Tesis sobre el tema "Signal processing. Cochlear implants"

Ett program som simulerar signalbehandlingen i ett cochleaimplantat med signalbehandlingsstrategin ACE (Advanced Combined Encoder) har konstruerats. Programmets främsta syfte är att i förväg prova ut inställningar i ett cochleaimplantat och genom detta försöka förutsäga olika användares individuella inställningar.   Programmet har validerats med utsignaler bearbetade av Cochlear Limited i deras egenkonstruerade Matlab-modul för forskning inom cochleaimplantat, NMT (Nucleus Matlab Toolbox). Samma insignal som använts av Cochlear Limited har processats av programmet och utsignalerna från detta jämfördes med utsignalerna från NMT. De bägge utsignalerna, producerade med samma inställningar, stämde bra överens.   Komprimeringsfunktionen i programmet, som är en vital del av signalbehandlingen, visade sig stämma bra överens med NMT:s komprimeringsfunktion, sånär som på en relativ minskning av värdet vid starka insignaler. Programmet ska nu användas på Cochleaimplantat-sektionen vid Uppsala Akademiska sjukhus för att pröva ut individuella inställningar till användare av cochleaimplantat. Förhoppningen är att bättre inställningar ska leda till bättre talförståelse och i förlängningen, bättre upplevelse av musik.
A program that simulates the signal processing in a cochlear implant using the signal processing strategy ACE (Advanced Combination Encoder) was constructed. Its main purpose is to, in advance, predict and test different implant settings with the purpose to be able to predict individual patient's differences in implant settings.   The program was validated using output signals processed by Cochlear Limited using their own Matlab Toolbox for implant research, NMT (Nucleus Matlab Toolbox). Identical signals were processed by the program and then compared with NMT:s output. The outputs, produced with several different identical settings matched each other well.   The amplitude compression function, a vital part of the signal processing, also matched well, apart from a relative loss of strength at high input amplitudes. The program will now be used by the cochlear implant section at Uppsala University Hospital to try out individual settings for cochlear implant users. The hope for the future is that better implant settings will lead to improved speech and sound experience, especially, in the long run, with regards to music.

Electrical stimulation of the retina affected by photoreceptor loss (e.g., cases of retinitis pigmentosa) elicits the perception of luminous spots (so-called phosphenes) in the visual field. This phenomenon, attributed to the relatively high survival rates of neurons comprising the retina's inner layer, serves as the cornerstone of efforts to provide a microelectronic retinal prosthesis -- a device analogous to the cochlear implant. This thesis concerns phosphenes -- their elicitation and modulation, and, in turn, image analysis for use in a prosthesis. This thesis begins with a comparative review of visual modelling of electrical epiretinal stimulation and analogous acoustic modelling of electrical cochlear stimulation. The latter models involve coloured noise played to normal listeners so as to investigate speech processing and electrode design for use in cochlear implants. Subsequently, four experiments (three psychophysical and one numerical), and two statistical analyses, are presented. Intrinsic signal optical imaging in cerebral cortex is canvassed appendically. The first experiment describes a visual tracking task administered to 20 normal observers afforded simulated prosthetic vision. Fixation, saccade, and smooth pursuit, and the effect of practice, were assessed. Further, an image analysis scheme is demonstrated that, compared to existing approaches, assisted fixation and pursuit (but not saccade) accuracy (35.8% and 6.8%, respectively), and required less phosphene array scanning. Subsequently, (numerical) information-theoretic reasoning is provided for the scheme's superiority. This reasoning was then employed to further optimise the scheme (resulting in a filter comprising overlapping Gaussian kernels), and may be readily extended to arbitrary arrangements of many phosphenes. A face recognition study, wherein stimuli comprised either size- or intensity-modulated phosphenes, is then presented. The study involved unpracticed observers (n=85), and showed no 'size' --versus--'intensity' effect. Overall, a 400-phosphene (100-phosphene) image afforded subjects 89.0% (64.0%) correct recognition (two-interval forced-choice paradigm) when five seconds' scanning was allowed. Performance fell (64.5%) when the 400-phosphene image was stabilised on the retina and presented briefly. Scanning was similar in 400- and 100-phosphene tasks. The final chapter presents the statistical effects of sampling and rendering jitter on the phosphene image. These results may generalise to low-resolution imaging systems involving loosely packed pixels.

