a reconstruction based chebyshev collocation method for the poisson equation an accurate treatment of the gibbs wilbraham phenomenon on irregular interfaces abstract an accurate numerical solution of the laplace equation and poisson equation is desirable in physical problems related to fluid flows and spectral methods are well suited towards this goal however in the presence of an interface the solution is discontinuous owing to the interface jump conditions and thus the numerical approximation is susceptible to the gibbs wilbraham phenomenon leading to a reduced solution accuracy in order to resolve the discontinuity at the interface we propose a reconstruction technique wherein the approximate solution is expressed as the sum of an infinitely differentiable smooth function and a modified heaviside function the smooth function is composed of chebyshev polynomials while the modified heaviside function is expressed in a weak form using the jump conditions across the interface the reconstruction framework allows us to impose the interfacial conditions exactly to motivate further application of the technique we first present the underlying rationale by revisiting the work of wilbraham 1848 13 followed by the numerical implementation we present numerical solution of four equations to illustrate the application of the proposed technique i an ordinary differential equation ode ii the euler bernoulli equation iii the laplace equation and iv the poisson equation the laplace and poisson equation are solved on a square domain embedded with complex interfacial geometries in two dimensions and on a cube with a spherical interface in three dimensions the method achieves error in the maximum norm o 10 14 using approximately 30 modes in each direction and requires a computational time of less than a second for all the problems we also conduct a sensitivity analysis to demonstrate the robustness of the algorithm under sub grid scale perturbations to interface location an alternate derivation for the modified heaviside function is proposed that is distinct from the taylor series expansion used to treat interface conditions in difference methods the pointwise error plots are subsequently presented to demonstrate that the gibbs wilbraham phenomenon is accurately resolved finally we present concluding thoughts on the potential applications of the reconstruction technique

convex entropies and hyperbolicity for general euler equations the compressible euler equations possess a family of generalized entropy densities of the form rho f sigma where rho is the mass density sigma is the specific entropy and f is an arbitrary function entropy inequalities associated with convex entropy densities characterize physically admissible shocks for polytropic gases harten has determined which rho f sigma are strictly convex in this paper we extend this determination to gases with an arbitrary equation of state moreover we show that at every state where the sound speed is positive i e where the euler equations are hyperbolic there exist rho f sigma that are strictly convex thereby establishing the converse of the general fact that the existence of a strictly convex entropy density implies hyperbolicity

stabilized high order galerkin methods based on a parameter free dynamic sgs model for les the high order spectral element approximation of the euler equations is stabilized via a dynamic sub grid scale model dyn sgs this model was originally designed for linear finite elements to solve compressible flows at large mach numbers we extend its application to high order spectral elements to solve the euler equations of low mach number stratified flows the major justification of this work is twofold stabilization and large eddy simulation are achieved via one scheme only because the diffusion coefficients of the regularization stresses obtained via dyn sgs are residual based the effect of the artificial diffusion is minimal in the regions where the solution is smooth the direct consequence is that the nominal convergence rate of the high order solution of smooth problems is not degraded to our knowledge this is the first application in atmospheric modeling of a spectral element model stabilized by an eddy viscosity scheme that by construction may fulfill stabilization requirements can model turbulence via les and is completely free of a user tunable parameter from its derivation it will be immediately clear that dyn sgs is independent of the numerical method it could be implemented in a discontinuous galerkin finite volume or other environments alike preliminary discontinuous galerkin results are reported as well the straightforward extension to non linear scalar problems is also described a suite of 1d 2d and 3d test cases is used to assess the method with some comparison against the results obtained with the most known lilly smagorinsky sgs model

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

the effect of spatial separation on informational and energetic masking of speech the effect of spatial separation of sources on the masking of a speech signal was investigated for three types of maskers ranging from energetic to informational normal hearing listeners performed a closed set speech identification task in the presence of a masker at various signal to noise ratios stimuli were presented in a quiet sound field the signal was played from 0 azimuth and a masker was played either from the same location or from 90 to the right signals and maskers were derived from sentences that were preprocessed by a modified cochlear implant simulation program that filtered each sentence into 15 frequency bands extracted the envelopes from each band and used these envelopes to modulate pure tones at the center frequencies of the bands in each trial the signal was generated by summing together eight randomly selected frequency bands from the preprocessed signal sentence three maskers were derived from the preprocessed masker sentences 1 different band sentence which was generated by summing together six randomly selected frequency bands out of the seven bands not present in the signal resulting in primarily informational masking 2 different band noise which was generated by convolving the different band sentence with gaussian noise and 3 same band noise which was generated by summing the same eight bands from the preprocessed masker sentence that were used in the signal sentence and convolving the result with gaussian noise resulting in primarily energetic masking results revealed that in the different band sentence masker the effect of spatial separation averaged 18 db at 51 correct while in the different band and same band noise maskers the effect was less than 10 db these results suggest that in these conditions the advantage due to spatial separation of sources is greater for informational masking than for energetic masking

the relationship between frequency selectivity and pitch discrimination sensorineural hearing loss this study tested the relationship between frequency selectivity and the minimum spacing between harmonics necessary for accurate fo discrimination fundamental frequency difference limens fo dls were measured for ten listeners with moderate sensorineural hearing loss snhl and three normal hearing listeners for sine and random phase harmonic complexes bandpass filtered between 1500 and 3500 hz with fo s ranging from 75 to 500 hz or higher all listeners showed a transition between small good fo dls at high fo s and large poor fo dls at low fo s although the fo at which this transition occurred fo tr varied across listeners three measures thought to reflect frequency selectivity were significantly correlated to both the fo tr and the minimum fo dl achieved at high fo s 1 the maximum fo for which fo dls were phase dependent 2 the maximum modulation frequency for which amplitude modulation and quasi frequency modulation were discriminable and 3 the equivalent rectangular bandwidth of the auditory filter estimated using the notched noise method these results provide evidence of a relationship between fo discrimination performance and frequency selectivity in listeners with snhl supporting spectral and spectro temporal theories of pitch perception that rely on sharp tuning in the auditory periphery to accurately extract fo information

modulation spectra of natural sounds and ethological theories of auditory processing the modulation statistics of natural sound ensembles were analyzed by calculating the probability distributions of the amplitude envelope of the sounds and their time frequency correlations given by the modulation spectra these modulation spectra were obtained by calculating the two dimensional fourier transform of the autocorrelation matrix of the sound stimulus in its spectrographic representation since temporal bandwidth and spectral bandwidth are conjugate variables it is shown that the joint modulation spectrum of sound occupies a restricted space sounds cannot have rapid temporal and spectral modulations simultaneously within this restricted space it is shown that natural sounds have a characteristic signature natural sounds in general are low passed showing most of their modulation energy for low temporal and spectral modulations animal vocalizations and human speech are further characterized by the fact that most of the spectral modulation power is found only for low temporal modulation similarly the distribution of the amplitude envelopes also exhibits characteristic shapes for natural sounds reflecting the high probability of epochs with no sound systematic differences across frequencies and a relatively uniform distribution for the log of the amplitudes for vocalizations it is postulated that the auditory system as well as engineering applications may exploit these statistical properties to obtain an efficient representation of behaviorally relevant sounds to test such a hypothesis we show how to create synthetic sounds with first and second order envelope statistics identical to those found in natural sounds

