effectiveness of a loudness model for time varying sounds in equating the loudness of sentences subjected to different forms of signal processing a model for the loudness of time varying sounds glasberg and moore 2012 j audio eng soc 50 331 342 was assessed for its ability to predict the loudness of sentences that were processed to either decrease or increase their dynamic fluctuations in a paired comparison task subjects compared the loudness of unprocessed and processed sentences that had been equalized in 1 root mean square rms level 2 the peak long term loudness predicted by the model 3 the mean long term loudness predicted by the model method 2 was most effective in equating the loudness of the original and processed sentences

quantifying the effects of fast acting compression on the envelope of speech fast acting compression has several effects on the envelope of speech signals especially when a target and a background are mixed together three measures of the envelope are described which can be used to quantify these changes 1 within signal modulation correlation or coherence the degree of correlation or coherence of the envelope on a db scale of a single source across different frequency regions which is reduced by fast acting compression 2 fidelity of envelope shape the degree to which the envelope shape of the target speech in different frequency channels is preserved following compression and 3 across signal modulation correlation or coherence asmc the extent to which the target and background acquire a common component of modulation when they are compressed together which becomes greater in absolute value more negative when the target and background are compressed together values of these measures are presented and compared with intelligibility scores obtained using stimuli processed with a noise vocoder so as to preserve mainly envelope cues in a limited number of frequency channels the results suggest that the dominant factor affecting intelligibility is asmc

syzygies of abelian varieties let a be an ample line bundle on an abelian variety x over an algebraically closed field a theorem of koizumi ko s developing mumford s ideas and results m states that if m 3 the line bundle l ao embeds x in projective space as a projectively normal variety moreover a celebrated theorem of mumford m2 slightly refined by kempf k4 asserts that the homogeneous ideal of x is generated by quadrics as soon as m 4 such results turn out to be particular cases of a statement conjectured by rob lazarsfeld concerning the minimal resolution of the graded algebra rl 0 ho x l h over the polynomial ring sl do symhho x l the purpose of this paper is to prove lazarsfeld s conjecture to put such matters into perspective it is useful to review the case of projective curves a classical theorem of castelnuovo states that a curve x embedded in projective space by a complete linear system ili is projectively normal as soon as deg l 2g x 1 and a theorem of mattuck fujita and saint donat states that if deg l 2g x 2 then the homogeneous ideal of x is generated by quadrics green g1 unified re interpreted and generalized these results to a statement about syzygies specifically given a smooth projective variety x and a very ample line bundle l on x a minimal resolution of rl as a graded sl module notation as above looks like

the effects of hearing loss on the contribution of high and low frequency speech information to speech understanding ii sloping hearing loss the speech understanding of persons with sloping high frequency hf hearing impairment hi was compared to normal hearing nh controls and previous research on persons with flat losses hornsby and ricketts 2003 j acoust soc am 113 1706 1717 to examine how hearing loss configuration affects the contribution of speech information in various frequency regions speech understanding was assessed at multiple low and high pass filter cutoff frequencies crossover frequencies defined as the cutoff frequencies at which low and high pass filtering yielded equivalent performance were significantly lower for the sloping hi compared to nh group suggesting that hf hi limits the utility of hf speech information speech intelligibility index calculations suggest this limited utility was not due simply to reduced audibility but also to the negative effects of high presentation levels and a poorer than normal use of speech information in the frequency region with the greatest hearing loss the hf regio

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

effects of spectro temporal modulation changes produced by multi channel compression on intelligibility in a competing speech task these experiments are concerned with the intelligibility of target speech in the presence of a background talker using a noise vocoder stone and moore j acoust soc am 114 1023 1034 2003 showed that single channel fast acting compression degraded intelligibility but slow compression did not stone and moore j acoust soc am 116 2311 2323 2004 showed that intelligibility was lower when fast single channel compression was applied to the target and background after mixing rather than before and suggested that this was partly due to compression after mixing introducing comodulation between the target and background talkers experiment 1 here showed a similar effect for multi channel compression in experiment 2 intelligibility was measured as a function of the speed of multi channel compression applied after mixing for both eight and 12 channel vocoders with one compressor per channel intelligibility decreased as compression speed increased for the eight channel vocoder a compresso

loudness perception and frequency discrimination in subjects with steeply sloping hearing loss possible correlates of neural plasticity loudness functions and frequency difference limens dlfs were measured in five subjects with steeply sloping high frequency sensorineural hearing loss the stimuli were pulsed pure tones encompassing a range of frequencies loudness data were obtained using a 2afc matching procedure with a 500 hz reference presented at a number of levels dlfs were measured using a 3afc procedure with intensities randomized within 6 db around an equal loudness level results showed significantly shallower loudness functions near the cutoff frequency of the loss than at a lower frequency where hearing thresholds were near normal dlfs were elevated on average relative to dlfs measured using the same procedure in five normally hearing subjects but showed a local reduction near the cutoff frequency in most subjects with high frequency loss the loudness data are generally consistent with recent models that describe loudness perception in terms of peripheral excitation patterns that are presumably restricted by a steeply sloping hearing loss however the dlf data are interpreted with reference to animal experiments that have shown reorganization in the auditory cortex following the introduction of restricted cochlear lesions such reorganization results in an increase in the spatial representation of lesion edge frequencies and is comparable with the functional reorganization observed in animals following frequency discrimination training it is suggested that similar effects may occur in humans with steeply sloping high frequency hearing loss and therefore the local reduction in dlfs in our data may reflect neural plasticity

