This thesis describes the application of adaptive equalization based on the constant modulus algorithm (CMA) and the decision-feedback minimum mean squared error (DF-MMSE) concept to the two compatible offset QPSK waveforms (FQPSK and SOQPSK-TG) that constitute the ARTM Tier-1 waveforms. An adaptive version of the DF-MMSE equalizer is developed and applied to this application. In the presence of frequency selective multipath interference typically encountered in aeronautical telemetry, both equalization techniques are shown to provide reliable performance for FQPSK and SOQPSK-TG. The performance of both waveforms with the DF-MMSE equalizer is slightly better than that using the CMA equalizer. Implementation trade-offs between the two types of equalizers are discussed.

Orientadora: Profa. Dra. Aline de Oliveira Neves Panazio / Dissertação (mestrado) - Universidade Federal do ABC, Programa de Pós-Graduação em Engenharia da Informação, 2014. / Redes de sensores sem o (WSN - Wireless Sensor Networks) têm sido usadas na observação de fenômenos, identicação de sistemas, equalização de canais, além de aplicações nas mais diversas áreas. Considerando o caso de redes homogêneas com protocolo ponto a ponto, nas quais os sensores são capazes de processar suas informações e se comunicar com sensores vizinhos, diversos algoritmos adaptativos vêm sendo aplicados no processamento dos dados medidos. Estes algoritmos podem ser supervisionados ou não supervisionados. Buscando estimar parâmetros comuns através de um processamento distribuído, a topologia da rede passa a ser uma característica importante e precisa ser levada em conta nos algoritmos utilizados. Tais algoritmos operam em modo de difusão, considerando a troca de informações entre sensores vizinhos na atualização dos coecientes dos ltros adaptativos de cada sensor. O mapeamento da topologia da rede é feito de forma matricial através das chamadas matrizes de combinação. Neste trabalho, estudamos o impacto da escolha da matriz de combinação no desempenho dos algoritmos supervisionados. No caso de algoritmos não supervisionados, como a única proposta encontrada na literatura considerava um caso bastante restrito em que o algoritmo só poderia ser aplicado a uma rede com topologia em anel e comunicação unidirecional entre os nós, propomos um novo algoritmo capaz de operar em modo de difusão em qualquer topologia, baseado no clássico critério do módulo constante. O algoritmo proposto é simulado em diversas situações, sempre apresentando vantagens em relação a uma rede sem cooperação entre os nós. / Wireless sensor networks (WSN) have been used in the observation of several phenomena, system identication, channel equalization, and others. Considering the case of homogeneous networks with point to point protocol, in which the sensors are able to process their information and communicate with neighbors, various adaptive algorithms have been applied in the processing of measured data. These algorithms can be supervised or unsupervised. Seeking to estimate common parameters across a distributed processing, network topology becomes an important feature and must be taken into account in the algorithms used. Such algorithms operate in diusion mode, that is, considering the exchange of information between sensors to update the coecients of the adaptive lters. Thenetwork topology is mapped through the use of a matrix, denoted combination matrix. In this work, we study the impact of the choice of the combination matrix on the performance of supervised algorithms. In the case of blind methods, the only technique found in the literature was applied to the specic case of a network with ring topology and unidirectional communication between nodes. Thus, we propose a new algorithm capable of operating in diusion mode on any topology, based on the classical constant modulus criterion. The proposed algorithm is simulated in several scenarios, always presenting advantages over a network without cooperation between nodes.

In this thesis, the problem of multiple access interference (MAI) suppression for the multi-carrier (MC) code division multiple access (CDMA) system, based on the multi-carrier modulation with modified Gaussian wavelet, associated with the combining process is investigated for Rayleigh fading channel. The main concern of this thesis is to derive a new scheme, based on the linearly constrained constant modulus (LCCM) criterion with the robust inverse QR decomposition (IQRD) recursive least squares (RLS) algorithm to improve the performance of the wavelet-based MC-CDMA system with combining process. To verify the merits of the new algorithm, the effect due to imperfect channel parameters estimation and near-far effect are investigated. We show that the proposed robust LCCM IQRD-RLS algorithm outperforms the conventional LCCM-gradient algorithm, in terms of output SINR, for MAI suppression under channel mismatch environment. Also, the performance of the modified Gaussian wavelet-based MC-CDMA system is superior to the one with wavelet-based MC-CDMA system. It is more robust to the channel mismatch and near-far effect. Moreover, the modified Gaussian wavelet-based MC-CDMA system with robust LCCM IQRD-RLS algorithm does have better performance over other conventional approaches, such as the LCCM-gradient algorithm, maximum ratio combining (MRC), and blind adaptation algorithm, in terms of the capability of MAI suppression and bit error rate (BER).

