Hearing aid devices are used to help people with hearing impairment. The number of people that requires hearingaid devices are possibly constant over the years, however the number of people that now have access to hearing aiddevices increasing rapidly. The hearing aid devices must be small, consume very little power, and be fairly accurate.Even though it is normally more important for the user that hearing impairment look good (are discrete). Once thehearing aid device prescribed to the user, she/he needs to train and adjust the device to compensate for the individualimpairment.We are within the framework of this project researching on hearing aid devices that can be trained by the hearingimpaired person her-/himself. This project is about finding suitable noise cancellation algorithm for the hearing-aiddevice. We consider several types of algorithms like, microphone array signal processing, Independent ComponentAnalysis (ICA) based on double microphone called Blind Source Separation (BSS) and DRNPE algorithm.We run this current and most sophisticated and robust algorithms in certain noise backgrounds like Cocktail noise,street, public places, train, babble situations to test the efficiency. The BSS algorithm was well in some situation andgave average results in some situations. Where one microphone gave steady results in all situations. The output isgood enough to listen targeted audio.The functionality and performance of the proposed algorithm is evaluated with different non-stationary noisebackgrounds. From the performance results it can be concluded that, by using the proposed algorithm we are able toreduce the noise to certain level. SNR, system delay, minimum error and audio perception are the vital parametersconsidered to evaluate the performance of algorithms. Based on these parameters an algorithm is suggested forheairng-aid. / Hearing-Aid

In this thesis, two novel broadband slot mobile phone antenna designs respectively for penta-band WWAN operation and eight-band LTE/WWAN operation are presented. The antennas are suitable to be mounted near the bottom edge of the system ground plane of the mobile phone. Good radiation characteristics for the antennas are obtained, and the two antennas respectively occupy a small printed area of 50 ¡Ñ 4 mm2 and 53 ¡Ñ 4 mm2. The first design uses a C-shaped strip connected to the bottom edge of the system ground plane to make the structure of the system ground plane close to a symmetric shorted dipole antenna. This makes it promising to excite a chassis mode to enhance the operating bandwidth of the antenna. The second one uses a microstrip feedline having a chip-inductor-loaded branch. The novel microstrip feedline can lead to more uniform distribution of the electric fields excited in the slot such that enhanced bandwidth of the antenna¡¦s lower band is obtained. Further, since the chip inductor performs like a low-pass filter, the original bandwidth of the antenna¡¦s upper band is not affected. Additionaly, the impedance matching of the lower frequencies of the upper band can be improved, which enhances the upper-band bandwidth of the antenna. Effects of the user¡¦s head and hand on the proposed antenna are also studied, and the simulated SAR and HAC issues are also analyzed in this thesis.

Hearing aids are tasked with the undesirable job of compensating an impaired, highly-nonlinear auditory system. Historically, these devices have either employed linear processing or relatively unsophisticated, nonlinear processing techniques. With increasingly more accurate models of the auditory system, expanding computational power, and many more objective measures which utilize these models, we are at a turning point in hearing aid design. Although subjective listener tests are often the most accepted methods for evaluating the quality and intelligibility of speech, they inherently treat the auditory system as a "black box." Conversely, model-based objective measures typically treat the auditory system as a cascade of physical processes. As a result, objective measures have the potential to provide more detailed information about how sound is processed and about where and why quality or intelligibility breaks down. Provided that we can generalize model-based objective measures, we can use the measures as tools for understanding how to best process degraded signals, and therefore, how to best design hearing aids. However, generalizability is a key requirement. Since many of the well-known objective measures have been developed for normal-hearing listeners in the context of audio codecs, we are unsure about the generalizability of these measures to predicting quality and intelligibility for hearing-impaired listeners with "unknown" datasets (i.e. a set on which it was not trained) and distortions which are specific to hearing aids. Relatively recently, however, Kates and Arehart (Journal of the Audio Engineering Society, 2010) proposed the Hearing Aid Speech Quality Index (HASQI), which is a model-based objective measure that predicts quality for normal-hearing and hearing-impaired listeners by taking into account many of the distortions which hearing aids introduce. HASQI solves many of our concerns of generalizability for predicting quality, but it still remains to test HASQI's ability to predict quality with datasets on which it was not trained. Thus, we explore the robustness of HASQI by testing its ability to predict quality for "unknown" de-noised speech, and we directly compare its performance to some other metrics in the literature.

