sources of variation in profile analysis ii component spacing dynamic changes and roving level profile analysis experiments have typically employed static profiles with constant frequency components spaced at equal intervals along a logarithmic frequency axis most periodic naturally occurring stimuli however have components that are harmonically related and vary dynamically in time one goal of these studies was to determine whether amplitude increment detection thresholds are different in dynamic harmonically spaced profiles compared to those for static log profiles and why such differences might exist a second goal was to determine the impact of roving levels within trial variation of level thresholds for static log profiles were on average 8 7 db lower than for static harmonic profiles a traditional filter bank model could not account for this result no consistent effect of dynamic contour an exponential rising frequency glide was observed thresholds were consistently poorer by 4 to 7 db when the level was roved but the differences in thresholds among the different profiles varied little it is proposed that the higher thresholds observed in static harmonic profiles may be accounted for by the more intense pitch strength associated with the harmonic profiles

relative contributions of specific frequency bands to the loudness of broadband sounds listeners with normal hearing nh and sensorineural hearing loss snhl were asked to compare pairs of noise stimuli and choose the louder noise in each pair each noise was made up of 15 two erbn equivalent rectangular bandwidth wide frequency bands that varied independently over a 12 db range from one presentation to the next mean levels of the bands followed the long term average speech spectrum ltass or were set to 43 51 or 59 db sound pressure level spl the relative contribution of each band to the total loudness of the noise was determined by computing the correlation between the difference in levels for a given band on every trial and the listener s decision on that trial weights for snhl listeners were governed by audibility and the spectrum of the noise stimuli with bands near the spectral peak of the ltass noise receiving greatest weight nh listeners assigned greater weight to the lowest and highest bands an effect that increased with overall level but did not assign greater weight to bands near the ltass peak additional loudness matching and paired comparison studies using stimuli missing one of the 15 bands showed a significant contribution by the highest band but properties other than loudness may have contributed to the decisions

spectral shape discrimination by hearing impaired and normal hearing listeners the ability to discriminate between sounds with different spectral shapes was evaluated for normal hearing and hearing impaired listeners listeners discriminated between a standard stimulus and a signal stimulus in which half of the standard components were decreased in level and half were increased in level in one condition the standard stimulus was the sum of six equal amplitude tones equal spl and in another the standard stimulus was the sum of six tones at equal sensation levels re audiometric thresholds for individual subjects equal sl spectral weights were estimated in conditions where the amplitudes of the individual tones were perturbed slightly on every presentation sensitivity was similar in all conditions for normal hearing and hearing impaired listeners the presence of perturbation and equal sl components increased thresholds for both groups but only small differences in weighting strategy were measured between the groups depending on whether the equal spl or equal sl condition was tested the average data suggest that normal hearing listeners may rely more on the central components of the spectrum whereas hearing impaired listeners may have been more likely to use the edges however individual weighting functions were quite variable especially for the hi listeners perhaps reflecting difficulty in processing changes in spectral shape due to hearing loss differences in weighting strategy without changes in sensitivity suggest that factors other than spectral weights such as internal noise or difficulty encoding a reference stimulus also may dominate performance

on global attractors of the 3d navier stokes equations abstract in view of the possibility that the 3d navier stokes equations nse might not always have regular solutions we introduce an abstract framework for studying the asymptotic behavior of multi valued dissipative evolutionary systems with respect to two topologies weak and strong each such system possesses a global attractor in the weak topology but not necessarily in the strong in case the latter exists and is weakly closed it coincides with the weak global attractor we give a sufficient condition for the existence of the strong global attractor which is verified for the 3d nse when all solutions on the weak global attractor are strongly continuous we also introduce and study a two parameter family of models for the navier stokes equations with similar properties and open problems these models always possess weak global attractors but on some of them every solution blows up in a norm stronger than the standard energy one in finite time

coordination capacity we develop elements of a theory of cooperation and coordination in networks rather than considering a communication network as a means of distributing information or of reconstructing random processes at remote nodes we ask what dependence can be established among the nodes given the communication constraints specifically in a network with communication rates ri j between the nodes we ask what is the set of all achievable joint distributions p x1 xm of actions at the nodes of the network several networks are solved including arbitrarily large cascade networks distributed cooperation can be the solution to many problems such as distributed games distributed control and establishing mutual information bounds on the influence of one part of a physical system on another

