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PubMed1991

Psychometric functions for level discrimination.

To determine the form of psychometric functions for 2I,2AFC level discrimination (commonly called intensity discrimination), ten increment levels were presented in random order within blocks of 100 trials. Stimuli were chosen to encompass a wide range of conditions and difference limens: eight 10-ms tones had frequencies of 0.25, 1, 8, or 14 kHz and levels of 30, 60, or 90 dB SPL; two 500-ms stimuli also were tested: a 1-kHz tone at 90 dB SPL and broadband noise at 63 dB SPL. For each condition, at least 20 blocks were presented in mixed order. Results for five normal listeners show that the sensitivity, d', is nearly proportional to delta L (= 20 log [(p + delta p)/p], where p is sound pressure) over the entire range of difference limens. When d' is plotted against Weber fractions for sound pressure, delta p/p, or intensity, delta I/I, exponents of the best-fitting power functions decrease with increasing difference limens and are less than unity for large difference limens. The approximately proportional relation between d' and delta L agrees with modern multichannel models of level discrimination and with psychometric functions derived for single auditory-nerve fibers. The results also support the notion that the difference limen, expressed as delta LDL and plotted on a logarithmic scale, is an appropriate representation of performance in level-discrimination experiments.

Adult↗
PubMed1991

An interleaved tracking procedure to monitor unstable psychometric functions.

In some experimental situations, the psychometric function underlying performance may not be stable, but instead may shift along the stimulus axis in response to changes in attention, learning, or task difficulty. When this occurs, the measured threshold may be influenced and the slope of the measured function will be inaccurately shallow. With commonly used experimental procedures, it is difficult to know whether a shallow psychometric function slope is a true reflection of the sensory process, or is a result of "averaging" a highly variable underlying function. Here, a new method is described of estimating psychometric function slope from the variability in two interleaved adaptive tracks, consulted on alternate trials, that is resistant to the effects of shifting performance levels. Further, a mechanism is described for assessing the likelihood that a threshold was, in fact, stable over the course of its measurement. Computer simulations are reported as well as verification of the method in measurements of human performance on a psychophysical task. Several conditions of externally imposed variability were simulated to establish the ability of these procedures to identify unstable functions and produce accurate slope estimates. The procedures worked well for thresholds shifting by as little as 4 dB if the variation did not occur too rapidly. The procedure and associated analyses are recommended as a relatively "free" means of calculating slope and quantifying threshold reliability with little extra experimental effort.

Attention↗
PubMed1991

Human brain-stem auditory evoked responses obtained by cross correlation to trains of clicks, noise bursts, and tone bursts.

Brain-stem auditory evoked responses (BAERs) were obtained in eight normal-hearing young adults. Stimuli included clicks, noise bursts, and tone bursts. Tone bursts included carrier frequencies of 1, 2, 4, and 8 kHz. All stimuli were presented at 60 dB nHL. BAERs were obtained by presenting stimuli in pseudorandom trains, called maximum length sequences (MLSs). BAERs were recovered by cross correlating the responses with a recovery sequence. MLS-BAERs were obtained with minimum pulse intervals (MPIs) of 6, 4, and 2 ms. Conventional BAERs were also obtained for stimuli presented at a rate of 30 Hz. BAERs were obtained for all stimuli, for both the conventional averaging technique and for the cross-correlation technique. BAERs were observed for MPIs as short as 2 ms for all stimuli. Wave V was the only peak consistently identifiable for these stimuli. For all stimuli, wave V latency increased and wave V amplitude decreased with decreasing MPI. This is the first demonstration of the use of maximum length sequences combined with cross correlation to obtain BAERs to noise burst and tone burst stimuli.

Adult↗
PubMed1991

A composite model of the auditory periphery for the processing of speech based on the filter response functions of single auditory-nerve fibers.

A composite model of the auditory periphery, based upon a unique analysis technique for deriving filter response characteristics from cat auditory-nerve fibers, is presented. The model is distinctive in its ability to capture a significant broadening of auditory-nerve fiber frequency selectivity as a function of increasing sound-pressure level within a computationally tractable time-invariant structure. The output of the model shows the tonotopic distribution of synchrony activity of single fibers in response to the steady-state vowel [e] presented over a 40-dB range of sound-pressure levels and is compared with the population-response data of Young and Sachs (1979). The model, while limited by its time invariance, accurately captures most of the place-synchrony response patterns reported by the Johns Hopkins group. In both the physiology and in the model, auditory-nerve fibers spanning a broad tonotopic range synchronize to the first formant (F1), with the proportion of units phase-locked to F1 increasing appreciably at moderate to high sound-pressure levels. A smaller proportion of fibers maintain phase locking to the second and third formants across the same intensity range. At sound-pressure levels of 60 dB and above, the vast majority of fibers with characteristic frequencies greater than 3 kHz synchronize to F1 (512 Hz), rather than to frequencies in the most sensitive portion of their response range. On the basis of these response patterns it is suggested that neural synchrony is the dominant auditory-nerve representation of formant information under "normal" listening conditions in which speech signals occur across a wide range of intensities and against a background of unpredictable and frequently intense acoustic interference.