temporal integration and compression near absolute threshold in normal and impaired ears the decrease in absolute threshold with increasing stimulus duration often referred to as temporal integration is greater for listeners with normal hearing than for listeners with sensorineural hearing loss it has been suggested that the difference is related to reduced basilar membrane bm compression in the impaired group the present experiment tested this hypothesis by comparing temporal integration and bm compression in normal and impaired ears at low levels absolute thresholds were measured for 4 24 and 44 ms pure tone signals with frequencies f s of 2 and 4 khz the difference between the absolute thresholds for the 4 and 24 ms signals was used as a measure of temporal integration compression near threshold was estimated by measuring the level of a 100 ms off frequency 0 45f s pure tone forward masker required to mask a 44 ms pure tone signal presented at sensation levels of 5 and 10 db there was a significant negative correlation between amount of temporal integration and absolute threshold however there was no correlation between absolute threshold and compression at low levels both normal and impaired ears showed a nearly linear response the results suggest that the differences in integration between normal and impaired ears cannot be explained by differences in bm compression

temporal integration of loudness measured using categorical loudness scaling and matching procedures temporal integration of loudness of 1 khz tones with 5 and 200 ms durations was assessed in four subjects using two loudness measurement procedures categorical loudness scaling cls and loudness matching cls provides a reliable and efficient procedure for collecting data on the temporal integration of loudness and previously reported nonmonotonic behavior observed at mid sound pressure level levels is replicated with this procedure stimuli that are assigned to the same category are effectively matched in loudness allowing the measurement of temporal integration with cls without curve fitting interpolation or assumptions concerning the form of the loudness growth function

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

response to articulation index predictions for hearing impaired listeners with and without cochlear dead regions j acoust soc am 111 2545 2548 2002 rankovic j acoust soc am 111 2545 2548 2002 argues that the ai calculated according to fletcher s method gives good predictions of speech intelligibility for subjects with high frequency hearing loss listening to amplified speech regardless of the presence or absence of dead regions she suggests further that it seems premature to introduce a clinical test for identifying dead regions because it is not clear what information is gained beyond what is obtainable by calculating the ai i argue that in fact the ai is not accurate in predicting the incremental benefit of amplifying frequencies well above the estimated edge frequency fe of a dead region for five out of nine cases application of the ai leads to the prediction that amplification of frequencies well above fe would improve speech scores by 10 15 in fact the largest improvement was 7

dependence of post masking on masker duration and its relation to temporal effects in loudness temporal masking of tones by noise was investigated using a post forward masking paradigm the masker level and duration were varied for every masker level employed the rate of decay of masking was found to depend on the duration of the masker specific loudness values were computed from the forward masking functions and an electronic device which simulates these loudness functions is presented in the simulation a series of integrators or a single integration with one long time constant was utilized the effect of the proposed integrators on loudness meters is discussed

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

the normalized interaural correlation accounting for nospi thresholds obtained with gaussian and low noise masking noise recently j acoust soc am 103 2578 2589 1998 and j acoust soc am 103 2573 2577 1998 independently reported that greater masking of interaurally phase reversed s tones was produced by diotic low noise noise than by diotic gaussian noise based on quantitative analyses eddins and barber suggested that their results could not be accounted for by assuming that listeners judgments were based on constant criterion changes in the normalized interaural correlation produced by adding the s signal to the diotic masker in particular they showed that a model like the one previously employed by j acoust soc am 100 3774 3784 1996 predicted an ordering of thresholds between the conditions of interest that was opposite to that observed bernstein and trahiotis computed the normalized interaural correlation subsequent to half wave square law rectification and low pass filtering the parameters of which were chosen to mimic peripheral auditory processing in this report it is demonstrat

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

binaural speech intelligibility in noise for hearing impaired listeners the effect of head induced interaural time delay itd and interaural level differences ild on binaural speech intelligibility in noise was studied for listeners with symmetrical and asymmetrical sensorineural hearing losses the material recorded with a kemar manikin in an anechoic room consisted of speech presented from the front 0 and noise presented at azimuths of 0 30 and 90 derived noise signals containing either only itd or only ild were generated using a computer for both groups of subjects speech reception thresholds srt for sentences in noise were determined as a function of 1 noise azimuth 2 binaural cue and 3 an interaural difference in overall presentation level simulating the effect of a monaural hearing aid comparison of the mean results with corresponding data obtained previously from normal hearing listeners shows that the hearing impaired have a 2 5 db higher srt in noise when both speech and noise are presented from the front and 2 6 5 1 db less binaural gain when the noise azimuth is changed from 0 to 90 the gain due to ild varies among the hearing impaired listeners between o db and normal values of 7 db or more it depends on the high frequency hearing loss at the side presented with the most favorable signal to noise s n ratio the gain due to itd is nearly normal for the symmetrically impaired 4 2 db compared with 4 7 db for the normal hearing but only 2 5 db in the case of asymmetrical impairment when itd is introduced in noise already containing ild the resulting gain is 2 2 5 db for all groups the only marked effect of the interaural difference in overall presentation level is a reduction of the gain due to ild when the level at the ear with the better s n ratio is decreased this implies that an optimal monaural hearing aid with a moderate gain will hardly interfere with unmasking through itd while it may increase the gain due to ild by preventing or diminishing threshold effects

a comparison of threshold estimation methods in children 6 11 years of age estimating detection threshold for auditory stimuli in children can be problematic because of lapses in attention and the time limits usually imposed by scheduling restrictions or fatigue data reported here were collected to compare the stability of threshold estimation procedures in testing children ages 6 to 11 in a three alternative forced choice paradigm stimuli consisted of a 1 khz tonal signal and a gaussian noise masker bandpass filtered between 500 2000 hz and presented at 25 db spectrum level the signal was either presented for 400 ms in the presence of a continuous masker simultaneous masking or for 10 ms just prior to a 400 ms masker backward masking for each masking paradigm the 79 correct threshold was assessed via each of three procedures 3 down 1 up adaptive staircase levitt maximum likelihood estimation mle and method of constant stimuli percent correct was measured at the end of the study for a signal 10 db above the previously determined threshold in order to estimate the most appropriate psychometric function asymptote for fitting data collected via the method of constant stimuli both the mle and levitt procedures produced equally stable threshold estimates for both conditions and age groups this was the case despite considerable variability in backward masking thresholds