a lagrange dual lower bound to the error exponent of the typical random code a lagrange dual gallager style lower bound is derived for the error exponent function of the typical random code trc pertaining to the i i d random coding ensemble and mismatched stochastic likelihood decoding while the original expression derived from the method of types the csiszar style expression involves minimization over probability distributions defined on the channel input output alphabets the new lagrange dual formula involves optimization of five parameters independently of the alphabet sizes for both stochastic and deterministic mismatched decoding including maximum likelihood decoding as a special case we provide a rather comprehensive discussion on the insight behind the various ingredients of this formula and describe how its behavior varies as the coding rate exhausts the relevant range among other things it is demonstrated that this expression simultaneously generalizes both the expurgated error exponent function at zero rate and the classical random coding exponent function at high rates where it also meets the sphere packing bound

detectability of increments and decrements in spectral profiles in auditory profile analysis the subject typically has to discriminate a standard spectrum consisting of widely spaced equal amplitude sinusoids from a comparison in which one or several components have been elevated relative to the others according to the widely accepted multichannel model however it should not matter whether the spectral shape alteration consists of incrementing or decrementing a component by a small number of decibels thus in order to investigate the generality of the hypothesized mechanism and to explore the role of pitch salience cues which might be thought to favor increments to single components two experiments comparing the detectability of intensity increments and decrements in spectral profiles were conducted in experiment 1 in which the number of components was varied from 3 to 15 within a fixed frequency range decrements to the central 800 hz component produced significantly larger thresholds than did increments experiment 2 which employed three tone complexes spanning different frequency ranges revealed that this effect is entirely due to the spectral density of the frequency components as the distance between the variable signal component and the flanking reference sinusoids decreased the increment decrement disparity became larger these effects may at least qualitatively be accounted for when peripheral excitation patterns are considered suggesting that local decrements are constrained more than increments by the excitation intruding from flanking stimulus components

variation in spectral shape discrimination weighting functions at different stimulus levels and signal strengths this study evaluated whether weights for spectral shape discrimination depend on overall stimulus level and signal strength the degree of spectral shape change between two stimuli five listeners discriminated between standard stimuli that were the sum of six equal amplitude tones and signal stimuli created by decreasing the amplitudes of three low frequency components and increasing the amplitudes of three high frequency components weighting functions were influenced by stimulus level in that the relative contribution of the low frequency decremented components to the high frequency incremented components decreased with increasing stimulus level although individual variability was present a follow up experiment suggested that the level dependence was due to greater reliance on high frequency components rather than incremented components excitation pattern analyses indicated that the level dependence is primarily but not solely driven by cochlear factors in general different signal strengths had no effect on the weighting functions when normalized but two of the five listeners showed variability in the shape of the weighting function across signal strengths results suggest that the effects of stimulus level on weighting functions and individual variability in the shapes of the weighting functions should be considered when comparing weighting functions across conditions and groups that might require different stimulus levels and signal strengths

nonasymptotic and second order achievability bounds for coding with side information we present a novel nonasymptotic or finite blocklength achievability bounds for three side information problems in network information theory these include 1 the wyner ahlswede korner wak problem of almost lossless source coding with rate limited side information 2 the wyner ziv wz problem of lossy source coding with side information at the decoder and 3 the gel fand pinsker gp problem of channel coding with noncausal state information available at the encoder the bounds are proved using ideas from channel simulation and channel resolvability our bounds for all three problems improve on all previous nonasymptotic bounds on the error probability of the wak wz and gp problems in particular those derived by verdu using our novel nonasymptotic bounds we recover the general formulas for the optimal rates of these side information problems finally we also present achievable second order coding rates by applying the multidimensional berry esseen theorem to our new nonasymptotic bounds numerical results show that the second order coding rates obtained using our nonasymptotic achievability bounds are superior to those obtained using existing finite blocklength bounds

