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Introduction: The ability to localize sound sources is crucial for humans. Due to specific hearing disorders, the affected individuals may have problems to accurately locate the sound sources, leading to other unwanted consequences. Nevertheless, a simple auditory localization test (that employs moving auditory stimuli) is currently lacking in clinical settings. Essentially, the objectives of the present study were to develop a virtual moving auditory localization (vMAL) test that is suitable for assessing children and assess the validity and the reliability of this test. Materials and Methods: This study consisted of two consecutive phases. In phase 1, the required stimulus and the test set up for the vMAL test were established. Two loudspeakers were employed to produce five virtual positions, and eight different moving conditions were constructed. In phase 2, 24 normal-hearing Malaysian children (aged 7-12 years) underwent the vMAL test. The validity and the reliability of this test were then assessed using several validation measures. Fleiss Kappa and Spearman correlation analyses were used to analyse the obtained data. Results: The vMAL test was found to have good convergent validity (kappa = 0.64) and good divergent validity (kappa = -0.06). Based on the item-total correlation and Spearman coefficient rho results, this test was found to have good internal reliability (rho = 0.36-0.75) and excellent external (test-retest) reliability (rho = 0.99). Conclusions: in this study a new vMAL test was developed and proven to be valid and reliable accordingly for its intended applications. This test can be useful in clinical settings since it is simple to administer, cost-effective, does not take up much room, and can assess auditory localization performance in children. The outcomes of the present study may serve as preliminary normative data as well as guidelines for future auditory localization research.

A signal processing approach combining beamforming with mask-informed speech enhancement was assessed by measuring sentence recognition in listeners with mild-to-moderate hearing impairment in adverse listening conditions that simulated the output of behind-the-ear hearing aids in a noisy classroom. Two types of beamforming were compared: binaural, with the two microphones of each aid treated as a single array, and bilateral, where independent left and right beamformers were derived. Binaural beamforming produces a narrower beam, maximising improvement in signal-to-noise ratio (SNR), but eliminates the spatial diversity that is preserved in bilateral beamforming. Each beamformer type was optimised for the true target position and implemented with and without additional speech enhancement in which spectral features extracted from the beamformer output were passed to a deep neural network trained to identify time-frequency regions dominated by target speech. Additional conditions comprising binaural beamforming combined with speech enhancement implemented using Wiener filtering or modulation-domain Kalman filtering were tested in normally-hearing (NH) listeners. Both beamformer types gave substantial improvements relative to no processing, with significantly greater benefit for binaural beamforming. Performance with additional mask-informed enhancement was poorer than with beamforming alone, for both beamformer types and both listener groups. In NH listeners the addition of mask-informed enhancement produced significantly poorer performance than both other forms of enhancement, neither of which differed from the beamformer alone. In summary, the additional improvement in SNR provided by binaural beamforming appeared to outweigh loss of spatial information, while speech understanding was not further improved by the mask-informed enhancement method implemented here. © The Author(s) 2022.

The Dichotic Digits Test (DDT) evaluates central auditory nervous system (CANS) dysfunction. The DDT is widely used in audiology clinics worldwide, because it is clinically efficient and has good sensitivity and specificity for CANS lesions. However, the DDT shows a strong ceiling effect, which can mitigate its ability to detect subtle CANS dysfunction. This study examines the effects of adding monaural and binaural speech-spectrum noise to the DDT, in an effort to make the test more challenging to the CANS and thereby reduce the observed ceiling effect. Effects of monaural and binaural noise on different DDT scoring indices, including Individual Ear Percent Correct Scores, Combined Total Percent Correct Scores, and Dichotic Difference Scores, as well as noise effects on the Right Ear Advantage for speech, are investigated here. Statistically significant differences on all indices were found between monaural and binaural noise-added conditions, suggesting a possible advantage for binaural listening in noise. These preliminary findings suggest that adding noise to tests of dichotic listening and assessing dichotic listening patterns in noise could add sensitivity to evaluations of CANS integrity and function. Keywords: binaural hearing, ceiling effect, central auditory processing, dichotic listening, neuroaudiology, Right Ear Advantage, speech-in-noise

Background. Advances in the design of the cochlear implant electrode arrays and improved surgical techniques have resulted in the successful preservation of residual hearing after implantation. The preservation of residual hearing following cochlear implantation provides the capacity for combining electrical and acoustic stimulation in the same ear to benefit the listener. The effectiveness of electric and acoustic stimulation in the implanted ear as compared with electrical and acoustic stimulation alone has been consistently demonstrated in the research literature. There is also accumulating research evidence that supports the provision of acoustic stimulation for residual hearing in the implanted ear together with the non-implanted ear of recipients of a unilateral cochlear implant. A variety of approaches have been used in research studies to fit devices that provide electric and acoustic stimulation in either the implanted ear or in opposite ears, however, there is no comprehensive study evaluating binaural fitting strategies in adult cochlear implant users with residual hearing in both ears. Investigation of a fitting procedure that could maximise hearing outcomes for adult recipients who use electric and acoustic stimulation in the implanted ear and acoustic stimulation in the non-implanted ear would be of benefit. Objective. The overarching goal of this thesis is to examine the fitting of devices that combine electric and acoustic stimulation for cochlear implant recipients with residual hearing in both ears. Aims 1. To systematically investigate how to adjust the hearing aid acoustic output in both ears, for adults who use electric and acoustic stimulation in the implanted ear together with acoustic stimulation in the non-implanted ear. 2. To investigate the effect of varying cross-over frequency settings for electric and acoustic stimulation in the implanted ear combined with acoustic stimulation in the non-implanted ear on speech perception in noise, loca

Sound localization is particularly important for maintaining situational awareness and overall safety, yet this listening skill is often neglected during clinical evaluations of the auditory system. Many people experience sound localization difficulties, including those with hearing loss (peripheral and central), the aging adult population (with or without substantial hearing loss), and those who have experience traumatic brain injury, such as concussion or stroke, for example. There are two main cues that are used to localize sounds on the horizontal plane: interaural level differences (ILD) and interaural timing differences (ITD). In both cases, comparisons are made by the central auditory nervous system for the sound arriving at both ears; these are called “binaural cues”. Much is known about how listeners utilize ILDs and ITDs for the purposes of sound localization for static or fixed location sound sources. However, some basic questions remain regarding the use of these binaural cues for auditory movement perception. The practical implication here, is that for the most part, our auditory environment is dynamic in the sense of relative movement; that is, either the listener and/or the sound sources are moving. While there is a substantial literature on localization, electrophysiology, and central auditory processing of fixed-position binaural cues, the literature about perception and electrophysiology for moving or dynamic cues is rather thin. The studies in this dissertation fill a gap in knowledge with systematic comparisons of fixed vs. moving sound sources, using both psychoacoustic and electrophysiologic measures. The motivation for the studies in this dissertation was born from both scientific as well as a clinical perspective. The clinical perspective was creating some novel stimulus paradigms that could be translated into methods that could be used in an auditory processing assessment. The specific aims of this dissertation were to 1) Evaluate the effect