Hearing impairment is the number one chronic disability affecting people in the world. Many people have great difficulty in understanding speech with background noise. This is especially true for a large number of elderly people and the sensorineural impaired persons. Several investigations on speech intelligibility have demonstrated that subjects with sensorineural loss may need a 5-15 dB higher signal-to-noise ratio than the normal hearing subjects. While most defects in transmission chain up to cochlea can nowadays be successfully rehabilitated by means of surgery, the great majority of the remaining inoperable cases are sensorineural hearing impaired, Recent statistics of the hearing impaired patients applying for a hearing aid reveal that 20% of the cases are due to conductive losses, more than 50% are due to sensorineural losses, and the rest 30% of the cases are of mixed origin. Presenting speech to the hearing impaired in an intelligible form remains a major challenge in hearing-aid research today. Even-though various methods have been suggested in the literature for the minimization of noise from the contaminated speech signals, they fail to give good SNR improvement and intelligibility improvement for moderate to-severe sensorineural loss subjects. So far, the power and capability of Newton's method, Nonlinear adaptive filtering methods and the feedback type artificial neural networks have not been exploited for this purpose. Hence we resort to the application of all these methods for improving SNR and intelligibility for the sensorineural loss subjects. Digital hearing aids frequently employ the concept of filter banks. One of the major drawbacks of this techniques is the complexity of computation requiring more number of multiplications. This increases the power consumption. Therefore this Thesis presents the new approach to speech enhancement for the hearing impaired and also the construction of filter bank in Digital hearing aid with minimum number of multiplications. The following are covered in this thesis. One of the most important application of adaptive systems is in noise cancellation using adaptive filters. The ANC setup requires two input signals (viz., primary and reference). The primary input consists of the sum of the desired signal and noise which is uncorrelated. The reference input consists of mother noise which is correlated in Some unknown way with noise of primary input. The primary signal is obtained by placing the omnidirectional microphone just above one ear on the head of the KEMAR mannikan and the reference signal is obtained by placing the hypercardioid microphone at the center of the vertebral column on the back. Conventional speech enhancement techniques use linear schemes for enhancing speech signals. So far Nonlinear adaptive filtering techniques are not used in hearing aid applications. The motivation behind the use of nonlinear model is that it gives better noise suppression as compared to linear model. This is because the medium through which signals reach the microphone may be highly nonlinear. Hence the use of linear schemes, though motivated by computational simplicity and mathematical tractability, may be suboptimal. Hence, we propose the use of nonlinear models to enhance the speech signals for the hearing impaired: We propose both Linear LMS and Nonlinear second order Volterra LMS schemes to enhance speech signals. Studies conducted for different environmental noise including babble, cafeteria and low frequency noise show that the second-order Volterra LMS performs better compared to linear LMS algorithm. We use measures such as signal-to-noise ratio (SNR), time plots, and intelligibility tests for performance comparison. We also propose an ANC scheme which uses Newton's method to enhance speech signals. The main problem associated with LMS based ANC is that their convergence is slow and hence their performance becomes poor for hearing aid applications. The reason for choosing Newton's method is that they have high performance adaptive-filtering methods that often converge and track faster than LMS method. We propose two models to enhance speech signals: one is conventional linear model and the other is a nonlinear model using a second order Volterra function. Development of Newton's type algorithm for linear mdel results in familiar Recursive least square (RLS) algorithm. The performance of both linear and non-linear Newton's algorithm is evaluated for babble, cafeteria and frequency noise. SNR, timeplots and intelligibility tests are used for performance comparison. The results show that Newton's method using Volterra nonlinearity performs better than RLS method. ln addition to the ANC based schemes, we also develop speech enhancement for the hearing impaired by using the feedback type neural network (FBNN). The main reason is that here we have parallel algorithm which can be implemented directly in hardware. We translate the speech enhancement problem into a neural network (NN) framework by forming an appropriate energy function. We propose both linear and nonlinear FBNN for enhancing the speech signals. Simulated studies on different environmental noise reveal that the FBNN using the Volterra nonlinearity is superior to linear FBNN in enhancing speech signals. We use SNR, time plots, and intelligibility tests for performance comparison. The design of an effective hearing aid is a challenging problem for sensorineural hearing impaired people. For persons with sensorineural losses it is necessary that the frequency response should be optimally fitted into their residual auditory area. Digital filter enhances the performance of the hearing aids which are either difficult or impossible to realize using analog techniques. The major problem in digital hearing aid is that of reducing power consumption. Multiplication is one of the most power consuming operation in digital filtering. Hence a serious effort has been made to design filter bank with minimum number of multiplications, there by minimizing the power consumption. It is achieved by using Interpolated and complementary FIR filters. This method gives significant savings in the number of arithmetic operations. The Thesis is concluded by summarizing the results of analysis, and suggesting scope for further investigation