the effect of enhanced spectral contrast on the internal representation of vowel shaped noise two steady state noise excited vowel sounds of 200 ms duration were synthesized one similar to a naturally produced whispered vowel the other with energy removed outside a nominal 50 hz band centered at each formant peak the threshold for a brief sinusoidal probe tone was measured at frequencies corresponding to the peaks and troughs in the vowel spectra in both forward and simultaneous masking the masking functions for a flat spectrum masker at each frequency were also measured so that the vowel masking patterns could be expressed in terms of the equivalent broadband masker level in the first experiment masking patterns for the two maskers were measured with three normal hearing subjects increasing spectral contrast was effective in increasing peak valley differences in the masking patterns up to 2 5 khz contrast was greater in forward than in simultaneous masking suggesting that the internal representation of a vowel may be enhanced by suppression in a second experiment masking patterns and flat spectrum masking functions were measured with three listeners with moderate sensorineural hearing impairments the enhancement produced by increasing spectral contrast was less than that obtained with normal hearing listeners but still present up to 2 5 khz a comparison of forward and simultaneous masking patterns again revealed an increase in peak valley differences suggesting that suppression mechanisms were still effective in these listeners the ability to enhance the internal representation of speechlike sounds in hearing impaired listeners may have implications for the design of signal processing hearing aids

spectral weights for overall level discrimination in listeners with sensorineural hearing loss a conditional on a single stimulus coss analysis procedure b g berg j acoust soc am 86 1743 1746 1989 was used to measure the weight or relative reliance that normal hearing and hearing impaired listeners give to different frequencies in the discrimination of the overall level of a multitone complex on each trial two multitone tone complexes comprised of six octave frequencies from 250 to 8000 hz were presented to subjects the levels of the frequencies for each complex were randomly varied the listeners task was to identify the complex with the higher overall intensity level normal hearing listeners used a variety of listening strategies to perform the task showing no general preference to weight one component over another hearing impaired listeners however showed a general tendency to give greatest weight to the spectral information in the region of their hearing loss thirteen of the 14 hearing impaired listeners all of whom had a high frequency sensorineural hearing loss weighted one or more of the high frequency components in the complex the greatest

sending a bivariate gaussian over a gaussian mac we study the power versus distortion tradeoff for the distributed transmission of a memoryless bivariate gaussian source over a two to one average power limited gaussian multiple access channel in this problem each of two separate transmitters observes a different component of a memoryless bivariate gaussian source the two transmitters then describe their source component to a common receiver via an average power constrained gaussian multiple access channel from the output of the multiple access channel the receiver wishes to reconstruct each source component with the least possible expected squared error distortion our interest is in characterizing the distortion pairs that are simultaneously achievable on the two source components we focus on the equal bandwidth case where the source rate in source symbols per second is equal to the channel rate in channel uses per second we present sufficient conditions and necessary conditions for the achievability of a distortion pair these conditions are expressed as a function of the channel signal to noise ratio snr and of the source correlation in several cases the necessary conditions and sufficient conditions are shown to agree in particular we show that if the channel snr is below a certain threshold then an uncoded transmission scheme is optimal moreover we introduce a source channel vector quantizer scheme which is asymptotically optimal as the snr tends to infinity

sharp bounds for mutual covering a fundamental tool in network information theory is the covering lemma which lower bounds the probability that there exists a pair of random variables among a given number of independently generated candidates falling within a given set we use a weighted sum trick and talagrand s concentration inequality to prove new mutual covering bounds we identify two interesting applications 1 when the probability of the set under the given joint distribution is bounded away from 0 and 1 the covering probability converges to 1 doubly exponentially fast in the blocklength which implies that the covering lemma does not induce penalties on the error exponents in the applications to coding theorems and 2 using hall s marriage lemma we show that the maximum difference between the probability of the set under the joint distribution and the covering probability equals half the minimum total variation distance between the joint distribution and any distribution that can be simulated by selecting a pair from the candidates thus we use the mutual covering bound to derive the exact error exponent in the joint distribution simulation problem in both applications the determination of the exact exponential or double exponential behavior relies crucially on the sharp concentration inequality used in the proof of the mutual covering lemma

