a composite auditory model for processing speech sounds a composite inner ear model containing the middle ear basilar membrane bm hair cells and hair cell nerve fiber synapses is presented the model incorporates either a linear bm stage or a nonlinear one the model with the nonlinear bm generally shows a high degree of success in reproducing the qualitative aspects of experimentally recorded cat auditory nerve fiber responses to speech in modeling fiber population responses to speech and speech in noise it was found that the bm nonlinearity allows bands of fibers in the model to synchronize strongly to a common spectral peak in the stimulus a cross channel correlation algorithm has been devised to further process the model s population outputs with output from the nonlinear bm model the cross channel correlation values are appreciably reduced only at those channels whose cfs coincide with the formant frequencies this observation also holds to a large extent for noisy speech

approximation by positive operators abstract if a b ic is a normal operator where b c are hermitian then in each unitarily invariant norm the positive part of b is a best approximation to a from the class of positive operators this generalizes results proved earlier by p r halmos t ando and r bouldin for special norms some related results are included

conewise linear systems non zenoness and observability conewise linear systems are dynamical systems in which the state space is partitioned into a finite number of nonoverlapping polyhedral cones on each of which the dynamics of the system is described by a linear differential equation this class of dynamical systems represents a large number of piecewise linear systems most notably linear complementarity systems with the p property and their generalizations to affine variational systems which have many applications in engineering systems and dynamic optimization the challenges of dealing with this type of hybrid system are due to two major characteristics mode switchings are triggered by state evolution and states are constrained in each mode in this paper we first establish the absence of zeno states in such a system based on this fundamental result we then investigate and relate several state observability notions short time and t time or finite time local global observability for the short time observability notions constructive finitely verifiable algebraic both sufficient and necessary conditions are derived due to their long time mode transitional behavior which is very difficult to predict only partial results are obtained for the t time observable states nevertheless we completely resolve the t time local observability for the bimodal conewise linear system for finite t and provide numerical examples to illustrate the difficulty associated with the long time observability

auditory filter nonlinearity in mild moderate hearing impairment sensorineural hearing loss has frequently been shown to result in a loss of frequency selectivity less is known about its effects on the level dependence of selectivity that is so prominent a feature of normal hearing the aim of the present study is to characterize such changes in nonlinearity as manifested in the auditory filter shapes of listeners with mild moderate hearing impairment notched noise masked thresholds at 2 khz were measured over a range of stimulus levels in hearing impaired listeners with losses of 20 50 db growth of masking functions for different notch widths are more parallel for hearing impaired than for normal hearing listeners indicating a more linear filter level dependent filter shapes estimated from the data show relatively little change in shape across level the loss of nonlinearity is also evident in the input output functions derived from the fitted filter shapes reductions in nonlinearity are clearly evident even in a listener with only 20 db hearing loss

relations between auditory functions in impaired hearing relations between auditory functions as expressed by coefficients of correlation were studied for a group of 22 sensorineurally hearing impaired subjects with moderate losses in addition to the audiogram we measured frequency resolution temporal resolution and speech reception in quiet and in noise frequency resolution was derived from masking with comb filtered noise and from the psychophysical tuning curve for both paradigms in simultaneous and in nonsimultaneous masking the critical ratio was also determined temporal resolution was determined with intensity modulated noise and from backward and forward masking all tests were performed at 1000 hz correlations among tests were gathered in a matrix and subjected to a principal components analysis it turned out that tests on frequency resolution form a cluster and are approximately independent of audiometric loss furthermore hearing loss for speech in noise is closely allied to frequency resolution whereas hearing loss for speech in quiet is governed by audiometric loss

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

discovering multiple lyapunov functions for switched hybrid systems in this paper we analyze local asymptotic stability of switched hybrid systems whose subsystems have polynomial vector fields by discovering multiple lyapunov functions in quadratic forms we start with an algebraizable sufficient condition for the existence of quadratic multiple lyapunov functions then since different discrete modes are considered we apply real root classification together with a projection operator to underapproximate this sufficient condition step by step arriving at a set of semialgebraic sets which only involve the coefficients of the pre assumed multiple lyapunov function note that for each step we additionally use the information on the different discrete modes to optimize our intermediate computation results afterward we compute a sample point in the corresponding semialgebraic set for the coefficients resulting in a multiple lyapunov function finally we test our approach on some examples using a prototypical implementation and compare it with the generic quantifier e

short term temporal integration evidence for the influence of peripheral compression thresholds for a 6 5 khz sinusoidal signal temporally centered in a 400 ms broadband noise masker were measured as a function of signal duration for normally hearing listeners and listeners with cochlear hearing loss over a range of masker levels for the normally hearing listeners the slope of the function relating signal threshold to signal duration integration function was steeper at medium masker levels than at low or high levels by a factor of nearly 2 for signal durations between 2 and 10 ms while no significant effect of level was found for signal durations of 20 ms and more no effect of stimulus level was found for the hearing impaired listeners at any signal duration for signal durations greater than 10 ms consistent with many previous studies the slope of the integration function was shallower for the hearing impaired listeners than for the normally hearing listeners however for shorter durations there was no significant difference in slope between the results from the hearing impaired listeners and those from the normally hearing listeners in the high and low level masker conditions a model incorporating a compressive nonlinearity representing the effect of basilar membrane bm compression and a short term temporal integrator postulated to be a more central process can account well for changes in the short term integration function with level if it is assumed that the compression is greater at medium levels than at low or high levels by a factor of about 4 this is in reasonable agreement with physiological measurements of bm compression and with previous psychophysical estimates

recovery from on and off frequency forward masking in listeners with normal and impaired hearing the aim of this study was to investigate the possible mechanisms underlying an effect reported earlier wojtczak m and oxenham a j 2009 j acoust soc am 125 270 281 in normal hearing listeners whereby recovery from forward masking can be slower for off frequency tonal maskers than for on frequency tonal maskers that produce the same amount of masking at a 0 ms masker signal delay to rule out potential effects of confusion between the tonal signal and tonal masker one condition used a noise band forward masker to test whether the effect involved temporal build up another condition used a short duration 30 ms forward masker to test whether the effect is dependent on normal cochlear function conditions were tested in five listeners with sensorineural hearing loss for the 150 ms noise maskers the data from normal hearing listeners replicated the findings from the previous study that used tonal maskers in contrast no significant difference in recovery from on and off frequency masking was observed for the 30 ms tonal maskers in normal hearing listeners or for the 150 ms tonal maskers in hearing impaired listeners overall the results are consistent with a mechanism based on efferent feedback that affects the recovery from forward masking in the normal auditory system

