Biomedical subjects

B C Moore

Publications and source records attributed to B C Moore.

At least 55 records · Page 3Linked to original sources

PubMed1999

Effects of frequency and duration on psychometric functions for detection of increments and decrements in sinusoids in noise.

Psychometric functions for detecting increments or decrements in level of sinusoidal pedestals were measured for increment and decrement durations of 5, 10, 20, 50, 100, and 200 ms and for frequencies of 250, 1000, and 4000 Hz. The sinusoids were presented in background noise intended to mask spectral splatter. A three-interval, three-alternative procedure was used. The results indicated that, for increments, the detectability index d' was approximately proportional to delta I/I. For decrements, d' was approximately proportional to delta L. The slopes of the psychometric functions increased (indicating better performance) with increasing frequency for both increments and decrements. For increments, the slopes increased with increasing increment duration up to 200 ms at 250 and 1000 Hz, but at 4000 Hz they increased only up to 50 ms. For decrements, the slopes increased for durations up to 50 ms, and then remained roughly constant, for all frequencies. For a center frequency of 250 Hz, the slopes of the psychometric functions for increment detection increased with duration more rapidly than predicted by a "multiple-looks" hypothesis, i.e., more rapidly than the square root of duration, for durations up to 50 ms. For center frequencies of 1000 and 4000 Hz, the slopes increased less rapidly than predicted by a multiple-looks hypothesis, for durations greater than about 20 ms. The slopes of the psychometric functions for decrement detection increased with decrement duration at a rate slightly greater than the square root of duration, for durations up to 50 ms, at all three frequencies. For greater durations, the increase in slope was less than proportional to the square root of duration. The results were analyzed using a model incorporating a simulated auditory filter, a compressive nonlinearity, a sliding temporal integrator, and a decision device based on a template mechanism. The model took into account the effects of both the external noise and an assumed internal noise. The model was able to account for the major features of the data for both increment and decrement detection.

Adult↗

PubMed1999

Comparison of different forms of compression using wearable digital hearing aids.

Four different compression algorithms were implemented in wearable digital hearing aids: (1) The slow-acting dual-front-end automatic gain control (AGC) system [B. C. J. Moore, B. R. Glasberg, and M. A. Stone, Br. J. Audiol. 25, 171-182 (1991)], combined with appropriate frequency response equalization, with a compression threshold of 63 dB sound pressure level (SPL) and with a compression ratio of 30 (DUAL-HI); (2) The dual-front-end AGC system combined with appropriate frequency response equalization, with a compression threshold of 55 dB SPL and with a compression ratio of 3 (DUAL-LO). This was intended to give some impression of the levels of sounds in the environment; (3) Fast-acting full dynamic range compression in four channels (FULL-4). The compression was designed to minimize envelope distortion due to overshoots and undershoots; (4) A combination of (2) and (3) above, where each applied less compression than when used alone (DUAL-4). Initial fitting was partly based on the concept of giving a flat specific-loudness pattern for a 65-dB SPL speech-shaped noise input, and this was followed by fine tuning using an adaptive procedure with speech stimuli. Eight subjects with moderate to severe cochlear hearing loss were tested in a counter-balanced design. Subjects had at least 2 weeks experience with each system in everyday life before evaluation using the Abbreviated Profile of Hearing Aid Benefit (APHAB) test and measures of speech intelligibility in quiet (AB word lists at 50 and 80 dB SPL) and noise (adoptive sentence lists in speech-shaped noise, or that same noise amplitude modulated with the envelope of speech from a single talker). The APHAB scores did not indicate clear differences between the four systems. Scores for the AB words in quiet were high for all four systems at both 50 and 80 dB SPL. The speech-to-noise ratios required for 50% intelligibility were low (indicating good performance) and similar for all the systems, but there was a slight trend for better performance in modulated noise with the DUAL-4 system than with the other systems. A subsequent trial where three subjects directly compared each of the four systems in their everyday lives indicated a slight preference for the DUAL-LO system. Overall, the results suggest that it is not necessary to compress fast modulations of the input signal.