distributed channel synthesis two familiar notions of correlation are rediscovered as the extreme operating points for distributed synthesis of a discrete memoryless channel in which a stochastic channel output is generated based on a compressed description of the channel input wyner s common information is the minimum description rate needed however when common randomness independent of the input is available the necessary description rate reduces to shannon s mutual information this paper characterizes the optimal tradeoff between the amount of common randomness used and the required rate of description we also include a number of related derivations including the effect of limited local randomness rate requirements for secrecy applications to game theory and new insights into common information duality our proof makes use of a soft covering lemma known in the literature for its role in quantifying the resolvability of a channel the direct proof achievability constructs a feasible joint distribution over all parts of the system using a soft covering from which the behavior of the encoder and decoder is inferred with no explicit reference to joint typicality or binning of auxiliary interest this paper also generalizes and strengthens this soft covering tool

detection of missing modal frequencies the amount of information about a source that can be recovered from sound is naturally limited by the ear s ability to resolve individual modal frequencies unique to that source to measure these limits listeners were asked to detect in a standard two interval forced choice task with feedback which of two sounds on each trial contained a missing partial the frequencies of the partials corresponded to the ideal modes of a simply supported rectangular plate plate surface areas and height wide ratios were chosen to produce the same bandwidths 125 1125 250 2250 or 500 4500 hz for different numbers of partials 11 16 or 24 overall level of the sounds was roved to discourage detection based on simple level differences detection of five highly practiced listeners was largely independent of the frequency of the lowest partial being best for partials 1 3 1 4 or 1 6 for 11 16 or 24 partials in total respectively mutual masking among the higher number partials is given as the likely cause the res

different auditory filter bandwidth estimates based on profile analysis notched noise and hybrid tasks auditory filter bandwidths were estimated in three experiments the first experiment was a profile analysis experiment the stimuli were composed of sinusoidal components ranging in frequency from 200 to 5000 hz the standard stimulus was the sum of equal amplitude tones and the signal stimulus had a power spectrum that varied up down up down the number of components ranged from four to 60 interval by interval level randomization prevented the change in level of a single component from reliably indicating the change from standard to signal the second experiment was a notched noise experiment in which the 1000 hz tone to be detected was added to a noise with a notch arithmetically centered at 1000 hz detection thresholds were estimated both in the presence of and in the absence of level randomization in the third hybrid experiment a 1000 hz tone was to be detected and the masker was composed of equal amplitude sinusoidal components ranging in frequency from 200 to 5000 hz for this experiment thr

secret key generation with limited interaction a basic two terminal secret key generation model is considered where the interactive communication rate between the terminals may be limited and in particular may not be enough to achieve the maximum key rate we first prove a multi letter characterization of the key communication rate region where the number of auxiliary random variables depends on the number of rounds of the communication and then provide an equivalent but simpler characterization in terms of concave envelopes in the case of unlimited number of rounds two extreme cases are given special attention first in the regime of very low communication rates the key bits per interaction bit kbib is expressed with a new symmetric strong data processing constant which has a concave envelope characterization analogous to that of the conventional strong data processing constant the symmetric strong data processing constant can be upper bounded by the supremum of the maximal correlation coefficient over a set of distributions which allows us to determine the kbib for binary symmetric sources and conclude in particular that the interactive scheme is not more efficient than the one way scheme at least in the low communication rate regime second a new characterization of the minimum interaction rate needed for achieving the maximum key rate mimk is given and we resolve a conjecture by tyagi regarding the mimk for possibly nonsymmetric binary sources we also propose a new conjecture for binary symmetric sources that the interactive scheme is not more efficient than the one way scheme at any communication rate

rate distortion theory for secrecy systems secrecy in communication systems is measured herein by the distortion that an adversary incurs the transmitter and receiver share secret key which they use to encrypt communication and ensure distortion at an adversary a model is considered in which an adversary not only intercepts the communication from the transmitter to the receiver but also potentially has side information in particular the adversary may have causal or noncausal access to a signal that is correlated with the source sequence or the receiver s reconstruction sequence the main contribution is the characterization of the optimal tradeoff among communication rate secret key rate distortion at the adversary and distortion at the legitimate receiver it is demonstrated that causal side information at the adversary plays a pivotal role in this tradeoff it is also shown that measures of secrecy based on normalized equivocation are a special case of the framework

