This thesis describes the data mining of a large set of patient records from the hearing aid clinic at James Cook University Hospital in Middlesbrough, UK. As typical of medical data in general, these audiology records are heterogeneous, containing the following three different types of data: Audiograms (graphs of hearing ability at different frequencies) Structured tabular data (such as gender, date of birth and diagnosis) Unstructured text (specific observations made about each patient in a free- text or comment field) This audiology data set is unique, as it contains records of patients prescribed with both ITE and BTE hearing aids. ITE hearing aids are not generally available on the British National Health Service in England, as they are more expensive than BTE hearing aids. However, both types of aids are prescribed at James Cook University Hospital in Middlesbrough, UK, which is also an important feature of this data. There are two research questions for this research: Which factors influence the choice of ITE (in the ear) as opposed to BTE (behind the ear) hearing aids? For patients diagnosed with tinnitus (ringing in the ear), which factors influence the decision whether to fit a tinnitus masker (a gentle sound source, worn like a hearing aid, designed to drown out tinnitus)? A number of data mining techniques, such as clustering of audiograms, association analysis of variables (such as, age, gender, diagnosis, masker, mould and free text keywords) using contingency tables and principal component analysis on audiograms were used to find candidate variables to be combined into a decision support system (OSS) where unseen patient records are presented to the system, and the relative likelihood that a patient should be fitted with an ITE as opposed to a BTE aid or a tinnitus with masker as opposed to tinnitus not with masker is returned. The DSS was created using the techniques of logistic regression, Nalve Bayesian analysis and Bayesian network, and these systems were tested using 5 fold cross validations to see which of the techniques produced the better results. The advantage of these techniques for the combination of evidence is that it is easy to see which variables contributed to the final d~~Jpion. The constructed models and the data behind them were validated by"presenting them to the Principal audiologist, Dr. Robertshaw at James Cook University Hospital in Middlesbrough for comments and suggestions for improvements. The techniques developed in this thesis for the construction of prediction models were also used successfully on a different audiology data set from Malaysia. These decisions are typically made by audiology technicians working in the out- patient clinics, on the basis of audiogram results and in consultation with the patients. In many cases, the choice is clear cut, but at other times the technicians might benefit from a second opinion given by an automatic system with an explanation of how that second opinion was arrived at.

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The published literature about measuring children's hearing aid outcomes in real-world contexts is limited, compared with that relating to adults. This research investigates two key issues: the validity of the data obtained when using a questionnaire and the extent of aqreement between child and adult proxy respondents. The research used a modified version of the Glasgow Hearing Aid Benefit Profile (GHABP) which was administered to 49 deaf children aged 11-13 years and their parents and Teachers of the Deaf (ToDs) from the Greater Manchester area. Audiological and demographic data were obtained for each child, including the aided and unaided Speech Intelligibility Index and scores on the Ling six sound test. A feature of the research is the use of both quantitative and qualitative methods. The research paradigm is post-positivist, i.e. it is postulated that the questionnaire data are, at best, only approximations to the reality of the children's experience with hearing aids. The quantitative data were explored by several statistical methods, including Generalised Linear Models and Rasch Rating Scale models: the latter are used here for the first time In audiology. The issue of proxy reporting was explored quantitatively by means of Psychometric Profile Analysis and other measures developed in the present research. The qualitative data were investigated by Interpretative Phenomenological Analysis.

