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B C Moore

Publications and source records attributed to B C Moore.

At least 127 records · Page 7Linked to original sources

Detection of combined frequency and amplitude modulation.

This article is concerned with the detection of mixed modulation (MM), i.e., simultaneously occurring amplitude modulation (AM) and frequency modulation (FM). In experiment 1, an adaptive two-alternative forced-choice task was used to determine thresholds for detecting AM alone. Then, thresholds for detecting FM were determined for stimuli which had a fixed amount of AM in the signal interval only. The amount of AM was always less than the threshold for detecting AM alone. The FM thresholds depended significantly on the magnitude of the coexisting AM. For low modulation rates (4, 16, and 64 Hz), the FM thresholds did not depend significantly on the relative phase of modulation for the FM and AM. For a high modulation rate (256 Hz) strong effects of modulator phase were observed. These phase effects are as predicted by the model proposed by Hartmann and Hnath [Acustica 50, 297-312 (1982)], which assumes that detection of modulation at modulation frequencies higher than the critical modulation frequency is based on detection of the lower sideband in the modulated signal's spectrum. In the second experiment, psychometric functions were measured for the detection of AM alone and FM alone, using modulation rates of 4 and 16 Hz. Results showed that, for each type of modulation, d' is approximately a linear function of the square of the modulation index. Application of this finding to the results of experiment 1 suggested that, at low modulation rates, FM and AM are not detected by completely independent mechanisms. In the third experiment, psychometric functions were again measured for the detection of AM alone and FM alone, using a 10-Hz modulation rate. Detectability was then measured for combined AM and FM, with modulation depths selected so that each type of modulation would be equally detectable if presented alone. Significant effects of relative modulator phase were found when detectability was relatively high. These effects were not correctly predicted by either a single-band excitation-pattern model or a multiple-band excitation-pattern model. However, the detectability of the combined AM and FM was better than would be predicted if the two types of modulation were coded completely independently.

Adult↗

Detection of temporal gaps in sinusoids by elderly subjects with and without hearing loss.

Thresholds for the detection of temporal gaps in sinusoidal signals were measured as a function of frequency (100-2000 Hz) and level in 15 elderly hearing-impaired subjects and 11 elderly subjects with near-normal hearing at frequencies below 2000 Hz. The sinusoids were presented in a background noise intended to mask spectral splatter associated with the gap. In a separate experiment, auditory filter shapes and detection efficiency were estimated for the same subjects using the notched-noise method, at center frequencies of 100, 200, 400, and 800 Hz. The gap thresholds at higher signal levels were similar for the two groups of subjects at all center frequencies tested. The mean gap thresholds were slightly higher than those obtained previously from young normally hearing subjects, but this was mainly due to the results of a few subjects with large gap thresholds; the majority of the elderly subjects had gap thresholds within the normal range. Thus reduced temporal resolution does not seem to be an inevitable consequence of aging. Gap thresholds at low center frequencies tended to be positively correlated with the equivalent rectangular bandwidth (ERB) of the auditory filter, the opposite of what would be expected if the auditory filter played a role in limiting gap detection. Detection efficiency, as estimated from the notched-noise experiment, was poorer for both groups of elderly subjects than for young normal listeners, but detection efficiency was not significantly correlated with gap thresholds.

Aged↗

Spectral feature enhancement for people with sensorineural hearing impairment: effects on speech intelligibility and quality.

People with sensorineural hearing loss often have difficulty understanding speech in background noise at speech-to-noise ratios (0 to +6 dB) for which normally hearing people have little difficulty. Spectral analysis of speech in noise at these ratios typically shows that the major spectral prominences in the speech (formants) are well represented, but the spectral valleys between the formants are filled with noise. Hearing impaired people have a reduced ability to pick out the spectral prominences, and are more affected by the noise filling in the valleys, partly because of their reduced frequency selectivity. This paper describes a 16-channel bandpass filter bank, implemented in analog electronics, that attempts to enhance spectral features of speech in noise to improve intelligibility for the hearing impaired. Each channel generates an 'activity function' that is proportional to the magnitude of the signal envelope in that channel, averaged over a short period of time. A positively weighted activity function from the nth channel is combined with negatively weighted functions from channels n-2, n-1, n+1, and n+2, giving a correction signal used to control the gain of the bandpass signal in the nth channel. Recombining the bandpass signals results in an enhancement of spectral features of the speech in noise. Two different experiments are described here, one using the activity function as described, and the other using a non-linear transform of the activity function. In both experiments, several different weighting patterns were used in calculating the correction signal. The intelligibility of speech in noise processed by the system was measured for subjects with moderate sensorineural hearing loss. In both experiments, no improvement in intelligibility was found. However, subjective ratings of the stimuli used in Experiment 2 indicated that some subjects judged the processed stimuli to have both higher quality and higher intelligibility than unprocessed stimuli.