auditory profile analysis frequency phase and weber s law this paper reports three separate experiments on different aspects of performance in auditory profile analysis the first experiment deals with the effects of the frequency and position of an increment in a single component of a multitonal complex the general results indicate that detection of the signal is easier for components in the mid frequency range around 1000 hz independent of signal position within the complex the second experiment investigates the effects of relative phase of the individual components of the complex regardless of the number of components our results indicate that phase has very little effect even when different phases are selected for each presentation the third experiment compares the detection of an increment in intensity of a single component the traditional weber fraction experiment and conditions where additional components are present a profile experiment the detection of the increment is measured as a function of the level of the standard the single tone condition shows the usual near miss to weber s law whereas the multitone condition does not in addition threshold for the increment is better for the multitone condition than for the single tone condition for levels of the standard up to 70 db spl this last result is investigated for ten observers five of whom were experienced in profile tasks and five of whom were not using a low level standard the five experienced observers replicate the results described above the inexperienced observers show the opposite result on average they are better able to detect the increment in the single component condition

the likelihood encoder for lossy compression a likelihood encoder is studied in the context of lossy source compression the analysis of the likelihood encoder is based on the soft covering lemma it is demonstrated that the use of a likelihood encoder together with the soft covering lemma yields simple achievability proofs for classical source coding problems the cases of the point to point rate distortion function the rate distortion function with side information at the decoder i e the wyner ziv problem and the multi terminal source coding inner bound i e the berger tung problem are examined in this paper furthermore a non asymptotic analysis is used for the point to point case to examine the upper bound on the excess distortion provided by this method the likelihood encoder is also related to a recent alternative technique using the properties of random binning

evaluation of simple models of auditory profile analysis using random reference spectra this article describes further study of the finding reported by green et al j acoust soc am 73 639 643 1983 and others that in certain conditions the threshold of detectability for an intensity increment to the center tone of a multitone reference spectrum decreased as the number of nonsignal tones increased that result was considered remarkable since critical band theory would predict that these nonsignal tones spaced outside the critical band containing the signal would have no effect on or at most slightly decrease within band detectability and certainly could not account for the result of improved detectability found in the study cited above recently henn and turner j acoust soc am 88 126 131 1990 were unable to replicate the result described above concluding that the phenomenon exists only in limited conditions and that it is highly individual in nature further they speculated that the most likely reason for the discrepancy between their study and previous s

spectral peak selection in spectral shape discrimination by normal hearing and hearing impaired listeners spectral shape discrimination thresholds were measured in the presence and absence of noise to determine whether normal hearing and hearing impaired listeners rely primarily on spectral peaks in the excitation pattern when discriminating between stimuli with different spectral shapes standard stimuli were the sum of 2 4 6 8 10 20 or 30 equal amplitude tones with frequencies fixed between 200 and 4000hz signal stimuli were generated by increasing and decreasing the levels of every other standard component the function relating the spectral shape discrimination threshold to the number of components n showed an initial decrease in threshold with increasing n and then an increase in threshold when the number of components reached 10 and 6 for normal hearing and hearing impaired listeners respectively the presence of a 50 db spl hz noise led to a 1 7db increase in threshold for normal hearing listeners and a 3 5db increase for hearing impaired listeners multichannel modeling and the relatively s

