Biomedical subjects

Brian C J Moore

Publications and source records attributed to Brian C J Moore.

At least 19 recordsLinked to original sources

PubMed2003

Enhanced frequency discrimination near the hearing loss cut-off: a consequence of central auditory plasticity induced by cochlear damage?

Patients with steeply sloping hearing losses of cochlear origin may exhibit enhanced difference limens for frequency (DLFs) near the cut-off frequency (Fc) of their hearing loss. This effect has been related to observations in deafened animals of an over-representation of Fc in the primary auditory cortex. However, alternative interpretations in terms of peripheral mechanisms have not been eliminated. In the present study, we assessed the possible role of two peripheral mechanisms [loudness cues and spontaneous otoacoustic emissions (SOAEs)] in a group of patients with high-frequency hearing loss. We tested first whether the DLF enhancement effect was still observed under conditions where subjects could not rely on loudness cues to perform the frequency discrimination task. To achieve this, we adjusted the nominal level of each stimulus so that it fell on an equal loudness contour measured at very fine (1/8 octave) frequency intervals, and we roved the level of each stimulus over a large range (12 dB). Under these conditions, the DLF enhancement was still observed in all patients; this demonstrates that the effect cannot be explained simply by loudness cues. We then screened the patients for SOAEs to test whether the DLF enhancement effect could be explained by the presence of such emissions in the vicinity of the Fc. None of the patients exhibited SOAEs. Finally, we tested whether the patients had cochlear dead regions, i.e. regions lacking functional inner hair cells and/or auditory nerve fibres. Using a refined version of a non-invasive clinical test for the identification of dead regions, we assessed the presence of such regions in fine frequency steps (1/4 octave) up to very high frequencies. All of the patients had cochlear dead regions. The first two findings support the hypothesis that DLF enhancement is due to injury-induced central reorganization in the auditory system. The last one is consistent with neurophysiological data in animals, which suggest that complete deprivation from auditory input at certain cochlear sites may be a necessary condition for the occurrence of injury-induced cortical reorganization.

Aged↗

PubMed2003

Behavioural measurement of level-dependent shifts in the vibration pattern on the basilar membrane at 1 and 2 kHz.

Physiological data suggest that the peak of the travelling wave on the basilar membrane evoked by a high-frequency sinusoid moves towards the base with increasing level. Previously, we used a forward-masking technique to provide evidence for a similar effect in humans at 4 and 6.5 kHz. In the present study, we used a similar technique to determine whether level-dependent shifts occur for mid-range frequencies. The signal was a brief 1-kHz or 2-kHz tone presented at 10 dB SL (approximately 30 dB SPL). For three fixed masker levels (75, 85 and 95 dB SPL), we measured the duration of the gap between the masker and signal required to give 79.4% correct detection of the signal (called the 'gap threshold') as a function of masker frequency; the longer the gap threshold, the more effective is the masker. The gap-threshold patterns nearly always showed a single peak close to the signal frequency. The gap-threshold patterns spread markedly towards lower frequencies with increasing masker level, but the frequency at the peak did not change systematically with level. We conclude that, for mid-range frequencies, the peak of the travelling wave does not shift significantly with increasing level over the range 30-95 dB SPL, but the envelope of the travelling wave becomes more shallow on its basal side.

Acoustic Stimulation↗

PubMed2003

Discrimination of the fundamental frequency of complex tones with fixed and shifting spectral envelopes by normally hearing and hearing-impaired subjects.

