on some characterisations of totally unimodular matrices a characterisation of totally unimodular matrices is derived from a result of hoffman and kruskal it is similar in spirit to a result of baum and trotter its relation with some other known characterisations is discussed and in the particular case where the matrices have 0 1 entries we derive some properties of the associated unimodular hypergraphs similar results for balanced and perfect matrices are also reviewed

on the continuity of the multivariate pade operator abstract continuity of the univariate pade operator was proved in 5 6 we discuss the limitations of a multivariate generalization and prove a multivariate analogon of the continuity property

multivariate pade approximation abstract this paper is a survey on the multivariate pade approximation two types of approximants are considered those which can approximate general meromorphic functions f h g where both h and g are holomorphic and those which are specialized to the approximation of functions of the same form where g is a polynomial algorithms are described together with the different techniques used for proving convergence

a decomposition property of polyhedra a polyhedronp x n d ax e is said to have the real decomposition property rdp if for any positivet and any realx tp there are positive coefficients 1 r and integers1 sr p withx lsl rsr andt l r we give a constructive proof that this property holds for a polyhedronp iffp is integral this construction is used to show that some classes of polyhedra have an integral decomposition property furthemore the rdp provides a generalization of the theorem of birkhoff von neumann so rdp may be used in some scheduling problems on parallel processors with preemptions

the d 2 transformation for infinite double series and the d 2 transformation for infinite double integrals new transformations for accelerating the convergence of infinite double series and infinite double integrals are presented these transformations are generalizations of the univariate d and d transformations the d 2 transformation for infinite double integrals is efficient if the integrand satisfies a p d e of a certain type similarly the d 2 transformation for double series works well for series whose terms satisfy a difference equation of a certain type in both cases the application of the transformation does not require an explicit knowledge of the differential or the difference equation asymptotic expansions for the remainders in the infinite double integrals and series are derived and nonlinear transformations based upon these expansions are presented finally numerical examples which demonstrate the efficiency of these transformations are given

role of mask pattern in intelligibility of ideal binary masked noisy speech intelligibility of ideal binary masked noisy speech was measured on a group of normal hearing individuals across mixture signal to noise ratio snr levels masker types and local criteria for forming the binary mask the binary mask is computed from time frequency decompositions of target and masker signals using two different schemes an ideal binary mask computed by thresholding the local snr within time frequency units and a target binary mask computed by comparing the local target energy against the long term average speech spectrum by depicting intelligibility scores as a function of the difference between mixture snr and local snr threshold alignment of the performance curves is obtained for a large range of mixture snr levels large intelligibility benefits are obtained for both sparse and dense binary masks when an ideal mask is dense with many ones the effect of changing mixture snr level while fixing the mask is significant whereas for more sparse masks the effect is small or insignificant

explicit construction of multivariate pade approximants we explicitly construct both homogeneous and nonhomogeneous multivariate pade approximants to some functions which satisfy some simple functional equations by using the residue theorem and the functional equation method which has been used successfully by borwein 1988 to construct one variable pade approximants to the q elementary functions

a multivariate convergence theorem of the de montessus de ballore type abstract the univariate theorem of de montessus de ballore proves the convergence of column sequences of pade approximants for functions f z meromorphic in a disk in case the number of poles of f z and their multiplicity is known in advance we prove here a multivariate anologon for the case of simple poles and for the general order pade approximants as introduced by cuyt and verdonk 1984

a polynomial algorithm for multiprocessor scheduling with two job lengths the following multiprocessor scheduling problem was motivated by scheduling maintenance periods for aircraft each maintenance period is a job and the maintenance facilities are machines in this context there are very few different types of maintenances performed so it is natural to consider the problem with only a small fixed number c of different types of jobs each job type has a processing time and each machine is available for the same length of time a machine can handle at most one job at a time all jobs are released at time zero there are no due dates or precedence constraints and preemption is not allowed the question is whether it is possible to finish all jobs we call this problem the multiprocessor scheduling problem with c job lengths mspc scheduling problems such as mspc where we can partition the jobs into relatively few types such that all jobs of a certain type are identical are often called high multiplicity problems high multiplicity problems are interesting because their input is very compact the input to mspc consists of only 2c 2 numbers for the case c 2 we present a polynomial time algorithm we show that this algorithm guarantees a schedule that uses the minimum possible number of different one machine schedules namely three further we extend this algorithm to the case of machine dependent deadlines uniform parallel machines to a multi parametric case that contains the case of unrelated parallel machines and to some related covering problems finally we give some counterexamples showing why our results do not extend to the case c 2