reduced frequency selectivity and the preservation of spectral contrast in noise reduced frequency selectivity associated with sensorineural hearing loss may pose particular problems for hearing impaired listeners in noisy environments in these situations broader than normal auditory filters may affect the perception of speech by reducing the contrast between spectral peaks and valleys in at least two ways first the peaks and valleys in the internal representation of the speech spectrum become smeared resulting in less precise frequency analysis second there may be a reduction in the signal to noise ratio s n at the output of each auditory filter in order to examine the relationship between frequency selectivity and identification of speechlike stimuli in noise hearing impaired and normal hearing listeners were trained to assign vowel labels to four harmonic complexes which differed in the frequency locations of four elevated peak harmonics peak harmonics were chosen to approximate first and second formant frequencies in four english vowels listeners were then tested to determine the spectral contrast necessary between peak and background components in order to maintain identification accuracy in the presence of various levels of broadband noise results indicated that for these stimuli normal hearing listeners required about 1 db of additional spectral contrast for every doubling of the intensity of noise the required increase in spectral contrast was generally greater for listeners with broader than normal auditory filters at 2000 hz this finding suggests indirectly that in the internal representations of speech sounds embedded in noise the signal to noise ratio for listeners with abnormal frequency selectivity is poorer than for listeners with normal frequency selectivity a poorer than normal internal s n may be one factor underlying the common observation that noise often is more degrading to speech understanding by hearing impaired listeners than by normal hearing listeners

quantum rate distortion reverse shannon theorems and source channel separation we derive quantum counterparts of two key theorems of classical information theory namely the rate distortion theorem and the source channel separation theorem the rate distortion theorem gives the ultimate limits on lossy data compression and the source channel separation theorem implies that a two stage protocol consisting of compression and channel coding is optimal for transmitting a memoryless source over a memoryless channel in spite of their importance in the classical domain there has been surprisingly little work in these areas for quantum information theory in this paper we prove that the quantum rate distortion function is given in terms of the regularized entanglement of purification we also determine a single letter expression for the entanglement assisted quantum rate distortion function and we prove that it serves as a lower bound on the unassisted quantum rate distortion function this implies that the unassisted quantum rate distortion function is nonnegative and generally not equal to the coherent information between the source and distorted output in spite of barnum s conjecture that the coherent information would be relevant here moreover we prove several quantum source channel separation theorems the strongest of these are in the entanglement assisted setting in which we establish a necessary and sufficient condition for transmitting a memoryless source over a memoryless quantum channel up to a given distortion

multiterminal source encoding with one distortion criterion the authors unify earlier investigations concerning the encoding of two correlated sources x sub k y sub k by means of separate encoders decoding is done by a single decoder which receives the outputs from both encoders the reconstruction of x sub k is required to be perfect in the usual shannon sense the authors determine the admissible rate region r d where d is the distortion of the reconstruction of y sub k the binary hamming case is investigated explicitly

the number of components in profile analysis tasks the central issue of this paper is how detection of a change in spectral shape depends on the number of components m that define the spectrum first a channel theory is reviewed and the predictions relating the detectability of such a change as a function of the number of components are derived second the existing literature is reviewed most of it concerns the detection of an increment in a single tone of an m component equal amplitude complex the data are close 2 db to the predictions of the theory which suggests that there is a relatively small change in threshold as the number of components increases the consistency of the data is also discussed and hypotheses for discrepancies are presented finally the results of an experiment measuring the detectability of a special spectral shape change is explored in this case all the changes are equal in size but some are opposite in sign so that the summed change is zero the results using this balanced spectral change are also consistent with the predictions of the theory and show that for this special kind of spectral change thresholds improve by nearly 7 db as m increases from 4 to 24 components

asymptotic regularity conditions for the strong convergence towards weak limit sets and weak attractors of the 3d navier stokes equations abstract the asymptotic behavior of solutions of the three dimensional navier stokes equations is considered on bounded smooth domains with no slip boundary conditions and on periodic domains asymptotic regularity conditions are presented to ensure that the convergence of a leray hopf weak solution to its weak limit set weak in the sense of the weak topology of the space h of square integrable divergence free velocity fields with the appropriate boundary conditions are achieved also in the strong topology it is proved that the weak limit set is strongly compact and strongly attracts the corresponding solution if and only if all the solutions in the weak limit set are continuous in the strong topology of h corresponding results for the strong convergence towards the weak global attractor of foias and temam are also presented in this case it is proved that the weak global attractor is strongly compact and strongly attracts the weak solutions uniformly with respect to uniformly bounded sets of weak solutions if and only if all the global weak solutions in the weak global attractor are strongly continuous in h

