Cognitive Radio (CR) is an emerging wireless communications technology that addresses the inefficiency of current radio spectrum usage. CR also supports the evolution of existing wireless applications and the development of new civilian and military applications. In military and public safety applications, there is no information available about the signal present in a frequency band and hence there is a need for a CR receiver to identify the modulation format employed in the signal. The automatic modulation classifier (AMC) is an important signal processing component that helps the CR in identifying the modulation format employed in the detected signal. AMC algorithms developed so far can classify only signals from a single user present in a frequency band. In a typical CR scenario, there is a possibility that more than one user is present in a frequency band and hence it is necessary to develop an AMC that can classify signals from multiple users simultaneously. One of the main objectives of this dissertation is to develop robust multiuser AMC's for CR. It will be shown later that multiple antennas are required at the receiver for classifying multiple signals. The use of multiple antennas at the transmitter and receiver is known as a Multi Input Multi Output (MIMO) communication system. By using multiple antennas at the receiver, apart from classifying signals from multiple users, the CR can harness the advantages offered by classical MIMO communication techniques like higher data rate, reliability, and an extended coverage area. While MIMO CR will provide numerous benefits, there are some significant challenges in applying conventional MIMO theory to CR. In this dissertation, open problems in applying classical MIMO techniques to a CR scenario are addressed. A blind equalizer is another important signal processing component that a CR must possess since there are no training or pilot signals available in many applications. In a typical wireless communication environment the transmitted signals are subjected to noise and multipath fading. Multipath fading not only affects the performance of symbol detection by causing inter symbol interference (ISI) but also affects the performance of the AMC. The equalizer is a signal processing component that removes ISI from the received signal, thus improving the symbol detection performance. In a conventional wireless communication system, training or pilot sequences are usually available for designing the equalizer. When a training sequence is available, equalizer parameters are adapted by minimizing the well known cost function called mean square error (MSE). When a training sequence is not available, blind equalization algorithms adapt the parameters of the blind equalizer by minimizing cost functions that exploit the higher order statistics of the received signal. These cost functions are non convex and hence the blind equalizer has the potential to converge to a local minimum. Convergence to a local minimum not only affects symbol detection performance but also affects the performance of the AMC. Robust blind equalizers can be designed if the performance of the AMC is also considered while adapting equalizer parameters. In this dissertation we also develop Single Input Single Output (SISO) and MIMO blind equalizers where the performance of the AMC is also considered while adapting the equalizer parameters. / Ph. D.

The central theme is the study of error propagation effects in decision feedback equalizers (DFEs). The thesis contains: a stochastic analysis of error propagation in a tuned DFE; an analysis of the effects of error propagation in a blindly adapted DFE; a deterministic analysis of error propagation through input-output stability ideas; and testing procedures for establishing correct tap convergence in blind adaptation. To a lesser extent, the decision directed equalizer (DDE) is also treated.¶ Characterizing error propagation using finite state Markov process (FSMP) techniques is first considered. We classify how the channel and DFE parameters affect the FSMP model and establish tight bounds on the error probability and mean error recovery time of a tuned DFE. These bounds are shown to be too conservative for practical use and highlight the need for imposing stronger hypotheses on the class of channels for which a DFE may be effectively used.¶ In blind DFE adaptation we show the effect of decision errors is to distort the adaptation relative to the use of a training sequence. The mean square error surface in a LMS type setting is shown to be a concatenation of quadratic functions exposing the possibility of false tap convergence to undesirable DFE parameter settings. Averaging analysis and simulation are used to verify this behaviour on some examples.¶ Error propagation in a tuned DFE is also examined in a deterministic setting. A finite error recovery time problem is set up as an input-output stability problem. Passivity theory is invoked to prove that a DFE can be effectively used on a channel satisfying a simple frequency domain condition. These results give performance bounds which relate well with practice.¶ Testing for false tap convergence in blind adaptation concludes our study. Simple statistic output tests are shown to be capable of discerning correct operation of a DDE. Similar tests are conjectured for the DFE, supported by proofs for the low dimensional cases.