The auditory system uses interaural time and level differences (ITD and ILD) as cues to localize and lateralize sounds. The availability of ITDs and ILDs in the auditory system is limited by neural phase-locking and by the head size, respectively. Although the frequency-specific limitations are well known, the relative contribution of ITDs and ILDs in individual frequency bands in broadband stimuli is unknown. To determine these relative contributions, or spectral weights, listeners were asked to lateralize stimuli consisting of eleven simultaneously presented 1-ERB-wide noise bands centered between 442 and 5544 Hz and separated by 1-ERB-wide gaps. Either ITDs or ILDs were varied independently across each noise band, while fixing the other interaural disparity to either 0 dB or 0 μs. The weights were obtained using a multiple linear regression analysis. In a second experiment, the effect of auditory enhancement on the spectral weights was investigated. The enhancement of single noise bands was realized by presenting ten of the noise bands as preceding and following sounds (pre- and post-cursors, respectively). Listeners were asked to lateralize the stimuli as in the first experiment. Results show that in the absence of pre- and post-cursors, only the lowest or highest frequency band received highest weight for ITD and ILD, respectively. Auditory enhancement led to significantly enhanced weights given to the band without the pre- and post-cursor. The weight enhancement could only be observed at low frequencies, when determined with ITD cues and for low and high frequencies for ILDs. Hence, the auditory system seems to be able to change the spectral weighting of binaural information depending on the information content.

Cochlear implants are a promising new treatment option for single-sided deafness. Cochlear implants for single-sided deafness have been shown to improve speech perception in noise and aid in sound localization. However, this intervention is not as good as acoustic hearing and listeners’ exhibit large amounts of variability in hearing outcomes. These limitations may be caused by certain distortions inherent in the processing of the sound signals by the cochlear implant. This dissertation examined the role that three key cochlear implant distortions might play in limiting speech perception in noise for listeners with single-sided deafness. The first distortion examined was the frequency mismatch between the cochlear implant and the acoustic ear. The next distortion examined was the effect of timing differences between the cochlear implant and the normal hearing ear. Finally, the effect of compression on hearing speech in spatial noise was investigated. These limitations and distortions could limit binaural processing ability in those with single-sided deafness who receive a cochlear implant. The goal of this dissertation was to examine the role of cochlear-implant distortions on binaural hearing using simulations of cochlear implant processing presented to normal-hearing listeners. Normal-hearing listeners were presented with vocoder simulations of cochlear-implant processing to one ear, and unprocessed signals to the other ear. These simulations were used to examine the ability to understand binaural speech signals in noisy environments and to examine auditory object formation in simulated free-field environments. These data provided insight into how CI distortions and mapping strategies can limit binaural benefits for those with single-sided deafness. Knowledge of these limitations could lead to better programming strategies to improve binaural hearing and quality of life for those with single-sided deafness who receive a cochlear implant.

Listeners can perceive interleaved sequences of sounds from two or more sources as segregated streams. In humans, physical separation of sound sources is a major factor enabling such stream segregation. Here, we examine spatial stream segregation with a psychophysical measure in domestic cats. Cats depressed a pedal to initiate a target sequence of brief sound bursts in a particular rhythm and then released the pedal when the rhythm changed. The target bursts were interleaved with a competing sequence of bursts that could differ in source location but otherwise were identical to the target bursts. This task was possible only when the sources were heard as segregated streams. When the sound bursts had broad spectra, cats could detect the rhythm change when target and competing sources were separated by as little as 9.4°. Essentially equal levels of performance were observed when frequencies were restricted to a high, 4-to-25-kHz, band in which the principal spatial cues presumably were related to sound levels. When the stimulus band was restricted from 0.4 to 1.6 kHz, leaving interaural time differences as the principal spatial cue, performance was severely degraded. The frequency sensitivity of cats in this task contrasts with that of humans, who show better spatial stream segregation with low- than with high-frequency sounds. Possible explanations for the species difference includes the smaller interaural delays available to cats due to smaller sizes of their heads and the potentially greater sound-level cues available due to the cat's frontally directed pinnae and higher audible frequency range.

Listeners can perceive interleaved sequences of sounds from two or more sources as segregated streams. In humans, physical separation of sound sources is a major factor enabling such stream segregation. Here, we examine spatial stream segregation with a psychophysical measure in domestic cats. Cats depressed a pedal to initiate a target sequence of brief sound bursts in a particular rhythm and then released the pedal when the rhythm changed. The target bursts were interleaved with a competing sequence of bursts that could differ in source location but otherwise were identical to the target bursts. This task was possible only when the sources were heard as segregated streams. When the sound bursts had broad spectra, cats could detect the rhythm change when target and competing sources were separated by as little as 9.4°. Essentially equal levels of performance were observed when frequencies were restricted to a high, 4-to-25-kHz, band in which the principal spatial cues presumably were related to sound levels. When the stimulus band was restricted from 0.4 to 1.6 kHz, leaving interaural time differences as the principal spatial cue, performance was severely degraded. The frequency sensitivity of cats in this task contrasts with that of humans, who show better spatial stream segregation with low- than with high-frequency sounds. Possible explanations for the species difference includes the smaller interaural delays available to cats due to smaller sizes of their heads and the potentially greater sound-level cues available due to the cat's frontally directed pinnae and higher audible frequency range.

A hearing aid using cartilage conduction (CC) has been proposed as an alternative to bone conduction (BC) hearing aids. The transducer developed for this application is lightweight, requires a much smaller fixation force than a BC hearing aid, and is more convenient to use. CC can be of great benefit to patients with fibrotic aural atresia. Fibrotic tissue connected to the ossicles provides an additional pathway (termed fibrotic tissue pathway) for sound to reach the cochlea by means of CC. To address the function of fibrotic tissue pathway, BC and CC thresholds were measured in six ears with fibrotic aural atresia. The relationship between the CC thresholds and the results of computed tomography was investigated. In the ears with the presence of a fibrotic tissue pathway, the CC thresholds were lower than the BC thresholds at 0.5 and 1.0 kHz. At 2.0 kHz, no significant difference was observed between the BC and CC thresholds. The current findings suggest that sound in the low to middle frequency range is transmitted more efficiently by CC via a fibrotic tissue pathway than BC. The development of hearing devices using CC can contribute to rehabilitation, particularly in patients with fibrotic aural atresia., 博士（医学）・甲621号・平成26年3月17日

Parametric methods for modeling the perceptually relevant features of head-related transfer functions (HRTFs) are very important for the development of low-cost immersive sound applications. This letter describes an efficient method based on a low-order infinite impulse response filter implemented by a chain of second order sections of conventional shelving and peak audio filters. The parameters (central frequency, gain, and quality factor) are numerically adjusted by iteratively fitting the frequency response of the filter to the desired HRTF. Besides allowing for low-order binaural models, the proposed approach provides an efficient way to synthesize HRTFs for non-measured angles by applying a simple interpolation between the parameters from neighboring responses. Additionally, the HRTF database size is significantly reduced. (C) 2013 Acoustical Society of America.