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

temporal integration and compression near absolute threshold in normal and impaired ears the decrease in absolute threshold with increasing stimulus duration often referred to as temporal integration is greater for listeners with normal hearing than for listeners with sensorineural hearing loss it has been suggested that the difference is related to reduced basilar membrane bm compression in the impaired group the present experiment tested this hypothesis by comparing temporal integration and bm compression in normal and impaired ears at low levels absolute thresholds were measured for 4 24 and 44 ms pure tone signals with frequencies f s of 2 and 4 khz the difference between the absolute thresholds for the 4 and 24 ms signals was used as a measure of temporal integration compression near threshold was estimated by measuring the level of a 100 ms off frequency 0 45f s pure tone forward masker required to mask a 44 ms pure tone signal presented at sensation levels of 5 and 10 db there was a significant negative correlation between amount of temporal integration and absolute threshold however there was no correlation between absolute threshold and compression at low levels both normal and impaired ears showed a nearly linear response the results suggest that the differences in integration between normal and impaired ears cannot be explained by differences in bm compression

frequency resolution and discrimination of constant and dynamic tones in normal and hearing impaired listeners frequency resolution and three tasks of frequency discrimination were measured at 500 and 4000 hz in 12 normal and 12 hearing impaired listeners a three interval two alternative forced choice procedure was used frequency resolution was measured with an abbreviated psychoacoustical tuning curve frequency discrimination was measured for 1 a fixed frequency standard and target 2 a fixed frequency standard and a frequency transition target and 3 frequency transition standard and a frequency transition target the 50 ms frequency transitions had the same final frequency as the standards but the initial frequency was lowered to obtain about 79 discrimination performance there was a strong relationship between poor frequency resolution and elevated pure tone thresholds but only a very weak relationship between poor frequency discrimination and elevated pure tone thresholds several hearing impaired listeners had normal discrimination performance together with pure tone thresholds of 80 90 db hl a slight correlation was found between word recognition and frequency discrimination but a detailed comparison of the phonetic errors and either the frequency discrimination or frequency resolution tasks failed to suggest any consistent interdependencies these results are consistent with previous work that has suggested that frequency resolution and frequency discrimination are independent processes

best approximation of a nonnormal operator in the trace norm on determine quand certains operateurs non normaux ont un approximant positif unique dans la norme trace on considere des operateurs sur un espace de hilbert complexe separable h

relating interaural difference sensitivities for several parameters measured in normal hearing and hearing impaired listeners just noticeable differences jnds in interaural time delay itd interaural level difference ild and interaural cross correlation icc were measured with low and high frequency noise bands over multiple sessions for 10 normal hearing nh and 11 hearing impaired hi listeners individual subject thresholds tended to improve with training then stabilize measured jnds varied over these experienced listeners for both subject groups and all tasks group jnds were seldom predictable from hearing level individual listeners jnds were highly correlated across frequency for each task and group except for icc in the hi listeners further itd jnds almost always significantly correlated with ild jnds within a group finally although the icc jnds always significantly correlated with the itd or ild jnds for the nh listeners they often did not for the hi listeners these findings suggest that little information about binaural sensitivity is added for nh listeners with multiple itd ild and icc measures for hi listeners however while itd and ild measures are well correlated information is added with icc measures in general the results suggest that less information is added with jnd measures for nh listeners 15 significant correlations than for hi listeners six significant correlations

semidefinite programming relaxations for semialgebraic problems a hierarchy of convex relaxations for semialgebraic problems is introduced for questions reducible to a finite number of polynomial equalities and inequalities it is shown how to construct a complete family of polynomially sized semidefinite programming conditions that prove infeasibility the main tools employed are a semidefinite programming formulation of the sum of squares decomposition for multivariate polynomials and some results from real algebraic geometry the techniques provide a constructive approach for finding bounded degree solutions to the positivstellensatz and are illustrated with examples from diverse application fields

relations between intelligibility of narrow band speech and auditory functions both in the 1 khz frequency region relations between perception of suprathreshold speech and auditory functions were examined in 24 hearing impaired listeners and 12 normal hearing listeners the speech intelligibility index sii was used to account for audibility the auditory functions included detection efficiency temporal and spectral resolution temporal and spectral integration and discrimination of intensity frequency rhythm and spectro temporal shape all auditory functions were measured at 1 khz speech intelligibility was assessed with the speech reception threshold srt in quiet and in noise and with the speech reception bandwidth threshold srbt previously developed for investigating speech perception in a limited frequency region around 1 khz the results showed that the elevated srt in quiet could be explained on the basis of audibility audibility could only partly account for the elevated srt values in noise and the deviant srbt values suggesting that suprathreshold deficits affected intelligibility in these conditions sii predictions for the srbt improved significantly by including the individually measured upward spread of masking in the sii model reduced spectral resolution reduced temporal resolution and reduced frequency discrimination appeared to be related to speech perception deficits loss of peripheral compression appeared to have the smallest effect on the intelligibility of suprathreshold speech

providing a basin of attraction to a target region of polynomial systems by computation of lyapunov like functions in this paper we present a method for computing a basin of attraction to a target region for polynomial ordinary differential equations this basin of attraction is ensured by a lyapunov like polynomial function that we compute using an interval based branch and relax algorithm this algorithm relaxes the necessary conditions on the coefficients of the lyapunov like function to a system of linear interval inequalities that can then be solved exactly it iteratively refines these relaxations in order to ensure that whenever a nondegenerate solution exists it will eventually be found by the algorithm application of an implementation to a range of benchmark problems shows the usefulness of the approach

correspondence between behavioral and individually optimized otoacoustic emission estimates of human cochlear input output curves previous studies have shown a high within subject correspondence between distortion product otoacoustic emission dpoae input output i o curves and behaviorally inferred basilar membrane bm i o curves for frequencies above 2 khz for lower frequencies dpoae i o curves contained notches and plateaus that did not have a counterpart in corresponding behavioral curves it was hypothesized that this might improve by using individualized optimal dpoae primary levels here data from previous studies are re analyzed to test this hypothesis by comparing behaviorally inferred bm i o curves and dpoae i o curves measured with well established group average primary levels and two individualized primary level rules one optimized to maximize dpoae levels and one intended for primaries to evoke comparable bm responses at the f2 cochlear region test frequencies were 0 5 1 and 4 khz behavioral i o curves were obtained from temporal forward masking curves results showed high within subject correspondence between behavioral and dpoae i o curves at 4 khz only regardless of the primary level rule plateaus and notches were equally common in low frequency dpoae i o curves for individualized and group average dpoae primary levels at 0 5 and 1 khz results are discussed in terms of the adequacy of dpoae i o curves for inferring individual cochlear nonlinearity characteristics

