Made available in DSpace on 2016-08-17T14:52:59Z (GMT). No. of bitstreams: 1 Jose Igor Santos de Oliveira.pdf: 3257339 bytes, checksum: 08e579c1308bf514ca6256e089894a3b (MD5) Previous issue date: 2005-09-21 / Coordenação de Aperfeiçoamento de Pessoal de Nível Superior / The level measurement, besides the applications in reservoirs of industrial processes, as in the industries chemical, pharmaceutical, chemical petroleum, of I refine of alumina etc, it is applicable also in reservoirs exposed outdoors, such as dikes, lakes and ponds, dams and other. To measure level in most of the cases has great impact in the people's safety, of the environment, and of the involved process, besides influencing in the quality of the final product. Level in several ways can be measured that space from a simple float to a sophisticated system for time of flight of waves that processes the information that returns after the emission of a sign. System as that can use several types of waves to take the information of the transmission, the most common are microwaves, infrared and the ultrasonic ones. In that work a method is described that is used of the technique of the time of flight with the use of ultrasonic waves in association with the adaptive filtering to determine the location of the levels of the liquids contained in a container of known height. It is made a study of robustness of the method with base in simulations, through the variation of the relationship sign noise and of the sampling tax, in comparison with the results obtained with the use of Hilbert Transform, Fourier Transform and Wavelet Transform, considering the time of processing and the measurement uncertainty. / A medição de nível, além das aplicações em reservatórios de processos industriais, como nas indústrias químicas, farmacêuticas, petroquímicas, de refino de alumina etc., é aplicável também em reservatórios expostos ao ar livre, tais como diques, lagos e lagoas, barragens e outros. Medir nível na maioria dos casos tem grande impacto na segurança das pessoas, do meio ambiente, e do processo envolvido, além de influenciar na qualidade do produto final. Pode-se medir nível de diversas formas que vão desde um simples flutuador até um sofisticado sistema por Tempo de Vôo de ondas que processa a informação que retorna na forma de eco refletido após a emissão de um sinal. Um Sistema como esse pode utilizar diversos tipos de ondas para levar a informação da transmissão, as mais comuns são microondas, infravermelhas e as ultra-sônicas. Nesse trabalho descreve-se um método que se utiliza da técnica do Tempo de Vôo com o uso de ondas ultra-sônicas em associação com a filtragem adaptativa para determinar a localização dos níveis dos líquidos contidos em um recipiente de altura conhecida. É feito um estudo de robustez do método com base em simulações, através da variação da relação sinal ruído e da taxa de amostragem. Compara-se com os resultados obtidos com o uso da Transformada de Hilbert, Transformada de Fourier e da Transformada Wavelets, considerando o tempo de processamento e a incerteza na medição.

Made available in DSpace on 2016-04-18T21:39:49Z (GMT). No. of bitstreams: 2 Carolina Duca Novaes1.pdf: 3168515 bytes, checksum: 4bc2a4d21f8e08b61653940548fcd0d3 (MD5) Carolina Duca Novaes2.pdf: 2186508 bytes, checksum: 51278edae485082fdb365b9e34ad3fda (MD5) Previous issue date: 2010-02-10 / This dissertation presents details of a structure needed to create a Single Frequency Network (SFN) using low-power transmitters, and a comparison with a network composed of a single transmitter with high power. Among the difficulties to implement networks which distribute signals in SFN, has prioritized the synchronization issue and equalization techniques to achieve better performance of this system. Thus, there was a theoretical research comparing the different methods of timing of single frequency networks and simulations of coverage and performance in order to discuss the implementation of such a network system in the Brazilian digital TV (SBTVD). / Este trabalho apresenta um detalhamento da estrutura necessária para se criar uma rede Single Frequency Network (SFN) utilizando transmissores de baixa potência, bem como um comparativo com uma rede composta por um transmissor único de alta potência. Dentre as dificuldades de se implementar as redes de redistribuição de sinais em SFN foi priorizada a questão do sincronismo e técnicas de equalização que permitam um melhor desempenho desse sistema. Para isso foi realizada uma pesquisa teórica comparativa entre os diferentes métodos de sincronismo de redes de frequência única e simulações de cobertura e desempenho, visando discutir a implementação desse tipo de rede no sistema Brasileiro de TV digital (SBTVD).