Electronics↗

Temporal modulation transfer functions for band-limited noise in subjects with cochlear hearing loss.

The modulation depth required for the detection of sinusoidal amplitude modulation was measured as a function of modulation rate, giving temporal modulation transfer functions (TMTFs). The carrier was a one-octave wide noise centred at 2 kHz, and it was presented in an unmodulated background noise lowpass filtered at 5 kHz. Three subjects with unilateral cochlear hearing loss were tested. For each subject, the normal ear was tested both at the same sound pressure level (SPL) and at the same sensation level (SL) as the impaired ear. The TMTFs were essentially the same for the normal and impaired ears, both at equal SPL and at equal SL. The better ears of three subjects with bilateral cochlear losses were also tested. Again, TMTFs were essentially the same as obtained for normal ears. These results suggest that temporal resolution is not necessarily adversely affected by cochlear hearing loss, at least as measured by this task.

Adult↗

Simplified measurement of auditory filter shapes using the notched-noise method.

The shape of the auditory filter at a given centre frequency can be estimated by measuring the threshold for detecting a sinusoid presented in a spectral notch in a noise masker, as a function of notch width. Laboratory studies using this method have typically been based on threshold measurements for between 13 and 19 notch widths. In this note, we describe how both the overall sharpness and asymmetry of the auditory filter in hearing-impaired subjects can be estimated with reasonable accuracy using only five notch widths. This considerably reduces the testing time needed, making it possible to apply the method in clinical testing.

Acoustic Stimulation↗

Syllabic compression: effective compression ratios for signals modulated at different rates.

Compression circuits are being used increasingly in hearing aids to reduce the dynamic range of signals. Their performance is usually characterized by: (1) the threshold sound level above which the compression starts to operate; (2) the compression ratio, which is the change in input level (in dB) required to achieve a 1 dB change in output level; and (3) the attack and release times over which the signal is integrated to determine the necessary gain change. In many practical situations, the effective compression ratio obtained with dynamically varying signals such as speech is less than the compression ratio obtained using standard test signals (slow square-wave modulation with large modulation depth). This article describes the effective compression ratios achieved with sinusoidal modulation, as a function of modulation rate, level relative to the compression threshold, compression ratio and time constants. The effects of compression on a typical speech signal are also discussed.

Acoustic Stimulation↗

Effects of the fitting parameters of a two-channel compression system on the intelligibility of speech in quiet and in noise.

These experiments were carried out to assess how accurately the gains and compression ratios in a two-channel compression system needed to be set. We used as a research tool a laboratory version of a two-channel full-dynamic-range compression system. The system was initially adjusted to suit each hearing-impaired subject according to the manufacturer's recommendations. Then, further adjustments were made to ensure that speech stimuli were both audible and comfortable over a wide range of sound levels. Finally, the settings of the gains and compression ratios were systematically varied from the adjusted values and the effects of this on the intelligibility of speech in quiet and in noise (12-talker babble, levels of 65 and 75 dB SPL) were measured. The results indicated that speech reception thresholds (SRTs) in quiet were significantly adversely affected by decreases in low-level gain. However, SRTs in noise were relatively unaffected by changes in low-level gain. An exception occurred at the higher noise level used, where increases in the low-level gains (with corresponding increases in compression ratios) had a significant adverse effect on the SRTs. It is concluded that, provided excessive low-level gains (associated with high compression ratios) are avoided, the main criteria for fitting such a system should be listening comfort (i.e. achieving an acceptable tonal balance, and avoiding uncomfortably loud sounds) and an appropriate value of the threshold for detecting speech in quiet (which should be a little below 50 dB SPL).

