efficient numerical schemes for viscoplastic avalanches part 2 the 2d case abstract this paper deals with the numerical resolution of a shallow water viscoplastic flow model viscoplastic materials are characterized by the existence of a yield stress below a certain critical threshold in the imposed stress there is no deformation and the material behaves like a rigid solid but when that yield value is exceeded the material flows like a fluid in the context of avalanches it means that after going down a slope the material can stop and its free surface has a non trivial shape as opposed to the case of water newtonian fluid the model involves variational inequalities associated with the yield threshold finite volume schemes are used together with duality methods namely augmented lagrangian and bermudez moreno to discretize the problem to be able to accurately simulate the stopping behavior of the avalanche new schemes need to be designed involving the classical notion of well balancing in the present context it needs to be extended to take into account the viscoplastic nature of the material as well as general bottoms with wet dry fronts which are encountered in geophysical geometries here we derive such schemes in 2d as the follow up of the companion paper treating the 1d case numerical tests include in particular a generalized 2d benchmark for bingham codes the bingham couette flow with two non zero boundary conditions on the velocity and a simulation of the avalanche path of taconnaz in chamonix mont blanc to show the usability of these schemes on real topographies from digital elevation models dem

quantifying the effects of fast acting compression on the envelope of speech fast acting compression has several effects on the envelope of speech signals especially when a target and a background are mixed together three measures of the envelope are described which can be used to quantify these changes 1 within signal modulation correlation or coherence the degree of correlation or coherence of the envelope on a db scale of a single source across different frequency regions which is reduced by fast acting compression 2 fidelity of envelope shape the degree to which the envelope shape of the target speech in different frequency channels is preserved following compression and 3 across signal modulation correlation or coherence asmc the extent to which the target and background acquire a common component of modulation when they are compressed together which becomes greater in absolute value more negative when the target and background are compressed together values of these measures are presented and compared with intelligibility scores obtained using stimuli processed with a noise vocoder so as to preserve mainly envelope cues in a limited number of frequency channels the results suggest that the dominant factor affecting intelligibility is asmc

the influence of age and high frequency hearing loss on sensitivity to temporal fine structure at low frequencies l sensitivity to temporal fine structure tfs at low frequencies may be adversely affected by hearing loss at high frequencies even when absolute thresholds at low frequencies are within the normal range however in several studies suggesting this the effects of hearing loss and age were confounded here interaural phase discrimination ipd thresholds for pure tones at 500 and 750 hz were measured for 39 subjects with ages from 61 to 83 yr all subjects had near normal audiometric thresholds at low frequencies but thresholds varied across subjects at high frequencies ipd thresholds were correlated with age ipd thresholds for the test frequency of 750 hz were weakly correlated with absolute thresholds at high frequencies but these correlations became non significant when the effect of age was partialed out the results do not confirm that sensitivity to tfs at low frequencies is influenced by hearing loss at high frequencies independently of age

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

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

numerical analysis of a frictional contact problem for viscoelastic materials with long term memory we consider a mathematical model which describes the bilateral contact between a deformable body and an obstacle the process is quasistatic the material is assumed to be viscoelastic with long memory and the friction is modeled with tresca s law the problem has a unique weak solution here we study spatially semi discrete and fully discrete schemes using finite differences and finite elements we show the convergence of the schemes under the basic solution regularity and we derive order error estimates finally we present an algorithm for the numerical realization and simulations for a two dimensional test problem

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

effects of spectro temporal modulation changes produced by multi channel compression on intelligibility in a competing speech task these experiments are concerned with the intelligibility of target speech in the presence of a background talker using a noise vocoder stone and moore j acoust soc am 114 1023 1034 2003 showed that single channel fast acting compression degraded intelligibility but slow compression did not stone and moore j acoust soc am 116 2311 2323 2004 showed that intelligibility was lower when fast single channel compression was applied to the target and background after mixing rather than before and suggested that this was partly due to compression after mixing introducing comodulation between the target and background talkers experiment 1 here showed a similar effect for multi channel compression in experiment 2 intelligibility was measured as a function of the speed of multi channel compression applied after mixing for both eight and 12 channel vocoders with one compressor per channel intelligibility decreased as compression speed increased for the eight channel vocoder a compresso

on the convergence of the bermudez moreno algorithm with constant parameters bermudez moreno 5 presents a duality numerical algorithm for solving variational inequalities of the second kind the performance of this algorithm strongly depends on the choice of two constant parameters assuming a further hypothesis of the inf sup type we present here a convergence theorem that improves on the one presented in 5 we prove that the convergence is linear and we give the expression of the asymptotic error constant and the explicit form of the optimal parameters as a function of some constants related to the variational inequality finally we present some numerical examples that confirm the theoretical results