Difference limens for the fundamental frequency (F0) of complex tones (DLCs) were measured for four normally hearing subjects and three subjects with cochlear hearing loss. The nominal F0 was 100, 200 or 400 Hz. The two tones to be compared contained either low resolved harmonics (RES), harmonics with intermediate resolvability (INT) or high unresolved harmonics (UNRES). For one set of stimuli (fixed harmonics, FH), the tones to be compared contained three harmonics of fixed number, so the changes in F0 were associated with spectral cues. For a second set of stimuli (Shaped), spectral cues were minimized by filtering stimuli through a fixed passband. For the INT and UNRES conditions, the excitation patterns evoked by the Shaped stimuli hardly changed when F0 was altered. To prevent subjects from comparing the frequencies of individual harmonics in the RES condition, subjects were required to detect F0 differences between two tones with non-overlapping harmonics. It was not possible to obtain repeatable results for the hearing-impaired subjects in this condition. The normally hearing subjects had smaller DLCs for the FH than for the Shaped stimuli for the RES condition, and the UNRES condition at the two higher F0s. However, DLCs were similar for the FH and Shaped stimuli for the INT condition and the UNRES condition at the 100-Hz F0, suggesting that spectral cues were not used in these conditions. Except for one subject with F0=400 Hz, the hearing-impaired subjects had smaller DLCs for the FH than for the Shaped stimuli, for both INT and UNRES conditions (although the difference was small for F0=200 Hz in the UNRES condition), suggesting that they used spectral cues for the FH stimuli. For the Shaped stimuli, DLCs were similar in the INT and UNRES conditions for the hearing-impaired subjects, but were smaller in the INT than the RES condition for the normally hearing subjects. We suggest that, in the INT condition with Shaped stimuli, normally hearing subjects used temporal fine structure cues to perform the task. The hearing-impaired subjects appeared to use only temporal envelope cues.

Acoustic Stimulation↗

PubMed2003

Coding of sounds in the auditory system and its relevance to signal processing and coding in cochlear implants.

OBJECTIVE: To review how the properties of sounds are "coded" in the normal auditory system and to discuss the extent to which cochlear implants can and do represent these codes. DATA SOURCES: Data are taken from published studies of the response of the cochlea and auditory nerve to simple and complex stimuli, in both the normal and the electrically stimulated ear. REVIEW CONTENT: The review describes: 1) the coding in the normal auditory system of overall level (which partly determines perceived loudness), spectral shape (which partly determines perceived timbre and the identity of speech sounds), periodicity (which partly determines pitch), and sound location; 2) the role of the active mechanism in the cochlea, and particularly the fast-acting compression associated with that mechanism; 3) the neural response patterns evoked by cochlear implants; and 4) how the response patterns evoked by implants differ from those observed in the normal auditory system in response to sound. A series of specific issues is then discussed, including: 1) how to compensate for the loss of cochlear compression; 2) the effective number of independent channels in a normal ear and in cochlear implantees; 3) the importance of independence of responses across neurons; 4) the stochastic nature of normal neural responses; 5) the possible role of across-channel coincidence detection; and 6) potential benefits of binaural implantation. CONCLUSIONS: Current cochlear implants do not adequately reproduce several aspects of the neural coding of sound in the normal auditory system. Improved electrode arrays and coding systems may lead to improved coding and, it is hoped, to better performance.

Cochlea↗

PubMed2003

Tolerable hearing aid delays. III. Effects on speech production and perception of across-frequency variation in delay.