Muitas pesquisas tem sido feitas em processamento digital de sinais (PDS) na tentativa de se avaliar o sinal de fala para diagnosticar doenças da laringe. Medidas acústicas têm sido propostas de forma a avaliar indiretamente o trato glotal por meio do sinal de voz coletado através de microfone convencional. Para isso, o modelo paramétrico Liljencrants-Fant (LF) foi desenvolvido para representar o sinal glotal em condições normais e patológicas. Tais parâmetros apresentam vantagens sobre medidas acústicas por possuírem características fisiológicas reais das pregas vocais. Assim, podendo ser empregados para identificação de doenças da laringe. Além da estimação dos parâmetros LF, no domínio do tempo (parâmetros T), a forma de onda da derivativa glotal também pôde ser quantificada através dos parâmetros identificados na literatura por parâmetros R (Rd, Ra, Rk e Rg), parâmetros quocientes Q (SQ, OQ, CQ, AQ e NAQ), parâmetros B1 e B2 que são as extensões de bandas do pulso derivativo LF, e o parâmetro ece, que relaciona os parâmetros &#946 e Ta. Os parâmetros B1 e B2 e ece apesar de serem propostos na literatura, não são encontrados resultados diferentes a essas duas medidas. Os resultados mostraram que os parâmetros B não foram confiáveis na discriminação entre as vozes, por outro lado, o parâmetro ece mostrou-se ser opção na discriminação entre as vozes normais, nódulo e Reinke. O objetivo deste trabalho é direcionar a atenção sobre o sinal glotal, estimando-o automaticamente mediante técnicas de PDS aplicadas ao sinal de fala, visando extrair parâmetros que identifiquem as condições normais e patológicas da laringe. Por fim foram propostos os parâmetros TRp e TRs, visando dissociar os efeitos de primeira ordem dos de ordem superior na fase de retorno do pulso glotal com a finalidade de estimar a real não-linearidade do sub-sistema glotal, retratando as condições normais e patológicas da laringe. Por fim foram propostos os parâmetros TRp e TRs, visando dissociar os efeitos de primeira ordem dos de ordem superior na fase de retorno do pulso glotal com a finalidade de estimar a real não-linearidade do sub-sistema glotal, retratando as condições fisiológicas do movimento das pregas vocais. Com um nível de confiança de 95%, o parâmetro de primeira ordem (TRp) é efetivo na discriminação do Edema de Reinke, porém mostrou-se ineficaz na detecção do nódulo. Em relação ao parâmetro de ordem superior, conclui-se que o TRs é um excelente detetor de vozes patológicas (nódulo e Edema de Reinke), porém não é capaz de discriminar as patologias. / Many researches has been conducted in digital signal processing (DSP) atempting to evaluate the physiological conditions of larynx. Acoustical parameters have been proposed to evaluate the glotal tract from voice signal. One technique proposed is the Liljencrants-Fant model (LF) developed to represent normal and pathologic conditions of the larynx. Those parameters compare favourably as far as real physiologic characteristic of vocal folds is concerned. So, a primary use of the model is the larynx pathologic identification. Beyond LF parameters estimation, (T parameters in the time domain), the waveform of glotal pulse derivative also can be quantified through, R parameters (Rd, Ra, Rk and Rg), quocient parameters (SQ, OQ, CQ, AQ and NAQ), B parameters (B1 and B2) that are band extension of the LF glotal pulse derivative and the ece parameter that in fact, is a relationship between &#946 and Ta. Although proposed in the literature, no results are found, related to B and ece parameters. Our founds show that B parameters do not present good results in voice discrimination, however, ece parameter seems to be good option to discriminate normal voice, nodulo and Reinke edema. The main purpose of this work is to estimate the glotal signal from the voice signal using DSP techniques in order to obtain parameters that identifies the physiological larynx condition. In order to estimate the shape of return phase of glotal pulse, twoparameters have been proposed in this work. The first one evaluates the pulse (TRp, in other words, the first order component of the return phase. The second is responsible to evaluate superior orders components of the return phase (TRs), i.e, the non-linear component of the glotal pulse. With 95% of confidence level, TRp is effective in Reinke edema discrimination however it is inefficient for nodule e dection. By the other hand, the TRs parameter works well to detect pathologic voice however is unable to discriminated them.