simple statistical explanation for the localization of energy in nonlinear lattices with two conserved quantities the localization of energy in the discrete nonlinear schrodinger equation is explained with statistical methods the partition function and the entropy of the system are computed for low amplitude initial conditions detailed predictions for the long time solution are derived localized high amplitude excitations absorb a surplus of energy when they emerge as a by product of the production of entropy in the small fluctuations the thermodynamic interpretation of this process applies to many dynamical systems with two conserved quantities

on the fundamental solution of a linearized uehling uhlenbeck equation in this paper we describe the fundamental solution of the equation that is obtained linearizing the uehling uhlenbeck equation around the steady state of kolmogorov type f k k 7 6 detailed estimates on its asymptotics are obtained

noisy spectra long correlations and intermittency in wave turbulence we study the k space fluctuations of the wave action about its mean spectrum in the turbulence of dispersive waves we use a minimal model based on the random phase approximation rpa and derive evolution equations for the arbitrary order one point moments of the wave intensity in the wave number space the first equation in this series is the familiar kinetic equation for the mean wave action spectrum whereas the second and higher equations describe the fluctuations about this mean spectrum the fluctuations exhibit a nontrivial dynamics if some long coordinate space correlations are present in the system as it is the case in typical numerical and laboratory experiments without such long range correlations the fluctuations are trivially fixed at their gaussian values and cannot evolve even if the wave field itself is non gaussian in the coordinate space unlike the previous approaches based on smooth initial k space cumulants the rpa model works even for extreme cases where the k space fluctuations are absent or very large and intermittent we show that any initial non gaussianity at small amplitudes propagates without change toward the high amplitudes at each fixed wave number at each fixed amplitude however the probability distribution function becomes gaussian at large time

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 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

a detailed study on the effects of noise on speech intelligibility a wavelet representation of speech was used to display the instantaneous amplitude and phase within 1 4 octave frequency bands representing the envelope and the carrier within each band adding stationary noise alters the wavelet pattern which can be understood as a combination of three simultaneously occurring subeffects two effects on the wavelet levels one systematic and one stochastic and one effect on the wavelet phases specific types of signal processing were applied to speech which allowed each effect to be either included or excluded the impact of each effect and of combinations on speech intelligibility was measured with cvc s it appeared that the systematic level effect i e the increase of each speech wavelet intensity with the mean noise intensity has the most degrading effect on speech intelligibility which is in accordance with measures such as the modulation transfer function and the speech transmission index however also the introduction of stochastic level fluctuations an

on the significance of phase in the short term fourier spectrum for speech intelligibility this paper investigates the significance of the magnitude or the phase in the short term fourier spectrum for speech intelligibility as a function of the time window length for a wide range of window lengths 1 16 2048 ms two hybrid signals were obtained by a cross wise combination of the magnitude and phase spectra of speech and white noise speech intelligibility data showed the significance of the phase spectrum for longer windows 256 ms and for very short windows 4 ms and that of the magnitude spectrum for medium range window lengths the hybrid signals used in the intelligibility test were analyzed in terms of the preservation of the original narrow band speech envelopes correlations between the narrow band envelopes of the original speech and the hybrid signals show a similar pattern as a function of window length this result illustrates the importance of the preservation of narrow band envelopes for speech intelligibility the observed significance of the phase spectrum in recovering the narrow band envelopes for the long term windows and for the very short term windows is discussed

psychometric functions for level discrimination in cochlearly impaired and normal listeners with equivalent threshold masking the purpose of this study was to determine whether the form of the psychometric functions for level discrimination also known as intensity discrimination measured in a two interval two alternative forced choice paradigm changes with hearing impairment measurements of sensitivity d as a function of delta l 20 log p delta p lp where p is pressure were obtained in six listeners with cochlear impairments five normal listeners tested in the quiet and two masked normal listeners stimuli were chosen to encompass a wide range of conditions and difference limens results show that the d is nearly proportional to delta l over the entire range of stimuli this simple relationship is not changed by cochlear impairments or masking noise these findings indicate that if the transformation from stimulus intensity to decision variable is affected by hearing impairment or masking the change affects both the mean and the standard deviation in the same manner i e the standard deviation may be proportional to the derivative of the transformation as was suggested by zwislocki and jordan j acoust soc am 79 772 780 1986 the results also lend support to the notion that delta l plotted on a logarithmic scale is an appropriate representation of level discrimination performance

