mesh refinement processes based on the generalized bisection of simplices mesh refinement processes based on the generalized bisection of simplices are discussed and characterized in the context of the general theory of rheinboldt siam j numer anal 17 1980 pp 766 778 then it is proved that such processes allow the construction of sequences of naturally smooth conforming and nested nonuniform meshes in fact it is shown that there exists a generalized mesh refinement operator that for any conforming triangulation delta and for any v subset delta produces a nested smooth conforming triangulation delta containing successors of all elements of v and such that the minimum cell size of delta is bounded from below by half the minimum cell size of delta two conforming mesh refinement algorithms that allow the selective refinement of computational triangulations are explicitly given

factors affecting the use of noise band vocoders as acoustic models for pitch perception in cochlear implants although in a number of experiments noise band vocoders have been shown to provide acoustic models for speech perception in cochlear implants ci the present study assesses in four experiments whether and under what limitations noise band vocoders can be used as an acoustic model for pitch perception in ci the first two experiments examine the effect of spectral smearing on simulated electrode discrimination and fundamental frequency f0 discrimination the third experiment assesses the effect of spectral mismatch in an f0 discrimination task with two different vocoders the fourth experiment investigates the effect of amplitude compression on modulation rate discrimination for each experiment the results obtained from normal hearing subjects presented with vocoded stimuli are compared to results obtained directly from ci recipients the results show that place pitch sensitivity drops with increased spectral smearing and that place pitch cues for multi channel stimuli can adequately be mimicked when

factors influencing electrical place pitch perception in bimodal listeners factors that might affect perceptual pitch match between acoustic and electric stimulation were examined in 25 bimodal listeners using magnitude estimation pre operative acoustic thresholds in both ears and duration of severe profound loss were first examined as correlates with degree of match between the measured pitch and that predicted by the spiral ganglion frequency position model the degree of match was examined with respect to 1 the ratio between the measured and predicted pitch percept on the most apical electrode and 2 the ratio between the slope of the measured and predicted pitch function second effect of listening experience was examined to assess whether adaptation occurred over time to match the frequency assignment to electrodes pre experience pitch estimates on the apical electrode were within the predicted range in only 28 of subjects and the slope of the electrical pitch function was lower than predicted in all except one subject subjects with poorer hearing tended to have a lower pitch and a shallower electrical pitch function than predicted by the model pre operative hearing thresholds in the contralateral ear and hearing loss duration were not correlated with the degree of pitch match and there was no significant group effect of listening experience

a fourier approach to nonparametric deconvolution of a density estimate we consider the problem of constructing a nonparametric estimate of a probability density function h from independent random samples of observations from densities a and f when a represents the convolution of h and f our approach is based on truncated fourier inversion in which the truncation point plays the role of a smoothing parameter we derive the asymptotic mean integrated squared error of the estimate and use this formula to suggest a simple practical method for choosing the truncation point from the data strikingly when the smoothing parameter is chosen in this way then in many circumstances the estimator behaves to first order as though the true f were known

a narrow band pattern matching model of vowel perception the purpose of this paper is to propose and evaluate a new model of vowel perception which assumes that vowel identity is recognized by a template matching process involving the comparison of narrow band input spectra with a set of smoothed spectral shape templates that are learned through ordinary exposure to speech in the present simulation of this process the input spectra are computed over a sufficiently long window to resolve individual harmonics of voiced speech prior to template creation and pattern matching the narrow band spectra are amplitude equalized by a spectrum level normalization process and the information bearing spectral peaks are enhanced by a flooring procedure that zeroes out spectral values below a threshold function consisting of a center weighted running average of spectral amplitudes templates for each vowel category are created simply by averaging the narrow band spectra of like vowels spoken by a panel of talkers in the present implementation separate templates are used for men women and children the pattern matching is implemented with a simple city block distance measure given by the sum of the channel by channel differences between the narrow band input spectrum level equalized and floored and each vowel template spectral movement is taken into account by computing the distance measure at several points throughout the course of the vowel the input spectrum is assigned to the vowel template that results in the smallest difference accumulated over the sequence of spectral slices the model was evaluated using a large database consisting of 12 vowels in hvd context spoken by 45 men 48 women and 46 children the narrow band model classified vowels in this database with a degree of accuracy 91 4 approaching that of human listeners