Submitted by Renata Lopes (renatasil82@gmail.com) on 2017-04-19T17:59:14Z No. of bitstreams: 1 carloshenriquenascimentomartins.pdf: 4613305 bytes, checksum: 54c109a7e2d55a96f74c2f40d23c074b (MD5) / Approved for entry into archive by Adriana Oliveira (adriana.oliveira@ufjf.edu.br) on 2017-04-20T12:29:45Z (GMT) No. of bitstreams: 1 carloshenriquenascimentomartins.pdf: 4613305 bytes, checksum: 54c109a7e2d55a96f74c2f40d23c074b (MD5) / Made available in DSpace on 2017-04-20T12:29:45Z (GMT). No. of bitstreams: 1 carloshenriquenascimentomartins.pdf: 4613305 bytes, checksum: 54c109a7e2d55a96f74c2f40d23c074b (MD5) Previous issue date: 2011-04-08 / Este trabalho tem por objetivo principal apresentar o desenvolvimento de plataformas eletrônicas de processamento de sinais de alto desempenho para monitoramento do sistema elétricos de potência. No trabalho são discutidas arquiteturas de hardware para três aplicações em sistemas de potência: analisador fasorial (Phasor Measurement Unit - PMU), analisador de qualidade de energia elétrica (QEE) e analisador de harmônicos variantes no tempo (AHVT). Além disso são todos os conceitos de eletrônica digital e analógica envolvidos na concepção deste projeto, tomando como base equipamentos de mercado, a literatura pertinente e as normas reguladoras para dispositivos de análise de parâmetros elétricos. No projeto é abordado principalmente a implementação de hardware, que envolve implementação de estruturas de conversão Analógico Digital, filtro anti-aliasing, condicionamento de sinais, processamento e gerenciamento de dados e finalmente meios de comunicação. O hardware foi testado utilizando algoritmos básicos de processamento de sinais sendo apresentado casos reais de monitoramento dos parâmetros do sinal elétrico e uma versão inicial do AHVT. / This work has the aim to present the development of electronic platforms for signal processing for high performance electric power monitoring system. At this work are discussed hardware architectures for three power systems applications: phasor measurement unit (PMU), Power Quality Analyzer (PQ) and Time Varying Harmonic Analyzer (TVHA). Also are explained all features of analog and digital electronics involved in the design of this project, based on commercial devices, the literature and regulatory standards for electrical parameters devices. The project is addressed principally to hardware implementation, which involves implementation of structures such as the Analog to Digital conversion, anti-aliasing filter, signal conditioning, processing and data management and communication. The hardware is tested using basic digital signal algorithms and real cases of parameters monitoring are presented. Furthermore prototype version of the TVHA is presented.

Submitted by Renata Lopes (renatasil82@gmail.com) on 2017-04-20T15:14:06Z No. of bitstreams: 1 bernardosottomaiorperalva.pdf: 4608167 bytes, checksum: c63c1f7fc453965f36158791fb85964e (MD5) / Approved for entry into archive by Adriana Oliveira (adriana.oliveira@ufjf.edu.br) on 2017-04-24T16:49:03Z (GMT) No. of bitstreams: 1 bernardosottomaiorperalva.pdf: 4608167 bytes, checksum: c63c1f7fc453965f36158791fb85964e (MD5) / Made available in DSpace on 2017-04-24T16:49:04Z (GMT). No. of bitstreams: 1 bernardosottomaiorperalva.pdf: 4608167 bytes, checksum: c63c1f7fc453965f36158791fb85964e (MD5) Previous issue date: 2012-05-07 / CAPES - Coordenação de Aperfeiçoamento de Pessoal de Nível Superior / Nesta dissertação, são apresentados métodos para detecção de sinais e estimação de energia para calorimetria de altas energias aplicados no calorímetro hadrônico (TileCal) do ATLAS. A energia depositada em cada célula do calorímetro é adquirida por dois canais eletrônicos de leitura e é estimada, separadamente, através da reconstrução da amplitude do pulso digitalizado amostrado a cada 25 ns. Este trabalho explora a aplicabilidade de uma aproximação do Filtro Casado no ambiente do TileCal para detectar sinais e estimar sua amplitude. Além disso, este trabalho explora o impacto na detecção de eventos válidos e estimação da amplitude quando somam-se os sinais referentes à mesma célula antes da aplicação do filtro. O método proposto é comparado com o Filtro Ótimo atualmente utilizado pelo TileCal para reconstrução de energia. Os resultados para dados simulados e de colisão mostram que, para condições em que a linha de base do sinal de entrada pode ser considerada estacionária, a técnica proposta apresenta uma melhor eficiência de detecção e estimação do que a alcançada pelo Filtro Ótimo empregada no TileCal. / The Tile Barrel Calorimeter (TileCal) is the central section of the hadronic calorimeter of ATLAS at LHC. The energy deposited in each cell of the calorimeter is read out by two electronic channels for redundancy and is estimated, per channel, by reconstructing the amplitude of the digitized signal pulse sampled every 25 ns. This work presents signal detection and energy estimation methods for high energy calorimetry, applied to the TileCal environment. It investigates the applicability of a Matched Filter and, furthermore, it explores the impact when summing the signals belonging to the same cell before the estimating and detecting procedures. The proposed method is compared to the Optimal Filter algorithm, that is currently been used at TileCal for energy reconstruction. The results for simulated and collision data sets showed that for conditions where the signal pedestal could be considered stationary, the proposed method achieves better detection and estimation efficiencies than the Optimal Filter technique employed in TileCal.

