a priori bounds for semilinear equations and a new class of critical exponents for lipschitz domains abstract a priori bounds for positive very weak solutions of semilinear elliptic boundary value problems u f x u on a bounded domain r n with u 0 on are studied where the nonlinearity 0 f x s grows at most like s p if is a lipschitz domain we exhibit two exponents p and p which depend on the boundary behavior of the green function and on the smallest interior opening angle of we prove that for 1 p p all positive very weak solutions are a priori bounded in l for p p we construct a nonlinearity f x s a x s p together with a positive very weak solution which does not belong to l finally we exhibit a class of domains for which p p for such domains we have found a true critical exponent for very weak solutions in the case of smooth domains p p n 1 n 1 is an exponent which is well known from classical work of brezis turner h brezis r e l turner on a class of superlinear elliptic problems comm partial differential equations 2 1977 601 614 and from recent work of quittner souplet p quittner ph souplet a priori estimates and existence for elliptic systems via bootstrap in weighted lebesgue spaces arch ration mech anal 174 2004 49 81

consonant and vowel confusions in speech weighted noise this paper presents the results of a closed set recognition task for 64 consonant vowel sounds 16c 4v spoken by 18 talkers in speech weighted noise 22 20 16 10 2 db and in quiet the confusion matrices were generated using responses of a homogeneous set of ten listeners and the confusions were analyzed using a graphical method in speech weighted noise the consonants separate into three sets a low scoring set c1 f v b m a high scoring set c2 t s z and set c3 n p g k d with intermediate scores the perceptual consonant groups are c1 f b v c2 s z and c3 m n while the perceptual vowel groups are ae and e the exponential articulation index ai model for consonant score works for 12 of the 16 consonants using a refined expression of the ai finally a comparison with past work shows that white noise masks the consonants more uniformly than speech weighted noise and shows

masking of speech by amplitude modulated noise the masking of speech by amplitude modulated and unmodulated speech spectrum noise has been evaluated by the measurement of monaural speech recognition in such noise on young and elderly subjects with normal hearing and elderly hearing impaired subjects with and without a hearing aid sinusoidal modulation with frequencies covering the range 2 100 hz as well as an irregular modulation generated by the sum of four sinusoids in random phase relation was used modulation degrees were 100 6 db and 12 db root mean square sound pressure level was equal for modulated and unmodulated maskers for the normal hearing subjects essentially all types of modulated noise provided some release of speech masking as compared to unmodulated noise sinusoidal modulation provided more release of masking than the irregular modulation the release of masking increased with modulation depth it is proposed that the number and duration of low level intervals are essential factors for the degree of masking the release of masking was found to reach a maximum at a modulation frequency between 10 and 20 hz for sinusoidal modulation for elderly hearing impaired subjects the release of masking obtained from amplitude modulation was consistently smaller than in the normal hearing groups presumably related to changes in auditory temporal resolution caused by the hearing loss the average speech to noise ratio required for 30 correct speech recognition varied greatly between the groups for young normal hearing subjects it was 15 db for elderly normal hearing it was 9 db for elderly hearing impaired subjects in the unaided listening condition it was 2 db and in the aided condition it was 3 db the results support the conclusion that within the methodological context of the study age as well as sensorineural hearing loss as such influence speech recognition in noise more than what can be explained by the loss of audibility according to the audiogram and the masking noise spectrum

the influence of age and high frequency hearing loss on sensitivity to temporal fine structure at low frequencies l sensitivity to temporal fine structure tfs at low frequencies may be adversely affected by hearing loss at high frequencies even when absolute thresholds at low frequencies are within the normal range however in several studies suggesting this the effects of hearing loss and age were confounded here interaural phase discrimination ipd thresholds for pure tones at 500 and 750 hz were measured for 39 subjects with ages from 61 to 83 yr all subjects had near normal audiometric thresholds at low frequencies but thresholds varied across subjects at high frequencies ipd thresholds were correlated with age ipd thresholds for the test frequency of 750 hz were weakly correlated with absolute thresholds at high frequencies but these correlations became non significant when the effect of age was partialed out the results do not confirm that sensitivity to tfs at low frequencies is influenced by hearing loss at high frequencies independently of age

optimal conditions for l regularity and a priori estimates for semilinear elliptic systems abstract in this paper we present a bootstrap procedure for semilinear elliptic systems with n 3 components combining with the l p l q estimates it yields the optimal l regularity conditions for the three well known types of weak solutions h 0 1 solutions l 1 solutions and l 1 solutions thanks to the linear theory in l p it also yields the optimal conditions for a priori estimates for l 1 solutions based on the a priori estimates we improve known existence theorems for some classes of elliptic systems

