Direct-Down Conversion (DDC) principle based transceiver architectures are of interest to meet the diverse needs of present and future wireless systems. DDC transceivers have a simple structure with fewer analog components and offer low-cost, flexible and multi-standard solutions. However, DDC transceivers have certain circuit impairments affecting their performance in wide-band, high data rate and multi-user systems. IQ imbalance is one of the problems of DDC transceivers that limits their image rejection capabilities. Compensation techniques for frequency independent IQI arising due to gain and phase mismatches of the mixers in the I/Q paths of the transceiver have been widely discussed in the literature. However for wideband multi-channel transceivers, it is becoming increasingly important to address frequency dependent IQI arising due to mismatches in the analog I/Q lowpass filters. A hardware-efficient and standard independent digital estimation and compensation technique for frequency dependent IQI is introduced which is also capable of tracking time-varying IQI changes. The technique is blind and adaptive in nature, based on the second order statistical properties of complex random signals such as properness/circularity. A detailed performance analysis of the introduced technique is executed through computer simulations for various real-time operating scenarios. A novel technique for finding the optimal number of taps required for the adaptive IQI compensation filter is proposed and the performance of this technique is validated. In addition, a metric for the measure of properness is developed and used for error power and step size analysis. / Master of Science / A wireless transceiver consists of two major building blocks namely the RF front-end and digital baseband. The front-end performs functions such as frequency conversion, filtering, and amplification. Impurities because of deep-submicron fabrication lead to non-idealities of the front-end components which limit their accuracy and affect the performance of the overall transceiver. Complex (I/Q) mixing of baseband signals is preferred over real mixing because of its inherent trait of bandwidth efficiency. The I/Q paths enabling this complex mixing in the front-end may not be exactly identical thereby disturbing the perfect orthogonality of inphase and quadrature components leading to IQ Imbalance. The resultant IQ imbalance leads to an image of the signal formed at its mirror frequencies. Imbalances arising from mixers lead to an image of constant strength whereas I/Q low-pass filter mismatches lead to an image of varying strength across the Nyquist range. In addition, temperature effects cause slow variation in IQ imbalance with time. In this thesis a hardware efficient and standard-independent technique is introduced to compensate for performance degrading IQ imbalance. The technique is blind and adaptive in nature and uses second order statistical signal properties like circularity or properness for IQ imbalance estimation. The contribution of this work, which gives a key insight into the optimal number of taps required for the adaptive compensation filter improves the state-of-the-art technique. The performance of the technique is evaluated under various scenarios of interest and a detailed analysis of the results is presented.

Les travaux de cette thèse portent sur l'égalisation des canaux de transmission pour des modulations mono-porteuses et multi-porteuses. Dans le cadre de l'égalisation, nous nous intéressons, plus précisément, à l'égalisation aveugle. Tout d'abord, nous décrivons les différents éléments constituants une chaîne de communication, et les différents types de modulations mono-porteuses et muti-porteuses (OFDM). Ensuite, nous faisons un état de l'art sur les méthodes de l'égalisation aveugle pour une modulation mono-porteuse. Nous proposons par la suite un algorithme d'égalisation aveugle en présence de perturbations bornées. Ensuite, nous fournissons une analyse de stabilité et de convergence de l'algorithme proposé. Dans le cadre de la modulation multi-porteuse, nous présentons, dans un premier temps, un état de l'art sur les techniques d'égalisation aveugle pour le système OFDM. Ensuite, nous adaptons l'algorithme proposé pour le système OFDM pour des canaux à trajets multiples, en particulier les canaux Raleigh et Rice. Les performances de l'algorithme proposé sont illustrées à travers plusieurs exemples en simulation tout au long de la thèse. / The work of this thesis deals with the equalization of the transmission channels for a single-carrier and multi-carrier modulation. In the context of equalization, we focus precisely on the blind equalization. First, we give a description of various elements constituting a communication chain, a description of different types of single-carrier modulations and a description of a multi-carrier modulation (OFDM). Then, we give an overview of the blind equalization methods for a single-carrier modulation. We propose subsequently a blind equalization algorithm in the presence of a bounded perturbation. Next, we provide stability and convergence analysis of the proposed method. In the context of multi-carrier modulation, we first present an overview of the blind equalization techniques for the OFDM system. Next, we adapt the proposed method for the OFDM system under multipath channels, especially the Raleigh and the Rice channels. Performance of the proposed algorithm have been illustrated in simulation by considering many examples throughout this thesis.