a posteriori error estimates for nonconforming finite element methods computable a posteriori error bounds for a large class of nonconforming finite element methods are provided for a model poisson problem in two and three space dimensions besides a refined residual based a posteriori error estimate an averaging estimator is established and an l 2 estimate is included the a posteriori error estimates are reliable and efficient the proof of reliability relies on a helmholtz decomposition

low frequency and high frequency cochlear nonlinearity in humans low and high frequency cochlear nonlinearity was studied by measuring distortion product otoacoustic emission input output dpoae i o functions at 0 5 and 4khz in 103 normal hearing subjects behavioral thresholds at both f2 s were used to set l2 in db sl for each subject primary levels were optimized by determining the l1 resulting in the largest ldp for each l2 for each subject and both f2 s dpoae i o functions were measured using l2 inputs from 10db sl 0 5khz or 20db sl 4khz to 65db sl both frequencies mean dpoae i o functions averaged across subjects differed between the two frequencies even when threshold was taken into account the slopes of the i o functions were similar at 0 5 and 4khz for high level inputs with maximum compression ratios of about 4 1 at both frequencies the maximum slope near dpoae threshold was approximately 1 which occurred at lower levels at 4khz compared to 0 5khz these results suggest that there is a wider dynamic range and perhaps greater cochlear ampl

a cochlear frequency position function for several species 29 years later accurate cochlear frequency position functions based on physiological data would facilitate the interpretation of physiological and psychoacoustic data within and across species such functions might aid in developing cochlear models and cochlear coordinates could provide potentially useful spectral transforms of speech and other acoustic signals in 1961 an almost exponential function was developed greenwood 1961b 1974 by integrating an exponential function fitted to a subset of frequency resolution integration estimates critical bandwidths the resulting frequency position function was found to fit cochlear observations on human cadaver ears quite well and with changes of constants those on elephant cow guinea pig rat mouse and chicken bekesy 1960 as well as in vivo behavioral anatomical data on cats schucknecht 1953 since 1961 new mechanical and other physiological data have appeared on the human cat guinea pig chinchilla monkey and gerbil it is shown here that the newer extended data on human cadaver ears and from living animal preparations are quite well fit by the same basic function the function essentially requires only empirical adjustment of a single parameter to set an upper frequency limit while a slope parameter can be left constant if cochlear partition length is normalized to 1 or scaled if distance is specified in physical units constancy of slope and form in dead and living ears and across species increases the probability that the function fitting human cadaver data may apply as well to the living human ear this prospect increases the function s value in plotting auditory data and in modeling concerned with speech and other bioacoustic signals since it fits the available physiological data well and consequently if those data are correct remains independent of and an appropriate means to examine psychoacoustic data and assumptions

finite element galerkin approximation of the eigenvalues and eigenvectors of selfadjoint problems abstract this paper establishes some refined estimates for the approximation of the eigenvalues and eigenvectors of selfadjoint eigenvalue problems by finite element or more generally galerkin methods suppose lambda is an eigenvalue of multiplicity q of a selfajoint problem and let m lambda denote the space of eigenvectors corresponding to lambda keywords hilbert space

nonparametric prediction in measurement error models predicting the value of a variable y corresponding to a future value of an explanatory variable x based on a sample of previously observed independent data pairs x1 y1 xn yn distributed like x y is very important in statistics in the error free case where x is observed accurately this problem is strongly related to that of standard regression estimation since prediction of y can be achieved via estimation of the regression curve e y x when the observed xis and the future observation of x are measured with error prediction is of a quite different nature here if t denotes the future contaminated available version of x prediction of y can be achieved via estimation of e y t in practice estimating e y t can be quite challenging as data may be collected under different conditions making the measurement errors on xi and x nonidentically distributed we take up this problem in the nonparametric setting and introduce estimators which allow a highly adaptive approach to smoothing

the use of acoustic cues for phonetic identification effects of spectral degradation and electric hearing although some cochlear implant ci listeners can show good word recognition accuracy it is not clear how they perceive and use the various acoustic cues that contribute to phonetic perceptions in this study the use of acoustic cues was assessed for normal hearing nh listeners in optimal and spectrally degraded conditions and also for ci listeners two experiments tested the tense lax vowel contrast varying in formant structure vowel inherent spectral change and vowel duration and the word final fricative voicing contrast varying in f1 transition vowel duration consonant duration and consonant voicing identification results were modeled using mixed effects logistic regression these experiments suggested that under spectrally degraded conditions nh listeners decrease their use of formant cues and increase their use of durational cues compared to nh listeners ci listeners showed decreased use of spectral cues like formant structure and formant change and consonant voicing and showed greater use of durational cues especially for the fricative contrast the results suggest that although nh and ci listeners may show similar accuracy on basic tests of word phoneme or feature recognition they may be using different perceptual strategies in the process