excess masking among listeners with a sensorineural hearing loss three experiments were conducted to determine whether listeners with a sensorineural hearing loss exhibited greater than normal amounts of masking at frequencies above the frequency of the masker excess masking was defined as the difference in db between the masked thresholds actually obtained from a hearing impaired listener and the expected thresholds calculated for the same individual the expected thresholds were the power sum of the listener s thresholds in quiet and the average masked thresholds obtained from a group of normal hearing subjects at the test frequency hearing impaired listeners with thresholds in quiet ranging from approximately 35 70 db spl at test frequencies between 500 3000 hz displayed approximately 12 15 db of maximum excess masking the maximum amount of excess masking occurred in the region where the threshold in quiet of the hearing impaired listener and the average normal masked threshold were equal these findings indicate that listeners with a sensorineural hearing loss display one form of reduced frequency selectivity i e abnormal upward spread of masking even when their thresholds in quiet are taken into account

multivariate pade approximation multivariate pade approximation is considered in the framework of the theory developed by a cuyt 1982 the denominators are determined from determinants of matrices whose entries are homogeneous polynomials the main difference to the univariate case is a typical shift of the degree of the polynomials singularities atx 0 are analyzed since it is the rule rather than the exception that simultaneous zeros of the numerator and the denominator cannot be removed

applying lehman s theorems to packing problems a 0 1 matrixa isideal if the polyhedronq a conv x qv a x 1 x 0 v denotes the column index set ofa is integral similarly a matrix isperfect ifp a conv x qv a x 1 x 0 is integral little is known about the relationship between these two classes of matrices we consider a transformation between the two classes which enables us to apply lehman s modified theorem about deletion minimal nonideal matrices to obtain new results about packing polyhedra this results in a polyhedral description for the stable set polytopes ofnear bipartite graphs the deletion of any neighbourhood produces a bipartite graph note that this class includes the complements of line graphs to date this is the only natural class besides the perfect graphs for which such a description is known for the graphs and their complements some remarks are also made on possible approaches to describing the stable set polyhedra of quasi line graphs and more generally claw free graphs these results also yield a new class oft perfect graphs

predicting binaural speech intelligibility using the signal to noise ratio in the envelope power spectrum domain this study proposes a binaural extension to the multi resolution speech based envelope power spectrum model mr sepsm jorgensen ewert and dau 2013 j acoust soc am 134 436 446 it consists of a combination of better ear be and binaural unmasking processes implemented as two monaural realizations of the mr sepsm combined with a short term equalization cancellation process and uses the signal to noise ratio in the envelope domain snrenv as the decision metric the model requires only two parameters to be fitted per speech material and does not require an explicit frequency weighting the model was validated against three data sets from the literature which covered the following effects the number of maskers the masker types speech shaped noise ssn speech modulated ssn babble and reversed speech the masker s azimuths reverberation on the target and masker and the interaural time difference of the target and masker the pearson correlation coefficient between the simulated spee

the qd algorithm and multivariate pade approximants the quotient difference qd algorithm can be used to construct univariate pade approximants 1 in this paper we see that it can also be used to construct the multivariate pade approximants introduced in 3 just by reformulating the quotient difference algorithm as in sect 1 the multivariate pade approximants and the multivariate qd scheme are treated in the sects 2 and 3 respectively thus for this type of multivariate pade approximants a link with the theory of multivariate continued fractions is established

cross correlation procedures for measuring noise and distortion in agc hearing aids the magnitude squared coherence function msc has been used to measure noise and distortion in linear and compression hearing aids however the msc will overestimate the distortion in a linear time varying system such as a compression amplifier the reduction in coherence caused by varying the gain in an otherwise linear system can be substantial and can lead to large errors in estimating the distortion present in a compression hearing aid the effects of gain changes in a linear system can be reduced by measuring the normalized system input output cross correlation which emphasizes the variance in the system phase response and deemphasizes the system gain fluctuations estimates of the total noise and distortion produced using the msc phase variance and notched noise measurement techniques are compared for additive noise clipping distortion and compression amplification the msc is found to give the most accurate results for estimating the noise and distortion in a linear time invariant system and the notched noise measurements are the most accurate for a compression system the phase variance is found to give reasonable measurements for a time varying gain as long as the system variations are slow relative to the length of the analysis data segments

integer round up property for the chromatic number of some h perfect graphs we prove that every h perfect line graph and every t perfect claw free graph g has the integer round up property for the chromatic number for every non negative integral weight function c on the vertices of g the weighted chromatic number of g c can be obtained by rounding up its fractional relaxation as a corollary we obtain that the weighted chromatic number can be computed in polynomial time for these graphs finally we show a new example of a graph operation which preserves the integer round up property for the chromatic number and use it to provide a first example of a t perfect 3 colorable graph which does not have this property