stability criteria for switched and hybrid systems the study of the stability properties of switched and hybrid systems gives rise to a number of interesting and challenging mathematical problems the objective of this paper is to outline some of these problems to review progress made in solving them in a number of diverse communities and to review some problems that remain open an important contribution of our work is to bring together material from several areas of research and to present results in a unified manner we begin our review by relating the stability problem for switched linear systems and a class of linear differential inclusions closely related to the concept of stability are the notions of exponential growth rates and converse lyapunov theorems both of which are discussed in detail in particular results on common quadratic lyapunov functions and piecewise linear lyapunov functions are presented as they represent constructive methods for proving stability and also represent problems in which significant progress has been made we also comment on the inherent difficulty in determining stability of switched systems in general which is exemplified by np hardness and undecidability results we then proceed by considering the stability of switched systems in which there are constraints on the switching rules through both dwell time requirements and state dependent switching laws also in this case the theory of lyapunov functions and the existence of converse theorems are reviewed we briefly comment on the classical lur e problem and on the theory of stability radii both of which contain many of the features of switched systems and are rich sources of practical results on the topic finally we present a list of questions and open problems which provide motivation for continued research in this area

detection thresholds for sinusoidal frequency modulation an adaptive forced choice procedure was used to measure in four normal hearing subjects detection thresholds for sinusoidal frequency modulation as a function of carrier frequency fc from 250 to 4000 hz and modulation frequency fmod from 1 to 64 hz the results show that for a wide range of fmod values fc and fmod have almost independent effects on the thresholds when the thresholds are expressed as just noticeable frequency swings and plotted on a log scale in two subjects the effect of fc on the thresholds was compared to the effect of standard frequency on the frequency just noticeable differences jnd s of successive and steady tones in agreement with previous data h fastl j acoust soc am 63 275 277 1978 it was found that the two effects are significantly different if the frequency jnd s are measured with long duration tones however it was also found that the two effects are similar if the frequency jnd s are measured with 25 ms tones these results support the idea that

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

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

noso and nosπ detection as a function of masker bandwidth in normal hearing and cochlear impaired listeners noso and nos detection thresholds for a 500 hz pure tone signal were measured as a function of masking noise bandwidth in normal hearing and cochlear hearing impaired subjects noso and nos critical bands were derived from the bandlimited noise functions a notched noise measure of the monaural critical band was also obtained for each ear one hypothesis tested was that an asymmetrical monaural critical band would result in a relatively steep improvement of the nos detection threshold as a function of decreasing masker bandwidth and would therefore be associated with a wider binaural critical band this was hypothesized because the outputs of the left and right auditory filters would be more decorrelated the greater the interaural difference in the monaural critical band however as the noise bandwidth was narrowed the decorrelation would lessen resulting in a relatively steep improvement in nos detection results indicated that the masking level difference mld was smaller and that the monaural critical bands were generally wider in cochlear impaired listeners noso and nos critical bands were somewhat larger in the cochlear hearing impaired listeners having relatively wide monaural critical bands there was a significant correlation between monaural critical band asymmetry and the nos critical band however this correlation was insignificant when a control was employed for the critical band in the worse ear therefore the present results did not support a strong association between monaural critical band asymmetry and the width of the nos critical band

brief on solution concepts and well posedness of linear relay systems in this paper we study the well posedness existence and uniqueness of solutions of linear relay systems with respect to two different solution concepts filippov solutions and forward solutions we derive necessary and sufficient conditions for well posedness in the sense of filippov of linear systems of relative degree one and two in closed loop with relay feedback to be precise uniqueness of filippov and also forward solutions follows in this case if the first non zero markov parameter is positive by means of an example it is shown that this intuitively clear condition is not true for systems with relative degree larger than two the influence of the zeno phenomenon an infinite number of relay switching times in a finite length time interval on well posedness is highlighted and although linear relay systems form a rather limited subclass of hybrid dynamical systems the consequences of the presence of the zeno behaviour is typical for many other classes of non smooth and hybrid systems

approximation in trace norm by positive semidefinite matrices abstract information is obtained about best approximation of a matrix by positive semidefinite ones in trace norm the situation is much different from approximation in spectral and euclidean norms

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

temporal masking functions for listeners with real and simulated hearing loss a functional simulation of hearing loss was evaluated in its ability to reproduce the temporal masking functions for eight listeners with mild to severe sensorineural hearing loss each audiometric loss was simulated in a group of age matched normal hearing listeners through a combination of spectrally shaped masking noise and multi band expansion temporal masking functions were obtained in both groups of listeners using a forward masking paradigm in which the level of a 110 ms masker required to just mask a 10 ms fixed level probe 5 10 db sl was measured as a function of the time delay between the masker offset and probe onset at each of four probe frequencies 500 1000 2000 and 4000 hz temporal masking functions were obtained using maskers that were 0 55 1 0 and 1 15 times the probe frequency the slopes and y intercepts of the masking functions were not significantly different for listeners with real and simulated hearing loss the y intercepts were positively correlated with level of hearing loss while the slopes were negatively correlated the ratio of the slopes obtained with the low frequency maskers relative to the on frequency maskers was similar for both groups of listeners and indicated a smaller compressive effect than that observed in normal hearing listeners

additivity of masking in normally hearing and hearing impaired subjects the effects of combining two equally effective maskers were studied in normally hearing and elderly hearing impaired subjects the additivity of nonsimultaneous masking was investigated by measuring thresholds for a brief 4 khz signal in the presence of a broadband noise forward masker a backward masker and a combination of both for the normally hearing subjects combining two equally effective nonsimultaneous maskers resulted in up to a 15 db greater increase in threshold than the 3 db predicted by an energy summation model excess masking however the hearing impaired subjects showed little or no excess masking the difference between the two groups is consistent with a theory linking excess masking to the compressive transfer function measured on the basilar membrane bm in the hearing impaired subjects the transfer function is more linear accounting for the lack of excess masking the additivity of simultaneous masking was investigated by measuring thresholds for a 100 ms 4 khz signal in the presence of either a 400 ms broadband noise masker or a 400 ms sinusoidal masker at the same frequency as the signal and then combining two equally effective maskers a noise and a tone the maximum amount of excess masking 3 to 4 db was similar across the two groups of subjects consistent with an explanation based on the use of different detection cues for the tonal and noise maskers it is argued that while peripheral compression may underlie excess masking for pairs of nonsimultaneous maskers it is unlikely that in simultaneous masking where the maskers are close in frequency to the signal the two maskers are compressed individually before their effects are combined it is further suggested that bm nonlinearity may underlie the effects of the upward spread of masking and the nonlinear growth of forward masking as well as accounting for the additivity of simultaneous masking when the masker frequencies are well below that of the signal