Parametric methods for modeling the perceptually relevant features of head-related transfer functions (HRTFs) are very important for the development of low-cost immersive sound applications. This letter describes an efficient method based on a low-order infinite impulse response filter implemented by a chain of second order sections of conventional shelving and peak audio filters. The parameters (central frequency, gain, and quality factor) are numerically adjusted by iteratively fitting the frequency response of the filter to the desired HRTF. Besides allowing for low-order binaural models, the proposed approach provides an efficient way to synthesize HRTFs for non-measured angles by applying a simple interpolation between the parameters from neighboring responses. Additionally, the HRTF database size is significantly reduced. (C) 2013 Acoustical Society of America.

Neurons in the medial superior olive (MSO) enable sound localization by their remarkable sensitivity to submillisecond interaural time differences (ITDs). Each MSO neuron has its own "best ITD" to whi

All Speech Processing Equipments like noise cancelling headphones and Hearing aids should be able to filter different kinds of interfering signals and present a clear sound to the listener. The effect of these interfering signals varies depending on the surroundings. Confining the area of interest to a closed room, ranging from a class room to an auditorium, the most significant interference that is dominant in the received signal is the “Reverberation Effect”. Reverberation is the combined effect of multiple echoes from the walls of the room and its intensity depends on the size, shape, dimensions and materials used in the construction of the room. This phenomenon is tolerable for some purposes like music recordings but it predominantly effects the speech. Eliminating this effect without affecting the original speech is a challenge and a topic of research in these days. This paper is an excerpt of the collective work done on reducing microphone inaccuracies using various adaptive beamformers and emphasizes on-the Elko’s Beamformer, the Weiner Beamformer, Max-SNIR Beamformer and the delay-and-sum beamformer. These beamformers are implemented with the fractional delay capability and then tested under reverberant conditions. This paper mainly concentrates on the effect of Reverberation on the outcome of a beamformer. A reverberant room environment is simulated and the effect of reverberation on an Elko’s beamformer output is compared with that of a Weiner beamformer. The size of the room is proportionate to the complexity of the implementation, i.e., as the size increases, so does the execution time of the algorithm. All these kinds of factors have been implemented and tested. Improvement in the Signal-to-Noise Ratio and PESQ scores of these beamformers with reverberant room and without reverberant room are noted. Results show that wiener beamformer has better average SNR and PESQ improvements both in reverberant and non-reverberant conditions with more microphones an, S/o P S N Murthy, Vizianagaram, 535003, India. Ph: 0046723362292 Ph: 00919492099315 (India)

All Speech Processing Equipments like noise cancelling headphones and Hearing aids should be able to filter different kinds of interfering signals and present a clear sound to the listener. The effect of these interfering signals varies depending on the surroundings. Confining the area of interest to a closed room, ranging from a class room to an auditorium, the most significant interference that is dominant in the received signal is the “Reverberation Effect”. Reverberation is the combined effect of multiple echoes from the walls of the room and its intensity depends on the size, shape, dimensions and materials used in the construction of the room. This phenomenon is tolerable for some purposes like music recordings but it predominantly effects the speech. Eliminating this effect without affecting the original speech is a challenge and a topic of research in these days. This paper is an excerpt of the collective work done on reducing microphone inaccuracies using various adaptive beamformers and emphasizes on-the Elko’s Beamformer, the Weiner Beamformer, Max-SNIR Beamformer and the delay-and-sum beamformer. These beamformers are implemented with the fractional delay capability and then tested under reverberant conditions. This paper mainly concentrates on the effect of Reverberation on the outcome of a beamformer. A reverberant room environment is simulated and the effect of reverberation on an Elko’s beamformer output is compared with that of a Weiner beamformer. The size of the room is proportionate to the complexity of the implementation, i.e., as the size increases, so does the execution time of the algorithm. All these kinds of factors have been implemented and tested. Improvement in the Signal-to-Noise Ratio and PESQ scores of these beamformers with reverberant room and without reverberant room are noted. Results show that wiener beamformer has better average SNR and PESQ improvements both in reverberant and non-reverberant conditions with more microphones an, S/o P S N Murthy, Vizianagaram, 535003, India. Ph: 0046723362292 Ph: 00919492099315 (India)

All Speech Processing Equipments like noise cancelling headphones and Hearing aids should be able to filter different kinds of interfering signals and present a clear sound to the listener. The effect of these interfering signals varies depending on the surroundings. Confining the area of interest to a closed room, ranging from a class room to an auditorium, the most significant interference that is dominant in the received signal is the “Reverberation Effect”. Reverberation is the combined effect of multiple echoes from the walls of the room and its intensity depends on the size, shape, dimensions and materials used in the construction of the room. This phenomenon is tolerable for some purposes like music recordings but it predominantly effects the speech. Eliminating this effect without affecting the original speech is a challenge and a topic of research in these days. This paper is an excerpt of the collective work done on reducing microphone inaccuracies using various adaptive beamformers and emphasizes on-the Elko’s Beamformer, the Weiner Beamformer, Max-SNIR Beamformer and the delay-and-sum beamformer. These beamformers are implemented with the fractional delay capability and then tested under reverberant conditions. This paper mainly concentrates on the effect of Reverberation on the outcome of a beamformer. A reverberant room environment is simulated and the effect of reverberation on an Elko’s beamformer output is compared with that of a Weiner beamformer. The size of the room is proportionate to the complexity of the implementation, i.e., as the size increases, so does the execution time of the algorithm. All these kinds of factors have been implemented and tested. Improvement in the Signal-to-Noise Ratio and PESQ scores of these beamformers with reverberant room and without reverberant room are noted. Results show that wiener beamformer has better average SNR and PESQ improvements both in reverberant and non-reverberant conditions with more microphones an, S/o P S N Murthy, Vizianagaram, 535003, India. Ph: 0046723362292 Ph: 00919492099315 (India)