the effect of tone vocoding on spatial release from masking for old hearing impaired listeners old hearing impaired listeners generally benefit little from lateral separation of multiple talkers when listening to one of them this study aimed to determine how spatial release from masking srm in such listeners is affected when the interaural time differences itds in the temporal fine structure tfs are manipulated by tone vocoding tvc at the ears by a master hearing aid system word recall was compared with and without tvc when target and masker sentences from a closed set were played simultaneously from the front loudspeaker co located and when the maskers were played 45 to the left and right of the listener separated for 20 hearing impaired listeners aged 64 to 86 srm was 3 7 db smaller with tvc than without tvc this difference in srm correlated with mean audiometric thresholds below 1 5 khz even when monaural tfs sensitivity discrimination of frequency shifts in identically filtered complexes was partialed out suggesting that low frequency audiometric thresholds may be a good indicator of candidacy for hearing aids that preserve itds the tvc difference in srm was not correlated with age pure tone itd thresholds nor fundamental frequency difference limens and only with monaural tfs sensitivity before control for low frequency audiometric thresholds

a unitary model of pitch perception a model of the mechanism of residue pitch perception is revisited it is evaluated in the context of some new empirical results and it is proposed that the model is able to reconcile a number of differing approaches in the history of theories of pitch perception the model consists of four sequential processing stages peripheral frequency selectivity within channel half wave rectification and low pass filtering within channel periodicity extraction and cross channel aggregation of the output the pitch percept is represented by the aggregated periodicity function using autocorrelation as the periodicity extraction method and the summary autocorrelation function sacf as the method for representing pitch information it is shown that the model can simulate new experimental results that show how the quality of the pitch percept is influenced by the resolvability of the harmonic components of the stimulus complex these include i the pitch of harmonic stimuli whose components alternate in phase ii the increased frequency difference limen of tones consisting of higher harmonics and iii the influence of a mistuned harmonic on the pitch of the complex as a function of its harmonic number to accommodate these paradigms it was necessary to compare stimuli along the length of the sacf rather than relying upon the highest peak alone these new results demonstrate that the model responds differently to complexes consisting of low and high harmonics as a consequence it is not necessary to postulate two separate mechanisms to explain different pitch percepts associated with resolved and unresolved harmonics

speech intelligibility as a function of the number of channels of stimulation for signal processors using sine wave and noise band outputs vowels consonants and sentences were processed through software emulations of cochlear implant signal processors with 2 9 output channels the signals were then presented as either the sum of sine waves at the center of the channels or as the sum of noise bands the width of the channels to normal hearing listeners for identification the results indicate as previous investigations have suggested that high levels of speech understanding can be obtained using signal processors with a small number of channels the number of channels needed for high levels of performance varied with the nature of the test material for the most difficult material vowels produced by men women and girls no statistically significant differences in performance were observed when the number of channels was increased beyond 8 for the least difficult material sentences no statistically significant differences in performance were observed when the number of channels was increased beyond 5 the nature of the output signal noise bands or sine waves made only a small difference in performance the mechanism mediating the high levels of speech recognition achieved with only few channels of stimulation may be the same one that mediates the recognition of signals produced by speakers with a high fundamental frequency i e the levels of adjacent channels are used to determine the frequency of the input signal the results of an experiment in which frequency information was altered but temporal information was not altered indicates that vowel recognition is based on information in the frequency domain even when the number of channels of stimulation is small

spectral integration of speech bands in normal hearing and hearing impaired listeners this investigation examined whether listeners with mild moderate sensorineural hearing impairment have a deficit in the ability to integrate synchronous spectral information in the perception of speech in stage 1 the bandwidth of filtered speech centered either on 500 or 2500 hz was varied adaptively to determine the width required for approximately 15 25 correct recognition in stage 2 these criterion bandwidths were presented simultaneously and percent correct performance was determined in fixed block trials experiment 1 tested normal hearing listeners in quiet and in masking noise the main findings were 1 there was no correlation between the criterion bandwidths at 500 and 2500 hz 2 listeners achieved a high percent correct in stage 2 approximately 80 and 3 performance in quiet and noise was similar experiment 2 tested listeners with mild moderate sensorineural hearing impairment the main findings were 1 the impaired listeners showed high variability in stage 1 with some listeners requiring narrower and others requiring wider bandwidths than normal and 2 hearing impaired listeners achieved percent correct performance in stage 2 that was comparable to normal the results indicate that listeners with mild moderate sensorineural hearing loss do not have an essential deficit in the ability to integrate across frequency speech information

the importance for speech intelligibility of random fluctuations in steady background noise spectrally shaped steady noise is commonly used as a masker of speech the effects of inherent random fluctuations in amplitude of such a noise are typically ignored here the importance of these random fluctuations was assessed by comparing two cases for one speech was mixed with steady speech shaped noise and n channel tone vocoded a process referred to as signal domain mixing sdm this preserved the random fluctuations of the noise for the second the envelope of speech alone was extracted for each vocoder channel and a constant was added corresponding to the root mean square value of the noise envelope for that channel this is referred to as envelope domain mixing edm it removed the random fluctuations of the noise sinusoidally modulated noise and a single talker were also used as backgrounds with both sdm and edm speech intelligibility was measured for n 12 19 and 30 with the target to background ratio fixed at 7 db for sdm performance was best for the speech background and worst for the steady noise for edm this pattern was reversed intelligibility with steady noise was consistently very poor for sdm but near ceiling for edm demonstrating that the random fluctuations in steady noise have a large effect

a time domain description for the pitch strength of iterated rippled noise two versions of a cascaded add attenuate and delay circuit were used to generate iterated rippled noise irn stimuli irn stimuli produce a repetition pitch whose strength relative to the noise can be varied by changing the type of circuit the attenuation or the number of iterations in the circuit listeners were asked to discriminate between various pairs of irn stimuli which differed in the type of network used to generate the sounds or the number of iterations n 1 2 3 4 7 and 9 performance was determined for irn stimuli generated with delays of 2 4 and 8 ms and for four bandpass filter conditions 0 2000 250 2000 500 2000 and 750 2000 hz some irn stimuli were extremely difficult to discriminate despite relatively large spectral differences while other irn stimuli produced readily discriminable changes in perception despite small spectral differences these contrasting results are inconsistent with simple spectral explanations for the perception of irn stimuli an explanation based on the first peak of the autocorrelation function of irn stimuli is consistent with the results simulations of the processing performed by the peripheral auditory system i e interval histograms and correlograms produce results which are consistent with the involvement of these temporal processes in the perception of irn stimuli