Objetivo: Identificar as contribuições tecnológicas do processador de fala Freedom® para pacientes implantados com Nucleus 22® e a satisfação dos usuários com a nova tecnologia. Entre os novos recursos disponíveis, foram analisados o efeito da tabela de alocação de frequências, o T-SPL e C-SPL e o ajuste de pré-processamento do som (ADRO®). Material: Este estudo foi prospectivo e exploratório. Foram incluídos adolescentes e adultos implantados com Nucleus 22® no Hospital das Clínicas da Faculdade de Medicina da Universidade de São Paulo, usuários efetivos do processador de fala Spectra®, com alguma percepção de frases em contexto fechado e sem experiência anterior com a nova tecnologia. Foram selecionados 17 pacientes, entre as idades de 15 e 82 anos, e implantados há mais de oito anos. Para determinar a contribuição do Freedom®, os limiares auditivos e os testes de percepção de fala foram realizados com o último mapa utilizado com o Spectra® e comparados os mapas criados com o Freedom®. Para identificar o efeito da tabela de alocação de frequências, ambos os mapas convertidos (mesma tabela) e atualizados (tabela nova) foram programados. A tabela escolhida foi mantida, e foram realizados três mapas com diferentes parâmetros: o programa 1 (P1) com T-SPL de 30 dB e do C-SPL de 70 dB, programa 2 (P2) com T-SPL de 25 dB e do C-SPL de 65 dB, e o programa 3 (P3) com ADRO®. A ordem de apresentação dos mapas e dos testes foi randomizada. Para avaliar a satisfação com seus dispositivos auditivos foram utilizados os questionários SADL e APHAB após um mês e um ano de uso do Freedom®. Resultados: A contribuição do processador de fala Freedom® para pacientes usuários do Nucleus 22® foi estatisticamente superior em comparação com o Spectra® em todos os testes de percepção da fala e em todos os limiares audiométricos, tanto individualmente quanto em média, com exceção de 8000 Hz. Em relação à escolha da tabela de frequência, 64,7% dos pacientes (n=11) mantiveram o mapa com a tabela de frequências do Spectra®. Comparando os mapas com diferentes T-SPL e C-SPL, houve diferença estatística tanto nos limiares audiométricos de 500, 1000, 1500 e 2000 Hz quanto na média. Não houve diferença estatística entre os testes de fala com ou sem o uso do ADRO®. Os questionários de satisfação mostraram uma melhora estatisticamente significativa, apenas na subescala que avalia o desempenho em ambiente ruidoso e uso do telefone. Conclusão: A tecnologia contribuiu no desempenho de percepção de fala e nos limiares audiométricos dos pacientes usuários de Nucleus22®. A maioria manteve a tabela de frequência original. As mudanças nos parâmetros de T-SPL e C-SPL mostraram uma melhora dos limiares audiométricos nas frequências principais da fala. As diferenças significantes foram sutis nos questionários de satisfação, demonstrando que os pacientes já estavam adaptados e satisfeitos com o implante coclear
Objective: To identify the technological contributions of the Freedom® speech processor to the patients implanted with Nucleus 22® and the satisfaction of users of the new technology. Among the new features available, we focused on the effect of the frequency allocation table, the T-SPL and C-SPL and the pre-processing gain adjustments (ADRO®). Methods: This study was prospective and exploratory. It included teenage and adult patients implanted with Nucleus 22® who effectively used the implant with no previous experience with the new technology and had at least some speech recognition on a closed set with the Spectra® processor. Seventeen patients met the inclusion criteria, ranging in age from 15 to 82 years and deployed for over 8 years. To determine the contribution of the Freedom®, thresholds and speech perception tests were performed with the last map used with the Spectra® and the maps created for Freedom®. To identify the effect of the frequency allocation table, both converted (same table) and upgraded (new table) maps were programmed. The table selected is maintained, and maps were performed with three different parameters: the first program (P1) was programmed with 30 dB T-SPL and 70 dB C-SPL; the second program (P2) with was programmed with 25 dB T-SPL and 65 dB C-SPL; and the program 3 (P3) with ADRO®. The order of presentation of the maps and the testing was randomized. To assess satisfaction were used SADL and APHAB after one moth and one year of using the Freedom®. Results: The contribution of the Freedom® speech processor to patients with the Nucleus 22® was statistically superior compared to the Spectra® in all tests of speech perception and in all audiometric thresholds, both individually and on average, except for 8000 Hz. Regarding the choice of a frequency allocation table, 64.7% of patients (n=11) maintained the same map that had been used with the Spectra® processor. The sound field threshold was statistically significant at 500, 1000, 1500 and 2000 Hz with 25 dB T-SPL/ 65 dB C-SPL. The patients\' satisfaction there was a statistically significant improvement, only in the sub-scale of speech in noise abilities and telephone use. Conclusions: The Freedom® technology improved the performance of patients with the Nucleus 22®. Most of the patients retained the original frequency table. The changes in the parameters of T-SPL and C-SPL showed an improvement in the audiometric thresholds for the main frequencies of speech. Significant differences were subtle in questionnaires of satisfaction, demonstrating that patients were already adapted and satisfied with the cochlear implant