All Speech Processing Equipments like noise cancelling headphones and Hearing aids should be able to filter different kinds of interfering signals and present a clear sound to the listener. The effect of these interfering signals varies depending on the surroundings. Confining the area of interest to a closed room, ranging from a class room to an auditorium, the most significant interference that is dominant in the received signal is the “Reverberation Effect”. Reverberation is the combined effect of multiple echoes from the walls of the room and its intensity depends on the size, shape, dimensions and materials used in the construction of the room. This phenomenon is tolerable for some purposes like music recordings but it predominantly effects the speech. Eliminating this effect without affecting the original speech is a challenge and a topic of research in these days. This paper is an excerpt of the collective work done on reducing microphone inaccuracies using various adaptive beamformers and emphasizes on-the Elko’s Beamformer, the Weiner Beamformer, Max-SNIR Beamformer and the delay-and-sum beamformer. These beamformers are implemented with the fractional delay capability and then tested under reverberant conditions. This paper mainly concentrates on the effect of Reverberation on the outcome of a beamformer. A reverberant room environment is simulated and the effect of reverberation on an Elko’s beamformer output is compared with that of a Weiner beamformer. The size of the room is proportionate to the complexity of the implementation, i.e., as the size increases, so does the execution time of the algorithm. All these kinds of factors have been implemented and tested. Improvement in the Signal-to-Noise Ratio and PESQ scores of these beamformers with reverberant room and without reverberant room are noted. Results show that wiener beamformer has better average SNR and PESQ improvements both in reverberant and non-reverberant conditions with more microphones an, S/o P S N Murthy, Vizianagaram, 535003, India. Ph: 0046723362292 Ph: 00919492099315 (India)

All Speech Processing Equipments like noise cancelling headphones and Hearing aids should be able to filter different kinds of interfering signals and present a clear sound to the listener. The effect of these interfering signals varies depending on the surroundings. Confining the area of interest to a closed room, ranging from a class room to an auditorium, the most significant interference that is dominant in the received signal is the “Reverberation Effect”. Reverberation is the combined effect of multiple echoes from the walls of the room and its intensity depends on the size, shape, dimensions and materials used in the construction of the room. This phenomenon is tolerable for some purposes like music recordings but it predominantly effects the speech. Eliminating this effect without affecting the original speech is a challenge and a topic of research in these days. This paper is an excerpt of the collective work done on reducing microphone inaccuracies using various adaptive beamformers and emphasizes on-the Elko’s Beamformer, the Weiner Beamformer, Max-SNIR Beamformer and the delay-and-sum beamformer. These beamformers are implemented with the fractional delay capability and then tested under reverberant conditions. This paper mainly concentrates on the effect of Reverberation on the outcome of a beamformer. A reverberant room environment is simulated and the effect of reverberation on an Elko’s beamformer output is compared with that of a Weiner beamformer. The size of the room is proportionate to the complexity of the implementation, i.e., as the size increases, so does the execution time of the algorithm. All these kinds of factors have been implemented and tested. Improvement in the Signal-to-Noise Ratio and PESQ scores of these beamformers with reverberant room and without reverberant room are noted. Results show that wiener beamformer has better average SNR and PESQ improvements both in reverberant and non-reverberant conditions with more microphones an, S/o P S N Murthy, Vizianagaram, 535003, India. Ph: 0046723362292 Ph: 00919492099315 (India)

All Speech Processing Equipments like noise cancelling headphones and Hearing aids should be able to filter different kinds of interfering signals and present a clear sound to the listener. The effect of these interfering signals varies depending on the surroundings. Confining the area of interest to a closed room, ranging from a class room to an auditorium, the most significant interference that is dominant in the received signal is the “Reverberation Effect”. Reverberation is the combined effect of multiple echoes from the walls of the room and its intensity depends on the size, shape, dimensions and materials used in the construction of the room. This phenomenon is tolerable for some purposes like music recordings but it predominantly effects the speech. Eliminating this effect without affecting the original speech is a challenge and a topic of research in these days. This paper is an excerpt of the collective work done on reducing microphone inaccuracies using various adaptive beamformers and emphasizes on-the Elko’s Beamformer, the Weiner Beamformer, Max-SNIR Beamformer and the delay-and-sum beamformer. These beamformers are implemented with the fractional delay capability and then tested under reverberant conditions. This paper mainly concentrates on the effect of Reverberation on the outcome of a beamformer. A reverberant room environment is simulated and the effect of reverberation on an Elko’s beamformer output is compared with that of a Weiner beamformer. The size of the room is proportionate to the complexity of the implementation, i.e., as the size increases, so does the execution time of the algorithm. All these kinds of factors have been implemented and tested. Improvement in the Signal-to-Noise Ratio and PESQ scores of these beamformers with reverberant room and without reverberant room are noted. Results show that wiener beamformer has better average SNR and PESQ improvements both in reverberant and non-reverberant conditions with more microphones an, S/o P S N Murthy, Vizianagaram, 535003, India. Ph: 0046723362292 Ph: 00919492099315 (India)

The acoustics of a concert hall, or any other room, are generally assessed by measuring room impulse responses for one or multiple source and receiver location(s). From these responses, objective parameters can be determined that should be related to various perceptual attributes of room acoustics. A set of these parameters is collected in ISO standard 3382. However, this method of assessing room acoustical quality has some major shortcomings. First of all, it is known that the perception of the acoustics of a room is dependent on the type of source signal. This is not taken into account when only impulse responses are considered. Furthermore, because of the type of test signals used to perform such a measurement, measurements are mostly carried out in empty rooms, while the acoustics can change drastically when a room is fully occupied with people. Finally, there is evidence in the literature of cases in which the parameters sometimes do not correlate well with perception. For example, it has been found that some parameters can fluctuate severely over small measurement intervals, whereas the perceptual attributes for which these parameters should be predictors remain constant. Apparently, some important properties of the human auditory system are not taken into account sufficiently. In this thesis, a new method is proposed. The method consists of the processing of arbitrary binaural audio recordings using a binaural, non-linear auditory model. These recordings (or simulations) should be performed with a dummy head. This model simulates the most important stages of the auditory system, such as the response of the inner ear, basilar membrane and hair cells, neural adaptation and binaural interaction. Using a peak detection algorithm, the output signals of the model are split into two separate streams: one related to the source (direct sound) and one related to the environment (reverberant sound). Together with the calculation of the amount of fluctuation in the Inter, Imaging Science and Technology, Applied Sciences

This thesis focuses on the mapping of the microphone array’s response to match the characteristics of a human subject’s Head-Related Transfer Function (HRTF). The mapping of the response is first explored with a ‘monaural HRTF matching’ that filters the response independent of the arrival angles. For arbitrary array geometry with the listener external to the acoustic, the monaural HRTF matching did not provide listeners with enough spatial information to precisely localize sound sources. To correct this, a preprocessor control algorithm was added to the HRTF matching, a ‘binaural HRTF matching’ process. The binaural HRTF matching increased the listeners’ performance in perceiving the location of a sound source. With the addition of simulated head movement, the listeners’ perception increased by 20%. An issue with this approach is the use of HRTFs other than the listeners’ measured HRTF, creating a psychoacoustic based error in localization, i.e., front/back confusion.

