Schr) or negative-Schroeder (−Schr) phase [see Schroeder, IEEE Trans. Inf. Theory 16, 85–89 (1970)] have identical long-term spectra and similar waveform envelopes. However, localized patterns of basilar-membrane (BM) excitation can be quite different in response to these two stimuli. Measurements in chinchillas showed more modulated (peakier) BM excitation for +Schr than −Schr complexes [Recio and Rhode, J. Acoust. Soc. Am. 108, 2281–2298 (2000)]. In the current study, laser velocimetry was used to examine BM responses at a location tuned to approximately 17 kHz in the basal turn of the guinea-pig cochlea, for +Schr and −Schr complexes with a 203-Hz fundamental frequency and including 101 equal-amplitude components from 2031 to 22 344 Hz. At 35-dB SPL, +Schr response waveforms showed greater amplitude modulation than −Schr responses. With increasing stimulation level, internal modulation decreased for both complexes. To understand the observed phenomena quantitatively, responses were predicted on the basis of a linearized model of the cochlea. Prediction was based on an “indirect impulse response” measured in the same animal. Response waveforms for Schroeder-phase signals were accurately predicted, provided that the level of the indirect impulse used in prediction closely matched the level of the Schroeder-phase stimulus. This result confirms that the underlying model, which originally was developed for noise stimuli, is valid for stimuli that produce completely different response waveforms. Moreover, it justifies explanation of cochlear filtering (i.e., differential treatment of different frequencies) in terms of a linear system.
Time-domain pressure response of arrays with periodic excitation (L)
Authors: Ketterling Jeffrey A. , Lizzi Frederic L.
Published: July 2003

A time-domain method utilizing the spatial impulse response is proposed to compute beam patterns for array transducers where each array element may have a unique continuously periodic (CP) excitation. The method is useful for cases when an efficient time-domain approach is desired for the steady-state and transient results. The pressures for each array element at a point P in the sound field are computed over the lowest period T and an appropriate time shift is applied. The resulting pressures are summed to give the total pressure at P.
Vertical-plane sound localization probed with ripple-spectrum noise
Authors: Macpherson Ewan A. , Middlebrooks John C.
Published: July 2003

Ripple-spectrum stimuli were used to investigate the scale of spectral detail used by listeners in interpreting spectral cues for vertical-plane localization. In three experiments, free-field localization judgments were obtained for 250-ms, 0.6–16-kHz noise bursts with log-ripple spectra that varied in ripple density, peak-to-trough depth, and phase. When ripple density was varied and depth was held constant at 40 dB, listeners’ localization error rates increased most (relative to rates for flat-spectrum targets) for densities of 0.5–2 ripples/oct. When depth was varied and density was held constant at 1 ripple/oct, localization accuracy was degraded only for ripple depths ⩾20 dB. When phase was varied and density was held constant at 1 ripple/oct and depth at 40 dB, three of five listeners made errors at consistent locations unrelated to the ripple phase, whereas two listeners made errors at locations systematically modulated by ripple phase. Although the reported upper limit for ripple discrimination is 10 ripples/oct [Supin et al., J. Acoust. Soc. Am. 106, 2800–2804 (1999)], present results indicate that details finer than 2 ripples/oct or coarser than 0.5 ripples/oct do not strongly influence processing of spectral cues for sound localization. The low spectral-frequency limit suggests that broad-scale spectral variation is discounted, even though components at this scale are among those contributing the most to the shapes of directional transfer functions.
Bifurcation of the Biot slow wave in a porous medium
Authors: Edelman Inna
Published: July 2003

The propagation of the Biot slow wave in a fluid-saturated porous medium at low frequencies is investigated by asymptotic methods. It is proven that the Biot wave has a bifurcation behavior depending on its wave number. The bifurcation occurs in a neighborhood of the critical value kcr, which depends on the permeability of a medium and the viscosity of a fluid. The P2 wave is fully attenuated if its wave number is smaller than kcr and it becomes propagatory with wave numbers bigger than kcr. Asymptotic formulas for the phase velocity and attenuation of the Biot wave are derived.
Ray stability in weakly range-dependent sound channels
Authors: Beron-Vera F. J. , Brown M. G.
Published: July 2003

Ray stability is investigated in environments consisting of a range-independent background sound-speed profile on which a range-dependent perturbation is superimposed. Theoretical arguments suggest and numerical results confirm that ray stability is strongly influenced by the background sound speed profile. Ray instability is shown to increase with increasing magnitude of α(I)=(I/ω)dω/dI, where 2π/ω(I) is the range of a ray double loop and I is the ray action variable. This behavior is illustrated using internal-wave-induced scattering in deep ocean environments and rough surface scattering in upward refracting environments.
Scattering of acoustic waves from immersed transversely isotropic cylinders (L)
Authors: Honarvar Farhang , Fan Ying , Sinclair Anthony N.
Published: July 2003

