Dynamic-range compression is used in hearing devices to reduce the wide range of environmental sound levels into a range better suited to the capability of the impaired ear. Its use is motivated by the fact that the healthy ear itself performs this function, but this natural compression is typically reduced or lost with sensorineural hearing loss. This thesis explores how dynamic-range compression influences aspects of spatial hearing that play an important role in everyday listening. Spatial hearing largely relies on comparing information from the two ears. The first two experiments investigated how spatial perception is affected when compression is applied independently at each ear, as occurs in traditional bilateral hearing-device fittings. This was found to have a variety of possib le adverse effects, such as altering the perceived position of sounds and making them appear more spatially diffuse. The effects are explained in terms of changes to the underlying acoustic cues. Some modern hearing devices incorporate a wireless link, allowing compression to be synchronized across the ears. The third experiment investigated how this might provide an advantage when listening to speech in the presence of a spatially separated noise. It was found that a small to moderate benefit was obtained, compared to unlinked compression, and that this was realized th rough changes to the monaural signal at the ear that had the more favourable ratio of speech-to-noise energy. The fourth experiment tested whether the natural compression that occurs within the healthy cochlea directly affects the use of the relative level difference between the two ears as a spatial cue. Contrary to the experimental hypothesis, it was found that the potency of this cue changes little as the overall sound intensity is varied over a wide range, raising interesting questions about how this cue is evaluated at a neural level.

Tone burst-evoked otoacoustic emissions (TBOAEs) may be reduced in amplitude (suppressed) by the simultaneous presentation of additional tone bursts. This simultaneous suppression of TBOAEs has not previously been studied in detail and the mechanisms responsible for it are not well understood. The literature suggests a number of possible underpinning mechanisms, including mechanisms involving TBOAE components generated at basal basilar membrane (BM)sites (i.e. basalsource components). However, the weight of evidence indicates that a mechanism wherein suppression caused by local nonlinear interactions (LNI) between BM vibration patterns, governed by the compressive nonlinearity of TBOAE generator channels, is primarily responsible. The aim of the work described in this thesis is to determine the extent to which this LNI-based mechanism can account for simultaneous suppression of TBOAEs. This was achieved by . testing four hypotheses of suppression behaviour derived from a simple mathematical model of the LNI-based mechanism. Each of these hypotheses was tested via a specific experiment, where suppression was measured from normal human ears. It was reasoned that a close agreement between the hypothesised behaviours and the suppression data measured from human ears could be held as support for the LNI-based mechanism being responsible for suppression. In contrast, any substantial differences might argue against the LNI-based mechanism. The results of the four experiments demonstrated some agreement between suppression data and the model-derived hypotheses, though some discrepancies were also observed. It was therefore reasoned that the LNI-based mechanism (as represented in the simple model) cannot account for all the behaviours exhibited by the suppression data. It was argued that some of the discrepancies could be understood in terms of limitations of the model of the LNI-based mechanism.

A community-based nationwide survey for hearing loss was conducted in Oman in between 1996 and 1997. Audiometric tests and ear examinations were conducted for 12,400 persons in phase I. In phase 11, otologists examined the hearing-impaired subjects to determine the cause. The prevalence of bilateral hearing impairment was 55/1000 (95% Cl 51.08-59.47). The prevalence of bilateral disabling hearing loss was 2111000 (95% Cl 18.07-23.29). Register of Omani paediatric cases with severe to profound deafness: The study includes a retrospective analysis of causes of deafness in 1400 Omani children who were detected to be suffering frorri severe to profound levels of hearing impairment. A standard form was used to collect various details from the otolaryngologists all over the country. The time period included is from 1986 to 2000. The rate of consanguineous marriage in the parents of the affected children was 70%, and 67.8% of these children had a sibling suffering from hearing impairment. Consanguinity and deafness in the Omani paediatric population: It was found that 70% of the deaf children were from parents of consanguineous marriages, and 30% from non-consanguineous unions. In those with consanguineous families 70.2% were first cousin marriages, 17.5% were second cousins, and 10.9% were from the same tribe. The 14 • proportion arising from first cousin marriages was higher than the background rate of first cousin marriages in Oman. Study to look for the GJB2 gene mutation in Omani population: This study was done using both PCR-RFLP and direct DNA sequencing methods. Two common GJB2 gene mutations (35de1G and 167de1T) were screened in 280 healthy controls and 95 deaf patients using two different PCR-RFLP methods. None of the samples studied, either by RFLP or sequencing, revealed any deafness-associated mutations in the coding region of the GJB2 gene. 15 •