identification of discontinuous nonlinear systems via a multivariate pade approach we present a nonlinear system identification technique based on multi dimensional rational polynomials a multi dimensional pade legendre approximation is developed to circumvent challenges in dealing with sharp shocks the purpose of this paper is to investigate the accuracy of such approximations for identification of various nonlinear systems particularly systems with a non smooth response surface this identification approach utilizes the generalized form of a pade legendre approximation for studying multivariable functions in the studied problems the nonlinearity is a function of state variables displacement and velocity which requires multi dimensional formulation furthermore a spatial filter is applied to minimize the effects of the singular points in the applicable rational function of the response surface this study presents different types of nonlinearities including smooth irregular and hysteretic functions in order to demonstrate the performance of the approach under different conditions in order to study the robustness of the method in comparison to other identification techniques based on plain polynomial representation a nonlinear system with a sharp discontinuous restoring force surface is considered the performance of both approaches is investigated for different degrees of sharpness in addition the accuracy of the identified models to represent the nonlinear system is verified by comparing the output of the system computed on the basis of the identified model from data sets corresponding to different excitations than those used for identification purposes it is shown that the proposed approach provides a robust identification technique for a broad class of highly nonlinear systems and it is particularly advantageous to use when dealing with systems incorporating discontinuous properties

a polynomial algorithm for integer programming covering problems satisfying the integer round up property leta be a non negative matrix with integer entries and no zero column the integer round up property holds fora if for every integral vectorw the optimum objective value of the generalized covering problem min 1y ya w y 0 integer is obtained by rounding up to the nearest integer the optimum objective value of the corresponding linear program a polynomial time algorithm is presented that does the following given any generalized covering problem with constraint matrixa and right hand side vectorw the algorithm either finds an optimum solution vector for the covering problem or else it reveals that matrixa does not have the integer round up property

general order multivariate pade approximants for pseudo multivariate functions ii although general order multivariate pade approximants were introduced some decades ago very few explicit formulas for special functions have been given we explicitly construct some general order multivariate pade approximants to the class of so called pseudo multivariate functions using the pade approximants to their univariate versions we also prove that the constructed approximants inherit the normality and consistency properties of their univariate relatives which do not hold in general for multivariate pade approximants examples include the multivariate forms of the exponential and the q exponential functions

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

on a diagonal pade approximation in two complex variables a special type diagonal pade approximation for a class of hermitian power series in two variables is related to a canonical strong operator topology finite rank approximation of cyclic operators the expected convergence of the process uniform or in measure is derived from operator theory facts

suggested formulae for calculating auditory filter bandwidths and excitation patterns recent estimates of auditory filter shape are used to derive a simple formula relating the equivalent rectangular bandwidth erb of the auditory filter to center frequency the value of the auditory filter bandwidth continues to decrease as center frequency decreases below 500 hz a formula is also given relating erb rate to frequency finally a method is described for calculating excitation patterns from filter shapes

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

general order newton pade approximants for multivariate functions pade approximants are a frequently used tool for the solution of mathematical problems one of the main drawbacks of their use for multivariate functions is the calculation of the derivatives off x 1 x p therefore multivariate newton pade approximants are introduced their computation will only use the value off at some points in sect 1 we shall repeat the univariate newton pade approximation problem which is a rational hermite interpolation problem in sect 2 we sketch some problems that can arise when dealing with multivariate interpolation in sect 3 we define multivariate divided differences and prove some lemmas that will be useful tools for the introduction of multivariate newton pade approximants in sect 4 a numerical example is given in sect 5 together with the proof that forp 1 the classical newton pade approximants for a univariate function are obtained

predicting speech intelligibility based on a correlation metric in the envelope power spectrum domain a speech intelligibility prediction model is proposed that combines the auditory processing front end of the multi resolution speech based envelope power spectrum model mr sepsm jorgensen ewert and dau 2013 j acoust soc am 134 1 436 446 with a correlation back end inspired by the short time objective intelligibility measure stoi taal hendriks heusdens and jensen 2011 ieee trans audio speech lang process 19 7 2125 2136 this hybrid model named sepsmcorr is shown to account for the effects of stationary and fluctuating additive interferers as well as for the effects of non linear distortions such as spectral subtraction phase jitter and ideal time frequency segregation itfs the model shows a broader predictive range than both the original mr sepsm which fails in the phase jitter and itfs conditions and stoi which fails to predict the influence of fluctuating interferers albeit with lower accuracy than the source models in some individual conditions similar to othe

a polynomial time opt 1 algorithm for the cutting stock problem with a constant number of object lengths in the cutting stock problem we are given a set of objects of different types and the goal is to pack them all in the minimum possible number of identical bins all objects have integer lengths and objects of different types have different sizes the total length of the objects packed in a bin cannot exceed the capacity of the bin in this paper we consider the version of the problem in which the number of different object types is constant and we present a polynomial time algorithm that computes a solution using at most one more bin than an optimum solution

mathematica evidence that ramanujan kills baker gammel wills a 1961 conjecture of baker gammel and wills asserts that if a function f is meromorphic in the unit ball and analytic at zero then a subsequence of its diagonal pade approximants converges uniformly in compact subsets omitting poles inasmuch as the denominators of the pade approximants are complex orthogonal polynomials and the convergence of sequences of pade approximants is determined largely by the behaviour of their poles the conjecture deals with distribution of zeros of complex orthogonal polynomials in this paper we present numerical evidence derived using the mathematica package that ramanujan s continued fractionh q z 1 qz 1 q 2z 1 q 3z 1 provides a counterexample provided q is appropriately chosen on the unit circle

on the continuity of the pade operator let r m n be the pade approximant of order m n for a given power series f let t m n be the operator that maps f on r m n it is known that t m n satisfies a local lipschitz condition in case r m n is normal but may fail to be continuous if r m n is not normal in this paper we prove that t m n is continuous if and only if r m n has defect zero we also prove a convergence result for pade approximants in the case where the defect is positive