models of the additivity of masking three models of masking additivity are reviewed which are referred to as the high compression model m j penner j acoust soc am 67 608 616 1980 m j penner and r m shiffrin j acoust soc am 67 617 627 1980 the power law model r a lutfi j acoust soc am 73 262 267 1983 80 422 428 1986 and the modified power law model with compressed internal noise humes et al j acoust soc am 83 188 202 1988 while the high compression model was derived from data for two or more nonsimultaneous maskers and the power law model was derived from data for two or more simultaneous maskers the modified power law model can be applied to both cases the modified power law model assumes that the threshold in quiet is equivalent to a masked threshold resulting from an internal noise that is continually present additional assumptions concern the interaction of two maskers prior to the addition of the masking effects most of the data on the additivity of masking are well described b

dynamical properties of hybrid automata hybrid automata provide a language for modeling and analyzing digital and analogue computations in real time systems hybrid automata are studied here from a dynamical systems perspective necessary and sufficient conditions for existence and uniqueness of solutions are derived and a class of hybrid automata whose solutions depend continuously on the initial state is characterized the results on existence uniqueness and continuity serve as a starting point for stability analysis lyapunov s theorem on stability via linearization and lasalle s invariance principle are generalized to hybrid automata

forward masking as a function of frequency masker level and signal delay the forward masking of a sinusoidal signal by a sinusoid of the same frequency was investigated for frequencies ranging from 125 to 4000 hz forward masking in db is proportional to both masker level and log signal delay at each frequency more forward masking occurs at very low frequencies than at high frequencies given equal sensation level maskers and masked thresholds are greater at low frequencies than at high frequencies given equal spl maskers the data can be described equally well by assuming that the difference in forward masking as a function of frequency is due to a change in the time course of recovery from masking or to a change in the growth of masking at each signal delay the frequency effect is not large enough to change the interpretation of forward masking data in studies of suppression or psychophysical tuning curves

a behavioral measure of basilar membrane nonlinearity in listeners with normal and impaired hearing this paper examines the possibility of estimating basilar membrane bm nonlinearity using a psychophysical technique the level of a forward masker required to mask a brief signal was measured for conditions where the masker was either at or one octave below the signal frequency the level of the forward masker at masked threshold provided an indirect measure of the bm response to the signal as follows consistent with physiological studies it was assumed that the bm responds linearly to frequencies well below the characteristic frequency cf thus the ratio of the slopes of the masking functions between a masker at the signal frequency and a masker well below the signal frequency should provide an estimate of bm compression at cf results obtained from normally hearing listeners were in quantitative agreement with physiological estimates of bm compression furthermore differences between normally hearing listeners and listeners with cochlear hearing impairment were consistent with the physiological effects of damage to the cochlea the results support the hypothesis that bm nonlinearity governs the nonlinear growth of the upward spread of masking and suggest that this technique provides a straightforward method for estimating bm nonlinearity in humans

estimating peripheral gain and compression using fixed duration masking curves estimates of human basilar membrane gain and compression obtained using temporal masking curve tmc and additivity of forward masking afm methods with long duration maskers or long masker signal silent intervals may be affected by olivocochlear efferent activation which reduces basilar membrane gain the present study introduces a fixed duration masking curve fdmc method which involves a comparison of off and on frequency forward masker levels at threshold as a function of masker and signal duration with the total masker signal duration fixed at 25 ms to minimize efferent effects gain and compression estimates from the fdmc technique were compared with those from tmc 104 ms maskers and afm 10 and 200 ms maskers methods compression estimates over an input masker range of 40 60 db sound pressure level were similar for the four methods maximum compression occurred at a lower input level for the fdmc compared to the tmc method estimates of gain were similar for tmc and fdmc methods the fdmc

effect of spectral change enhancement for the hearing impaired using parameter values selected with a genetic algorithm chen et al j acoust soc am 131 2987 2998 2012 evaluated the effectiveness of an algorithm for enhancing spectral changes over time in improving the intelligibility of speech in background sounds for hearing impaired subjects the processing improved intelligibility for speech in steady speech spectrum noise ssn but tended to impair intelligibility in a background of two talker speech tts large individual differences were found the present study assessed whether the effectiveness of the algorithm was improved when the parameters that controlled the degree and type of enhancement were chosen individually for each subject using a genetic algorithm based on subjective preferences for speech clarity the parameter values selected by the genetic algorithm varied markedly across subjects speech intelligibility was measured for unprocessed stimuli and stimuli processed using the selected parameters with ssn and tts maskers and two signal to masker ratios smrs for each subject the intelligibi

discrimination of interaural temporal disparities by normal hearing listeners and listeners with high frequency sensorineural hearing loss thresholds of ongoing interaural time difference itd were obtained from normal hearing and hearing impaired listeners who had high frequency sensorineural hearing loss several stimuli a 500 hz sinusoid a narrow band noise centered at 500 hz a sinusoidally amplitude modulated 4000 hz tone and a narrow band noise centered at 4000 hz and two criteria equal sound pressure level eq spl and equal sensation level eq sl for determining the level of stimuli presented to each listener were employed the itd thresholds and slopes of the psychometric functions were elevated for hearing impaired listeners for the two high frequency stimuli in comparison to a the listener s own low frequency thresholds and b data obtained from normal hearing listeners for stimuli presented with eq spl interaurally the two groups of listeners required similar itds to reach threshold when stimuli were presented at eq sls to each ear for low frequency stimuli the itd thresholds of the hearing impaired listener were

speech reception by listeners with real and simulated hearing impairment effects of continuous and interrupted noise the effects of audibility and age on masking for sentences in continuous and interrupted noise were examined in listeners with real and simulated hearing loss the absolute thresholds of each of ten listeners with sensorineural hearing loss were simulated in normal hearing listeners through a combination of spectrally shaped threshold noise and multi band expansion for octave bands with center frequencies from 0 25 8 khz each individual hearing loss was simulated in two groups of three normal hearing listeners an age matched and a non age matched group the speech to noise ratio s n for 50 correct identification of hearing in noise test hint sentences was measured in backgrounds of continuous and temporally modulated 10 hz square wave noise at two overall levels for unprocessed speech and for speech that was amplified with the nal rp prescription the s n in both continuous and interrupted noise of the hearing impaired listeners was relatively well simulated in both groups of normal hearing listeners thus release from masking the difference in s n obtained in continuous versus interrupted noise appears to be determined primarily by audibility minimal age effects were observed in this small sample observed values of masking release were compared to predictions derived from intelligibility curves generated using the extended speech intelligibility index esii rhebergen et al 2006 j acoust soc am 120 3988 3997