effect of loudness recruitment on the perception of amplitude modulation people with hearing loss of cochlear origin usually display loudness recruitment the rate of growth of loudness level with increasing sound level is greater than for a normally hearing person loudness recruitment has usually been studied with steady sounds of relatively long duration the present study examines how recruitment affects the perception of dynamically varying sounds namely amplitude modulated sinusoids the modulation rates used 4 8 16 and 32 hz were chosen to span the range of the most prominent modulations present in the envelope of speech three subjects with unilateral cochlear hearing loss were used in experiment 1 subjects were required to make loudness matches between 1 khz tones presented alternately to the two ears this was done over a wide range of sound levels experiment 2 used 1 khz carriers that were amplitude modulated the modulation was sinusoidal on a db scale the modulated tones were presented alternately to the two ears and were approximately equally loud in the two ears the modulation depth was fixed in one ear and the subject was required to adjust the modulation depth in the other ear so that the modulation depth appeared equal in the two ears this was done for a range of modulation depths and with the fixed tone presented to both the normal and the impaired ear a given modulation depth in the impaired ear was matched by a greater modulation depth in the normal ear to a first approximation the modulation matching functions were independent of modulation rate furthermore the functions could be predicted reasonably well from the loudness matching results of experiment 1 obtained with steady tones the results are consistent with the idea that loudness recruitment results from the loss of a fast acting compressive nonlinearity that operates in the normal peripheral auditory system possible implications of the results for the use of fast acting compression in hearing aids are discussed

loudness of tone pulses in a free field investigations of temporal loudness summation of tone pulses have been performed the investigations comprised equal loudness determinations between pairs of tone pulses with a duration ratio of 1 2 and threshold determinations of the same tone pulses pulse durations ranged from 5 to 640 ms the frequencies were 500 1000 and 4000 hz all pulses were shaped by means of 1 3 octave filters for 25 normal hearing observers the investigations were performed at the observer s threshold and at 35 and 55 db spl fitting of the experimental data to a single exponential function yields a time constant of about 200 ms near and at the threshold whereas is about 100 ms at levels well above threshold discrepancies exist nevertheless between this single time constant model and the experimental data obtained for the pulses of shortest duration to account for this a model is proposed comprising a combination of two exponential functions this yields a short time constant of 5 to 10 ms combined with the longer time constant mentioned above

loudness of subcritical sounds as a function of bandwidth center frequency and level level differences at equal loudness between band pass noise and pure tones with a frequency equal to the center frequency of the noise were measured in normal hearing listeners using a loudness matching procedure the center frequencies were 750 1500 and 3000 hz and noise bandwidths from 5 to 1620 hz were used the level of the reference pure tone was 30 50 or 70 db for all center frequencies and reference levels the level at equal loudness was close to 0 db for the narrowest bandwidth increased with bandwidth for bandwidths smaller than the critical bandwidth and decreased for bandwidths larger than the critical bandwidth for bandwidths considerably larger than the critical bandwidth the level difference was negative the maximum positive level difference was measured for a bandwidth close to the critical bandwidth this maximum level difference decreased with increasing reference level a similar effect was found when the level differences were derived from data of an additional categorical loudness scaling experiment the results indicate that the decrease of loudness at equal level with increasing subcritical bandwidth is a common property of the auditory system which is not taken into account in current loudness models

loudness of brief tones measured by magnitude estimation and loudness matching mcfadden j acoust soc am 57 702 704 1975 questioned the accuracy and reliability of magnitude estimation for measuring loudness of tones that vary both in duration and level whereas stevens and hall percept psychophys 1 319 327 1966 reported reasonable group data to gain insight into this discrepancy the present study compares loudness measures for 5 and 200 ms tones using magnitude estimation and equal loudness matches from the same listeners results indicate that both procedures provide rapid and accurate assessments of group loudness functions for brief tones but may not be reliable enough to reveal specific characteristics of loudness in individual listeners

the loudness of sounds whose spectra differ at the two ears moore and glasberg 2007 j acoust soc am 121 1604 1612 developed a model for predicting the loudness of dichotic sounds the model gave accurate predictions of data in the literature except for an experiment of zwicker and zwicker 1991 j acoust soc am 89 756 764 in which sounds with non overlapping spectra were presented to the two ears the input signal was noise with the same intensity in each critical band bark this noise was filtered into 24 bands each 1 bark wide the bands were then grouped into wider composite bands consisting of 1 2 4 or 12 successive sub bands and each composite band was presented either to one ear or the other loudness estimates obtained using a scaling procedure decreased somewhat as the number of composite bands increased and their width decreased but the predictions of the model showed the opposite pattern this experiment was similar to that of zwicker and zwicker except that the widths of the bands were based on the erb n number scale and a loudness matching procedure was used the pattern of the results was consistent with the predictions of the model showing an increase in loudness as the number of composite bands increased and their spacing decreased