OBJECTIVE: Signal processing strategies that attempt to mimic the frequency resolution of the healthy cochlea require finer frequency resolution at low frequencies than at high. The filtering required will cause more delay to the low-frequency end of the spectrum than to the high-frequency end, which may have disturbing effects. In a real-time application, making the delay constant across frequency would require the higher frequency components to be subjected to a compensating delay, leading to a larger overall delay, which may also give rise to disturbing effects (Stone & Moore, 1999, 2002). This experiment assessed the effects of frequency-dependent delay on subjective and objective measures of speech production and perception. DESIGN: Digital filtering was used to introduce across-frequency delays of 0, 4, 9, 15, or 24 msec. The low frequencies were always more delayed than the high frequencies, with a smooth variation in delay between the low and the high frequencies. In addition, there was a 2.5 msec delay that did not vary with frequency arising from the processes of analog-digital-analog conversion and equalization. Ten subjects with symmetric, bilateral, moderate hearing impairment of cochlear origin were fitted binaurally with Phonak PicoNet2 BTE linear hearing aids. The aids were programmed to provide insertion gains closely approximating the NAL (RP) (Byrne & Dillon, 1986) prescription for each subject and ear. The output of an omnidirectional microphone, attached on top of each aid, fed a digital filtering unit that introduced the across-frequency delay. Audio shoes reintroduced the signal back into the BTE aids. Subjects were tested in a counterbalanced order for their identification of vowel-consonant-vowel (VCV) syllables. In a second session, the same subjects spoke from a script: their utterances were recorded on tape, speech production rates were measured and subjective ratings of the disturbance of the delay were obtained. Subjects required some training to recognize the effects of the delay and rate them consistently. RESULTS: Subjective disturbance increased progressively with increasing across-frequency delay. A 9-msec delay was significantly more disturbing than a 0-msec delay. A delay of about 20 msec led to a mean rating of "disturbing." VCV identification decreased significantly once the across-frequency delay was 15 msec or greater. However, word production rates were not significantly affected by across-frequency delay over the range tested. CONCLUSIONS: Relatively small across-frequency delays (9 to 15 msec) give rise to significant changes in speech identification and subjective disturbance. A delay of 9 to 15 msec, constant across frequency, would have a smaller effect (Stone & Moore, 1999). It appears desirable to compensate, at least in part, for across-frequency delays introduced by any processing.

Aged↗

PubMed2003

Perception of the low pitch of frequency-shifted complexes.

When all of the components in a harmonic complex tone are shifted in frequency by delta f, the pitch of the complex shifts roughly in proportion to delta 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 (FOs) 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 stimulus had a passband centered above that of the test stimulus, to avoid common partials. For resolved and intermediate conditions, pitch shifts were observed that could generally be predicted from the frequencies of the partials. The shifts were unaffected by addition of noise to mask combination tones. For the unresolved condition, no pitch shift was observed, which suggests that pitch is not based on temporal fine structure for stimuli containing only high unresolved harmonics. Experiment 2 used three-component complexes resembling those of Schouten [J. Acoust. Soc. Am. 34, 1418-1424 (1962)]. Nominal harmonic numbers were 3, 4, 5 (resolved), 8, 9, 10 (intermediate), or 13, 14, 15 (unresolved) and F0s were 50, 100, 200, or 400 Hz. Clear shifts in the matches were found for all conditions, including unresolved. For the latter, subjects may have matched the "center of gravity" of the excitation patterns of the test and matching stimuli.

Adult↗

PubMed2003

Testing the concept of a modulation filter bank: the audibility of component modulation and detection of phase change in three-component modulators.

Two experiments were performed to test the concept that the auditory system contains a "modulation filter bank" (MFB). Experiment 1 examined the ability to "hear out" the modulation frequency of the central component of a three-component modulator applied to a 4-kHz sinusoidal carrier. On each trial, three modulated stimuli were presented. The modulator of the first stimulus contained three components. Within a run the frequencies of the outer two components were fixed and the frequency of the central ("target") component was drawn randomly from one of five values. The modulators of second and third stimuli contained one component. One had a frequency equal to that of the target and the other had a frequency randomly selected from one of the other possible values. Subjects indicated whether the target corresponded to the second or third stimulus. Scores were around 80% correct when the components in the three-component modulator were widely spaced and when the frequencies of the target and comparison differed sufficiently. Experiment 2 examined the ability to hear a change in the relative phase of the components in a three-component modulator with harmonically spaced components, using a 31FC task. The frequency of the central component, f(c), was either 50 or 100 Hz. Scores were 80%-90% correct when the component spacing was < or = 0.5 f(c), but decreased markedly for greater spacings. Performance was only slightly impaired by randomizing the overall modulation depth from one stimulus to the next. The results of both experiments are broadly consistent with what would be expected from a MFB with a Q value of 1 or slightly less.

