A method for degrading sound localization while preserving binaural advantages for speech reception in noise.

This study developed and tested a real-time processing algorithm designed to degrade sound localization (LocDeg algorithm) without affecting binaural benefits for speech reception in noise. Input signals were divided into eight frequency channels. The odd-numbered channels were mixed between the ears to confuse the direction of interaural cues while preserving interaural cues in the even-numbered channels. The LocDeg algorithm was evaluated for normal-hearing listeners performing sound localization and speech-reception tasks. Results showed that the LocDeg algorithm successfully degraded sound-localization performance without affecting speech-reception performance or spatial release from masking for speech in noise. The LocDeg algorithm did, however, degrade speech-reception performance in a task involving spatially separated talkers in a multi-talker environment, which is thought to depend on differences in perceived spatial location of concurrent talkers. This LocDeg algorithm could be a valuable tool for isolating the importance of sound-localization ability from other binaural benefits in real-world environments.

Can monaural temporal masking explain the ongoing precedence effect?

The precedence effect for transient sounds has been proposed to be based primarily on monaural processes, manifested by asymmetric temporal masking. This study explored the potential for monaural explanations with longer ("ongoing") sounds exhibiting the precedence effect. Transient stimuli were single lead-lag noise burst pairs; ongoing stimuli were trains of 63 burst pairs. Unlike with transients, monaural masking data for ongoing sounds showed no advantage for the lead, and are inconsistent with asymmetric audibility as an explanation for ongoing precedence. This result, along with supplementary measurements of interaural time discrimination, suggests different explanations for transient and ongoing precedence.

Strength of onset and ongoing cues in judgments of lateral position.

This study describes the contributions to auditory image position of an interaural time delay (ITD) cue at onset relative to subsequent ITDs during the ongoing part of a stimulus. Test stimuli were trains of 1-ms binaural noise bursts; lateral position was measured with a wideband acoustic pointer that subjects adjusted to match the intracranial position of test stimuli. In different conditions the ongoing part of the stimulus (the bursts following the first one) either had a consistent ITD (the same ITD on each ongoing burst), or had alternating leading and lagging components with ITDs that opposed one another. As duration of the ongoing part was increased from 4 to 250 ms, with the initial ITD fixed, lateral position changed from being dominated by the onset ITD to being dominated by the ongoing consistent or leading ITD. With alternating ongoing ITDs equal contributions from onset and ongoing parts were obtained at an ongoing duration of about 40 ms; with consistent ongoing ITDs equal contributions were obtained at about 15 ms. The results point up the increased dominance of onset cues when ongoing cues are ambiguous, as they often are in real-world settings.

Threshold of the precedence effect in noise.

Three effects that show a temporal asymmetry in the influence of interaural cues were studied through the addition of masking noise: (1) The transient precedence effect-the perceptual dominance of a leading transient over a similar lagging transient; (2) the ongoing precedence effect-lead dominance with lead and lag components that extend in time; and (3) the onset capture effect-determination by an onset transient of the lateral position of an otherwise ambiguous extended trailing sound. These three effects were evoked with noise-burst stimuli and were compared in the presence of masking noise. Using a diotic noise masker, detection thresholds for stimuli with lead/lag interaural delays of 0/500 µs were compared to those with 500/0 µs delays. None of the three effects showed a masking difference between those conditions, suggesting that none of the effects is operative at masked threshold. A task requiring the discrimination between stimuli with 500/0 and 0/500 µs interaural delays was used to determine the threshold for each effect in noise. The results showed similar thresholds in noise (10-13 dB SL) for the transient and ongoing precedence effects, but a much higher threshold (33 dB SL) for onset capture of an ambiguous trailing sound.

Investigating the Utility of a Compact Loudspeaker Array for Audiometric Testing.