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

effect of adaptive psychophysical procedure on loudness matches large variability in equal loudness matches has been observed across studies the purpose of the present study was to gain insight into the extent to which this variability results from differences in psychophysical procedures and or differences among listeners four adaptive two interval two alternatives forced choice procedures were used to obtain equal loudness matches between 5 and 200 ms 1 khz tones as a function of level for each of six normal listeners the procedures differed primarily in the sequence in which the stimuli were presented the variations tested were the ordering of stimuli by amplitude across blocks of trials both increasing and decreasing amplitudes randomizing the order across those blocks and randomizing the order within blocks the random within block procedure which sought to randomize any intertrial information yielded a significantly greater amount of temporal integration than the other three procedures the results show significant differences in temporal integration

the detailed shapes of equal loudness level contours at low frequencies high resolution equal loudness level contours elcs were measured over the frequency range 10 250 hz using 19 normal hearing subjects three levels of the 50 hz reference sound were used corresponding to the levels at 50 hz of the 30 50 and 70 phon standardized elcs given in iso 226 2003 the dynamic range of the contours generally decreased with increasing reference level and the slope was shallow between 10 and 20 hz consistent with previous studies for the lowest level the elcs were sometimes but not always smooth and on average followed the standardized 30 phon contour for frequencies above 40 hz for the two higher levels the individual elcs showed a distinct non monotonic feature in a transition region between about 40 and 100 hz where the slope could reach near zero or even positive values the pattern of the non monotonic feature was similar across levels for the subjects for whom it was observed but the pattern varied across subjects below 40 hz the slopes of the elcs increased m

temporal integration at 6 khz as a function of masker bandwidth thresholds were measured for a 6 khz sinusoidal signal presented within a 500 ms masker the masker was either a bandpass gaussian noise of varying bandwidth or a sinusoid of the same frequency as the signal the spectrum level of the noise masker was kept constant at 20 db spl and the level of the sinusoidal masker was 40 db spl thresholds for signal durations between 2 and 300 ms were measured for masker bandwidths ranging from 60 to 12 000 hz the masker was spectrally centered around 6 khz for masker bandwidths less than 600 hz the slope of the temporal integration function decreased with decreasing masker bandwidth the results are not consistent with current models of temporal integration or temporal resolution it is suggested that the results at narrow bandwidths can be understood in terms of changes in the power spectrum of the stimulus envelope or modulation spectrum according to this view the onset and offset ramps of the signal introduce detectable high frequency components into the modulation spectrum which provide a salient cue in narrowband maskers for broadband maskers these high frequency components are masked by the inherent rapid fluctuations in the masker envelope additionally for signal durations between 7 and 80 ms signal thresholds decreased by up to 5 db as the masker bandwidth increased from 1200 to 12 000 hz the mechanisms underlying this effect are not yet fully understood

effect of spectral change enhancement for the hearing impaired using parameter values selected with a genetic algorithm chen et al j acoust soc am 131 2987 2998 2012 evaluated the effectiveness of an algorithm for enhancing spectral changes over time in improving the intelligibility of speech in background sounds for hearing impaired subjects the processing improved intelligibility for speech in steady speech spectrum noise ssn but tended to impair intelligibility in a background of two talker speech tts large individual differences were found the present study assessed whether the effectiveness of the algorithm was improved when the parameters that controlled the degree and type of enhancement were chosen individually for each subject using a genetic algorithm based on subjective preferences for speech clarity the parameter values selected by the genetic algorithm varied markedly across subjects speech intelligibility was measured for unprocessed stimuli and stimuli processed using the selected parameters with ssn and tts maskers and two signal to masker ratios smrs for each subject the intelligibi

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

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

symmetric explicit numerical integrator for molecular dynamics equations of motion with a generalized friction a general mathematical scheme to construct symmetric explicit numerical integrators of newtonian equations of motion endowed with a generalized friction is provided for a molecular dynamics md study the exact integrations are done for all the decomposed vector fields including the one that contains the friction term on the basis of the symmetric composition scheme with the adjoint for the resulting maps integrators with any local order of accuracy can be systematically constructed among them the second order p2s1 integrator gives the least evaluation of atomic force and potential which are most time consuming in md simulations as examples of the friction function three functional types are considered constant laurent polynomial and exponential with respect to the kinetic energy several md equations of motion fall into these categories and the numerical examinations of their integrators using model systems give positive results on the accuracy and efficiency the extended phase space scheme also presents an invariant function which allows us to easily detect numerical errors in the integration process by monitoring the function value

the effect of amplitude comodulation on extracting sentences from noise evidence from a variety of contexts while signal to noise ratios range from about 50 90 db in laboratory experiments measurements of speech levels in most natural environments show much poorer listening conditions for example teder hear instrum 11 32 33 1990 measured signal to noise ratios ranged from a high of 13 db in a carpeted office to low of 1 db in a 1986 chevy nova traveling at 55 mph these numbers indicate an immense difference between speech signals presented to listeners in laboratory experiments as opposed to the real world environment although speech intelligibility has been well studied in noise there has been little study of the acoustic characteristics of speech that allow the message to be separated from the background noise one characteristic of voiced speech that shows great promise in this regard is amplitude comodulation in the present research amplitude comodulated tone analog sentences carrell and opie percept psychophys 52 437 445 1989 were presented with simultaneous white noise and multispeaker babble at 0 5 and 10 db signal to noise ratios it was found that the beneficial effect of amplitude comodulation was greater at the lower signal to noise ratios

effect of spatial separation extended bandwidth and compression speed on intelligibility in a competing speech task the benefit for speech intelligibility of extending the bandwidth of hearing aids was assessed when the target speech sentences and background two talkers were co located or spatially separated also the relative benefits of slow and fast compression were assessed sixteen hearing impaired hi subjects with mild to moderate high frequency hearing loss and eight normal hearing nh subjects were tested the target and interfering sounds were recorded using a kemar manikin and were located at 60 azimuth either co located or spatially separated simulated binaural hearing aid processing using five channel slow or fast compression was performed offline with gains set individually for each hi subject upper cutoff frequencies were 5 7 5 or 10 khz processed stimuli were presented via headphones for both nh unaided and hi subjects there was no significant effect of cutoff frequency for the co located condition but a small but significant benefit from increasing the cutoff frequency from 5 to 7