Auditory Perception↗

PubMed2003

Perceived naturalness of spectrally distorted speech and music.

We determined how the perceived naturalness of music and speech (male and female talkers) signals was affected by various forms of linear filtering, some of which were intended to mimic the spectral "distortions" introduced by transducers such as microphones, loudspeakers, and earphones. The filters introduced spectral tilts and ripples of various types, variations in upper and lower cutoff frequency, and combinations of these. All of the differently filtered signals (168 conditions) were intermixed in random order within one block of trials. Levels were adjusted to give approximately equal loudness in all conditions. Listeners were required to judge the perceptual quality (naturalness) of the filtered signals on a scale from 1 to 10. For spectral ripples, perceived quality decreased with increasing ripple density up to 0.2 ripple/ERB(N) and with increasing ripple depth. Spectral tilts also degraded quality, and the effects were similar for positive and negative tilts. Ripples and/or tilts degraded quality more when they extended over a wide frequency range (87-6981 Hz) than when they extended over subranges. Low- and mid-frequency ranges were roughly equally important for music, but the mid-range was most important for speech. For music, the highest quality was obtained for the broadband signal (55-16,854 Hz). Increasing the lower cutoff frequency from 55 Hz resulted in a clear degradation of quality. There was also a distinct degradation as the upper cutoff frequency was decreased from 16,845 Hz. For speech, there was a marked degradation when the lower cutoff frequency was increased from 123 to 208 Hz and when the upper cutoff frequency was decreased from 10,869 Hz. Typical telephone bandwidth (313 to 3547 Hz) gave very poor quality.

Adolescent↗

PubMed2003

Asymmetry of masking between complex tones and noise: partial loudness.

This experiment examined the partial masking of periodic complex tones by a background of noise, and vice versa. The tones had a fundamental frequency (F0) of 62.5 or 250 Hz, and components were added in either cosine phase (CPH) or random phase (RPH). The tones and the noise were bandpass filtered into the same frequency region, from the tenth harmonic up to 5 kHz. The target alone was alternated with the target and the background; for the mixture, the background and target were either gated together, or the background was turned on 400 ms before, and off 200 ms after, the target. Subjects had to adjust the level of either the target alone or the target in the background so as to match the loudness of the target in the two intervals. The overall level of the background was 50 dB SPL, and loudness matches were obtained for several fixed levels of the target alone or in the background. The resulting loudness-matching functions showed clear asymmetry of partial masking. For a given target-to-background ratio, the partial loudness of a complex tone in a noise background was lower than the partial loudness of a noise in a complex tone background. Expressed as the target-to-background ratio required to achieve a given loudness, the asymmetry typically amounted to 12-16 dB. When the F0 of the complex tone was 62.5 Hz, the asymmetry of partial masking was greater for CPH than for RPH. When the F0 was 250 Hz, the asymmetry was greater for RPH than for CPH. Masked thresholds showed the same pattern as for partial masking for both F0's. Onset asynchrony had some effect on the loudness matching data when the target was just above its masked threshold, but did not significantly affect the level at which the target in the background reached its unmasked loudness. The results are interpreted in terms of the temporal structure of the stimuli.

Adult↗

PubMed2003

Effect of the speed of a single-channel dynamic range compressor on intelligibility in a competing speech task.

Using a "noise-vocoder" cochlear implant simulator [Shannon et al., Science 270, 303-304 (1995)], the effect of the speed of dynamic range compression on speech intelligibility was assessed, using normal-hearing subjects. The target speech had a level 5 dB above that of the competing speech. Initially, baseline performance was measured with no compression active, using between 4 and 16 processing channels. Then, performance was measured using a fast-acting compressor and a slow-acting compressor, each operating prior to the vocoder simulation. The fast system produced significant gain variation over syllabic timescales. The slow system produced significant gain variation only over the timescale of sentences. With no compression active, about six channels were necessary to achieve 50% correct identification of words in sentences. Sixteen channels produced near-maximum performance. Slow-acting compression produced no significant degradation relative to the baseline. However, fast-acting compression consistently reduced performance relative to that for the baseline, over a wide range of performance levels. It is suggested that fast-acting compression degrades performance for two reasons: (1) because it introduces correlated fluctuations in amplitude in different frequency bands, which tends to produce perceptual fusion of the target and background sounds and (2) because it reduces amplitude modulation depth and intensity contrasts.