Animals↗
PubMed1991

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↗
PubMed1991

Effect of amplitude modulation on profile detection.

The effect of amplitude modulation on profile detection was examined in three experiments. The observer's task was to determine in which observation interval an increment was added to the 1000-Hz target component of a multitone complex in which the components were equally spaced on a logarithmic frequency scale from 200 to 5000 Hz. The target was unmodulated throughout the study. In some conditions, all nontarget components of the standard were modulated in phase; in other conditions, they were modulated with random phase. In experiment 1, the threshold was measured as a function of the modulation rate. The results show that, at low modulation rates, 5 Hz for example, modulation elevates threshold by about 13 dB. The threshold decreases as the modulation rate increases, with the threshold elevation being only 3 dB at 80 Hz. In experiment 2, threshold was measured as a function of modulation depth for both 21- and 5-component complexes. The results show that for a 5-Hz modulation rate the threshold decreases as the modulation depth decreases, and that the rate of decrease is greater for the 21-component complex than for the 5-component complex. In experiment 3, the effects of random-phase modulation were explored; the phase of the modulation waveform was randomly chosen for each component. The results show that there is no difference between in-phase and random-phase modulation when each component occupies a different critical band. If, however, two or more components occupy the same critical band, then randomizing the phase of modulation reduces the effective depth of modulation within that critical band, and the effect of modulation is thereby lessened.

Adult↗
PubMed1991

Temporal integration and multiple looks.

The decrease in detection and discrimination thresholds with increases in signal duration has often been taken to indicate that a process of relatively long-term temporal integration occurs in hearing. Two experiments are reported that suggest that no such process occurs. The first experiment is similar to the two-pulse experiment reported by Zwislocki [J. Zwislocki, J. Acoust. Soc. Am. 32, 1046-1059 (1960)] in which the threshold in quiet for a pair of brief pulses is measured as a function of the temporal separation between them. Our data indicate that power integration occurs only for separations less than approximately 5 ms. For separations larger than 5-10 ms, thresholds do not change with separation and the pulses appear to be processed independently. In the second experiment, brief 1-kHz tone pulses separated by 100 ms are presented during gaps in a wideband noise. The threshold for a pair of pulses is lower than that for either pulse presented alone, indicating that some type of "integration" occurs. However, the threshold for the pulse pair is not affected by changes in the level of the noise during the interval between the pulses. These data are inconsistent with the classical view of temporal integration that involves long-term integration. They are consistent with the notion that the input is sampled at a fairly high rate and that these samples or "looks" are stored in memory and can be accessed and processed selectively. This multiple-look model can account for the data from the present experiment and also can account for the data on temporal integration for tones and noise.(ABSTRACT TRUNCATED AT 250 WORDS)

Attention↗
PubMed1991

Turning on a tone.

It is possible to choose the starting phase of a pure tone in a way that minimizes the onset noise when the tone is turned on abruptly. A spectral model shows that when the tone has a low frequency, minimum onset noise is expected for a starting phase of zero (turning on a sine tone) but when the tone has a high frequency, minimum onset noise is expected for a starting phase of +/- 90 deg (turning on a cosine tone). Listening experiments confirm the above expectations and show that the transition between low- and high-frequency domains is sharp and depends upon both the electroacoustical transducer and the individual listener.

Adult↗
PubMed1991

The energy spectrum of an impulse: its relation to hearing loss.

Permanent threshold shifts obtained from 242 chinchillas that were exposed to various impulse noise paradigms have been related to the energy spectra of the impulses. The impulses were generated by three different shock tubes that produced impulse noise spectra whose A-weighted energies showed peaks at 0.25, 1, and 2 kHz. The results show that there is an increasing susceptibility to NIPTS as the audiometric test frequency increases from 0.5 to 16 kHz. This increase in susceptibility to NIPTS is further accentuated by approximately 5 to 10 dB for impulses whose spectra peak at 2 kHz.

Animals↗
PubMed1991

The importance of "temporal pattern" in traumatic impulse noise exposures.