The Decision Feedback Equalizer (DFE) is a known method of channel equalization which has performance superiority over linear equalizer. The best performance of DFE is obtained, commonly, with training period which is used for initial acquisiton of channel or recovering changes in the channel. The training period requires a training sequence which reduces the bit transmission rate or is not possible to send in most of the situations. So, it is desirable to skip the training period. The Unsupervised (Blind) DFE (UDFE) is such a DFE scheme which has no training period. The UDFE has two modes of operation. In one mode, the UDFE uses Constant Modulus Algorithm (CMA) to perform channel acquisition, blindly. The other mode is the same as classical decision-directed DFE. This thesis compares the performances of the classical trained DFE method and the UDFE. The performance comparison is done in some channel environments with the problem of timing error present in the received data bearing signal. The computer aided simulations are done for two stationary channels, a time-varying channel and a frequency selective Rayleigh fading channel to test the performance of the relevant equalizers. The test results are evaluted according to mean square error (MSE), bit-error rate (BER), residual intersymbol interference (RISI) performances and equalizer output diagrams. The test results show that the UDFE has an equal or, sometimes, better performance compared to the trained DFE methods. The two modes of UDFE enable it to solve the absence of training sequence.

Please read the abstract in the section 00front of this document / Dissertation (M Eng (Electronic Engineering))--University of Pretoria, 2006. / Electrical, Electronic and Computer Engineering / unrestricted

We consider the problem of equalizing data blocks of signals, which have been transmitted over an aeronautical channel using two different modulation schemes. The equalization is performed using the block-processing constant modulus algorithm (CMA), and in order to achieve real-time processing a Fast Fourier Transform (FFT) is used to compute the gradient of this cost function during equalization. The equalizer length is chosen to be five times of the channel length. For the first experiment, we present the result of equalizing a set of measured data, which was modulated and transmitted using the iNET packet structure with SOQPSK modulation. In this case, the CMA equalizer is first initialized using MMSE and the equalizer coefficients are then updated once, using each entire block (iNET packet). In the second experiment, we apply the FFT-based block processing equalizer to received data blocks of QPSK signals, which have been randomly generated and transmitted over an aeronautical channel. A modified constant modulus algorithm and alphabet matched algorithm (CMA + AMA) equalizer is used to recover these data blocks. For this case of QPSK signals, the equalizer performance is evaluated over 500 Monte Carlo runs, using the average symbol error rate (SER).

Approved for public release, distribution is unlimited / Direct-Sequence Spread Spectrum (DSSS) modulation is being advanced as the physical-layer basis for Seaweb undersea acoustic networking. DSSS meets the need for channel tolerance, transmission security, and multi-user access. This thesis investigates the performance of subspace-decomposition blind-equalization algorithms as alternatives to RAKE processing of DSSS signals. This approach is tailored for superior performance in time-dispersive and frequency-dispersive channels characteristic of ocean acoustic propagation. Transmitter and receiver structures are implemented in Matlab and evaluated with a statistics-based model of a doubly spread channel with additive noise. Receiver performance is examined using Monte Carlo simulation. Biterror rates versus signal-to-noise ratio are presented for various multipath assumptions, noise assumptions, and receiver synchronization assumptions. / Lieutenant, Hellenic Navy