A channel change in digital television is usually conducted at a reference frame, which are sent at certain intervals. A higher compression ratio could however be obtained by sending reference frames at arbitrary long intervals. This would on the other hand increase the average channel change time for the end user. This thesis investigates various approaches for reducing the average channel change time while using arbitrary long intervals between reference frames, and presents an implementation and evaluation of one of these methods, called Baydec. The approach of Baydec for solving the channel switch problem is to statistically estimate what the original image looked like, starting with an incoming P-frame and estimate an image between the original and current image. Baydec gathers statistical data from typical video sequences and calculates expected likelihood for estimation. Further on it uses the Simulated Annealing search method to maximise the likelihood function. This method is more general than the requirements of this thesis. It is not only applicable to channel switches between video streams, but can also be used for random access in general. Baydec could also be used if an I-frame is dropped in a video stream. However, Baydec has so far shown only theoretical result, but very small visual improvements. Baydec produces images with better PSNR than without the method in some cases, but the visual impression is not better than for the motion compensated residual images. Some examples of future work to improve Baydec is also presented.

Les travaux de recherche de cette thèse de doctorat s’inscrivent dans le cadre de la conception d’unconvertisseur analogique-numérique (ADC, Analog-to-Digital Converter) large bande et à haute résolution afinde numériser plusieurs standards de communications sans fil. Il répond ainsi au concept de la radio logiciellerestreinte (SDR, Software Defined Radio). L’objectif visé est la reconfigurabilité par logiciel et l’intégrabilité envue d’un système radio multistandard. Les ADCs à sur-échantillonnage de type sigma-delta () s’avèrent debons candidats dans ce contexte de réception SDR multistandard en raison de leur précision accrue. Bien queleur bande passante soit réduite, il est possible de les utiliser dans une architecture en parallèle permettantd’élargir la bande passante. Nous nous proposons alors dans cette thèse de dimensionner et d’implanter unADC parallèle à décomposition fréquentielle (FBD) basé sur des modulateurs  à temps-discret pour unrécepteur SDR supportant les standards E-GSM, UMTS et IEEE802.11a. La nouveauté dans l’architectureproposée est qu’il est programmable, la numérisation d’un signal issu d’un standard donné se réalise enactivant seulement les branches concernées de l’architecture parallèle avec des sous-bandes defonctionnement et une fréquence d’échantillonnage spécifiée. De plus, le partage fréquentiel des sous-bandesest non uniforme. Après validation du dimensionnement théorique par simulation, l’étage en bande de base aété dimensionné. Cette étude conduit à la définition d’un filtre anti-repliement passif unique d’ordre 6 et detype Butterworth, permettant l’élimination du circuit de contrôle de gain automatique (AGC). L’architectureFBD requière un traitement numérique permettant de combiner les signaux à la sortie des branches enparallèle pour reconstruire le signal de sortie finale. Un dimensionnement optimisé de cet étage numérique àbase de démodulation a été proposé. La synthèse de l’étage en bande de base a montré des problèmes destabilité des modulateurs . Pour y remédier, une solution basée sur la modification de la fonction detransfert du signal (STF) afin de filtrer les signaux hors bande d’intérêt par branche a été élaborée. Unediscontinuité de phase a été également constatée dans le signal de sortie reconstruit. Une solution deraccordement de phase a été proposée. L’étude analytique et la conception niveau système ont étécomplétées par une implantation de la reconstruction numérique de l’ADC parallèle. Deux flots de conceptionont été considérés, un associé au FPGA et l’autre indépendant de la cible choisie (VHDL standard).L’architecture proposée a été validée sur un FPGA Xilinx de type VIRTEX6. Une dynamique de 74 dB a étémesurée pour le cas d’étude UMTS, ce qui est compatible avec celle requise du standard UMTS. / The work presented in this Ph.D. dissertation deals with the design of a wideband and accurate Analog-to-Digital Converter (ADC) able to digitize signals of