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

on claw free t perfect graphs a graph is called t perfect if its stable set polytope is defined by non negativity edge and odd cycle inequalities we characterise the class of all claw free t perfect graphs by forbidden t minors and show that they are 3 colourable moreover we determine the chromatic number of claw free h perfect graphs and give a polynomial time algorithm to compute an optimal colouring

convergence of diagonal pade approximants for functions analytic near 0 for functions analytic in a neighbourhood of 0 we show that at least for a subsequence of the diagonal pade approximants the point 0 attracts a zero proportion of the poles the same is true for every sufficiently dense diagonal subsequence consequently these subsequences have a convergence in capacity type property which is possibly the correct analogue of the nuttallpommerenke theorem in this setting

the convergence of diagonal pade approximants and the pade conjecture questions related to the convergence problem of diagonal pade approximants are discussed a central place is taken by the pade conjecture also known as the baker gammel wills conjecture partial results concerning this conjecture are reviewed and weaker and more special versions of the conjecture are formulated and their plausibility is investigated great emphasis is given to the role of spurious poles of the approximants a conjecture by nuttall 1970 about the number and distribution of such poles is stated and its importance for the pade conjecture is analyzed

near perfect matrices a 0 1 matrixa isnear perfect if the integer hull of the polyhedron x 0 ax bar 1 can be obtained by adding one extra rank constraint we show that in general such matrices arise as the cliquenode incidence matrices of graphs we give a colouring like characterization of the corresponding class of near perfect graphs which shows that one need only check integrality of a certain linear program for each 0 1 2 valued objective function this in contrast with perfect matrices where it is sufficient to check 0 1 valued objective functions we also make the following conjecture a graph is near perfect if and only if sequentially lifting any rank inequality associated with a minimally imperfect graph results in the rank inequality for the whole graph we show that the conjecture is implied by the strong perfect graph conjecture it is also shown to hold for graphs with no stable set of size eleven our results are used to strengthen and give a new proof of a theorem of padberg this results in a new characterization of minimally imperfect graphs a graph is minimally imperfect if and only if both the graph and its complement are near perfect

the effects of hearing loss on the contribution of high and low frequency speech information to speech understanding the speech understanding of persons with flat hearing loss hi was compared to a normal hearing nh control group to examine how hearing loss affects the contribution of speech information in various frequency regions speech understanding in noise was assessed at multiple low and high pass filter cutoff frequencies noise levels were chosen to ensure that the noise rather than quiet thresholds determined audibility the performance of hi subjects was compared to a nh group listening at the same signal to noise ratio and a comparable presentation level although absolute speech scores for the hi group were reduced performance improvements as the speech and noise bandwidth increased were comparable between groups these data suggest that the presence of hearing loss results in a uniform rather than frequency specific deficit in the contribution of speech information measures of auditory thresholds in noise and speech intelligibility index sii calculations were also performed these data suggest that differences in performance between the hi and nh groups are due primarily to audibility differences between groups measures of auditory thresholds in noise showed the effective masking spectrum of the noise was greater for the hi than the nh subjects

characterizing and bounding the imperfection ratio for some classes of graphs perfect graphs constitute a well studied graph class with a rich structure which is reflected by many characterizations with respect to different concepts perfect graphs are for instance precisely those graphs g where the stable set polytope stab g equals the fractional stable set polytope qstab g the dilation ratio rm min t rm qstab g subseteq t rm stab g of the two polytopes yields the imperfection ratio of g it is np hard to compute and for most graph classes it is even unknown whether it is bounded for graphs g such that all facets of stab g are rank constraints associated with antiwebs we characterize the imperfection ratio and bound it by 3 2 outgoing from this result we characterize and bound the imperfection ratio for several graph classes including near bipartite graphs and their complements namely quasi line graphs by means of induced antiwebs and webs respectively

a direct approach to convergence of multivariate nonhomogeneous pade approximants we present a direct approach for proving convergence in measure product capacity of multivariate nonhomogeneous pade approximants previous approaches have involved projection onto pade type approximation in one variable and only yielded convergence in lebesgue measure

packing and covering with linear programming a survey this paper considers the polyhedral results and the min max results on packing and covering problems of the decade since the strong perfect graph theorem published in 2006 the main such results are available for the packing problem however there are still important polyhedral questions that remain open for the covering problem the main questions are still open although there has been important progress we survey some of the main results with emphasis on those where linear programming and graph theory come together they mainly concern the covering of cycles or dicycles in graphs or signed graphs either with vertices or edges this includes the multicut and integral multiflow problems