International Telemetering Conference Proceedings / October 25-28, 1999 / Riviera Hotel and Convention Center, Las Vegas, Nevada / Blind adaptive equalization with application for Non-Linearly Amplified (NLA) quadrature amplitude modulation (QAM) systems in multipath selective fading channels is presented. With an offset sampling strategy in the receiver, the proposed blind equalization using Constant Modulus Algorithm (CMA) exhibits a fast convergent speed for a family of quadrature modulated systems in NLA and multipath fading channels. Feher’s patented Quadrature Phase Shift Keying (FQPSK) and Feher’s Quadrature Amplitude Modulation (FQAM) which correspond respectively to 4-state and 16-state QAM are used due to their higher Radio Frequency (RF) power and spectral efficiency in NLA channel. It has been shown that blind adaptive equalization can significantly open the eye signals in multipath frequency selective fading channels.

At a communications receiver the observed signal is a corrupted version of the transmitted signal. This distortion in the received signal is due to the physical characteristics of the channel, including multipath propagation, the non-idealities of copper wires and impulse noise. Equalization is a process to combat these distortions in order to recover the original transmitted signal. Roughly stated, the equalizer tries to implement the inverse transfer function of the channel while taking into account the channel noise. The equalizer parameters can be tuned to this inverse transfer function using an adaptive algorithm. In many cases, the algorithm uses a training sequence to drive the equalizer parameters to the optimum solution. But, for time-varying channels or multiuser channels the use of a training sequence is inefficient in terms of bandwidth, as bandwidth is wasted due to the periodic re-transmission of the training sequence. A blind equalization algorithm is a practical method to eliminate this training sequence. An equalizer adapted using a blind algorithm is a key component of a bandwidth efficient receiver for broadcast and point-to-multipoint communications. The initial convergence performance of a blind adaptive equalizer depends on the higher-order statistics of the transmitted signal. In modern digital systems, Tomlinson-Harashima precoding (THP) is often used for signal shaping and to mitigate the error propagation problem of a decision feedback equalizer (DFE). The concept of THP comes from pre-equalization. In fact, it is a nonlinear form of pre-equalization, which bounds the higher-order statistics of the transmitted signal. But, THP and blind equalization are often viewed as incompatible equalization techniques. In this research, we give multiple scenarios where blind equalization of a THP-encoded signal might arise. With this motivation we set out to answer the question, can a blind equalizer successfully acquire a THP-encoded signal? We investigate the combination of a Tomlinson-Harashima precoder on the transmitter side and a blind equalizer on the receiver side. By bounding the kurtosis of the THP-encoded signal, we show that THP actually aids the initial convergence of blind equalization. We find that, as the symbol constellation size increases, the THP-encoded signal kurtosis approaches that of a uniform distribution, not a Gaussian. We investigate the compatibility of blind equalization with THP-encoded signals for both SISO and MIMO systems. In a SISO system, conventional blind algorithms can be used to counter the distortions introduced in the received signal. However, in a MIMO system with multiple users, the other users act as interferers on the desired user's signal. Hence, modified blind algorithms need to be applied to mitigate these interferers. For both SISO and MIMO systems, we show that the THP encoder ensures that the signal distribution approaches a non-Gaussian distribution. Using Monte Carlo simulations, we study the effects of Tomlinson-Harashima precoding on the performance of Bussgang-type blind algorithms and verify our theoretical analysis. The major contributions of this thesis are: • A demonstration that a blind equalizer can successfully acquire a THP-encoded signal for both SISO and MIMO systems. We show that THP actually aids blind equalization, as it ensures that the transmitted signal is non-Gaussian. • An analytical quantification of the effects of THP on the transmitted signal statistics. We derive a novel bound on the kurtosis of the THP-encoded signal. • An extension of the results from a single-user SISO scenario to multiple users and a MIMO scenario. We demonstrate that our bound and simulated results hold for these more general cases. Through our work, we have opened the way for a novel application of training sequence-less equalization: to acquire and equalize THP-encoded signals. Using our proposed system, periodic training sequences for a broadcast or point-to-multipoint system can be avoided, improving the bandwidth efficiency of the transceiver. Future modem designs with THP encoding can make use of our advances for bandwidth efficient communication systems.