Adolescent↗

PubMed2003

Louder sounds can produce less forward masking: effects of component phase in complex tones.

The influence of the degree of envelope modulation and periodicity on the loudness and effectiveness of sounds as forward maskers was investigated. In the first experiment, listeners matched the loudness of complex tones and noise. The tones had a fundamental frequency (F0) of 62.5 or 250 Hz and were filtered into a frequency range from the 10th harmonic to 5000 Hz. The Gaussian noise was filtered in the same way. The components of the complex tones were added either in cosine phase (CPH), giving a large crest factor, or in random phase (RPH), giving a smaller crest factor. For each F0, subjects matched the loudness between all possible stimulus pairs. Six different levels of the fixed stimulus were used, ranging from about 30 dB SPL to about 80 dB SPL in 10-dB steps. Results showed that, at a given overall level, the CPH and the RPH tones were louder than the noise, and that the CPH tone was louder than the RPH tone. The difference in loudness was larger at medium than at low levels and was only slightly reduced by the addition of a noise intended to mask combination tones. The differences in loudness were slightly smaller for the higher than for the lower F0. In the second experiment, the stimuli with the lower F0s were used as forward maskers of a 20-ms sinusoid, presented at various frequencies within the spectral range of the maskers. Results showed that the CPH tone was the least effective forward masker, even though it was the loudest. The differences in effectiveness as forward maskers depended on masker level and signal frequency; in order to produce equal masking, the level of the CPH tone had to be up to 35 dB above that of the RPH tone and the noise. The implications of these results for models of loudness are discussed and a model is presented based on neural activity patterns in the auditory nerve; this predicts the general pattern of loudness matches. It is suggested that the effects observed in the experiments may have been influenced by two factors: cochlear compression and suppression.

Adult↗

PubMed2003

Evaluation of the noise reduction system in a commercial digital hearing aid.

We evaluated the effectiveness of a noise reduction system implemented in a commercial digital multichannel compression hearing aid. Eight experienced hearing aid wearers with moderate sensorineural hearing loss were fitted bilaterally according to the manufacturer's fitting guidelines. After a 3-month period of regular use of two programs, one with and one without the noise reduction system, speech recognition thresholds (SRTs) were measured in four types of background noise, including steady noise, and noises with spectral and/or temporal dips. SRTs were very similar with and without the noise reduction system; in both cases, SRTs were markedly lower than for unaided listening. SRTs were lower for the noises with dips than for the steady noise, especially for the aided conditions, indicating that amplification can help to 'listen in the dips'. Ratings of sound quality and listening comfort in the aided conditions were uniformly high and very similar with and without the noise reduction system.

Adult↗

PubMed2003

Use of a loudness model for hearing aid fitting. V. On-line gain control in a digital hearing aid.

Many researchers have proposed that hearing aids should process sounds so as to restore loudness perception to 'normal'. We describe how a model for predicting loudness for people with cochlear hearing loss can be implemented in a digital hearing aid so as to calculate the frequency-dependent gains that would be required to achieve that goal. It is assumed that the input signal is processed using brief segments or 'frames'. For each frame, the spectrum is calculated, usually via a fast Fourier transform (FFT). From the spectrum, an excitation pattern is calculated for a normal car and for the impaired ear of the patient. The loudness model is then used to calculate the gain required at the centre frequency of each channel in the aid, so as to match the specific loudness in the normal and impaired ears. The whole process is repeated for each successive frame, with overlap of frames and with smoothing of the gain changes across frames. We describe both an 'exact' model, which prescribes a 'curvilinear' compression characteristic at each frequency, and an approximation using 'straight' compression, which is computationally less intensive. Limitations of the present approach are described, and the approach is compared with more traditional approaches using multichannel compression, and with previous approaches using loudness models for fitting hearing aids.

