h adaptive least squares finite element methods for the 2d stokes equations of any order with optimal convergence rates abstract the adaptive least squares finite element method ls fem for the stokes equations has recently been based on alternative error estimators in bringmann and carstensen 2016 for the lowest order case since the first order div ls fem measures the flux errors in h div the data resolution error measures the l 2 norm of the right hand side f minus the piecewise polynomial approximation k f without a mesh size factor this enforces separate marking with an overall abstract theory carstensen and rabus 2016 this paper contributes a a discussion of the scaling of the ls fem b optimal rates of the adaptive h version of any order k and c comparing numerical results in 2d with all details on inhomogeneous dirichlet data covered by the analysis

uniqueness of the multiplicative cyclotomic trace abstract making use of the theory of noncommutative motives we characterize the topological dennis trace map as the unique multiplicative natural transformation from algebraic k theory to topological hochschild homology thh and the cyclotomic trace map as the unique multiplicative lift through topological cyclic homology tc moreover we prove that the space of all multiplicative structures on algebraic k theory is contractible we also show that the algebraic k theory functor from small stable categories to spectra is lax symmetric monoidal which in particular implies that e n ring spectra give rise to e n 1 ring algebraic k theory spectra along the way we develop a multiplicative morita theory establishing a symmetric monoidal equivalence between the category of small idempotent complete stable categories and the morita localization of the category of spectral categories

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

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

nonmonotonicity of informational masking neff and green percept psychophys 41 409 415 1987 report that the masking of single tones by random frequency multitone maskers varies nonmonotonically with number of masker components peaking at 10 50 components in this paper it is shown that such results are well predicted by a model the component relative entropy model core wherein thresholds increase linearly with the ensemble variance of masker spectra smoothed by peripheral auditory filters r a lutfi j acoust soc am 94 748 758 1993 three experiments were conducted in the first the nonmonotonic relation was replicated for 9 of 11 listeners in conditions similar to those of neff and green in the second the frequencies of masker components were fixed and the levels of components were varied randomly across presentations to simulate gaussian noise in this case the nonmonotonicity and the total amount of masking for these listeners were shown to be significantly reduced in the third experiment masked thresholds for the signal were found to vary monotonically with the frequency spacing of masker components for a fixed number of masker components large individual differences among listeners were obtained in some experimental conditions individual as well as mean thresholds were well predicted by the core model with an appropriate selection of the values of the two free parameters of the model for each listener

factors affecting masking release for speech in modulated noise for normal hearing and hearing impaired listeners the speech reception threshold for sentences in stationary noise and in several amplitude modulated noises was measured for 8 normal hearing listeners 29 sensorineural hearing impaired listeners and 16 normal hearing listeners with simulated hearing loss this approach makes it possible to determine whether the reduced benefit from masker modulations as often observed for hearing impaired listeners is due to a loss of signal audibility or due to suprathreshold deficits such as reduced spectral and temporal resolution which were measured in four separate psychophysical tasks results show that the reduced masking release can only partly be accounted for by reduced audibility and that when considering suprathreshold deficits the normal effects associated with a raised presentation level should be taken into account in this perspective reduced spectral resolution does not appear to qualify as an actual suprathreshold deficit while reduced temporal resolution does temporal resolution and age are shown to be the main factors governing masking release for speech in modulated noise accounting for more than half of the intersubject variance their influence appears to be related to the processing of mainly the higher stimulus frequencies results based on calculations of the speech intelligibility index in modulated noise confirm these conclusions

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

a posteriori error estimators for the stokes equations we present two a posteriori error estimators for the mini element discretization of the stokes equations one is based on a suitable evaluation of the residual of the finite element solution the other one is based on the solution of suitable local stokes problems involving the residual of the finite element solution both estimators are globally upper and locally lower bounds for the error of the finite element discretization numerical examples show their efficiency both in estimating the error and in controlling an automatic self adaptive mesh refinement process the methods presented here can easily be generalized to the navier stokes equations and to other discretization schemes

auditory temporal integration and the power function model the auditory temporal integration function was studied with the objective of improving both its quantitative description and the specification of its principle independent variable stimulus duration in sec i temporal integration data from 20 studies were subjected to uniform analyses using standardized definitions of duration and two models of temporal integration analyses revealed that these data were best described by a power function model used in conjunction with a definition of duration termed assigned duration that de emphasized the rise fall portions of the stimuli there was a strong effect of stimulus frequency and in general the slope of the temporal integration function was less than 10 db per decade of duration i e a power function exponent less than 1 0 in sec ii an experimental study was performed to further evaluate the models and definitions detection thresholds were measured in 11 normal hearing human subjects using a total of 24 single burst and multiple burst acoustic stimuli of 3 125 khz the issues addressed are the quantitative description of the temporal integration function the definition of stimulus duration the similarity of the integration processes for single burst and multiple burst stimuli and the contribution of rise fall time to the integration process a power function in conjunction with the assigned duration definition was again most effective in describing the data single and multiple burst stimuli both seemed to be integrated by the same central mechanism with data for each type of stimulus being described by a power function exponent of approximately 0 6 at 3 125 khz it was concluded that the contribution of the rise fall portions of the stimuli can be factored out from the rest of the temporal integration process in sec iii the conclusions that emerged from the review of published work and the present experimental work suggested that auditory temporal integration is best described by a power function in conjunction with the assigned duration definition the exponent for the power function is typically less than 1 0 and varies with frequency and hearing level second a means of empirically assaying the contribution of the rise fall portions of the stimuli is presented and evaluated finally properties of a central auditory integrator are hypothesized abstract truncated at 250 words

median plane sound localization as a function of the number of spectral channels using a channel vocoder using a vocoder median plane sound localization performance was measured in eight normal hearing listeners as a function of the number of spectral channels the channels were contiguous and logarithmically spaced in the range from 0 3 to 16 khz acutely testing vocoded stimuli showed significantly worse localization compared to noises and 100 pulses click trains both of which were tested after feedback training however localization for the vocoded stimuli was better than chance a second experiment was performed using two different 12 channel spacings for the vocoded stimuli now including feedback training one spacing was from experiment 1 the second spacing called the speech localization spacing assigned more channels to the frequency range associated with speech there was no significant difference in localization between the two spacings however even with training localizing 12 channel vocoded stimuli remained worse than localizing virtual wideband noises by 4 8 degrees in local root mean square error and 5 2 in quadrant error rate speech understanding for the speech localization spacing was not significantly different from that for a typical spacing used by cochlear implant users these experiments suggest that current cochlear implants have a sufficient number of spectral channels for some vertical plane sound localization capabilities albeit worse than normal hearing listeners without loss of speech understanding