Blind equalization schemes are used to cancel the effects of a channel on the received signal when the transmission of a training sequence in a predefined time slot is not possible. In the absence of a training sequence, blind equalization schemes can also increase the throughput of the overall system. A general problem with blind adaptation techniques is that they have poor convergence properties compared to the traditional techniques using training sequences. Having a multi-modal cost surface, blind adaptation techniques may force the equalizer to converge to a false minimum, depending on the initialization. The most commonly used blind adaptation algorithm is the Constant Modulus Algorithm (CMA). It is shown by simulation that a logarithmic error equation can make CMA converge to a global minimum, if a differential encoding scheme is used. The performance of CMA with different error equations is also investigated for different channel conditions. For a time varying channel, the performance of an equalizer not only depends on the convergence behavior but also on the tracking property, which indicates the ability of an equalizer to track changes in the channel. The tracking property of a blind equalizer with CMA has been investigated under different channel conditions. It is also shown that the tracking property of a blind equalizer can be improved by using a recursive linear predictor at the output of the equalizer to predict the amplitude of the equalizer output. The predicted value of the amplitude is then used to adjust the instantaneous gain of the overall system. A recursive linear predictor is designed to predict a colored signal without having a priori knowledge about the correlation function of the input sequence. The performance of the designed predictor is also investigated by predicting the envelope of a flat fading channel under constant mobile velocity and constant acceleration conditions. / Master of Science

Nesta tese analisamos e propomos métodos para a equalização não-treinada (cega) de canais de comunicação lineares FIR baseados em filtros de partículas, que são técnicas recursivas para a solução Bayesiana de problemas de filtragem estocástica. Iniciamos propondo novos métodos para equalização sob ruído gaussiano que prescindem do uso de codificação diferencial, ao contrário dos métodos existentes. Empregando técnicas de evolução artificial de parâmetros, estendemos estes resultados para o caso de ruído aditivo com distribuição não-gaussiana. Em seguida, desenvolvemos novos métodos baseados nos mesmos princípios para equalizar e decodificar conjuntamente sistemas de comunicação que empregam códigos convolucionais ou em bloco. Através de simulações numéricas, observamos que os algoritmos propostos apresentam desempenhos, medidos em termos de taxa média de erro de bit e velocidade de convergência, marcadamente superiores aos de métodos tradicionais, freqüentemente aproximando o desempenho dos algoritmos ótimos (MAP) treinados. Além disso, observamos que os métodos baseados em filtros de partículas determinísticos exibem desempenhos consistentemente superiores aos dos demais métodos, sendo portanto a melhor escolha caso o modelo de sinal empregado permita a marginalização analítica dos parâmetros desconhecidos do canal. / In this thesis, we propose and analyze blind equalization methods suitable for linear FIR communications channels, focusing on the development of algorithms based on particle filters - recursive methods for approximating Bayesian solutions to stochastic filtering problems. Initially, we propose new equalization methods for signal models with gaussian additive noise that dispense with the need for differentially encoding the transmitted signals, as opposed to the previously existing methods. Next, we extend these algorithms to deal with non-gaussian additive noise by deploying artificial parameter evolution techniques. We next develop new joint blind equalization and decoding algorithms, suitable for convolutionally or block-coded communications systems. Via numerical simulations we show that the proposed algorithms outperform traditional approaches both in terms of mean bit error rate and convergence speed, and closely approach the performance of the optimal (MAP) trained equalizer. Furthermore, we observed that the methods based on deterministic particle filters consistently outperform those based on stochastic approaches, making them preferable when the adopted signal model allows for the analytic marginalization of the unknown channel parameters.