Acoustic Stimulation↗

PubMed2002

Interference effects and phase sensitivity in hearing.

This paper reviews interference effects in the auditory system, particularly effects occurring in the outer ear and the inner ear (cochlea). Sounds enter the ear canal both directly and after reflections from the pinna. This results in complex spectral patterns, which vary systematically with the direction of incidence of the sound source relative to the head. Evidence is described indicating that these spectral patterns are used in the localization of sounds in space. The cochlea behaves like a limited-resolution frequency analyser. When the components of a complex sound are closely spaced in frequency, they can interfere on the basilar membrane (BM) within the cochlea. Interference effects on the BM are complex, as they are influenced by a physiologically active mechanism which introduces strong nonlinearities, including level-dependent amplification. Interference effects on the BM play a role in many aspects of auditory perception, including the perception of consonance and dissonance, the perception of pitch, the perception of changes in phase, and the perception of timbre. Interference effects in the cochlea may also play a role in producing the spectral regularity observed in sounds reflected from the ear (otoacoustic emissions).

Animals↗

PubMed2002

Behavioural measurement of level-dependent shifts in the vibration pattern on the basilar membrane.

Physiological data suggest that the travelling wave on the basilar membrane evoked by a sinusoid of fixed frequency moves towards the base with increasing level. We describe two psychoacoustic experiments that attempted to provide evidence for and quantify the extent of such a shift in humans. In experiment 1, masking patterns were measured in forward masking using a fixed 6-kHz tone presented at 65 or 85 dB sound pressure level. The threshold for detecting a brief sinusoidal signal was measured as a function of signal frequency for several time delays of the signal relative to the end of the masker. A background noise was included to reduce 'off-frequency listening'. As the signal delay was increased, the signal level at the peaks of the masking patterns decreased and the signal frequency at the peak of the patterns moved progressively towards higher frequencies. The pattern of results was consistent with the idea of a basalward shift of the travelling wave with increasing level. The estimated shift corresponds to about 0.25 octaves for a 40-dB change in level. Experiment 2 also used forward masking. The signal was a 4-kHz tone presented at 10 dB sensation level. For three fixed masker levels (65, 85 and 95 dB), we measured the duration of the gap between the masker and signal required to give 79.4% correct detection of the signal (called the 'gap threshold') as a function of masker frequency; the longer the gap threshold, the more effective is the masker. The gap threshold patterns sometimes showed two peaks. One occurred just below the signal frequency and the frequency at the peak was hardly affected by masker level. The second peak fell at a lower frequency, and this frequency tended to decrease with increasing masker level. The gap threshold patterns tended to spread markedly towards lower frequencies with increasing masker level. The shift with level provides further evidence for a basalward spread of the travelling wave with increasing level.

Acoustic Stimulation↗

PubMed2002

The relative role of beats and combination tones in determining the shapes of masking patterns: II. Hearing-impaired listeners.