intensity discrimination for precedence effect stimuli when two identical stimuli are presented from two loudspeakers with a brief delay between them a single image is heard near the source of the leading sound the delayed sound or echo appears to be suppressed whereas the preceding sound determines perceived location hence the name the precedence effect this study investigated normal hearing listeners sensitivity to changes in the intensity of the lagging sound pairs of 2 ms white noise bursts with a 2 ms delay between the onsets of lead and lag were presented from two loudspeakers 45 left and right of midline in an anechoic chamber a 2afc procedure was used to test discrimination of intensity changes in the lead lag and both sounds together the untreated results showed discrimination to be poorest for changes in the lag stimulus however when the intensity differences were transformed into predictions of equivalent monaural level based on kemar measurements and binaural loudness summation discrimination for the lag was equal to the other two

efficient across frequency integration evidence from psychometric functions across frequency integration of complex signals was investigated by measuring psychometric functions log d versus signal level in db spl for detection of brief and long signals presented in broadband noise the signals were tones at 630 1600 and 4000 hz and a nine tone complex with components spaced at one third octave frequencies between 630 and 4000 hz the phase relationship of the components in the complex was varied such that adjacent components were in phase at 0 degrees 90 or 180 degrees out of phase signal durations defined in terms of the number of cycles between the half amplitude points of the gaussian envelopes of 4 7 and 150 cycles were tested results for six normal hearing listeners showed that the slopes of the psychometric functions were steeper for the brief than for the long signals and steeper for the tone complexes than for the tones particularly for the brief signals this suggests that the transformation from signal intensity to decision variable may be different fo

speech intelligibility is best predicted by intensity not cochlea scaled entropy cochlea scaled entropy cse is a measure of spectro temporal change that has been reported to predict the contribution of speech segments to overall intelligibility this paper confirms that cse is highly correlated with intensity making it impossible to determine empirically whether it is cse or simply intensity that determines speech importance a more perceptually relevant version of cse that uses db scaled differences rather than differences in linear amplitude failed to predict speech intelligibility overall a parsimonious account of the available data is that the importance of speech segments to overall intelligibility is best predicted by their relative intensity not by cse

simulation of the effect of threshold elevation and loudness recruitment combined with reduced frequency selectivity on the intelligibility of speech in noise the effect of loudness recruitment and threshold elevation together with reduced frequency selectivity have been simulated to examine the combined effect of the two major consequences of cochlear hearing loss on the intelligibility of speech in speech shaped noise in experiment 1 four conditions were simulated a moderate flat loss with auditory filters broadened by a factor of three b3r2 a moderate to severe sloping loss with auditory filters broadened by a constant factor of three b3rx and these conditions with linear amplification applied prior to the simulation processing b3r2 b3rx for conditions b3r2 and b3rx performance was markedly worse than for a control condition normal hearing condition r1 tested in a previous study for conditions b3r2 and b3rx linear amplification improved performance considerably however performance remained below that for condition r1 by between 5 and 19 in experiment 2 the broadening of the auditory filters was made more realistic by making it a function of the absolute threshold at the center frequency of the auditory filter three different hearing losses were simulated a moderate to severe sloping loss with variable broadening of the auditory filters bxrx the same moderate to severe sloping loss with linear amplification bxrx and the same broadening of the auditory filters but without the simulation of loudness recruitment and threshold elevation bx for condition bxrx performance was markedly worse than in condition r1 while performance in condition bx was somewhat worse than for condition r1 for condition bxrx linear amplification according to the nal procedure improved performance to a large extent but it remained worse than for condition r1 the results are consistent with previous evidence indicating that only part of the decrease of performance produced by actual cochlear hearing loss can be compensated by conventional linear hearing aids

level discrimination of tones as a function of duration difference limens for level ls db 20 log p p p where p is the pressure were measured as a function of duration for tones at 250 500 and 8000 hz stimulus durations ranged from 2 ms to 2 s and the stimulus power was held constant rise and fall times were 1 ms the interstimulus interval was 250 ms at each frequency three levels were tested 85 65 and approximately 40 db spl an adaptive two alternative forced choice procedure with feedback was used for three normal listeners ls decreased as duration increased up to at least 2 s except at 250 hz at 250 hz l stopped decreasing at durations between 0 5 and 1 s in a double logarithmic plot of l versus duration the rate of decrease is generally well fitted by a sloping line the average slope is 0 28 it is steeper at high levels than at low levels because the average slope is shallower than the 0 5 slope predicted for an optimum detector it may be that fast adaptation of auditory nerve activity and or memory effects interfere w

informational masking by everyday sounds tone in noise detection is severely degraded when only a few spectral components of the noise are presented at random on each trial neff and green percept psychophys 41 409 415 1987 the elevations in threshold are attributed to uncertainty regarding the noise caused by sparse sampling of noise components informational masking is the term used to describe the result the present experiment was undertaken to determine how informational masking is affected when sparse sampling is from a set of common everyday sounds instead of noise on each presentation a different masker waveform of constant total power was synthesized from the magnitude and phase of a fixed number m of spectral components m 2 921 across blocks of trials the components were selected at random from 1 of 50 common environmental sounds e g baby crying door slamming phone ringing or 1 of 50 samples of gaussian noise masked thresholds for a 1 0 khz signal in the presence of both types of maskers were obtained using an adaptive two interval forced choice procedure results with noise replicated earlier results showing largest elevations in threshold for 10 20 sampled components results with everyday sounds showed a similar pattern with thresholds elevated above those for noise by as much as 10 db at the larger values of m the differences in masked thresholds were systematically related to differences in the ensemble variance of masker spectra as predicted by a model previously applied to noise lutfi j acoust soc am 94 748 758 1993 not predicted was the result of a subsequent trial by trial analysis in which 9 11 db less masking was observed for samples from everyday sounds rated as easily recognized by listeners the results suggest that listeners fail to exploit lawful dependencies among spectral components of everyday sounds to aid detection unless enough information is available for the sound to be easily recognized

psychometric functions for level discrimination to determine the form of psychometric functions for 2i 2afc level discrimination ten increment levels were presented in random order within blocks of 100 trials stimuli were chosen to encompass a wide range of conditions and difference limens eight 10 ms tones had frequencies of 0 25 1 8 or 14 khz and levels of 30 60 or 90 db spl two 500 ms stimuli also were tested a 1 khz tone at 90 db spl and a white noise at 63 db spl for each condition at least 20 blocks were presented in mixed order results for three normal listeners show that the sensitivity d is nearly proportional to l 20 log p p p where p is pressure over the entire range of difference limens when d is plotted against weber fractions for pressure p p or intensity i i exponents of the best fitting power functions decrease with increasing difference limens and are less than one for large difference limens these results indicate that the transformation from stimulus intensity to decision variable may be approximately lo

on the algebraic k theory of the complex k theory spectrum let p 5 be a prime let ku be the connective complex k theory spectrum and let k ku be the algebraic k theory spectrum of ku in this paper we study the p primary homotopy type of the spectrum k ku by computing its mod p v1 homotopy groups we show that up to a finite summand these groups form a finitely generated free module over the polynomial algebra mathbb f p b where b is a class of degree 2p 2 defined as a higher bott element