Neste trabalho, investigam-se técnicas autodidatas baseadas em estatísticas de ordem superior, aplicadas à equalização de canais de comunicação. Inicialmente, obtém-se um intervalo do passo de adaptação que assegura a convergência do algoritmo do Módulo Constante com o gradiente exato. Algoritmos como o CMA (Constant Modulus Algorithm) e o SWA (Shalvi-Weinstein Algorithm) são revisitados e suas capacidades de tracking analisadas, utilizando-se uma relação de conservação de energia. Além disso, é proposto um algoritmo autodidata denominado AC-CMA (Accelerated Constant Modulus Algorithm) que utiliza a segunda derivada (“aceleração") da estimativa dos coeficientes. Esse algoritmo pode apresentar um compromisso mais favorável entre complexidade computacional e velocidade de convergência que o CMA e o SWA. Esses resultados são estendidos para o caso multiusuário. Através de simulações, os algoritmos são comparados e as análises de convergência e tracking validadas. Considerando o DFE (Decision Feedback Equalizer) no caso monousuário com o critério do módulo constante, é proposto um algoritmo concorrente que evita soluções degeneradas e apresenta um desempenho melhor do que os existentes na literatura. Com o intuito de evitar propagação de erros, é proposta uma estrutura híbrida que utiliza uma rede neural recorrente na malha de realimentação. Resultados de simulações indicam que seu uso pode ser vantajoso para canais lineares e não-lineares. / The equalization of communication channels is addressed by using blind techniques based on higher order statistics. A step-size interval is obtained to ensure the convergence of Steepest-Descent Constant Modulus Algorithm. The Shalvi-Weinstein Algorithm (SWA) and Constant Modulus Algorithm (CMA) are revisited and their tracking capabilities are analyzed by using an energy conservation relation. Moreover, a novel blind algorithm named Accelerated Constant Modulus Algorithm (AC-CMA) is proposed. It adjusts the second derivative (“acceleration") of the coefficient estimates and presents a more favorable compromise between computational complexity and convergence rate than CMA or SWA. These results are extended to the MIMO (Multiple-Input Multiple-Output) case. By means of simulations, the algorithms are compared and the convergence and tracking analysis are validated. The Decision Feedback Equalizer (DFE) is considered in the SISO (Single-Input Single-Output) case with the Constant Modulus criterion and a concurrent algorithm is proposed. It avoids degenerated solutions and shows better behavior than the others presented in the literature. In order to avoid error propagation, a hybrid DFE is also proposed. It includes a recurrent neural network in the feedback filter and may be advantageously used to equalize linear and nonlinear channels.

Neste trabalho, investigam-se técnicas autodidatas baseadas em estatísticas de ordem superior, aplicadas à equalização de canais de comunicação. Inicialmente, obtém-se um intervalo do passo de adaptação que assegura a convergência do algoritmo do Módulo Constante com o gradiente exato. Algoritmos como o CMA (Constant Modulus Algorithm) e o SWA (Shalvi-Weinstein Algorithm) são revisitados e suas capacidades de tracking analisadas, utilizando-se uma relação de conservação de energia. Além disso, é proposto um algoritmo autodidata denominado AC-CMA (Accelerated Constant Modulus Algorithm) que utiliza a segunda derivada (“aceleração”) da estimativa dos coeficientes. Esse algoritmo pode apresentar um compromisso mais favorável entre complexidade computacional e velocidade de convergência que o CMA e o SWA. Esses resultados são estendidos para o caso multiusuário. Através de simulações, os algoritmos são comparados e as análises de convergência e tracking validadas. Considerando o DFE (Decision Feedback Equalizer) no caso monousuário com o critério do módulo constante, é proposto um algoritmo concorrente que evita soluções degeneradas e apresenta um desempenho melhor do que os existentes na literatura. Com o intuito de evitar propagação de erros, é proposta uma estrutura híbrida que utiliza uma rede neural recorrente na malha de realimentação. Resultados de simulações indicam que seu uso pode ser vantajoso para canais lineares e não-lineares. / The equalization of communication channels is addressed by using blind techniques based on higher order statistics. A step-size interval is obtained to ensure the convergence of Steepest-Descent Constant Modulus Algorithm. The Shalvi-Weinstein Algorithm (SWA) and Constant Modulus Algorithm (CMA) are revisited and their tracking capabilities are analyzed by using an energy conservation relation. Moreover, a novel blind algorithm named Accelerated Constant Modulus Algorithm (AC-CMA) is proposed. It adjusts the second derivative (“acceleration”) of the coefficient estimates and presents a more favorable compromise between computational complexity and convergence rate than CMA or SWA. These results are extended to the MIMO (Multiple-Input Multiple-Output) case. By means of simulations, the algorithms are compared and the convergence and tracking analysis are validated. The Decision Feedback Equalizer (DFE) is considered in the SISO (Single-Input Single-Output) case with the Constant Modulus criterion and a concurrent algorithm is proposed. It avoids degenerated solutions and shows better behavior than the others presented in the literature. In order to avoid error propagation, a hybrid DFE is also proposed. It includes a recurrent neural network in the feedback filter and may be advantageously used to equalize linear and nonlinear channels.