effects of degradation of intensity time or frequency content on speech intelligibility for normal hearing and hearing impaired listeners many hearing impaired listeners suffer from distorted auditory processing capabilities this study examines which aspects of auditory coding i e intensity time or frequency are distorted and how this affects speech perception the distortion sensitivity model is used the effect of distorted auditory coding of a speech signal is simulated by an artificial distortion and the sensitivity of speech intelligibility to this artificial distortion is compared for normal hearing and hearing impaired listeners stimuli speech plus noise are wavelet coded using a complex sinusoidal carrier with a gaussian envelope 14 octave bandwidth intensity information is distorted by multiplying the modulus of each wavelet coefficient by a random factor temporal and spectral information are distorted by randomly shifting the wavelet positions along the temporal or spectral axis respectively measured were 1 detection thresholds for each type of distortion and 2 speech reception thresholds for various degrees

implementation of the entropy viscosity method with the discontinuous galerkin method the notion of entropy viscosity method introduced in guermond and pasquetti 21 is extended to the discontinuous galerkin framework for scalar conservation laws and the compressible euler equations

on the symmetric form of systems of conservation laws with entropy abstract this paper reviews the symmetrizability of systems of conservation laws which possess entropy functions symmetric formulations in conservation form for the equations of gas dynamics are presented

an unconditionally stable method for the euler equations we discuss how to combine a front tracking method with dimensional splitting to solve systems of conservation laws numerically in two space dimensions in addition we present an adaptive grid refinement strategy the method is unconditionally stable and allows for moderately highcflnumbers typically 1 4 and thus it is highly efficient the method is applied to the euler equations of gas dynamics in particular it is tested on an expanding circular gas front a wind tunnel with a step a double mach reflection and a shock bubble interaction the method shows very sharp resolution of shocks

a central spectrum model a synthesis of auditory nerve timing and place cues in monaural communication of frequency spectrum a probabilistic psychophysical model for monaural communication from the auditory nerve to the brain is given in the form of a tonotopic display of stimulus spectrum termed central spectrum the model builds upon prior research demonstrating the potential of neural timing cues from the auditory nerve for conveying information on complex spectra and was designed to meet the quantified demands of the psychophysics of frequency measurement the central spectrum magnitude at each frequency is determined by the response of the auditory nerve fiber with characteristic frequency matching that frequency an interval histogram from each fiber is passed through a filter matched to the characteristic frequency of the fiber this output versus characteristic frequency defines the central spectrum detailed analysis demonstrates that efficient probabilistic processing of the central spectrum describes known psychophysical properties of frequency measurement in discrimination and periodicity pitch experiments psycho

a maximum principle preserving finite element method for scalar conservation equations this paper introduces a first order viscosity method for the explicit approximation of scalar conservation equations with lipschitz fluxes using continuous finite elements on arbitrary grids in any space dimension provided the lumped mass matrix is positive definite the method is shown to satisfy the local maximum principle under a usual cfl condition the method is independent of the cell type for instance the mesh can be a combination of tetrahedra hexahedra and prisms in three space dimensions

the relative strength of the tone and noise components in iterated rippled noise rippled noise is constructed by delaying a random noise and adding it back to the original iterated rippled noise irn is constructed by repeating the delay and add process irn produces a two component perception i e a buzzy tone with a pitch equal to the reciprocal of the delay and a background noise that sounds like the original random noise the perceived tone noise ratio increases with the number of iterations the effective tone noise ratio in irn sounds with 1 16 iterations was measured in a discrimination matching experiment each irn was paired with a range of standard sounds having varying proportions of a broadband noise and a complex tone to find the point where their perceived tone noise ratios are the same the experiment shows that the tone noise ratio of the matching standard increases 3 8 db per doubling of the number of iterations in the irn stimulus spectral models of auditory perception explain the pitch of irn in terms of peaks in the region of the first five to eight harmonic

an autocorrelation model with place dependence to account for the effect of harmonic number on fundamental frequency discrimination fundamental frequency f0 difference limens dls were measured as a function of f0 for sine and random phase harmonic complexes bandpass filtered with 3 db cutoff frequencies of 2 5 and 3 5 khz low region or 5 and 7 khz high region and presented at an average 15 db sensation level approximately 48 db spl per component in a wideband background noise fundamental frequencies ranged from 50 to 300 hz and 100 to 600 hz in the low and high spectral regions respectively in each spectral region f0 dls improved dramatically with increasing f0 as approximately the tenth harmonic appeared in the passband generally f0 dls for complexes with similar harmonic numbers were similar in the two spectral regions the dependence of f0 discrimination on harmonic number presents a significant challenge to autocorrelation ac models of pitch in which predictions generally depend more on spectral region than harmonic number a modification involving a lag window is proposed and tested restricting the ac repr

modeling formant frequency discrimination of female vowels the present investigations were designed to establish the features of vowel spectra that mediate formant frequency discrimination thresholds for detecting frequency shifts in the first and second formants of two steady state vowels were initially measured for conditions in which the amplitudes of all harmonics varied in accordance with a model of cascade formant synthesis in this model changes in formant frequency produce level variations in components adjacent to the altered formant as well as in harmonics spectrally remote from the shifted resonant frequency discrimination thresholds determined with the cascade synthesis procedure were then compared to difference limens dls obtained when the number of harmonics exhibiting level changes was limited to the frequency region surrounding the altered formant results indicated that amplitude variations could be restricted to one to three components near the shifted formant before significant increases in formant frequency dls were observed in a second

the case of the missing delay lines synthetic delays obtained by cross channel phase interaction temporal models of pitch and harmonic segregation call for delays of up to 30ms to cover the full range of existence of musical pitch to date there is little anatomical or physiological evidence for delays that long we propose a mechanism by which delays may be synthesized from cross channel phase interaction phases of adjacent cochlear filter channels are shifted by an amount proportional to frequency and then combined as a weighted sum to approximate a delay synthetic delays may be used by pitch perception models such as autocorrelation segregation models such as harmonic cancellation and binaural processing models to explain sensitivity to large interaural delays the maximum duration of synthetic delays is limited by the duration of the impulse responses of cochlear filters itself inversely proportional to cochlear filter bandwidth maximum delay is thus frequency dependent this may explain the fact puzzling for temporal pitch models such as autocorrelation that pitch is more salient and eas