Acoustic Stimulation↗

Across-channel masking of changes in modulation depth for amplitude- and frequency-modulated signals.

This study examines a form of masking that can take place when the signal and masker are widely separated in frequency and cannot be explained in terms of the traditional concept of the auditory filter or critical band. We refer to this as across-channel masking. The task of the subject was to detect an increment in modulation depth of a 1000-Hz sinusoidal carrier. The carrier could either be sinusoidally amplitude modulated or sinusoidally frequency modulated at a 10-Hz rate. Modulation increment thresholds of this "target" signal were measured for the target alone, and in the presence of two interfering sounds with carrier frequencies of 230 and 3300 Hz. When the interfering sounds were unmodulated, they had no effect on modulation increment thresholds. When the interfering sounds were either amplitude or frequency modulated, thresholds increased. Amplitude modulation (AM) increment thresholds were affected by both amplitude-modulated and frequency-modulated interference. Similarly, frequency modulation (FM) increment thresholds were affected by both amplitude-modulated and frequency-modulated interference. For both types of signal, the interference was tuned for modulation rate; across-channel masking was greatest when the interfering sounds were modulated at rates close to 10 Hz, and declined for higher or lower rates. However, the tuning was rather broad. When the target and interfering sounds were modulated at the same rate, there was no effect of the relative phase of the modulators. Two possible explanations for the results are discussed. One is based on the idea that carriers that are modulated in a similar way tend to be perceptually "grouped". The other is based on the idea that there are "channels" in the auditory system tuned for AM and FM rate. Neither explanation appears completely satisfactory.

Adult↗

Characterization and simulation of impaired hearing: implications for hearing aid design.

This article reviews several aspects of auditory perception that are affected by hearing loss of cochlear origin. It is argued that most of the observed effects can be understood in terms of damage to a physiologically vulnerable active process in the cochlea. In a normal ear, this process enhances sensitivity and frequency selectivity and reduces the slope of the input-output function on the basilar membrane. Damage to the active process in impaired ears leads to reduced sensitivity and frequency selectivity, an abnormally rapid rate of growth of loudness with intensity, reduced temporal integration and, for certain types of stimuli, reduced temporal resolution. The implications and relevance of each of these effects to the design and fitting of hearing aids are discussed.

Acoustics↗

Decrement detection in normal and impaired ears.

The smallest detectable duration of a brief decrement in the intensity of wideband noise was measured as a function of the depth of the decrement. In the first experiment, conditions were tested in which the noise before the decrement was more intense than the noise after the decrement, and vice-versa. These data were used to estimate the shape of an intensity-weighting function, or temporal window, describing the temporal resolution of the ear. The equivalent rectangular durations (ERDs) of the temporal windows measured in this way had values of about 5.5, 4.6, and 6.6 ms for noise spectrum levels of 10, 30, and 50 dB, respectively. In a second experiment, decrement detection was measured in subjects with unilateral sensorineural hearing loss. One set of thresholds was measured in the impaired ear, and two sets of thresholds were measured in the normal ear; one with the noise level at equal SPL to the level in the impaired ear, and one with the noise at equal SL. Temporal window shapes were also estimated from these data. Only one of the subjects showed reduced temporal resolution in the impaired ear, the other two subjects having similar ERD values for all three conditions.

Aged↗

Comodulation masking release as a function of level.

These experiments examine the effects of masker level on the magnitude of comodulation masking release (CMR). In experiment 1, threshold was measured for detecting a 2000-Hz signal in noise bands 100 or 3200 Hz wide, centered at the signal frequency. The noise was either amplitude modulated by a low-pass-filtered noise, or was unmodulated. At noise spectrum levels of 30 and 50 dB, thresholds were lower in the 3200-Hz-wide modulated noise than in the 100-Hz-wide modulated noise or the 3200-Hz-wide unmodulated noise, indicating a CMR. The magnitude of this CMR decreased at a noise spectrum level of 10 dB, and was very small at a spectrum level of -10 dB. In experiment 2, threshold was measured for a 700-Hz signal centered in a 20-Hz wide band of noise (the on-frequency band, OFB), both in the presence and absence of eight flanking bands (FBs) whose envelopes were either identical with that of the OFB (correlated condition) or were uncorrelated. Thresholds were lower in the correlated than in the uncorrelated condition, indicating a CMR. When the OFB and the FBs were presented to the same ear, the CMR decreased when the spectrum level of all bands was below 30 dB, or when the spectrum level of the FBs was decreased below 40 dB keeping the level of the OFB constant at 40 dB. When the OFB and the FBs were presented to opposite ears, the CMR decreased when the spectrum level of all bands was decreased below 30 dB or when the spectrum level of the FBs was decreased below 40 dB, keeping the level of the OFB fixed at 40 or 60 dB. However, the CMR was almost independent of the spectrum level of the OFB (over the range 10-70 dB) when the spectrum level of the FBs was held constant at 60 dB. The results are interpreted in terms of perceptual grouping mechanisms. Implications for the measurement of CMR in hearing-impaired subjects are also discussed.