the relative phonetic contributions of a cochlear implant and residual acoustic hearing to bimodal speech perception the addition of low passed lp speech or even a tone following the fundamental frequency f0 of speech has been shown to benefit speech recognition for cochlear implant ci users with residual acoustic hearing the mechanisms underlying this benefit are still unclear in this study eight bimodal subjects ci users with acoustic hearing in the non implanted ear and eight simulated bimodal subjects using vocoded and lp speech were tested on vowel and consonant recognition to determine the relative contributions of acoustic and phonetic cues including f0 to the bimodal benefit several listening conditions were tested ci vocoder lp tf0 env ci vocoder lp ci vocoder tf0 env compared with ci vocoder performance lp significantly enhanced both consonant and vowel perception whereas a tone following the f0 contour of target speech and modulated with an amplitude envelope of the maximum frequency of the f0 contour tf0 env enhanced only consonant perception information transfer analysis revealed a dual mechanism in the bimodal benefit the tone representing f0 provided voicing and manner information whereas lp provided additional manner place and vowel formant information the data in actual bimodal subjects also showed that the degree of the bimodal benefit depended on the cutoff and slope of residual acoustic hearing

a posteriori error estimates for nonconforming approximations of steklov eigenvalue problems this paper deals with a posteriori error estimators for the non conforming crouzeix raviart finite element approximations of the steklov eigenvalue problem first we define an error estimator of the residual type which can be computed locally from the approximate eigenpair and we prove the equivalence between this estimator and the broken energy norm of the error with constants independent of the corresponding eigenvalue next we prove that edge residuals dominate the volumetric part of the residual and that the volumetric part of the residual terms dominate the normal component of the jumps of the discrete fluxes across interior edges finally based on these results we introduce two simpler equivalent error estimators the analysis shows that these a posteriori error estimates are optimal up to higher order terms and that may be used for the design of adaptive algorithms

effects of upper frequency boundary and spectral warping on speech intelligibility in electrical stimulation speech understanding was tested for seven listeners using 12 electrode med el cochlear implants cis and six normal hearing listeners using a ci simulation eighteen different types of processing were evaluated which varied the frequency to tonotopic place mapping and the upper boundary of the frequency and stimulation range spectrally unwarped and warped conditions were included unlike previous studies on this topic the lower boundary of the frequency and stimulation range was fixed while the upper boundary was varied for the unwarped conditions only eight to ten channels were needed in both quiet and noise to achieve no significant degradation in speech understanding compared to the normal 12 electrode speech processing the unwarped conditions were often the best conditions for understanding speech however small changes in frequency to place mapping 0 77 octaves for the most basal electrode yielded no significant degradation in performance from the nearest unwarped condition a second exper

updated parameters and expanded simulation options for a model of the auditory periphery a phenomenological model of the auditory periphery in cats was previously developed by zilany and colleagues j acoust soc am 126 2390 2412 2009 to examine the detailed transformation of acoustic signals into the auditory nerve representation in this paper a few issues arising from the responses of the previous version have been addressed the parameters of the synapse model have been readjusted to better simulate reported physiological discharge rates at saturation for higher characteristic frequencies liberman j acoust soc am 63 442 455 1978 this modification also corrects the responses of higher characteristic frequency cf model fibers to low frequency tones that were erroneously much higher than the responses of low cf model fibers in the previous version in addition an analytical method has been implemented to compute the mean discharge rate and variance from the model s synapse output that takes into account the effects of absolute refractoriness

error estimators for nonconforming finite element approximations of the stokes problem in this paper we define and analyze a posteriori error estimators for nonconforming approximations of the stokes equations we prove that these estimators are equivalent to an appropriate norm of the error for the case of piecewise linear elements we define two estimators both of them are easy to compute but the second is simpler because it can be computed using only the right hand side and the approximate velocity we show how the first estimator can be generalized to higher order elements finally we present several numerical examples in which one of our estimators is used for adaptive refinement