Acoustic Stimulation↗

PubMed1999

Use of a loudness model for hearing aid fitting: II. Hearing aids with multi-channel compression.

A model for predicting loudness for people with cochlear hearing loss was applied to the problem of the initial fitting of a multi-channel compression hearing aid. The fitting was based on two constraints: (1) The specific loudness pattern evoked by speech of a moderate level (65 dB SPL) should be reasonably flat (equal loudness per critical band), and the overall loudness should be similar to that evoked in a normal listener by 65-dB speech (about 23 sones for binaural listening); (2) Speech with an overall level of 45 dB SPL should just be audible in all frequency bands from 500 Hz up to about 4 kHz, provided that this does not require compression ratios exceeding about 3. These two constraints were used to determine initial values for the gain, compression ratio and compression threshold in each channel of a multi-channel compression system. This initial fitting was based entirely on audiometric thresholds; it does not require suprathreshold loudness measures. The fitting method was evaluated using an experimental fast-acting four-channel compression system. The initial fitting was followed by an adaptive procedure to 'fine tune' the fitting, and the aids were then used in everyday life. Performance was evaluated by use of questionnaires and by measures of speech intelligibility. Although the fine tuning resulted in modest changes in the fitting parameters for some subjects, on average the frequency response shapes and compression ratios were similar before and after the fine tuning. The fittings led to satisfactory loudness impressions in everyday life and to high speech intelligibility over a wide range of levels. It was concluded that the initial fitting method gives reasonable starting values for the fine tuning.

Acoustic Stimulation↗

PubMed1999

Use of a loudness model for hearing aid fitting: III. A general method for deriving initial fittings for hearing aids with multi-channel compression.

A model for predicting loudness for people with cochlear hearing loss is applied to the problem of the initial fitting of multi-channel fast-acting compression hearing aids. The fitting is based entirely on the pure tone audiogram, and does not require measures of loudness growth. One constraint is always applied: the specific loudness pattern evoked by speech of a moderate level (65 dB SPL) should be reasonably flat (equal loudness per critical band), and the overall loudness should be similar to that evoked in a normal listener by 65-dB speech. This is achieved using the 'Cambridge' formula. For hearing aids where the compression threshold in each channel can be set to a very low value, an additional constraint is used: speech with an overall level of 45 dB SPL should be audible over its entire dynamic range in all frequency channels from 500 Hz up to about 4 kHz. For hearing aids where the compression thresholds cannot be set to very low values, a different additional constraint is used: the specific loudness pattern evoked by speech of a high level (85 dB SPL, and with the spectral characteristics of shouted speech) should be reasonably flat, and the overall loudness should be similar to that evoked in a normal listener by 85-dB speech. For both cases, compression ratios are limited to values below 3. For each of these two cases, we show how to derive compression ratios and gains, and for the first case, compression thresholds, for each channel. The derivations apply to systems with any number of channels. A computer program implementing the derivations is described. The program also calculates target insertion gains at the centre frequency of each channel for input levels of 50, 65 and 80 dB SPL, and target gains at the eardrum measured relative to the level at the reference microphone of a probe microphone system.

Auditory Threshold↗

PubMed1998

Psychoacoustic consequences of compression in the peripheral auditory system.

Input-output functions on the basilar membrane of the cochlea show a strong compressive nonlinearity at midrange levels for frequencies close to the characteristic frequency of a given place. This article shows how many different phenomena can be explained as consequences of this nonlinearity, including the "excess" masking produced when 2 nonsimultaneous maskers are combined, the nonlinear growth of forward masking with masker level, the influence of component phase on the effectiveness of complex forward maskers, changes in the ability to detect increments and decrements with level, temporal integration, and the influence of component phase and level on the perception of vowellike sounds. Cochlear hearing loss causes basilar-membrane responses to become more linear. This can account for loudness recruitment, linear additivity of nonsimultaneous masking, linear growth of forward masking, reduced temporal resolution for sounds with fluctuating envelopes, and reduced temporal integration.