La mayor parte de los sistemas de comunicaciones requieren el conocimiento previo del canal, el cual se suele estimar a partir de una secuencia de entrenamiento. Sin embargo, la transmisión de símbolos piloto se traduce en una reducción de la eficiencia espectral del sistema, lo que imposibilita que se alcancen los límites predichos por la Teoría de la Información. Este problema ha motivado el desarrollo de un gran número de técnicas para la estima e igualación ciega de canal, es decir, para la obtención del canal o la fuente sin necesidad de transmitir una señal de entrenamiento. Normalmente, estas técnicas se basan en el conocimiento previo de ciertas características de la señal, tales como su pertenencia a un alfabeto finito, o sus estadísticos de orden superior. Sin embargo, en el caso de sistemas de múltiples entradas y salidas (MIMO), se ha demostrado que los estadísticos de segundo orden de las observaciones proporcionan la información suficiente para resolver el problema ciego.El objetivo de esta Tesis consiste en la obtención de nuevas técnicas para la estima e igualación ciega de canales MIMO, tanto en sistemas con redundancia espacial, como en casos más generales en los que las fuentes no presentan ningún tipo particular de estructura. De manera general, los métodos propuestos se basan en los estadísticos de segundo orden de las observaciones. Sin embargo, las técnicas se presentan desde un punto de vista determinista, es decir, los algoritmos propuestos explotan directamente la estructura de las matrices de datos, lo que permite obtener resultados más precisos cuando se dispone de un número reducido de observaciones. Adicionalmente, la reformulación de los criterios propuestos como problemas clásicos del análisis estadístico de señales, ha permitido la obtención de algoritmos adaptativos eficientes para la estima e igualación de canales MIMO. En primer lugar se aborda el caso de sistemas sin redundancia. Más concretamente, se analiza el problema de igualación ciega de canales MIMO selectivos en frecuencia, el cual se reformula como un conjunto de problemas de análisis de correlaciones canónicas (CCA). La solución de los problemas CCA se puede obtener de manera directa mediante un problema de autovalores generalizado. Además, en esta Tesis se presenta un algoritmo adaptativo basado en la reformulación de CCA como un conjunto de problemas de regresión lineal acoplados. De esta manera, se obtienen nuevos algoritmos bloque y adaptativos para la igualación ciega de canales MIMO de una manera sencilla. Finalmente, el método propuesto se basa, como muchas otras técnicas ciegas, en el conocimiento a priori del orden del canal, lo que constituye un problema casi tan complicado como el de la estima o igualación ciega. Así, en el caso de canales de una entrada y varias salidas (SIMO), la combinación de la técnica propuesta con otros métodos para la estima ciega del canal permite obtener un nuevo criterio para extracción del orden de este tipo de canalesEn segundo lugar se considera el problema de estima ciega de canal en sistemas con algún tipo de redundancia o estructura espacial, con especial interés en el caso de sistemas con codificación espacio-temporal por bloques (STBC). Específicamente, se propone una nueva técnica para la estima ciega del canal, cuya complejidad se reduce a la extracción del autovector principal de una matriz de correlación modificada. El principal problema asociado a este tipo de sistemas viene dado por la existencia de ciertas ambigüedades a la hora de estimar el canal. En esta Tesis se plantea el problema de identificabilidad de una manera general, y en el caso de códigos ortogonales (OSTBCs) se presentan varios nuevos teoremas que aseguran la identificabilidad del canal en un gran número de casos. Adicionalmente, se proponen varias técnicas para la resolución de las ambigüedades, tanto en el caso OSTBC como para códigos más generales. En concreto, se introduce el concepto de diversidad de código, que consiste en la combinación de varios códigos STBC. Esta técnica permite resolver las indeterminaciones asociadas a un gran número de problemas, y en su versión más sencilla se reduce a una precodificación no redundante consistente en una simple rotación o permutación de las antenas transmisoras.En definitiva, en esta Tesis se abordan los problemas de estima e igualación ciega de canal en sistemas MIMO, y se presentan varias técnicas ciegas, cuyas prestaciones se evalúan mediante un gran número de ejemplos de simulación. / The majority of communication systems need the previous knowledge of the channel, which is usually estimated by means of a training sequence. However, the transmission of pilot symbols provokes a reduction in bandwidth efficiency, which precludes the system from reaching the limits predicted by the Information Theory. This problem has motivated the development of a large number of blind channel estimation and equalization