Masking patterns were measured for hearing-impaired subjects with varying degrees of hearing loss. In one set of conditions, three subjects were tested using narrowband noise ('noise') and sinusoidal ('tone') maskers and narrowband noise signals. The maskers had centre frequencies of 0.25, 0.5, 1.0 and 4.0 kHz and levels of 60, 80 and 100 dB SPL. Masking patterns for both the noise and tone maskers showed irregularities ('dips'), especially for signal frequencies up to 500 Hz above the masker frequency. The irregularities occurred for all masker levels and for all subjects for at least one masker frequency and they occurred for a relatively constant range of masker-signal frequency separations, suggesting that they were the result of beat detection. In another set of conditions, masking patterns were measured using two subjects, for a 2.0-kHz tone masker with a level of 100 dB SPL and tone and noise signals. For the tone masker alone (baseline condition), the masking patterns again exhibited prominent dips above, and sometimes below, the masker frequency. The addition of a lowpass noise to the masker, intended to mask combination tones, had little effect for one subject. For the other subject, who had near-normal absolute thresholds at low frequencies, the noise elevated thresholds for masker-signal frequency separations between 500 and 1500 Hz. For this subject, an extra tone with a frequency equal to the masker-signal frequency separation, added in place of the lowpass noise, had a very similar effect to that produced by the lowpass noise, suggesting that he was detecting a simple difference tone in the baseline condition. The addition of a pair of high-frequency tones (MDI tones - intended to reduce the detectability of beats) to the masker elevated thresholds for signal frequencies from 1500 to 2500 Hz for one subject and from 1500 to 3500 Hz for another subject. The addition of lowpass noise and MDI tones to the masker produced masking patterns very similar to those observed when the MDI tones alone were added to the masker. Overall, the results suggest that the irregularities in the masking patterns were caused mainly by the detection of beats and not by the detection of combination tones.

Acoustic Stimulation↗

PubMed2002

Psychoacoustics of normal and impaired hearing.

Recent developments in the field of psychoacoustics are presented, focusing on areas which have application in the diagnosis and understanding of impaired hearing. Cochlear hearing loss often results in a loss of the compressive non-linearity that operates in normal ears; this loss is probably the main cause of loudness recruitment. Forward masking can be used as a tool to assess the strength of cochlear compression in human listeners. Hearing impairment can sometimes be associated with complete loss of function of inner hair cells over a certain region of the cochlea, resulting in a 'dead region'. Two psychoacoustic methods for detecting dead regions and defining their limits are described. The implications of the results for fitting hearing aids are discussed. Finally, the effect of cochlear hearing loss on the perception of rapid sequences of sounds (stream segregation) is described.

Auditory Perception↗

PubMed2002

Tolerable hearing aid delays. II. Estimation of limits imposed during speech production.

OBJECTIVE: We used real-time processing in a wearable digital hearing aid to examine the effect of processing delay on normal-hearing participants while speaking. Objective and subjective data were recorded so as to permit analysis of both the production and perception of speech read aloud from a script. We also asked participants to rate the disturbance of the echo introduced by the delay. DESIGN: Thirty-two (16M, 16F) participants were fitted binaurally with behind-the-ear (BTE) aids connected to a digital processor. A 4 mm Libby horn, surrounded by an expanding foam earplug, conducted processed sound into each ear canal. The processor provided either linear processing or three-channel, fast-acting wide dynamic range compression, independently to each ear. Insertion gains were set, using a KEMAR manikin, to be 0 dB over a wide frequency range, for frontally presented speech with a free field level of 65 dB SPL. Additionally, the aids introduced one of four selectable delays (7 to 43 msec) between the BTE microphone and receiver. After a short period of acclimatization, each participant read 16 prose passages of about 500 words in length in each of two similar-sized rooms with markedly different acoustics: reverberant and nonreverberant. For each passage, a subjective rating of the level of disturbance of the perceived echo was recorded, as well as simultaneous recordings from a microphone and a Laryngograph, which directly records glottal pulses. RESULTS: Disturbance ratings generally increased monotonically with increasing delay. Averaged results show that a delay between 25 and 30 msec is rated as "disturbing." Measures were also taken of word production rate, speech level and range of level as well as fundamental frequency and range of fundamental frequency. For these measures of speech production, there was no significant effect until the delay exceeded 30 msec. There was little effect of acoustic environment or aid processing (linear or compression). CONCLUSIONS: The acceptability of delays introduced by digital hearing aids is primarily determined by aspects of the perception of self-generated speech. Speech production, on average, is hardly affected unless the processing delay exceeds 30 msec. The permissible limit of 20 to 30 msec is smaller than the delays at which audio-visual integration is disrupted.

Adolescent↗