contributions of cochlea scaled entropy and consonant vowel boundaries to prediction of speech intelligibility in noise recent evidence suggests that spectral change as measured by cochlea scaled entropy cse predicts speech intelligibility better than the information carried by vowels or consonants in sentences motivated by this finding the present study investigates whether intelligibility indices implemented to include segments marked with significant spectral change better predict speech intelligibility in noise than measures that include all phonetic segments paying no attention to vowels consonants or spectral change the prediction of two intelligibility measures normalized covariance measure ncm coherence based speech intelligibility index csii is investigated using three sentence segmentation methods relative root mean square rms levels cse and traditional phonetic segmentation of obstruents and sonorants while the cse method makes no distinction between spectral changes occurring within vowels consonants the rms level segmentation method places more emphasis on the vowel consonant boundaries wherein the spectral change is often most prominent and perhaps most robust in the presence of noise higher correlation with intelligibility scores was obtained when including sentence segments containing a large number of consonant vowel boundaries than when including segments with highest entropy or segments based on obstruent sonorant classification these data suggest that in the context of intelligibility measures the type of spectral change captured by the measure is important

the completion of locally refined simplicial partitions created by bisection recently in found comput math 7 2 2007 245 269 we proved that an adaptive finite element method based on newest vertex bisection in two space dimensions for solving elliptic equations which is essentially the method from sinum 38 2000 466 488 by morin nochetto and siebert converges with the optimal rate the number of triangles n in the output partition of such a method is generally larger than the number m of triangles that in all intermediate partitions have been marked for bisection because additional bisections are needed to retain conforming meshes a key ingredient to our proof was a result from numer math 97 2004 219 268 by binev dahmen and devore saying that n n 0 cm for some absolute constant c where n 0 is the number of triangles from the initial partition that have never been bisected in this paper we extend this result to bisection algorithms of n simplices with that generalizing the result concerning optimality of the adaptive finite element method to general space dimensions

estimates of effective frequency selectivity based on the detection of a tone added to complex maskers in experiment 1 the validity of parameters associated with the roex p r auditory filter shape was examined for three different types of maskers a a noise masker b a random 12 tone masker whose frequencies varied on a burst by burst basis multiple burst different mbd and c a random 12 tone masker whose frequencies were the same across bursts multiple burst same mbs first the power spectrum model of masking was used to estimate auditory filter shapes for four observers second the resulting auditory filter shapes were used in a computer simulation that provided an estimate of internal noise for each observer third relative weights across frequency were estimated for each observer and each masker type for the noise masker these analyses provided predictions and relative weights that were consistent across the three analyses for the mbd and mbs maskers there was little consistency neither the estimated internal noise nor the estimated relative weights reliably supported a single f

an a posteriori error estimate for a first kind integral equation in this paper we present a new a posteriori error estimate for the boundary element method applied to an integral equation of the first kind the estimate is local and sharp for quasi uniform meshes and so improves earlier work of ours the mesh dependence of the constants is analyzed and shown to be weaker than expected from our previous work besides the galerkin boundary element method the collocation method and the qualocation method are considered a numerical example is given involving an adaptive feedback algorithm

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

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

detection and intensity discrimination of gaussian shaped tone pulses as a function of duration j acoust soc am 105 3425 3435 1999 proposed that the auditory system partitions the spectro temporal domain into frequency time f t windows and that the characteristics of these windows could be explored by measuring intensity discrimination for gaussian shaped tone pulses presented just above their detection threshold in noise they reasoned that for a long duration tone pulse the auditory representation would be maximally compact in the frequency domain but would spread across several f t windows in the time domain for a very short duration tone pulse the auditory representation would be maximally compact in the time domain but would spread across several f t windows in the frequency domain there should be some intermediate duration at which the auditory representation is compact in both the time and frequency domains and for which intensity discrimination performance should worsen due to the limited opportunity for multiple looks their data for signal frequencies of 1 and 4 khz were consistent with this expectation intensity discrimination was poorest at a duration of about 3 5 ms at 1 khz and 1 ms at 4 khz durations are specified between 6 8 db down points on the envelope this experiment attempted to replicate those results and to extend them to a wider range of frequencies and levels intensity discrimination of gaussian shaped tone pulses was measured at three levels 10 db above absolute threshold or above masked threshold in a pink noise with a spectrum level of either 15 or 40 db at 1 khz the signal frequency was 0 25 khz durations from 2 to 320 ms 1 khz durations from 0 5 to 80 ms or 4 khz durations from 0 1 to 20 ms three normally hearing subjects were tested at 1 and 4 khz performance was poorest overall for the 15 db pink noise level and thresholds showed a peak at intermediate durations about 3 5 ms at 1 khz and 1 ms at 4 khz such peaks were still apparent but smaller in the no noise condition and were almost absent at the higher noise level for the 0 25 khz signal frequency peaks were not observed consistently at any level although two subjects showed small peaks for durations around 10 ms an explanation is offered for the results in terms of the level and frequency dependence of basilar membrane input output functions

quasi optimal adaptive pseudostress approximation of the stokes equations the pseudostress velocity formulation of the stationary stokes problem allows some quasi optimal raviart thomas mixed finite element formulation for any polynomial degree the adaptive algorithm employs standard residual based explicit a posteriori error estimation from carstensen kim and park siam j numer anal 49 2011 pp 2501 2523 for the lowest order raviart thomas finite element functions in a simply connected lipschitz domain this paper proves optimal convergence rates in terms of the number of unknowns of the adaptive mesh refining algorithm based on the concept of approximation classes the proofs use some novel equivalence to first order nonconforming crouzeix raviart discretization plus a particular helmholtz decomposition of deviatoric tensors

convergence of natural adaptive least squares finite element methods the first order div least squares finite element methods provide inherent a posteriori error estimator by the elementwise evaluation of the functional in this paper we prove q linear convergence of the associated adaptive mesh refining strategy for a sufficiently fine initial mesh with some sufficiently large bulk parameter for piecewise constant right hand sides in a poisson model problem the proof relies on some modification of known supercloseness results to non convex polygonal domains plus the flux representation formula the analysis is carried out for the lowest order case in two dimensions for the simplicity of the presentation