rate of loudness growth for pure tones in normal and impaired hearing the present article provides an analysis of loudness growth rates in normal and cochlear impaired hearing for diverse groups with respect to age and backgrounds slopes are obtained from absolute magnitude estimation and magnitude production of loudness measured values and from cross modality matching and absolute magnitude estimation of apparent length predicted values consistent with an earlier study r p hellman and c h meiselman j acoust soc am 88 2596 2606 1990 slopes calculated within the 15 30 db stimulus range above the elevated threshold increase in size with the degree of hearing loss the corresponding range of loudness values in normal hearing yields a slope near 0 60 independent of the threshold levels this pattern of loudness growth holds for individuals and groups but the intersubject variability of the slope is more labile being larger across than within groups and larger for the measured slopes than for the predicted values determined from the predicted slopes the coefficient of variation sigma m is approximately constant in normal and impaired hearing ranging from 20 27 in contrast sigma m obtained from the measured slopes increases with the degree of hearing loss to a value of almost 50 for a 75 db loss the overall stability and systematics of the observed effects further demonstrate that the loudness intensity relation can be specified with reasonable precision and accuracy from cross modality matching

on syzygies of calabi yau varieties and varieties of general type abstract we establish projective normality property n p and effective very ampleness for line bundles on a calabi yau variety as well as for pluricanonical bundles of a variety of general type we also prove a result on the degrees of generators of canonical rings under the condition that the image of the canonical map has minimal degree

testing the concept of softness imperception loudness near threshold for hearing impaired ears buus and florentine j assoc res otolaryngol 3 120 139 2002 have proposed that loudness recruitment in cases of cochlear hearing loss is caused partly by an abnormally large loudness at absolute threshold this has been called softness imperception to evaluate this idea loudness matching functions were obtained using tones at very low sensation levels for subjects with asymmetrical hearing loss matches were obtained for a single frequency across ears for subjects with sloping hearing loss matches were obtained between tones at two frequencies one where the absolute threshold was nearly normal and one where there was a moderate hearing loss loudness matching was possible for sensation levels sls as low as 2 db when the fixed tone was presented at a very low sl in an ear or at a frequency where there was hearing impairment it was matched by a tone with approximately the same sl in an ear or at a frequency where hearing was normal e g 2 db sl matched 2 db sl this relationship held for sls up to 4 10 db depending on the subject these results are not consistent with the concept of softness imperception

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

a note on the dynamics of an oscillator in the presence of strong friction we study the longtime behavior of the solutions of a second order autonomous differential equation differing from the one of a harmonic oscillator by a nonlinear friction term being only holder continuous in particular we show that there are two solution curves reaching the rest point in finite time

effects of low pass filtering on the intelligibility of speech in noise for people with and without dead regions at high frequencies people with high frequency sensorineural hearing loss differ in the benefit they gain from amplification of high frequencies when listening to speech using vowel consonant vowel vcv stimuli in quiet that were amplified and then low pass filtered with various cutoff frequencies vickers et al j acoust soc am 110 1164 1175 2001 found that the benefit from amplification of high frequency components was related to the presence or absence of a cochlear dead region at high frequencies for hearing impaired subjects without dead regions performance improved with increasing cutoff frequency up to 7 5 khz the highest value tested subjects with high frequency dead regions showed no improvement when the cutoff frequency was increased above about 1 7 times the edge frequency of the dead region the present study was similar to that of vickers et al but used vcv stimuli presented in background noise ten subjects with high frequency hearing loss including eight from the study of vickers et al were tested five had dead regions starting below 2 khz and five had no dead regions speech stimuli at a nominal level of 65 db were mixed with spectrally matched noise amplified according to the cambridge prescriptive formula for each subject and then low pass filtered the noise level was chosen separately for each subject to give a moderate reduction in intelligibility relative to listening in quiet for subjects without dead regions performance generally improved with increasing cutoff frequency up to 7 5 khz on average more so in noise than in quiet for most subjects with dead regions performance improved with cutoff frequency up to 1 5 2 times the edge frequency of the dead region but hardly changed with further increases calculations of speech audibility using a modified version of the articulation index showed that application of the cambridge formula was at least partially successful in making high frequency components of the speech audible for subjects with dead regions and that such subjects often failed to benefit from increased audibility of the speech at high frequencies

a note on the nonlinear schrodinger equation in a general domain abstract we consider the cauchy problem for nonlinear schrodinger equations in a general domain r n construction of solutions has been only done by classical compactness method in previous results here we construct solutions by a simple alternative approach more precisely solutions are constructed by proving that approximate solutions form a cauchy sequence in some banach space we discuss three different types of nonlinearities power type nonlinearities logarithmic nonlinearities and damping nonlinearities

stabilization of oscillators subject to dry friction finite time convergence versus exponential decay results we investigate the dynamics of an oscillator subject to dry friction via the following differential inclusion s x t x t f x t 0 t 0 where f r n r is a smooth potential and r n r is a convex function the friction is modelized by the subdifferential term x when 0 int 0 dry friction condition it was shown by adly attouch and cabot 2006 that the unique solution to s converges in a finite time toward an equilibrium state too provided that f x int 0 in this paper we study the delicate case where the vector f x belongs to the boundary of the set 0 we prove that either the solution converges in a finite time or the speed of convergence is exponential when a b 2 2 a 0 b 0 we obtain the existence of a critical coefficient be 0 below which every solution stabilizes in a finite time it is also shown that the geometry of the set 0 plays a central role in the analysis