different wireless communications standards. Thereby, itresponds to the Software Defined Radio concept (SDR). The purpose is reconfigurability by software andintegrability of the multistandard radio terminal. Oversampling  (Sigma Delta) ADCs have been interestingcandidates in this context of multistandard SDR reception thanks to their high accuracy. Although they presentlimited operating bandwidth, it is possible to use them in a parallel architecture thus the bandwidth isextended. Therefore, we propose in this work the design and implementation of a parallel frequency banddecomposition ADC based on Discrete-time  modulators in an SDR receiver handling E-GSM, UMTS andIEEE802.11a standard signals. The novelty of this proposed architecture is its programmability. Where,according to the selected standard digitization is made by activating only required branches are activated withspecified sub-bandwidths and sampling frequency. In addition the frequency division plan is non-uniform.After validation of the theoretical design by simulation, the overall baseband stage has been designed. Resultsof this study have led to a single passive 6th order Butterworth anti-aliasing filter (AAF) permitting theelimination of the automatic gain control circuit (AGC) which is an analog component. FBD architecturerequires digital processing able to recombine parallel branches outputs signals in order to reconstruct the finaloutput signal. An optimized design of this digital reconstruction signal stage has been proposed. Synthesis ofthe baseband stage has revealed  modulators stability problems. To deal with this problem, a solution basedon non-unitary STF has been elaborated. Indeed, phase mismatches have been shown in the recombinedoutput signal and they have been corrected in the digital stage. Analytic study and system level design havebeen completed by an implementation of the parallel ADC digital reconstruction stage. Two design flows havebeen considered, one associated to the FPGA and another independent of the chosen target (standard VHDL).Proposed architecture has been validated using a VIRTEX6 FPGA Xilinx target. A dynamic range over 74 dB hasbeen measured for UMTS use case, which responds to the dynamic range required by this standard.

Software Defined Radios (SDRs) and Cognitive Radios (CRs) pave the way for next-generation radio technology. They promise versatility, flexibility and cognition which can revolutionize communications systems. However they present greater challenges to the design of radio frequency (RF) front-ends. RF front-ends for the radios in use today are narrow-band in their frequency response and are optimized and tuned to the carrier frequency of interest. SDRs and CRs demand front-ends which are versatile, configurable, tunable and be capable of transmitting and receiving signals with different bandwidths and modulation schemes. Integrating power amplifiers (PAs) with transmitters in CMOS has many advantages and challenges. This thesis deals with the design of an RF transmitter front-end for SDRs and CRs in CMOS. The thesis begins with an introduction to SDRs and the requirements they place on transmitters and the challenges involved in designing them in CMOS. After a brief overview of the existing techniques, the proposed architecture is presented and explained. A digitally intensive transmitter solution is proposed. The transmitter covers a wide frequency range of 750 MHz to 2.5 GHz. The inputs to the proposed transmitter are in-phase and quadrature (I & Q) data bit streams. Multiple stages of up-sampling and filtering are used to remove all spurs in the spectrum such that only the harmonics of the carrier remain. Differential rail-to-rail quadrature clocks are generated from a continuous wave signal at twice the carrier frequency. The clocks are corrected for their duty cycle and quadrature impairments. The heart of the transmitter is an integrated reconfigurable CMOS power amplifier (PA). A methodology to design reconfigurable Class E PAs with a series fixed inductor has been presented. A CMOS power amplifier that can span a wide frequency range with sufficient output power and efficiency, supporting varying envelope complex modulation signals, with good linearity has been designed. Digital pre-distortion (DPD) is used to linearize the PA. The full transmitter and the clock correction blocks have been designed and fabricated in a commercial 130-nm CMOS process and experimentally characterized. The PA delivers a maximum power of 13 dBm with an efficiency of 27% at 1 GHz. While transmitting a 16-QAM signal at 1 GHz, the measured EVM is 4%. It delivers a maximum power of around 11-13 dBm from 750 MHz to 1.5 GHz and up to 6.5 dBm of power till 2.5 GHz. Comparing the proposed system with recently published literature, it can be seen that the proposed design is one of the very few transmitters which has an integrated matching network, tunable across the frequency range. The proposed PA produces the highest output power and with largest efficiency for systems with on-chip output networks.