The thesis investigated the importance of cognition for speech understanding in experienced and new hearing aid users. The aims were 1) to develop a cognitive test (Sentence-final Word Identification and Recall, or SWIR test) to measure the effects of a noise reduction algorithm on processing of highly intelligible speech (everyday sentences); 2) to investigate, using the SWIR test, whether hearing aid signal processing would affect memory for heard speech in experienced hearing aid users; 3) to test whether the effects of signal processing on the ability to recall speech would interact with background noise and individual differences in working memory capacity; 4) to explore the potential clinical application of the SWIR test; and 5) to examine the relationship between cognition and speech recognition in noise in new users over the first six months of hearing aid use. Results showed that, for experienced users, noise reduction freed up cognitive resources and alleviated the negative  impact of noise on memory when speech stimuli were presented in a background of speech babble spoken in the listener’s native language. The possible underlying mechanisms are that noise reduction facilitates auditory stream segregation between target and irrelevant speech and reduces the attention captured by the linguistic information in irrelevant speech. The effects of noise reduction and SWIR performance were modulated by individual differences in working memory capacity. SWIR performance was related to the self-reported outcome of hearing aid use. For new users, working memory capacity played a more important role in speech recognition in noise before acclimatization to hearing aid amplification than after six months. This thesis demonstrates for the first time that hearing aid signal processing can significantly improve the ability of individuals with hearing impairment to recall highly intelligible speech stimuli presented in babble noise. It also adds to the literature showing the key role of working memory capacity in listening with hearing aids, especially for new users. By virtue of its relation to subjective measures of hearing aid outcome, the SWIR test can potentially be used as a tool in assessing hearing aid outcome. / Avhandlingens övergripande mål var att studera kognitionens betydelse för talförståelse hos vana och nya hörapparatsanvändare. Syftena var att 1) utveckla ett kognitivt test (Sentence-final Word Identification and Recall, eller SWIR test) för att mäta en brusreducerande algoritms effekt på bearbetningen av tydligt tal (vardagsmeningar); 2) att med hjälp av SWIR testet undersöka huruvida hörapparatens signalbehandling påverkade återgivningen av uppfattat tal hos vana hörapparatsanvändare; 3) att utvärdera om effekten av signalbehandling på förmågan att komma ihåg tal påverkas av störande bakgrundsljud samt individuella skillnader i arbetsminnets kapacitet; 4) att undersöka den potentiella kliniska tillämpningen av SWIR testet och 5) att undersöka förhållandet mellan kognition och taluppfattning i störande bakgrundsljud hos nya hörapparatsanvändare under de första sex månaderna med hörapparater. Resultaten visade att för vana hörapparatsanvändare lindrade brusreduceringen det störande ljudets negativa inverkan på minnet när meningar presenterades i form av irrelevant tal på deltagarnas modersmål. De möjliga underliggande mekanismerna är att brusreducering underlättar diskriminering av de auditiva informationsflödena mellan det som ska uppfattas och det som är irrelevant, samt minskar graden av uppmärksamhet som fångas av den språkliga informationen i det irrelevanta talet. Effekterna av brusreducering och resultaten av SWIR var beroende av individuella skillnader i arbetsminnets kapacitet. Resultaten av SWIR har också samband med det självrapporterade utfallet av  hörapparatsanvändning. För nya användare spelar arbetsminnets kapacitet initialt en viktigare roll för taluppfattning i störande bakgrundsljud, innan anpassningen till hörapparatens förstärkning skett, än efter sex månader. Denna avhandling visar för första gången att hörapparatens signalbehandling kan signifikant förbättra möjligheten för individer med hörselnedsättning att minnas tydligt tal, som presenteras i störande bakgrundsljud. Avhandlingen bidrar till litteraturen med en diskussion om hur arbetsminnets kapacitet spelar roll i taluppfattning med hörapparat, i synnerhet för nya användare. Med stöd av dess samband med det självrapporterade utfallet, kan SWIR testet användas som redskap i bedömning av hörapparaters effekt.