limit cycle oscillations in a nonlinear state space model of the human cochlea it is somewhat surprising that linear analysis can account for so many features of the cochlea when it is inherently nonlinear for example the commonly detected spacing between adjacent spontaneous otoacoustic emissions soaes is often explained by a linear theory of coherent reflection zweig and shera 1995 j acoust soc am 98 2018 2047 the nonlinear saturation of the cochlear amplifier is however believed to be responsible for stabilizing the amplitude of a soae in this investigation a state space model is used to first predict the linear instabilities that arise given distributions of cochlear inhomogeneities and then subsequently to simulate the time varying spectra of the nonlinear models by comparing nonlinear simulation results to linear predictions it is demonstrated that nonlinear effects can have a strong impact on the steady state response of an unstable cochlear model sharply tuned components that decay away exponentially within 100 ms are shown to be due to linearly resonant modes of the model generated by the cochlear inhomogeneities some oscillations at linearly unstable frequencies are suppressed over a longer time scale whereas those that persist are due to linear instabilities and their distortion products

adaptive finite element methods with convergence rates adaptive finite element methods for numerically solving elliptic equations are used often in practice only recently 12 17 have these methods been shown to converge however this convergence analysis says nothing about the rates of convergence of these methods and therefore does in principle not guarantee yet any numerical advantages of adaptive strategies versus non adaptive strategies the present paper modifies the adaptive method of morin nochetto and siebert 17 for solving the laplace equation with piecewise linear elements on domains in 2 by adding a coarsening step and proves that this new method has certain optimal convergence rates in the energy norm which is equivalent to the h1 norm namely it is shown that whenever s 0 and the solution u is such that for each n 1 it can be approximated to accuracy o n s in the energy norm by a continuous piecewise linear function on a triangulation with n cells using complete knowledge of u then the adaptive algorithm constructs an approximation of the same type with the same asymptotic accuracy while using only information gained during the computational process moreover the number of arithmetic computations in the proposed method is also of order o n for each n 1 the construction and analysis of this adaptive method relies on the theory of nonlinear approximation

pitch matching in bimodal cochlear implant patients effects of frequency spectral envelope and level this study systematically investigated the effects of frequency level and spectral envelope on pitch matching in twelve bimodal cochlear implant ci users the participants were asked to vary the frequency and level of a pure or complex tone adjustable sounds presented in the non implanted ear to match the pitch and loudness of different reference stimuli presented to the implanted ear three reference sounds were used single electrode pulse trains pure tones and piano notes the data showed a significant effect of the frequency and complexity of the reference sounds no significant effect of the level of the reference sounds was found the magnitude of effect of frequency was compressed in the implanted ear on average a difference of seven semitones in the non implanted ear induced the same pitch change as a difference of 19 to 24 semitones for a stimulus presented to the implanted ear the spectral envelope of the adjustable sound presented to the non implanted ear also had a significant effect the matched frequencies were higher by an average of six semitones for the pure tone compared to a complex tone overall the ci listeners might have matched the stimuli based on timbre characteristics such as brightness

the effects of neural synchronization and peripheral compression on the acoustic reflex threshold this study investigates the acoustic reflex threshold art dependency on stimulus phase utilizing low level reflex audiometry neumann et al audiol neuro otol 1 359 369 1996 the goal is to obtain optimal broadband stimuli for elicitation of the acoustic reflex and to obtain objective determinations of cochlear hearing loss three types of tone complexes with different phase characteristics were investigated a stimulus that compensates for basilar membrane dispersion thus causing a large overall neural synchrony basilar membrane tone complex bmtc the temporally inversed stimulus ibmtc and random phase tone complexes rtc the arts were measured in eight normal hearing and six hearing impaired subjects five different conditions of peak amplitude and stimulus repetition rate were used for each stimulus type the results of the present study suggest that the art is influenced by at least two different factors a the degree of synchrony of neural activity across frequency and b the fast acting compression mechanism in the cochlea that is reduced in the case of a sensorineural hearing loss the results allow a clear distinction of the two subjects groups based on the different art for the utilized types and conditions of the stimuli these differences might be useful for objective recruitment detection in clinical diagnostics