on the gap between positive polynomials and sos of polynomials this note investigates the gap existing between positive polynomials and sum of squares sos of polynomials which affects several analysis and synthesis tools in control systems based on polynomial sos relaxations and about which almost nothing is known in particular a matrix characterization of the pns that is the positive homogeneous forms that are not sos is proposed which allows to show that any pns is the vertex of an unbounded cone of pns moreover a complete parametrization of the set of pns is introduced

frequency dependence of binaural performance in listeners with impaired binaural hearing binaural performance was measured as a function of stimulus frequency for four impaired listeners each with bilaterally symmetric audiograms the subjects had various degrees and configurations of audiometric losses two had high frequency sensorineural losses one had a flat sensorineural loss and one had multiple sclerosis with normal audiometric thresholds just noticeable differences jnd s in interaural time interaural intensity and interaural correlation as well as detection thresholds for noso and nos conditions were obtained for narrow band noise stimuli at octave frequencies from 250 4000 hz performance of the impaired listeners was generally poorer than that of normal hearing listeners although it was comparable to normal in a few instances the patterns of binaural performance showed no apparent relation to the audiometric patterns even the two subjects with similar degree and configuration of hearing loss have very different binaural performance both in the level and frequency depen

growth of masking as a measure of response growth in hearing impaired listeners growth of masking functions were obtained from 19 normal and 5 hearing impaired listeners using a simultaneous masking paradigm when masker and probe frequency are identical the slope of masking approximates 1 0 for both normal hearing and impaired listeners for masker frequencies less than or greater than probe frequency the slopes for impaired listeners are shallower than those of normals these findings are consistent with previously reported physiological data single fiber rate versus level and ap masking functions for animals with induced cochlear lesions results are discussed in terms of a potential masking technique to estimate the growth of response in normal and impaired ears

use of psychometric function slopes for forward masked tones to investigate cochlear nonlinearity schairer et al 2003 effects of peripheral nonlinearity on psychometric functions for forward masked tones j acoust soc am 133 1560 1573 demonstrated that cochlear nonlinearity is reflected in psychometric function pf slopes for 4 khz forward masked tones the goals of the current study were to use pf slopes to compare the degree of compression between signal frequencies of 0 25 and 4 khz in listeners with normal hearing lnh and between lnh and listeners with cochlear hearing loss lhl forward masked thresholds were estimated in lnh and lhl using on and off frequency maskers and 0 25 and 4 khz signals in three experiments pfs were reconstructed from adaptive procedure data for each subject in each condition trends in pf slopes across conditions suggest comparable compression at 0 25 and 4 khz and potentially a wider bandwidth of compression in relative frequency at 0 25 khz this is consistent with other recent behavioral studies that revise earlier estimates of less compression at lower frequencies the preliminary results in lhl demonstrate that pf slopes are abnormally steep at frequencies with hl but are similar to those for lnh at frequencies with nh overall the results are consistent with the notion that pf slopes reflect degree of cochlear nonlinearity and can be used as an additional measure of compression across frequency

approximation by matrices with restricted spectra we investigate the properties of the approximation of a matrix by matrices whose spectra are in a closed convex set of the complex plane we explain why the khalil and maher characterization of an approximant which spectrum is in a strip is not quite correct we prove that their characterization is valid but for another kind of approximation we formulate a conjecture which leads to some algorithm for computing approximants the conjecture is motivated by numerical experiments and some theoretical considerations separately we consider the approximation of normal matrices

relation between derived band auditory brainstem response latencies and behavioral frequency selectivity derived band click evoked auditory brainstem responses abrs were obtained for normal hearing nh and sensorineurally hearing impaired hi listeners the latencies extracted from these responses as a function of derived band center frequency and click level served as objective estimates of cochlear response times for the same listeners auditory filter bandwidths at 2 khz were estimated using a behavioral notched noise masking paradigm generally shorter derived band latencies were observed for the hi than for the nh listeners only at low click sensation levels prolonged latencies were obtained for some of the hi listeners the behavioral auditory filter bandwidths accounted for the across listener variability in the abr latencies cochlear response time decreased with increasing filter bandwidth consistent with linear system theory the results link cochlear response time and frequency selectivity in human listeners and offer a window to better understand how hearing impairment affects the spatiotemporal cochlear response pattern

effect of auditory nerve response variability on estimates of tuning curves near poisson variability in auditory nerve an responses limits the accuracy of automated tuning curve algorithms here a typical adaptive tuning curve algorithm was used with a physiologically realistic an model with and without the inclusion of neural randomness response randomness produced variability in q 10 estimates that was nearly as large as in an data results suggest that it is sufficient for an models to specify frequency selectivity based on mean q 10 values at each characteristic frequency cf errors in estimates of cf which decreased from 0 2 octaves at low frequencies to 0 05 octaves at high frequencies are significant for studies of spatiotemporal coding

cochlear nonlinearity in normal hearing subjects as inferred psychophysically and from distortion product otoacoustic emissions the aim was to investigate the correlation between estimates of compression ratio compression threshold and cochlear gain for normal hearing subjects inferred from temporal masking curves tmc and distortion product otoacoustic emission dpoaes input output i o curves special care was given to reduce the influence of dpoae fine structure on the i o curves data were collected for 10 subjects and for cochlear places with characteristic frequencies cfs of 0 5 1 2 and 4 khz except at 4 khz little correlation was found between ratio estimates obtained with the two methods one reason is that the dpoae i o curves show plateaus or dips which result in unexpectedly high compression estimates moderately high correlation was found between the compression threshold estimates obtained with the two methods although dpoae based values were systematically lower than those based on the tmcs both methods confirmed that compression ratio and threshold are approximately constant across the cf range considered cochlear gain could be estimated from tmcs only and it was found to increase slightly with increasing cf it is discussed that dpoae i o curves may be inappropriate to infer precise individual nonlinear cochlear characteristics in normal hearing listeners particularly at low cfs work supported by imserso 131 06 profit cit 390000 2005 4 mec bfu 2006 07536 and the oticon foundation