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

preconditioning in h div and applications we consider the solution of the system of linear algebraic equations which arises from the finite element discretization of boundary value problems associated to the differential operator i grad div the natural setting for such problems is in the hilbert space h div and the variational formulation is based on the inner product in h div we show how to construct preconditioners for these equations using both domain decomposition and multigrid techniques these preconditioners are shown to be spectrally equivalent to the inverse of the operator as a consequence they may be used to precondition iterative methods so that any given error reduction may be achieved in a finite number of iterations with the number independent of the mesh discretization we describe applications of these results to the efficient solution of mixed and least squares finite element approximations of elliptic boundary value problems

an optimal adaptive mixed finite element method various applications in fluid dynamics and computational continuum mechanics motivate the development of reliable and efficient adaptive algorithms for mixed finite element methods in order to save degrees of freedom not all but just some selected set of finite element domains are refined hence the fundamental question of convergence as well as the question of optimality require new mathematical arguments the presented adaptive algorithm for raviart thomas mixed finite element methods solves the poisson model problem with optimal convergence rate chen holst and xu presented convergence and optimality of adaptive mixed finite element methods 2008 following arguments of rob stevenson for the conforming finite element method their algorithm reduces oscillations separately before approximating the solution by some adaptive algorithm in the spirit of w dorfler 1996 the algorithm proposed here appears more natural in switching to either reduction of the edge error estimator or of the oscillations

level dependence of auditory filters in nonsimultaneous masking as a function of frequency auditory filter bandwidths were measured using nonsimultaneous masking as a function of signal level between 10 and 35db sl for signal frequencies of 1 2 4 and 6khz the brief sinusoidal signal was presented in a temporal gap within a spectrally notched noise two groups of normal hearing subjects were tested one using a fixed masker level and adaptively varying signal level the other using a fixed signal level and adaptively varying masker level in both cases auditory filters were derived by assuming a constant filter shape for a given signal level the filter parameters derived from the two paradigms were not significantly different at 1khz the equivalent rectangular bandwidth erb decreased as the signal level increased from 10to20db sl after which it remained roughly constant in contrast at 6khz the erb increased consistently with signal levels from 10to35db sl the results at 2 and 4khz were intermediate showing no consistent change in erb with signal level overall the results sugge

comparing the perceptual contributions of cochlear scaled entropy and speech level cochlear scaled entropy cse has been suggested to be a reliable predictor of speech intelligibility previous studies showed that speech segments with high root mean square rms levels h levels contained primarily vowels which carry important information for speech recognition the present work compared the contributions of high cse h entropy and h level segments to speech intelligibility the natural speech was edited to generate two types of noise replaced stimuli which preserved the same percentages of largest cse segments and highest rms level segments and played to normal hearing listeners in a recognition experiment experimental results showed that the nature of the noise replaced stimulus h entropy and h level made a small difference in intelligibility performance cses and rms levels showed a moderately high correlation r 0 79 suggesting that many speech segments may have both large cses and high rms levels which might account partially for the small intelligibility difference between the two types of stimuli in addition the vowel duration proportion differed between h entropy and h level segments of the same length suggesting that vowels play different roles in contributing to the intelligibility of h entropy and h level stimuli

auditory filters and the benefit measured from spectral enhancement algorithms designed to improve speech intelligibility for those with sensorineural hearing loss snhl by enhancing peaks in a spectrum have had limited success since testing of such algorithms cannot separate the theory of the design from the implementation itself the contribution of each of these potentially limiting factors is not clear therefore psychophysical paradigms were used to test subjects with either normal hearing or snhl in detection tasks using well controlled stimuli to predict and assess the limits in performance gain from a spectrally enhancing algorithm a group of normal hearing nh and hearing impaired hi subjects listened in two experiments auditory filter measurements and detection of incremented harmonics in a harmonic spectrum the results show that nh and hi subjects have an improved ability to detect incremented harmonics when there are spectral decrements surrounding the increment various decrement widths and depths were compared against subjects equivalent rectangula

least squares methods for incompressible newtonian fluid flow linear stationary problems this paper develops and analyzes two least squares methods for the numerical so lution of linear stationary incompressible newtonian fluid flow in two and three dimensions both approaches use the l 2 norm to define least squares functionals one is based on the stress velocity formulation see section 3 2 and it applies to general boundary conditions the other is based on an equivalent formulation for the pseudostress and velocity see section 4 2 and it applies to pure velocity dirichlet boundary conditions the velocity gradient and vorticity can be obtained alge braically from this new tensor variable it is shown that the homogeneous least squares functionals are elliptic and continuous in the h div d h 1 d norm this immediately implies optimal error estimates for conforming finite element approximations as well it admits optimal multigrid solution methods if raviart thomas finite element spaces are used to approximate the stress or the pseudostress tensor

relationship between frequency selectivity and frequency discrimination measured in subjects with unilateral and bilateral cochlear impairments frequency selectivity refers to the filtering processes which underlie our ability to detect a sound of one frequency in the presence of another sound with a different frequency frequency discrimination is our ability to detect changes in frequency over time it is a basic tenet of the place theory that frequency discrimination depends upon frequency selectivity in this paper we describe an experimental test of a place model put forward by zwicker 1956 in which frequency discrimination is explained in terms of excitation patterns the model is illustrated schematically in fig 1 which shows excitation patterns calculated using the method described by moore and glasberg 1983 for two sinusoids with frequencies 1180 and 1220 hz zwicker assumed that the frequencies of the two sinusoids can be discriminated if their excitation patterns differ at any point by more than a criterion amount he assumed a criterion of 1 db

representation and character theory in 2 categories we develop a 2 categorical generalization of the theory of group representations and characters we categorify the concept of the trace of a linear transformation associating to any endofunctor of any small category a set called its categorical trace in a linear situation the categorical trace is a vector space and we associate to any two commuting self equivalences a number called their joint trace for a group acting on a linear category we define an analog of the character which is the function on commuting pairs of group elements given by the joint traces of the corresponding functors we call this function the 2 character of such functions of commuting pairs and more generally n tuples appear in the work of hopkins kuhn and ravenel michael j hopkins nicholas j kuhn douglas c ravenel generalized group characters and complex oriented cohomology theories j amer math soc 13 3 2000 553 594 electronic on equivariant morava e theories we define the concept of induced categorical representation and show that the corresponding 2 character is given by the same formula as was obtained in michael j hopkins nicholas j kuhn douglas c ravenel generalized group characters and complex oriented cohomology theories j amer math soc 13 3 2000 553 594 electronic for the transfer map in the second equivariant morava e theory

axioms of adaptivity this paper aims first at a simultaneous axiomatic presentation of the proof of optimal convergence rates for adaptive finite element methods and second at some refinements of particular questions like the avoidance of discrete lower bounds inexact solvers inhomogeneous boundary data or the use of equivalent error estimators solely four axioms guarantee the optimality in terms of the error estimators compared to the state of the art in the temporary literature the improvements of this article can be summarized as follows first a general framework is presented which covers the existing literature on optimality of adaptive schemes the abstract analysis covers linear as well as nonlinear problems and is independent of the underlying finite element or boundary element method second efficiency of the error estimator is neither needed to prove convergence nor quasi optimal convergence behavior of the error estimator in this paper efficiency exclusively characterizes the approximation classes involved in terms of the best approximation error and data resolution and so the upper bound on the optimal marking parameters does not depend on the efficiency constant third some general quasi galerkin orthogonality is not only sufficient but also necessary for the r linear convergence of the error estimator which is a fundamental ingredient in the current quasi optimality analysis due to stevenson 2007 finally the general analysis allows for equivalent error estimators and inexact solvers as well as different non homogeneous and mixed boundary conditions