Adolescent↗

Free intracellular Ca2+ concentration ([Ca2+]i) and growth hormone release from purified rat somatotrophs. I. GH-releasing factor-induced Ca2+ influx raises [Ca2+]i.

This study was carried out to investigate the role of free intracellular Ca2+ ([Ca2+]i) in the action of GH-releasing factor (GRF) by determining whether GRF causes and increase in [Ca2+]i and whether this increase results from changes in Ca2+ influx/efflux and/or mobilization of intracellular Ca2+ stores. We used a purified preparation of normal rat somatotrophs and examined the changes in 45Ca uptake, [Ca2+]i measured with indo-1, intracellular cAMP, and GH release induced by GRF. GRF stimulated a concentration-related biphasic increase in [Ca2+]i. Both the GRF-dependent increase in [Ca2+]i and GH release were blocked by incubation in low Ca2+ medium and by the organic Ca2+ antagonists nifedipine and diltiazem. The measurement of 45Ca uptake, in both steady state and nonsteady state conditions, demonstrated directly that GRF stimulates Ca2+ influx into somatotrophs. These data demonstrate that the GRF-stimulated increase in [Ca2+]i is dependent on Ca2+ influx. Redistribution of intracellularly stored Ca2+ could not be detected, even though intracellular Ca2+ stores were present. Therefore, the increase is due to Ca2+ influx, and the biphasic nature of the increase in [Ca2+]i induced by GRF is due to a difference in the rate of activation of Ca2+ influx and Ca2+ removal from the cytosol.

Animals↗

Optimization of a slow-acting automatic gain control system for use in hearing aids.

This paper describes experiments evaluating and optimizing an automatic gain control system, called dual front-end AGC (abbreviated as D), intended for use in hearing aids. This system has two purposes: (1) to compensate for variations in the overall level of speech from one situation to another by slowly changing its gain; (2) to protect the user from sudden intense transients without affecting the long-term gain. This is achieved by using two control voltages to determine the gain. One changes slowly as the input varies in level. Normally this component determines the overall gain. The other comes into operation when an intense transient occurs. It acts rapidly to reduce the gain, avoiding over-amplification of the transient, but its action ceases quickly after the end of the transient. We describe four experiments measuring speech intelligibility for subjects with cochlear hearing loss in which we determine optimum values for two of the time constants of the D system, namely the recovery time of the fast component and the attack time of the slow component. The experiments also compare the D system with linear amplification (L) and 'adaptive compression' (A). The results show: (1) for the D system, optimum values are about 80-150 ms for the recovery time of the fast component and 150-325 ms for the attack time of the slow component; (2) in situations where intense transient sounds are present, and there is either no background sound (experiment 1) or continuous speech-shaped noise as a background (experiment 2), the D system gives significantly better performance than the L or A systems. When the background noise is a single voice, reversed in time (experiment 3), the D and L systems give similar performance, and both are markedly superior to the A system; (3) when the level of speech is varied over a range of 30 dB (experiment 4), both D and A systems allow good performance over the whole range of levels. Performance for the L system worsens markedly at the lower levels.

Aged↗

Derivation of auditory filter shapes from notched-noise data.