The acoustics of a concert hall, or any other room, are generally assessed by measuring room impulse responses for one or multiple source and receiver location(s). From these responses, objective parameters can be determined that should be related to various perceptual attributes of room acoustics. A set of these parameters is collected in ISO standard 3382. However, this method of assessing room acoustical quality has some major shortcomings. First of all, it is known that the perception of the acoustics of a room is dependent on the type of source signal. This is not taken into account when only impulse responses are considered. Furthermore, because of the type of test signals used to perform such a measurement, measurements are mostly carried out in empty rooms, while the acoustics can change drastically when a room is fully occupied with people. Finally, there is evidence in the literature of cases in which the parameters sometimes do not correlate well with perception. For example, it has been found that some parameters can fluctuate severely over small measurement intervals, whereas the perceptual attributes for which these parameters should be predictors remain constant. Apparently, some important properties of the human auditory system are not taken into account sufficiently. In this thesis, a new method is proposed. The method consists of the processing of arbitrary binaural audio recordings using a binaural, non-linear auditory model. These recordings (or simulations) should be performed with a dummy head. This model simulates the most important stages of the auditory system, such as the response of the inner ear, basilar membrane and hair cells, neural adaptation and binaural interaction. Using a peak detection algorithm, the output signals of the model are split into two separate streams: one related to the source (direct sound) and one related to the environment (reverberant sound). Together with the calculation of the amount of fluctuation in the Inter, Imaging Science and Technology, Applied Sciences

Bilateral amplification seems to be the best solution for bilaterally hearing-impaired persons. Nevertheless, some individuals are unsuccessful with this strategy. The goals of the present study were to develop tests to improve the diagnostic test battery before rehabilitation of hearing-impaired persons with bilateral or unilateral amplification, and to evaluate the tests with normally-hearing subjects and with two groups of hearing-impaired persons. The latter two groups contained 11 successful and 11 unsuccessful users of bilateral amplification respectively. Hearing thresholds, speech recognition in noise, signal analysis ability, binaural abilities, and dichotic tests were used in the investigation. The subjects answered a questionnaire and hearing aid gain curves were measured. The results for the two groups were similar for peripheral hearing functions and binaural performance. The unilateral amplification group showed significantly worse results in speech-in-noise and dichotic tests. Spatial aspects within the questionnaire were correlated to amplification preference. We therefore suggest the inclusion of speech-in-noise, dichotic tests, and questions on spatial orientation into the diagnostic scheme before rehabilitation with hearing aids.

Contains table of contents for Section 2, an introduction and reports on fifteen research projects., National Institutes of Health Grant RO1 DC00117, National Institutes of Health Grant RO1 DC02032, National Institutes of Health Contract P01-DC00361, National Institutes of Health Contract N01-DC22402, National Institutes of Health/National Institute on Deafness and Other Communication Disorders Grant 2 R01 DC00126, National Institutes of Health Grant 2 R01 DC00270, National Institutes of Health Contract N01 DC-5-2107, National Institutes of Health Grant 2 R01 DC00100, U.S. Navy - Office of Naval Research/Naval Air Warfare Center Contract N61339-94-C-0087, U.S. Navy - Office of Naval Research/Naval Air Warfare Center Contract N61339-95-K-0014, U.S. Navy - Office of Naval Research/Naval Air Warfare Center Grant N00014-93-1-1399, U.S. Navy - Office of Naval Research/Naval Air Warfare Center Grant N00014-94-1-1079, U.S. Navy - Office of Naval Research Subcontract 40167, U.S. Navy - Office of Naval Research Grant N00014-92-J-1814, National Institutes of Health Grant R01-NS33778, U.S. Navy - Office of Naval Research Grant N00014-88-K-0604, National Aeronautics and Space Administration Grant NCC 2-771, U.S. Air Force - Office of Scientific Research Grant F49620-94-1-0236, U.S. Air Force - Office of Scientific Research Agreement with Brandeis University

Contains research objectives and summary of research on five research projects, with ten sub-topics., National Institutes of Health (Grant 1 RO1 NS10916-01), National Institutes of Health (Grant 5 RO1 NS11000-03), National Institutes of Health (Grant 1 RO1 NS11153-01), Harvard-M.I.T. Rehabilitation Engineering Center, U. S. Department of Health, Education, and Welfare (Grant 23-P-55854), National Institutes of Health (Grant 1 RO1 NS11680-01), National Institutes of Health (Grant 5 ROI NS11080-02), M.I.T. Health Sciences Fund, National Aeronautics and Space Administration (Grant NSG-2032), National Institutes of Health (Grant 5 TO1 GM01555-09), Massachusetts General Hospital Purchase Order F63853, Boston City Hospital Purchase Order 4338-75431

The present study developed simple and rapid school-based binaural hearing tests as part of an experimental assessment for Central Auditory Processing Disorder (CAPD), and investigated the outcomes in relation to socioeconomic status. The investigations were carried out in a population study on 256 primary school children from fourteen schools within the Perth Metropolitan area. It was hypothesized that the tests results would show a significant association with socioeconomic level based on the reported association between CAPD and history of otitis media with effusion (OME), a disease known to be prevalent in disadvantaged populations. A significant association would raise concern of an underlying physiological auditory problem for the educational and social-emotional issues often reported in this population. The tests of binaural hearing used in this study were subjective measure of directional hearing (DH) ability and the objective measure of binaural interaction using the postauricular muscle response (PAMR). Both of these tests have been shown to be sensitive to identifying subjects diagnosed with CAPD in a laboratory set-up (Zakaria, 2007). The tests were adapted for use within the school environment and implemented in a protocol that allowed for large number of children to be tested in a relatively short amount of time. This was done by incorporating the use of inexpensive MP3 players, simple score sheets and automatic video marking for the lateralization test; and custom-made portable two-channel equipment for the PAMR test., Significant associations with socioeconomic status was found for both the experimental binaural hearing tests, as well as for the standard CAPD test carried out in this study, although only weak correlations (r = 0.16 to 0.47 ) were found between the tests themselves. This was not unexpected, considering the heterogeneity of the disorder and the many related issues in CAPD assessment. Despite the weak correlations, significant differences were found between normal and at-risk-for-CAPD subjects especially for the PAMR test, using the ratio measure of the binaural response over the sum of the contralateral and the ipsilateral responses (T = 2.91, p = 0.004). With further optimization of the test protocol, the two simple binaural hearing tests may prove useful as part of a comprehensive clinical assessment for the disorder., Thesis (Ph.D.)--University of Western Australia, 2011