The equal energy hypothesis (EEH) was evaluated for impulse noise. Specifically, the experiments evaluated the importance of the temporal distribution of impulses; the trading relation between the number of impulses and peak level and the difference between continuous and impulse noise. Monaural chinchillas were exposed to one of seven conditions. Their hearing was evaluated before, immediately after, and 30 days after the exposure. Hair cell damage was reported in the form of a cochleogram. The experiments show that the EEH is more appropriate for low-level impulse (135-dB peak); for equal amounts of energy, 150-dB impulses produce more hearing loss and hair cell damage than 135-dB impulses; for equal amounts of energy, impulses presented in rapid bursts cause less damage than impulses presented at "1/s" and 50 microseconds. Pairs of impulses presented at "1/s" produce the largest amount of damage. The results are discussed in terms of implications for the EEH.

Animals↗
PubMed1991

The effect of frequency-selective attenuation on the speech-reception threshold of sentences in conditions of low-frequency noise.

Within a study on the merits of a multichannel automatic gain control in hearing aids, the effect of frequency-selective amplification on the masked speech-reception threshold (SRT) for sentences is measured in conditions of seriously disturbing low-frequency noise, with the effect of wideband amplification as a reference. Speech and noise are both spectrally shaped according to the bisector line of the listener's dynamic-range of hearing, but with the noise in a single octave band (0.25-0.5 or 0.5-1 kHz) increased by 20 dB relative to this line. The increase of noise level is steady state in the first experiment, and time varying in the second experiment. Results for 12 normal-hearing and 12 hearing-impaired listeners indicate that, in both experiments, frequency-selective compression of the signal in the octave band with the 20-dB increase of noise is more beneficial than wideband compression. For the hearing-impaired group, wideband compression does not give any systematic change in intelligibility. Frequency-selective compression in steady-state conditions may, for both groups of listeners, give a decrease of masked SRT (relative to a condition without compression) of up to 4 dB for a compression factor of 100%. Roughly comparable effects are seen for frequency-selective compression in time-varying conditions. The superiority of frequency-selective over wideband compression is attributed to a more effective reduction of upward spread of masking.

Adult↗
PubMed1992

Evidence for two discrete sources of 2f1-f2 distortion-product otoacoustic emission in rabbit: I. Differential dependence on stimulus parameters.

The results of studies of the physiological vulnerability of distortion-product otoacoustic emissions (DPOAEs) suggest that the DPOAE at 2f1-f2 in vertebrate ears is generated by more than one source. The principal aims of the present study were to provide independent evidence for the existence of more than one DPOAE source, and to determine the contributions of each to the ear-canal 2f1-f2 signal. To accomplish these aims, specific stimulus parameters were separately and systematically varied to provide detailed parametric information regarding 2f1-f2 DPOAE amplitude and phase in normal ears of awake rabbits. The findings indicate that two discrete sources, demonstrating differential dependence on stimulus parameters, dominate the generation of the 2f1-f2 DPOAE. One source of distortion is dominant above 60-70 dB SPL at moderate primary-frequency separations, and at all stimulus levels when the primary tones are closely spaced. The other source is dominant below 60-70 dB SPL at moderate primary-frequency separations, and may be dominant at all stimulus levels when the primary tones are widely separated in frequency. The results suggest that by varying stimulus parameters, it may be possible to independently study the two generator mechanisms.

Animals↗
PubMed1992

An auditory psychometric function from newborn chicks.

This study shows how the hearing of newborn chicks can be quantified with receiver operating characteristics (ROCs). A detailed auditory psychometric function was constructed from areas under ROCs, derived from delays in ongoing vocalizations elicited by pulsing pure tones. Over 20,000 delays were analyzed from 160 birds tested at four frequencies (0.25-2 kHz), eight intensities (-3 to 18 dB above estimated threshold), and two ages (0 and 4 days posthatch). Areas under these ROCs increase in an S-shaped function over intensity, as expected. Maximum responsiveness at approximately 75% correct shows that these neonates never achieve the near-perfect performance of mature listeners. Thresholds appear best estimated by the level required for 65% correct. Results replicate previous findings, showing that thresholds mature at low frequencies before high. Although there is no simple way to alleviate decreasing responsiveness caused by habituation over trials, improved performance results from presenting different frequencies, but only in the younger birds, and primarily with louder stimuli.

Animals↗
PubMed1992

The number of components in profile analysis tasks.

The central issue of this paper is how detection of a change in spectral shape depends on the number of components m that define the spectrum. First, a channel theory is reviewed and the predictions relating the detectability of such a change as a function of the number of components are derived. Second, the existing literature is reviewed. Most of it concerns the detection of an increment in a single tone of an m-component, equal-amplitude complex. The data are close (+/- 2 dB) to the predictions of the theory, which suggests that there is a relatively small change in threshold as the number of components increases. The consistency of the data is also discussed and hypotheses for discrepancies are presented. Finally, the results of an experiment measuring the detectability of a special spectral shape change is explored. In this case, all the changes are equal in size but some are opposite in sign, so that the summed change is zero. The results using this "balanced" spectral change are also consistent with the predictions of the theory and show that for this special kind of spectral change thresholds improve by nearly 7 dB as m increases from 4 to 24 components.