evaluating human neural tuning curves from a mechanical model of the cochlea by relating them to psychophysical masking data human neural threshold tuning curves are estimated by scaling the parameters of allen s j acoust soc am 68 1660 1670 1980 resonant tectorial membrane model of the cat cochlea a way to evaluate the derived tuning curves using psychophysical data is developed based on a psychophysical detection model which relates the physiological tuning curves to psychophysical masking data a detection criterion defined by a relationship among the bandwidth of the frequency tuning curves expressed as an equivalent rectangular bandwidth erb the width of the excitation patterns expressed as an equivalent rectangular spread ers and the psychophysical critical ratio is explored and verified using cat data the detection criterion is then used to test the derived human curves by making predictions of psychophysical masking and comparing the predictions to experimental data the detection model may also provide a deeper understanding of the frequency resolving properties of the cochlea

an automated procedure for identifying spontaneous otoacoustic emissions an algorithm is described for objectively identifying and measuring spontaneous otoacoustic emissions soaes using the spectrum that results from transformation of the acoustic waveform measured in the outer ear canal prior to spectral analysis the rms level is calculated for successive short segments of the waveform and only the weakest 25 of the segments are retained for the spectral analysis the quietest 150 when using 16k point fast fourier transforms ffts the resulting initial spectrum is scanned for peaks potential soaes which are then deleted from the spectrum new values are estimated for the deleted values using linear extrapolations from frequency ranges on either side of the deleted values the end result is a smoothed spectrum devoid of all local peaks the initial spectrum is then compared peak by peak with the smoothed spectrum and those peaks having differences that exceed an objectively determined decision criterion are identified as likely soaes the effects of varying some of the important parameter values of the algorithm are described and the sensitivity of the procedure is evaluated by measuring the detection rate for a lorentzian peak of known amplitude added to a spectrum otherwise devoid of soaes

pitch shifts of mistuned partials a time domain model mistuning one partial of a complex harmonic tone makes that partial easier to hear as a tone separate from the complex at the same time two pitch shifts may be observed first the low pitch of the complex is shifted in the direction of the mistuning as if it were pulled by the partial second the mistuning of the partial is perceptually exaggerated as if the pitch of the partial were pushed away from the harmonic series defined by the complex this paper shows how the latter effect can emerge within a hypothetical neural circuit the circuit involves a gating neuron fed by three pathways one direct and excitatory and the other two delayed and inhibitory the neuron responds to any excitatory input spike unless it is accompanied by an inhibitory input spike on either delayed input thus acting as a kind of anticoincidence counter the first delay is fixed and tuned to the period of the background harmonic complex its purpose is to weaken correlates of in tune components and allow the mistuned partial to stand out the second delay is variable and used to estimate the period of the mistuned partial by searching for a minimum output as a function of delay with an appropriate choice of parameters the estimate is subject to shifts that are of the same sign as the mistuning and that peak at about 4 mistuning and decrease beyond as observed experimentally

invariance principles for cochlear mechanics hearing phases a functional model of the cochlea is devised on the basis of the results from classical experiments the basilar membrane filter is investigated in detail its phase is close to linear in the region around the peak of the amplification on one side this has consequences for the time analysis and on the other side this has led to a prediction on phase perception for very simple combinations of tones a prediction which is now confirmed by experiments equivariance under the dilation group permits one to describe the model by a wavelet transform daubechies ten lectures on wavelets siam philadelphia 1992 the wavelet is discussed in reference to the phase analysis of the basilar membrane filter

the lower bound property of the morley element eigenvalues in this paper we prove that the morley element eigenvalues approximate the exact ones from below on regular meshes including adaptive local refined meshes for the fourth order elliptic eigenvalue problems with the clamped boundary condition in any dimension and we implement the adaptive computation with the morley element to obtain lower bounds of eigenvalues for the vibration problem of clamped plate under tension and a fourth order elliptic eigenvalue problem with variable coefficients

a posterior error estimator and lower bound of a nonconforming finite element method in this paper we present an a posteriori error estimator and the lower bound for a nonconforming finite element approximation i e the extended crouzeix raviart element of the laplace eigenvalue problem under the guideline of the analysis to the laplace source problem we first give out an error indicator and prove it as the global upper and local lower bounds of the approximation error we also give the lower bound analysis for this type of nonconforming element on the adaptive meshes some numerical experiments are presented to verify our theoretical results