Contains table of contents for Section 2, an introduction and reports on fifteen research projects., National Institutes of Health Grant RO1 DC00117, National Institutes of Health Grant RO1 DC02032, National Institutes of Health Contract P01-DC00361, National Institutes of Health Contract N01-DC22402, National Institutes of Health/National Institute on Deafness and Other Communication Disorders Grant 2 R01 DC00126, National Institutes of Health Grant 2 R01 DC00270, National Institutes of Health Contract N01 DC-5-2107, National Institutes of Health Grant 2 R01 DC00100, U.S. Navy - Office of Naval Research/Naval Air Warfare Center Contract N61339-94-C-0087, U.S. Navy - Office of Naval Research/Naval Air Warfare Center Contract N61339-95-K-0014, U.S. Navy - Office of Naval Research/Naval Air Warfare Center Grant N00014-93-1-1399, U.S. Navy - Office of Naval Research/Naval Air Warfare Center Grant N00014-94-1-1079, U.S. Navy - Office of Naval Research Subcontract 40167, U.S. Navy - Office of Naval Research Grant N00014-92-J-1814, National Institutes of Health Grant R01-NS33778, U.S. Navy - Office of Naval Research Grant N00014-88-K-0604, National Aeronautics and Space Administration Grant NCC 2-771, U.S. Air Force - Office of Scientific Research Grant F49620-94-1-0236, U.S. Air Force - Office of Scientific Research Agreement with Brandeis University

Humans are highly sensitive to Interaural Time Differences (ITDs) in stimuli presented via headphones. For broadband noise stimuli of long durations, ITD detection thresholds can be as low as 10 to 15 ms. When the stimulus duration is shortened, thresholds increase by about a factor 2 for a tenfold decrease in duration. ITD thresholds also increase, when the probe carrying an ITD is surrounded by diotic fringes. When a 5-ms probe is combined with preceding or trailing fringes, the effect of a fringe preceding the probe is stronger than that of a trailing fringe for fringe durations <35 ms. The effect of fringes surrounding the probe is equal to the addition of the effects of the individual fringes. In this contribution, we present behavioral data for the same experimental condition, called dynamically varying ITD detection, but for a wider range of probe and fringe durations. Probe durations varied between 5 and 400 ms, and fringe durations had values of 5, 20, 100 or 200 ms. In contrast to earlier findings, we observed for most duration combinations a stronger effect of the trailing fringe than of the preceding fringe. For these configurations, the effect of surrounding fringes was dominated by the trailing fringe. Only for the combination of 5-ms fringes with 5-ms probes did we see the clear dominance of the preceding fringe. These results are not easy to align with the concept of onset emphasis often used to explain binaural localization data for short stimuli. In fact the data seem to be difficult to predict with a purely signal-driven model of perception and thus form an interesting challenge for modeling human localization.

Humans are highly sensitive to Interaural Time Differences (ITDs) in stimuli presented via headphones. For broadband noise stimuli of long durations, ITD detection thresholds can be as low as 10 to 15 ms. When the stimulus duration is shortened, thresholds increase by about a factor 2 for a tenfold decrease in duration. ITD thresholds also increase, when the probe carrying an ITD is surrounded by diotic fringes. When a 5-ms probe is combined with preceding or trailing fringes, the effect of a fringe preceding the probe is stronger than that of a trailing fringe for fringe durations <35 ms. The effect of fringes surrounding the probe is equal to the addition of the effects of the individual fringes. In this contribution, we present behavioral data for the same experimental condition, called dynamically varying ITD detection, but for a wider range of probe and fringe durations. Probe durations varied between 5 and 400 ms, and fringe durations had values of 5, 20, 100 or 200 ms. In contrast to earlier findings, we observed for most duration combinations a stronger effect of the trailing fringe than of the preceding fringe. For these configurations, the effect of surrounding fringes was dominated by the trailing fringe. Only for the combination of 5-ms fringes with 5-ms probes did we see the clear dominance of the preceding fringe. These results are not easy to align with the concept of onset emphasis often used to explain binaural localization data for short stimuli. In fact the data seem to be difficult to predict with a purely signal-driven model of perception and thus form an interesting challenge for modeling human localization.

The purpose of this project is to desig a systematical method in order to measure human directionality ability in horizontal plane with a single sound source. A completely virtual auditory model has been created in Matlab. The project consists of modeling binaural cues, designing digital filters, designing a test workbench, measuring listener's directionality and analyzing the data. The head related transfer function (HRTF) is computed by calculating the two most important binaural cues, interaural level difference (ILD) and interaural time difference (ITD). The platform is made in Matlab and all results have been shown by plots produced from Matlab code. The directionality test has been done with real human subjects and the results have been analyzed and presented.

The purpose of this project is to desig a systematical method in order to measure human directionality ability in horizontal plane with a single sound source. A completely virtual auditory model has been created in Matlab. The project consists of modeling binaural cues, designing digital filters, designing a test workbench, measuring listener's directionality and analyzing the data. The head related transfer function (HRTF) is computed by calculating the two most important binaural cues, interaural level difference (ILD) and interaural time difference (ITD). The platform is made in Matlab and all results have been shown by plots produced from Matlab code. The directionality test has been done with real human subjects and the results have been analyzed and presented.

The purpose of this project is to desig a systematical method in order to measure human directionality ability in horizontal plane with a single sound source. A completely virtual auditory model has been created in Matlab. The project consists of modeling binaural cues, designing digital filters, designing a test workbench, measuring listener's directionality and analyzing the data. The head related transfer function (HRTF) is computed by calculating the two most important binaural cues, interaural level difference (ILD) and interaural time difference (ITD). The platform is made in Matlab and all results have been shown by plots produced from Matlab code. The directionality test has been done with real human subjects and the results have been analyzed and presented.

The purpose of this project is to desig a systematical method in order to measure human directionality ability in horizontal plane with a single sound source. A completely virtual auditory model has been created in Matlab. The project consists of modeling binaural cues, designing digital filters, designing a test workbench, measuring listener's directionality and analyzing the data. The head related transfer function (HRTF) is computed by calculating the two most important binaural cues, interaural level difference (ILD) and interaural time difference (ITD). The platform is made in Matlab and all results have been shown by plots produced from Matlab code. The directionality test has been done with real human subjects and the results have been analyzed and presented.