speech perception in gated noise the effects of temporal resolution previous investigations have suggested that hearing impaired hi listeners have reduced masking release mr compared to normal hearing listeners nh when they listen in modulated noise the current study examined the following questions that have not been clearly answered first when hi listeners are amplified so that their performance is equal to that of nh listeners in quiet and in steady noise do hi listeners still show reduced mr with modulated noise when compared to nh listeners second is the masking release the same for sentences and cv syllables third does forward masking significantly contribute to the variability in performance among hi listeners to compensate for reduced hearing sensitivity for hi listeners the spectrum levels of both speech and noise were adjusted based on the individual hearing loss there was no significant difference between the performance of nh listeners and that of hi listeners in steady noise and in quiet however the amount of mr for sentences and for cv syllables was significantly reduced for hi listeners for sentence recognition the amount of mr seemed to be more related to hearing sensitivity for low to mid frequencies than to forward masking in contrast forward masking thresholds appear to be a major contributor to the amount of mr for syllable recognition

the effect of tone vocoding on spatial release from masking for old hearing impaired listeners old hearing impaired listeners generally benefit little from lateral separation of multiple talkers when listening to one of them this study aimed to determine how spatial release from masking srm in such listeners is affected when the interaural time differences itds in the temporal fine structure tfs are manipulated by tone vocoding tvc at the ears by a master hearing aid system word recall was compared with and without tvc when target and masker sentences from a closed set were played simultaneously from the front loudspeaker co located and when the maskers were played 45 to the left and right of the listener separated for 20 hearing impaired listeners aged 64 to 86 srm was 3 7 db smaller with tvc than without tvc this difference in srm correlated with mean audiometric thresholds below 1 5 khz even when monaural tfs sensitivity discrimination of frequency shifts in identically filtered complexes was partialed out suggesting that low frequency audiometric thresholds may be a good indicator of candidacy for hearing aids that preserve itds the tvc difference in srm was not correlated with age pure tone itd thresholds nor fundamental frequency difference limens and only with monaural tfs sensitivity before control for low frequency audiometric thresholds

nonexistence of positive solutions to a semilinear elliptic system and blow up estimates for a reaction diffusion system abstract we get simple conditions under which the elliptic system u u p 1 v q 1 v u p 2 v q 2 in r n n 3 with p i q i 1 i 1 2 has no positive radially symmetric solutions then by using this nonexistence result we establish blow up estimates for reaction diffusion system of fujita type u t u u p 1 v q 1 v t v u p 2 v q 2 with the homogeneous dirichlet boundary value condition

prediction of speech intelligibility in spatial noise and reverberation for normal hearing and hearing impaired listeners binaural speech intelligibility of individual listeners under realistic conditions was predicted using a model consisting of a gammatone filter bank an independent equalization cancellation ec process in each frequency band a gammatone resynthesis and the speech intelligibility index sii hearing loss was simulated by adding uncorrelated masking noises according to the pure tone audiogram to the ear channels speech intelligibility measurements were carried out with 8 normal hearing and 15 hearing impaired listeners collecting speech reception threshold srt data for three different room acoustic conditions anechoic office room cafeteria hall and eight directions of a single noise source speech in front artificial ec processing errors derived from binaural masking level difference data using pure tones were incorporated into the model except for an adjustment of the sii to intelligibility mapping function no model parameter was fitted to the srt data of this study the overall correlation coefficient between predicted and observed srts was 0 95 the dependence of the srt of an individual listener on the noise direction and on room acoustics was predicted with a median correlation coefficient of 0 91 the effect of individual hearing impairment was predicted with a median correlation coefficient of 0 95 however for mild hearing losses the release from masking was overestimated

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

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

abnormal processing of temporal fine structure in speech for frequencies where absolute thresholds are normal the identification of nonsense syllables that were lowpass filtered at 1 5khz was compared for subjects with normal hearing and subjects with mild to severe hearing loss at high frequencies but with normal or near normal hearing at low frequencies absolute thresholds were mostly within the normal range 20db hearing level for both groups for frequencies below 1 5khz performance was assessed with intact speech speech that had been processed to preserve only temporal envelope cues in a few frequency bands e speech and speech that had been processed to remove envelope cues as far as possible while preserving temporal fine structure cues again in a few frequency bands tfs speech for the intact speech and e speech the hearing impaired subjects performed slightly more poorly than the normal hearing subjects but this effect was significant only for the intact speech for the tfs speech the hearing impaired subjects performed significantly more poorly than the normal hearing subjects with 12 out o

a unitary model of pitch perception a model of the mechanism of residue pitch perception is revisited it is evaluated in the context of some new empirical results and it is proposed that the model is able to reconcile a number of differing approaches in the history of theories of pitch perception the model consists of four sequential processing stages peripheral frequency selectivity within channel half wave rectification and low pass filtering within channel periodicity extraction and cross channel aggregation of the output the pitch percept is represented by the aggregated periodicity function using autocorrelation as the periodicity extraction method and the summary autocorrelation function sacf as the method for representing pitch information it is shown that the model can simulate new experimental results that show how the quality of the pitch percept is influenced by the resolvability of the harmonic components of the stimulus complex these include i the pitch of harmonic stimuli whose components alternate in phase ii the increased frequency difference limen of tones consisting of higher harmonics and iii the influence of a mistuned harmonic on the pitch of the complex as a function of its harmonic number to accommodate these paradigms it was necessary to compare stimuli along the length of the sacf rather than relying upon the highest peak alone these new results demonstrate that the model responds differently to complexes consisting of low and high harmonics as a consequence it is not necessary to postulate two separate mechanisms to explain different pitch percepts associated with resolved and unresolved harmonics