techniques, which are able to obtain the channel or the source without the need of transmitting a training signal. Usually, these techniques are based on the previous knowledge of certain properties of the signal, such as its belonging to a finite alphabet, or its higher-order statistics. However, in the case of multiple-input multiple-output (MIMO) systems, it has been proven that the second order statistics of the observations provide the sufficient information for solving the blind problem.The aim of this Thesis is the development of new blind MIMO channel estimation and equalization techniques, both in systems with spatial redundancy, and in more general cases where the sources do not have any particular spatial structure. In general, the proposed methods are based on the second order statistics of the observations. However, the techniques are presented from a deterministic point of view, i.e., the proposed algorithms directly exploit the structure of the data matrices, which allows us to obtain more accurate results when only a reduced number of observations is available. Additionally, the reformulation of the proposed criteria as classical statistical signal processing problems is exploited to obtain efficient adaptive algorithms for MIMO channel estimation and equalization.Firstly, we consider the case of systems without spatial redundancy. Specifically, we analyze the problem of blind equalization of frequency selective MIMO channels, which is reformulated as a set of canonical correlation analysis (CCA) problems. The solution of the CCA problems can be obtained by means of a generalized eigenvalue problem. In this Thesis, we present a new adaptive algorithm based on the reformulation of CCA as a set of coupled linear regression problems. Therefore, new batch and adaptive algorithms for blind MIMO channel equalization are easily obtained. Finally, the proposed method, as well as many other blind techniques, is based on the previous knowledge of the channel order, which is a problem nearly as complicated as the blind channel estimation or equalization. Thus, in the case of single-input multiple-output (SIMO) channels, the combination of the proposed technique with other blind channel estimation methods provides a new criterion for the order extraction of this class of channels.Secondly, we consider the problem of blind channel estimation in systems with some kind of redundancy or spatial structure, with special interest in space-time block coded (STBC) systems. Specifically, a new blind channel estimation technique is proposed, whose computational complexity reduces to the extraction of the principal eigenvector of a modified correlation matrix. The main problem in these cases is due to the existence of certain ambiguities associated to the blind channel estimation problem. In this Thesis the general identifiability problem is formulated and, in the case of orthogonal codes (OSTBCs), we present several new theorems which ensure the channel identifiability in a large number of cases. Additionally, several techniques for the resolution of the ambiguities are proposed, both in the OSTBC case as well as for more general codes. In particular, we introduce the concept of code diversity, which consists in the combination of several STBCs. This technique avoids the ambiguities associated to a large number of problems, and in its simplest version it reduces to a non-redundant precoding consisting of a single rotation or permutation of the transmit antennas.In summary, in this Thesis the blind MIMO channel estimation and equalization problems are analyzed, and several blind techniques are presented, whose performance is evaluated by means of a large number of simulation examples.

Submitted by Geyciane Santos (geyciane_thamires@hotmail.com) on 2015-07-23T12:57:35Z No. of bitstreams: 1 Dissertação - Mauro Lopes de Freitas.pdf: 1646151 bytes, checksum: a4f17991e9db7da2871b0c711b18f484 (MD5) / Approved for entry into archive by Divisão de Documentação/BC Biblioteca Central (ddbc@ufam.edu.br) on 2015-07-23T18:44:37Z (GMT) No. of bitstreams: 1 Dissertação - Mauro Lopes de Freitas.pdf: 1646151 bytes, checksum: a4f17991e9db7da2871b0c711b18f484 (MD5) / Approved for entry into archive by Divisão de Documentação/BC Biblioteca Central (ddbc@ufam.edu.br) on 2015-07-23T18:47:54Z (GMT) No. of bitstreams: 1 Dissertação - Mauro Lopes de Freitas.pdf: 1646151 bytes, checksum: a4f17991e9db7da2871b0c711b18f484 (MD5) / Made available in DSpace on 