Thesis (MSL and HT (Interdisciplinary Health Sciences. Speech-Language and Hearing Therapy))--Stellenbosch University, 2008.
This study aims to describe the speech processing skills of three children ages 6;0, 6;10 and 8; 10, with cochlear implants. A psycholinguistic framework was used to profile each child’s strengths and weaknesses, using a single case study approach. Each child’s speech processing skills are described based on detailed psycholinguistically-orientated assessments. In addition, retrospective data from 1-2 years post-implantation were examined in the light of the psycholinguistic framework in order to describe each child’s development over time and in relation to time of implantation. Results showed each child to have a unique profile of strengths and weaknesses, and widely varying outcomes in terms of speech processing even though all three children had the same initial difficulty (congenital bilateral hearing loss). Links between speech processing and other aspects of development as well as contextual factors are discussed in relation to outcomes for each child. The case studies contribute to knowledge of speech processing skills in children with cochlear implants, and have clinical implications for those who work with children with cochlear implants and their families.

Modelling of the mammalian auditory system is valuable in understanding perception processes and has benefits in the design of signal processing systems and human prosthetic implants. However, as models increase in complexity, traditional methods of modelling using general purpose computers become very slow. One method of overcoming this is to use electronic implementations of these models. This thesis looks into the feasibility of auditory system implementations in digital technology, through the implementation of the Four-Stage Pitch System for pitch detection in hearing proposed by Hewitt and Meddis.

Many people around the world suffer from neurological injuries of various sorts that cause serious difficulties in their lives, due to the loss of important sensory and motor functions. Functional electrical stimulation (FES) provides a possible solution to these difficulties by means of a feedback connection allowing the target organ (or organs) to be controlled by electrical stimulation. The control signals can be provided using recorded data extracted from the nerves (electroneurogram, ENG). The most common and safe approaches for interfacing with nerves is called cuff electrodes which deliver the required feedback path for the implantable system with minimum risk. The amount of recorded information can be improved by increasing the number of electrodes within a single cuff known as multi-electrode cuffs (MECs) configuration. This strategy can increase the signal to noise ratio for the recorded signals which have typically very low amplitude (less than 5μV). Consequently multiple high gain amplifiers are used in order to amplify the signals and supply a multi-channel recorded data stream for signal processing or monitoring applications. The signal processing unit within the implantable system or outside the body is employed for classification and sorting the action potential signals (APs) depending on their conduction velocities. This method is called velocity selective recording (VSR). Basically, the idea of this approach is that the conduction velocity of AP can be determined by timing the appearance of the signal at two or more points along the nerve and then dividing the distance between the points by the delay. The purpose of this thesis to investigate an alternative approach using artificial network for APs detection and extraction in neural recording applications to increase the velocity selectivity based on VSR using MECs. The prototype systems impose four major requirements which are high velocity selectivity, small size, low power consumption and high reliability. The proposed method has been developed for applications which require online AP classification. A novel time delay neural network (TDNN) approach is used to decompose the recorded data into several matched velocity bands to allow for individual velocity selectivity at each band to be increased. Increasing the velocity selectivity leads to more accurate recording from the target fibre (or fibres) within the nerve bundle which can be used for applications that require AP classification such as bladder control and the adjustment of foot drop. The TDNN method was developed to obtain more information from an individual cuff without increasing the number of electrodes or the sampling rate. Moreover, the optimization of the hardware implementation for the proposed signal processing method permits savings in power consumption and silicon area. Finally, a nerve signal synthesiser and noise generator for the evaluation of the VSRmethod is described. This system generates multiple artificial AP signals with a time offset between the channels with additive white Gaussian noise (AWGN) to simulate the MEC and hence reduce the cost and the number of the animals required for experimental tests.