optimal regularity conditions for elliptic problems via l p_ delta spaces we consider positive solutions of the dirichlet problem for nonlinear elliptic equations where the nonlinearity is assumed to satisfy the polynomial upper growth condition f x u c 1 up it is known from a classical work of brezis and turner that the condition p pbt for a suitable nonlinearity f x u a x up with a x 0 and a l we prove the existence of an unbounded weak solution we next consider the case of systems and show that the polynomial growth conditions for a priori estimates recently obtained by quittner and the author are also optimal our results are strongly connected with the regularity theory of the laplace operator in the spaces l p delta omega the lebesgue spaces weighted by the distance to the boundary as a by product we in turn establish the optimality of the known linear l p delta regularity estimates our proofs are based on the construction of a solution of the laplace equation with a suitable boundary singularity with conical support and we use recent results on the boundary behavior of heat kernels

the importance for speech intelligibility of random fluctuations in steady background noise spectrally shaped steady noise is commonly used as a masker of speech the effects of inherent random fluctuations in amplitude of such a noise are typically ignored here the importance of these random fluctuations was assessed by comparing two cases for one speech was mixed with steady speech shaped noise and n channel tone vocoded a process referred to as signal domain mixing sdm this preserved the random fluctuations of the noise for the second the envelope of speech alone was extracted for each vocoder channel and a constant was added corresponding to the root mean square value of the noise envelope for that channel this is referred to as envelope domain mixing edm it removed the random fluctuations of the noise sinusoidally modulated noise and a single talker were also used as backgrounds with both sdm and edm speech intelligibility was measured for n 12 19 and 30 with the target to background ratio fixed at 7 db for sdm performance was best for the speech background and worst for the steady noise for edm this pattern was reversed intelligibility with steady noise was consistently very poor for sdm but near ceiling for edm demonstrating that the random fluctuations in steady noise have a large effect

weakly split spherical tits systems in quasi reductive groups we will prove that any weakly split spherical tits system b n in g bf g k bf g a quasi reductive k group such as a connected reductive k group satisfying some natural conditions is standard in particular if bf g is anisotropic over k then such a tits system is trivial i e b g n

modeling the identification of concurrent vowels with different fundamental frequencies human listeners are better able to identify two simultaneous vowels if the fundamental frequencies of the vowels are different a computational model is presented which for the first time is able to simulate this phenomenon at least qualitatively the first stage of the model is based upon a bank of bandpass filters and inner hair cell simulators that simulate approximately the most relevant characteristics of the human auditory periphery the output of each filter hair cell channel is then autocorrelated to extract pitch and timbre information the pooled autocorrelation function acf based on all channels is used to derive a pitch estimate for one of the component vowels from a signal composed of two vowels individual channel acfs showing a pitch peak at this value are combined and used to identify the first vowel using a template matching procedure the acfs in the remaining channels are then combined and used to identify the second vowel model recognition performance shows a rapid improvement in

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

prediction of binaural speech intelligibility with frequency dependent interaural phase differences the aim of this study was to test the hypothesis of independent processing strategies in adjacent binaural frequency bands underlying current models for binaural speech intelligibility in complex configurations and to investigate the effective binaural auditory bandwidth in broad band signals speech reception thresholds srts were measured for binaural conditions with frequency dependent interaural phase differences ipds of speech and noise srt predictions with the binaural speech intelligibility model by beutelmann and brand 2006 j acoust soc am 120 331 342 were compared with the observed data the ipds of speech and noise had a sinusoidal shape on a logarithmic frequency scale the bandwidth between zeros of the ipd function was varied from 1 8 to 4 octaves speech and noise had either the same ipd function reference condition or opposite signs of the ipd function binaural condition each condition had two subconditions with alternating and non alternating signs respectively of the ip

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

the effect of interaural delay of the masker on masking level differences in young and old adults diotic sono thresholds and dichotic s pi n pi tau thresholds were measured for young and old adults using a 500 hz pure tone signal and broadband burst masking noise at 37 db spl hz in the dichotic condition both the signal and the masker were phase reversed and the masker was presented with an interaural delay of 0 25 0 5 0 75 1 1 25 1 5 1 75 2 3 or 5 ms masking level differences mlds were determined by subtracting dichotic thresholds from diotic thresholds the sono thresholds for the old subjects did not differ significantly from those for the young subjects however when mlds were plotted as a function of delay the pattern of results differed significantly between young and old subjects this difference in pattern was completely accounted for in terms of a delay line version of durlach s equalization and cancellation ec model n i durlach in foundations of modern auditory theory edited by j v tobias academic new york 1972 b a schneider and p m zurek j acoust soc am 86 1756 1763 1989 by assuming that temporal jitter increases with internal delay in young subjects but that it does not vary with the amount of internal delay in old subjects