Adult↗
PubMed1992

Minimum audible movement angle in the horizontal plane as a function of stimulus frequency and bandwidth, source azimuth, and velocity.

Minimum audible movement angles (MAMAs) were measured in the horizontal plane for four normal-hearing adult subjects in a darkened anechoic chamber. On each trial, a single stimulus was presented, and the subject had to say whether it came from a stationary loudspeaker or from a loudspeaker that was moving at a constant angular velocity around him. Thresholds were established by adaptively varying stimulus duration. In experiment 1, MAMAs were measured as a function of center frequency (500-5000 Hz), velocity (10 degrees-180 degrees/s), and direction of motion (left versus right). There was no effect of direction of motion. MAMAs increased with velocity, from an average of 8.8 degrees of arc for a target moving at 10 degrees/s to an average of 20.2 degrees of arc for a target moving at 180 degrees/s. MAMAs were higher for a 3000-Hz tone than for tones of lower or higher frequencies, as has been previously reported [D. R. Perrott and J. Tucker, J. Acoust. Soc. Am. 83, 1522-1527 (1988)]. In experiment 2, minimum audible angles (MAAs) were measured with sequentially presented stationary tone pulses (500-5000 Hz), and were shown to exhibit the same dependence on signal frequency that the MAMAs showed (average MAA at 3000 Hz: 8.4 degrees; average MAA at the other frequencies: 3.4 degrees). In experiment 3, MAMAs and MAAs were measured as a function of stimulus bandwidth (centered at 3000 Hz) and listening azimuth (0 degrees vs 60 degrees). Average MAAs decreased monotonically as stimulus bandwidth increased from 0 Hz to wideband (from 8.4 degrees to 1.2 degrees at 0 degrees azimuth; from 11.3 degrees to 1.5 degrees at 60 degrees azimuth). As in experiment 1, MAMAs increased with stimulus velocity, from values comparable to the MAAs for the slowest-velocity (10 degrees/s) targets to 70 degrees of arc or more in the poorest condition (third-octave band of noise presented at a velocity of 180 degrees/s and an azimuth of 60 degrees). MAMAs obtained in the slower-velocity conditions depended in the same way on stimulus bandwidth and listening azimuth that MAAs depended on these variables. In no case was the MAMA ever smaller than the MAA. It is hypothesized that a minimum integration time is required to achieve optimal performance in a dynamic spatial resolution task. Average estimates of this minimum time based on the current data vary from 336 ms (for targets presented at midline) to 1116 ms (for narrow-band targets presented at 60 degrees azimuth).(ABSTRACT TRUNCATED AT 400 WORDS)

Adult↗
PubMed1992

The dominant role of low-frequency interaural time differences in sound localization.

Two experiments are described in which listeners judge the apparent directions of virtual sound sources-headphone-presented sounds that are processed in order to simulate free-field sounds. Previous results suggest that when the cues to sound direction are preserved by the simulation, the apparent directions of virtual sources are nearly the same as the apparent directions of real free-field sources. In the experiments reported here, the interaural phase relations in the processing algorithms are manipulated in order to produce stimuli in which the interaural time difference cues signal one direction and interaural intensity and pinna cues signal another direction. The apparent directions of these conflicting cue stimuli almost always follow the interaural time cue, as long as the wideband stimuli include low frequencies. With low frequencies removed from the stimuli, the dominance of interaural time difference disappears, and apparent direction is determined primarily by interaural intensity difference and pinna cues.

Adult↗
PubMed1992

Evaluation of an adaptive beamforming method for hearing aids.

In this paper evaluations of a two-microphone adaptive beamforming system for hearing aids are presented. The system, based on the constrained adaptive beamformer described by Griffiths and Jim [IEEE Trans. Antennas Propag. AP-30, 27-34 (1982)], adapts to preserve target signals from straight ahead and to minimize jammer signals arriving from other directions. Modifications of the basic Griffiths-Jim algorithm are proposed to alleviate problems of target cancellation and misadjustment that arise in the presence of strong target signals. The evaluations employ both computer simulations and a real-time hardware implementation and are restricted to the case of a single jammer. Performance is measured by the spectrally weighted gain in the target-to-jammer ratio in the steady state. Results show that in environments with relatively little reverberation: (1) the modifications allow good performance even with misaligned arrays and high input target-to-jammer ratios; and (2) performance is better with a broadside array with 7-cm spacing between microphones than with a 26-cm broadside or a 7-cm endfire configuration. Performance degrades in reverberant environments; at the critical distance of a room, improvement with a practical system is limited to a few dB.

Equipment Design↗