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

relations between frequency selectivity temporal fine structure processing and speech reception in impaired hearing frequency selectivity temporal fine structure tfs processing and speech reception were assessed for six normal hearing nh listeners ten sensorineurally hearing impaired hi listeners with similar high frequency losses and two listeners with an obscure dysfunction od tfs processing was investigated at low frequencies in regions of normal hearing through measurements of binaural masked detection tone lateralization and monaural frequency modulation fm detection lateralization and fm detection thresholds were measured in quiet and in background noise speech reception thresholds were obtained for full spectrum and lowpass filtered sentences with different interferers both the hi listeners and the od listeners showed poorer performance than the nh listeners in terms of frequency selectivity tfs processing and speech reception while a correlation was observed between the monaural and binaural tfs processing deficits in the hi listeners no relation was found between tfs processing and frequency selectivity the effect of noise on tfs processing was not larger for the hi listeners than for the nh listeners finally tfs processing performance was correlated with speech reception in a two talker background and lateralized noise but not in amplitude modulated noise the results provide constraints for future models of impaired auditory signal processing

a cross linguistic study of speech modulation spectra languages show systematic variation in their sound patterns and grammars accordingly they have been classified into typological categories such as stress timed vs syllable timed or head complement hc vs complement head ch to date it has remained incompletely understood how these linguistic properties are reflected in the acoustic characteristics of speech in different languages in the present study the amplitude modulation am and frequency modulation fm spectra of 1797 utterances in ten languages were analyzed overall the spectra were found to be similar in shape across languages however significant effects of linguistic factors were observed on the am spectra these differences were magnified with a perceptually plausible representation based on the modulation index a measure of the signal to noise ratio at the output of a logarithmic modulation filterbank the maximum value distinguished between hc and ch languages with the exception of turkish while the exact frequency of this ma

modeling speech intelligibility in quiet and noise in listeners with normal and impaired hearing the speech intelligibility index sii is an often used calculation method for estimating the proportion of audible speech in noise for speech reception thresholds srts measured in normally hearing listeners using various types of stationary noise this model predicts a fairly constant speech proportion of about 0 33 necessary for dutch sentence intelligibility however when the sii model is applied for srts in quiet the estimated speech proportions are often higher and show a larger inter subject variability than found for speech in noise near normal speech levels 65 db sound pressure level spl the present model attempts to alleviate this problem by including cochlear compression it is based on a loudness model for normally hearing and hearing impaired listeners of moore and glasberg 2004 hear res 188 70 88 it estimates internal excitation levels for speech and noise and then calculates the proportion of speech above noise and threshold using similar spectral weighting as used in the sii the present model and the standard sii were used to predict sii values in quiet and in stationary noise for normally hearing and hearing impaired listeners the present model predicted siis for three listener types normal hearing noise induced and age induced hearing loss with markedly less variability than the standard sii

speech reception in quiet and in noisy conditions by individuals with noise induced hearing loss in relation to their tone audiogram tone thresholds and speech reception thresholds were measured in 200 individuals 400 ears with noise induced hearing loss the speech reception thresholds were measured in a quiet condition and in noise with a speech spectrum at levels of 35 50 65 and 80 dba the tone audiograms could be described by three principal components hearing loss in the regions above 3 khz from 1 to 3 khz and below 1 khz the speech thresholds could be described by two components speech reception in quiet and speech reception in noise at 50 80 dba hearing loss above 1 khz was related to speech reception in noise hearing loss at and below 1 khz to speech reception in quiet the correlation between the speech thresholds in quiet and in noise was only r 0 45 an adequate predictor of the speech threshold in noise the primary factor in the hearing handicap was the pure tone average at 2 and 4 khz pta2 4 r 0 72 the minimum value of the prediction error for any tone audiometric predictor of this speech threshold was 1 2 db standard deviation the prediction could not be improved by taking into account the critical ratio for low frequency noise nor by its upward spread of masking the prediction error is due to measurement error and to a factor common to both ears the latter factor is ascribed to cognitive skill in speech reception hearing loss above 10 to 15 db hl hearing level already shows an effect on the speech threshold in noise a noticeable handicap is found at pta2 4 30 db hl

the effect of interaural delay of the masker on masking level differences in young and old adults diotic sono thresholds and dichotic s pi n pi tau thresholds were measured for young and old adults using a 500 hz pure tone signal and broadband burst masking noise at 37 db spl hz in the dichotic condition both the signal and the masker were phase reversed and the masker was presented with an interaural delay of 0 25 0 5 0 75 1 1 25 1 5 1 75 2 3 or 5 ms masking level differences mlds were determined by subtracting dichotic thresholds from diotic thresholds the sono thresholds for the old subjects did not differ significantly from those for the young subjects however when mlds were plotted as a function of delay the pattern of results differed significantly between young and old subjects this difference in pattern was completely accounted for in terms of a delay line version of durlach s equalization and cancellation ec model n i durlach in foundations of modern auditory theory edited by j v tobias academic new york 1972 b a schneider and p m zurek j acoust soc am 86 1756 1763 1989 by assuming that temporal jitter increases with internal delay in young subjects but that it does not vary with the amount of internal delay in old subjects