cancellation model of pitch perception a model of pitch perception is presented involving an array of delay lines and inhibitory gating neurons in response to a periodic sound a minimum appears in the pattern of outputs of the inhibitory neurons at a lag equal to the period of the sound the position of this minimum is the cue to pitch the model is similar to the autocorrelation model of pitch multiplication being replaced by an operation similar to subtraction and maxima by minima the two models account for a wide class of pitch phenomena in very much the same way the principal goal of this paper is to demonstrate this fact several features of the cancellation model may be to its advantage it is closely related to the operation of harmonic cancellation that can account for segregation of concurrent harmonic stimuli it can be generalized to explain the perception of multiple pitches and it shows a greater degree of sensitivity to phase than autocorrelation which may allow it to explain certain phenomena that autocorrelation cannot account for

on the stability of the dual problem for high reynolds number flow past a circular cylinder in two dimensions in this paper we present a computational study of the stability of time dependent dual problems for compressible flow at high reynolds numbers in two dimensions the dual problem measures the sensitivity of an output functional with respect to numerical errors and is a key part of goal oriented a posteriori error estimation our investigation shows that the dual problem associated with the computation of the drag force for the compressible euler navier stokes equations which are approximated numerically using different temporal discretization and stabilization techniques is unstable and exhibits blow up for several mach regimes considered in this paper

psychophysical evidence against the autocorrelation theory of auditory temporal processing nowadays it is widely believed that the temporal structure of the auditory nerve fibers response to sound stimuli plays an important role in auditory perception an influential hypothesis is that information is extracted from this temporal structure by neural operations akin to an autocorrelation algorithm the goal of the present work was to test this hypothesis the stimuli consisted of sequences of unipolar clicks that were high pass filtered and mixed with low pass noise so as to exclude spectral cues in experiment 1 interfering clicks were inserted in an otherwise periodic isochronous click sequence each click belonging to the periodic sequence was followed after a random portion of the period by one interfering click this disrupted the detection of temporal regularity even when the interfering clicks were 5 db less intense than the periodic clicks experiments 2 4 used click sequences that showed a single peak in their autocorrelation functions for some sequences this peak originated from first order temporal regularities that is from the temporal relations between consecutive clicks for other sequences the peak originated instead from second order regularities relative to nonconsecutive clicks the detection of second order regularities appeared to be much more difficult than the detection of comparable first order regularities overall these results do not tally with the current autocorrelation models of temporal processing they suggest that the extraction of temporal information from a group of closely spaced spectral components makes no use of time intervals between nonconsecutive peaks of the amplitude envelope

the effect of intensity perturbations on speech intelligibility for normal hearing and hearing impaired listeners hearing impaired listeners are known to suffer from reduced speech intelligibility in noise even if sounds are above their hearing thresholds this study examined the possible contribution of reduced acuity of intensity coding to this problem the distortion sensitivity model was used the effect of reduced acuity of auditory intensity coding on intelligibility was mimicked by an artificial distortion of the speech intensity coding and the sensitivity to this distortion for hearing impaired listeners was compared with that for normal hearing listeners stimuli speech plus noise were wavelet coded using a gaussian wavelet 1 4 octave bandwidth the intensity coding was distorted by multiplying the modulus of each wavelet coefficient by a random factor speech reception thresholds srts were measured for various degrees of intensity perturbation hearing impaired listeners were classified as suffering from suprathreshold deficits if intelligibility of undistorted speech was worse than predicted from audibility by the speech intelligibility index model ansi ansi s3 5 1997 1997 hearing impaired listeners without suprathreshold deficits were as sensitive to the intensity distortion as the normal hearing listeners hearing impaired listeners with suprathreshold deficits appeared to be less sensitive results indicate that reduced acuity of auditory intensity coding may be a factor underlying reduced speech intelligibility in noise for the hearing impaired hearing impaired listeners are known to suffer from reduced speech intelligibility in noise even if sounds are above their hearing thresholds this study examined the possible contribution of reduced acuity of intensity coding to this problem the distortion sensitivity model was used the effect of reduced acuity of auditory intensity coding on intelligibility was mimicked by an artificial distortion of the speech intensity coding and the sensitivity to this distortion for hearing impaired listeners was compared with that for normal hearing listeners stimuli speech plus noise were wavelet coded using a gaussian wavelet 1 4 octave bandwidth the intensity coding was distorted by multiplying the modulus of each wavelet coefficient by a random factor speech reception thresholds srts were measured for various degrees of intensity perturbation hearing impaired listeners were classified as suffering from suprathreshold deficits if intelligibility of undistorted speech was worse than predicted from

a second order maximum principle preserving lagrange finite element technique for nonlinear scalar conservation equations this paper proposes an explicit at least second order maximum principle sat isfying lagrange finite element method for solving nonlinear scalar conservation equations the technique is based on a new viscous bilinear form introduced in guermond and nazarov com put methods appl mech engrg 272 2014 pp 198 213 a high order entropy viscosity method and the boris book zalesak flux correction technique the algorithm works for arbitrary meshes in any space dimension and for all lipschitz fluxes the formal second order accuracy of the method and its convergence properties are tested on a series of linear and nonlinear benchmark problems

slip with friction and penetration with resistance boundary conditions for the navier stokes equations numerical tests and aspects of the implementation we consider slip with friction and penetration with resistance boundary conditions in the steady state navier stokes equations this paper describes some aspects of the implementation of these boundary conditions for finite element discretizations numerical tests on two and three dimensional channel flows across a step using the slip with friction boundary condition study the influence of the friction parameter on the position of the reattachment point and the reattachment line of the recirculating vortex respectively

benefit of high rate envelope cues in vocoder processing effect of number of channels and spectral region in cochlear implants or vocoder simulations of cochlear implants the transmission of envelope cues at high rates related to voice fundamental frequency f0 may be limited by the widths of the filters used to form the channels and or by the cutoff frequency flp of the low pass filters used for envelope extraction the effect of varying flp in tone and noise vocoders was investigated for channel numbers n from 6 to 18 as n increased the widths of the channels decreased the value of flp was 45hz envelope or e filter or 180hz pitch or p filter the following combinations of cutoff frequencies were used for channels below and above 1500hz respectively ee pe ep and pp results from a competing talker task showed that the tone vocoder led to better intelligibility than the noise vocoder the pp condition led to the best intelligibility and the ee condition to the worst for n 6 intelligibility was better for condition pe than for condition ep for n 18 the reverse was true the results

contribution of temporal fine structure information and fundamental frequency separation to intelligibility in a competing speaker paradigm the speech reception threshold srt for identifying a target speaker in a background speaker was measured as a function of the difference f0sep in fundamental frequency f0 between the two speakers the amount of original temporal fine structure tfs information in the mixed signals was manipulated by tone vocoding channels above a certain cutoff channel co when the natural variations in f0 of both speakers were preserved the srt did not decrease with increasing f0sep indicating that short term differences in f0 can allow perceptual segregation of two speakers even when their f0s cross when f0 variations were removed from both speakers increasing f0sep led to decreased better srts the decrease was greater for unprocessed signals than for fully tone vocoded signals however the decrease was similar for unprocessed signals and for signals with original tfs below 1600 hz suggesting that most of the benefit from increasing f0 difference depends on the use of tfs information at lower frequenci