extension of a multivariate convergence theorem of the de montessus de ballore type to multipoles abstract in this journal 1990 we proved a multivariate version of the de montessus de ballore theorem stating the convergence of general order pade approximants for multivariate meromorphic functions with so called simple poles that result is extended here to the case of multipoles

integral decomposition in polyhedra we say that a polyhedronp satisfies weak integral decomposition if whenever an integral vector is the sum ofk vectors inp it is also the sum ofk integral vectors inp this property is related to rounding results for packing and covering problems we study the property and two related properties and give results concerning integral polymatroids totally unimodular matrices and network flows pairs of strongly base orderable matroids and branchings in directed graphs

nuttall pommerenke theorems for homogeneous pade approximants we investigate nuttall pommerenke theorems for several variable homogeneous pade approximants using ideas of goncar karlsson and wallin

how well can the concept of pade approximant be generalized to the multivariate case what we know about multivariate pade approximation has been developed in the last 25 years in the next sections we compare and discuss many of these results it will become clear that simple properties or requirements such as the uniqueness of the pade approximant and consequently its consistency property can play a crucial role in the development of the multivariate theory a separate section is devoted to a discussion of the convergence properties at the end we include an extensive reference list on the topic

loudness growth in 1 2 octave bands lgob a procedure for the assessment of loudness in this paper a method that has been developed for the assessment and quantification of loudness perception in normal hearing and hearing impaired persons is described the method has been named lgob which stands for loudness growth in 1 2 octave bands the method uses 1 2 octave bands of noise centered at 0 25 0 5 1 0 2 0 and 4 0 khz with subjective levels between a subject s threshold of hearing and the too loud level the noise bands are presented to the subject randomized over frequency and level and the subject is asked to respond with a loudness rating one of very soft soft ok loud very loud too loud subject responses normal and hearing impaired are then compared to the average responses of a group of normal hearing subjects this procedure allows one to estimate the subject s loudness growth relative to normals as a function of frequency and level the results may be displayed either as isoloudness contours or as recruitment curves in its present form the measurements take less than 30 min the signal presentation and analysis is done using a pc and a pc plug in board having a digital to analog converter

quality ratings for frequency shaped peak clipped speech peak clipping is a common form of distortion in hearing aids and can reduce the subjective quality of the amplified speech in a typical hearing aid frequency response shaping precedes symmetric peak clipping the effect of peak clipping on speech quality ratings was therefore studied using sentence test materials that were processed using different frequency response contours and then clipped at different clipping thresholds the quality of each processed sentence was rated on a ten point scale by normal hearing subjects the experimental results indicate that the clipping threshold and the interaction of the frequency response shaping with the clipping threshold significantly affect speech quality it is also shown that the distortion effects on speech quality can be modeled by a distortion index computed from the magnitude squared coherence of the speech processing system in response to a shaped noise input signal

uncomodulated glimpsing in checkerboard noise the ability of listeners to glimpse acoustic cues during the quieter sections of an interrupted noise has primarily been studied using maskers with interruptions occurring simultaneously across the entire frequency range of the masker broadband comodulated interruptions here the possibility of uncomodulated glimpsing the glimpsing of acoustic cues separated both in time and frequency was investigated to achieve this speech reception thresholds for a set of intervocalic consonants were adaptively measured in 100 hz to 10 khz pink noise divided into a varying number of frequency bands of equal energy in uncomodulated noise conditions the odd and even numbered bands were switched on and off alternately at a rate of 10 hz the spectrograms of such noises on log frequency scales resemble portions of a checkerboard glimpsing in checkerboard noise was found with maskers divided into two and four bands but not into eight bands or more further investigations showed that in the two band case this release from masking was indeed due to uncomodulated glimpsing and not simply attributable to glimpsing in one of the modulated bands in the four band case the release from masking in checkerboard noise can be accounted for without recourse to uncomodulated glimpsing interestingly conditions which allowed glimpsing resulted in greater intersubject variability the implications of these results for quantitative analyses of masker fluctuations are discussed

computation and efficiency of potential function minimizers of combinatorial congestion games we study the computation and efficiency of pure nash equilibria in combinatorial congestion games where the strategies of each player i are given by the binary vectors of a polytope p i our main goal is to understand which structural properties of such polytopal congestion games enable us to derive an efficient equilibrium selection procedure to compute pure nash equilibria with attractive social cost approximation guarantees to this aim we identify two general properties of the underlying aggregation polytope p n sum i p i which are sufficient for our results to go through namely the integer decomposition property idp and the box totally dual integrality property box tdi our main results for polytopal congestion games satisfying idp and box tdi are as follows i we show that pure nash equilibria can be computed in polynomial time in fact we obtain this result through a general framework for separable convex function minimization which might be of independent interest ii we bound the inefficiency of these equilibria and show that this provides a tight bound on the price of stability iii we also prove that these results extend to strong equilibria for the bottleneck variant of polytopal congestion games examples of polytopal congestion games satisfying idp and box tdi include common source network congestion games symmetric totally unimodular congestion games non symmetric matroid congestion games and symmetric matroid intersection congestion games in particular r arborescences and strongly base orderable matroids