A débit de données élevé, typiquement supérieur à 10 Gsymboles/s, les lignes de<p>télécommunication optique à fibre monomode souffrent de façon accrue des distorsions<p>inhérentes à la fibre et à l’architecture de transmission. Nous pouvons classer les<p>effets de fibre en plusieurs catégories:<p>– Les effets linéaires. La dispersion chromatique est entraînée par la dépendance en<p>fréquence de l’indice de réfraction de la fibre. Il en résulte un élargissement des<p>bits optiques. La dispersion des modes de polarisation prend son origine dans<p>la biréfringence de la fibre. La modélisation de cet effet est compliquée par son<p>caractère stochastique et variable dans le temps.<p>– Les effets non linéaires prennent leur origine dans un indice de réfraction de<p>fibre qui dépend du champ optique. Ces effets peuvent être classés en deux<p>catégories. Premièrement, les effets intérieurs à un canal dont le plus influant<p>est l’automodulation de phase qui découle de l’effet Kerr optique :l’intensité<p>d’une impulsion lumineuse influence sa propre propagation. Deuxièmement, il<p>existe des conséquences de l’effet Kerr par lesquelles les différents canaux, se<p>propageant au sein de la même fibre, s’influencent mutuellement. Le phénomène<p>le plus influent parmi ces derniers est la modulation de phase croisée :l’intensité<p>d’un canal influence la propagation dans un canal voisin.<p>– Les pertes par diffusion Rayleigh sont compensées par les amplificateurs distribués<p>le long de la ligne de transmission. L’amplification optique par l’intermédiaire<p>d’émission stimulée dans des dispositifs dopés aux ions Erbium est<p>accompagnée d’émission spontanée amplifiée. Ceci entraîne la présence d’un<p>bruit blanc gaussien se superposant au signal à transmettre.<p>– La gestion des canaux dans le réseau optique implique la présence dans les noeuds<p>du réseau de filtres de sélection, des multiplexeurs et démultiplexeurs.<p>Nous examinerons aussi les effets de ligne non inhérents à la fibre mais à l’architecture<p>de transmission. Les modèles de l’émetteur et du récepteur représentent les imperfections<p>d’implémentation des composants optiques et électroniques.<p>Un premier objectif est de définir et évaluer un format de modulation robuste aux<p>imperfections introduites sur le signal par la fibre optique et par l’émetteur/récepteur.<p>Deux caractéristiques fondamentales du format de modulation, determinants pour la<p>performance du système, sont étudiés dans ce travail :<p>– La forme d’ onde. Les symboles complexes d’information sont mis en forme par<p>un filtre passe-bas dont le profil influence la robustesse du signal vis-à-vis des<p>effets de ligne.<p>– La distribution des fréquences porteuses. Les canaux de communication sont<p>disposés sur une grille fréquentielle qui peut être définie de manière électronique<p>par traitement de signal, de manière optique ou dans une configuration hybride.<p>Lorsque des porteuses optiques sont utilisées, le bruit de phase relatif entre lasers<p>entraîne des effets d’ influence croisée entre canaux. En revanche, les limites des<p>implémentations électroniques sont données par la puissance des architectures<p>numériques.<p>Le deuxième objectif est de concevoir des techniques de traitement numérique du<p>signal implémentées après échantillonnage au récepteur afin de retrouver l’information<p>transmise. Les fonctions suivantes seront implémentées au récepteur :<p>– Les techniques d’estimation et d’égalisation des effets linéaires introduits par la<p>fibre optique et par l’émetteur et le récepteur. Le principe de l’égalisation dans<p>le domaine fréquentiel est de transformer le canal convolutif dans le domaine<p>temporel en un canal multiplicatif qui peut dès lors être compensé à une faible<p>complexité de calcul par des multiplications scalaires. Les blocs de symboles<p>émis doivent être rendus cycliques par l’ajout de redondance sous la forme d’un<p>préfixe cyclique ou d’une séquence d’apprentissage. Les techniques d’égalisation<p>seront comparées en termes de performance (taux d’erreurs binaires, efficacité<p>spectrale) et en termes de complexité de calcul. Ce dernier aspect est particulièrement<p>crucial en vue de l’optimisation de la consommation énergétique du<p>système conçu.