pitch discrimination of diotic and dichotic tone complexes harmonic resolvability or harmonic number three experiments investigated the relationship between harmonic number harmonic resolvability and the perception of harmonic complexes complexes with successive equal amplitude sine or random phase harmonic components of a 100 or 200 hz fundamental frequency f0 were presented dichotically with even and odd components to opposite ears or diotically with all harmonics presented to both ears experiment 1 measured performance in discriminating a 3 5 5 frequency difference between a component of a harmonic complex and a pure tone in isolation listeners achieved at least 75 correct for approximately the first 10 and 20 individual harmonics in the diotic and dichotic conditions respectively verifying that only processes before the binaural combination of information limit frequency selectivity experiment 2 measured fundamental frequency difference limens f0 dls as a function of the average lowest harmonic number similar results at both f0 s provide further evidence that harmonic number not absolute frequency underlies the order of magnitude increase observed in f0 dls when only harmonics above about the 10th are presented similar results under diotic and dichotic conditions indicate that the auditory system in performing f0 discrimination is unable to utilize the additional peripherally resolved harmonics in the dichotic case in experiment 3 dichotic complexes containing harmonics below the 12th or only above the 15th elicited pitches of the f0 and twice the f0 respectively together experiments 2 and 3 suggest that harmonic number regardless of peripheral resolvability governs the transition between two different pitch percepts one based on the frequencies of individual resolved harmonics and the other based on the periodicity of the temporal envelope

entropy viscosity method for nonlinear conservation laws a new class of high order numerical methods for approximating nonlinear conservation laws is described entropy viscosity method the novelty is that a nonlinear viscosity based on the local size of an entropy production is added to the numerical discretization at hand this new approach does not use any flux or slope limiters applies to equations or systems supplemented with one or more entropy inequalities and does not depend on the mesh type and polynomial approximation various benchmark problems are solved with finite elements spectral elements and fourier series to illustrate the capability of the proposed method

on the number of channels needed to understand speech recent studies have shown that high levels of speech understanding could be achieved when the speech spectrum was divided into four channels and then reconstructed as a sum of four noise bands or sine waves with frequencies equal to the center frequencies of the channels in these studies speech understanding was assessed using sentences produced by a single male talker the aim of experiment 1 was to assess the number of channels necessary for a high level of speech understanding when sentences were produced by multiple talkers in experiment 1 sentences produced by 135 different talkers were processed through n 2 n 16 number of channels synthesized as a sum of n sine waves with frequencies equal to the center frequencies of the filters and presented to normal hearing listeners for identification a minimum of five channels was needed to achieve a high level 90 of speech understanding asymptotic performance was achieved with eight channels at least for the speech material used in this study the outcome of experiment 1 demonstrated that the number of channels needed to reach asymptotic performance varies as a function of the recognition task and or need for listeners to attend to fine phonetic detail in experiment 2 sentences were processed through 6 and 16 channels and quantized into a small number of steps the purpose of this experiment was to investigate whether listeners use across channel differences in amplitude to code frequency information particularly when speech is processed through a small number of channels for sentences processed through six channels there was a significant reduction in speech understanding when the spectral amplitudes were quantized into a small number 8 of steps high levels 92 of speech understanding were maintained for sentences processed through 16 channels and quantized into only 2 steps the findings of experiment 2 suggest an inverse relationship between the importance of spectral amplitude resolution number of steps and spectral resolution number of channels

the recognition of sentences in noise by normal hearing listeners using simulations of cochlear implant signal processors with 6 20 channels sentences were processed through simulations of cochlear implant signal processors with 6 8 12 16 and 20 channels and were presented to normal hearing listeners at 2 db s n and at 2 db s n the signal processing operations included bandpass filtering rectification and smoothing of the signal in each band estimation of the rms energy of the signal in each band computed every 4 ms and generation of sinusoids with frequencies equal to the center frequencies of the bands and amplitudes equal to the rms levels in each band the sinusoids were summed and presented to listeners for identification at issue was the number of channels necessary to reach maximum performance on tests of sentence understanding at 2 db s n the performance maximum was reached with 12 channels of stimulation at 2 db s n the performance maximum was reached with 20 channels of stimulation these results in combination with the outcome that in quiet asymptotic performance is reached with five channels of stimulation demonstrate that more channels are needed in noise than in quiet to reach a high level of sentence understanding and that as the s n becomes poorer more channels are needed to achieve a given level of performance

the importance of temporal fine structure information in speech at different spectral regions for normal hearing and hearing impaired subjects speech reception thresholds srts were measured for target and competing speech signals processed to contain variable amounts of temporal fine structure tfs information signals were filtered into 30 1 erb n wide channels where erb n refers to the bandwidth of normal auditory filters which were either tone vocoded preserving temporal envelope information extracted using the hilbert transform or left unprocessed containing both tfs and envelope information improvements in srt were compared when tfs was progressively introduced starting with the high or low frequency channels the results suggest some redundancy in the tfs information across frequency regions in a second experiment the signal was divided into five 6 erb n wide spectral regions four of which were tone vocoded the remaining region was either absent creating a spectral notch or was present and unprocessed srts were measured for normal hearing and hearing impaired subjects conditions where all channels were vocoded or unprocessed were also included normal hearing subjects benefited similarly when tfs information was added to each region suggesting that tfs information is important over a wide frequency range hearing impaired subjects benefited less although the benefit varied across subjects benefit from tfs information in speech was correlated with a psychophysical measure of tfs sensitivity obtained at two center frequencies

convergence of a residual based artificial viscosity finite element method we present a residual based artificial viscosity finite element method to solve conservation laws the galerkin approximation is stabilized by only residual based artificial viscosity without any least squares supg or streamline diffusion terms we prove convergence of the method applied to a scalar conservation law in two space dimensions toward an unique entropy solution for implicit time stepping schemes

weighting the edge stabilization a modification of the edge stabilization technique is proposed to improve the behavior of the method when solving conservation equations with nonsmooth data or nonsmooth solutions the key ingredient is tempering the edge stabilization in regions of large gradients through appropriate weights the new method is shown to preserve the convergence properties of the original method on smooth solutions and numerical tests indicate that it performs better on nonsmooth solutions