PURPOSE: This project addressed the uses of a loudspeaker array for audiometric measurements. It sought to evaluate a prototype compact array in terms of the reliability of test results across sound booths. METHOD: A seven-loudspeaker array was developed to deliver sounds from -60° to +60° on an arc with a radius of 0.5 m. The system was equipped with a head position sensing system to maintain the listener's head near the optimal test position. Three array systems were distributed to each of the two test sites for within-subject assessments of booth equivalence on tests of sound localization, speech reception in noise, and threshold detection. A total of 36 subjects participated, 18 at each test site. RESULTS: Results showed excellent interbooth consistency on tests of sound localization using speech and noise signals, including conditions in which one or both ears were covered with a muff. Booth consistency was also excellent on sound field threshold measurements for detecting quasi-diffuse noise bands. Nonequivalence was observed in some cases of speech-in-noise tests, particularly with a small one-person booth. Acoustic analyses of in situ loudspeaker responses indicated that some of the nonequivalent comparisons on speech-in-noise tests could be traced to the effects of reflections. CONCLUSIONS: Overall, the results demonstrate the utility and reliability of a compact array for the assessment of localization ability, speech reception in noise, and sound field thresholds. However, the results indicate that researchers and clinicians should be aware of the reflection effects that can influence the results of sound field tests in which signal and noise levels from separate loudspeakers are critical.

Lateralization of noise-burst trains based on onset and ongoing interaural delays.

The lateralization of 250-ms trains of brief noise bursts was measured using an acoustic pointing technique. Stimuli were designed to assess the contribution of the interaural time delay (ITD) of the onset binaural burst relative to that of the ITDs in the ongoing part of the train. Lateralization was measured by listeners' adjustments of the ITD of a pointer stimulus, a 50-ms burst of noise, to match the lateral position of the target train. Results confirmed previous reports of lateralization dominance by the onset burst under conditions in which the train is composed of frozen tokens and the ongoing part contains multiple ambiguous interaural delays. In contrast, lateralization of ongoing trains in which fresh noise tokens were used for each set of two alternating (left-leading/right-leading) binaural pairs followed the ITD of the first pair in each set, regardless of the ITD of the onset burst of the entire stimulus and even when the onset burst was removed by gradual gating. This clear lateralization of a long-duration stimulus with ambiguous interaural delay cues suggests precedence mechanisms that involve not only the interaural cues at the beginning of a sound, but also the pattern of cues within an ongoing sound.

Review article: review of the literature on temporal resolution in listeners with cochlear hearing impairment: a critical assessment of the role of suprathreshold deficits.

A critical review of studies of temporal resolution in listeners with cochlear hearing impairment is presented with the aim of assessing evidence for suprathreshold deficits. Particular attention is paid to the roles of variables-such as stimulus audibility, overall stimulus level, and participant's age-which may complicate the interpretation of experimental findings in comparing the performance of hearing-impaired (HI) and normal-hearing (NH) listeners. On certain temporal tasks (e.g., gap detection), the performance of HI listeners appears to be degraded relative to that of NH listeners when compared at equal SPL (sound pressure level). For other temporal tasks (e.g., forward masking), HI performance is degraded relative to that of NH listeners when compared at equal sensation level. A relatively small group of studies exists, however, in which the effects of stimulus audibility and level (and occasionally participant's age) have been controlled through the use of noise-masked simulation of hearing loss in NH listeners. For some temporal tasks (including gap-detection, gap-duration discrimination, and detection of brief tones in modulated noise), the performance of HI listeners is well reproduced in the results of noise-masked NH listeners. For other tasks (i.e., temporal integration), noise-masked hearing-loss simulations do not reproduce the results of HI listeners. In three additional areas of temporal processing (duration discrimination, detection of temporal modulation in noise, and various temporal-masking paradigms), further studies employing control of stimulus audibility and level, as well as age, are necessary for a more complete understanding of the role of suprathreshold deficits in the temporal-processing abilities of HI listeners.

Lateralization of two-transient stimuli.