The aim of the experiments described within this thesis was to measure binaural temporal and spectral resolution. Previous investigations that have studied temporal resolution (e.g. Bernstein et al., 2001) assumed that interfering noise dilutes delayed noise within the temporal window. The first two experiments described in this thesis have validated the dilution concept for correlated interfering noise, but not for uncorrelated interfering noise, the presence of which has a more detrimental effect than interfering correlated noise. The study by Bernstein et al. (2001) suggested that the equivalent rectangular bandwidth (ERD) of the binaural temporal window is considerably smaller than estimates made in previous studies (e.g. Kollmeier and Gilkey, 1990 Culling and Summerfield, 1998). The results from the experiments in this thesis disagree with those of Bernstein et al., and suggest that several factors led to their findings, including lack of control over the coherence of the stimulus due to the use of a detection task, the short duration of their stimuli, and the use of diotic interfering noise. The ERD of the binaural temporal window was found to range from 110-349 ms across listeners, a finding consistent with binaural sluggishness. In the frequency domain, a study by Sondhi and Guttman (1966) that investigated the frequency selectivity of the binaural system found evidence suggesting that binaural auditory filters are substantially wider than monaural auditory filters. Conversely, Kohlrausch (1988) measured auditory filters that were comparable to monaural filters. The results from the experiment conducted in this thesis found that binaural auditory filters are substantially wider than monaural auditory filters. Best fits were found to be 2-parameter asymmetric Gaussian filters with an ERB that ranged from 99-198 Hz at a centre frequency (CF) of 250 Hz, 138-215 Hz at a CF of 500 Hz, and 229-285 Hz at a CF of 750 Hz.

No description available.

Tinnitus affects 10-15% of the population and causes a diminished quality of life for 1-3% of people. Without a clear understanding of the mechanisms that generate and sustain tinnitus, treatment plans can only attempt to manage the problem rather than address the underlying causes of tinnitus. As 85% of tinnitus sufferers have hearing loss, tinnitus was originally considered to be a problem of the ear. Moreover the remaining tinnitus sufferers are postulated to have hidden hearing loss. Subsequent research has found that tinnitus is a problem extending beyond the ear. Experimental findings have identified tinnitus related activity throughout the auditory system. It is unclear how this activity is created and how it leads to the perception of a phantom sound. This thesis aims to identify the factors involved in the development of tinnitus related activity and the manifestation of a phantom sound; computational models of the auditory system are developed to address this primary objective. In this thesis, a biologically inspired model of the auditory periphery is created, called the peripheral model, which emulates tinnitus related activity in the auditory brainstem and accurately models hidden hearing loss in line with empirical data. The peripheral model is extended to model correlates of tinnitus associated with the thalamocortical network. Finally a perceptual model of tinnitus is developed using a Linear Mixed Effects (LME) approach to show how tinnitus related activity leads to the perception of a phantom sound. The outcomes from the development of the peripheral model include an accurate model of cochlear synaptopathy to replicate hidden hearing loss as well as the finding that hidden hearing loss can instigate adaptive changes that result in tinni­tus related activity. Extending the peripheral model to include the thalamocortical network led to the discovery that tinnitus requires changes to both the bottom-up and top-down signals in the auditory system. The result provides a significant step forward towards understanding the mechanisms underpinning tinnitus related activity. The peripheral model and the result from the investigation of the thalamocortical network provide the underpinning basis upon which to develop the perceptual model of tinnitus using a LME approach. The LME model provides a state of the art perceptual model of tinnitus that accurately models characteristics of tinnitus such as pitch and loudness. The results provide a significant advance­ment in the understanding of tinnitus generation and the evidence to motivate and direct further studies, which could lead to improved treatment methods for this condition.