free and forced vibration of a cantilever beam contacting with a rigid cylindrical foundation abstract in this paper the free and forced vibration of a fixed free euler bernoulli beam in contact with a rigid cylindrical foundation is studied one end of the beam is clamped at the top of the rigid cylindrical foundation and the other end is free the vibrations are separated into upward and downward configurations since a unilateral constraint is added by the cylindrical foundation the partial differential equation describing the transverse vibration of the cantilever beam and the transversality condition describing the contact position between the beam and the cylindrical foundation are derived by calculus of variation and hamilton s principle this is a moving boundary problem since the unknown contact position has to be determined as part of the solutions galerkin approximation is used to reduce the partial differential equation to a set of non linear ordinary differential equations the transient amplitudes and the phase planes of the vibration and the contact length are simulated by runge kutta algorithm the effects of initial condition radius parameter of the cylindrical foundation externally static and harmonic excitations are investigated and discussed

the resolution of complex spectral patterns by cochlear implant and normal hearing listeners the differences in spectral shape resolution abilities among cochlear implant ci listeners and between ci and normal hearing nh listeners when listening with the same number of channels 12 was investigated in addition the effect of the number of channels on spectral shape resolution was examined the stimuli were rippled noise signals with various ripple frequency spacings an adaptive 41fc procedure was used to determine the threshold for resolvable ripple spacing which was the spacing at which an interchange in peak and valley positions could be discriminated the results showed poorer spectral shape resolution in ci compared to nh listeners average thresholds of approximately 3000 and 400 hz respectively and wide variability among ci listeners range of approximately 800 to 8000 hz there was a significant relationship between spectral shape resolution and vowel recognition the spectral shape resolution thresholds of nh listeners increased as the number of channels increased from 1 to 16 while the ci listeners showed a performance plateau at 4 6 channels which is consistent with previous results using speech recognition measures these results indicate that this test may provide a measure of ci performance which is time efficient and non linguistic and therefore if verified may provide a useful contribution to the prediction of speech perception in adults and children who use cis

further evidence that fundamental frequency difference limens measure pitch discrimination difference limens for complex tones dlcs that differ in f0 are widely regarded as a measure of periodicity pitch discrimination however because f0 changes are inevitably accompanied by changes in the frequencies of the harmonics dlcs may actually reflect the discriminability of individual components to test this hypothesis dlcs were measured for complex tones the component frequencies of which were shifted coherently upward or downward by f 0 25 37 5 or 50 of the f0 yielding fully harmonic f 0 strongly inharmonic f 25 37 5 or odd harmonic f 50 tones if dlcs truly reflect periodicity pitch discriminability they should be larger worse for inharmonic tones than for harmonic and odd harmonic tones because inharmonic tones have a weaker pitch consistent with this prediction the results of two experiments showed a non monotonic dependence of dlcs on f with larger dlcs for f s of 25 or 37 5 than for f s of 0 or 50 of f0 these findings are consistent with

weak imposition of no slip conditions in finite element methods abstract we begin the analysis of two schemes for imposing weakly the no slip boundary conditions in the navier stokes equations the first is via a lagrange multiplier constraining no tangential velocity on the boundary and the second is via penalization of tangential velocities along boundaries our analysis in this first step is for the stokes problem although the motivation for relaxing this essential boundary condition comes from higher reynolds number flow

the role of excitation pattern temporal fine structure and envelope cues in the discrimination of complex tones the discrimination of bandpass filtered harmonic h from inharmonic i tones produced by shifting all components of the h tones upwards by a fixed amount in hz could be based on shifts in the pattern of ripples in the excitation pattern ep or on changes in the temporal fine structure evoked by the tones the predictions of two computational ep models were compared with measured performance one model used auditory filters with bandwidth values specified by glasberg and moore 1990 hear res 47 103 138 and one used filters that were twice as sharp stimulus variables were passband width fundamental frequency harmonic rank n of the lowest component within the passband component phase cosine or random signal to noise ratio snr and random perturbation in level of each component in the tones while the ep models correctly predicted the lack of an effect of phase and some of the trends in the data as a function of fundamental frequency and n neither model predicted the worsening in perfo

invariant domains and first order continuous finite element approximation for hyperbolic systems we propose a numerical method for solving general hyperbolic systems in any space dimension using forward euler time stepping and continuous finite elements on nonuniform grids the properties of the method are based on the introduction of an artificial dissipation that is defined so that any convex invariant set containing the initial data is an invariant domain for the method the invariant domain property is proved for any hyperbolic system provided a cfl condition holds the solution is also shown to satisfy a discrete entropy inequality for every admissible entropy of the system the method is formally first order accurate in space and can be made high order in time by using strong stability preserving algorithms this technique extends to continuous finite elements the work of d hoff math comp 33 1979 pp 1171 1193 d hoff trans amer math soc 289 1985 pp 591 610 and h frid arch ration mech anal 160 2001 pp 245 269

viscous regularization of the euler equations and entropy principles this paper investigates a general class of viscous regularizations of the compressible euler equations a unique regularization is identified that is compatible with all the generalized entropies la harten et al siam j numer anal 35 1998 pp 2117 2127 and satisfies the minimum entropy principle a connection with a recently proposed phenomenological model by h brenner phys a 370 2006 pp 190 224 is made

the effect of hearing loss on the resolution of partials and fundamental frequency discrimination the relationship between the ability to hear out partials in complex tones discrimination of the fundamental frequency f0 of complex tones and frequency selectivity was examined for subjects with mild to moderate cochlear hearing loss the ability to hear out partials was measured using a two interval task each interval included a sinusoid followed by a complex tone one complex contained a partial with the same frequency as the sinusoid while in the other complex that partial was missing subjects had to indicate the interval in which the partial was present in the complex the components in the complex were uniformly spaced on the erbn number scale performance was generally good for the two edge partials but poorer for the inner partials performance for the latter improved with increasing spacing f0 discrimination was measured for a bandpass filtered complex tone containing low harmonics the equivalent rectangular bandwidth erb of the auditory filter was estimated using the notched noise

second order solutions for the dynamics of a semi infinite cable on a unilateral substrate we present an asymptotic solution of a moving boundary problem which describes the nonlinear oscillations of semi infinite cables resting on an elastic substrate reacting in compression only and subjected to a constant distributed load and to a small harmonic displacement applied to the finite boundary our solution is correct through the second order terms in a smallness parameter which we identify with the amplitude of the harmonic oscillation at the boundary and it complements the first order solution presented in an earlier work the second order analysis confirms the existence of two different regimes in the behaviour of the system one below called subcritical and one above called supercritical a certain critical cutoff excitation frequency in the latter energy is lost by radiation at infinity while in the former this phenomenon does not occur and various resonances are observed instead we show that these two regimes exist at all orders in the expansion parameter and that the cutoff frequency decreases at each order we also perform a limited comparison of our asymptotic results with a numerical solution the two approaches show very good agreement