the deterioration of hearing with age frequency selectivity the critical ratio the audiogram and speech threshold the frequency selectivity of the auditory system was measured by masking a sinusoidal signal 0 5 2 0 or 4 0 khz or a filtered speech signal with a wideband noise having a notch or stopband centered on the signal as the notch was widened performance improved for both types of signal but the rate of improvement decreased as the age of the 16 listeners increased from 23 to 75 years indicating a loss in frequency selectivity with age auditory filter shapes derived from the tone in noise data show a that the passband of the filter broadens progressively with age and b that the dynamic range of the filter ages like the audiogram that is the range changes little with age before 55 but beyond this point there is an accelerating rate of loss the speech experiment shows comparable but smaller effects the filter width measurements show that the critical ratio is a poor estimator of frequency selectivity because it confounds the tuning of the system with the efficiency of the signal detection and speech processing mechanisms that follow the filter an alternative one point measure of frequency selectivity which is both sensitive and reliable is developed via the filter shape model of masking

real solution isolation using interval arithmetic we propose a complete algorithm for real solution isolation for semialgebraic systems by using interval arithmetic the algorithm is implemented as a maple program nrealzero and its performance on several examples is reported

age related differences in discrimination of an interval separating onsets of successive tone bursts as a function of interval duration the study measured listener sensitivity to increments in the inter onset interval ioi separating pairs of successive 20 ms 4000 hz tone pulses a silent interval between the tone pulses was adjusted across conditions to create reference tonal ioi values of 25 600ms for each condition a duration dl for increments of the tonal ioi was measured in listeners comprised of young normal hearing adults and two groups of older adults with and without high frequency hearing loss discrimination performance of all listeners was poorest for the shorter reference iois and improved to stable levels for longer reference intervals exceeding about 200ms temporal sensitivity of the young listeners was significantly better than that of the elderly listeners in each condition with the largest age related differences observed for the shortest reference interval age related differences were also observed for duration dls measured using single 4000 hz tone bursts set to three reference durations in the range 50 200ms t

characterizations and criteria for synchronization of heterogeneous networks to linear subspaces in this paper we investigate synchronization issues in heterogeneous complex networks we start with a necessary condition for the existence of synchronous trajectories by analyzing the vector fie

relative dominance of individual partials in determining the pitch of complex tones these experiments were conducted to determine the dominance of each partial in determining the residue pitch of a complex tone subjects were required to make pitch matches to a complex tone which had one partial slightly mistuned from its correct harmonic value the shift in residue pitch was measured as a function of the frequency shift of the harmonic for each harmonic in turn for mistunings up to 2 3 the shift in residue pitch was approximately a linear function of the shift in the harmonic but for greater mistunings the shift in residue pitch was reduced the degree to which a given harmonic can influence residue pitch gives a measure of the dominance of that harmonic the dominant harmonics were always contained within the lowest six harmonics for fundamental frequencies of 100 200 and 400 hz but there were marked individual differences in the exact distribution of dominance across harmonics the level of a harmonic relative to adjacent harmonics can have a significant effect on its dominance the implications of the results for theories of pitch perception are discussed

characterizing auditory processing and perception in individual listeners with sensorineural hearing loss this study considered consequences of sensorineural hearing loss in ten listeners the characterization of individual hearing loss was based on psychoacoustic data addressing audiometric pure tone sensitivity cochlear compression frequency selectivity temporal resolution and intensity discrimination in the experiments it was found that listeners with comparable audiograms can show very different results in the supra threshold measures in an attempt to account for the observed individual data a model of auditory signal processing and perception jepsen et al j acoust soc am 124 422 438 2008 was used as a framework the parameters of the cochlear processing stage of the model were adjusted to account for behaviorally estimated individual basilar membrane input output functions and the audiogram from which the amounts of inner hair cell and outer hair cell losses were estimated as a function of frequency all other model parameters were left unchanged the predictions showed a reasonably good agreement with the measured individual data in the frequency selectivity and forward masking conditions while the variation of intensity discrimination thresholds across listeners was underestimated by the model the model and the associated parameters for individual hearing impaired listeners might be useful for investigating effects of individual hearing impairment in more complex conditions such as speech intelligibility in noise

condition number based complexity estimate for solving polynomial systems by modifying and combining algorithms in symbolic and numerical computation we propose a real root counting based method for deciding the feasibility of systems of polynomial equations along with this method we also use a modified newton operator to efficiently approximate the real solutions when the systems are feasible the complexity of our method can be measured by a number of arithmetic operations which is singly exponential in the number of variables

comparing different estimates of cochlear compression in listeners with normal and impaired hearing a loss of cochlear compression may underlie many of the difficulties experienced by hearing impaired listeners two behavioral forward masking paradigms that have been used to estimate the magnitude of cochlear compression are growth of masking gom and temporal masking tm the aim of this study was to determine whether these two measures produce within subjects results that are consistent across a range of signal frequencies and if so to compare them in terms of reliability or efficiency gom and tm functions were measured in a group of five normal hearing and five hearing impaired listeners at signal frequencies of 1000 2000 and 4000 hz compression values were derived from the masking data and confidence intervals were constructed around these estimates both measures produced comparable estimates of compression but both measures have distinct advantages and disadvantages so that the more appropriate measure depends on factors such as the frequency region of interest and the degree of hearing loss because of the long testing times needed neither measure is suitable for clinical use in its current form

on and off frequency compression estimated using a new version of the additivity of forward masking technique on and off frequency compression at the 4000 and 8000 hz cochlear places were estimated using a new version of the additivity of forward masking afm technique that measures the effects of combining two non overlapping forward maskers instead of measuring signal thresholds to estimate compression of the signal as in the original afm technique the decrease in masker threshold in the combined masker condition compared to the individual masker conditions is used to estimate compression of the masker at the signal place by varying masker frequency it is possible to estimate off frequency compression the maskers were 500 hz wide bands of noise and the signal was a brief pure tone compression at different levels was estimated using different overall signal levels or different masker signal intervals it was shown that the new afm technique and the original afm technique produce consistent results considerable compression was observed for maskers well below the signal frequency suggesting that the assumption of off frequency linearity used in other techniques may not be valid reducing the duration of the first masker from 200 to 20 ms reduced the compression exponent in some cases suggesting a possible influence of olivocochlear efferent activity

robust stability analysis of nonlinear hybrid systems we present a methodology for robust stability analysis of nonlinear hybrid systems through the algorithmic construction of polynomial and piecewise polynomial lyapunov like functions using convex optimization and in particular the sum of squares decomposition of multivariate polynomials several improvements compared to previous approaches are discussed such as treating in a unified way polynomial switching surfaces and robust stability analysis for nonlinear hybrid systems