2015-07-23T18:47:54Z (GMT). No. of bitstreams: 1 Dissertação - Mauro Lopes de Freitas.pdf: 1646151 bytes, checksum: a4f17991e9db7da2871b0c711b18f484 (MD5) Previous issue date: 2014-09-25 / CAPES - Coordenação de Aperfeiçoamento de Pessoal de Nível Superior / systems, mostly due to their welldefined structure and blockwise encoding. Among the main challenges encountered by mobile applications, there is an inherent interblock interference, due to superpositions of delayed signal copies, which is commonly eliminated with the addition of redundancy between adjacent data blocks. In addition to that, channel equalization is also usually employed, in order to further mitigate channel interferences. However, the amount of redundancy may be overestimated, which opens an opportunity for reduced-redundancy superfast transceivers, whose features include high spectral efficiency and low computational cost. Although the superfast approach aims at achieving low complexity, equalizer-coefficient updates are still very complex tasks due to channel variations, and most designs do not employ methodologies for computational-effort reduction. The present work addresses this problem and proposes a new design strategy for block-based transceivers, which provides semiblind equalization with data-selective update, besides the possibility of a generalized approach, based on the fast Fourier transform and diagonal matrices. Simulation results show that our approach updates less than 60% of the equalizer coefficients duringsupervised and blind period and maintain a competitive throughput for single-carrier and multicarrier transmissions. / Atualmente, os transceptores multicanais baseados em blocos são largamente utilizados em sistemas de comunicação sem fio, muito devido a sua estrutura bem definida e ao blockwise encoding. A respeito dos principais problemas encontrados em aplicações móveis, podemos destacar a interferência entre blocos, em decorrência da superposição de cópias atrasadas do sinal, a qual é usualmente eliminada com a adição de uma quantidade de redundância entre blocos de dados adjacentes. Adicionalmente, a equalização é comumente aplicada para mitigar o efeito do canal. Entretanto, a quantidade de redundância pode estar superestimada, abrindo oportunidade para a utilização de transceptores multicanais super-rápidos e com redundância reduzida, que possuem como característica uma alta eficiência espectral e baixa complexidade computacional. Entretanto, a abordagem super-rápida ainda possui uma alta complexidade para atualizar os coeficientes de equalização e a maioria das arquiteturas propostas não utilizam metodologias visando à redução do número de operações. O trabalho atual trata este problema e propõe uma nova arquitetura para tranceptores multicanais com transmissão em blocos, que se utiliza de uma equalização semi-cega com seleção de dados, além da abordagem generalizada, baseadas em transformadas rápidas de Fourier e matrizes diagonais. Os resultados das simulações demonstram que a abordagem permite atualizar menos de 60% dos coeficientes de equalização durante o período supervisionado e não supervisionado de equalização e manter a taxa de saída competitiva para sistemas monoportadora e multiportadora.

Submitted by Patrícia Cerveira (pcerveira1@gmail.com) on 2017-06-06T18:58:56Z No. of bitstreams: 1 Felipe Frasson- Dissertação.pdf: 984658 bytes, checksum: 05ae4f112679292aefe890dc2f563010 (MD5) / Rejected by Biblioteca da Escola de Engenharia (bee@ndc.uff.br), reason: Patrícia, o formulário de submissão apresenta vários erros, informações duplicadas e fora da formatação (orientador, coorientador, resumo, dentre outros). Atenciosamente, Catarina Ribeiro Bibliotecária BEE - Ramal 5992 on 2017-06-29T16:53:14Z (GMT) / Submitted by Patrícia Cerveira (pcerveira1@gmail.com) on 2017-06-29T19:32:38Z No. of bitstreams: 1 Felipe Frasson- Dissertação.pdf: 984658 bytes, checksum: 05ae4f112679292aefe890dc2f563010 (MD5) / Approved for entry into archive by Biblioteca da Escola de Engenharia (bee@ndc.uff.br) on 2017-07-03T13:00:12Z (GMT) No. of bitstreams: 1 Felipe Frasson- Dissertação.pdf: 984658 bytes, checksum: 05ae4f112679292aefe890dc2f563010 (MD5) / Made available in DSpace on 2017-07-03T13:00:12Z (GMT). No. of bitstreams: 1 Felipe Frasson- Dissertação.pdf: 984658 bytes, checksum: 05ae4f112679292aefe890dc2f563010 (MD5) / Em um sistema de comunica c~oes, os sinais s~ao transmitidos atrav es de canais de comunica c~ao que, idealmente, deveriam