linear and nonlinear heat equations in l q_ delta spaces and universal bounds for global solutions we develop a theory of both linear and nonlinear heat equations in the weighted lebesgue spaces l q delta where delta is the distance to the boundary in particular we prove an optimal l q delta l r delta estimate for the heat semigroup and we establish sharp results on local existence uniqueness and local nonexistence of solutions for semilinear heat equations with initial values in those spaces this theory enables us to obtain new types of results concerning positive global solutions of superlinear parabolic problems namely under certain assumptions we prove that any global solution is uniformly bounded for t geq tau 0 by a universal constant independent of the initial data in all previous results the bounds for global solutions were depending on the initial data

effects of sensorineural hearing loss on interaural discrimination and virtual localization cross frequency binaural processing was investigated in listeners with normal hearing nh and with bilateral high frequency sensorineural hearing impairment ih in experiment 1 just noticeable differences for interaural time and interaural intensity were measured using 1 3 octave narrow band noises nbns centered at 0 5 and 4 khz these stimuli were presented in isolation and in different cross frequency interaural combinations ih listeners displayed the best interaural time discrimination when the 0 5 khz nbn was dichotic and the best intensity discrimination when both bands were dichotic both nh listeners time and ih listeners time and intensity displayed the poorest interaural discrimination when the nbns were presented simultaneously with interaural differences in only the 4 khz nbn 0 5 khz nbn dichotic localization accuracy was measured in experiment 2 using the 0 5 and 4 khz nbns in isolation and with 0 5 khz target 4 khz interferer and 4 khz target 0 5 khz interferer conditions best localization of nh and ih subjects was seen for the 0 5 khz target with or without an interferer poorest localization of ih subjects was observed for the 4 khz target and 0 5 khz interferer results suggest that for these ih subjects localization is most difficult when they are forced to rely on interaural information in a higher frequency region with conflicting interaural information at low frequencies

variations on the baer suzuki theorem the baer suzuki theorem says that if p is a prime x is a p element in a finite group g and langle x x g rangle is a p group for all g in g then the normal closure of x in g is a p group we consider the case where x g is replaced by y g for some other p element y while the analog of baer suzuki is not true we show that some variation is we also answer a closely related question of pavel shumyatsky on commutators of conjugacy classes of p elements

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

the influence of different segments of the ongoing envelope on sensitivity to interaural time delays the auditory system is sensitive to interaural timing disparities in the fine structure and the envelope of sounds each contributing important cues for lateralization in this study psychophysical measurements were conducted with customized envelope waveforms in order to investigate the isolated effect of different segments of a periodic ongoing envelope on lateralization one envelope cycle was composed of the four segments attack flank hold duration decay flank and pause duration which were independently varied to customize the envelope waveform the envelope waveforms were applied to a 4 khz sinusoidal carrier and just noticeable envelope interaural time differences were measured in six normal hearing subjects the results indicate that attack durations and pause durations prior to the attack are the most important stimulus characteristics for processing envelope timing disparities the results were compared to predictions of three binaural lateralization models based on the normalized cross correlation coefficient two of the models included an additional stage to mimic neural adaptation prior to binaural interaction involving either a single short time constant 5 ms or a combination of five time constants up to 500 ms it was shown that the model with the single short time constant accounted best for the data

consonant identification in noise using hilbert transform temporal fine structure speech and recovered envelope speech for listeners with normal and impaired hearing consonant identification ability was examined in normal hearing nh and hearing impaired hi listeners in the presence of steady state and 10 hz square wave interrupted speech shaped noise the hilbert transform was used to process speech stimuli 16 consonants in a c a syllables to present envelope cues temporal fine structure tfs cues or envelope cues recovered from tfs speech the performance of the hi listeners was inferior to that of the nh listeners both in terms of lower levels of performance in the baseline condition and in the need for higher signal to noise ratio to yield a given level of performance for nh listeners scores were higher in interrupted noise than in steady state noise for all speech types indicating substantial masking release for hi listeners masking release was typically observed for tfs and recovered envelope speech but not for unprocessed and envelope speech for both groups of listeners tfs and recovered envelope speech yielded similar levels of performance and consonant confusion patterns the masking release observed for tfs and recovered envelope speech may be related to level effects associated with the manner in which the tfs processing interacts with the interrupted noise signal rather than to the contributions of tfs cues per se

mathematical treatment of context effects in phoneme and word recognition percent recognition of phonemes and whole syllables measured in both consonant vowel consonant cvc words and cvc nonsense syllables is reported for normal young adults listening at four signal to noise s n ratios similar data are reported for the recognition of words and whole sentences in three types of sentence high predictability hp sentences with both semantic and syntactic constraints low predictability lp sentences with primarily syntactic constraints and zero predictability zp sentences with neither semantic nor syntactic constraints the probability of recognition of speech units in context pc is shown to be related to the probability of recognition without context pi by the equation pc 1 1 pi k where k is a constant the factor k is interpreted as the amount by which the channels of statistically independent information are effectively multiplied when contextual constraints are added empirical values of k are approximately 1 3 and 2 7 for word and sentence context respe