The experiments described in this thesis investigate the temporal relationship between frequency bands in a cochlear implant sound processor. Initial studies were of cochlea-based traveling wave delays for cochlear implant sound processing strategies. These were later broadened into studies of an ensemble of across-frequency delays.
Before incorporating cochlear delays into a cochlear implant processor, a set of suitable delays was determined with a psychoacoustic calibration to pitch perception, since normal cochlear delays are a function of frequency. The first experiment assessed the perception of pitch evoked by electrical stimuli from cochlear implant electrodes. Six cochlear implant users with acoustic hearing in their non-implanted ears were recruited for this, since they were able to compare electric stimuli to acoustic tones. Traveling wave delays were then computed for each subject using the frequencies matched to their electrodes. These were similar across subjects, ranging over 0-6 milliseconds along the electrode array.
The next experiment applied the calibrated delays to the ACE strategy filter outputs before maxima selection. The effects upon speech perception in noise were assessed with cochlear implant users, and a small but significant improvement was observed. A subsequent sensitivity analysis indicated that accurate calibration of the delays might not be necessary after all; instead, a range of across-frequency delays might be similarly beneficial.
A computational investigation was performed next, where a corpus of recorded speech was passed through the ACE cochlear implant sound processing strategy in order to determine how across-frequency delays altered the patterns of stimulation. A range of delay vectors were used in combination with a number of processing parameter sets and noise levels. The results showed that additional stimuli from broadband sounds (such as the glottal pulses of vowels) are selected when frequency bands are desynchronized with across-frequency delays. Background noise contains fewer dominant impulses than a single talker and so is not enhanced in this way.
In the following experiment, speech perception with an ensemble of across-frequency delays was assessed with eight cochlear implant users. Reverse cochlear delays (high frequency delays) were equivalent to conventional cochlear delays. Benefit was diminished for larger delays. Speech recognition scores were at baseline with random delay assignments. An information transmission analysis of speech in quiet indicated that the discrimination of voiced cues was most improved with across-frequency delays. For some subjects, this was seen as improved vowel discrimination based on formant locations and improved transmission of the place of articulation of consonants.
A final study indicated that benefits to speech perception with across-frequency delays are diminished when the number of maxima selected per frame is increased above 8-out-of-22 frequency bands.

Speech processing in the human cochlea introduces travelling waves on the basilar membrane. These travelling waves have largely been ignored in most processing strategies. This study implements a hydrodynamical model in a speech processing strategy in order to investigate the neural spike train patterns for a travelling wave processing strategy. In cochlear implants a trade-off remains between the simulation rate and the number of electrode channels. This trade-off was investigated in the proposed travelling wave strategy. Taking into consideration existing current spread and electrical stimulation models, predicted neural spike train responses have shown that stimulating fewer channels (six and four) at stimulation rates of 2 400 pps and 3 600 pps gives better approximations of predicted normal hearing responses for input frequencies of 200 Hz, 600 Hz and 1 kHz, compared to stimulating more channels at lower channel stimulation rates. The predicted neural spike train patterns suggest that these resulting neural patterns might contain both spatial and temporal information that could be extracted by the auditory system. For a frequency of 4 kHz the predicted neural patterns for a channel-number stimulation-rate configuration of 2 - 7 200 pps suggested that although there is no travelling wave delay information, the predicted neural patterns still contain temporal information. The predicted ISI histograms show peaks at the input tone period and multiples thereof, with clusters of spikes evident around the tone period in the predicted spatio-temporal neural spike train patterns. Similar peaks at the tone period were observed for calculated ISI histograms for predicted normal hearing neural patterns and measured neural responses. The predicted spatio-temporal neural patterns for the input frequency of 200 Hz show the travelling wave delay with clusters of spikes at the tone period. This travelling wave delay can also be seen from predicted normal hearing neural responses. The current spread, however, shows a significant distortion effect around the characteristic frequency place where the travelling wave delay increases rapidly. Spacing electrodes more closely results in an increase in this distortion, with the nerve fibre threshold decreasing in adjacent populations of nerve fibres, increasing the probability of firing. The current spread showed a more limited distortion effect on travelling wave delays when electrodes were spaced across the cochlea, at an electrode spacing of 6.08 mm. ISI histogram results also showed increased peaks around the tone period and multiples thereof. These predicted neural spike train patterns suggest that travelling waves in processing strategies, although mostly ignored, might provide the auditory system with both the spatial and temporal information needed for better pitch perception.
Dissertation (MEng)--University of Pretoria, 2012.
Electrical, Electronic and Computer Engineering
MEng
Unrestricted