long time behavior of solutions of nonlinear classical field equations the example of nls defocusing abstract hamiltonian classical systems are the farthest from variational systems formally they cannot decay because the equation of motion is reversible thus the long time behavior of their solutions is naturally described by statistical methods for instance a large assembly of interacting particles in a box follow the gibbs boltzmann statistics as far as average properties are concerned as noticed already by jeans this statistical approach does not hold for the long time behavior of classical nonlinear field equations because of the infinite heat capacity of a classical field thus nonlinear and nonintegrable classical fields cannot relax for long times toward an ill defined thermal equilibrium the specific example considered here is the long time dynamics of solutions of the nonlinear schrodinger equation in the defocusing case i show under some assumptions that for long times a cascade of energy toward smaller and smaller scales introduces a kind of dissipation in a system that is formally reversible and i give the scaling laws for this besides this relaxation by weak interaction between phonons there is also a relaxation of localized vortical structures emitting phonons

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

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

efficient numerical schemes for viscoplastic avalanches part 1 the 1d case this paper deals with the numerical resolution of a shallow water viscoplastic flow model viscoplastic materials are characterized by the existence of a yield stress below a certain critical threshold in the imposed stress there is no deformation and the material behaves like a rigid solid but when that yield value is exceeded the material flows like a fluid in the context of avalanches it means that after going down a slope the material can stop and its free surface has a non trivial shape as opposed to the case of water newtonian fluid the model involves variational inequalities associated with the yield threshold finite volume schemes are used together with duality methods namely augmented lagrangian and bermudez moreno to discretize the problem to be able to accurately simulate the stopping behavior of the avalanche new schemes need to be designed involving the classical notion of well balancing in the present context it needs to be extended to take into account the viscoplastic nature of the material as well as general bottoms with wet dry fronts which are encountered in geophysical geometries we derived such schemes and numerical experiments are presented to show their performances

frequency weighting functions for broadband speech as estimated by a correlational method the relative contributions of various regions of the frequency spectrum to speech recognition were assessed with a correlational method k a doherty and c w turner j acoust soc am 100 3769 3773 1996 the speech materials employed were the 258 item set of the nonsense syllable test the speech was filtered into four frequency bands and a random level of noise was added to each band on each trial a point biserial correlation was computed between the signal to noise ratio in each band on the trials and the listener s responses and these correlations were then taken as estimates of the relative weights for each frequency band when the four bands were presented separately the correlations for each band were approximately equal however when the four bands were presented in combination the correlations were quite different from one another implying that in the broadband case listeners relied much more on some bands than on others it is hypothesized that these differences reflect the way in which listeners combine and attend to speech information across various frequency regions the frequency weighting functions as determined by this method were highly similar across all subjects suggesting that normal hearing listeners use similar frequency weighting strategies in recognizing speech

numerical solution of cavitation problems in lubrication abstract a combination of the method of characteristics and of the finite element method is applied to solve numerically a stationary free boundary problem of hydrodynamic lubrication with cavitation the discretization leads to a system of nonlinear equations to solve it a duality iterative algorithm is used numerical results are presented

investigation of perceptual constancy in the temporal envelope domain the ability to discriminate complex temporal envelope patterns submitted to temporal compression or expansion was assessed in normal hearing listeners an xab matching to sample procedure was used x the reference stimulus is obtained by applying the sum of two inharmonically related sinusoids to a broadband noise carrier a and b are obtained by multiplying the frequency of each modulation component of x by the same time expansion compression factor 0 35 2 83 for each trial a or b is a time reversed rendering of x and the listeners task is to choose which of the two is matched by x overall the results indicate that discrimination performance degrades for increasing amounts of time expansion compression i e when departs from 1 regardless of the frequency spacing of modulation components and the peak to trough ratio of the complex envelopes an auditory model based on envelope extraction followed by a memory limited template matching process accounted for results obtained withou

duality methods for solving variational inequalities abstract methods of maximal monotone operators are used in order to study from a general point of view duality numerical algorithms for solving variational inequalities with classical algorithms such as uzawa s method for the standard and augmented lagrangian this paper presents some new algorithms which appear to have very good numerical performances

intensity discrimination of gaussian windowed tones indications for the shape of the auditory frequency time window the just noticeable difference in intensity jnd i was measured for 1 khz tones with a gaussian shaped envelope as a function of their spectro temporal shape the stimuli with constant energy and a constant product of bandwidth and duration ranged from a long duration narrow band tone to a short duration broadband click the jnd i was measured in three normal hearing listeners at sensation levels of 0 10 20 and 30 db in 35 db a spl pink noise at intermediate sensation levels jnd i depends on the spectro temporal shape at the extreme shapes tones and clicks intensity discrimination performance is best whereas at intermediate shapes the jnd i is larger similar results are observed at a higher overall sound level and at a higher carrier frequency the maximum jnd i is observed for stimuli with an effective bandwidth of about 1 3 octave and an effective duration of 4 ms at 1 khz 1 ms at 4 khz a generalized multiple window model is proposed that assumes that the spectro temporal domain is partitioned into internal auditory frequency time windows the model predicts that intensity discrimination thresholds depend upon the number of windows excited by a signal jnd i is largest for stimuli covering one window