decrement detection in normal and impaired ears the smallest detectable duration of a brief decrement in the intensity of wideband noise was measured as a function of the depth of the decrement in the first experiment conditions were tested in which the noise before the decrement was more intense than the noise after the decrement and vice versa these data were used to estimate the shape of an intensity weighting function or temporal window describing the temporal resolution of the ear the equivalent rectangular durations erds of the temporal windows measured in this way had values of about 5 5 4 6 and 6 6 ms for noise spectrum levels of 10 30 and 50 db respectively in a second experiment decrement detection was measured in subjects with unilateral sensorineural hearing loss one set of thresholds was measured in the impaired ear and two sets of thresholds were measured in the normal ear one with the noise level at equal spl to the level in the impaired ear and one with the noise at equal sl temporal window shapes were also estimated from these data only one of the subjects showed reduced temporal resolution in the impaired ear the other two subjects having similar erd values for all three conditions

the additivity of traces in triangulated categories abstract we explain a fundamental additivity theorem for euler characteristics and generalized trace maps in triangulated categories the proof depends on a refined axiomatization of symmetric monoidal categories with a compatible triangulation the refinement consists of several new axioms relating products and distinguished triangles the axioms hold in the examples and shed light on generalized homology and cohomology theories

indices of frequency and temporal resolution as a function of level several investigators have suggested that between subject variations in hearing aid benefit may be related to between subject variations in auditory resolution evaluation of this relationship is complicated by the need to assess resolution at several frequencies and levels in hearing impaired listeners who may not tolerate lengthy test procedures this paper reports data from normal hearing listeners for abbreviated indices of frequency and temporal resolution for a particular frequency stimulus level combination both indices may be derived from three pure tone thresholds obtained with different masking stimuli norms are reported as a function of masker level for each index at four test frequencies changes in the indices as a function of masker level were consistent with expectations based on related studies by other investigators norms were obtained using a supra aural earphone comparison of results from supra aural and insert earphones suggested that low frequency sound leakage substantially affected the norms for 500 hz reliability of individual indices was good to excellent

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

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

efficient across frequency integration in short signal detection a series of experiments was performed on the influence of bandwidth on the masked threshold of brief deterministic signals in continuous broadband noise the signal bandwidth is quantified by the number n of constituent 1 3 oct bands for n increasing from 1 to typically 9 the masked threshold level in the constituent 1 3 oct bands is found to decrease by 8 log n this integration rule is obtained when each of the 1 3 oct bands covered by the signal equally contributes to detection i e that for each of these 1 3 oct bands the difference between signal level and the individual masked threshold level is the same it was found that this integration rule also applies to noncontiguous signal spectra and that it remains intact over a broad range of masker levels commonly the masked threshold of compound signals for instance n frequency components with a spacing of typically 1 3 oct relative to the masked threshold of single component signals has been described by a 5 log n integration rule however this rule was obtained for signal durations of typically 100 ms or more for the present brief signals typically 10 ms or less the across frequency integration is found to be more effective

gluing semi orthogonal decompositions abstract we introduce preordered semi orthogonal decompositions psod s of dg categories we show that homotopy limits of dg categories equipped with compatible psod s carry a natural psod this gives a way to glue semi orthogonal decompositions along faithfully flat covers extending the main result of 4 as applications we will construct semi orthogonal decompositions for root stacks of log pairs x d where d is a not necessarily simple normal crossing divisor generalizing results from 17 and 3 further we will compute the kummer flat k theory of general log pairs x d generalizing earlier results of hagihara and niziol in the simple normal crossing case 15 23

additivity of auditory masking using gaussian shaped tones both temporal and spectral masking have been studied extensively in the literature mostly they have been regarded as separate phenomena very little is known about the interaction between these two effects i e masking in the time frequency domain data on the time frequency spread of masking evoked by a single gaussian shaped tone pulse are presented in an accompanying study at the same conference necciari et al the current study gathers data on the additivity of masking by up to four approximately equally effective gaussian maskers erb 600 hz separated either along the time or the frequency axis for temporal separation the amount of masking increases with the number of maskers with excess masking exceeding linear additivity of up to 25 db for frequency separation preliminary data excess masking amounts up to 15 db and the higher frequency masker relative to the target contributes more to the additivity than the lower frequency maskers experiments with multiple maskers combining both temporal and frequency separation are underway combined with the single maker data these data may serve as a basis for modeling time frequency masking effects in complex signals work partly supported by oead wtz project amadeus and anr

excitation based and informational masking of a tonal signal in a four tone masker this study examined contributions of peripheral excitation and informational masking to the variability in masking effectiveness observed across samples of multi tonal maskers detection thresholds were measured for a 1000 hz signal presented simultaneously with each of 25 four tone masker samples using a two interval forced choice adaptive task thresholds were measured with each sample fixed throughout trial blocks for ten listeners average thresholds differed by as much as 26 db across samples an excitation based model of partial loudness moore b c j et al 1997 j audio eng soc 45 224 237 was used to predict thresholds these predictions accounted for a significant portion of variance in the data of several listeners but no relation between the model and data was observed for many listeners moreover substantial individual differences on the order of 41 db were observed for some maskers the largest individual differences were found for maskers predicted to produce minimal excitat

bayesian adaptive estimation of the auditory filter a bayesian adaptive procedure for estimating the auditory filter shape was proposed and evaluated using young normal hearing listeners at moderate stimulus levels the resulting quick auditory filter qaf procedure assumed the power spectrum model of masking with the auditory filter shape being modeled using a spectrally symmetric two parameter rounded exponential roex function during data collection using the qaf procedure listeners detected the presence of a pure tone signal presented in the spectral notch of a noise masker dependent on the listener s response on each trial the posterior probability distributions of the model parameters were updated and the resulting parameter estimates were then used to optimize the choice of stimulus parameters for the subsequent trials results showed that the qaf procedure gave similar parameter estimates to the traditional threshold based procedure in many cases and was able to reasonably predict the masked signal thresholds additional measurements suggested that occasional failures of the qaf procedure to reliably converge could be a consequence of incorrect responses early in a qaf track the addition of a parameter describing lapses of attention reduced the likelihood of such failures

effects of frequency on the detection of decrements and increments in sinusoids thresholds for the detection of decrements in level of sinusoidal signals were measured as a function of decrement duration and of frequency 250 1000 and 4000 hz in nine normally hearing subjects thresholds for detecting a brief increment in level were also measured the sinusoids were presented in a background noise intended to mask spectral splatter associated with the decrement or increment performance tended to improve with increasing frequency for all decrement durations and for increment detection the results were analyzed using a model consisting of an array of bandpass filters the auditory filters each followed by a nonlinearity a sliding temporal integrator and a decision device the analysis indicated that the worsening in performance with decreasing frequency can be attributed mainly to changes in the efficiency of the decision device following the temporal integrator at lower frequencies a larger change is required at the output of the integrator for threshold to be reached at 25