frequency discrimination of complex signals frequency selectivity and speech perception in hearing impaired subjects frequency discrimination of spectral envelopes of complex stimuli frequency selectivity measured with psychophysical tuning curves and speech perception were determined in hearing impaired subjects each having a relatively flat sensory neural loss both the frequency discrimination and speech perception measures were obtained in quiet and noise most of these subjects showed abnormal susceptibility to ambient noise with regard to speech perception frequency discrimination in quiet and frequency selectivity did not correlate significantly at low signal to noise ratios frequency discrimination correlated significantly with frequency selectivity speech perception in noise correlated significantly with frequency selectivity and with frequency discrimination at low signal to noise ratios the frequency discrimination data are discussed in terms of an excitation pattern model however they neither support nor refute the model

comparison of the roex and gammachirp filters as representations of the auditory filter although the rounded exponential roex filter has been successfully used to represent the magnitude response of the auditory filter recent studies with the roex p w t filter reveal two serious problems the fits to notched noise masking data are somewhat unstable unless the filter is reduced to a physically unrealizable form and there is no time domain version of the roex p w t filter to support modeling of the perception of complex sounds this paper describes a compressive gammachirp cgc filter with the same architecture as the roex p w t which can be implemented in the time domain the gain and asymmetry of this parallel cgc filter are shown to be comparable to those of the roex p w t filter but the fits to masking data are still somewhat unstable the roex p w t and parallel cgc filters were also compared with the cascade cgc filter patterson et al j acoust soc am 114 1529 1542 2003 which was found to provide an equivalent fit with 25 fewer coefficients moreover the fits were stable the advantage of the cascade cgc filter appears to derive from its parsimonious representation of the high frequency side of the filter it is concluded that cgc filters offer better prospects than roex filters for the representation of the auditory filter

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

random effects models for longitudinal data models for the analysis of longitudinal data must recognize the relationship between serial observations on the same unit multivariate models with general covariance structure are often difficult to apply to highly unbalanced data whereas two stage random effects models can be used easily in two stage models the probability distributions for the response vectors of different individuals belong to a single family but some random effects parameters vary across individuals with a distribution specified at the second stage a general family of models is discussed which includes both growth models and repeated measures models as special cases a unified approach to fitting these models based on a combination of empirical bayes and maximum likelihood estimation of model parameters and using the em algorithm is discussed two examples are taken from a current epidemiological study of the health effects of air pollution

some inequalities for norm ideals several inequalities for norms of operators are extended to more operators and or to more norms these include results of halmos and bouldin on approximating a normal operator by another with restricted spectrum the powers stormer and the van hemmen ando inequalities for the distance between the square roots of two positive operators and also some recent generalisations of these latter results by kittaneh

a multi resolution envelope power based model for speech intelligibility the speech based envelope power spectrum model sepsm presented by jorgensen and dau 2011 j acoust soc am 130 1475 1487 estimates the envelope power signal to noise ratio snrenv after modulation frequency selective processing changes in this metric were shown to account well for changes of speech intelligibility for normal hearing listeners in conditions with additive stationary noise reverberation and nonlinear processing with spectral subtraction in the latter condition the standardized speech transmission index 2003 iec 60268 16 fails however the sepsm is limited to conditions with stationary interferers due to the long term integration of the envelope power and cannot account for increased intelligibility typically obtained with fluctuating maskers here a multi resolution version of the sepsm is presented where the snrenv is estimated in temporal segments with a modulation filter dependent duration the multi resolution sepsm is demonstrated to account for intelligibility

condition number based complexity estimate for computing local extrema in this paper we present a new algorithm for computing local extrema by modifying and combining algorithms in symbolic and numerical computation this new algorithm improves the classical steepest descent method that may not terminate by combining a sturm s theorem based separation method and a sufficient condition on infeasibility in addition we incorporate a grid subdivision method into our algorithm to approximate all local extrema the complexity of our algorithm is polynomial in a newly defined condition number and singly exponential in the number of variables

linear complementarity systems zeno states a linear complementarity system lcs is a hybrid dynamical system defined by a linear time invariant ordinary differential equation coupled with a finite dimensional linear complementarity problem lcp the present paper is the first of several papers whose goal is to study some fundamental issues associated with an lcs specifically this paper addresses the issue of zeno states and the related issue of finite number of mode switches in such a system the cornerstone of our study is an expansion of a solution trajectory to the lcs near a given state in terms of an observability degree of the state on the basis of this expansion and an inductive argument we establish that an lcs satisfying the p property has no strongly zeno states we next extend the analysis for such an lcs to a broader class of problems and provide sufficient conditions for a given state to be weakly non zeno while related mode switch results have been proved by brunovsky and sussmann for more general hybrid systems our analysis exploits the special structure of the lcs and yields new results for the latter that are of independent interest and complement those by these two and other authors

towards a measure of auditory filter phase response this study investigates how the phase curvature of the auditory filters varies with center frequency cf and level harmonic tone complex maskers were used with component phases adjusted using a variant of an equation proposed by schroeder ieee trans inf theory 16 85 89 1970 in experiment 1 the phase curvature of the masker was varied systematically and sinusoidal signal thresholds were measured at frequencies from 125 to 8000 hz at all signal frequencies threshold differences of 20 db or more were observed between the most effective and least effective masker phase curvature in experiment 2 the effect of overall masker level on masker phase effects was studied using signal frequencies of 250 1000 and 4000 hz the results were used to estimate the phase curvature of the auditory filters the estimated relative phase curvature decreases dramatically with decreasing cf below 1000 hz at frequencies above 1000 hz relative auditory filter phase curvature increases only slowly with increasing cf or may remain constant the phase curvature of the auditory filters seems to be broadly independent of overall level most aspects of the data are in qualitative agreement with peripheral physiological findings from other mammals which suggests that the phase responses observed here are of peripheral origin however in contrast to the data reported in a cat auditory nerve study carney et al j acoust soc am 105 2384 2391 1999 no reversal in the sign of the phase curvature was observed at very low frequencies overall the results provide a framework for mapping out the phase curvature of the auditory filters and provide constraints on future models of peripheral filtering in the human auditory system