A mathematical model for the scattering of acoustic waves from immersed transversely isotropic cylinders was developed by the authors [J. Acoust. Soc. Am. 100, 57 (1996)]. Ahmad and Rahman further elaborated on the mathematical aspects of this model and based on mathematical discussions showed that there exist two distinct types of transversely isotropic materials, which they named type I and type II [Int. J. Eng. Sci. 38, 325 (2000)]. They also described the expected behavior of the form functions of such materials. The purpose of this article is to distinguish the physical differences between these two types of transversely isotropic materials.
The effects of the amplitude distribution of equal energy exposures on noise-induced hearing loss: The kurtosis metric
Authors: Hamernik Roger P. , Qiu Wei , Davis Bob
Published: July 2003

Seventeen groups of chinchillas with 11 to 16 animals/group (∑N=207) were exposed for 5 days to either a Gaussian (G) noise or 1 of 16 different non-Gaussian (non-G) noises at 100 dB(A) SPL. All exposures had the same total energy and approximately the same flat spectrum but their statistical properties were varied to yield a series of exposure conditions that varied across a continuum from G through various non-G conditions to pure impact noise exposures. The non-G character of the noise was produced by inserting high level transients (impacts or noise bursts) into the otherwise G noise. The peak SPL of the transients, their bandwidth, and the intertransient intervals were varied, as was the rms level of the G noise. The statistical metric, kurtosis (β), computed on the unfiltered noise β(t), was varied 3⩽β(t)⩽105. Brainstem auditory evoked responses were used to estimate hearing thresholds and surface preparation histology was used to determine sensory cell loss. Trauma, as measured by asymptotic and permanent threshold shifts (ATS, PTS) and by sensory cell loss, was greater for all of the non-G exposure conditions. Permanent effects of the exposures increased as β(t) increased and reached an asymptote at β(t)∼40. For β(t)>40 varying the interval or peak histograms did not alter the level of trauma, suggesting that, in the chinchilla model, for β(t)>40 an energy metric may be effective in evaluating the potential of non-G noise environments to produce hearing loss. Reducing the probability of a transient occurring could reduce the permanent effects of the non-G exposures. These results lend support to those standards documents that use an energy metric for gauging the hazard of exposure but only after applying a “correction factor” when high level transients are present. Computing β on the filtered noise signal [β(f)] provides a frequency specific metric for the non-G noises that is correlated with the additional frequency specific outer hair cell loss produced by the non-G noise. The data from the abundant and varied exposure conditions show that the kurtosis of the amplitude distribution of a noise environment is an important variable in determining the hazards to hearing posed by non-Gaussian noise environments.
Asymmetry of masking between complex tones and noise: Partial loudness
Authors: Gockel Hedwig , Moore Brian C. J. , Patterson Roy D.
Published: July 2003

This experiment examined the partial masking of periodic complex tones by a background of noise, and vice versa. The tones had a fundamental frequency (F0) of 62.5 or 250 Hz, and components were added in either cosine phase (CPH) or random phase (RPH). The tones and the noise were bandpass filtered into the same frequency region, from the tenth harmonic up to 5 kHz. The target alone was alternated with the target and the background; for the mixture, the background and target were either gated together, or the background was turned on 400 ms before, and off 200 ms after, the target. Subjects had to adjust the level of either the target alone or the target in the background so as to match the loudness of the target in the two intervals. The overall level of the background was 50 dB SPL, and loudness matches were obtained for several fixed levels of the target alone or in the background. The resulting loudness-matching functions showed clear asymmetry of partial masking. For a given target-to-background ratio, the partial loudness of a complex tone in a noise background was lower than the partial loudness of a noise in a complex tone background. Expressed as the target-to-background ratio required to achieve a given loudness, the asymmetry typically amounted to 12–16 dB. When the F0 of the complex tone was 62.5 Hz, the asymmetry of partial masking was greater for CPH than for RPH. When the F0 was 250 Hz, the asymmetry was greater for RPH than for CPH. Masked thresholds showed the same pattern as for partial masking for both F0’s. Onset asynchrony had some effect on the loudness matching data when the target was just above its masked threshold, but did not significantly affect the level at which the target in the background reached its unmasked loudness. The results are interpreted in terms of the temporal structure of the stimuli.