wavelet and matching pursuit estimates of the transient evoked otoacoustic emission latency different time frequency techniques may be used to investigate the relation between latency and frequency of transient evoked otoacoustic emissions in this work the optimization of these techniques and the interpretation of the experimental result are discussed time frequency analysis of click evoked otoacoustic emissions of 42 normal hearing young subjects has been performed using both wavelet and matching pursuit algorithms wavelet techniques are very effective to provide fast and reliable evaluation of the average latency of large samples of subjects a major advantage of the matching pursuit technique as observed by jedrzejczak et al j acoust soc am 115 2148 2158 2004 is to provide detailed information about the time evolution of the response of single ears at selected frequencies a hybrid matching pursuit algorithm that includes fourier spectral information was developed capable of speeding up computation times and of identifying spurious atoms whose latency frequency relation i

estimating bone conduction transfer functions using otoacoustic emissions a technique for estimating the nonparametric bone conduction transfer function using distortion product otoacoustic emissions dpoaes is presented individual transfer functions were obtained using dpoaes recorded from a single ear of five normal hearing adults repeatability of the technique was investigated by performing measurements on at least three dates functions were reasonably repeatable and were unique to each individual as expected from subjective measurements input force and dpoae measurements were made for each individual and a model of the auditory periphery representative of an average person was employed the technique is objective and requires only passive cooperation but robust dpoaes are needed and the measurement time can be onerous for a wide frequency band or fine frequency resolution with appropriate adjustments to the model of the auditory periphery the method could be applied with animal models

simultaneous latency estimations for distortion product otoacoustic emissions and envelope following responses the purpose of this research was to simultaneously estimate processing delays in the cochlea and brainstem using the same acoustic stimuli apparent latencies were estimated from ear canal measurements of 2f1 f2 distortion product otoacoustic emissions dpoaes and scalp recordings of the f2 f1 envelope following response efr the stimuli were equal level tone pairs 65db spl with the upper tone f2 set at either 900 or 1800hz to fix the initiation site of the dpoae and efr the frequency of f1 was swept continuously between frequency limits chosen to keep the efr response between 150 and 170hz the average dpoae latencies were 9 6 and 6 2ms for f2 900 and 1800hz and the corresponding efr latencies were 12 4 and 8 8ms in a control condition a third suppressor tone was added near the dpoae response frequency to evaluate whether the potential source at fdp was contributing significantly to the measured emission dpoae latency is the sum of both inward and outward cochlear delays the efr apparent la

the mammalian cochlear map is optimally warped the form of the mammalian cochlear frequency position map has been well described by greenwood and empirical values found for its coefficients for a number of species the apical portion of the mammalian map is spatially compressed relative to the base and this nonuniformity in the representation of frequency is evidently consistent across species however an evolutionary reason for this consistency encompassing critical band behavior with respect to position is conspicuously missing likewise the length of the cochlea in any mammal including echolocating species is related to body size but attempts to explain the length in terms of frequency limits range or resolution have no general explanation new insight stems from a hypothesis in which the map curvature may be appreciated as an adaptation for optimal frequency resolution over the auditory range it is demonstrated numerically that the mammalian curve may be considered a member of a family of curves which vary in their degree of warp the warp factor found to be common across mammals is an optimal trade off between four conflicting constraints 1 enhancing high frequency resolution 2 setting a lower bound on loss of existing low frequency resolution 3 minimizing map nonuniformity and 4 keeping the whole map smooth thereby avoiding reflections

the effect of stimulus frequency ratio on distortion product otoacoustic emission components a detailed measurement of distortion product otoacoustic emission dpoae fine structure was used to extract estimates of the two major components believed to contribute to the overall dpoae level in the ear canal a fixed ratio paradigm was used to record dpoae fine structure from three normal hearing ears over a range of 400 hz for 12 different stimulus frequency ratios between 1 053 and 1 36 and stimulus levels between 45 and 75 db spl inverse fourier transforms of the amplitude and phase data were filtered to extract the early component from the generator region of maximum stimulus overlap and the later component reflected from the characteristic frequency region of the dpoae after filtering the data were returned to the frequency domain to evaluate the impact of the stimulus frequency ratio and stimulus level on the relative levels of the components although there were significant differences between data from different ears some consistent patterns could be detected the component from the overlap region of the stimulus tones exhibits a bandpass shape with the maximum occurring at a ratio of 1 2 the mean data from the dpoae characteristic frequency region also exhibits a bandpass shape but is less sharply tuned and exhibits greater variety across ears and stimulus levels the component from the dpoae characteristic frequency region is dominant at ratios narrower than approximately 1 1 the transition varies between ears the relative levels of the two components are highly variable at ratios greater than 1 3 and highly dependent on the stimulus level the reflection component is larger at all ratios at the lowest stimulus level tested 45 45 db spl we discuss the factors shaping dpoae component behavior and some cursory implications for the choice of stimulus parameters to be used in clinical protocols