The purpose of this project is to desig a systematical method in order to measure human directionality ability in horizontal plane with a single sound source. A completely virtual auditory model has been created in Matlab. The project consists of modeling binaural cues, designing digital filters, designing a test workbench, measuring listener's directionality and analyzing the data. The head related transfer function (HRTF) is computed by calculating the two most important binaural cues, interaural level difference (ILD) and interaural time difference (ITD). The platform is made in Matlab and all results have been shown by plots produced from Matlab code. The directionality test has been done with real human subjects and the results have been analyzed and presented.

The purpose of this project is to desig a systematical method in order to measure human directionality ability in horizontal plane with a single sound source. A completely virtual auditory model has been created in Matlab. The project consists of modeling binaural cues, designing digital filters, designing a test workbench, measuring listener's directionality and analyzing the data. The head related transfer function (HRTF) is computed by calculating the two most important binaural cues, interaural level difference (ILD) and interaural time difference (ITD). The platform is made in Matlab and all results have been shown by plots produced from Matlab code. The directionality test has been done with real human subjects and the results have been analyzed and presented.

In complex acoustic environments including command and control settings multiple acoustic sources vie for attention. The ability to locate and understand each acoustic signal is degraded by the presence of reverberant energy as well as the presence of competing sound sources. This study explored the effects of echoic energy on the ability to segregate analyze and localize communication signals. Results demonstrate that the human auditory system is able to cope with many of the degradations caused by room acoustics. When selectively attending to one sound in a mixture of sound the greatest perceptual contribution of spatially separating competing sound sources is in enhancing sound source segregation and allowing selective attention to be directed effectively. In anechoic space any salient feature that differentiates target from masker is sufficient to enable accurate segregation and selective attention. However features that would be redundant in anechoic space yield additive gains in reverberant space. Directional localization accuracy is degraded by reverberation while distance accuracy is enhanced. Collaborations with neurophysiologists and computational researchers lend insights into the mechanisms that contribute to how listeners process sound in complex reverberant environments.

OBJECTIVE: The Glasgow Benefit Plot (GBP) is a graph showing the pure-tone thresholds preoperatively and postoperatively, and is meant to visualize the possible outcome of binaural hearing for individual patients. We used our database comprising a large number of patient data to test the usefulness of the GBP. STUDY DESIGN: Evaluation based on retrospective clinical data. SETTING: Tertiary referral center, University clinic. MAIN OUTCOME MEASURES: Audiometric assessment included bilateral preoperative and postoperative evaluation using conventional audiometry. Three preoperative groups (unilateral, asymmetric, and symmetric hearing impairment) were split into 6 postoperative categories in the GBP diagram. RESULTS: Considering the 509 operations, including 34 bilateral operations, the total outcome was distributed as follows: 34% resulted in bilateral normal hearing, 24% unilateral normal hearing (in operated ear), 14% still unilateral hearing impairment (in operated ear), 13% symmetric hearing impairment, 10% asymmetric hearing impairment (operated ear best), 5% still asymmetric hearing impairment (nonoperated ear best). The distribution of outcomes depended in part on the bone conduction level in the operated ear and in part on the hearing status in the contralateral ear. CONCLUSION: The GBP is a useful instrument that provides a means for judging the binaural hearing status. However, the outcome of stapes surgery for individual persons and for a group depends critically on the preoperative audiometric criteria for patients who are chosen for surgery. In cases of depressed bone conductions, the method does not indicate the upper limit for possible hearing improvement. A number of patients with combined hearing impairments were included in the present study population. In preoperative counseling, the GBP must be complemented with information with regard to the limitation of possible hearing improvement owing to the individual bone conduction level.

When a target speech signal is obscured by an interfering speech wave form, comprehension of the target message depends both on the successful detection of the energy from the target speech wave form and on the successful extraction and recognition of the spectro-temporal energy pattern of the target out of a background of acoustically similar masker sounds. This study attempted to isolate the effects that energetic masking, defined as the loss of detectable target information due to the spectral overlap of the target and masking signals, has on multi-talker speech perception. This was achieved through the use of ideal time-frequency binary masks that retained those spectro-temporal regions of the acoustic mixture that were dominated by the target speech but eliminated those regions that were dominated by the interfering speech. The results suggest that energetic masking plays a relatively small role in the overall masking that occurs when speech is masked by interfering speech but a much more significant role when speech is masked by interfering noise., Prepared in collaboration with Department of Computer Science and Engineering and Center for Cognitive Science, Ohio State Univ., Columbus, OH. Sponsored in part by Air Force Office of Scientific Research (AFOSR). Published in the Journal of the Acoustical Society of America, v120 n6 p4007-4018, Dec 2006.

Purpose: Speech perception in participants with auditory neuropathy (AN) was systematically studied to answer the following 2 questions: Does noise present a particular problem for people with AN? Can clear speech and cochlear implants alleviate this problem? Method: The researchers evaluated the advantage in intelligibility of clear speech over conversational speech in 13 participants with AN. Of these participants, 7 had received a cochlear implant. Eight sentence-recognition experiments were conducted to examine the clear speech advantage in 2 listening conditions (quiet and noise) using 4 stimulation modes (monaural acoustic, diotic acoustic, monaural electric, and binaurally combined acoustic and electric stimulation). Results: Participants with AN performed more poorly in speech recognition in noise than did the normal-hearing, cochlear-impaired, and cochlear implant controls. A significant clear speech advantage was observed, ranging from 9 to 23 percentage points in intelligibility for all listening conditions and stimulation modes. Electric stimulation via a cochlear implant produced significantly higher intelligibility than acoustic stimulation in both quiet and in noise. Binaural hearing with either diotic acoustic stimulation or combined acoustic and electric stimulation produced significantly higher intelligibility than monaural stimulation in quiet but not in noise. Conclusions: Participants with AN most likely derive the clear speech advantage from enhanced temporal properties in clear speech and improved neural synchrony with electric stimulation. Although the present result supports cochlear implantation as one treatment choice for people with AN, it suggests that the use of innovative hearing aids may be another viable option to improve speech perception in noise.