Auditory Perception↗

PubMed1998

Evaluation of the effect of speech-rate slowing on speech intelligibility in noise using a simulation of cochlear hearing loss.

The effect of digital processing, which slows the speed of speech (speech-rate) without changing its pitch, has been examined. The processing is intended to make speech communication easier by allowing more time for cognitive processing when the listening situation is difficult, for example, when listening to a foreign language, or when the user has a hearing loss. The speech-rate slowing makes use of a pitch-synchronous partial expansion of the waveform in the time domain. The processing was evaluated using a simulation of hearing loss which has been shown to lead to reduced intelligibility for normally hearing subjects. The simulation included the major consequences of cochlear hearing loss; loudness recruitment, threshold elevation, and reduced frequency selectivity. Two simulations were used: a moderate flat hearing loss with auditory filters broadened by a constant factor of three (B3R2); and the same loss with linear amplification applied prior to the simulation processing (B3R2+). Two expansion rates were used for the speech-rate slowing, 1.25 and 1.50. The intelligibility of sentences in speech-shaped noise was measured. For both simulation conditions, the speech-rate slowing did not give any improvement in intelligibility. Rather, in condition B3R2+ the slowing produced statistically significant deleterious effects on intelligibility. The results suggest that artificial speech-rate slowing will not improve the intelligibility of speech in noise for hearing-impaired people who have the type of cochlear damage simulated in this test.

Adult↗

PubMed1998

Speech reception thresholds in noise with and without spectral and temporal dips for hearing-impaired and normally hearing people.

People with cochlear hearing loss often have considerable difficulty in understanding speech in the presence of background sounds. In this paper the relative importance of spectral and temporal dips in the background sounds is quantified by varying the degree to which they contain such dips. Speech reception thresholds in a 65-dB SPL noise were measured for four groups of subjects: (a) young with normal hearing; (b) elderly with near-normal hearing; (c) young with moderate to severe cochlear hearing loss; and (d) elderly with moderate to severe cochlear hearing loss. The results indicate that both spectral and temporal dips are important. In a background that contained both spectral and temporal dips, groups (c) and (d) performed much more poorly than group (a). The signal-to-background ratio required for 50% intelligibility was about 19 dB higher for group (d) than for group (a). Young hearing-impaired subjects showed a slightly smaller deficit, but still a substantial one. Linear amplification combined with appropriate frequency-response shaping (NAL amplification), as would be provided by a well-fitted "conventional" hearing aid, only partially compensated for these deficits. For example, group (d) still required a speech-to-background ratio that was 15 dB higher than for group (a). Calculations of the articulation index indicated that NAL amplification did not restore audibility of the whole of the speech spectrum when the speech-to-background ratio was low. For unamplified stimuli, the SRTs in background sounds were highly correlated with absolute thresholds, but not with age. For stimuli with NAL amplification, the correlations of SRTs with absolute thresholds were lower, but SRTs in backgrounds with spectral and/or temporal dips were significantly correlated with age. It is proposed that noise with spectral and temporal dips may be especially useful in evaluating possible benefits of multi-channel compression.

Adult↗

PubMed1998

Discrimination of frequency glides with superimposed random glides in level.

These experiments were designed to test the hypothesis that glides in frequency are detected and discriminated by monitoring changes in excitation level on the low-frequency side of the excitation pattern. Thresholds were measured for detecting an increase in the extent of a frequency glide, for various standard extents (transition spans). The center frequency of each stimulus was roved, to prevent subjects from using the start or endpoint frequencies of the stimuli as cues. The level was either fixed at 70 dB SPL, or changed linearly in dB/s by an amount that varied randomly in extent and direction, keeping the level at the midpoint of the glide at 70 dB SPL. These random changes in level were intended to disrupt cues based on monitoring changes in excitation level on one side of the excitation pattern. For some conditions, performance was too good to be explained by subjects monitoring the start or endpoint frequencies of the stimuli. Performance was also too good to be explained in terms of the discrimination of changes in excitation level on one side of the excitation pattern. Thresholds, expressed as a proportion of the equivalent rectangular bandwidth (ERB) of the auditory filter, did not vary greatly with center frequency (0.5, 2, or 6 kHz), suggesting that discrimination did not depend strongly on information derived from phase locking. Glide duration (50 or 400 ms) and glide direction (upward or downward) also had little effect. Thresholds increased with increasing standard transition span, when that span was increased beyond 0.5 ERB. It is concluded that changes in glide extent per se can be discriminated, but this is not done by monitoring just one side of the excitation pattern.