masked threshold difference due to masker harmonicity uniform modulation rates of auditory filter outputs versus periodicity in waveform fine structure in a previous paper treurniet and boucher j acoust soc am 109 306 320 2001 a model was presented which successfully accounted for the lower threshold obtained using a harmonic masker instead of a similar inharmonic masker for the harmonic masker the frequencies of partials were equally spaced so the envelope modulation rates of auditory filter outputs were invariant across filters however for the inharmonic masker the intervals between adjacent partials were not equal and this resulted in a decreased uniformity of modulation rates across filters the model proposed that the lower uniformity interferes with detection of a probe induced change in the modulation rates thus accounting for the masked threshold difference this paper shows that an inharmonic masker yields thresholds similar to a harmonic masker provided that the modulation rates are uniform across the auditory filters involved in processing the masker and probe thus the lowered threshold associated with a harmonic masker appears to arise from uniform modulation rates across auditory filters and does not require periodicity in the fine structure of the waveform produced by partial frequencies that are integer multiples of a fundamental

rapid estimation of high parameter auditory filter shapes a bayesian adaptive procedure the quick auditory filter qaf procedure was used to estimate auditory filter shapes that were asymmetric about their peaks in three experiments listeners who were naive to psychoacoustic experiments detected a fixed level pure tone target presented with a spectrally notched noise masker the qaf procedure adaptively manipulated the masker spectrum level and the position of the masker notch which was optimized for the efficient estimation of the five parameters of an auditory filter model experiment i demonstrated that the qaf procedure provided a convergent estimate of the auditory filter shape at 2 khz within 150 to 200 trials approximately 15 min to complete and for a majority of listeners excellent test retest reliability in experiment ii asymmetric auditory filters were estimated for target frequencies of 1 and 4 khz and target levels of 30 and 50 db sound pressure level the estimated filter shapes were generally consistent with published norms especially at the low target level it is known that the auditory filter estimates are narrower for forward masking than simultaneous masking due to peripheral suppression a result replicated in experiment iii using fewer than 200 qaf trials

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

op lax natural transformations twisted quantum field theories and even higher morita categories abstract motivated by the challenge of defining twisted quantum field theories in the context of higher categories we develop a general framework for lax and oplax transformations and their higher analogs between strong n functors we construct a double n category built out of the target n category governing the desired diagrammatics we define op lax transformations as functors into parts thereof and an op lax twisted field theory to be a symmetric monoidal op lax natural transformation between field theories we verify that lax trivially twisted relative field theories are the same as absolute field theories as a second application we extend the higher morita category of e d algebras in a symmetric monoidal n category c to an n d category using the higher morphisms in c

an adaptive least squares fem for linear elasticity with optimal convergence rates adaptive mesh refining is of particular importance in computational mechanics and established here for the lowest order locking free least squares finite element scheme which solely employs conforming p 1 approximations for the displacement and lowest order raviart thomas approximations for the stress variables this forms a competitive discretization in particular in three dimensional linear elasticity with traction boundary conditions although the stress approximation does not satisfy the symmetry condition exactly the paper introduces an adaptive mesh refining algorithm based on separate marking and exact solve with some novel explicit a posteriori error estimator and proves optimal convergence rates the point is robustness in the sense that the crucial input parameters theta for the dorfler marking and kappa for the separate marking as well as the equivalence constants in the asymptotic convergence rates do not degenerate as the lame parameter lambda tends to infty

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

effect of masker level on overshoot overshoot refers to the phenomenon where signal detectability improves for a short duration signal as the onset of that signal is delayed relative to the onset of a longer duration masker a popular explanation for overshoot is that it reflects short term adaptation in auditory nerve fibers in this study overshoot was measured for a 10 ms 4 khz signal masked by a broadband noise in the first experiment masker duration was 400 ms and signal onset delay was 1 or 195 ms masker spectrum level ranged from 10 50 db spl overshoot was negligible at the lowest masker levels grew to about 10 15 db at the moderate masker levels but declined and approached 0 db at the highest masker levels in the second experiment the masker duration was reduced to 100 ms and the signal was presented with a delay of 1 or 70 ms masker spectrum level was 10 30 or 50 db spl overshoot was about 10 db for the two lower masker levels but about 0 db at the highest masker level the results from the second experiment suggest that the decline in overshoot at high masker levels is probably not due to auditory fatigue it is suggested instead that the decline may be attributable to the neural response at high levels being dominated by those auditory nerve fibers that do not exhibit short term adaptation i e those with low spontaneous rates and high thresholds

fast computation in adaptive tree approximation adaptive methods of approximation arise in many settings including numerical methods for pdes and image processing they can usually be described by a tree which records the adaptive decisions this paper is concerned with the fast computation of near optimal trees based on n adaptive decisions the best tree based on n adaptive decisions could be found by examining all such possibilities however this is exponential in n and could be numerically prohibitive the main result of this paper is to show that it is possible to find near optimal trees using computations linear in n

rapid efficient encoding of correlated complex acoustic properties early sensory processing is long thought to extract and exploit redundancy and regularity in the input but perceptual evidence supporting this hypothesis has been limited experiments are reported demonstrating efficient encoding of multiple acoustic attributes resulting in the reduction of perceptual dimensionality stimuli varied across two complex independent dimensions attack decay ad and spectral shape ss a subset of 24 stimuli captured a near perfect correlation between these dimensions r2 0 95 participants passively listened to 25 presentations of these correlated stimuli over 7 5 min before discriminating sound pairs in axb trials without feedback for trials where both ss and ad varied listeners successfully discriminated sounds consistent with the correlation of ss and ad during exposure but they were at chance discriminating pairs varying orthogonal to the correlation presented in exposure after exposure participants were also significantly poorer at detecting differences among s

convergence and optimality of adaptive least squares finite element methods the first order div least squares finite element methods lsfems allow for an immediate a posteriori error control by the computable residual of the least squares functional this paper establishes an adaptive refinement strategy based on some equivalent refinement indicators since the first order div lsfem measures the flux errors in h div the data resolution error measures the l 2 norm of the right hand side f minus the piecewise polynomial approximation pi f without a mesh size factor hence the data resolution term is neither an oscillation nor of higher order and consequently requires a particular treatment e g by the thresholding second algorithm due to binev and devore the resulting novel adaptive lsfem with separate marking converges with optimal rates relative to the notion of a nonlinear approximation class