This research focuses on biologically inspired audio signal processing using programmable analog circuitry. This research is inspired by the biology of the human cochlea since biology far outperforms any engineered system at converting audio signals into meaningful electrical signals. The human cochlea efficiently decomposes any sound into the respective frequency components by harnessing the resonance nature of the basilar membrane, essentially forming a bank of bandpass filters. In a similar fashion, this work revolves around developing a filter bank composed of continuous-time, low-power, analog bandpass filters that serve as the core front end to this silicon audio-processing system. Like biology, the individual bandpass filters are tuned to have narrow bandwidths, moderate amounts of resonance, and exponentially spaced center frequencies. This audio front end serves to efficiently convert incoming sounds into information useful to subsequent signal-processing elements, and it does so by performing a frequency decomposition of the waveform with extremely low-power consumption and real-time operation. To overcome mismatch and offsets inherent in CMOS processes, floating-gate transistors are used to precisely tune the time constants in the filters and to allow programmability of analog components.

La prothèse cochléaire est un appareillage destiné à la réhabilitation des surdités profondes et totales dont un appareillage conventionnel est inefficace. Elle assure la stimulation directe des neurones cochléaires à travers un faisceau d’électrodes. Différents travaux de recherches ont été établis afin d'évaluer l'intelligibilité de la parole chez les sujets bilatéralement implantés en environnements silencieux et bruité. Les résultats ont montré une bonne intelligibilité de la parole en milieu silencieux. Toutefois, les capacités de perception de la parole chez les patients implantés se dégradent en environnement bruité. Nous avons de ce fait proposé trois approches de traitement du signal en vue d'une amélioration de l'intelligibilité de la parole dans le cas de l'implant cochléaire bilatéral: la stimulation bilatérale décalée, l'algorithme de la soustraction spectrale bi-voie et l'algorithme de la soustraction interspectrale. Des améliorations de l'intelligibilité de la parole entre 4% et 10% ont été notées dans le cas de la stimulation bilatérale décalée par rapport à la stimulation bilatérale symétrique. L'approche basée sur l'algorithme de la soustraction spectrale bi-voie présentait des améliorations variables entre 10% et 17%. De meilleures performances ont été obtenues lorsque l'approche basée sur l'algorithme de la soustraction interspectrale est considérée où les améliorations étaient entre 15% et 27%
Cochlear prostheses are intended for persons suffering from deep or total deafness where conventional prostheses proved ineffective. In quiet listening conditions, most bilateral cochlear implant (BCI) users can now achieve even more than 80% word recognition scores regardless the used device. However, under more challenging listening conditions, BCI recipients perform poorly, compared to normal-hearing listeners. In this work, we proposed three speech processing approaches for speech intelligibility improvement. The first is based on shifted bilateral cochlear implant stimulation; the second is based on dual-channel spectral subtraction algorithm and finally the cross power spectral subtraction algorithm was considered. Experimental results showed a speech intelligibility improvement between 4% and 10% when the shifted bilateral cochlear implant stimulation is considered. Performance amelioration was observed when the dual-channel spectral subtraction based speech enhancement algorithm was considered and the improvement was between 10% and 17%. The better performance was obtained when noisy speech signals were processed using cross power spectral subtraction algorithm and the improvement was between 15% and 27%

碩士
國立交通大學
生醫工程研究所
101
Cochlear implants can provide an opportunity for severe to profound hearing impaired patients to perceive hearing sensation again. However, the hearing resolutions of traditional speech processing strategies are constrained by the limited number of electrodes. For cochlear implant users, the electrical field interaction between the electrodes would distort speeches. In acoustic model or vocoder implementations, two stimulating strategies (FAME and DRNL), incorporating current steering schemes to provide a higher frequency spectrum resolution than the two original strategies, are studied. To simulate the electrical field interaction in the acoustic model or vocoder, the revised SPREAD model combined with common strategies is introduced to reflect the electric field interaction between cochlear implant electrodes. Result of the first experiment shows that the two strategies with current steering scheme performed better in hearing tests. Result of the second experiment shows that the stimulating strategies with SPREAD model would reduce the gap between the results from normal hearing test subjects and those from cochlear implant users.