Nesta tese analisamos e propomos métodos para a equalização não-treinada (cega) de canais de comunicação lineares FIR baseados em filtros de partículas, que são técnicas recursivas para a solução Bayesiana de problemas de filtragem estocástica. Iniciamos propondo novos métodos para equalização sob ruído gaussiano que prescindem do uso de codificação diferencial, ao contrário dos métodos existentes. Empregando técnicas de evolução artificial de parâmetros, estendemos estes resultados para o caso de ruído aditivo com distribuição não-gaussiana. Em seguida, desenvolvemos novos métodos baseados nos mesmos princípios para equalizar e decodificar conjuntamente sistemas de comunicação que empregam códigos convolucionais ou em bloco. Através de simulações numéricas, observamos que os algoritmos propostos apresentam desempenhos, medidos em termos de taxa média de erro de bit e velocidade de convergência, marcadamente superiores aos de métodos tradicionais, freqüentemente aproximando o desempenho dos algoritmos ótimos (MAP) treinados. Além disso, observamos que os métodos baseados em filtros de partículas determinísticos exibem desempenhos consistentemente superiores aos dos demais métodos, sendo portanto a melhor escolha caso o modelo de sinal empregado permita a marginalização analítica dos parâmetros desconhecidos do canal. / In this thesis, we propose and analyze blind equalization methods suitable for linear FIR communications channels, focusing on the development of algorithms based on particle filters - recursive methods for approximating Bayesian solutions to stochastic filtering problems. Initially, we propose new equalization methods for signal models with gaussian additive noise that dispense with the need for differentially encoding the transmitted signals, as opposed to the previously existing methods. Next, we extend these algorithms to deal with non-gaussian additive noise by deploying artificial parameter evolution techniques. We next develop new joint blind equalization and decoding algorithms, suitable for convolutionally or block-coded communications systems. Via numerical simulations we show that the proposed algorithms outperform traditional approaches both in terms of mean bit error rate and convergence speed, and closely approach the performance of the optimal (MAP) trained equalizer. Furthermore, we observed that the methods based on deterministic particle filters consistently outperform those based on stochastic approaches, making them preferable when the adopted signal model allows for the analytic marginalization of the unknown channel parameters.

A Bussgang based blind deconvolution algorithm called self-correcting multi-channel Bussgang (SCMB) blind deconvolution algorithm was proposed. Unlike the original Bussgang blind deconvolution algorithm where the probability density function (pdf) of the signal being recovered is assumed to be completely known, the proposed SCMB blind deconvolution algorithm relaxes this restriction by parameterized the pdf with a Gaussian mixture model and expectation maximization (EM) algorithm, an iterative maximum likelihood approach, is employed to estimate the parameter side by side with the estimation of the equalization filters of the original Bussgang blind deconvolution algorithm. A feedback loop is also designed to compensate the effect of the parameter estimation error on the estimation of the equalization filters. Application of the SCMB blind deconvolution framework for binary image restoration, multi-pass synthetic aperture radar (SAR) autofocus and inverse synthetic aperture radar (ISAR) autofocus are exploited with great results.