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

temporal integration of forward masking in listeners having sensorineural hearing loss findings are described from three experiments that were designed to contrast estimates of auditory time and frequency analysis in highly practiced young adult subjects having sensorineural hearing loss of cochlear origin with subjects having normal hearing in one experiment pure tone detection threshold was measured for two durations of tones for the hearing impaired subjects smaller differences were found between thresholds for the two durations of tones than were found for the normally hearing subjects at test frequencies where the sensitivity loss was greatest in a subsequent forward masking experiment masked thresholds were obtained at 3 khz as a function of masker level at several masker durations the results indicated that for the range of values tested masked probe thresholds changed less as a function of masker duration for the hearing impaired subjects than for normally hearing subjects however forward masking grew as a function of masker level more steeply for the hearing impaired listeners than for the normally hearing listeners we believe that this result indicates abnormal temporal processing of sequential sounds in a third experiment psychophysical tuning curves were measured in a forward masking procedure for two maskers that differed in duration by approximately an order of magnitude again the hearing impaired subjects demonstrated less change in forward masking as a function of masker duration than normally hearing subjects further the sharpness of the tuning curves from the hearing impaired listeners was markedly reduced as compared to normal it is believed that the results reflect a disruption of the normal temporal and spectral representation of sounds in the hearing impaired subjects

benefit of high rate envelope cues in vocoder processing effect of number of channels and spectral region in cochlear implants or vocoder simulations of cochlear implants the transmission of envelope cues at high rates related to voice fundamental frequency f0 may be limited by the widths of the filters used to form the channels and or by the cutoff frequency flp of the low pass filters used for envelope extraction the effect of varying flp in tone and noise vocoders was investigated for channel numbers n from 6 to 18 as n increased the widths of the channels decreased the value of flp was 45hz envelope or e filter or 180hz pitch or p filter the following combinations of cutoff frequencies were used for channels below and above 1500hz respectively ee pe ep and pp results from a competing talker task showed that the tone vocoder led to better intelligibility than the noise vocoder the pp condition led to the best intelligibility and the ee condition to the worst for n 6 intelligibility was better for condition pe than for condition ep for n 18 the reverse was true the results

green function for schrodinger operator and conditioned feynman kac gauge consider the schrodinger operator h 2 q in a bounded c1 1 domain d with q klocd d q satisfies the condition sup spec 2 q d v x y g x y e y x exp 0 t q x t d t x y d where xt is the conditioned brownian motion starting at x and ending at y with the life time t and v g can be extended to a continuous and positive function on d d specifically we give the comparison theorem there exists a constant c c d q 0 such that 1 c g x y v x y c g x y x y d another result is that v nz x z x d z d exists and represents the poisson kernel for the dirichlet problem corresponding to schrodinger operator h

behavioral manifestations of audiometrically defined slight or hidden hearing loss revealed by measures of binaural detection this study assessed whether audiometrically defined slight or hidden hearing losses might be associated with degradations in binaural processing as measured in binaural detection experiments employing interaurally delayed signals and maskers thirty one listeners participated all having no greater than slight hearing losses i e no thresholds greater than 25 db hl across the 31 listeners and consistent with the findings of bernstein and trahiotis 2015 j acoust soc am 138 el474 el479 binaural detection thresholds at 500 hz and 4 khz increased with increasing magnitude of interaural delay suggesting a loss of precision of coding with magnitude of interaural delay binaural detection thresholds were consistently found to be elevated for listeners whose absolute thresholds at 4 khz exceeded 7 5 db hl no such elevations were observed in conditions having no binaural cues available to aid detection i e monaural conditions partitioning and analyses of the data revealed that those elevated thresholds 1 were more attributable to hearing level than to age and 2 result from increased levels of internal noise the data suggest that listeners whose high frequency monaural hearing status would be classified audiometrically as being normal or slight loss may exhibit substantial and perceptually meaningful losses of binaural processing

development of a quick speech in noise test for measuring signal to noise ratio loss in normal hearing and hearing impaired listeners this paper describes a shortened and improved version of the speech in noise sin test etymotic research 1993 in the first two of four experiments the level of a female talker relative to that of four talker babble was adjusted sentence by sentence to produce 50 correct scores for normal hearing subjects in the second two experiments those sentences in babble that produced either lack of equivalence or high across subject variability in scores were discarded these experiments produced 12 equivalent lists each containing six sentences with one sentence at each adjusted signal to noise ratio of 25 20 15 10 5 and 0 db six additional lists were also made equivalent when the scores of particular pairs were averaged the final lists comprise the quicksin test that measures the snr a listener requires to understand 50 of key words in sentences in a background of babble the standard deviation of single list scores is 1 4 db snr for hearing impaired subjects based on test retest data a single quicksin list takes approximately one minute to administer and provides an estimate of snr loss accurate to 2 7 db at the 95 confidence level

consonant recognition loss in hearing impaired listeners this paper presents a compact graphical method for comparing the performance of individual hearing impaired hi listeners with that of an average normal hearing nh listener on a consonant by consonant basis this representation named the consonant loss profile clp characterizes the effect of a listener s hearing loss on each consonant over a range of performance the clp shows that the consonant loss which is the signal to noise ratio snr difference at equal nh and hi scores is consonant dependent and varies with the score this variation in the consonant loss reveals that hearing loss renders some consonants unintelligible while it reduces noise robustness of some other consonants the conventional snr loss metric snr50 defined as the snr difference at 50 recognition score is insufficient to capture this variation the snr50 value is on average 12 db lower when measured with sentences using standard clinical procedures than when measured with nonsense syllables a listener with symmetric hearing loss may not have identical clps for both ears some consonant confusions by hi listeners are influenced by the high frequency hearing loss even at a presentation level as high as 85 db sound pressure level