the perceptual tone noise ratio of merged iterated rippled noises iterated rippled noise irn is constructed by delaying a random noise by d ms adding it back to the same noise and repeating the process iteratively when two irns with the same power but slightly different delays are added together the perceptual tone noise ratio of the merged irn is markedly reduced with respect to that of either of the component irns in this paper the reduction in the perceptual tone noise ratio is measured for irns in which one of the delays is always 16 ms and the other is either 16 0 1 ms or 16 1 1 ms the component irns have the same number of iterations and the number varies across conditions from 4 to 256 the perceptual tone noise ratio is measured using a discrimination matching procedure developed for single irns each merged irn is compared with a range of standard stimuli having varying proportions of a complex tone and a broadband noise patterson et al j acoust soc am 100 3286 3294 1996 for single irns the function relating the signal to noise ratio of the matching standard to the number of iterations in the irn was found to be essentially straight this relationship was explained in terms of the height of the first peak in the autocorrelation of the stimulus wave or by the first peak in the summary autocorrelogram produced by a time domain auditory model for the merged irns in the current experiment the matching point functions are found to have pronounced downward curvature in addition to being well below the function for single irns to account for the reduction in the perceptual tone noise ratio of merged irns the autocorrelation model was extended to include a simple rule for combining adjacent peaks in the autocorrelation function of the wave and the autocorrelogram model was revised to improve the simulation of the loss of phase locking at higher frequencies in the autocorrelogram

acoustic and perceptual effects of amplitude and frequency compression on high frequency speech this study investigated how six different amplification methods influence acoustic properties and subsequently perception of high frequency cues in fricatives that have been processed with conventional full bandwidth amplification or nonlinear frequency compression nfc 12 conditions total amplification methods included linear gain fast slow acting wide dynamic range compression crossed with fixed individualized compression parameters and a method with adaptive time constants twenty one hearing impaired listeners identified seven fricatives in nonsense syllables produced by female talkers for nfc stimuli frequency compressed filters that precisely aligned 1 3 octave bands between input and output were used to quantify effective compression ratio audibility and temporal envelope modulation relative to the input results indicated significant relationships between these acoustic properties each of which contributed significantly to fricative recognition across the entire corpus of stimuli recog

importance of tonal envelope cues in chinese speech recognition recent studies have shown that temporal waveform envelope cues can provide significant information for english speech recognition this study investigated the use of temporal envelope cues in a tonal language mandarin chinese in this study the speech was divided into several frequency analysis bands the amplitude envelope was extracted from each band by half wave rectification and low pass filtering and was used to modulate a noise of the same bandwidth as the analysis band these manipulations preserved temporal and amplitude cues in each frequency band but removed the spectral detail within each band chinese vowels consonants tones and sentences were identified by 12 native chinese speaking listeners with 1 2 3 and 4 noise bands the results showed that the recognition score of vowels consonants and sentences increased monotonically with the number of bands a pattern similar to that observed in english speech recognition in contrast tones were consistently recognized at about 80 correct level independent of the number of bands this high level of tone recognition produced a significant difference in the open set sentence recognition between chinese 11 0 and english 2 9 for the one band condition where no spectral information was available the data also revealed that with primarily temporal cues the falling rising tone tone 3 and the falling tone tone 4 were more easily recognized than the flat tone tone 1 and the rising tone tone 2 this differential pattern in tone recognition resulted in a similar pattern in word recognition words having either tone 3 or 4 were more likely to be recognized while words having tone 1 and 2 were not the quantitative role of tones in chinese speech recognition was further explored using a power function model and found to play a significant role in relating phoneme recognition to sentence recognition

modeling the dynamics of a vibrating string with a finite distributed unilateral constraint application to the sitar the free vibration response of an ideal string impacting a distributed parabolic obstacle located at its boundary has been analyzed the goal being to understand and simulate a sitar string the portion of the string in contact with the obstacle is governed by a different partial differential equation pde from the free portion represented by the classical string equation these two pdes and corresponding boundary conditions along with the transversality condition that governs the dynamics of the moving boundary are obtained using hamilton s principle a galerkin approximation is used to convert them into a system of nonlinear ordinary differential equations with lower mode shapes parametrized with respect to the location of the moving boundary as basis functions this system is solved numerically and the behavior of the string studied from simulations the advantages and disadvantages of the proposed method are discussed in comparison to the penalty approach for simulating wrapping contacts simulati

temporal envelope and fine structure cues for speech intelligibility this paper describes a number of listening experiments to investigate the relative contribution of temporal envelope modulations and fine structure to speech intelligibility the amplitude envelopes of 24 1 4 oct bands covering 100 6400 hz were processed in several ways e g fast compression in order to assess the importance of the modulation peaks and troughs results for 60 normal hearing subjects show that reduction of modulations by the addition of noise is more detrimental to sentence intelligibility than the same degree of reduction achieved by direct manipulation of the envelope in some cases the benefit in speech reception threshold srt is almost 7 db two crossover levels can be defined in dividing the temporal envelope into two equally important parts the first crossover level divides the envelope into two perceptually equal parts removing modulations either x db below or above that level yields the same intelligibility score the second crossover level divides the envelope into two acoustically equal peak and trough parts the perceptual level is 9 12 db higher than the acoustic level indicating that envelope peaks are perceptually more important than troughs further results showed that 24 intact temporal speech envelopes with noise fine structure retain perfect intelligibility in general for the present type of signal manipulations no one to one relation between the modulation transfer function and the intelligibility scores could be established

the effect of smoothing filter slope and spectral frequency on temporal speech information it is known that information contained within the filter skirts can provide cues important to speech intelligibility however the role of filter slope during temporal smoothing has received little attention in experiment 1 smoothing filter slope angle was found to have a large effect on the intelligibility of sentences represented by three amplitude modulated sinusoids in experiment 2 the use of temporal cues above 16hz was examined across various regions of the spectrum when increases in rate were presented to individual spectral bands intelligibility only increased when presented in the higher spectral region this result suggests a greater reliance on higher rate cues in this region however intelligibility was greatest when these cues were distributed across the spectrum indicating that their effective use is not restricted solely to this region

searching for the time constant of neural pitch extraction multichannel auditory models have been repeatedly used to explain many aspects of human pitch perception among the most successful ones are models where pitch is estimated based on an analysis of periodicity in the simulated auditory nerve firing this periodicity analysis is typically implemented as a running autocorrelation i e the autocorrelation is calculated within a temporal window which is shifted along the time axis the window was suggested to have an exponential decay with time constant estimates between 1 5 and 100 ms the window length determines the minimal integration time of pitch extraction the present experiments are designed to quantify the temporal window of pitch extraction using regular interval noises rins rins were generated by concatenating equal duration noise samples which produce a pitch corresponding to the reciprocal of the sample duration when the samples are identical periodic noise when the samples are independent the stimulus is gaussian noise and produces no pitch using rin stimuli where periodic portions interchange with aperiodic portions it is shown that the temporal window of pitch extraction cannot be modeled using a single time constant but that the size of the temporal window depends on the pitch itself