k theory and the bridge from motives to noncommutative motives abstract in this work we present a new approach to the theory of noncommutative motives and use it to explain the different flavors of algebraic k theory of schemes and dg categories the work is divided into three main parts in the first part we use the techniques of higher algebra developed in 63 to provide a universal characterization for the symmetric monoidal 1 category underlying the motivic stable a 1 homotopy theory of morel voevodsky 107 67 more precisely given a symmetric monoidal model category v together with an object x v we characterize the underlying symmetric monoidal 1 category of the symmetric monoidal model category sp v x introduced by hovey in 43 by means of a universal property amongst symmetric monoidal 1 categories this characterization trivializes the problem of finding motivic monoidal realizations in the second part we introduce a new approach to the theory of noncommutative motives by constructing a stable motivic homotopy theory for the noncommutative spaces of kontsevich 56 55 54 the key ingredient is a notion of nisnevich topology in the noncommutative setting compatible with the classical notion this compatibility together with the universal property proved in the first part ensures the existence of a canonical monoidal map from the stable motivic theory of morel voevodsky towards these new noncommutative motives that allow us to compare the two theories in the last part of this paper we explain how this bridge can be used to explain the various flavors of algebraic k theory of dg categories more precisely we prove that the non connective k theory of dg categories introduced by schlichting 82 is the non commutative nisnevich sheafification of connective algebraic k theory then we prove that its further non commutative a 1 localization is a tensor unit in our noncommutative motives as a corollary we obtain a precise proof for an original conjecture of kontsevich claiming that k theory gives the correct mapping spaces in noncommutative motives our major application is the discovery of a canonical factorization of our motivic bridge through the 1 category of modules over the commutative algebra object representing homotopy invariant algebraic k theory of schemes the results in 77 imply that this bridge is fully faithful over a field k with resolutions of singularities so that at the motivic level no information below k theory is lost by passing to the noncommutative world

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

sample discrimination of frequency differences with distracters this study examined the effect of the type and frequency range of remote frequency distracters on sample discrimination of frequency differences sd f for baseline sd f conditions normal hearing listeners judged frequency differences between pairs of target tones drawn from gaussian frequency distributions near 2000 hz in experiment 1 the distracters were pairs of random frequency tones fixed frequency tones or noise bands with one distracter above and one below the target region three frequency separations of targets and distracters were tested none overlapping the target region effects of fixed frequency or noise band distracters were small compared to that of random frequency distracters which drove performance to near chance in experiment 2 dominance of the low frequency distracter was supported by the effects of changing distracter level by presenting only the higher or lower frequency distracter and by the pattern of weights derived from trial by trial responses performance recovered only when the lower frequency distracter was attenuated 40 50 db relative to the targets in experiment 3 all stimulus distributions were shifted 2 octaves higher in frequency the stronger influence of the distracter frequency below the target remained the results demonstrate the importance of both stimulus variability and frequency relationships in the interaction of targets and distracters for sd f

optimal adaptive nonconforming fem for the stokes problem this paper presents an optimal nonconforming adaptive finite element algorithm and proves its quasi optimal complexity for the stokes equations with respect to natural approximation classes the proof does not explicitly involve the pressure variable and follows from a novel discrete helmholtz decomposition of deviatoric functions

auditory filter shapes of cba caj mice behavioral assessments auditory filter shape and frequency tuning may be derived by measuring changes in pure tone thresholds as a function of the bandwidth of notched noise maskers when these psychophysical methods were applied to cba caj mice the resulting filter shapes were well fit by roex p r functions originally developed for human subjects the equivalent rectangular bandwidths erbs of the filter shapes ranged from 16 to 19 of test frequencies between 8 to 16khz these erbs correspond well to the performance of humans at high frequencies and the limited number of mammalian species that have been characterized with notched noise procedures frequency tuning was maintained throughout most of the adult lifespan and then showed a selective high frequency loss at ages beyond 2 years these results suggest that auditory filtering effects in adult cba caj mice are similar to normal processes in other mammalian species and provide an excellent model of human presbycusis when they begin to degrade in aging individuals

higher traces noncommutative motives and the categorified chern character abstract we propose a categorification of the chern character that refines earlier work of toen and vezzosi and of ganter and kapranov if x is an algebraic stack our categorified chern character is a symmetric monoidal functor from a category of mixed noncommutative motives over x which we introduce to s 1 equivariant perfect complexes on the derived free loop stack l x as an application of the theory we show that toen and vezzosi s secondary chern character factors through secondary k theory our techniques depend on a careful investigation of the functoriality of traces in symmetric monoidal n categories which is of independent interest

effects of level and frequency on the detection of decrements and increments in sinusoids thresholds for the detection of decrements in level of sinusoidal signals were measured as a function of decrement duration level 25 40 55 and 70 db spl and frequency 250 1000 and 4000 hz in eleven normally hearing subjects thresholds for detecting a brief increment in level were also measured the sinusoids were presented in a background noise intended to mask spectral splatter associated with the decrement or increment performance tended to worsen with decreasing frequency for all decrement durations and for increment detection performance also worsened with decreasing level the results were analyzed using a model consisting of a compressive nonlinearity a sliding temporal integrator and a decision device the analysis indicated that the worsening in performance with decreasing frequency and decreasing level can be attributed partly to increases in the equivalent rectangular duration erd of the temporal integrator but mainly to changes in the efficiency of the detection process following the temporal integrator at lower frequencies and levels a larger change is required at the output of the integrator for threshold to be reached at each frequency the erd was relatively invariant with level for levels more than about 20 db above the absolute threshold

spectral and temporal integration of brief tones both spectral and temporal integration of tones have been explored in detail but integration of tones varying across both dimensions has received little attention this study explores temporal integration of tone pulses that vary over a range of frequencies baseline thresholds were obtained for both spectral and temporal integration with the same signals and compared with prior research the signals were then varied on both dimensions in several ways with equivalent spectral and temporal step sizes different spectral and temporal step sizes and a random pattern of frequency presentation the data were also analyzed by spectral step size temporal step size frequency range direction and slope of frequency change and predictability the spectral and temporal integration conditions showed that the current procedures and signals yielded the same improvement in detection thresholds as prior studies the spectrotemporal integration conditions showed the improvement for overall detection of the signals to be limited by spectral integration with improvement related primarily to the number of tones regardless of timing and frequency surprisingly trial by trial random presentation of signal frequencies did not negatively influence detection these results support the multiple looks hypothesis viemeister n f and wakefield g h 1991 temporal integration and multiple looks j acoust soc am 90 858 865 as applied to spectrotemporal integration

intensity discrimination and detection of amplitude modulation thresholds for detection of low rate sinusoidal amplitude modulation and for detection of intensity increments were measured over a wide range of levels in an examination of the relationship between these fundamental aspects of intensity processing as expected thresholds measured with a continuous 1 khz tone decrease with increasing carrier pedestal level for levels between 6 and 85 db spl the data are well described by 10 log i i 0 44 20 log m d fm where i i is the weber fraction for increment detection m is the modulation index at threshold and d fm depends on modulation rate fm the relationship between the psychometric functions for modulation and increment detection is also consistent with this equation the data indicate a clear relationship between modulation and increment detection and thus provide an important additional consideration for models of modulation processing no existing models provide an adequate account of this relationship