effects of compression and onset offset asynchronies on the detection of one tone in the presence of another the weaker of two temporally overlapping complex tones can be easier to hear when the tones are asynchronous than when they are synchronous this study explored how the use of fast and slow five channel amplitude compression as might be used in hearing aids affected the ability to use onset and offset asynchronies to detect one signal complex tone when another masking complex tone was presented almost simultaneously a 2 1 compression ratio was used with normal hearing subjects and individual compression ratios and gains recommended by the cam2 hearing aid fitting method were used for hearing impaired subjects when the signal started before the masker there was a benefit of compression for both normal hearing and hearing impaired subjects when the signal finished after the masker there was a benefit of fast compression for the normal hearing subjects but no benefit for most of the hearing impaired subjects except when the offset asynchrony was relatively large 100 ms the benefit of compression probably occurred because the compression improved the effective signal to masker ratio hence reducing backward and forward masking this apparently outweighed potential deleterious effects of distortions in envelope shape and the introduction of partially correlated envelopes of the signal and masker

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

on and off frequency compression estimated using a new version of the additivity of forward masking technique on and off frequency compression at the 4000 and 8000 hz cochlear places were estimated using a new version of the additivity of forward masking afm technique that measures the effects of combining two non overlapping forward maskers instead of measuring signal thresholds to estimate compression of the signal as in the original afm technique the decrease in masker threshold in the combined masker condition compared to the individual masker conditions is used to estimate compression of the masker at the signal place by varying masker frequency it is possible to estimate off frequency compression the maskers were 500 hz wide bands of noise and the signal was a brief pure tone compression at different levels was estimated using different overall signal levels or different masker signal intervals it was shown that the new afm technique and the original afm technique produce consistent results considerable compression was observed for maskers well below the signal frequency suggesting that the assumption of off frequency linearity used in other techniques may not be valid reducing the duration of the first masker from 200 to 20 ms reduced the compression exponent in some cases suggesting a possible influence of olivocochlear efferent activity

forward masking adaptation or integration the aim of this study was to attempt to distinguish between neural adaptation and persistence or temporal integration as possible explanations of forward masking thresholds were measured for a sinusoidal signal as a function of signal duration for conditions where the delay between the masker offset and the signal offset the offset offset interval was fixed the masker was a 200 ms broadband noise presented at a spectrum level of 40 db re 20 pa and the signal was a 4 khz sinusoid gated with 2 ms ramps the offset offset interval was fixed at various durations between 4 and 102 ms and signal thresholds were measured for a range of signal durations at each interval a substantial decrease in thresholds was observed with increasing duration for signal durations up to about 20 ms at short offset offset intervals the amount of temporal integration exceeded that normally found in quiet the results were simulated using models of temporal integration the temporal window model and adaptation for both models the inclusion of a peripheral nonlinearity similar to that observed physiologically in studies of the basilar membrane was essential in producing a good fit to the data both models were about equally successful in accounting for the present data however the temporal window model provided a somewhat better account of similar data from a simultaneous masking experiment using the same parameters this suggests that the linear time invariant properties of the temporal window approach are appropriate for modeling forward masking overall the results confirm that forward masking can be described in terms of peripheral nonlinearity followed by linear temporal integration at higher levels in the auditory system however the difference in predictions between the adaptation and integration models is relatively small meaning that influence of adaptation cannot be ruled out

interaural fluctuations and the detection of interaural incoherence iv the effect of compression on stimulus statistics the purpose of this experiment was to determine whether the normalized interaural cross correlation cc model or a model based on interaural phase and level differences can better describe incoherence detection data the ability to detect interaural incoherence in three sets of reproducible dichotic noises was tested in six listeners the first set contained noises with a constrained value of the cc and the cc including signal compression the second set contained noises with a constrained value of the cc including signal compression the third set contained noises with constrained values in the fluctuations in the interaural differences modeling showed that neither the cc model nor the model using the interaural differences could account for the data in any set examination of the statistical properties of the stimuli showed that including compression before the calculation of the interaural cc causes a substantial correlation of this metric to the fluctuations in the interaural phase difference this finding implies that it may be more difficult to discriminate between the common types of binaural models than previously thought

comparing different estimates of cochlear compression in listeners with normal and impaired hearing a loss of cochlear compression may underlie many of the difficulties experienced by hearing impaired listeners two behavioral forward masking paradigms that have been used to estimate the magnitude of cochlear compression are growth of masking gom and temporal masking tm the aim of this study was to determine whether these two measures produce within subjects results that are consistent across a range of signal frequencies and if so to compare them in terms of reliability or efficiency gom and tm functions were measured in a group of five normal hearing and five hearing impaired listeners at signal frequencies of 1000 2000 and 4000 hz compression values were derived from the masking data and confidence intervals were constructed around these estimates both measures produced comparable estimates of compression but both measures have distinct advantages and disadvantages so that the more appropriate measure depends on factors such as the frequency region of interest and the degree of hearing loss because of the long testing times needed neither measure is suitable for clinical use in its current form

loudness functions for long and short tones 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 long and short 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 200 and 5 ms tones at 1 khz 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 function is 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 low levels where it tends to increas

further evidence that fundamental frequency difference limens measure pitch discrimination difference limens for complex tones dlcs that differ in f0 are widely regarded as a measure of periodicity pitch discrimination however because f0 changes are inevitably accompanied by changes in the frequencies of the harmonics dlcs may actually reflect the discriminability of individual components to test this hypothesis dlcs were measured for complex tones the component frequencies of which were shifted coherently upward or downward by f 0 25 37 5 or 50 of the f0 yielding fully harmonic f 0 strongly inharmonic f 25 37 5 or odd harmonic f 50 tones if dlcs truly reflect periodicity pitch discriminability they should be larger worse for inharmonic tones than for harmonic and odd harmonic tones because inharmonic tones have a weaker pitch consistent with this prediction the results of two experiments showed a non monotonic dependence of dlcs on f with larger dlcs for f s of 25 or 37 5 than for f s of 0 or 50 of f0 these findings are consistent with