multiband product rule and consonant identification the multiband product rule also known as band independence is a basic assumption of articulation index and its extension the speech intelligibility index previously fletcher showed its validity for a balanced mix of 20 consonant vowel cv 20 vowel consonant vc and 60 consonant vowel consonant cvc sounds this study repeats miller and nicely s version of the hi lo pass experiment with minor changes to study band independence for the 16 miller nicely consonants the cut off frequencies are chosen such that the basilar membrane is evenly divided into 12 segments from 250 to 8000 hz with the high pass and low pass filters sharing the same six cut off frequencies in the middle results show that the multiband product rule is statistically valid for consonants on average it also applies to subgroups of consonants such as stops and fricatives which are characterized by a flat distribution of speech cues along the frequency it fails for individual consonants

spectral dynamics of sibilant fricatives are contrastive and language specific previous research has extensively investigated the spectral properties of sibilant fricatives with little consideration to how these properties vary over time to investigate such spectro temporal variation productions of english s and and of japanese s and in word initial prevocalic position were elicited from adult native speakers the spectral dynamics of these productions were analyzed in terms of a psychoacoustic measure of peak frequency peak erbn number peak erbn number was computed at 17 evenly spaced points across each fricative production the resulting peak erbn number trajectories were analyzed with orthogonal polynomial growth curve models to determine how peak frequency varied temporally within each fricative three analyses compared 1 the english sibilants to each other 2 the japanese sibilants to each other and 3 english s to japanese s the results indicated that in both english and japanese the sibilant fricatives differ acoustically in terms of both static i e overall level and dynamic i e shape aspects of the peak erbn number trajectories furthermore english s and japanese s exhibited language specific differences in the shape but not overall level of peak erbn number trajectories

frequency representation in the emu basilar papilla the frequency representation on the basilar papilla of the emu was investigated by recording from single auditory afferents and labeling them iontophoretically with horseradish peroxidase successfully labeled fibers covered a range of characteristic frequencies from 0 05 to 3 3 khz which corresponded to nearly 90 of the total papillar length the termination sites of labeled fibers within the basilar papilla correlated with their characteristic frequency the lowest frequencies being represented apically the highest basally the best mathematical description of the frequency distribution was given by an exponential regression with a uniform mapping constant of 0 8 mm octave this function deviates from the more commonly observed pattern of increasing space constant towards higher frequencies most labeled fibers innervated only one hair cell those contacting 4 or more hair cells were all of low characteristic frequency 0 11 khz all labeled fibers terminated within the neural 56 of the basilar papilla s width this innervation pattern is consistent with that observed in other bird species there was no conclusive evidence for a change of sensitivity and or frequency selectivity with innervational position across the papilla s width

simulation extrapolation estimation in parametric measurement error models we describe a simulation based method of inference for parametric measurement error models in which the measurement error variance is known or at least well estimated the method entails adding add

a convergent adaptive algorithm for poisson s equation we construct a converging adaptive algorithm for linear elements applied to poisson s equation in two space dimensions starting from a macro triangulation we describe how to construct an initial triangulation from a priori information then we use a posteriors error estimators to get a sequence of refined triangulation and approximate solutions it is proved that the error measured in the energy norm decreases at a constant rate in each step until a prescribed error bound is reached extension to higher order elements in two space dimension and numerical results are included

correlated amplitude fluctuations of spontaneous otoacoustic emissions the frequency spectrum of a spontaneous otoacoustic emission soae recording may contain multiple narrow peaks these peaks correspond to multiple soaes emitted by the inner ear the individual soaes exhibit small amplitude fluctuations the correlation between the amplitude fluctuations of multiple soaes in an individual ear was studied in 19 soae recordings from humans for 424 pairs of soaes the normalized amplitude cross correlation function was computed significant correlation was found in 63 pairs 15 34 pairs showed positive and 29 pairs showed negative correlation some soae pairs were part of an equally spaced triplet of emission peaks with one soae apparently generated as a distortion product of the other two primary soaes the occurrence of correlation between a distortion product soae and a primary soae was 56 this is significantly higher than the occurrence of correlation for the entire set of soae pairs in addition the spectral width of the distortion product soae was correlat

