multiple bursts multiple looks and stream coherence in the release from informational masking in the simultaneous multitone masking paradigm introduced by neff and green percept psychophys 41 409 415 1987 the masker typically is a small number of tones having frequencies and levels that are randomly drawn on every presentation large amounts of masking for a pure tone signal often occur that are thought to reflect central rather than peripheral limitations on processing previous work from this laboratory has indicated that playing a rapid succession of randomly drawn multitone maskers in each observation interval dramatically reduces the amount of masking that is observed relative to a single burst sb in this multiple bursts different mbd procedure the signal tone is the only constant frequency component during the sequence of bursts and tends to perceptually segregate from the masker in this study the number of masker bursts and the interburst interval ibi were varied the goals were to determine how the release from masking relative to the sb condition depends on the number of bursts and to examine whether increasing the ibi would cause each burst to be processed independently if the latter were true it might disrupt the perception of signal stream coherence thereby diminishing the mbd advantage however multiple independent looks could also lead to an improvement in performance for those subjects showing large amounts of informational masking in the sb condition substantial reduction in masked thresholds occurred as the number of masker bursts increased while masking increased as ibi lengthened the results were not consistent with a simple version of a multiple look model in which the information from each burst was combined optimally but instead appear to be attributable to mechanisms involved in the perceptual organization of sounds

temporal integration and multiple looks revisited weights as a function of time this study tests the hypothesis that temporal integration for detection of tone bursts with various durations can be explained by optimally combining multiple looks of brief signal segments whose contribution to detection increases over time detectability was measured for signals consisting of six consecutive 25 ms 1 khz tone pulses presented in a 50 hz wide masker or in maskers consisting of seven 50 hz wide noises one critical band apart with either coherent or incoherent envelopes the level of each signal pulse varied randomly around masked threshold according to a gaussian distribution the slopes of conditional psychometric functions plotted in terms of d 2 as a function of the squared signal pulse intensity for pulses in a particular temporal position yielded estimates of the contribution to detection provided by each pulse results for three normal listeners showed a small but significant effect of the temporal location of the pulse multiple looks predictions of temporal integration functio

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

effect of variability in level on forward masking and on increment detection in the first of four experiments all with the same four subjects varying the level of a forward masker from interval to interval in a two interval forced choice 2ifc adaptive procedure had little effect on threshold in the second experiment the signal level was fixed and performance was measured in units of d varying the level of the forward masker again had little effect analyses of trial by trial data indicated that subjects did not vote for the interval with the higher level masker as would an energy detector performance was better on trials where the masker level in the interval with the signal was lower and was relatively independent of masker level in the nonsignal interval in the third experiment these results were replicated for a wider range of masker variability and with maskers lower in frequency than the signal in the fourth experiment the same range of variability from interval to interval was imposed on the level of the pedestal in an increment detection task results were sim

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

new low roundoff noise realizations of second order digital filter sections new structures for second order filter sections are proposed that yield lower output roundoff noise than many well known structures and are free of zero input limit cycle oscillations these new low roundoff structures are based on these suggested by bomar 1989 and are obtained by putting the constraint that some of the state space coefficients be sum of two power of two terms numerical results are included to illustrate the performance of the proposed structures for narrow band filters and to compare them with other structures

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

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

measures of intensity discrimination several different measures have been employed in the literature to describe performance by human observers in intensity discrimination tasks these measures include the difference limen dl or i in db the weber fraction i i and signal to masker ratio ps pm this article describes the mathematical relationship among these measures for conditions in which masker and signal are 1 sinusoids of the same frequency 2 sinusoids of different frequencies and 3 noise waveforms we provide a table which contains values of each of these measures describing equivalent performance over the typical range obtained in most intensity discrimination experiments we express some cautions that should be observed in choosing and reporting particular measures of intensity discrimination

low roundoff noise and normal realizations of fixed point iir digital filters explicit design and performance equations are given in terms of the parameters of the desired transfer function h z for two classes of realizations these realizations offer the digital circuit designer an alternative to the usual designs based on direct forms in addition to low roundoff noise these new realizations offer other desirable properties such as low coefficient sensitivities and freedom from overflow oscillations numerical results are presented which detail the design process and verify the theory