interaural fluctuations and the detection of interaural incoherence iii narrowband experiments and binaural models in the first two articles of this series reproducible noises with a fixed value of interaural coherence 0 992 were used to study the human ability to detect interaural incoherence it was found that incoherence detection is strongly correlated with fluctuations in interaural differences especially for narrow noise bandwidths but it remained unclear what function of the fluctuations best agrees with detection data in the present article ten different binaural models were tested against detection data for 14 and 108 hz bandwidths these models included different types of binaural processing independent interaural phase difference interaural level difference lateral position and short term cross correlation several preprocessing transformations of the interaural differences were incorporated compression of binaural cues temporal averaging and envelope weighting for the 14 hz bandwidth data the most successful model postulated that incoherence is detected via fluctuations of interaural phase and interaural level processed by independent centers that model correlated with detectability at r 0 87 that model proved to be more successful than short term cross correlation models incorporating standard physiologically based model features r 0 78 for the 108 hz bandwidth data detection performance varied much less among different waveforms and the data were less able to distinguish between models

equations of kummer varieties let x 0 be a principally polarized abelian variety over a field k of characteristic 2 we may assume that 0 1 0 and that we have an involution i ox 0 ox 0 covering the 1 morphism of x thus we have iln yx n0 yx n0 for all n which has global effect i f x yx n0 r f x yx n0 let f x yx no and f x yx n0 be the 1 and 1 eigenspace of i clearly d3n 0 f x yx n0 is a ring and d3n 0 f x yx n0 is a module over this ring in this paper we will prove

prediction of absolute thresholds and equal loudness contours using a modified loudness model the loudness model described by moore et al j audio eng soc 45 224 240 1997 forms the basis for a recent ansi standard for the calculation of the loudness of steady sounds however the model does not give accurate predictions of the absolute thresholds published in a recent iso standard here it is described how the assumed middle ear transfer function in the model can be modified to give more accurate absolute threshold predictions the modified model also gives reasonably accurate predictions of the equal loudness contours published in a recent iso standard

equal loudness level contours for pure tones equal loudness level contours provide the foundation for theoretical and practical analyses of intensity frequency characteristics of auditory systems since 1956 equal loudness level contours based on the free field measurements of robinson and dadson br j appl phys 7 166 181 1956 have been widely accepted however in 1987 some questions about the general applicability of these contours were published h fastl and e zwicker fortschritte der akustik daga 87 pp 189 193 1987 as a result a new international effort to measure equal loudness level contours was undertaken the present paper brings together the results of 12 studies starting in the mid 1980s to arrive at a new set of contours the new contours estimated in this study are compared with four sets of classic contours taken from the available literature the contours described by fletcher and munson j acoust soc am 5 82 108 1933 exhibit some overall similarity to our proposed estimated contours in the mid frequency range up to 60 phons the contours described by robinson and dadson exhibit clear differences from the new contours these differences are most pronounced below 500 hz and the discrepancy is often as large as 14 db

acoustic and perceptual effects of amplitude and frequency compression on high frequency speech this study investigated how six different amplification methods influence acoustic properties and subsequently perception of high frequency cues in fricatives that have been processed with conventional full bandwidth amplification or nonlinear frequency compression nfc 12 conditions total amplification methods included linear gain fast slow acting wide dynamic range compression crossed with fixed individualized compression parameters and a method with adaptive time constants twenty one hearing impaired listeners identified seven fricatives in nonsense syllables produced by female talkers for nfc stimuli frequency compressed filters that precisely aligned 1 3 octave bands between input and output were used to quantify effective compression ratio audibility and temporal envelope modulation relative to the input results indicated significant relationships between these acoustic properties each of which contributed significantly to fricative recognition across the entire corpus of stimuli recog

on loudness at threshold absolute thresholds for and loudness matches between pure tones and four and ten tone complexes were used to assess the form of the function relating loudness to sensation level sl at low and moderate levels the components of the tone complexes had equal sls and were separated by one two four or six critical bands six listeners with normal hearing were tested the thresholds for the multitone complexes indicate that they generally can be detected even when the level of a single component is a few db below the threshold the average detection advantage is consistent with predictions for multiple observations in independent frequency selective auditory channels but differences among listeners are apparent the loudness matches also vary somewhat among listeners five of the six listeners matched tone complexes composed of subthreshold components to a pure tone a few db above threshold this indicates that the loudness of tones at or even below threshold is greater than zero for these five listeners a simple model of loudness summation was used to obtain loudness functions from the individual listeners loudness matches the slopes of the loudness functions log loudness plotted as a function of log intensity generally exceed unity at low levels and are near 0 2 at 40 db sl this shallow slope at moderate levels agrees with loudness functions derived from data on temporal integration of loudness the average loudness function derived from the present data also is in good agreement with a variety of previous data obtained by magnitude estimation magnitude production ratio production and measurements of binaural loudness summation