transportar os dados de maneira a n~ao causar distor c~ao alguma. Por em, em sistemas reais, existem limita c~oes que interferem neste processo causando degrada c~ao nas informa c~oes transmitidas, podendo comprometer sua recep c~ao. Tais limita c~oes ocorrem devido a presen ca de ru do aditivo, e principalmente por interfer^encia intersimb olica, esta caracterizada pela sobreposi c~ao de s mbolos gerados por uma mesma fonte transmissora. A equaliza c~ao de canal e uma das t ecnicas existentes que reduzem os efeitos da interfer^encia intersimb olica, dando maior con abilidade e robustez aos sistemas de comunica c~oes. Dentre as t ecnicas utilizadas para equaliza c~ao de canal, o uso de algoritmo adaptativos vem sendo amplamente utilizados devido as suas propriedades de se auto-ajustarem as varia c~oes que ocorrem ao longo do tempo. Este trabalho tem como objetivo veri car o comportamento de diferentes tipos de algoritmos adaptativos cegos ou semicegos, assim denominados por n~ao utilizarem sequ^encias de treinamento, aplicados a equaliza c~ao de canais esparsos. Canais esparsos s~ao encontrados em diversos sistemas de comunica c~oes como, por exemplo, na comunica c~ao sem o (telefonia m ovel, transmiss~ao de r adio e TV), ou, ainda, em canais subaqu aticos. Os algoritmos foram escolhidos com base em recentes estudos desta aplica c~ao, que operam em modo cego ou semicego e utilizam estat sticas de alta ordem, como os algoritmos Bussgang e Matching Pursuit. Os algoritmos foram implementados em ambiente de simula c~ao computacional no qual foram utilizados canais esparsos simples e de resposta ao impulso conhecida, permitindo comparar o comportamento dos diferentes algoritmos, em termos do sinal recuperado, e da inversa da resposta ao impulso do canal original. / In communications systems, information signals are transmitted through communications channels that, ideally, are delivered without distortions. However, on real communications channels there are limitations that interferes on the process, reducing the probability to recover the original signal at receiver. These distortions are basically thermal noise and Intersymbol Interference (ISI), caused by superposition on the received symbols received from the same source. Channel Equalization acts reducing these distortions, bringing more reliability to communications systems. The objective of this work is to verify di erent adaptive algorithms behavior, applied to sparse channel equalization problem. Many communications systems have sparse channels, like broadcast radio, television, mobile telephony and underwater communications. The selected algorithms used in this work includes high order statistics algorithms family, like Bussgang and Matching Pursuit. This kind of algorithms are widely used, with high relevance, for blind channel equalization. The selected algorithms were submitted to computer simulations using simple sparse channels and knowledge about their impulse response, in order to analyze their behavior in therms of bit error rate and the inverse impulse response of the channel.

Conselho Nacional de Desenvolvimento CientÃfico e TecnolÃgico / Este trabalho trata da proposiÃÃo de algoritmos para equalizaÃÃo cega de canais lineares e nÃao-lineares inspirados no Algoritmo do MÃdulo Constante (CMA). O CMA funciona de maneira bastante eficiente com constelaÃÃes nas quais todos os pontos possuem a mesma amplitude, como em modulaÃÃes do tipo Phase Shift Keying (PSK). Entretanto, quando os pontos da constelaÃÃo podem assumir diferentes valores de amplitudes, como em modulaÃÃes do tipo Quadrature Amplitude Modulation (QAM), o CMA e seus derivados muitas vezes nÃo funcionam de forma satisfatÃria. Desta forma, as tÃcnicas aqui propostas sÃo projetadas para melhorar a performance do CMA em termos de velocidade de convergÃncia e precisÃo, quando operando em sinais transmitidos com diversos mÃdulos, em particular para a modulaÃÃo QAM. Assim como o CMA, para possuir um bom apelo prÃtico, essas tÃcnicas devem apresentar bom compromisso entre complexidade, robustez e desempenho. Para tanto, as tÃcnicas propostas utilizam o Ãltimo sÃmbolo decidido para definir uma estimaÃÃo de raio de referÃncia para a saÃda do equalizador. De fato, esses algoritmos podem ser vistos como generalizaÃÃes do CMA e de alguns derivados do CMA para constelaÃÃes com mÃltiplos raios. A proposiÃÃo de algoritmos do tipo gradiente estocÃstico à concluÃda com o desenvolvimento de tÃcnicas