The modern cochlear implant is one of the most successful neural stimulation devices, which partially mimics the workings of the auditory periphery. In the last few decades it has created a paradigm shift in hearing restoration of the deaf population, which has led to more than 324,000 cochlear implant users today. Despite its great success there is great disparity in patient outcomes without clear understanding of the aetiology of this variance in implant performance. Furthermore speech recognition in adverse conditions or music appreciation is still not attainable with today's commercial technology. This motivates the research for the next generation of cochlear implants that takes advantage of recent developments in electronics, neuroscience, nanotechnology, micro-mechanics, polymer chemistry and molecular biology to deliver high fidelity sound. The main difficulties in determining the root of the problem in the cases where the cochlear implant does not perform well are two fold: first there is not a clear paradigm on how the electrical stimulation is perceived as sound by the brain, and second there is limited understanding on the plasticity effects, or learning, of the brain in response to electrical stimulation. These significant knowledge limitations impede the design of novel cochlear implant technologies, as the technical specifications that can lead to better performing implants remain undefined. The motivation of the work presented in this thesis is to compare and contrast the cochlear implant neural stimulation with the operation of the physiological healthy auditory periphery up to the level of the auditory nerve. As such design of novel cochlear implant systems can become feasible by gaining insight on the question 'how well does a specific cochlear implant system approximate the healthy auditory periphery?' circumventing the necessity of complete understanding of the brain's comprehension of patterned electrical stimulation delivered from a generic cochlear implant device. A computational model, termed Digital Cochlea Stimulation and Evaluation Tool ('DiCoStET') has been developed to provide an objective estimate of cochlear implant performance based on neuronal activation measures, such as vector strength and average activation. A patient-specific cochlea 3D geometry is generated using a model derived by a single anatomical measurement from a patient, using non-invasive high resolution computed tomography (HRCT), and anatomically invariant human metrics and relations. Human measurements of the neuron route within the inner ear enable an innervation pattern to be modelled which joins the space from the organ of Corti to the spiral ganglion subsequently descending into the auditory nerve bundle. An electrode is inserted in the cochlea at a depth that is determined by the user of the tool. The geometric relation between the stimulation sites on the electrode and the spiral ganglion are used to estimate an activating function that will be unique for the specific patient's cochlear shape and electrode placement. This 'transfer function', so to speak, between electrode and spiral ganglion serves as a 'digital patient' for validating novel cochlear implant systems. The novel computational tool is intended for use by bioengineers, surgeons, audiologists and neuroscientists alike. In addition to 'DiCoStET' a second computational model is presented in this thesis aiming at enhancing the understanding of the physiological mechanisms of hearing, specifically the workings of the auditory synapse. The purpose of this model is to provide insight on the sound encoding mechanisms of the synapse. A hypothetical mechanism is suggested in the release of neurotransmitter vesicles that permits the auditory synapse to encode temporal patterns of sound separately from sound intensity. DiCoStET was used to examine the performance of two different types of filters used for spectral analysis in the cochlear implant system, the Gammatone type filter and the Butterworth type filter. The model outputs suggest that the Gammatone type filter performs better than the Butterworth type filter. Furthermore two stimulation strategies, the Continuous Interleaved Stimulation (CIS) and Asynchronous Interleaved Stimulation (AIS) have been compared. The estimated neuronal stimulation spatiotemporal patterns for each strategy suggest that the overall stimulation pattern is not greatly affected by the temporal sequence change. However the finer detail of neuronal activation is different between the two strategies, and when compared to healthy neuronal activation patterns the conjecture is made that the sequential stimulation of CIS hinders the transmission of sound fine structure information to the brain. The effect of the two models developed is the feasibility of collaborative work emanating from various disciplines; especially electrical engineering, auditory physiology and neuroscience for the development of novel cochlear implant systems. This is achieved by using the concept of a 'digital patient' whose artificial neuronal activation is compared to a healthy scenario in a computationally efficient manner to allow practical simulation times.

Cochlear implant (CI) users regularly perform as well as normal-hearing (NH) listeners in quiet conditions. However, CI users have reduced speech perception in noise. CI users suffer more in terms of speech intelligibility than NH listeners in the same noisy environment. Speech coding strategies with noise reduction algorithms for CI devices play an important role, allowing CI users to benefit more from their implants. This thesis investigates a wavelet packet-based speech coding strategy with envelope-based noise reduction algorithms to enhance speech intelligibility in noisy conditions. The advantages of wavelet packet transforms (WPTs), in terms of time-frequency analysis, the sparseness property, and low computational complexity, might make WPT appropriate for speech coding and denoising in CI devices. In cases with an optimal set of parameters for a wavelet packet-based speech coding strategy, the 23- and 64-band WPTs with sym8 and frame length of 8 ms were found to be more suitable than other combinations for this strategy. These parameters can optimise speech intelligibility to benefit CI users. However, both the standard ACE strategy and the wavelet packet-based strategy provided almost the same results in either quiet or noisy conditions. Cases using envelope-based denoising techniques in a wavelet packet-based strategy, namely time-adaptive wavelet thresholding (TAWT) and time-frequency spectral subtraction (TFSS) were developed and evaluated by objective and subjective intelligibility measures and compared to ideal binary masking (IdBM) as a baseline for denoising performance. IdBM can restore intelligibility to nearly the same level as NH listeners in all noisy conditions. Both TAWT and TFSS showed slight intelligibility improvements in some noisy conditions. This may result from noise estimation in denoising techniques. Noise level may be under- or overestimated, and this results in distortion in enhanced speech and difficult in speech discrimination. Both objective and subjective intelligibility measures can predict the trend of the performance of different denoising techniques for CI users. However, NH listeners can achieve better intelligibility at higher SNR levels without noise reduction, since they are less sensitive to noise but more sensitive to speech distortion when compared to CI listeners. Therefore, denoising techniques may work well for CI users in further investigations.