algebraic k theory of topological k theory we are interested in the arithmetic of ring spectra to make sense of this we must work with structured ring spectra such as s algebras ekmm symmetric ring spectra hss or rings ly we will refer to these as salgebras the commutative objects are then commutative s algebras the category of rings is embedded in the category of s algebras by the eilenberg maclane functor r hr we may therefore view an s algebra as a generalization of a ring in the algebraic sense the added flexibility of s algebras provides room for new examples and constructions which may eventually also shed light upon the category of rings itself in algebraic number theory the arithmetic of the ring of integers in a number field is largely captured by its picard group its unit group and its brauer group these are

a compressive gammachirp auditory filter for both physiological and psychophysical data a gammachirp auditory filter was developed by irino and patterson j acoust soc am 101 412 419 1997 to provide a level dependent version of the linear gammatone auditory filter with which to explain the level dependent changes in cochlear filtering observed in psychophysical masking experiments in this analytical gammachirp filter the chirp varied with level and there was no explicit representation of the change in filter gain or compression with level subsequently carney et al j acoust soc am 105 2384 2391 1999 reviewed carney and yin s j neurophysiol 60 1653 1677 1988 reverse correlation revcor data and showed that the frequency glide of the chirp does not vary with level in their data in this article the architecture of the analytical gammachirp is reviewed with respect to cochlear physiology and a new form of gammachirp filter is described in which the magnitude response the gain and the compression vary with level but the chirp does not this new compressive gammachirp filter is used to fit the level dependent revcor data reported by carney et al 1999 and the level dependent masking data reported by rosen and baker hear res 73 231 243 1994

an adaptive least squares fem for the stokes equations with optimal convergence rates this paper introduces the first adaptive least squares finite element method ls fem for the stokes equations with optimal convergence rates based on the newest vertex bisection with lowest order raviart thomas and conforming p 1 p1 discrete spaces for the divergence least squares formulation in 2d although the least squares functional is a reliable and efficient error estimator the novel refinement indicator stems from an alternative explicit residual based a posteriori error control with exact solve particular interest is on the treatment of the data approximation error which requires a separate marking strategy the paper proves linear convergence in terms of the levels and optimal convergence rates in terms of the number of unknowns relative to the notion of a non linear approximation class it extends and generalizes the approach of carstensen and park siam j numer anal 53 43 62 2015 from the poisson model problem to the stokes equations

detection of time and bandlimited increments and decrements in a random level noise the purpose of this study was to compare detection of increments and decrements occurring over limited regions of time and frequency within a 500 ms broadband 0 6000 hz noise three listeners tracked detection thresholds adaptively in a two interval two alternative forced choice task thresholds were measured for both increments and decrements in level l 10 log10 1 n0 n0 db where n0 is the spectral power density of the noise as a function of signal duration t 30 500 ms for a range of signal bandwidths w 62 6000 hz that were logarithmically centered around 2500 hz listeners were forced to rely on temporal and spectral profile cues for detection due to randomization of overall presentation level from interval to interval which rendered overall energy an inconsistent cue increments were detectable for all combinations of w and t whereas decrements were not consistently detectable for w 500 hz narrow band decrements were not detectable due to spread of excitation from the spectral edges of

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

cochlea scaled spectral entropy predicts rate invariant intelligibility of temporally distorted sentences some evidence mostly drawn from experiments using only a single moderate rate of speech suggests that low frequency amplitude modulations may be particularly important for intelligibility here two experiments investigated intelligibility of temporally distorted sentences across a wide range of simulated speaking rates and two metrics were used to predict results sentence intelligibility was assessed when successive segments of fixed duration were temporally reversed exp 1 and when sentences were processed through four third octave band filters the outputs of which were desynchronized exp 2 for both experiments intelligibility decreased with increasing distortion however in exp 2 intelligibility recovered modestly with longer desynchronization across conditions performances measured as a function of proportion of utterance distorted converged to a common function estimates of intelligibility derived from modulation transfer functions predict a substantial proportion of the variance in listeners responses in exp 1 but fail to predict performance in exp 2 by contrast a metric of potential information quantified as relative dissimilarity change between successive cochlear scaled spectra is introduced this metric reliably predicts listeners intelligibility across the full range of speaking rates in both experiments results support an information theoretic approach to speech perception and the significance of spectral change rather than physical units of time

on least squares finite element methods for the poisson equation and their connection to the dirichlet and kelvin principles least squares finite element methods for first order formulations of the poisson equation are not subject to the inf sup condition and lead to stable solutions even when all variables are approximated by equal order continuous finite element spaces for such elements one can also prove optimal convergence in the energy norm equivalent to a norm on h 1 omega times h omega mbox rm div for all variables and optimal l2 convergence for the scalar variable however showing optimal l2 convergence for the flux has proven to be impossible without adding the redundant curl equation to the first order system of partial differential equations in fact numerical evidence strongly suggests that nodal continuous flux approximations do not posses optimal l2 accuracy in this paper we show that optimal l2 error rates for the flux can be achieved without the curl constraint provided that one uses the div conforming family of brezzi douglas marini or brezzi douglas duran fortin elements then we proceed to reveal an interesting connection between a least squares finite element method involving h omega mbox rm div conforming flux approximations and mixed finite element methods based on the classical dirichlet and kelvin principles we show that such least squares finite element methods can be obtained by approximating through an l2 projection the hodge operator that connects the kelvin and dirichlet principles our principal conclusion is that when implemented in this way a least squares finite element method combines the best computational properties of finite element methods based on each of the classical principles

the effects of frequency region and bandwidth on the temporal modulation transfer function temporal modulation transfer functions tmtfs were measured using narrow band am and qfm noises with upper spectral edges from 0 6 to 4 8 khz and spectrum levels of 10 and 40 db spl the cutoff frequency of the tmtf increases as the upper spectral edge is increased up to 4 8 khz at low levels and is constant at higher levels sensitivity increases with bandwidth if frequency region is constant in a second experiment these results were compared to predictions of a model incorporating peripheral and central limitations to modulation detection to obtain an estimate of peripheral filtering frequency selectivity was measured using the notched noise method with probe frequencies and levels chosen to parallel those in the first experiment the tmtf data were then predicted using the model predicted cutoff frequencies as a function of the upper spectral edge of the test stimulus were lower than but parallel to those of the subjects at the lower stimulus level the model predicted only a slight increase in cutoff frequency with level and thus predicted an increase in cutoff frequency with frequency region at the higher level as well in contrast to the measured data these results suggest that there are peripheral and central limitations to temporal resolution but the psychoacoustically derived auditory filter may be only an indirect measure of peripheral filtering and or a more complex model may be needed

derived azumaya algebras and twisted k theory abstract we construct a relative version of topological k theory in the sense of 11 over an arbitrary quasi compact quasi separated c scheme x this has as input a perf x linear stable category and output a sheaf of spectra on x c the space of complex points of x we then characterize the values of this functor on inputs of the form perf x a for a a derived azumaya algebra over x in such cases we show that this coincides with the twisted topological k theory of x c for some appropriately defined twist of k theory we use this to provide a topological analogue of a classical result of quillen s on the algebraic k theory of severi brauer varieties

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