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

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

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

a priori estimates and existence for elliptic systems via bootstrap in weighted lebesgue spaces we present a new general method to obtain regularity and a priori estimates for solutions of semilinear elliptic systems in bounded domains this method is based on a bootstrap procedure used alternatively on each component in the scale of weighted lebesgue spaces lp lp x dx where x is the distance to the boundary using this method we significantly improve the known existence results for various classes of elliptic systems

products of conjugacy classes in finite and algebraic simple groups abstract we prove the arad herzog conjecture for various families of finite simple groups if a and b are nontrivial conjugacy classes then a b is not a conjugacy class we also prove that if g is a finite simple group of lie type and a and b are nontrivial conjugacy classes either both semisimple or both unipotent then a b is not a conjugacy class we also prove a strong version of the arad herzog conjecture for simple algebraic groups and in particular show that almost always the product of two conjugacy classes in a simple algebraic group consists of infinitely many conjugacy classes as a consequence we obtain a complete classification of pairs of centralizers in a simple algebraic group which have dense product a special case of this has been used by prasad to prove a uniqueness result for tits systems in quasi reductive groups our final result is a generalization of the baer suzuki theorem for p elements with p 5

measurements of the binaural temporal window using a detection task two experiments investigated the shape of the binaural temporal window using a detection task in experiment 1 a 10 ms tone burst was presented binaurally out of phase during a burst of white noise which changed from being interaurally uncorrelated to correlated and back to uncorrelated the tone occurred during the correlated portion of the noise in one interval of each 2i fc trial detection thresholds were recorded using a 2 down 1 up adaptive procedure thresholds were measured for different durations of correlated noise 0 960 ms frequencies of tone burst 125 250 500 and 1000 hz and levels of noise 20 30 40 and 50 db spl hz window shapes based on nine candidate functions were fitted to the data using the assumption that the binaural masking release was related to the overall interaural correlation of noise admitted by the window fitted windows included both a forward and a backward lobe gaussian functions tended to give closer fits than exponential and rounded exponential functions and simple functions gave more parsimonious fits that those which included dynamic range limiting terms using simple gaussian fits the shape of the window was largely independent of frequency and level and the windows for individual listeners had equivalent rectangular durations ranging from 55 to 188 ms the asymmetry was variable although forward lobes were generally shorter than backward lobes experiment 2 ruled out the possibility that the forward lobe might be an artefact caused by distraction of the listener when the interaural phase change in the noise closely followed the signal in this experiment the out of phase tone was presented during a burst of partially correlated noise which changed after a variable interval to a fully correlated noise thresholds for detecting the tone rose i e performance worsened as the interval was increased distraction would have produced the opposite effect

perceived lateral position of narrow band noise in hearing impaired and normal hearing listeners under conditions of equal sensation level and sound pressure level the perceived lateral position of narrow band noise nbn was studied in a graphic pointer task as a function of the method of compensation for interaural threshold asymmetries in hearing impaired and normal hearing subjects the method of compensation consisted of equal sensation level eqsl or equal sound pressure level eqspl at the two ears within the same subject the nbn signals were presented at 11 center frequencies with interaural intensity differences iids that varied from 20 to 20 db when equalizing by sl the perceived lateral position is essentially linearly dependent on the degree and direction of asymmetry in asymmetric normal hearing and hearing impaired listeners equalizing by spl shows no such dependency but produces images that are lateralized close to the midline these results reveal that subjects may have adapted to their threshold asymmetries these results will be discussed in terms of the fitting of binaural hearing aids

the importance of temporal fine structure information in speech at different spectral regions for normal hearing and hearing impaired subjects speech reception thresholds srts were measured for target and competing speech signals processed to contain variable amounts of temporal fine structure tfs information signals were filtered into 30 1 erb n wide channels where erb n refers to the bandwidth of normal auditory filters which were either tone vocoded preserving temporal envelope information extracted using the hilbert transform or left unprocessed containing both tfs and envelope information improvements in srt were compared when tfs was progressively introduced starting with the high or low frequency channels the results suggest some redundancy in the tfs information across frequency regions in a second experiment the signal was divided into five 6 erb n wide spectral regions four of which were tone vocoded the remaining region was either absent creating a spectral notch or was present and unprocessed srts were measured for normal hearing and hearing impaired subjects conditions where all channels were vocoded or unprocessed were also included normal hearing subjects benefited similarly when tfs information was added to each region suggesting that tfs information is important over a wide frequency range hearing impaired subjects benefited less although the benefit varied across subjects benefit from tfs information in speech was correlated with a psychophysical measure of tfs sensitivity obtained at two center frequencies