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

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

masked detection thresholds and temporal integration for noise band signals masked detection thresholds were measured at a center frequency of 2500 hz for a range of noise signal bandwidths w 62 to 6000 hz and durations t 10 to 480 ms approximating that found acoustically in speech the signals were presented in an uncorrelated 480 ms 6000 hz wide masker the masker was presented 1 at a constant spectrum level 53 db spl hz or 2 with the overall level varied randomly over a 50 db range from interval to interval of a trial performance was disrupted in the random level masker only for the condition where the signal and uncorrelated masker were gated on and off simultaneously and were matched spectrally time constants estimated from temporal integration functions fit to the masked detection threshold data were related inversely to w for w broader than the critical bandwidth sensitivity to the noise signals was evaluated in the context of an optimum detection model green 1960 the results did not follow the prediction of a constant bandwidth duration wt product but may be understood in terms of cues available to the listener from the relative combination of signal and masker parameters at least three cues for detection were identified in these experiments 1 a relative timing cue 2 a spectral shape cue and 3 a traditional energy cue compared across observation intervals the relative timing cue and spectral shape cue together contributed as much as a 10 to 12 db advantage relative to detection based on the traditional energy cue alone a new multi cue detection model for predicting the masked detection thresholds was derived predictions from the new model were highly correlated r 0 95 with the empirically measured masked detection thresholds

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

synthesis of symmetric and nonsymmetric lattice digital filters which are free of constant input limit cycles syntheses procedures for lattice digital filter structures which can be free of constant input limit cycles are presented as a result the realizations derived in most cases maintain the desirable properties of the original lattice while being free of constant input limit cycles the symmetric lattice structures which are amenable to efficient vlsi implementation are considered regarding their freedom from zero input and constant input limit cycles the pertaining syntheses procedures are presented experimental results are included to compare the several realizations discussed with respect to output roundoff noise and multiplier coefficients sensitivity

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

effects of degradation of intensity time or frequency content on speech intelligibility for normal hearing and hearing impaired listeners many hearing impaired listeners suffer from distorted auditory processing capabilities this study examines which aspects of auditory coding i e intensity time or frequency are distorted and how this affects speech perception the distortion sensitivity model is used the effect of distorted auditory coding of a speech signal is simulated by an artificial distortion and the sensitivity of speech intelligibility to this artificial distortion is compared for normal hearing and hearing impaired listeners stimuli speech plus noise are wavelet coded using a complex sinusoidal carrier with a gaussian envelope 14 octave bandwidth intensity information is distorted by multiplying the modulus of each wavelet coefficient by a random factor temporal and spectral information are distorted by randomly shifting the wavelet positions along the temporal or spectral axis respectively measured were 1 detection thresholds for each type of distortion and 2 speech reception thresholds for various degrees

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

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

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

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

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

analysis of coefficient quantization errors in state space digital filters in this paper we examine the sensitivity of state space digital filters to coefficient quantization errors a cost function using an l p norm criteria is used to measure the deviation in frequency response of the filter the cost function is used to indicate the connection between roundoff noise and filter sensitivity to coefficient quantization errors then based on this cost function it is shown that low roundoff noise state space digital filters filters in balanced coordinates and filters based on polynomials orthogonal on the unit circle have low sensitivity to coefficient quantization errors furthermore the sensitivity of the filters is shown to depend on the second order modes thereby exhibiting certain robustness and immunity to filter bandwidth

design of state digital filters in this correspondence we develop a very simple algorithm for the design of a second order state space digital filter the design equations presented here were derived on the rendition that the sensitivity of the coefficients in the state space digital filter sdf are spread uniformly to all coefficients of the sdf the design algorithm has been shown to provide sdf realizations having a minimum output round off noise while preserving low coefficient sensitivity and freedom from limit cycles

detection of partially filled gaps in noise and the temporal modulation transfer function results of experiments on the detection of silent intervals or gaps in broadband noise are reported for normal hearing listeners in some preliminary experiments a gap threshold of about 2 ms was measured this value was independent of the duration of the noise burst variation of the noise level on each presentation or the temporal position of the gap within the noise burst in the main experiments the thresholds for partial decrements in the noise waveform as well as brief increments were determined as predicted by a model that assumes a single fixed peak to valley detection ratio thresholds for increments are slightly higher than thresholds for decrements when the signal is measured as the change in rms noise level a first order model describes the temporal properties of the auditory system as a low pass filter with a 7 to 8 ms time constant temporal modulation transfer functions were determined for the same subjects and the estimated temporal parameters agreed well with those estimated from the gap detection data more detailed modeling was carried out by simulating viemeister s three stage temporal model simulations using an initial stage bandwidth of 4000 hz and a 3 ms time constant for the low pass filter generate data that are very similar to those obtained from human subjects in both modulation and gap detection