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

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

effects of amplitude nonlinearity on phoneme recognition by cochlear implant users and normal hearing listeners it is widely assumed that the proper transformation of acoustic amplitude to electric amplitude is a critical factor affecting speech recognition in cochlear implant users and normal hearing listeners a four channel noise band speech processor was implemented reducing spectral information to four bands a power law transformation was applied to the amplitude mapping stage in the speech processor design and the exponent of the power function varied from a strongly compressive p 0 05 to a weakly compressive p 0 75 for implant listeners and from 0 3 to 3 0 for acoustic listeners results for implants showed that the best performance was achieved with an exponent of about 0 2 and performance gradually deteriorated when either more compressive or less compressive exponents were applied the loudness growth functions of the four activated electrodes in each subject were measured and those data were well fit by a power function with a mean exponent of 2 72 the results indicated that the best performance was achieved when the normal loudness growth was restored for acoustic listeners results were similar to those observed with cochlear implant listeners except that best performance was achieved with no amplitude nonlinearity p 1 0 the similarity of results in both acoustic and electric stimulation indicated that the performance deterioration observed for extreme nonlinearity was due to similar perceptual effects the function relating amplitude mapping exponent and performance was relatively flat indicating that phoneme recognition was only mildly affected by amplitude nonlinearity

integration across frequency bands for consonant identification a comparison of the predictions of models of integration to data on the reception of consonants filtered into a variety of frequency bands is reported new data on the consonant identification are presented three experiments were conducted testing the following bands experiment i 0 2100 hz and 2100 4500 hz experiment ii 0 700 hz combined with 700 1400 1400 2100 2100 2800 and 2800 4500 hz experiment iii all combinations of 700 1400 1400 2100 2100 2800 and 2800 4500 hz the predictions of four models fletcher s speech and hearing in communication van nostrand new york 1950 independent errors model massaro s fuzzy logical model of perception proc int congress of phonetic sciences stockholm vol 3 pp 106 113 1987 and braida s pre labelling and post labelling models of integration q j exp psychol a 43 647 677 1991 were compared in terms of their ability to predict combined band scores at least two models were capable of predicting performance for each combined band condition for experiment i all models were able to make satisfactory predictions for experiment ii a variant of the pre labelling model was able to make satisfactory predictions for experiment iii no model was able to make satisfactory predictions but the fuzzy logical model of perception and a variant of the pre labelling model made relatively good predictions thus the ability of the models to predict performance depended more on whether the condition included the lowest frequency band than on the adjacency or frequency separation

measuring the threshold for speech reception by adaptive variation of the signal bandwidth ii hearing impaired listeners in a previous study noordhoek et al j acoust soc am 105 2895 2902 1999 an adaptive test was developed to determine the speech reception bandwidth threshold srbt i e the width of a speech band around 1 khz required for a 50 intelligibility score in this test the band filtered speech is presented in complementary bandstop filtered noise in the present study the performance of 34 hearing impaired listeners was measured on this srbt test and on more common srt speech reception threshold tests namely the srt in quiet the standard srt in noise standard speech spectrum and the spectrally adapted srt in noise fitted to the individual s dynamic range the aim was to investigate to what extent the performance on these tests could be explained simply from audibility as estimated with the sii speech intelligibility index model or require the assumption of suprathreshold deficits for most listeners an elevated srt in quiet or an elevated standard srt in noise could be explained on the basis of audibility for the spectrally adapted srt in noise and especially for the srbt the data of most listeners could not be explained from audibility suggesting that the effects of suprathreshold deficits may be present possibly such a deficit is an increased downward spread of masking