effect of enhancement of spectral changes on speech intelligibility and clarity preferences for the hearing impaired most information in speech is carried in spectral changes over time rather than in static spectral shape per se a form of signal processing aimed at enhancing spectral changes over time was developed and evaluated using hearing impaired listeners the signal processing was based on the overlap add method and the degree and type of enhancement could be manipulated via four parameters two experiments were conducted to assess speech intelligibility and clarity preferences three sets of parameter values one corresponding to a control condition two types of masker steady speech spectrum noise and two talker speech and two signal to masker ratios smrs were used for each masker type generally the effects of the processing were small although intelligibility was improved by about 8 percentage points relative to the control condition for one set of parameter values using the steady noise masker at 6 db smr the processed signals were not preferred over those for the control condition except for the steady noise masker at 6 db smr further work is needed to determine whether tailoring the processing to the characteristics of the individual hearing impaired listener is beneficial

horizontal directivity of low and high frequency energy in speech and singing speech and singing directivity in the horizontal plane was examined using simultaneous multi channel full bandwidth recordings to investigate directivity of high frequency energy in particular this method allowed not only for accurate analysis of running speech using the long term average spectrum but also for examination of directivity of separate transient phonemes several vocal production factors that could affect directivity were examined directivity differences were not found between modes of production speech vs singing and only slight differences were found between genders and production levels soft vs normal vs loud more pronounced in the higher frequencies large directivity differences were found between specific voiceless fricatives with s more directional than f in the 4 8 16 khz octave bands

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

temporal integration of loudness in listeners with hearing losses of primarily cochlear origin to investigate how hearing loss of primarily cochlear origin affects the loudness of brief tones loudness matches between 5 and 200 ms tones were obtained as a function of level for 15 listeners with cochlear impairments and for seven age matched controls three frequencies usually 0 5 1 and 4 khz were tested in each listener using a two interval two alternative forced choice 2i 2afc paradigm with a roving level up down adaptive procedure results for the normal listeners generally were consistent with published data e g j acoust soc am 99 1633 1644 1996 the amount of temporal integration defined as the level difference between equally loud short and long tones varied nonmonotonically with level and was largest at moderate levels no consistent effect of frequency was apparent the impaired listeners varied widely but most showed a clear effect of level on the amount of temporal integration overall their results appear consistent with expectations based on knowledge of the general properties of their loudness growth functions and the equal loudness ratio hypothesis which states that the loudness ratio between equal spl long and brief tones is the same at all spls the impaired listeners amounts of temporal integration at high spls often were larger than normal although it was reduced near threshold when evaluated at equal sls the amount of temporal integration well above threshold usually was in the low end of the normal range two listeners with abrupt high frequency hearing losses slopes 50 db octave showed larger than normal maximal amounts of temporal integration 40 to 50 db this finding is consistent with the shallow loudness functions predicted by our excitation pattern model for impaired listeners in modeling sensorineural hearing loss edited by w jesteadt erlbaum mahwah nj 1997 pp 187 198 loudness functions derived from impaired listeners temporal integration functions indicate that restoration of loudness in listeners with cochlear hearing loss usually will require the same gain whether the sound is short or long

loudness relations for individuals and groups in normal and impaired hearing individual and group loudness relations were obtained at a frequency in the region of impaired hearing for 100 people 98 with bilateral cochlear impairment slope distributions were determined from absolute magnitude estimation ame and absolute magnitude production amp of loudness they were also derived from cross modality matching cmm and ame of apparent length with respect to both the means and the individual slope values the two distributions closely agree more than half of the measured deviations are less than 20 with an overall average of 1 5 meaning that transitivity is preserved for bilaterally impaired individuals moreover over the stimulus range where cochlear impairment steepens the loudness function both the group means and the individual slope values are clearly larger than in normal hearing the results also show that for groups of people with approximately similar losses the standard deviation is a nearly constant proportion of the mean slope value giving a coefficient of variation of about 27 in normal and impaired hearing this indicates in accord with loudness matching that the size of the slopes depends directly on the degree of hearing loss the results disclose that loudness measurements obtained by magnitude scaling are able to reveal the operating characteristic of the ear for individuals

relation between the growth of loudness and high frequency excitation to test the conjecture that cochlear impairment and a masking noise in normal ears produce analogous effects on the growth of loudness loudness was measured for 21 listeners all with bilateral symmetrical high frequency cochlear losses beyond the normal hearing region thresholds increased by 30 49 db octave for 10 listeners in group i and by 51 71 db octave for 11 listeners in group ii measured by absolute magnitude estimation and production at two frequencies where thresholds were normal the dynamic range over which loudness grows was not altered by the adjacent hearing loss this was not true however for the slope whereas the overall shape and slope of the loudness functions were essentially the same for group i the mid to high level slope for group ii was noticeably flatter for a tone at the cutoff frequency than for a lower frequency tone in contrast markedly steeper loudness functions characteristic of loudness recruitment were obtained for both groups at a frequency in the region of impaired hearing the reduced rate of loudness growth observed in the region of normal hearing for listeners with steep low pass losses like the reduction usually found for a tone masked by an adjacent high frequency noise is ascribed to the tone s restricted excitation pattern these results indicate that high frequency auditory cues are effectively eliminated by a steeply sloping high frequency hearing loss just as they are eliminated by the addition of an external noise to test the conjecture that cochlear impairment and a masking noise in normal ears produce analogous effects on the growth of loudness loudness was measured for 21 listeners all with bilateral symmetrical high frequency cochlear losses beyond the normal hearing region thresholds increased by 30 49 db octave for 10 listeners in group i and by 51 71 db octave for 11 listeners in group ii measured by absolute magnitude estimation and production at two frequencies where thresholds were normal the dynamic range over which loudness grows was not altered by the adjacent hearing loss this was not true however for the slope whereas the overall shape and slope of the loudness functions were essentially the same for group i the mid to high level slope for group ii was noticeably flatter for a tone at the cutoff frequency than for a lower frequency tone in contrast markedly steeper loudness functions characteristic of loudness recruitment were obtained for both groups at a frequency in the region of im