Analysis of Variance↗

PubMed1998

Masking patterns for sinusoidal and narrow-band noise maskers.

The masking patterns produced by narrow-band maskers can show distinct irregularities. These experiments attempted to clarify the relative importance of factors contributing to these irregularities. A three-alternative adaptive forced-choice method with feedback was used, to promote use of the optimal detection cues. The masker and signal were either a sinusoid or a band of noise that was 80 Hz wide, giving four possible combinations of masker and signal type. In experiment 1, masking patterns were measured for maskers centered at 1 kHz, for all combinations of masker and signal type (tone or noise). The masking patterns showed irregularities (dips or "shoulders") above the masker frequency, and the irregularities were larger for the sinusoidal than for the noise masker. Experiment 2 was similar to experiment 1, except that low-pass noise was added to mask combination products. For the noise masker, the low-pass noise slightly increased thresholds, and largely eliminated the irregularities in the patterns, but for the tone masker, the irregularities persisted. Experiment 3 used a noise signal with tone and noise maskers centered at 250, 1000, and 4000 Hz. The tone masker produced less masking than the noise masker for masker-signal frequency separations of 150-250 Hz, regardless of masker frequency. Experiment 4 used an additional masker tone to introduce beats similar to those produced by the interaction of the signal and (main) masker, and to mask combination products. This largely eliminated the dips in the masking patterns for both the noise and tone maskers. Experiment 5 used an additional pair of high-frequency tones to introduce beats, with similar results. We conclude that temporal fluctuations (beats) have a strong influence on the masking patterns for sinusoidal maskers, for masker-signal frequency separations up to a few hundred Hz. Beats may also have some influence on the masking patterns for noise maskers. The detection of combination products also plays a role.

Auditory Perception↗

PubMed1998

The role of excitation-pattern cues and temporal cues in the frequency and modulation-rate discrimination of amplitude-modulated tones.

These experiments examine the influence of excitation-pattern cues and temporal-fine-structure cues on frequency difference limens (FDLs) measured as a function of duration. In the first three conditions, listeners were required to detect a change in carrier frequency from a baseline of 250 or 2000 Hz, for stimuli with half-amplitude durations ranging from 5 to 320 ms. In the "steady" condition, duration was manipulated by increasing the steady-state portion of the envelope between two 5-ms linear onset and offset ramps. This resulted in spectra and excitation patterns that broadened with decreasing duration. In the "modulated" condition, the carrier was amplitude modulated with a triangle function (period 10 ms) and duration was manipulated by varying the number of cycles of the modulator. In this case, the spectral envelope did not vary with duration, but the width of individual spectral lobes broadened with decreasing duration. The "low-peak-constant" condition was similar to the modulated condition, except that an increase in carrier frequency was accompanied by a decrease in the period of the modulator, so as to hold constant the frequency of the spectral lobe located roughly 100 Hz below the carrier frequency. In this condition, changes in carrier frequency resulted in minimal changes in excitation level on the low-frequency side of the excitation pattern, but changes on the high-frequency side were larger than for the first two conditions. Both the values of the FDLs, and their variation with frequency and duration, were similar in these three conditions. The fourth, "modulator varying," condition differed in that the carrier frequency was held constant and listeners were required to detect a change in modulator frequency. Thresholds were expressed as the change in frequency of the spectral side lobes adjacent to the carrier frequency ("equivalent" FDLs). Excitation-pattern cues in this condition were at least as large as in the first three conditions. However, equivalent FDLs were significantly higher than for the other three conditions. The higher FDLs are attributed to the lack of temporal fine-structure cues related to the carrier frequency. Overall, the results suggest that FDLs in the first three conditions were determined by temporal cues rather than by excitation-pattern cues. The increase of the FDLs with decreasing duration did not arise from increasing spectral splatter.