level discrimination of single tones in a multitone complex by normal hearing and hearing impaired listeners a conditional on a single stimulus coss analysis procedure j acoust soc am 86 1743 1746 1989 was used to estimate how well normal hearing and hearing impaired listeners selectively attend to individual spectral components of a broadband signal in a level discrimination task on each trial two multitone complexes consisting of six octave frequencies from 250 to 8000 hz were presented to listeners the levels of the individual tones were chosen independently and at random on each presentation the target tone was selected within a block of trials as the 250 1000 or 4000 hz component on each trial listeners were asked to indicate which of the two complex sounds contained the higher level target as a group normal hearing listeners exhibited greater selectivity than hearing impaired listeners to the 250 hz target while hearing impaired listeners showed greater selectivity than normal hearing listeners to the 4000 hz target which is in the region of their hearing loss both groups of listeners displayed large variability in their ability to selectively weight the 1000 hz target trial by trial analysis showed a decrease in weighting efficiency with increasing frequency for normal hearing listeners but a relatively constant weighting efficiency across frequency for hearing impaired listeners interestingly hearing impaired listeners selectively weighted the 4000 hz target which was in the region of their hearing loss more efficiently than did the normal hearing listeners

spectral weights in profile listening the coss analysis suggested by berg b g berg j acoust soc am 86 1743 1746 1989 is applied to a profile listening task the listener s task is to detect an increment in the level of the middle component of an n component spectrum the overall level of the components is randomly selected from a 20 db range on each presentation thus the detection task is essentially one of detecting a change in spectral shape to implement the coss analysis a small perturbation in level is added to each component of the complex coss functions are generated from these perturbations and the spectral weight that the listener assigns to each component is estimated data are reported for n 3 5 and 11 components and for perturbations with standard deviations of 0 5 1 and 2 db the estimated weights are similar to those derived for an optimum detector namely the level at the signal component is compared with the average level of the nonsignal components this result supports the view that profile analysis invol

blow up in finite time for the dyadic model of the navier stokes equations we study the dyadic model of the navier stokes equations introduced by katz and pavlovic they showed a finite time blow up in the ca se where the dissipation degreeis less than 1 4 in this paper we prove the existence of weak solutions for all energy inequality for every weak solution with nonnegative initial data starting from any time local regularity for 1 3 and global regularity for 1 2 in addition we prove a finite time blow up in the case where 1 3 it is remarkable that the model with 1 3 enjoys the same estimates on the nonlinear term as the 4d navier stokes equations finally we discuss a weak global attractor which coincides with a maximal bounded invariant set for alland becomes a strong global attractor for 1 2

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

sensitivity to changes in overall level and spectral shape an evaluation of a channel model two experiments involving level and spectral shape discrimination which test an optimal channel model developed by durlach et al j acoust soc am 80 63 72 1986 are described the model specifies how the auditory system compares and or combines intensity information in different frequency channels in the first experiment psychometric functions were obtained for the discrimination of changes in level and discrimination of changes in spectral shape for an eight tone complex sound a variety of different base spectral shapes were tested in some conditions level randomization was introduced to reduce the reliability of across interval changes in level increasing the amount of level variation degraded performance for the level discrimination task but had no effect on the shape discrimination task in all conditions sensitivity to changes in spectral shape was superior to sensitivity to changes in level consequently two models of central noise are evaluated in an attempt to explain these results one in which central noise acts prior to the formation of the likelihood ratio and one in which central noise degrades the likelihood ratio the former model is more successful in accounting for the data in a second experiment the detectability of a level increment to one component of a multitone complex was measured the frequency content of the complex was varied by systematically removing six components from a 23 component complex thresholds were measured for increments at three different signal frequencies a common trend in the data was that when there was a spectral gap directly above the signal frequency thresholds were lowest this result differs from the predictions of a simple channel model and contrasts with results presented by green and berg q j exp psychol 43a 449 458 1991