Hearing impairment (HI) is a growing health issue in today’s ageing society. Research has suggested that individuals with HI may experience increased levels of anxiety. Previous research has mainly focused on anxiety as a trait; recent research, however, has looked at state anxiety in the hearing impaired population. Cognitive anxiety is a state anxiety that occurs when people encounter a situation which does not lie within their construct system. As a result, they may experience anxiety as they are unable, or only partially able, to interpret the event meaningfully and are therefore unable to judge the implications of this event. The following study aimed to use the Cognitive Anxiety Scale to investigate relationships between cognitive anxiety and client variables in hearing impaired individuals, adding to the small amount of research currently available in this area. The following research questions were investigated: (1) Is there a relationship between cognitive anxiety level and (a) age, (b) gender, (c) audiometric variables, and (d) quality of life? (2) Is there a significant difference between the level of cognitive anxiety for the participants who purchased and kept hearing aids and those who did not? Twenty-five hearing impaired individuals who were consulting an audiologist for the first time participated in this study, with the cognitive anxiety interview conducted prior to the audiological assessment. The results indicated that cognitive anxiety was significantly related to an ability to understand speech in noise and quality of life, and that hearing aid adopters exhibited greater levels of cognitive anxiety than non-adopters. These results confirm that cognitive anxiety is indeed experienced by adults with HI, and suggest that it may be a factor which motivates people to adopt hearing aids. Further research is needed to confirm and further investigate the relationships with client variables. By listening for signs of cognitive anxiety, an audiologist may be able to gauge if a client is ready for rehabilitation, and encourage the process by exploring the effects of HI on communication situations, employing speech in noise testing, and including the significant other in the process.

In the future, audiological rehabilitation of adults with hearing loss will be more available, personalized and thorough due to the possibilities offered by the internet. By using the internet as a platform it is also possible to perform the process of rehabilitation in a cost-effective way. With tailored online rehabilitation programs containing topics such as communication strategies, hearing tactics and how to handle hearing aids it might be possible to foster behavioral changes that will positively affect hearing aid users. Four studies were carried out in this thesis. The first study investigated internet usage among adults with hearing loss. In the second study the administration format, online vs. paper- and pencil, of four standardized questionnaires was evaluated. Finally two randomized controlled trials were performed evaluating the efficacy of online rehabilitation programs including professional guidance by an audiologist. The programs lasted over five weeks and were designed for experienced adult hearing-aid users. The effects of the online programs were compared with the effects of a control group. It can be concluded that the use of computers and the internet overall is at least at the same level for people with hearing loss as for the general age-matched population in Sweden. Furthermore, for three of the four included questionnaires, the participants’ scores remained the same across formats. It is however recommended that the administration format remain consistent across assessment points. Finally, results from the two concluding intervention studies provide preliminary evidence that the internet can be used to deliver education and rehabilitation to experienced hearing aid users who report residual hearing problems and that their problems are reduced by the intervention; however the content and design of the online rehabilitation program requires further investigation.

A vida de milhões de pessoas é afetada por problemas de deficiência auditiva, incapacitando-as de ouvirem os sons naturalmente. O uso de aparelhos de auxílio auditivo minimiza o efeito das deficiências, pois possibilita tratamento dos sinais auditivos através de sofisticados algoritmos que eliminam ruídos e amplificam os sinais de interesse. Este trabalho propõem a especificação de um sistema integrado, otimizado em termos de consumo de potência, para realizar o processamento de sinais digitais em aparelhos de auxílio auditivo digital. Foram desenvolvidos dois blocos para processamento hardwired, que substituem o processamento realizado por software, cuja finalidade é filtrar os sinais sonoros digitalizados com menor consumo. Um dos blocos, um filtro FIR de até 128 coeficientes, pode ser utilizado como filtro do tipo passa baixa ou passa altas frequências. O outro bloco, para executar o algoritmo ALE, é utilizado para eliminar ruídos periódicos. Os blocos desenvolvidos e implementados foram compilados e simulados para comprovar a funcionalidade. Os resultados das simulações mostraram que eles atendem as especificações de funcionalidade. Os blocos foram também sintetizados em uma tecnologia CMOS de 0,35 &#956m, três níveis de metal, para assim se ter as estimativas de área do circuito e de consumo de potência. A área do layout final foi de 14 mm². O consumo de potência estimado é de 0,30 mW para frequência de clock de 300 kHz (o que permite que um filtro FIR processe uma amostra a cada 240 &#956s, no pior caso, e o ALE, uma a cada 36 &#956s), e de 5,06 mW para frequência de clock de 5,0 MHz (filtro FIR processa uma amostra a cada 14,4 &#956s e o ALE, uma a cada 2,2 &#956s). As estimativas de consumo foram feitas considerando os dois blocos operando simultaneamente e com tensão de alimentação de 1,8 V. Para todo o sistema integrado proposto, obtive-se, com um cenário específico, o consumo de potência de 1,1 mW, considerando dois Filtros Configuráveis, um Filtro ALE e um DSP. / The live of millions of people are affected by hearing problems, disabling them from hearing the sounds naturally. The use of hearing aids devices minimizes the effect of deficiencies, since it allows processing of auditory signals through sophisticated algorithms that eliminate noise and amplify the signals of interest. This work proposes the specification of an integrated system, optimized in terms of power consumption, to perform digital signal processing in digital hearing aid devices. Were developed two blocks of hardwired processing, replacing software processing, whose purposes are to filter the digitized audio signals with lower consumption. One of the blocks, an FIR filter up to 128 coefficients can be used as a low pass or high pass filter. The other block, to run the ALE algorithm, is used to eliminate periodic noises. The blocks developed and implemented were compiled and simulated to demonstrate their functionality. The simulation results show that they meet the specifications of functionality. The blocks were also synthesized in a 0.35 &#956m CMOS technolog, three metal levels, in order to have estimatives of circuit area and power consumption. The area of the final layout was 14,0 mm². The estimated power consumption is 0.30 mW for clock frequency of 300 kHz (which allows a FIR filter to process one sample every 240 &#956s in the worst case, and ALE, one every 36 &#956s), and 5.06 mW for clock frequency of 5.0 MHz (FIR filter processing one sample every 14.4 &#956s, and ALE, one every 2.2 &#956s). Consumption estimates were made considering the two blocks operating simultaneously and supply voltage of 1.8 V. For all the proposed integrated system, it was found, for a specific scenario, the power consumption of 1.1 mW, considering two configurable filters, one filter ALE and one DSP.