Purpose: Speech perception in participants with auditory neuropathy (AN) was systematically studied to answer the following 2 questions: Does noise present a particular problem for people with AN? Can clear speech and cochlear implants alleviate this problem? Method: The researchers evaluated the advantage in intelligibility of clear speech over conversational speech in 13 participants with AN. Of these participants, 7 had received a cochlear implant. Eight sentence-recognition experiments were conducted to examine the clear speech advantage in 2 listening conditions (quiet and noise) using 4 stimulation modes (monaural acoustic, diotic acoustic, monaural electric, and binaurally combined acoustic and electric stimulation). Results: Participants with AN performed more poorly in speech recognition in noise than did the normal-hearing, cochlear-impaired, and cochlear implant controls. A significant clear speech advantage was observed, ranging from 9 to 23 percentage points in intelligibility for all listening conditions and stimulation modes. Electric stimulation via a cochlear implant produced significantly higher intelligibility than acoustic stimulation in both quiet and in noise. Binaural hearing with either diotic acoustic stimulation or combined acoustic and electric stimulation produced significantly higher intelligibility than monaural stimulation in quiet but not in noise. Conclusions: Participants with AN most likely derive the clear speech advantage from enhanced temporal properties in clear speech and improved neural synchrony with electric stimulation. Although the present result supports cochlear implantation as one treatment choice for people with AN, it suggests that the use of innovative hearing aids may be another viable option to improve speech perception in noise.

Over the past two decades, the fascinating possibilities of cochlear implants for congenitally deaf or deafened children and adults developed tremendously and created a rapidly developing interdisciplinary research field.The main advancements of cochlear implantation in the past decade are marked by significant improvement of hearing and speech understanding in CI users. These improvements are attributed to the enhancement of speech coding strategies.The Implantation of more (and increasingly younger) children as well as the possibilities of the restoration of binaural hearing abilities with cochlear implants reflect the high standards reached by this development. Despite this progress, modern cochlear implants do not yet enable normal speech understanding, not even for the best patients. In particular speech understanding in noise remains problematic . Until the mid 1990ies research concentrated on unilateral implantation. Remarkable and effective improvements have been made with bilateral implantation since 1996. Nowadays an increasing numbers of patients enjoy these benefits.

Over the past two decades, the fascinating possibilities of cochlear implants for congenitally deaf or deafened children and adults developed tremendously and created a rapidly developing interdisciplinary research field.The main advancements of cochlear implantation in the past decade are marked by significant improvement of hearing and speech understanding in CI users. These improvements are attributed to the enhancement of speech coding strategies.The Implantation of more (and increasingly younger) children as well as the possibilities of the restoration of binaural hearing abilities with cochlear implants reflect the high standards reached by this development. Despite this progress, modern cochlear implants do not yet enable normal speech understanding, not even for the best patients. In particular speech understanding in noise remains problematic . Until the mid 1990ies research concentrated on unilateral implantation. Remarkable and effective improvements have been made with bilateral implantation since 1996. Nowadays an increasing numbers of patients enjoy these benefits.

The illusion of auditory apparent motion (AAM) was examined in order to determine the burst durations and interstimulus onset intervals (ISOIs) at which AAM is heard when spatial information regarding source location was varied. In the first experiment AAM was examined in the free field under monaural and binaural listening conditions. AAM was heard at the same burst duration-ISOI combinations for both listening conditions, but the location of the lead source could be determined only under binaural listening. In the second experiment AAM was measured with two and three sound sources. The number of sources did not affect the burst duration-ISOI combinations that produced AAM, but did affect the determination of the location of the lead source. In the third experiment AAM was tested when the sources were located in the median plane. The sources were located either at 0 degrees and 180 degrees azimuth, or both at 0 degrees azimuth, one in the horizontal plane and one 20 degrees above. The location of the speakers did not affect the timing requirements for the perception of AAM, only the timing requirements for the detection of the lead source. In the fourth experiment, AAM was measured when the vertical separation between the sources was either 2.5 degrees or 20 degrees. AAM was heard at both separations, even though 2.5 degrees is less than the vertical MAA. In each of these experiments only burst duration and ISOI determined whether motion was heard. Localization cues were important only for the determination of the direction of motion. Copyright 1994 Acoustical Society of America

Characterization of the function of single neurons has become essential for developing advanced nodes for massively parallel computational structures. The lateral superior olive (LSO) is a nucleus located early in the ascending auditory pathway. Anatomically, each LSO neuron receives input from both ears; this fact led researchers to hypothesize that the LSO is involved in binaural hearing. Recordings from this nucleus not only confirm that LSO neurons are responsive to sounds presented in either ear, but in a particularly interesting and simple way: sounds presented in one ear excite an LSO neuron (increase its discharge rate) while sounds present at the other ear inhibit the neuron (discharge rate decreases). This mode of operation-the interaction of excitatory an inhibitory inputs to produce neural output-is the fundamental mode of neural processing. Thus, modeling LSO neurons represents a particularly clear-cut opportunity to understand basic neural processing. The approach taken in this research is to create anatomical and biophysical models of single LSO neurons and demand that the discharge patterns thus produced statistically match single-neuron recordings. With this strategy, this project seeks to (1) understand the neural mechanisms that underly basic LSO response patterns (transient chopping response to tone bursts and serial interspike interval correlations), (2) understand the transformation of fractal inputs by the LSO neuron, and (3) characterize excitatory/inhibitory interactions.

With 253 normal-hearing Navy enlisted men examined in groups of 20 or fewer, speech reception was tested on a sentence intelligibility test masked by voices coming seemingly from R and L (by the stereophonic illusion; this is the well-known 'cocktail party' effect). Performance was not affected until the input to one ear was reduced by 25+ dB re the other ear. This rejection criterion of 25+ dB was corroborated on 92 patients with audiometric asymmetries. If this criterion is also found on other binaural tasks now being considered in this Laboratory, it will support a recommendation to exclude those with such asymmetries from certain Navy duties.

Theory and some empirical evidence are cited which predict that humans should exhibit little or no ability to localize sounds underwater; however, other evidence is presented also which seems to contradict this position. In order to provide relevant data on the issue, a pilot study was conducted with results suggesting that man can localize sounds underwater--at least to some degree. Accordingly, a series of experiments were carried out; the first is reported. It utilized 17 divers who were free to move their heads but not their bodies in an underwater localization task. In order to allow precision in the experiment, a Diver Auditory Localization System (DALS) was developed. DALS consists of an open polyvinyl chloride (PVC) framework to which five 8.5-foot arms and other equipment can be attached. The five arms allow placement--at ear level--of underwater projectors at angles (to the diver) of 0, 45, 90, 27 and 315 deg. Subjects responded to four different signals--250, 1000, 6000 Hz sinusoids and thermal noise--at an SPL of 110 dB (approximately 40 dB re: underwater hearing thresholds) by means of a specially constructed five- position underwater switch coupled to an IBM key punch on the surface. The obtained correct mean scores were (1) 250 Hz: 51%; (2) 1 kHz: 39%; (3) 6 kHz: 33%; (4) Noise: 52% and (5) Overall: 44%. These data demonstrate that humans are capable of at least some primitive sound localization underwater; mechanisms are postulated that may explain this ability., Report on Underwater Speech Communication. See also Progress rept. no. 21, AD716793.