a mean field statistical theory for the nonlinear schrodinger equation abstract a statistical model of self organization in a generic class of one dimensional nonlinear schrodinger nls equations on a bounded interval is developed the main prediction of this model is that the statistically preferred state for such equations consists of a deterministic coherent structure coupled with fine scale random fluctuations or radiation the model is derived from equilibrium statistical mechanics by using a mean field approximation of the conserved hamiltonian and particle number l2 norm squared for finite dimensional spectral truncations of the nls dynamics the continuum limits of these approximated statistical equilibrium ensembles on finite dimensional phase spaces are analyzed holding the energy and particle number at fixed finite values the analysis shows that the coherent structure minimizes total energy for a given value of particle number and hence is a solution to the nls ground state equation and that the remaining energy resides in gaussian fluctuations equipartitioned over wave numbers some results of direct numerical integration of the nls equation are included to validate empirically these properties of the most probable states for the statistical model moreover a theoretical justification of the mean field approximation is given in which the approximate ensembles are shown to concentrate on the associated microcanonical ensemble in the continuum limit

a generalized duality method for solving variational inequalities applications to some nonlinear dirichlet problems in 2 bermudez and moreno introduced a duality algorithm for the numerical solution of variational inequalities this algorithm is based on some properties of the yosida regularization of maximal monotone operators the performances of this algorithm strongly depend on the choice of two constant parameters a generalization of the algorithm with automatic choice of parameters was discussed in 13 where the constant parameters were replaced by scalar functions thus improving the convergence of the algorithm in this article we present a generalization of the bermudez moreno algorithm that allows the use of very general operators as parameters extending some of the results in 2 13 and 14 as a particular case we analyze the use of scalar and matrix valued parameters in a lp m context we apply the results developed to some boundary value problems involving the p laplacian operator where it is shown that the use of matrix valued parameters improves the convergence of the algorithm

a numerical algorithm for a signorini problem associated with maxwell norton materials by using generalized newton s methods abstract the goal of this work is to present new numerical approaches to obtain computationally efficient adaptive procedures to solve contact problems associated with maxwell norton materials we will describe in detail the numerical methods used to solve such problems and derive iterative algorithms to validate the proposed algorithms we will present some test examples with known analytical solutions finally we will compare the efficiency of the algorithms presented in this paper with others previously proposed by the authors

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

the negative effect of amplitude compression in multichannel hearing aids in the light of the modulation transfer function the article deals with the question of why multichannel amplitude compression appears to have a negative rather than a positive effect on speech intelligibility by hearing impaired listeners it is argued that the small time constants of amplitude compression diminish the temporal as well as the spectral contrasts in the speech signal according to the modulation transfer function concept this results in reduced intelligibility scores experimental evidence is reviewed indicating that the following two arguments in favor of amplitude compression in case of sensorineural hearing loss are not valid 1 to compensate for the effects of loudness recruitment and 2 to get weak consonants above threshold the author concludes that in multichannel hearing aids automatic gain control with time constants of 0 25 0 5 s should be given preference to amplitude compression

use of a compound approach to derive auditory filter wide frequency importance functions for vowels and consonants speech recognition in noise presumably relies on the number and spectral location of available auditory filter outputs containing a relatively undistorted view of local target signal properties the purpose of the present study was to estimate the relative weight of each of the 30 auditory filter wide bands between 80 and 7563 hz because previous approaches were not compatible with this goal a technique was developed similar to the hole approach the weight of a given band was assessed by comparing intelligibility in two conditions differing in only one aspect the presence or absence of the band of interest in contrast to the hole approach however random gaps were also created in the spectrum these gaps were introduced to render the auditory system more sensitive to the removal of a single band and their location was randomized to provide a general view of the weight of each band i e irrespective of the location of information elsewhere in the spectrum frequency weighting functions derived using this technique confirmed the main contribution of the 400 2500 hz frequency region however they revealed a complex microstructure contrasting with the bell curve shape typically reported

sentence reception in noise from one versus two sources effects of aging and hearing loss sentence reception thresholds srts and babble detection thresholds bdts were measured for signals presented from loudspeakers located at 0 deg and 90 deg azimuth in addition the s b ratios necessary to achieve srt in the presence of a 12 talker babble were determined under three conditions in which the speech and babble were presented from the same or spatially separated loudspeakers in the first experiment normal hearing subjects were tested on two occasions the results were highly reliable with standard errors of the test retest differences of less than 1db for all conditions in the second experiment srts bdts and s b ratios were measured for three normal hearing groups 39 40 54 and 55 years and for a fourth group with presbycusis 55 years presbycusics had higher srts and bdts than all other groups regardless of age and the oldest normals had higher thresholds than younger subjects presbycusics required higher s b ratios than any of the normal groups and the oldest normal group needed a higher s b ratio compared to the youngest group all groups demonstrated a significant advantage in s b ratio when the sentences and babble sources were spatially separated compared to when they were presented from the same loudspeaker there was no significant difference in the magnitude of advantage due to spatial separation of speech and noise between the normal groups regardless of age but the presbycusics had significantly smaller advantages than all normals these findings suggest that aging alone does not impair the ability to take advantage of the spatial separation of speech and noise and that the speech reception problems of aging individuals involve various components that depend upon the nature of the listening task