碩士
義守大學
電機工程學系
91
ABSTRACT A prosthetic device, called the cochlear implant, can be implanted in the inner ear and can restore partial hearing to profoundly deaf people. According to the clinical data, the effect is vary different from individual. Therefore, improved efficacy to cochlear implant is such a important work. There are two ways can ameliorate the situation. One is to find the new stimulation strategy, the other is to design new electrode dimension over again. When new stimulation strategy or new electrode be developed. Apply to patient is only way to distinguish weather the new design fine or inferior. We are going to find some method called modeling evoked compound action potential in cochlear implant that we can know the performance on new design without any surgery. Currently there is no easy ways to model evoked compound action potential (EAP) of the auditory nerve fibers. This thesis presents a method to model the EAP using finite element method, Schwarz-Eikhof nerve fiber model, and equivalent circuit models on a 3D cochlea model. The simulation results are validated by EAP measured in cochlear implant users.

Cochlear implants (CIs) provide coarse representations of pitch, which are adequate for speech but not for music. Despite increasing interest in music processing by CI users, the available information is fragmentary. The present experiment attempted to fill this void by conducting a comprehensive assessment of music processing in adult CI users. CI users (n =6) and normally hearing (NH) controls (n = 12) were tested on several tasks involving melody and rhythm perception, recognition of familiar music, and emotion of recognition in speech and music. CI performance was substantially poorer than NH performance and at chance levels on pitch processing tasks. Performance was highly variable, however, with one individual achieving NH performance levels on some tasks, probably because of low-frequency residual hearing in his unimplanted ear. Future research with a larger sample of CI users can shed light on factors associated with good and poor music processing in this population.

Cochlear implant listeners are presented with a time and frequency-quantized version of speech signals. In the frequency domain, resolution is limited by the number of electrodes in each listener's array. Current cochlear implant speech processing strategies implicitly assume that the information presented to each one of these electrodes is perceived as unique and independent. However, previous research suggests that stimuli presented on different electrodes can be indiscriminable (e.g. Zwolan et al., 1997; Throckmorton and Collins, 1999; Henry et al., 2000) . Additional studies suggest that stimuli presented on one electrode can influence the perception of stimuli on neighboring electrodes (e.g. Shannon, 1990; Chatterjee and Shannon, 1998; Boëx et al., 2003). Removing this redundant or occluded information could cause more distinct or perceivable information to be presented to the listener and possibly result in improved speech recognition.

Previous studies have used psychophysical data to identify the electrodes with the highest potential to confound speech recognition (Zwolan et al., 1997, Boëx et al., 2003, and Garadat et al., 2012). In order to minimize electrode interactions and maximize the amount of perceivable information, each of these studies used a single psychophysical metric to deactivate the electrodes across all time windows of the speech processing strategy. For some listeners, these reduced electrode sets resulted in improved speech recognition over using the of the electrodes in their array. These studies did not compare the results of using different psychophysical metrics to exclude electrodes for a group of listeners nor did they investigate speech recognition performance as a function of the number of electrodes excluded from the array.

In this work, three different psychophysical metrics were used to obtain a multidimensional estimate of the potential "usefulness'' of each electrode. These results were then used to inform two different methods of psychophysics-motivated electrode selection. The first method incorporated individual data into each listener's energy-driven speech processing strategy. For each time window, the electrodes with the highest energy that were also most likely to be perceived, according to the psychophysical data, were selected for stimulation. The second method sequentially excluded the electrodes with the highest potential to confound from the array across all time windows, resulting in a group of psychophysics-motivated electrode sets for each metric. Evaluating each of these electrode sets exhaustively would require a prohibitive amount of experimental time. To mitigate this problem, an adaptive procedure was developed to estimate performance as a function of cochlear implant parameters in a time-efficient manner. For each metric, the procedure estimated the set with the highest estimated probability of correct phoneme identification. Listeners' speech recognition performance using this electrode set was then compared to their performance using their full electrode array. For both electrode selection methods, listeners' speech recognition scores were generally comparable to those obtained in the clinical condition. This finding supports the hypothesis that listeners were not perceiving all the information presented to them using their clinical speech processing strategy and their complete set of electrodes. Additionally, these results suggest that improvements to the proposed electrode selection strategies should be in investigated in order to increase the amount of perceivable information presented to cochlear implant listeners.