otoacoustic emission latency cochlear tuning and hearing functionality in neonates transient evoked otoacoustic emissions of a large population of neonates 466 ears are analyzed with the aim of establishing if a significant latency difference can be observed between pass 333 and fail 133 ears discriminated with a screening protocol based on band and global reproducibility the ears that did not pass the test in at least one frequency band are named fail for simplicity but they should be more appropriately defined as partial pass in a previous study significantly different average latencies had been observed in adult subjects comparing normal hearing and hearing impaired ears j acoust soc am 111 297 308 2002 but no similar study has been conducted on neonates yet an improved wavelet technique was applied to transient evoked otoacoustic emission data to get accurate experimental measurements of the otoacoustic emission latency the results show statistically significant differences between the latency distributions of the pass and fail populations with the fail latencies longer however non gaussianity of the distributions and systematic errors associated with low reproducibility levels suggested using a conservative approach in the data analysis and interpretation new otoacoustic estimates of cochlear tuning in neonates are also provided

time frequency domain filtering of evoked otoacoustic emissions time domain filtering is a standard analysis technique which is used to disentangle the two main vector components of the distortion product otoacoustic emission response exploiting their different phase frequency relation in this study a time frequency filtering technique based on the continuous wavelet transform is proposed to overcome the intrinsic limitations of the time domain filtering technique and to extend it also to the analysis of stimulus frequency and transient evoked otoacoustic emissions the advantages of the proposed technique are first discussed on a theoretical basis then practically demonstrated by applying it to the analysis of synthesized and real otoacoustic data the results show that the time frequency approach can be empirically optimized to get effective separation of the components of the otoacoustic response associated with either different generation mechanisms or different generation places focusing on a single component of the otoacoustic response with a given time frequency signature may also improve significantly the signal to noise ratio because the random noise contribution tends to be uniformly distributed on the time frequency plane

interaural envelope correlation change discrimination in bilateral cochlear implantees effects of mismatch centering and onset of deafness bilateral cochlear implant ci listeners can perform binaural tasks but they are typically worse than normal hearing nh listeners to understand why this difference occurs and the mechanisms involved in processing dynamic binaural differences interaural envelope correlation change discrimination sensitivity was measured in real and simulated ci users in experiment 1 11 ci eight late deafened three early deafened and eight nh listeners were tested in an envelope correlation change discrimination task just noticeable differences jnds were best for a matched place of stimulation and increased for an increasing mismatch in experiment 2 attempts at intracranially centering stimuli did not produce lower jnds in experiment 3 the percentage of correct identifications of antiphasic carrier pulse trains modulated by correlated envelopes was measured as a function of mismatch and pulse rate sensitivity decreased for increasing mismatch and increasing pulse rate the experiments led to two conclusions first envelope correlation change discrimination necessitates place of stimulation matched inputs however it is unclear if previous experience with acoustic hearing is necessary for envelope correlation change discrimination second nh listeners presented with ci simulations demonstrated better performance than real ci listeners if the simulations are realistic representations of electrical stimuli real ci listeners appear to have difficulty processing interaural information in modulated signals

features of stimulation affecting tonal speech perception implications for cochlear prostheses tone languages differ from english in that the pitch pattern of a single syllable word conveys lexical meaning in the present study dependence of tonal speech perception on features of the stimulation was examined using an acoustic simulation of a cis type speech processing strategy for cochlear prostheses contributions of spectral features of the speech signals were assessed by varying the number of filter bands while contributions of temporal envelope features were assessed by varying the low pass cutoff frequency used for extracting the amplitude envelopes ten normal hearing native mandarin chinese speakers were tested when the low pass cutoff frequency was fixed at 512 hz consonant vowel and sentence recognition improved as a function of the number of channels and reached plateau at 4 to 6 channels subjective judgments of sound quality continued to improve as the number of channels increased to 12 the highest number tested tone recognition i e recognition of the four mandarin tone patterns depended on both the number of channels and the low pass cutoff frequency the trade off between the temporal and spectral cues for tone recognition indicates that temporal cues can compensate for diminished spectral cues for tone recognition and vice versa an additional tone recognition experiment using syllables of equal duration showed a marked decrease in performance indicating that duration cues contribute to tone recognition a third experiment showed that recognition of processed fm patterns that mimic mandarin tone patterns was poor when temporal envelope and duration cues were removed