abstract: Distributed inference has applications in a wide range of fields such as source localization, target detection, environment monitoring, and healthcare. In this dissertation, distributed inference schemes which use bounded transmit power are considered. The performance of the proposed schemes are studied for a variety of inference problems. In the first part of the dissertation, a distributed detection scheme where the sensors transmit with constant modulus signals over a Gaussian multiple access channel is considered. The deflection coefficient of the proposed scheme is shown to depend on the characteristic function of the sensing noise, and the error exponent for the system is derived using large deviation theory. Optimization of the deflection coefficient and error exponent are considered with respect to a transmission phase parameter for a variety of sensing noise distributions including impulsive ones. The proposed scheme is also favorably compared with existing amplify-and-forward (AF) and detect-and-forward (DF) schemes. The effect of fading is shown to be detrimental to the detection performance and simulations are provided to corroborate the analytical results. The second part of the dissertation studies a distributed inference scheme which uses bounded transmission functions over a Gaussian multiple access channel. The conditions on the transmission functions under which consistent estimation and reliable detection are possible is characterized. For the distributed estimation problem, an estimation scheme that uses bounded transmission functions is proved to be strongly consistent provided that the variance of the noise samples are bounded and that the transmission function is one-to-one. The proposed estimation scheme is compared with the amplify and forward technique and its robustness to impulsive sensing noise distributions is highlighted. It is also shown that bounded transmissions suffer from inconsistent estimates if the sensing noise variance goes to infinity. For the distributed detection problem, similar results are obtained by studying the deflection coefficient. Simulations corroborate our analytical results. In the third part of this dissertation, the problem of estimating the average of samples distributed at the nodes of a sensor network is considered. A distributed average consensus algorithm in which every sensor transmits with bounded peak power is proposed. In the presence of communication noise, it is shown that the nodes reach consensus asymptotically to a finite random variable whose expectation is the desired sample average of the initial observations with a variance that depends on the step size of the algorithm and the variance of the communication noise. The asymptotic performance is characterized by deriving the asymptotic covariance matrix using results from stochastic approximation theory. It is shown that using bounded transmissions results in slower convergence compared to the linear consensus algorithm based on the Laplacian heuristic. Simulations corroborate our analytical findings. Finally, a robust distributed average consensus algorithm in which every sensor performs a nonlinear processing at the receiver is proposed. It is shown that non-linearity at the receiver nodes makes the algorithm robust to a wide range of channel noise distributions including the impulsive ones. It is shown that the nodes reach consensus asymptotically and similar results are obtained as in the case of transmit non-linearity. Simulations corroborate our analytical findings and highlight the robustness of the proposed algorithm. / Dissertation/Thesis / Ph.D. Electrical Engineering 2013

O objetivo da desconvolução cega de imagens é restaurar uma imagem degradada sem usar informação da imagem real ou da função de degradação. O mapeamento dos níveis de cinza de uma imagem em um sinal de comunicação possibilita o uso de técnicas de equalização cega de canais para a restauração de imagens. Neste trabalho, propõe-se o uso de um esquema para desconvolução cega de imagens baseado na combinação convexa de um equalizador cego com um equalizador no modo de decisão direta. A combinação também é adaptada de forma cega, o que possibilita o chaveamento automático entre os filtros componentes. Dessa forma, o esquema proposto é capaz de atingir o desempenho de um algoritmo de filtragem adaptativa supervisionada sem o conhecimento prévio da imagem original. O desempenho da combinação é ilustrado por meio de simulações, que comprovam a eficiência desse esquema quando comparado a outras soluções da literatura. / The goal of blind image deconvolution is to restore a degraded image without using information from the actual image or from the point spread function. The mapping of the gray levels of an image into a communication signal enables the use of blind equalization techniques for image restoration. In this work, we use a blind image deconvolution scheme based on the convex combination of a blind equalizer with an equalizer in the decision-directed mode. The combination is also blindly adapted, which enables automatic switching between the component filters. Thus, the proposed scheme is able to achieve the performance of a supervised adaptive filtering algorithm without prior knowledge of the original image. The performance of the combination is illustrated by simulations, which show the efficiency of this scheme when compared to other solutions in the literature.