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

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

roundoff noise in signal estimation and minimization of total real estimator output noise statistical analysis of error due to roundoffs in the arithmetic in signal estimation is given the convenience of modeling this error by an equivalent roundoff noise source is shown matrix relations for computing the roundoff noise in nonrecursive and recursive estimators implemented with fixed point and floating point arithmetic are derived the derived relations are verified by a statistical simulation procedure furthermore relations for determining estimator coefficients which ensure minimum total real estimator output noise are derived the basic performances and advantages of the doubly optimized estimators are shown and verified by characteristic examples the proposed new classes of estimators ensure lower total output noise 5 30 which was the aim of the optimization the doubly optimized recursive estimator shows significantly greater robustness than the standard estimator in relation to the finite accuracy of the coefficients

more economical state space digital filter structures which are free of constant input limit cycles a theorem that establishes conditions for the elimination of constant input limit cycles is used as the basis for a procedure for the synthesis of state space structures three different second order structures are obtained and in each granularity and overflow limit cycles can be eliminated under zero and constant input conditions detailed procedures are then presented for the application of one of the proposed structures to the design of parallel and cascade filters the paper concludes with extensive comparisons which show that more economical low noise and low sensitivity state space filter implementations are possible through the use of one of the proposed structures

alternative discrete time operators an algorithm for optimal selection of parameters we consider the issue of parameter sensitivity in models based on alternative discrete time operators adtos a generic first order adto is proposed that encompasses all the known first order adtos introduced so far in the literature new bounds on the operator parameters are derived and a new algorithm is given for optimally selecting the parameters to give minimum parameter sensitivity

intensity discrimination determined with two paradigms in normal and hearing impaired subjects the literature on auditory intensity jnd s is ambiguous with respect to the relationship between the jnd s measured with gated and continuous pedestals and with respect to changes in this relationship in the presence of loudness recruitment accompanying cochlear pathology in an attempt to clarify these issues and to lay a foundation for systematic investigations of the dependence on the jnd s on loudness functions the jnd s for pure tones with gated and continuous pedestal paradigms of two groups of subjects one with normal hearing and one with hearing loss of cochlear origin were measured the experiments were performed at 0 5 2 and 6 khz and at a wide range of sensation levels sls by means of an adaptive two alternative forced choice 2ifc procedure the jnd s obtained with the continuous pedestal method were smaller than those obtained with the gated pedestal method for both groups of subjects they also had smaller intersubject standard deviations when jnd s of the two groups were compared on the basis of equal sls the group with hearing loss showed smaller jnd values than the group with normal hearing for both pedestal paradigms when the comparisons were made on the basis of equal sound pressure levels spls both groups showed similar values for moderate and high spls at relatively low spls the group with hearing loss tended to have somewhat higher values

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

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

roving level tone in noise detection the detectability of tones or of intensity increments to tones in bands of random noise was measured for conditions in which the overall level was fixed or was randomly roved from interval to interval of every experimental trial the purpose of the within trial rove was to limit the usefulness of a detection strategy based on overall level or level within a single critical band at supracritical bandwidths the functions relating masked threshold to noise bandwidth for the roved conditions were similar to those obtained when no rove was employed at subcritical bandwidths thresholds were higher in some roved conditions but for the largest rove were still lower than would be predicted from arguments based purely on level detection with one exception a comparison of observer performance relative to the statistical limits imposed by the roving level procedure indicated that the traditional critical band energy detector model could not account for the results which are attributed to discrimination based on spectral shape or on waveshape

relative effects of increment and pedestal duration on the detection of intensity increments the detection of a brief increment in the intensity of a longer duration pedestal is commonly used as a measure of intensity resolution increment detection is known to improve with increasing duration of the increment and also with increasing duration of the pedestal but the relative effects of these two parameters have not been explored in the same study in several past studies of the effects of increment duration pedestal duration was increased as increment duration increased in the present study increment and pedestal duration were independently manipulated increment detection thresholds were determined for four subjects with normal hearing using a 500 or 4000 hz pedestal presented at 60 db sound pressure level spl increment durations were 10 20 40 80 160 and 320 ms pedestal durations were 20 40 80 160 and 320 ms each increment duration was combined with all pedestals of equal or greater duration multiple regression analyses indicate that increment detection under these conditions is determined primarily by pedestal duration follow up experiments ruled out effects of off frequency listening or overshoot the results suggest that effects of increment duration have been confounded by effects of pedestal duration in studies that co varied increment and pedestal duration implications for models of temporal integration are discussed