The human brain is regarded as an ensemble of dynamic systems in which communication between neural centres is very important. In order to perceive sounds many different cortical and subcortical brain areas have to coordinate their activity. After hearing loss, the connections and information pathways between these areas may rearrange and this may be one of the reasons for unsatisfactory speech perception after cochlear implantation (CI). It remains unclear how the brain connectivity and its re-organisation contribute to this, and this provides the motivation for the current study. The brain organisation can be quantified by connectivity measures, which may give the strength, direction, and timing information on the connections between brain areas. This research project aims to assess different methods of brain connectivity and response detection in the Electroencephalogram (EEG) and use these to investigate brain responses during tone, word, and sentence perception in normal hearing adults and CI users. The initial focus of this project was Dynamic Causal Modelling connectivity method, but early results raised questions regarding its reliability. DCM was then replaced by simpler and more established linear multi-variate auto regressive (MVAR) based models such as Coherence, Directed Transfer Function (DTF), and Partial Directed Coherence (PDC), as well as classical non-parametric power-spectral and coherence analysis. Both latter approaches could find changes in the brain activity in different time-frequency windows after the stimulus onset, which depended on the stimulus type and electrode positions. MVAR-based models were then employed and showed promising results when applied on synthetic data but on recorded data only model-based coherence (both pair-wise and multi-channel models) was able to detect connectivity changes in response to the stimuli presented to normal hearing participants; DTF and PDC appeared insufficiently sensitive. Different artefact rejection methods were also employed to remove the CI artefact from the EEG, prior to performing connectivity analyses. While connectivity changes could be identified, results need to be interpreted with caution, due to remaining uncertainty about the removal of all CI artefacts. This work is original in analysing and detecting changes in connectivity following repeated stimulation with words and sentences. Finding these changes proved challenging, with many pitfalls with established methods, but the current results and methodological approaches are promising for the continuing study of higher level responses to speech stimulation.

Pitch perception for cochlear implant (CI) users is known to vary between individuals due to differences of insertion depth, of the function of neural tissue in the cochlea, of acclimatisation and CI stimulation parameters. In this study, frequency-to-electrode allocation was adjusted in a group of 12 adult cochlear implant users, to ascertain if the use of a default setting results in optimum perception of speech and music for individual recipients. Participants in the experiment trialled a map in which the frequency allocation was adjusted to the frequency-position function of the normal cochlea and a map which allocated sounds to a limited area of the cochlea, in addition to the default. Performance with the two alternative maps did not exceed that of the default allocation and was poorer for the majority of participants: [F(2,14) = 51.3, p<0.001] for a sentence test in noise. Performance was negatively correlated with the magnitude of the adjustment from the default [r=0.838, p=0.002 and r=-0.700, p=0.024] for the two maps, suggesting that participants had acclimatised to their clinical maps. Electrode discrimination was found to be at chance levels for some participants at the apical end of the array but above chance in the middle of the array. Another alternative map, with logarithmic frequency spacing and some basal shift was trialled and gave improved performance on a sentence test in noise for three participants with poor electrode discrimination at the apical end of the array. A second experiment was conducted, with 13 adult CI users, in which perception of speech and music was assessed with ten frequency allocations, including the default. The ability to follow a pitch contour was measured for centre frequencies of neighbouring filters. Performance with the different allocations varied between individuals; some individuals performed better with alternative allocations from the default. A strategy was developed for the selection of frequency allocation for individuals, based on pitch contour scores for different electrodes, which offered improved performance on the sentence test for the group [t(12)=-3.31, p=0.006, r=0.69]. The overall results show that optimisation of frequency allocation for individuals can be achieved by adjustment of the frequency-to-electrode allocation based on pitch perception ability in different areas of the cochlea.