Management of earwax problems is inconsistent and under-researched with a paucity of evidence currently available to inform ear care strategy. This study aimed to determine if daily lubrication of the external ear canal with olive oil enhances the normal expulsion of earwax. The study comprised four components: • A pilot questionnaire survey of 55 ear care exhibition attendees • A cross-sectional ear survey of 483 people in Essex and South Yorkshire. • An experimental study of 50 people over 50 years old, presenting with bilateral occlusive earwax. Subjects instilled nightly measure of olive oil into one ear over 24 weeks, the untreated ear acting as control. • Pre- and post-experiment interviews with experimental subjects. A Visual measure of earwax accumulation throughout the experimental period demonstrated a greater increase in treated ears. This was confirmed by the mean weight of earwax removed from treated ears being 72.6% greater than control ears. All earwax was successfully removed before and after the experimental study without adverse effects, by a novel procedure using olive oil application immediately before earwax removal. The mass of earwax accumulated was much greater in males than females. A slower rate of earwax accumulation with increasing age was demonstrated. A history of prior earwax problems correlated with increased earwax accumulation. Interview data illuminated changes in personal ear care post 11 study and linked some earwax problems to the availability of General Practice care and treatment provided. It is concluded that normal expulsion of earwax from the ear canal is not enhanced by regular lubrication with olive oil: on the contrary the accumulating mass of earwax is increased. Olive oil is, however, beneficial in alleviating itchiness and dry ear skin and to aid earwax removal when applied immediately before this procedure. This study contributes new knowledge, which should encourage future studies and guide development of ear care practice. 12

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Cochlear Implants (Cis) are surgically implanted devices that can provide access to the hearing world for individuals with bilateral ‘severe to profound’ hearing loss who do not gain benefit from conventional hearing aids. However, some patients who have received Cis derive more benefit from them than others. There is a relative paucity of work involving those individuals who have had poor outcomes and are performing (with regard to speech understanding) below the level expected for them pre-operatively. The aim of this thesis was to identify potentially remedial factors that contribute to poor performance with a CI, in an attempt to develop practical clinical protocols for the management of ‘poor performers’. Hypotheses: la. The ability to discriminate pitch reliably across the electrode array predicts overall speech discrimination levels in Cochlear Implant users. lb. Using pitch discrimination findings as a ‘site-selection strategy’ would lead to improvements in speech understanding in poor performers. 2. Areas of poor pitch discrimination may correlate with specific electrode position within the cochlea measured by post-operative computerised tomography imaging.

Cochlear implants have been shown to successfully restore hearing to the profoundly deaf. Despite this achievement, issues remain concerning the power consumption and the accuracy of stimulation. This thesis is mainly concerned with investigating the spread of stimulation voltage within the cochlea. The power required to generate the stimulus is also investigated, as is the feasibility of powering a fully implanted cochlear implant by harvesting energy from head motion. Several different models have been used to study the voltage distribution within the cochlea due to electrical stimulation from individual electrodes of a cochlear implant. A resistive cable model is first used to illustrate the fall-off of the voltage with distance at the electrode positions along the cochlea. A three-dimensional finite element model of the cochlea is then developed to obtain the voltage distribution at positions closer to the site of neural stimulation. This model is used to demonstrate the way that the voltage distribution varies with the geometry of the cochlea and the electrode array. It was found that placing the return electrode of the implant within the modiolus, as opposed to outside the cochlea, resulted in higher stimulation for the same current input, which reduces the power requirements. The model has also been used to investigate the consequences of a current-steering, or stimulation focussing, strategy that has previously been proposed. A generalisation of this strategy is suggested, whereby impedance information at the neural level, along the path of the spiral ganglion, was used to optimise the focussed voltage distribution at the target neurons. The power consumption of various stimulation strategies is then estimated in order to assess their energy efficiency. Strategies are defined by parameters such as stimulation rate and number of active channels. The feasibility has also been investigated of harvesting electrical energy from head motion, to power a fully-implanted cochlear implant. It was demonstrated that more power could be harvested from higher harmonics but that this would be sensitive to walking speed. The practical approach is to have a heavily damped device that is insensitive.