breakdown of weak turbulence and nonlinear wave condensation abstract the formation of a large scale coherent structure a condensate as a result of the long time evolution of the initial value problem of a classical partial differential nonlinear wave equation is considered we consider the nonintegrable and unforced defocusing nonlinear schrodinger nls equation as a representative model in spite of the formal reversibility of the nls equation the nonlinear wave exhibits an irreversible evolution towards a thermodynamic equilibrium state the equilibrium state is characterized by a homogeneous solution condensate with small scale fluctuations superposed uncondensed particles which store the information necessary for time reversal we analyze the evolution of the cumulants of the random wave as originally formulated by d j benney and p g saffman d j benney p g saffman proc roy soc london a 289 1966 301 and a c newell a c newell rev geophys 6 1968 1 this allows us to provide a self consistent weak turbulence theory of the condensation process in which the nonequilibrium formation of the condensate is a natural consequence of the spontaneous regeneration of a non vanishing first order cumulant in the hierarchy of the cumulants equations more precisely we show that in the presence of a small condensate amplitude all relevant statistical information is contained in the off diagonal second order cumulant as described by the usual weak turbulence theory conversely in the presence of a high amplitude condensate the diagonal second order cumulants no longer vanish in the long time limit which signals a breakdown of the weak turbulence theory however we show that an asymptotic closure of the hierarchy of the cumulants equations is still possible provided one considers the bogoliubov s basis rather than the standard fourier s free particle basis the nonequilibrium dynamics turns out to be governed by the bogoliubov s off diagonal second order cumulant while the corresponding diagonal cumulants as well as the higher order cumulants are shown to vanish asymptotically the numerical discretization of the nls equation implicitly introduces an ultraviolet frequency cut off the simulations are in quantitative agreement with the weak turbulence theory without adjustable parameters despite the fact that the theory is expected to breakdown nearby the transition to condensation the fraction of condensed particles vs energy is characterized by two distinct regimes for small energies h h c the bogoliubov s regime is established whereas for h h c the small amplitude condensate regime is described by the weak turbulence theory in both regimes we derive coupled kinetic equations that describe the coupled evolution of the condensate amplitude and the incoherent field component the influence of finite size effects and of the dimensionality of the system are also considered it is shown that beyond the thermodynamic limit wave condensation is reestablished in two spatial dimensions in complete analogy with uniform and ideal 2d bose gases

using speech sounds to test functional spectral resolution in listeners with cochlear implants in this study spectral properties of speech sounds were used to test functional spectral resolution in people who use cochlear implants cis specifically perception of the ba da contrast was tested using two spectral cues formant transitions a fine resolution cue and spectral tilt a coarse resolution cue higher weighting of the formant cues was used as an index of better spectral cue perception participants included 19 ci listeners and 10 listeners with normal hearing nh for whom spectral resolution was explicitly controlled using a noise vocoder with variable carrier filter widths to simulate electrical current spread perceptual weighting of the two cues was modeled with mixed effects logistic regression and was found to systematically vary with spectral resolution the use of formant cues was greatest for nh listeners for unprocessed speech and declined in the two vocoded conditions compared to nh listeners ci listeners relied less on formant transitions and more on spectral tilt cue weighting results showed moderately good correspondence with word recognition scores the current approach to testing functional spectral resolution uses auditory cues that are known to be important for speech categorization and can thus potentially serve as the basis upon which ci processing strategies and innovations are tested

the effects of age and cochlear hearing loss on temporal fine structure sensitivity frequency selectivity and speech reception in noise temporal fine structure tfs sensitivity frequency selectivity and speech reception in noise were measured for young normal hearing nhy old normal hearing nho and hearing impaired hi subjects two measures of tfs sensitivity were used the tfs lf test interaural phase difference discrimination and the tfs2 test discrimination of harmonic and frequency shifted tones these measures were not significantly correlated with frequency selectivity after partialing out the effect of audiometric threshold suggesting that insensitivity to tfs cannot be wholly explained by a broadening of auditory filters the results of the two tests of tfs sensitivity were significantly but modestly correlated suggesting that performance of the tests may be partly influenced by different factors the nho group performed significantly more poorly than the nhy group for both measures of tfs sensitivity but not frequency selectivity suggesting that tfs sensitivity declines with age in the absence of elevated audiometric thresholds or broadened auditory filters when the effect of mean audiometric threshold was partialed out speech reception thresholds in modulated noise were correlated with tfs2 scores but not measures of frequency selectivity or tfs lf test scores suggesting that a reduction in sensitivity to tfs can partly account for the speech perception difficulties experienced by hearing impaired subjects