Auditory Perception↗

PubMed1998

Development and evaluation of a procedure for fitting multi-channel compression hearing aids.

Hearing aids with multi-channel compression are often fitted on the basis of loudness scaling data obtained using narrow bands of noise or tones. Here, we report the development and evaluation of an alternative fitting procedure based on the use of speech signals. The parameters of the hearing aid (the gains in each channel for high and low input levels) are adjusted adaptively under computer control on the basis of the listener's responses. The goal is that speech at 85 dB SPL should be judged as 'loud', speech at 60 dB SPL should be judged as 'quiet', and speech at both levels should be judged as 'neither tinny nor boomy'. The procedure was evaluated using a two-channel compression hearing aid, the remote control of which allowed two programs to be stored. One program was based on our fitting procedure. The other was either based on the manufacturer's recommended full fitting procedure (which included loudness scaling with bands of noise), or was based on the audiogram alone, using the manufacturer's algorithm. After an acclimatization period of at least two weeks, subjects were then asked to fill in a questionnaire about their experiences with the two programs in different listening situations. The results generally indicated a preference for the program based on our adaptive fitting procedure. We also conducted laboratory measurements of speech intelligibility, in quiet and in a background of a single competing talker. These showed no clear difference between programs, although scores overall were very high. We conclude that our adaptive procedure gives very satisfactory results in everyday life. Parameter values giving good comfort also give good intelligibility. The procedure typically takes between five and 10 minutes per ear, which is quicker than most loudness scaling procedures.

Hearing Aids↗

PubMed1998

Use of a loudness model for hearing-aid fitting. I. Linear hearing aids.

A model for predicting loudness for people with cochlear hearing loss is applied to the problem of prescribing the frequency-gain characteristic of a linear hearing aid. It is argued that a reasonable goal is to make all frequency bands of speech equally loud while achieving a comfortable overall loudness; this can maximize the proportion of the speech spectrum that is above the absolute threshold for a given loudness. In terms of the model this means that the specific loudness pattern evoked by speech of a moderate level (65 dB SPL) should be reasonably flat (equal loudness per critical band), and the overall loudness should be similar to that evoked in a normal listener by 65 dB speech (about 23 sones). The model is used to develop a new formula - the 'Cambridge formula' - for prescribing insertion gain from audiometric thresholds. It is shown that, for a fixed overall loudness of 23 sones, the Cambridge formula leads to a higher calculated articulation index than three other commonly used prescriptive methods: NAL(R), FIG6 and DSL.

Auditory Threshold↗

PubMed1997

Dependence of frequency modulation detection on frequency modulation coherence across carriers: effects of modulation rate, harmonicity, and roving of the carrier frequencies.