There has been explosion in the number of digital hearing aids on the market in the last five years. This master thesis deals with some basic issues related to Digital Hearing Aids and more specifically, with the matter of compression in hearing aid devices. The classic frequency-domain compression uses FFT processing and the ideal and practical FFT systems are described. The underlying theory of Multirate Filter Banks and the Polyphase Decomposition as an efficient way of implementing them are presented. A different prototype filter design is thoroughly described and is proposed since it provides a minimum combined approximation error. Moreover, the approximation of the time-domain post filter with gain coefficients being adapted at the frequency domain is done by an all pole filter of lower degree. The simulation results provide us an evaluation of the proposed technique. Our contribution has been the design of a low-delay FIR filter which is extremely crucial for real-time speech processing applications. / Η τεχνολογία ακουστικών βαρηκοΐας έχει προχωρήσει δραματικά τα τελευταία χρόνια. Η εισαγωγή της Ψηφιακής Επεξεργασίας Σήματος (DSP) στα ακουστικά βαρηκοΐας έχει οδηγήσει σε προχωρημένους αλγορίθμους επεξεργασίας σήματος στους τομείς της έρευνας και της αγοράς. Στη παρούσα μεταπτυχιακή εργασία, ασχολούμαστε με το ζήτημα της συμπίεσης στα Ψηφιακά Ακουστικά Βαρηκοΐας δηλαδή τη μείωση του εύρους των επιπέδων του ήχου που βρίσκονται στο περιβάλλον έτσι ώστε να «ταιριάζουν» μέσα στο περιορισμένο εύρος ενός ατόμου με μειωμένη ακοή. Παραθέτουμε μια μελέτη πάνω στις υπάρχουσες τεχνικές συμπίεσης στο συχνοτικό πεδίο με χρήση FFT (Fast Fourier Transform) για τη σχεδίαση συστημάτων συμπίεσης. Στη συνέχεια, κάνοντας μια εισαγωγή στις τράπεζες φίλτρων (Filter Banks) προτείνουμε τη σχεδίαση ενός prototype φίλτρου με μικρό σφάλμα προσέγγισης, απότομη ζώνη μετάβασης και υψηλή εξασθένιση στη ζώνη αποκοπής. Με αυτό τον τρόπο, η προτεινόμενη τεχνική σχεδίασης παρέχει σημαντικό κέρδος σε αντιπαράθεση με την προτεινόμενη τεχνική με χρήση FFT καθότι οδηγεί σε μείωση της διαρροής ενέργειας μεταξύ γειτονικών συχνοτικών ζωνών. Επιπλέον, μελετήθηκε και αξιολογήθηκε η προσέγγιση του FIR φίλτρου των κερδών συμπίεσης από ένα IIR Auto-Regressive φίλτρο μικρότερης τάξης. Η συμβολή μας στο σημείο αυτό είναι η σχεδίαση ενός FIR φίλτρου συμπίεσης με χαμηλή καθυστέρηση. Το χαρακτηριστικό της χαμηλής καθυστέρησης είναι ιδιαίτερα σημαντικό σε εφαρμογές σημάτων ομιλίας πραγματικού χρόνου όπως στη περίπτωση των ακουστικών βαρηκοΐας.