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

the influence of non spatial factors on measures of spatial release from masking this study tested the hypothesis that the reduction in spatial release from masking srm resulting from sensorineural hearing loss in competing speech mixtures is influenced by the characteristics of the interfering speech a frontal speech target was presented simultaneously with two intelligible or two time reversed unintelligible speech maskers that were either colocated with the target or were symmetrically separated from the target in the horizontal plane the difference in srm between listeners with hearing impairment and listeners with normal hearing was substantially larger for the forward maskers deficit of 5 8 db than for the reversed maskers deficit of 1 6 db this was driven by the fact that all listeners regardless of hearing abilities performed similarly and poorly in the colocated condition with intelligible maskers the same conditions were then tested in listeners with normal hearing using headphone stimuli that were degraded by noise vocoding reducing the number of available spectral channels systematically reduced the measured srm and again more so for forward reduction of 3 8 db than for reversed speech maskers reduction of 1 8 db the results suggest that non spatial factors can strongly influence both the magnitude of srm and the apparent deficit in srm for listeners with impaired hearing

nonequilibrium statistical behavior of nonlinear schrodinger equations this paper explores the equilibrium and near equilibrium statistical behavior of nonlinear schrodinger equations through numerical simulations the investigation centers on a class of finite dimensional hamiltonian systems obtained by discretizing nls equations defined on a bounded interval with saturated focusing nonlinearities these nonlinear systems govern a wave dynamics that is nonintegrable and nonsingular and for which typical solutions are composed of large scale coherent structures interacting within a background of small scale fluctuations the equilibrium statistics of such a system is most naturally defined by a gibbs distribution that is canonical in energy and microcanonical in particle number by sampling these equilibrium distributions using a metropolis type algorithm the typical excitations of the system are determined over the full range of system parameters nonequilibrium relaxations from initial perturbations of these equilibrium distributions are then computed by means of linear response theory the computations reveal a surprising dependence of the qualitative properties of relaxation toward equilibrium on the temperature and the nonlinearity on the one hand a simple exponential decay to equilibrium is observed for sufficiently high temperature or sufficiently strong nonlinearity for low temperature and moderate nonlinearity on the other hand the system rapidly adjusts to a metaequilibrium state that equilibrates extremely slowly this two stage thermalization may be related to quasiperiodicity in the underlying dynamics indeed the metaequilibria are found to occur in a range of system parameters for which approximately quasiperiodic solutions are prevalent in equilibrium

nonlinear frequency compression influence of start frequency and input bandwidth on consonant and vowel recognitiona by varying parameters that control nonlinear frequency compression nfc this study examined how different ways of compressing inaudible mid and or high frequency information at lower frequencies influences perception of consonants and vowels twenty eight listeners with mild to moderately severe hearing loss identified consonants and vowels from nonsense syllables in noise following amplification via a hearing aid simulator low pass filtering and the selection of nfc parameters fixed the output bandwidth at a frequency representing a moderately severe 3 3 khz group ms or a mild to moderate 5 0 khz group mm high frequency loss for each group n 14 effects of six combinations of nfc start frequency sf and input bandwidth by varying the compression ratio cr were examined for both groups the 1 6 khz sf significantly reduced vowel and consonant recognition especially as cr increased whereas recognition was generally unaffected if sf increased at the expense of a higher cr vowel recognition detriments for group ms were moderately correlated with the size of the second formant frequency shift following nfc for both groups significant improvement 33 50 with nfc was confined to final s and z and to some vcv tokens perhaps because of listeners limited exposure to each setting no set of parameters simultaneously maximized recognition across all tokens

use of a correlational method to estimate a listener s weighting function for speech the purpose of this study was to determine if it is feasible to use the correlational method lutfi 1995 richards and zhu 1994 to estimate how listeners use or weight the information contained within various frequency bands of speech three naturally spoken vowel consonant vowel vcv syllables aba aga and ada were presented monaurally to listeners each of the vcv waveforms were filtered into three separate frequency bands i e low mid and high each band was then independently and randomly degraded at various signal to noise s n levels 7 5 3 1 or 1 on each trial listeners were asked to identify the vcv that was presented to them for each trial the s n level of each frequency band the stimulus that was presented and the listener s responses were all recorded and stored in a file from this trial by trial data a point biserial correlation was computed between the listener s response correct or incorrect identification and the degradation within each frequency band the stronger the correlation the greater influence that given frequency band had on the listener s performance on the task from these relations it was shown that it is possible to obtain a listener s weighting function for speech results showed that although most listeners weighted the mid frequency band the greatest several of the listeners used different weighting strategies to perform the task several methodological issues are discussed in regard to improving the future application of the correlational method to speech

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

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

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

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

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

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

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