Furukawa and Moore [S. Furukawa and B. C. J. Moore, J. Acoust. Soc. Am. 100, 2299-2312 (1996)] found that the detection of frequency modulation (FM) imposed on two inharmonically related carriers was better when the FM was coherent across carriers than when it was incoherent. Here, "coherence" refers to whether the pattern of frequency change over time was identical or different across carriers. The present paper was designed to explore three possible mechanisms underlying this effect. Thresholds were measured for the detection of a single cycle of sinusoidal FM imposed on two sinusoidal carriers. The FM of each carrier was equally detectable, as determined in preliminary experiments. A continuous pink-noise background was used to mask the outputs of auditory filters tuned between the two carrier frequencies. The modulation rate was either 2.5, 5, or 10 Hz. Three combinations of carrier frequencies were used, varying in the extent to which the carriers were harmonically related (1050 and 2069 Hz; 1100 and 2000 Hz; and 1100 and 1925 Hz). The carrier frequencies were either fixed at these values, or were randomly varied (roved) from one trial to another (+/-10%) keeping the frequency ratio constant. Performance for coherent FM was generally better than for incoherent FM. The effect of FM coherence was greater at the lowest modulation rate and was slightly greater when the carrier frequencies were fixed throughout a block of trials than when they were roved. For the two lowest modulation rates, the effect of FM coherence was greater for carriers that were (nearly) harmonically related. It is proposed that sensitivity to FM coherence depends partly on comparing patterns of phase locking to the carriers; this is done most effectively at low modulation rates. However, two other factors may play a small role. These are: sensitivity to the coherence of amplitude modulation induced in the auditory system by the FM (which is somewhat disrupted by roving the carrier frequencies); and sensitivity to fluctuations in the residue pitch evoked by the two carriers (the residue pitch being less salient for inharmonically related carriers.

Auditory Perception↗

PubMed1997

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.

Adult↗

PubMed1997

The role of spread excitation and suppression in simultaneous masking.

This experiment was intended to clarify the relative role of spread of excitation and suppression in simultaneous masking, for masker frequencies just below and well below the signal frequency. The experiment had two stages. In stage 1, growth-of-masking functions were measured in simultaneous masking for a 2200-Hz sinusoidal signal and a sinusoidal masker with frequency of either 1800 Hz or 500 Hz. Straight lines fitted to these data were used to determine masker levels that would give 10, 20, and 30 dB of masking. In stage 2, thresholds for detecting a brief 2200-Hz signal were measured using forward masking. It was reasoned that the threshold of the signal would give an indication of the amount of excitation evoked by the masker in the frequency region of the signal. Three forward maskers were used: (1) a 2200-Hz sinusoid at 10, 20, or 30 dB sensation level (SL); (2) a 2200-Hz sinusoid at the same levels as in (1) together with a sinusoid with frequency 500 or 1800 Hz at a level just sufficient to mask the 2200-Hz sinusoid. We refer to this as the "combined masker," (3) a 500-Hz or 1800-Hz sinusoid at the same levels as in (2) above. The 1800-Hz combined masker produced slightly less forward masking than the 2200-Hz masker (1), which might be explained in terms of suppression or as perceptual cueing. Both the 1800-Hz combined masker and the 1800-Hz component alone (3) gave significant amounts of forward masking (up to 18 dB), indicating that these maskers produced substantial excitation at 2200 Hz. This is consistent with the idea that the simultaneous masking of the 2200-Hz component in stage 1 was produced by spread of excitation rather than by suppression. The 500-Hz combined masker produced much less forward masking than the 2200-Hz component alone, indicating strong suppression of the 2200-Hz component of the combined masker by the 500-Hz component. However, both the 500-Hz combined masker and the 500-Hz component alone produced some forward masking. This is not consistent with the idea that masking of the 2200-Hz component in stage 1 (simultaneous masking) was produced solely by suppression.

Acoustic Stimulation↗

PubMed1997

Simulation of the effect of threshold elevation and loudness recruitment combined with reduced frequency selectivity on the intelligibility of speech in noise.