pade legendre approximants for uncertainty analysis with discontinuous response surfaces a novel uncertainty propagation method for problems characterized by highly non linear or discontinuous system responses is presented the approach is based on a pade legendre pl formalism which does not require modifications to existing computational tools non intrusive approach and it is a global method the paper presents a novel pl method for problems in multiple dimensions which is non trivial in the pade literature in addition a filtering procedure is developed in order to minimize the errors introduced in the approximation close to the discontinuities the numerical examples include fluid dynamic problems characterized by shock waves a simple dual throat nozzle problem with uncertain initial state and the turbulent transonic flow over a transonic airfoil where the flight conditions are assumed to be uncertain results are presented in terms of statistics of both shock position and strength and are compared to monte carlo simulations

finite checkability for integer rounding properties in combinatorial programming problems leta be a nonnegative integral matrix with no zero columns theinteger round up property holds fora if for each nonnegative integral vectorw the solution value to the integer programming problem min 1 y ya w y 0 y integer is obtained by rounding up to the nearest integer the solution value to the corresponding linear programming problem min 1 y ya w y 0 theinteger round down property is similarly defined for a nonnegative integral matrixb with no zero rows by considering max 1 y yb w y 0 y integer and its linear programming correspondent it is shown that the integer round up and round down properties can be checked through a finite process the method of proof motivates a new and elementary proof of fulkerson s pluperfect graph theorem

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

channel selection in the modulation domain for improved speech intelligibility in noise background noise reduces the depth of the low frequency envelope modulations known to be important for speech intelligibility the relative strength of the target and masker envelope modulations can be quantified using a modulation signal to noise ratio s n mod measure such a measure can be used in noise suppression algorithms to extract target relevant modulations from the corrupted target masker envelopes for potential improvement in speech intelligibility in the present study envelopes are decomposed in the modulation spectral domain into a number of channels spanning the range of 0 30 hz target dominant modulations are identified and retained in each channel based on the s n mod selection criterion while modulations which potentially interfere with perception of the target i e those dominated by the masker are discarded the impact of modulation selective processing on the speech reception threshold for sentences in noise is assessed with normal hearing listeners results indicate that the intelligibility of noise masked speech can be improved by as much as 13 db when preserving target dominant modulations present up to a modulation frequency of 18 hz while discarding masker dominant modulations from the mixture envelopes

coherence and the speech intelligibility index the speech intelligibility index sii ansi s3 5 1997 provides a means for estimating speech intelligibility under conditions of additive stationary noise or bandwidth reduction the sii concept for estimating intelligibility is extended in this paper to include broadband peak clipping and center clipping distortion with the coherence between the input and output signals used to estimate the noise and distortion effects the speech intelligibility predictions using the new procedure are compared with intelligibility scores obtained from normal hearing and hearing impaired subjects for conditions of additive noise and peak clipping and center clipping distortion the most effective procedure divides the speech signal into low mid and high level regions computes the coherence sii separately for the signal segments in each region and then estimates intelligibility from a weighted combination of the three coherence sii values

efficient adaptive procedures for threshold and concurrent slope estimates for psychophysics and speech intelligibility tests the minimum standard deviations achievable for concurrent estimates of thresholds and psychometric function slopes as well as the optimal target values for adaptive procedures are calculated as functions of stimulus level and track length on the basis of the binomial theory the optimum pair of targets for a concurrent estimate is found at the correct response probabilities p1 0 19 and p2 0 81 for the logistic psychometric function an adaptive procedure that converges at these optimal targets is introduced and tested with monte carlo simulations the efficiency increases rapidly when each subject s response consists of more than one statistically independent bernoulli trial sentence intelligibility tests provide more than one bernoulli trial per sentence when each word is scored separately the number of within sentence trials can be quantified by the j factor boothroyd and nittrouer j acoust soc am 84 101 114 1988 the adaptive procedure was evaluated with 10 normal hearing and 11 hearing imp

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

some generalized max flow min cut problems in the plane we consider variants of the max flow min cut problems in the plane motivated by models of damage where certain sets of edges of the graph can be simultaneously removed rather than one edge at a time as in the standard min cut problem our main contributions are a polynomial algorithm for the generalized min cut problem an np completeness result for the generalized max flow problem and an approximate generalized max flow min cut theorem