a note on the strong convergence towards attractors of damped forced kdv equations abstract this note is a continuation of a previous article which appeared in this journal 1988 74 369 390 in the previous work we showed that a weak dissipation is suffiient to impose a finite dimensional behavior for korteweg de vries equations with space periodic boundary conditions more precisely it was shown that all the trajectories are captured by a finite dimensional attractor however we were only able to show that the convergence of the trajectories holds good in the sense of weak topology the goal of this note is to show that in fact this convergence holds true in the sense of strong topology

successive versus simultaneous comparison in auditory intensity discrimination the ability to discriminate between two brief sounds having identical frequency components but differing in the intensity of one or more of the components is studied the stimulus manipulations include randomizing the overall intensity of the sounds varying the number and spacing of the components and varying the interval of time between the sounds the results from the experiment in which interstimulus interval is manipulated clearly support a profile analysis mechanism that computes two rough spectral analyses stores a classification of these spectra in memory and compares the two stores this profile mechanism involves a simultaneous comparison of different components of the complex some remote from the signal frequency rather than a successive comparison of the difference in intensity at the signal frequency the simultaneous comparison process is more sensitive when the profile is composed of many components spaced over a wide frequency range

strong functional representation lemma and applications to coding theorems this paper shows that for any random variables x and y it is possible to represent y as a function of x z such that z is independent of x and i x z y le log i x y 1 4 bits we use this strong functional representation lemma sfrl to establish a bound on the rate needed for one shot exact channel simulation for general discrete or continuous random variables strengthening the results by harsha et al and braverman and garg and to establish new and simple achievability results for one shot variable length lossy source coding multiple description coding and gray wyner system we also show that the sfrl can be used to reduce the channel with state noncausally known at the encoder to a point to point channel which provides a simple achievability proof of the gelfand pinsker theorem

on possible cues in profile analysis identification of the incremented component the ability to identify the frequency of the incremented component in a multitone complex was assessed in two conditions both using the same 11 tone complexes fixed frequency equilog spaced random phases in the standard plus signal complex all components had equal level except one of the inner nine which was incremented in level by an in phase addition of a sinusoid of the same frequency the signal the overall level and signal position were randomly varied on each presentation in the standard absent condition the standard plus signal complex was presented first and it was followed by a single sinusoid either at the signal frequency or at that of an adjacent component below or above in the standard present condition a complex with equal level components the standard complex was also presented on each trial either before or after the standard plus signal complex in random order these two complexes were followed by the single sinusoid in both conditions the subjects had to indica

weak and strong attractors for the 3d navier stokes system abstract we study in this paper the asymptotic behaviour of the weak solutions of the three dimensional navier stokes equations on the one hand using the weak topology of the usual phase space h of square integrable divergence free functions we prove the existence of a weak attractor in both autonomous and nonautonomous cases on the other we obtain a conditional result about the existence of the strong attractor which is valid under an unproved hypothesis also with this hypothesis we obtain continuous weak solutions with respect to the strong topology of h

the global attractor for the 2d navier stokes flow on some unbounded domains we extend previous results on the existence of the global attractor for the 2d navier stokes equations on some unbounded domains in the sense that the forcing term need not lie in any weighted space nor is the boundary of the domain required to be smooth the existence of the global attractor is obtained on arbitrary open sets such that the poincare inequality holds and for forces in the natural dual space v the proof is based on the energy equation and the concept of asymptotic compactness an estimate of the hausdorff and fractal dimensions of the attractor is also given

error exponents of typical random codes we define the error exponent of the typical random code trc as the long block limit of the negative normalized expectation of the logarithm of the error probability of the random code as opposed to the traditional random coding error exponent which is the limit of the negative normalized logarithm of the expectation of the error probability for the ensemble of uniformly randomly drawn fixed composition codes we provide exact error exponents of trcs for a general discrete memoryless channel and a wide class of stochastic decoders collectively referred to as the generalized likelihood decoder gld this ensemble of fixed composition codes is shown to be no worse than any other ensemble of independent codewords that are drawn under a permutation invariant distribution e g independent identically distributed codewords we also present relationships between the error exponent of the trc and the ordinary random coding error exponent as well as the expurgated exponent for the gld finally we demonstrate that our analysis technique is applicable also to more general communication scenarios such as list decoding for fixed size lists as well as the decoding with an erasure list option in forney s sense