psychoacoustic and phonetic temporal processing in normal and hearing impaired listeners four measures of auditory temporal processing were obtained from 16 normals and 16 individuals with a hearing loss of heterogeneous origin these measures were 1 temporal integration the difference in detection thresholds between signals of 10 and 1000 ms duration which was determined to provide an estimate of the ability to integrate energy over time 2 gap detection the shortest duration of silence between two noise bursts that can be discriminated from an uninterrupted noise 3 temporal difference limen the increment in duration necessary to detect a difference in the duration of a noise burst 4 gap difference limen the increment in duration necessary to detect a difference in the duration of a silent interval between two noise bursts each measure was obtained for stimuli centered both at 500 and at 4000 hz using a three alternative forced choice procedure in addition measures of identification and discrimination were obtained for two sets of synthetic speech syllables varying chiefly in a temporal parameter voice onset time from ba to pa and from bi to pi finally speech identification in noise was measured with the faaf test most of the hearing impaired listeners displayed poorer temporal analysis than the normals on all of the psychoacoustical tasks regardless of whether the two groups were compared at similar sound pressure levels or at similar sensation levels although the hearing impaired listeners displayed a reduction in the ability to discriminate subphonemic cues for the voiced voiceless distinction their identification of that distinction in stop consonants appeared to be normal the hearing impaired group made about twice as many errors as did the normals on each of the consonant features of place manner and voicing when identifying speech in noise increased temporal difference limen and longer gap detection thresholds were found to correlate significantly with reduced speech intelligibility in noise even when the effects of the pure tone threshold loss were partialed out

new second order state space structures for realizing low roundoff noise digital filters several new second order state space digital filter structures are introduced each of the new structures requires one or two less multiplies than the minimum noise structure while achieving a round off noise performance which is nearly as good thus they provide an excellent means of realizing second order subfilters when such subfilters are used to construct nth order filters the reduction in multiplies as compared to using minimum noise subfilters is from about 4n to about 3n an approach to developing simple algebraic real arithmetic design equations for l 2 norm scaled second order structures is presented and then applied to obtaining design equations for the new structures and for the minimum noise structure

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

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

finite wordlength design of digital kalman filters for state estimation the optimal design of a kalman filter is considered with respect to its finite wordlength fwl characteristics taking into account the roundoff noise due to state quantization the issues are particularly relevant in the design of fwl kalman filters for continuous time systems operating under a fast sampling rate in this respect the results demonstrate one compromise between the selection of the sampling rate and the selection of the state wordlength the optimum filter structure includes state residue feedback compensation which can result in the saving of many bits of additional state wordlength

a test of the equal loudness ratio hypothesis using cross modality matching functions this study tests the equal loudness ratio hypothesis florentine et al j acoust soc am 99 1633 1644 1996 which states that the loudness ratio between equal spl long and short tones is independent of spl the amount of temporal integration i e the level difference between equally loud short and long sounds is maximal at moderate levels therefore the equal loudness ratio hypothesis predicts that the loudness function is shallower at moderate levels than at low and high levels equal loudness matches and cross modality string length matches were used to assess the form of the loudness function for 5 and 200 ms tones at 1 khz and the loudness ratio between them results from nine normal listeners show that 1 the amount of temporal integration is largest at moderate levels in agreement with previous studies and 2 the loudness functions are shallowest at moderate levels for eight of the nine listeners the loudness ratio between the 200 and 5 ms tones is approximately constant except at

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

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

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

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

a topological index condition for conical controllability covers the interior of a convex set containing e q and a neighborhood of a y w we say that p1 is controllable at 4 if it is w conically controllable at w 0 these concepts are motivated by problems of the optimal control theory and are closely related to the investigation of attainable sets certain higher order sufficient conditions for controllability and conical controllability were derived in l 21 these conditions also ensure that controllability respectively conical controllability is preserved when 9 p2 is replaced by v e pz ez for all higher order perturbations et ej in the present paper we generalize the basic result of 2 by replacing a homeomorphism condition with a more general nonzero index condition let v be an open subset of ky h i r continuous and w e r h av we refer to the elements of g w as w points of g let 71 be a bounded connected component of h w admitting an open neighborhood u whose closure is contained in i and is disjoint from other components of h w we refer to such a component rc as a bounded isolated w patch of h and denote by i h rrw w the topological degree d h u w for any choice of u as described above 3 we refer to i h rr w as the index of h at w with respect to rc we also denote by jh x the jacobian determinant det h x

higher order conditions with and without lagrange multipliers let q be a convex subset of a vector space mathcal u subset q mathcal s a topological vector space c a convex subset of mathcal z with a nonempty interior phi phi 1 phi 2 q to mathbb r m times mathcal z bar q in q and phi 2 bar q in c we assume that phi has a pth order taylor approximation at bar q when it is restricted to an arbitrary finite dimensional simplex in q with a vertex at bar q in the case when mathcal u is a proper subset of q we also assume that qis a uniform space phi continuous and mathcal u abundant we establish a number of higher order sufficient conditions not involving any lagrange multipliers for the existence of neighborhoods g 1 and g 2 of the origins such that phi 1 bar q g 1 subset phi 1 u u in mathcal u phi 2 u g 2 subset c these sufficient conditions involving nonconvex sets of variations are shown by an example to be stronger than those in the literature we