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

birkhoff gustavson normal form and the semiclassical energies of a double pendulum although the double pendulum is a well known system the literature is sparse with regard to the study of its full dynamical behaviour as a simple conservative system that displays both ordered and chaotic motion it poses an interesting problem for the study of classical quantum correspondence in this work we present its birkhoff gustavson normal form and semiclassical energies obtained from the normal form the pade summation technique is used to re sum the normal form series

factors influencing intelligibility of ideal binary masked speech implications for noise reduction the application of the ideal binary mask to an auditory mixture has been shown to yield substantial improvements in intelligibility this mask is commonly applied to the time frequency t f representation of a mixture signal and eliminates portions of a signal below a signal to noise ratio snr threshold while allowing others to pass through intact the factors influencing intelligibility of ideal binary masked speech are not well understood and are examined in the present study specifically the effects of the local snr threshold input snr level masker type and errors introduced in estimating the ideal mask are examined consistent with previous studies intelligibility of binary masked stimuli is quite high even at 10db snr for all maskers tested performance was affected the most when the masker dominated t f units were wrongly labeled as target dominated t f units performance plateaued near 100 correct for snr thresholds ranging from 20to5db the existence of the plateau region suggests that

an approximation scheme for finding steiner trees with obstacles we consider the problem of constructing a steiner minimal tree connecting a given set k of points and lying inside a polygonally bounded not necessarily simply connected region r in the plane we first define the path convex hull of k in r which is a sufficiently small subregion of r guaranteed to contain the steiner minimal tree we then give an varepsilon approximation scheme to find the steiner minimal tree in r by reducing it to a steiner tree problem on a visibility graph associated with k and the path convex hull of r this will be a fully polynomial approximation scheme when k is restricted to lie on a small number of interior points and boundary polygons of r several techniques are given which further reduce the region in which the steiner minimal tree is known to lie and which extend known results for the steiner minimal tree problem without obstacles

effects of manipulating the signal to noise envelope power ratio on speech intelligibility jorgensen and dau 2011 j acoust soc am 130 1475 1487 suggested a metric for speech intelligibility prediction based on the signal to noise envelope power ratio snrenv calculated at the output of a modulation frequency selective process in the framework of the speech based envelope power spectrum model sepsm the snrenv was demonstrated to account for speech intelligibility data in various conditions with linearly and nonlinearly processed noisy speech as well as for conditions with stationary and fluctuating interferers here the relation between the snrenv and speech intelligibility was investigated further by systematically varying the modulation power of either the speech or the noise before mixing the two components while keeping the overall power ratio of the two components constant a good correspondence between the data and the corresponding sepsm predictions was obtained when the noise was manipulated and mixed with the unprocessed speech consistent with the hypothesis that sn

multivariate pade approximants to a meromorphic function abstract we explicitly construct the general multivariate pade approximants to the function g q x y j 1 1 xy q j x q 2j q 1 q c by using the residue theorem and the functional equation method then we prove some convergence properties of the approximants

multivariate pade approximants abstract for an operator f r n r analytic in the origin the notion of abstract multivariate pade approximant apa is introduced by making use of abstract polynomials the classical pade approximant n 1 is a special case of the multivariate theory and many interesting properties of classical pade approximants remain valid such as covariance properties and the block structure annie a m cuyt j oper theory6 2 1981 207 209 of the pade table also a projection property for multivariate pade approximants is proved

lovasz schrijver sdp operator near perfect graphs and near bipartite graphs we study the lovasz schrijver lift and project operator mathrm text ls ls based on the cone of symmetric positive semidefinite matrices applied to the fractional stable set polytope of graphs the problem of obtaining a combinatorial characterization of graphs for which the mathrm text ls ls operator generates the stable set polytope in one step has been open since 1990 we call these graphs mathrm text ls ls perfect in the current contribution we pursue a full combinatorial characterization of mathrm text ls ls perfect graphs and make progress towards such a characterization by establishing a new close relationship among mathrm text ls ls perfect graphs near bipartite graphs and a newly introduced concept of full support perfect graphs

the concept of signal to noise ratio in the modulation domain and speech intelligibility a new concept is proposed that relates to intelligibility of speech in noise the concept combines traditional estimations of signal to noise ratios s n with elements from the modulation transfer function model which results in the definition of the signal to noise ratio in the modulation domain the sn mod it is argued that this sn mod quantifying the strength of speech modulations relative to a floor of spurious modulations arising from the speech noise interaction is the key factor in relation to speech intelligibility it is shown that by using a specific test signal the strength of these spurious modulations can be measured allowing an estimation of the sn mod for various conditions of additive noise noise suppression and amplitude compression by relating these results to intelligibility data for these same conditions the relevance of the sn mod as the key factor underlying speech intelligibility is clearly illustrated for instance it is shown that the commonly observed limited effect of noise suppression on speech intelligibility is correctly predicted by the sn mod whereas traditional measures such as the speech transmission index considering only the changes in the speech modulations fall short in this respect it is argued that sn mod may provide a relevant tool in the design of successful noise suppression systems