originais, baseadas no CMA, para equalizaÃÃo de canais do tipo Wiener, que consiste em um filtro linear com memÃria, seguido por um filtro nÃo-linear sem memÃria. As expressÃes para a adaptaÃÃo do equalizador sÃo encontradas com o auxÃlio de uma notaÃÃo unificada para trÃs diferentes estruturas: i) um filtro de Hammerstein; ii) um filtro de Volterra diagonal; e iii) um filtro de Volterra completo. Um estudo teÃrico acerca do comportamento do principal algoritmo proposto, o Decision Directed Modulus Algorithm (DDMA) à realizado. SÃo analisadas a convergÃncia e a estabilidade do algoritmo atravÃs de uma anÃlise dos pontos de mÃnimo de sua funÃÃo custo. Outro objetivo à encontrar o valor teÃrico do Erro MÃdio QuadrÃtico MÃdio em Excesso - Excess Mean Square Error (EMSE) fornecido pelo DDMA considerando-se o caso sem ruÃdo. Ao final, à feito um estudo em que se constata que o algoritmo DDMA possui fortes ligaÃÃes com a soluÃÃo de Wiener e com o CMA. VersÃes normalizadas, bem como versÃes do tipo Recursive Least Squares (RLS), dos algoritmos do tipo gradiente estocÃstico estudados sÃo tambÃm desenvolvidas. Cada famÃlia de algoritmos estudada fie composta por quatro algoritmos com algumas propriedades interessantes e vantagens sobre as tÃcnicas clÃssicas, especialmente quando operando em sinais QAM de ordem elevada. TambÃm sÃo desenvolvidas versÃes normalizadas e do tipo RLS dos algoritmos do tipo CMA estudados para equalizaÃÃo de canais nÃo-lineares. O comportamento de todas as famÃlias de algoritmos desenvolvidos à testado atravÃs de simulaÃÃes computacionais, em que à verificado que as tÃcnicas propostas fornecem ganhos significativos em desempenho, em termos de velocidade de convergÃncia e erro residual, em relaÃÃo Ãs tÃcnicas clÃssicas. / This work studies and proposes algorithms to perform blind equalization of linear and nonlinear channels inspired on the Constant Modulus Algorithm (CMA). The CMA works very well for modulations in which all points of the signal constellation have the same radius, like in Phase Shift Keying (PSK) modulations. However, when the constellation points are characterized by multiple radii, like in Quadrature Amplitude Modulation (QAM) signals, the CMA does not work properly in many situations. Thus, the techniques proposed here are designed to improve the performance of the CMA, in terms of speed of convergence and residual error, when working with signals transmitted with multiple magnitude, in particular with QAM signals. As well as for the CMA, these techniques should have a good compromise among performance, complexity and robustness. To do so, the techniques use the last decided symbol to estimate reference radius to the output of the equalizer. In fact, they can be seen as modifications of the CMA and of some of its derivatives for constellations with multiple radii. The proposition of stochastic gradient algorithms is concluded with the development of new adaptive blind techniques to equalize channels with a Wiener structure. A Wiener filter consists of a linear block with memory followed by a memoryless nonlinearity, by using the CMA. We develop expressions for the adaptation of the equalizer using a unified notation for three different equalizer filter structures: i) a Hammerstein filter, ii) a diagonal Volterra filter and iii) a Volterra filter. A theoretical analysis of the main proposed technique, the Decision Directed Modulus Algorithm (DDMA), is also done. We study the convergence and the stability of the DDMA by means of an analysis of the minima of the DDM cost function. We also develop an analytic expression for the Excess Mean Square Error (EMSE) provided by the DDMA in the noiseless case. Then, we nd some interesting relationships among the DDM, the CM and the Wiener cost functions. We also develop a class of normalized algorithms and a class of Recursive Least Squares (RLS)-type algorithms for blind equalization inspired on the CMA-based techniques studied. Each family is composed of four algorithms with desirable properties and advantages over the original CM algorithms, specially when working with high-level QAM signals. Normalized and RLS techniques for equalization of Wiener channels are also developed. The behavior of the proposed classes of algorithms discussed is tested by computational simulations. We verify that the proposed techniques provide significative gains in performance, in terms of speed of convergence and residual error, when compared to the classical algorithms.