behavioral measures of cochlear compression and temporal resolution as predictors of speech masking release in hearing impaired listeners hearing impaired hi listeners often show less masking release mr than normal hearing listeners when temporal fluctuations are imposed on a steady state masker even when accounting for overall audibility differences this difference may be related to a loss of cochlear compression in hi listeners behavioral estimates of compression using temporal masking curves tmcs were compared with mr for band limited 500 4000 hz speech and pure tones in hi listeners and age matched noise masked normal hearing nmnh listeners compression and pure tone mr estimates were made at 500 1500 and 4000 hz the amount of mr was defined as the difference in performance between steady state and 10 hz square wave gated speech shaped noise in addition temporal resolution was estimated from the slope of the off frequency tmc no significant relationship was found between estimated cochlear compression and mr for either speech or pure tones nmnh listeners had significantly steeper off frequency temporal masking recovery slopes than did hi listeners and a small but significant correlation was observed between poorer temporal resolution and reduced mr for speech the results suggest either that the effects of hearing impairment on mr are not determined primarily by changes in peripheral compression or that the tmc does not provide a sufficiently reliable measure of cochlear compression

stability region analysis using polynomial and composite polynomial lyapunov functions and sum of squares programming we propose using bilinear sum of squares programming for obtaining inner bounds of roas for dynamical systems with polynomial vector fields we search for polynomial as well as composite lyapunov functions comprised of pointwise maximums of polynomial functions results for several examples from the literature are presented using the proposed methods and the penbmi solver

stability and stabilizability of switched linear systems a survey of recent results during the past several years there have been increasing research activities in the field of stability analysis and switching stabilization for switched systems this paper aims to briefly survey recent results in this field first the stability analysis for switched systems is reviewed we focus on the stability analysis for switched linear systems under arbitrary switching and we highlight necessary and sufficient conditions for asymptotic stability after a brief review of the stability analysis under restricted switching and the multiple lyapunov function theory the switching stabilization problem is studied and a variety of switching stabilization methods found in the literature are outlined then the switching stabilizability problem is investigated that is under what condition it is possible to stabilize a switched system by properly designing switching control laws note that the switching stabilizability problem has been one of the most elusive problems in the switched systems literature a necessary and sufficient condition for asymptotic stabilizability of switched linear systems is described here

remarque sur l approximation positive contractante abstract in this note we show that if a b i c is normal operator then for some decomposition of the hilbert space h o h 0 b h 1 i h 2 is an approximant of a in the class of operators with spectrum in 0 1

a nonconservative lmi condition for stability of switched systems with guaranteed dwell time ensuring stability of switched linear systems with a guaranteed dwell time is an important problem in control systems several methods have been proposed in the literature to address this problem but unfortunately they provide sufficient conditions only this technical note proposes the use of homogeneous polynomial lyapunov functions in the non restrictive case where all the subsystems are hurwitz showing that a sufficient condition can be provided in terms of an lmi feasibility test by exploiting a key representation of polynomials several properties are proved for this condition in particular that it is also necessary for a sufficiently large degree of these functions as a result the proposed condition provides a sequence of upper bounds of the minimum dwell time that approximate it arbitrarily well some examples illustrate the proposed approach

centennial history of hilbert s 16th problem the second part of hilbert s 16th problem deals with polynomial differential equations in the plane it remains unsolved even for quadratic polynomials there were several attempts to solve it that failed yet the problem inspired significant progress in the geometric theory of planar differential equations as well as bifurcation theory normal forms foliations and some topics in algebraic geometry the dramatic history of the problem as well as related developments are presented below 1 the problem and its counterparts what may be said about the number and location of limit cycles of a planar polynomial vector field of degree n the limit cycle is an isolated closed orbit of a vector field this second part of hilbert s 16th problem appears to be one of the most persistent in the famous hilbert list h second only to the riemann function conjecture traditionally hilbert s question is split into three each one requiring a stronger answer problem 1 is it true that a planar polynomial vector field has but a finite number of limit cycles problem 2 is it true that the number of limit cycles of a planar polynomial vector field is bounded by a constant depending on the degree of the polynomials only the bound on the number of limit cycles in problem 2 is denoted by h n and known as the hilbert number linear vector fields have no limit cycles hence h 1 0 it is still unknown whether or not h 2 exists problem 3 give an upper bound for h n a solution to any of these problems implies a solution for the previous ones only the first problem is solved now the positive answer was established in e92 i91 there are analytic counterparts of problems 1 and 2 received by the editors december 2001 2000 mathematics subject classification primary 34cxx 34mxx 37f75

effect of enhancement of spectral changes on speech intelligibility and clarity preferences for the hearing impaired most information in speech is carried in spectral changes over time rather than in static spectral shape per se a form of signal processing aimed at enhancing spectral changes over time was developed and evaluated using hearing impaired listeners the signal processing was based on the overlap add method and the degree and type of enhancement could be manipulated via four parameters two experiments were conducted to assess speech intelligibility and clarity preferences three sets of parameter values one corresponding to a control condition two types of masker steady speech spectrum noise and two talker speech and two signal to masker ratios smrs were used for each masker type generally the effects of the processing were small although intelligibility was improved by about 8 percentage points relative to the control condition for one set of parameter values using the steady noise masker at 6 db smr the processed signals were not preferred over those for the control condition except for the steady noise masker at 6 db smr further work is needed to determine whether tailoring the processing to the characteristics of the individual hearing impaired listener is beneficial

a phenomenological model for the responses of auditory nerve fibers i nonlinear tuning with compression and suppression a phenomenological model was developed to describe responses of high spontaneous rate auditory nerve an fibers including several nonlinear response properties level dependent gain compression bandwidth and phase properties were implemented with a control path that varied the gain and bandwidth of tuning in the signal path filter by making the bandwidth of the control path broad with respect to the signal path the wide frequency range of two tone suppression was included by making the control path filter level dependent and tuned to a frequency slightly higher than the signal path filter other properties of two tone suppression were also included these properties included the asymmetrical growth of suppression above and below the characteristic frequency and the frequency offset of the suppression tuning curve with respect to the excitatory tuning curve the implementation of this model represents a relatively simple phenomenological description of a single mechanism that underlies several important nonlinear response properties of an fibers the model provides a tool for studying the roles of these nonlinearities in the encoding of simple and complex sounds in the responses of populations of an fibers

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

normal approximants to binormal operators abstract let t tij i j 1 2 m be a block operator whose entries tij are commuting normal operators on a hilbert space we give a simple proof of the known fact that such operators can be reduced to an upper triangular form via a unitary conjugation our proof brings out some useful features of the triangular form when m 2 we find the closest normal operator to the binormal operator t with respect to every unitarily invariant norm this is a generalization of a result of j phillips who solved this approximation problem for the operator bound norm