relations among some psychoacoustic parameters in normal and cochlearly impaired listeners frequency resolution viz masking by low pass filtered noise and broadband noise and temporal resolution viz masking by interrupted noise were compared with hearing thresholds and acoustic reflex thresholds for four normally hearing and 13 cochlearly impaired subjects two models one for frequency resolution model i and one for temporal resolution model ii were introduced and these provided a means of predicting individual frequency and temporal resolution from hearing thresholds for both normal hearing and hearing impaired listeners model i is based on the assumption that the upward spread of masking increases in cochlearly impaired hearing with an amount proportional to the hearing threshold in db hl model ii is based on the assumption that the poststimulatory masked thresholds return to the level of the hearing threshold within a duration of 200 ms independent of the level of the masker and the amount of cochlear hearing loss model parameters were determined from results from other st

masking release for hearing impaired listeners the effect of increased audibility through reduction of amplitude variability the masking release i e better speech recognition in fluctuating compared to continuous noise backgrounds observed for normal hearing nh listeners is generally reduced or absent in hearing impaired hi listeners one explanation for this lies in the effects of reduced audibility elevated thresholds may prevent hi listeners from taking advantage of signals available to nh listeners during the dips of temporally fluctuating noise where the interference is relatively weak this hypothesis was addressed through the development of a signal processing technique designed to increase the audibility of speech during dips in interrupted noise this technique acts to i compare short term and long term estimates of energy ii increase the level of short term segments whose energy is below the average energy and iii normalize the overall energy of the processed signal to be equivalent to that of the original long term estimate evaluations of this energy equalizing eeq technique included consonant identification and sentence reception in backgrounds of continuous and regularly interrupted noise for hi listeners performance was generally similar for processed and unprocessed signals in continuous noise however superior performance for eeq processing was observed in certain regularly interrupted noise backgrounds

a generalization of antiwebs to independence systems and their canonical facets we consider independence system polytopes i e polytopes whose extreme points are the incidence vectors of the sets of an independence system we first give a sufficient condition for recognizing boolean facets then the notion of antiweb introduced by trotter for graphs is generalized to independence systems and used for obtaining canonical facets of the associated polytopes we also point out how our results relate with known ones for knapsack set covering and matroid polytopes

the role of short time intensity and envelope power for speech intelligibility and psychoacoustic masking the generalized power spectrum model gpsm biberger and ewert 2016 j acoust soc am 140 1023 1038 combining the classical concept of the power spectrum model psm and the envelope power spectrum model epsm was demonstrated to account for several psychoacoustic and speech intelligibility si experiments the psm path of the model uses long time power signal to noise ratios snrs while the epsm path uses short time envelope power snrs a systematic comparison of existing si models for several spectro temporal manipulations of speech maskers and gender combinations of target and masker speakers schubotz et al 2016 j acoust soc am 140 524 540 showed the importance of short time power features conversely jorgensen et al 2013 j acoust soc am 134 436 446 demonstrated a higher predictive power of short time envelope power snrs than power snrs using reverberation and spectral subtraction here the gpsm was extended to utilize short time power snrs and was shown to account

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

revision extension and evaluation of a binaural speech intelligibility model this study presents revision extension and evaluation of a binaural speech intelligibility model beutelmann r and brand t 2006 j acoust soc am 120 331 342 that yields accurate predictions of speech reception thresholds srts in the presence of a stationary noise source at arbitrary azimuths and in different rooms the modified model is based on an analytical expression of binaural unmasking for arbitrary input signals and is computationally more efficient while maintaining the prediction quality of the original model an extension for nonstationary interferers was realized by applying the model to short time frames of the input signals and averaging over the predicted srt results binaural srts from 8 normal hearing and 12 hearing impaired subjects incorporating all combinations of four rooms three source setups and three noise types were measured and compared to the model s predictions depending on the noise type the parametric correlation coefficients between observed and predicted srts were 0 80 0 93 for normal hearing subjects and 0 59 0 80 for hearing impaired subjects the mean absolute prediction error was 3 db for the mean normal hearing data and 4 db for the individual hearing impaired data 70 of the variance of the srts of hearing impaired subjects could be explained by the model which is based only on the audiogram

clique circulants and the stable set polytope of fuzzy circular interval graphs in this paper we give a complete and explicit description of the rank facets of the stable set polytope for a class of claw free graphs recently introduced by chudnovsky and seymour proceedings of the bristish combinatorial conference 2005 called fuzzy circular interval graphs the result builds upon the characterization of minimal rank facets for claw free graphs by galluccio and sassano j combinatorial theory 69 1 38 2005 and upon the introduction of a superclass of circulant graphs that are called clique circulants the new class of graphs is invetigated in depth we characterize which clique circulants c are facet producing i e are such that sum v in v c x v leq alpha c is a facet of stab c thus extending a result of trotter discrete math 12 373 388 1975 for circulants we show that a simple clique family inequality oriolo discrete appl math 132 2 185 201 2004 may be associated with each clique circulant c subseteq g when g is fuzzy circular interval we show that these inequalities provide all the rank facets of stab g if g is a fuzzy circular interval graph moreover we conjecture that in this case they also provide all the non rank facets of stab g and offer evidences for this conjecture