Dissertation

Yu, Shing.
Thesis (M.Phil.)--Chinese University of Hong Kong, 2011.
Includes bibliographical references (leaves 87-93).
Abstracts in English and Chinese; includes Chinese.
Abstract --- p.i
Acknowledgement --- p.vi
Chapter 1 --- Introduction --- p.1
Chapter 1.1 --- Hearing impairment --- p.1
Chapter 1.2 --- Limitations of existing CI --- p.2
Chapter 1.3 --- Objectives --- p.3
Chapter 1.4 --- Thesis Outline --- p.4
Chapter 2 --- Background --- p.6
Chapter 2.1 --- Signal Processing in CI --- p.6
Chapter 2.1.1 --- Continuous Interleaved Sampler (CIS) --- p.7
Chapter 2.1.2 --- Advanced Combination Encoder (ACE) --- p.12
Chapter 2.2 --- Tone perception by cochlear implantees --- p.15
Chapter 2.2.1 --- Pitch and Tone --- p.15
Chapter 2.2.2 --- Mechanisms of pitch perception by cochlear im- plantees --- p.20
Chapter 3 --- Tone-enhanced ACE Strategy for CI --- p.23
Chapter 3.1 --- Basic principles --- p.23
Chapter 3.2 --- Acoustical simulation with noise excited vocoder --- p.26
Chapter 3.3 --- Implementation in a real CI system --- p.29
Chapter 3.3.1 --- Technical details --- p.30
Chapter 3.3.2 --- Visual comparison --- p.31
Chapter 4 --- Robust Generation of F0 Trajectory --- p.33
Chapter 4.1 --- Requirement on the F0 contour --- p.33
Chapter 4.2 --- Extraction of F0 contour --- p.34
Chapter 4.3 --- Post-processing of F0 contour --- p.36
Chapter 4.3.1 --- Removal of octave-jump --- p.36
Chapter 4.3.2 --- Interpolation --- p.36
Chapter 4.3.3 --- Prediction --- p.36
Chapter 4.3.4 --- Smoothing --- p.38
Chapter 4.4 --- Performance evaluation --- p.38
Chapter 5 --- Design of Listening Tests --- p.41
Chapter 5.1 --- Speech Materials --- p.41
Chapter 5.2 --- Testing modes --- p.43
Chapter 5.2.1 --- Sound field mode --- p.45
Chapter 5.2.2 --- Direct stimulation mode --- p.46
Chapter 5.3 --- Test Interface --- p.47
Chapter 6 --- Sound-field Tests --- p.49
Chapter 6.1 --- Materials and Methods --- p.50
Chapter 6.1.1 --- Subjects --- p.50
Chapter 6.1.2 --- Signal processing and test stimuli --- p.52
Chapter 6.1.3 --- Procedures --- p.52
Chapter 6.2 --- Results --- p.54
Chapter 6.3 --- Discussion --- p.57
Chapter 7 --- Evaluation of Tone-enhanced Strategy --- p.59
Chapter 7.1 --- Materials and Methods --- p.60
Chapter 7.1.1 --- Subjects --- p.60
Chapter 7.1.2 --- Signal processing and test stimuli --- p.60
Chapter 7.1.3 --- Procedures --- p.62
Chapter 7.2 --- Results --- p.63
Chapter 7.3 --- Discussion --- p.66
Chapter 8 --- Use of Automatically Generated F0 Contour --- p.72
Chapter 8.1 --- Materials and Methods --- p.73
Chapter 8.2 --- Results --- p.74
Chapter 8.3 --- Discussion --- p.76
Chapter 9 --- Conclusions --- p.80
Chapter A --- LSHK Cantonese Romanization Scheme --- p.85
Bibliography --- p.87