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

the effects of age and cochlear hearing loss on temporal fine structure sensitivity frequency selectivity and speech reception in noise temporal fine structure tfs sensitivity frequency selectivity and speech reception in noise were measured for young normal hearing nhy old normal hearing nho and hearing impaired hi subjects two measures of tfs sensitivity were used the tfs lf test interaural phase difference discrimination and the tfs2 test discrimination of harmonic and frequency shifted tones these measures were not significantly correlated with frequency selectivity after partialing out the effect of audiometric threshold suggesting that insensitivity to tfs cannot be wholly explained by a broadening of auditory filters the results of the two tests of tfs sensitivity were significantly but modestly correlated suggesting that performance of the tests may be partly influenced by different factors the nho group performed significantly more poorly than the nhy group for both measures of tfs sensitivity but not frequency selectivity suggesting that tfs sensitivity declines with age in the absence of elevated audiometric thresholds or broadened auditory filters when the effect of mean audiometric threshold was partialed out speech reception thresholds in modulated noise were correlated with tfs2 scores but not measures of frequency selectivity or tfs lf test scores suggesting that a reduction in sensitivity to tfs can partly account for the speech perception difficulties experienced by hearing impaired subjects

products and commutators of classes in algebraic groups we classify pairs of conjugacy classes in almost simple algebraic groups whose product consists of finitely many classes this leads to several interesting families of examples which are related to a generalization of the baer suzuki theorem for finite groups we also answer a question of pavel shumyatsky on commutators of pairs of conjugacy classes in simple algebraic groups it turns out that the resulting examples are exactly those for which the product also consists of only finitely many classes

phoneme recognition in vocoded maskers by normal hearing and aided hearing impaired listeners this study compared modulation benefit for phoneme recognition obtained by normal hearing nh and aided hearing impaired hi listeners consonant and vowel recognition scores were measured using nonsense syllables in the presence of a steady state noise and four vocoded speech maskers vocoded maskers were generated by modulating the steady state noise in either one or six frequency channels with the speech envelope extracted from the speech of either a single talker or a four talker babble aided hi listeners obtained lower consonant recognition scores than nh listeners in all masker conditions vowel recognition scores for aided hi listeners were comparable to nh scores except in the six channel vocoded masker conditions where they were relatively lower analysis using the extended speech intelligibility index developed by rhebergen versfeld and dreschler 2006 j acoust soc am 120 6 3988 3997 suggested that the signal to noise ratio deficit observed in aided hi listeners was largely due to uncompensated audibility loss there was no significant difference between modulation masking release obtained by nh and aided hi listeners for both consonant and vowel recognition

accounting quantitatively for sensitivity to envelope based interaural temporal disparities at high frequencies bernstein and trahiotis 2009 j acoust soc am 125 3234 3242 reported threshold interaural temporal disparities itds conveyed by the envelopes of 4 khz centered raised sine stimuli a raised sine stimulus consists of a carrier modulated by a sinusoid raised to an exponent such stimuli permitted bernstein and trahiotis to vary independently stimulus modulation frequency modulation depth and relative peakedness deadtime an interaural correlation based model that included stages mimicking peripheral auditory processing captured most of the data save for an overestimation of threshold itds obtained when the depth of modulation was 25 and the raised sine exponent was 8 0 the purpose of the present study was 1 to present a quantitative evaluation of how well other measures including normalized envelope fourth moment envelope peakwidth and envelope deadtime might also account for the data reported by bernstein and trahiotis and 2 to present new threshold itds measured while varying factorially depth of modulation raised sine exponent and modulation frequency quantitative analyses of both the prior and the new data showed that the normalized interaural correlation computed subsequent to peripheral auditory processing provided the most accurate predictions importantly the overestimation of threshold itds did not occur when it was assumed that listeners can employ information within off frequency auditory filters

auditory filter characteristics and consonant recognition for hearing impaired listeners to examine the association between frequency resolution and speech recognition auditory filter parameters and stop consonant recognition were determined for 9 normal hearing and 24 hearing impaired subjects in an earlier investigation the relationship between stop consonant recognition and the articulation index ai had been established on normal hearing listeners based on ai predictions speech presentation levels for each subject in this experiment were selected to obtain a wide range of recognition scores this strategy provides a method of interpreting speech recognition performance among listeners who vary in magnitude and configuration of hearing loss by assuming that conditions which yield equal audible spectra will result in equivalent performance it was reasoned that an association between frequency resolution and consonant recognition may be more appropriately estimated if hearing impaired listeners performance was measured under conditions that assured equivalent audibility of the speech stimuli derived auditory filter parameters indicated that filter widths and dynamic ranges were strongly associated with threshold stop consonant recognition scores for most hearing impaired listeners were not significantly poorer than predicted by the ai model furthermore differences between observed recognition scores and those predicted by the ai were not associated with auditory filter characteristics suggesting that frequency resolution and speech recognition may appear to be associated primarily because both are degraded by threshold elevation

existence and nonexistence results for critical growth polyharmonic elliptic systems abstract in this work we consider semilinear elliptic systems for the polyharmonic operator having a critical growth nonlinearity we establish conditions for existence and nonexistence of nontrivial solutions to these systems