In this study of the precedence effect in binaural hearing, subjects adjusted the interaural delay of a wideband acoustic pointer to match the perceived intracranial position of transient test stimuli presented over headphones. The test stimuli had leading and lagging components (either brief noise bursts or clicks), each with its own interaural delay. In some test conditions, the leading and lagging stimuli were coherent copies of one another, whereas in others, they were independent samples of noise. The duration of the stimuli and the delay from the leading component to the lagging component were also varied. All the stimulus conditions showed a moderate or strong precedence effect (i.e., covariation of perceived lateral position of the composite two-transient stimulus with the interaural delay of the leading component). Predictions of the lateralization data are presented for variants of models based on temporal weighting and/or bandpass correlation. In one model variant, the binaural stimuli are temporally weighted to emphasize the onset and then subjected to bandpass correlation analysis. In another variant, it is assumed that the onset mechanism provides a rough estimate of the initial interaural delay that guides a slower and more focused bandpass correlation analysis. The accuracies of these two model's predictions were equivalent and superior to those of models that either represent leading and lagging cues equally (bandpass correlation with no onset effect) or do not represent lagging cues at all (a complete precedence effect). The results of these analyses show the need for both a strong onset effect and for bandpass correlation analysis and suggest two modeling approaches for achieving that goal.

Evaluation of array-processing algorithms for a headband hearing aid.

Several array-processing algorithms were implemented and evaluated with experienced hearing-aid users. The array consisted of four directional microphones mounted broadside on a headband worn on the top of the listener's head. The algorithms included two adaptive array-processing algorithms, one fixed array-processing algorithm, and a reference condition consisting of binaural directional microphones. The algorithms were evaluated under conditions with both one and three independent noise sources. Performance metrics included quantitative speech reception thresholds and qualitative subject preference ratings for ease-of-listening measured using a paired-comparison procedure. On average, the fixed algorithm improved speech reception thresholds by 2 dB, while the adaptive algorithms provided 7-9-dB improvement over the reference condition. Subjects judging ease-of-listening generally preferred all array-processing algorithms over the reference condition. The results suggest that these adaptive algorithms should be evaluated further in more realistic acoustic environments.

Auditory target detection in reverberation.

Measurements and theoretical predictions of auditory target detection in simulated reverberant conditions are reported. The target signals were pulsed 1/3-octave bands of noise and the masker signal was a continuous wideband noise. Target and masker signals were passed through a software simulation of a reverberant room with a rigid sphere modeling a listener's head. The location of the target was fixed while the location of the masker was varied in the simulated room. Degree of reverberation was controlled by varying the uniform acoustic absorption of the simulated room's surfaces. The resulting target and masker signals were presented to the listeners over headphones in monaural-left, monaural-right, or binaural listening modes. Changes in detection performance in the monaural listening modes were largely predictable from the changes in target-to-masker ratio in the target band, but with a few dB of extra masking in reverberation. Binaural detection performance was generally well predicted by applying Durlach's [in Foundations of Modern Auditory Theory (Academic, New York, 1972)] equalization-cancellation theory to the direct-plus-reverberant ear signals. Predictions in all cases were based on a statistical description of room acoustics and on acoustic diffraction by a sphere. The success of these detection models in the present well-controlled reverberant conditions suggests that they can be used to incorporate listening mode and source location as factors in speech-intelligibility predictions.

Robustness of an Adaptive Beamforming Method for Hearing Aids.

We describe the results of computer simulations of a multimicrophone adaptive-beamforming system as a noise reduction device for hearing aids. Of particular concern was the system's sensitivity to violations of the underlying assumption that the target signal is identical at the microphones. Two-and four-microphone versions of the system were tested in simulated anechoic and modestly-reverberant environments with one and two jammers, and with deviations from the assumed straight-ahead target direction. Also examined were the effects of input target-to-jammer ratio and adaptive-filter length. Generally, although the noise-reduction performance of the system is degraded by target misalignment and modest reverberation, the system still provides positive advantage at input target-to-jammer ratios up to about 0 dB. This is in contrast to the degrading target-cancellation effect that the system can have when the equal-target assumption is violated and the input target-to-jammer ratio is greater than zero.