coding of vowellike signals in cochlear implant listeners neural population interactions resulting from excitation overlap in multi channel cochlear implants ci may cause blurring of the internal auditory representation of complex sounds such as vowels in experiment i confusion matrices for eight german steady state vowellike signals were obtained from seven ci listeners identification performance ranged between 42 and 74 correct on the basis of an information transmission analysis across all vowels pairs of most and least frequently confused vowels were selected for each subject in experiment ii vowel masking patterns vmps were obtained using the previously selected vowels as maskers the vmps were found to resemble the electrical vowel spectra to a large extent indicating a relatively weak effect of neural population interactions correlation between vowel identification data and vmp spectral similarity measured by means of several spectral distance metrics showed that the ci listeners identified the vowels based on differences in the betwe

on the convergence of shock capturing streamline diffusion finite element methods for hyperbolic conservation laws we extend our previous analysis of streamline diffusion finite element methods for hyperbolic systems of conservation laws to include a shock capturing term adding artificial viscosity depending on the local absolute value of the residual of the finite element solution and the meh size with this term present we prove a maximum norm bound for finite element solutionsof burgers equation an thus complete an earlier convergence proof for this equation we further prove using entropy variables that a strong limit of finite element solutions is a weak solution of the system of conservation laws and satisfies the entropy inequality asociated with the entropy variables results of some numerical experiments for the time dependent compressible euler equations in two dimensions are also reported

the intelligibility of speech with holes in the spectrum the intelligibility of speech having either a single hole in various bands or having two holes in disjoint or adjacent bands in the spectrum was assessed with normal hearing listeners in experiment 1 the effect of spectral holes on vowel and consonant recognition was evaluated using speech processed through six frequency bands and synthesized as a sum of sine waves results showed a modest decrease in vowel and consonant recognition performance when a single hole was introduced in the low and high frequency regions of the spectrum respectively when two spectral holes were introduced vowel recognition was sensitive to the location of the holes while consonant recognition remained constant around 70 correct even when the middle and high frequency speech information was missing the data from experiment 1 were used in experiment 2 to derive frequency importance functions based on a least squares approach the shapes of the frequency importance functions were found to be different for consonants and vowels in agreement with the notion that different cues are used by listeners to identify consonants and vowels for vowels there was unequal weighting across the various channels while for consonants the frequency importance function was relatively flat suggesting that all bands contributed equally to consonant identification

moderate cochlear hearing loss leads to a reduced ability to use temporal fine structure information the ability of normally hearing and hearing impaired subjects to use temporal fine structure information in complex tones was measured subjects were required to discriminate a harmonic complex tone from a tone in which all components were shifted upwards by the same amount in hz in a three alternative forced choice task the tones either contained five equal amplitude components non shaped stimuli or contained many components but were passed through a fixed bandpass filter to reduce excitation pattern changes shaped stimuli components were centered at nominal harmonic numbers n 7 11 and 18 for the shaped stimuli hearing impaired subjects performed much more poorly than normally hearing subjects with most of the former scoring no better than chance when n 11 or 18 suggesting that they could not access the temporal fine structure information performance for the hearing impaired subjects was significantly improved for the non shaped stimuli presumably because they could benefit from spectral cues it is proposed that normal hearing subjects can use temporal fine structure information provided the spacing between fine structure peaks is not too small relative to the envelope period but subjects with moderate cochlear hearing loss make little use of temporal fine structure information for unresolved components

exploring the temporal mechanism involved in the pitch of unresolved harmonics this paper continues a line of research initiated by kaernbach and demany j acoust soc am 104 2298 2306 1998 who employed filtered click sequences to explore the temporal mechanism involved in the pitch of unresolved harmonics in a first experiment the just noticeable difference jnd for the fundamental frequency f0 of high pass filtered and low pass masked click trains was measured with f0 100 to 250 hz and the cut frequency 0 5 to 6 khz being varied orthogonally the data confirm the result of houtsma and smurzynski j acoust soc am 87 304 310 1990 that a pitch mechanism working on the temporal structure of the signal is responsible for analyzing frequencies higher than ten times the fundamental using high pass filtered click trains however the jnd for the temporal analysis is at 1 2 as compared to 2 3 found in studies using band pass filtered stimuli two further experiments provide evidence that the pitch of this stimulus can convey musical information a fourth experiment replicates the finding of kaernbach and demany on first and second order regularities with a cut frequency of 2 khz and extends the paradigm to binaural aperiodic click sequences the result suggests that listeners can detect first order temporal regularities in monaural click streams as well as in binaurally fused click streams

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

use of high rate envelope speech cues and their perceptually relevant dynamic range for the hearing impaired the ability of hearing impaired hi listeners to use high rate envelope information in a competing talker situation was assessed in experiment 1 signals were tone vocoded and the cutoff frequency fc of the envelope extraction filter was either 50 hz e filter or 200 hz p filter the channels for which the p or e filter was used were varied intelligibility was higher with the p filter regardless of whether it was used for low or high center frequencies performance was best when the p filter was used for all channels experiment 2 explored the dynamic range over which hi listeners made use of high rate cues in each channel of a vocoder the envelope extracted using fc 16 hz was replaced by the envelope extracted using fc 300 hz either at the peaks or valleys with a parametrically varied switching threshold for a target to background ratio of 5 db changes in speech intelligibility occurred mainly when the switching threshold was between 8 and 8 db relative to the channel root mean square level this range is similar in width to but about 3 db higher in absolute level than that found for normal hearing listeners despite the reduced dynamic range of the hi listeners

effects of spectral smearing and temporal fine structure degradation on speech masking release this study assessed the effects of spectral smearing and temporal fine structure tfs degradation on masking release mr the improvement in speech identification in amplitude modulated compared to steady noise observed for normal hearing listeners syllables and noise stimuli were processed using either a spectral smearing algorithm or a tone excited vocoder the two processing schemes simulated broadening of the auditory filters by factors of 2 and 4 simulations of the early stages of auditory processing showed that the two schemes produced comparable excitation patterns however fundamental frequency f0 information conveyed by tfs was degraded more severely by the vocoder than by the spectral smearing algorithm both schemes reduced mr but for each amount of spectral smearing the vocoder produced a greater reduction in mr than the spectral smearing algorithm consistent with the effects of each scheme on f0 representation moreover the effects of spectral smearing on mr produced by the two schemes were different for manner and voicing finally mr data for listeners with moderate hearing loss were well matched by mr data obtained for normal hearing listeners with vocoded stimuli suggesting that impaired frequency selectivity alone may not be sufficient to account for the reduced mr observed for hearing impaired listeners