A well established method for estimating the shape of the auditory filter is based on the measurement of the threshold of a sinusoidal signal in a notched-noise masker, as a function of notch width. To measure the asymmetry of the filter, the notch has to be placed both symmetrically and asymmetrically about the signal frequency. In previous work several simplifying assumptions and approximations were made in deriving auditory filter shapes from the data. In this paper we describe modifications to the fitting procedure which allow more accurate derivations. These include: 1) taking into account changes in filter bandwidth with centre frequency when allowing for the effects of off-frequency listening; 2) correcting for the non-flat frequency response of the earphone; 3) correcting for the transmission characteristics of the outer and middle ear; 4) limiting the amount by which the centre frequency of the filter can shift in order to maximise the signal-to-masker ratio. In many cases, these modifications result in only small changes to the derived filter shape. However, at very high and very low centre frequencies and for hearing-impaired subjects the differences can be substantial. It is also shown that filter shapes derived from data where the notch is always placed symmetrically about the signal frequency can be seriously in error when the underlying filter is markedly asymmetric. New formulae are suggested describing the variation of the auditory filter with frequency and level. The implication of the results for the calculation of excitation patterns are discussed and a modified procedure is proposed. The appendix list FORTRAN computer programs for deriving auditory filter shapes from notched-noise data and for calculating excitation patterns. The first program can readily be modified so as to derive auditory filter shapes from data obtained with other types of maskers, such as rippled noise.

Acoustic Stimulation↗

Monaural envelope correlation perception, revisited: effects of bandwidth, frequency separation, duration, and relative level of the noise bands.

This article presents the results of two experiments investigating performance on a monaural envelope correlation discrimination task. Subjects were asked to discriminate pairs of noise bands that had identical envelopes (referred to as correlated stimuli) from pairs of noise bands that had envelopes which were independent (uncorrelated stimuli). In the first experiment, a number of stimulus parameters were varied: the center frequency of the lower frequency noise band in a pair, f1; the frequency separation between component noise bands; the duration of the stimuli; and the bandwidth of the component noise bands. For a long stimulus duration (500 ms) and a relatively wide bandwidth (100 Hz), subjects could easily discriminate correlated from uncorrelated stimuli for a wide range of frequency separations between the component noise bands. This was true both when f1 was 350 Hz, and when f1 was 2500 Hz. In each case, narrowing the bandwidth to 25 Hz, or shortening the duration to 100 ms, or both, made the task more difficult, but not impossible. In the second experiment, the level of the higher frequency noise band in a pair was varied. Performance did not decrease monotonically as the level of this band was decreased below the level of the other band, and only showed marked impairment when the level of the higher frequency band was at least 60 dB below that of the lower frequency band. The pattern of results in these two experiments is different from that which is obtained when the same stimulus parameters are varied in experiments investigating comodulation masking release (CMR). This suggests that the mechanisms underlying CMR and those underlying the discrimination of envelope correlation are not identical.

Acoustic Stimulation↗

Frequency discrimination of complex tones with overlapping and non-overlapping harmonics.

These experiments address the following issues. (1) When two complex tones contain different harmonics, do the differences in timbre between them impair the ability to discriminate the pitches of the tones? (2) When two complex tones have only a single component in common, and that component is the most discriminable component in each tone, is the frequency discrimination of the component affected by differences in residue pitch between the two tones? (3) How good is the pitch discrimination of complex tones with no common components when each tone contains multiple harmonics, so as to avoid ambiguity of pitch? (4) Is the pitch discrimination of complex tones with common harmonics impaired by shifting the component frequencies to nonharmonic values? In all experiments, frequency difference limens (DLCs) were measured for multiple-component complex tones, using an adaptive two-interval, two-alternative, forced-choice task. Three highly trained subjects were used. The main conclusions are as follows. (1) When two tones have the first six harmonics in common, DLCs are larger when the upper harmonics are different than when the upper harmonics are in common or are absent. It appears that differences in timbre impair DLCs. (2) Discrimination of the frequency of a single common partial in two complex tones is worse when the two tones have different residue pitches than when they have the same residue pitch. (3) DLCs for complex tones with no common harmonics are generally larger than those for complex tones with common harmonics. For the former, large individual differences occur, probably because subjects are affected differently by differences in timbre. (4) DLCs for harmonic complex tones are smaller than DLCs for complex tones in which the components are mistuned from harmonic values. This can probably be attributed to the less distinct residue pitch of the inharmonic complexes, rather than to reduced discriminability of partials. Overall, the results support the idea that DLCs for complex tones with common harmonics depend on residue pitch comparisons, rather than on comparisons of the pitches of partials.

Adult↗