<p>– Les techniques de synchronisation des signaux en temps/fréquence. Avant de<p>pouvoir égaliser les effets linéaires introduits dans la fibre, le signal reçu devra<p>être synchronisé en temps et en fréquence sur le signal envoyé. La synchronisation<p>est généralement accomplie en deux étapes principales :l’acquisition réalisée<p>avant de recevoir les symboles d’information don’t l’objectif est une première<p>estimation/compensation des effets de manière "grossière", le tracking réalisé en<p>parallèle à l’estimation des symboles d’information dont l’objectif est l’estimation<p>/compensation des effets de manière "fine". Les algorithmes d’acquisition et<p>de tracking peuvent nécessiter l’envoi d’informations connues du récepteur.<p>– Les techniques d’estimation et de compensation des imperfections de fonctionnement<p>de l’émetteur et du récepteur. Une structure de compensation des effets<p>introduits par les composants optiques et électroniques sera développée afin de<p>relâcher les contraintes d’implémentation de l’émetteur et du récepteur.<p>Etant donné la très haute cadence à laquelle les échantillons du signal sont produits<p>(plusieurs dizaines de Gech/s), une attention particulière est portée à la complexité de<p>calcul des algorithmes proposés. / Doctorat en Sciences de l'ingénieur / info:eu-repo/semantics/nonPublished

Conventional hearing aid circuitry is often unable to provide children with hearing loss with sufficient high frequency information in order to develop adequate oral language skills due to the risk of acoustic feedback and the narrower frequency spectrum of conventional amplification. The purpose of this study was to investigate word recognition abilities of children with moderate-to-severe hearing loss using hearing aids with linear frequency transposition. Seven children with moderate-to-severe sensorineural hearing loss between the ages of 5 years 0 months and 7 years 11 months were selected for the participant group. Word recognition assessments were first performed with the participants using their own previous generation digital signal processing hearing aids. Twenty-five-word lists from the Word Intelligibility by Picture Identification (WIPI) test were presented to the participants in three test conditions, namely: at 55 dB HL in quiet, 55 dB HL with a +5 dB signal-to-noise ratio (SNR) and at 35 dB HL. The participants were then fitted with an ISP-based hearing aid without linear frequency transposition, and the word recognition assessments were repeated with different WIPI word lists under the same conditions as the first assessment. Linear frequency transposition was then activated in the ISP-based hearing aid and different WIPI word lists were presented once more under identical conditions as the previous assessments. A 12-day acclimatization period was allowed between assessments, and all fittings were verified according to the DSL v5 fitting algorithm. Results indicated a significant increase of more than 12% in word recognition score for some of the participants when they used the ISP-based hearing aid with linear frequency transposition. A significant decrease was also seen for some of the participants when they used the ISP-based hearing aid with linear frequency transposition, but all participants presented with better word recognition scores when they used the ISP-based hearing aids without linear frequency transposition compared to their previous generation digital signal processing hearing aids. This study has shown that linear frequency transposition may improve the word recognition skills of some children with moderate-to-severe sensorineural hearing loss, and more research is needed to explore the criteria that can be used to determine candidacy for linear frequency transposition. / Dissertation (MCommunication Pathology)--University of Pretoria, 2010. / Speech-Language Pathology and Audiology / Unrestricted