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

a generalisation of a theorem of wielandt abstract in 1974 helmut wielandt proved that in a finite group g a subgroup a is subnormal if and only if it is subnormal in a g for all g g in this paper we prove that the subnormality of an odd order nilpotent subgroup a of g is already guaranteed by a seemingly weaker condition a is subnormal in g if for every conjugacy class c of g there exists c c for which a is subnormal in a c we also prove the following property of finite non abelian simple groups if a is a subgroup of odd prime order p in a finite almost simple group g then there exists a cyclic p subgroup of f g which does not normalise any non trivial p subgroup of g that is generated by conjugates of a

perception of the low pitch of frequency shifted complexes when all of the components in a harmonic complex tone are shifted in frequency by f the pitch of the complex shifts roughly in proportion to f for tones with a small number of components the shift is usually somewhat larger than predicted from pitch theories which has been attributed to the influence of combination tones smoorenburg j acoust soc am 48 924 941 1970 experiment 1 assessed whether combination tones influence the pitch of complex tones with more than five harmonics by using noise to mask the combination tones the matching stimulus was a harmonic complex test complexes were bandpass filtered with passbands centered on harmonic numbers 5 resolved 11 intermediate or 16 unresolved and fundamental frequencies f0s were 100 200 or 400 hz for the intermediate and unresolved conditions the matching stimuli were filtered with the same passband to minimize differences in the excitation patterns of the test and matching stimuli for the resolved condition the matching st

an existence theorem for some semilinear elliptic systems abstract an existence theorem is obtained for a class of semilinear second order uniformly elliptic systems obtained formally from a variational principle and modeled on nonlinear helmholtz systems superlinear growth of the nonlinear term precludes application of standard methods to these systems indeed we permit very rapid growth of the nonlinear term so the underlying functional is not defined on the hilbert space within which a solution is naturally sought mollification of the nonlinear term nonetheless results in the resulting functional satisfying the palais smale condition critical points are determined by solution of a dynamical system the limit of vanishing mollification then produces a weak solution of the original problem

moderate cochlear hearing loss leads to a reduced ability to use temporal fine structure information the ability of normally hearing and hearing impaired subjects to use temporal fine structure information in complex tones was measured subjects were required to discriminate a harmonic complex tone from a tone in which all components were shifted upwards by the same amount in hz in a three alternative forced choice task the tones either contained five equal amplitude components non shaped stimuli or contained many components but were passed through a fixed bandpass filter to reduce excitation pattern changes shaped stimuli components were centered at nominal harmonic numbers n 7 11 and 18 for the shaped stimuli hearing impaired subjects performed much more poorly than normally hearing subjects with most of the former scoring no better than chance when n 11 or 18 suggesting that they could not access the temporal fine structure information performance for the hearing impaired subjects was significantly improved for the non shaped stimuli presumably because they could benefit from spectral cues it is proposed that normal hearing subjects can use temporal fine structure information provided the spacing between fine structure peaks is not too small relative to the envelope period but subjects with moderate cochlear hearing loss make little use of temporal fine structure information for unresolved components

evaluation of a method for enhancing interaural level differences at low frequencies a method called binaural enhancement for enhancing interaural level differences at low frequencies based on estimates of interaural time differences was developed and evaluated five conditions were compared all using simulated hearing aid processing 1 linear amplification with frequency response shaping 2 binaural enhancement combined with linear amplification and frequency response shaping 3 slow acting four channel amplitude compression with independent compression at the two ears agc4ch 4 binaural enhancement combined with four channel compression be agc4ch and 5 four channel compression but with the compression gains synchronized across ears ten hearing impaired listeners were tested and gains and compression ratios for each listener were set to match targets prescribed by the cam2 fitting method stimuli were presented via headphones using virtualization methods to simulate listening in a moderately reverberant room the intelligibility of speech at 60 azimuth in the presence of competing speech on the opposite side of the head at 60 azimuth was not affected by the binaural enhancement processing sound localization was significantly better for condition be agc4ch than for condition agc4ch for a sentence but not for broadband noise lowpass noise or lowpass amplitude modulated noise the results suggest that the binaural enhancement processing can improve localization for sounds with distinct envelope fluctuations

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

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

use of high rate envelope speech cues and their perceptually relevant dynamic range for the hearing impaired the ability of hearing impaired hi listeners to use high rate envelope information in a competing talker situation was assessed in experiment 1 signals were tone vocoded and the cutoff frequency fc of the envelope extraction filter was either 50 hz e filter or 200 hz p filter the channels for which the p or e filter was used were varied intelligibility was higher with the p filter regardless of whether it was used for low or high center frequencies performance was best when the p filter was used for all channels experiment 2 explored the dynamic range over which hi listeners made use of high rate cues in each channel of a vocoder the envelope extracted using fc 16 hz was replaced by the envelope extracted using fc 300 hz either at the peaks or valleys with a parametrically varied switching threshold for a target to background ratio of 5 db changes in speech intelligibility occurred mainly when the switching threshold was between 8 and 8 db relative to the channel root mean square level this range is similar in width to but about 3 db higher in absolute level than that found for normal hearing listeners despite the reduced dynamic range of the hi listeners

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