The effect of loudness recruitment and threshold elevation together with reduced frequency selectivity have been simulated to examine the combined effect of the two major consequences of cochlear hearing loss on the intelligibility of speech in speech-shaped noise. In experiment 1, four conditions were simulated: a moderate flat loss with auditory filters broadened by a factor of three (B3R2); a moderate-to-severe sloping loss with auditory filters broadened by a constant factor of three (B3RX); and these conditions with linear amplification applied prior to the simulation processing (B3R2+, B3RX+). For conditions B3R2 and B3RX, performance was markedly worse than for a control condition (normal hearing, condition R1) tested in a previous study. For conditions B3R2+ and B3RX+, linear amplification improved performance considerably. However, performance remained below that for condition R1 by between 5% and 19%. In experiment 2 the broadening of the auditory filters was made more realistic by making it a function of the absolute threshold at the center frequency of the auditory filter. Three different hearing losses were simulated: a moderate-to-severe sloping loss with variable broadening of the auditory filters (BXRX); the same moderate-to-severe sloping loss with linear amplification (BXRX+); and the same broadening of the auditory filters but without the simulation of loudness recruitment and threshold elevation (BX). For condition BXRX, performance was markedly worse than in condition R1, while performance in condition BX was somewhat worse than for condition R1. For condition BXRX+, linear amplification according to the NAL procedure improved performance to a large extent but it remained worse than for condition R1. The results are consistent with previous evidence indicating that only part of the decrease of performance produced by actual cochlear hearing loss can be compensated by conventional linear hearing aids.

Auditory Threshold↗

PubMed1997

Perceptual grouping of tone sequences by normally hearing and hearing-impaired listeners.

This study examined the perceptual grouping of rapid tone sequences for listeners with normal hearing and listeners with unilateral and bilateral cochlear hearing loss. The sequence ABA-ABA- was used, where A and B represent sinusoidal tones bursts (10-ms rise/fall, 80-ms steady state, 20-ms interval between tones) and - represents a silent interval of 120 ms. Tone A was fixed in frequency at 250, 500, 1000, or 2000 Hz. Tone B started with a frequency well above or below that of tone A, and its frequency was swept towards that of tone A so that the frequency separation between them decreased in an exponential manner. Listeners were required to indicate when they could no longer hear the tones A and B as two separate streams, but heard only a single stream with a "gallop" rhythm. This is called the fission boundary. For the normally hearing listeners, the separation between tones A and B at the fission boundary was roughly independent of the frequency of tone A when expressed as the difference in number of ERBs (delta E) between A and B, which is consistent with a recent model of stream segregation [M. W. Beauvois and R. Meddis, J. Acoust. Soc. Am. 99, 2270-2280 (1996)]. For the unilaterally hearing-impaired listeners, there was no consistent difference in the delta E magnitudes across ears, even though the auditory filters were broader in the impaired ears. This is not consistent with the theory of Beauvois and Meddis. The bilaterally hearing-impaired listeners sometimes showed delta E magnitudes within the normal range, and sometimes showed larger than normal delta E magnitudes. The results are discussed in terms of the factors that might influence perceptual stream formation in hearing-impaired listeners.

Adult↗

PubMed1997

Effect of the relative phase of amplitude modulation on the detection of modulation on two carriers.

This study examined how effectively information about amplitude modulation (AM) on two carriers is combined, and whether the detection of AM depends on the relative phase of the AM across carriers. Psychometric functions were measured for detecting 5-Hz sinusoidal AM of carriers with frequencies 1100 and 1925 Hz, with a mean level 65 dB SPL for each carrier. The carriers had a duration of 400 ms with 50-ms raised-cosine ramps on either side of this. A single cycle of 5-Hz sinusoidal AM (200 ms in duration) was imposed on the temporal center of the stimulus, with 100-ms steady-state fringes before and after the modulation. The modulators for the two carriers were either in phase or in antiphase. The modulation of each carrier was equally detectable, as determined in a preliminary experiment. A continuous pink noise background was used to mask the outputs of auditory filters tuned between the two carrier frequencies. There was no effect of relative modulator phase. However, performance was consistently better than predicted from the assumption that information about AM from the two carriers is processed independently and combined optimally. The results are discussed in terms of (1) predictions using Dau's "modulation filter bank model" [T. Dau et al., in Psychoacoustics, Speech and Hearing Aids, edited by B. Kollmeier (World Scientific, Singapore, 1996), pp. 45-48], and (2) the fact that relative modulator phase does have an effect on the detection of frequency modulation on two carriers, as found by Furukawa and Moore [J. Acoust. Soc. Am. 100, 2299-2311 (1996)].

Auditory Perception↗