Cochlear implants provide hearing to people who are deaf, by electrically stimulating the auditory nerve. Children with a single cochlear implant suffer deficiencies inherent to unilateral hearing, including inability to locate sounds. A second cochlear implant may improve sound localization, which normally requires interpretation of differences in sound intensity and time of arrival between two ears. Currently, it is unknown whether these cues are available to children who were provided with a second cochlear implant after a period of using one implant alone. We asked whether such children could interpret inter-implant level and timing cues. Results indicated that children using two cochlear implants detected level cues but had difficulty interpreting timing cues. Further, children rarely reported that sounds were perceived to come from the middle. Children receiving bilateral cochlear implants sequentially do not process bilateral auditory cues normally but can use inter-implant level cues to make judgments about where sound is coming from.

The purpose of this thesis is to provide a review and discussion of the current literature on auditory processing, speech discrimination, word recognition, and early lexical representations in children with normal hearing and those with hearing impairment in addition to identifying areas in which current research is lacking. This information is needed to consider potential interactions between various factors affecting the development of spoken word recognition. This will also provide a starting point for identifying further research needs. Since children with hearing loss do not receive the same amount of exposure to speech and language as typically developing children, it can be expected that the development of speech and word recognition in this population may progress differently. If we can identify differences in auditory processing and phonological development in children with hearing impairment, we can modify speech and language therapy to focus on more specific and effective targets. The subsequent chapters will provide a critical review of the current literature on the aforementioned topics. In Chapters 2 and 3, studies assessing differences in processing, attention to sound, intersensory perception, and sound discrimination abilities in children with normal hearing and hearing impairment will be discussed. Chapters 4 and 5 focus on word recognition skills, and early lexical representations. Chapter 6 will synthesize results of available studies and suggest areas in which more research is needed. Together, these chapters will help us gain a better understanding of the complex interactions between auditory processing, executive functioning, phonological development and later word recognition outcomes. By identifying which avenues have the greatest effect on outcomes in cochlear implant users, we can modify speech and language therapy in order to address the unique needs of this special population.
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It is well established that cochlear implants (CIs) are able to provide many users with excellent speech recognition ability in quiet conditions; however, the ability to correctly identify speech in noisy conditions or appreciate music is generally poor for implant users with respect to normal-hearing listeners. This discrepancy has been hypothesized to be in part a function of the relative decrease in spectral information available to implant users (Rubinstein and Turner, 2003; Wilson et al., 2004). One method that has been proposed for increasing the amount of spectral information available to CI users is to include time-varying stimulation rate in addition to changes in the place of stimulation. However, previous implementations of multi-rate strategies have failed to result in an improvement in speech recognition over the clinically available, fixed-rate strategies (Fearn, 2001; Nobbe, 2004). It has been hypothesized that this lack of success was due to a failure to consider the underlying perceptual responses to multi-rate stimulation.

In this work, psychophysical experiments were implemented with the goal of achieving a better understanding of the interaction of place and rate of stimulation and the effects of duration and context on CI listeners' ability to detect changes in stimulation rate. Results from those experiments were utilized in the implementation of a tuned multi-rate sound processing strategy for implant users in order to potentially ``tune" multi-rate strategies and improve speech recognition performance.

In an acute study with quiet conditions, speech recognition performance with a tuned multi-rate implementation was better than performance with a clinically available, fixed-rate strategy, although the difference was not statistically significant. These results suggest that utilizing time-varying pulse rates in a subject-specific implementation of a multi-rate algorithm may offer improvements in speech recognition over clinically available strategies. A longitudinal study was also performed to investigate the potential benefit from training to speech recognition. General improvements in speech recognition ability were observed as a function of time; however, final scores with the tuned multi-rate algorithm never surpassed performance with the fixed-rate algorithm for noisy conditions.

The ability to improve upon speech recognition scores for quiet conditions with respect to the fixed-rate algorithm suggests that using time-varying stimulation rates potentially provides additional, usable information to listeners. However, performance with the fixed-rate algorithm proved to be more robust to noise, even after three weeks of training. This lack of robustness to noise may be in part a result of the frequency estimation technique used in the multi-rate strategy, and thus more sophisticated techniques for real-time frequency estimation should be explored in the future.

Dissertation