The reduced speech intelligibility caused by feed feedback oscillation is a major problem for hearing aid users. The demand for improved signal quality has led researchers to look for feedback reduction techniques. In this Thesis, we studied several feedback reduction schemes with emphasis in adaptive feedback cancellation algorithms. The main goal was to develop a system for feedback cancellation that is able to adapt to non-stationary environments while having reasonable computational complexity. This requirement is imposed by the need to implement the feedback cancellation scheme in low power DSP systems. In Chapter 1, we briefly introduced hearing aid systems. We examined the parts that are made of and the types of hearing aids that are available in the market. Then, we described the mechanism that causes feedback oscillation in hearing aids and the adverse effects it has on signal quality. Chapter 2 contains some theoretical results on the field of adaptive linear system identification algorithms and simulation results that support this theory. The chapter begins by giving a derivation of the popular LMS algorithm. A theoretical analysis of LMS using the independence assumption is also provided. Then we are concerned with the least squares filter. We described the RLS algorithm and a linear complexity version of it, the FAEST algorithm. Subsequently, we discussed the FNTF algorithm that trades computational complexity for performance in solving the system identification problem. Next, we developed a new algorithm, the FLMS, by making simplifications to FNTF. We also proved that the proposed algorithm outperforms LMS at least when the input signal is an AR process. Finally, we provided simulation results which prove the superiority of FLMS over LMS. Chapter 3 is devoted in using some algorithms described in Chapter 2 for feedback cancellation in hearing aids. The chapter begins with a hearing aid model that includes an acoustic feedback mechanism. On this system, a linear filter is added that estimates the acoustic feedback so that it can be removed from he signal captured by the microphone. The feedback estimation is performed with LMS and FLMS. Using simulation results, we saw that FLMS can be successfully used in feedback systems and continues to outperform LMS. We also saw that, contrary to the open loop case, when feedback is present, the stochastic approximation theory does not satisfactorily predict the mean learning curves of LMS. / Ένα από τα σημαντικότερα προβλήματα που πρέπει να αντιμετωπιστούν κατά το σχεδιασμό ενός ακουστικού βαρηκοΐας είναι αυτό της ακουστικής ανάδρασης. Με τον όρο ακουστική ανάδραση αναφερόμαστε στο φαινόμενο κατά το οποίο ένα μέρος της εξόδου του ακουστικού επιστρέφει στην είσοδο και ενισχύεται εκ νέου. Γνωστό και ως μικροφωνισμός, το φαινόμενο αυτό γίνεται αντιληπτό από τους ασθενείς ως ένα συνεχές σφύριγμα και είναι ιδιαίτερα ενοχλητικό. Για την αντιμετώπιση του φαινομένου έχουν προταθεί διάφορες τεχνικές. Για παράδειγμα, ο περιορισμός του κέρδους ενίσχυσης στις συχνότητες όπου εμφανίζεται ο μικροφωνισμός είναι μια λύση που συναντάται συχνά σε αναλογικά ακουστικά βαρηκοΐας. Η μέθοδος αυτή, όμως, απαιτεί τον προσδιορισμό των επικίνδυνων συχνοτήτων κατά τη διαδικασία προσαρμογής του ακουστικού στον εκάστοτε ασθενή. Ακόμα και αν ο προσδιορισμός γίνει με μεγάλη ακρίβεια, οι συχνότητες στις οποίες εμφανίζεται ο μικροφωνισμός αλλάζουν κατά τη διάρκεια χρήσης του ακουστικού, περιορίζοντας έτσι την αποτελεσματικότητα της μεθόδου. Με την καθιέρωση της ψηφιακής τεχνολογίας στα ακουστικά βαρηκοΐας, εμφανίζονται νέες δυνατότητες για την αντιμετώπιση του μικροφωνισμού. Είμαστε σε θέση, πλέον, να μοντελοποιήσουμε το σύστημα της ακουστικής ανάδρασης και να χρησιμοποιήσουμε το μοντέλο αυτό για εξαλείψουμε το μικροφωνισμό. Για την μοντελοποίηση αυτή χρησιμοποιείται κατά κόρον ο αλγόριθμος LMS. Η χαμηλή υπολογιστική πολυπλοκότητα που τον χαρακτηρίζει τον κάνει ιδανικό για ακουστικά βαρηκοΐας. Στην εργασία αυτή παρουσιάζουμε έναν νέο αλγόριθμο, επίσης χαμηλής πολυπλοκότητας, για το πρόβλημα της αναγνώρισης γραμμικών συστημάτων. Αποδεικνύουμε με μαθηματικό τρόπο ότι είναι πιο αποτελεσματικός από τον LMS για συγκεκριμένα μοντέλα σημάτων εισόδου, ενώ με εξομοιώσεις ότι υπερτερεί του LMS και για πολύ πιο γενικές εισόδους. Επιπρόσθετα, δείχνουμε ότι ο νέος αλγόριθμος μπορεί να χρησιμοποιηθεί για την ακύρωση της ανάδρασης σε ακουστικά βαρηκοΐας, όπου παραμένει πιο αποτελεσματικός από τον LMS.