auditory filter shapes of normal hearing and hearing impaired listeners in continuous broadband noise listeners with sensorineural hearing impairment typically exhibit auditory processing deficits such as reduced frequency and or temporal resolution such deficits may represent separate sequela of auditory pathology or may result directly from the sensitivity loss and the requirement to listen at high levels to assess the impact of increased thresholds on frequency resolution auditory filter characteristics were determined for hearing impaired and normal hearing listeners at 500 and 2000 hz in the presence of continuous broadband noise meant as a rough simulation of hearing loss in the fitting procedure the low frequency skirt of the derived auditory filter was allowed to vary as a function of signal level permitting different filter shapes to be estimated at high versus low signal levels listeners with moderate hearing losses at 2000 hz demonstrated near normal auditory filter shapes for lower signal levels but increasingly broad and asymmetric filters as signal level was raised at 500 hz where hearing losses were mild filter bandwidths increased little at the higher signal levels the presence of broadband noise had essentially no effect on filter shapes of either listener group the filter shape abnormalities demonstrated by listeners with moderate hearing loss which were not observed in normal hearing listeners at the same signal levels indicate that poor frequency resolution in these patients for high intensity stimuli does not follow directly from decreased sensitivity but instead reflects an independent pathology

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

the psychophysics of absolute threshold and signal duration a probabilistic approach the absolute threshold for a tone depends on its duration longer tones have lower thresholds this effect has traditionally been explained in terms of temporal integration involving the summation of energy or perceptual information over time an alternative probabilistic explanation of the process is formulated in terms of simple equations that predict not only the time duration dependence but also the shape of the psychometric function at absolute threshold it also predicts a tight relationship between these two functions measurements made using listeners with either normal or impaired hearing show that the probabilistic equations adequately fit observed threshold duration functions and psychometric functions the mathematical formulation implies that absolute threshold can be construed as a two valued function a gain and b sensory threshold and both parameters can be estimated from threshold duration data sensorineural hearing impairment is sometimes associated with a smaller threshold duration effect and sometimes with steeper psychometric functions the equations explain why these two effects are expected to be linked the probabilistic approach has the potential to discriminate between hearing deficits involving gain reduction and those resulting from a raised sensory threshold

summation bandwidths at threshold in normal and hearing impaired listeners the bandwidths for summation at threshold were measured for subjects with normal hearing and subjects with sensorineural hearing loss thresholds in quiet and in the presence of a masking noise were measured for complex stimuli consisting of 1 to 40 pure tone components spaced 20 hz apart the single component condition consisted of a single pure tone at 1100 hz additional components were added below this frequency in a replication of the gassler acustica 4 408 414 1954 procedure for the normal subjects thresholds increased approximately 3 db per doubling of bandwidth for signal bandwidths exceeding the critical bandwidth this slope was less for the hearing impaired subjects summation bandwidths as estimated from two line fits were wider for the hearing impaired than for the normal subjects these findings provide evidence that hearing impaired subjects integrate sound energy over a wider than normal frequency range for the detection of complex signals a second experiment used stimuli simila

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

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

higher order conditions for conical controllability let q be a convex subset of a linear space mathcal u subset q mathcal z a topological vector space c subset mathcal z convex with a nonempty interior bar q in q varphi varphi 1 varphi 2 q to mathbb r m times mathcal z and varphi 2 bar q in c also let mathcal a u in mathcal u varphi 2 u in c this framework can be used to model many optimization problems including control problems with unilateral state inequality constraints sufficient conditions are derived for varphi 1 mathcal a to cover the interior of a convex set containing varphi 1 bar q and an open neighborhood of a point varphi 1 bar q w for w in mathbb r m these sufficient conditions can be used to provide information about the attainable sets of control problems and in some cases to show that varphi 1 mathcal a contains a neighborhood of varphi 1 bar q several illustrative examples are also discussed

stability of digital filters implemented with two s complement truncation quantization the stability of digital filters implemented with two s complement truncation quantization is investigated both the direct form and normal form realizations are considered and regions in the parameter plane are obtained where the filters are globally asymptotically stable regions within the stability triangle of a direct form digital filter are also studied where limit cycles of period one and two are possible the results are illustrated with several examples

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