Advances in hearing instrument technology have permitted the development of non-linear prescriptive methods to prescribe amplification characteristics for the hearing- impaired individual. The dispenser’s task in selecting the most appropriate prescriptive procedure for the young child is of utmost importance to ensure optimum hearing aid benefit for communication development. It was the aim of this study to compare and describe the effect of the two most widely used methods, DSL (i/o) and NAL-NL1, on speech recognition and loudness perception. An exploratory, descriptive research design was selected to realise this goal. Ten participants were selected using a convenient non-probability method of sampling. Articulation index calculations and a closed set speech recognition test were utilised in the evaluation of speech recognition, whereas functional gain results and loudness rating measurements provided an opportunity to describe loudness perception. The obtained results were analysed using the SAS (Statistical Analysis System). The study concluded that, although significant statistical differences existed in loudness perception, no statistical difference was observed in actual speech recognition measures. This effect may contribute to the individual amplification approaches of the two methods, which seem to reflect the uncertainties expressed by researchers as to the contribution of high frequency amplification to speech recognition in young children. / Dissertation (M (Communication Pathology))--University of Pretoria, 2006. / Speech-Language Pathology and Audiology / Unrestricted

People with presbyacusis commonly report difficulties in conversation in everyday settings. Although previous research has focused on self-report inventories concerning conversation difficulties in age related hearing difficulties, there is a lack of published work describing the interactions between people with presbyacusis and their conversational partners. The aim of this study is to describe conversational interactions between people with presbyacusis and their main everyday conversational partner and to determine whether there is evidence of change in interaction before and after the fitting of hearing aids. Ten participants recruited from a larger cohort were included in this study, consisting of 5 participants with diagnosed presbyacusis and 5 frequent conversation partners. A battery of audiological assessments was completed for each participant with presbyacusis. Each participant with presbyacusis was videotaped in conversation at home with their main everyday conversational partner: once before hearing aid fitting and once two months following hearing aid fitting. The conversational interactions before and after hearing aid fitting were analysed using Conversation Analysis. The results of the study revealed that both the people with presbyacusis and the conversation partners used patterns of interaction in instances of mishearings in conversation. The person with presbyacusis shifted gaze direction to show a need for repair. In addition, the conversation partner used physical prompting to gain gaze directed attention from the person with presbyacusis. The person with presbyacusis also made verbal requests for a repair as a result of mishearings. These patterns in interaction showed co-ordination and timing of the repair recognition, initiation and completion by both parties. The phenomena uncovered in this study indicate that the responsibility to monitor and maintain conversation was increasingly placed on the conversation partner of the person with presbyacusis. This could explain why people with presbyacusis and their conversation partners frequently complain of frustration in conversation activities. In the postamplification conversations, no mishearings occurred, suggesting a trend towards fewer mishearings on conversation as a result of amplification of hearing. The research findings contribute to the evidence base concerning the real benefit of digital hearing aids to these elderly clients. The findings of this study can be used to design assessment and intervention tools in the future. / Dissertation (M (Communication Pathology))--University of Pretoria, 2007. / Speech-Language Pathology and Audiology / M (Communication Pathology) / Unrestricted