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.

Neste trabalho, são propostos e analisados algoritmos autodidatas eficientes para a equalização de canais de comunicação, considerando a transmissão de sinais QAM (quadrature amplitude modulation). Suas funções de erro são construídas de forma a fazer com que o erro de estimação seja igual a zero nas coordenadas dos símbolos da constelação. Essa característica os possibilita ter um desempenho similar ao de um algoritmo de equalização supervisionada como o NLMS (normalized least mean-square), independentemente da ordem da constelação QAM. Verifica-se analiticamente que, sob certas condições favoráveis para a equalização, os vetores de coeficientes dos algoritmos propostos e a correspondente solução de Wiener são colineares. Além disso, usando a informação da estimativa do símbolo transmitido e de seus símbolos vizinhos, esquemas de baixo custo computacional são propostos para aumentar a velocidade de convergência dos algoritmos. No caso do algoritmo baseado no critério do módulo constante, evita-se sua divergência através de um mecanismo que descarta estimativas inconsistentes dos símbolos transmitidos. Adicionalmente, apresenta-se uma análise de rastreio (tracking), que permite obter expressões analíticas para o erro quadrático médio em excesso dos algoritmos propostos em ambientes estacionários e não-estacionários. Através dessas expressões, verifica-se que com sobreamostragem, ausência de ruído e ambiente estacionário, os algoritmos propostos podem alcançar a equalização perfeita, independentemente da ordem da constelação QAM. Os algoritmos são estendidos para a adaptação conjunta dos filtros direto e de realimentação do equalizador de decisão realimentada, levando-se em conta um mecanismo que evita soluções degeneradas. Resultados de simulação sugerem que a utilização dos esquemas aqui propostos pode ser vantajosa na recuperação de sinais QAM, fazendo com que seja desnecessário o chaveamento para o algoritmo de decisão direta. / In this work, we propose efficient blind algorithms for equalization of communication channels, considering the transmission of QAM (quadrature amplitude modulation) signals. Their error functions are constructed in order to make the estimation error equal to zero at the coordinates of the constellation symbols. This characteristic enables the proposed algorithms to have a similar performance to that of a supervised equalization algorithm as the NLMS (normalized least mean-square), independently of the QAM order. Under some favorable conditions, we verify analytically that the coefficient vector of the proposed algorithms are collinear with the Wiener solution. Furthermore, using the information of the symbol estimate in conjunction with its neighborhood, we propose schemes of low computational cost in order to improve their convergence rate. The divergence of the constant-modulus based algorithm is avoided by using a mechanism, which disregards nonconsistent estimates of the transmitted symbols. Additionally, we present a tracking analysis in which we obtain analytical expressions for the excess mean-square error in stationary and nonstationary environments. From these expressions, we verify that using a fractionally-spaced equalizer in a noiseless stationary environment, the proposed algorithms can achieve perfect equalization, independently of the QAM order. The algorithms are extended to jointly adapt the feedforward and feedback filters of the decision feedback equalizer, taking into account a mechanism to avoid degenerative solutions. Simulation results suggest that the proposed schemes may be advantageously used to recover QAM signals and make the switching to the decision direct mode unnecessary.

O objetivo da desconvolução cega de imagens é restaurar uma imagem degradada sem usar informação da imagem real ou da função de degradação. O mapeamento dos níveis de cinza de uma imagem em um sinal de comunicação possibilita o uso de técnicas de equalização cega de canais para a restauração de imagens. Neste trabalho, propõe-se o uso de um esquema para desconvolução cega de imagens baseado na combinação convexa de um equalizador cego com um equalizador no modo de decisão direta. A combinação também é adaptada de forma cega, o que possibilita o chaveamento automático entre os filtros componentes. Dessa forma, o esquema proposto é capaz de atingir o desempenho de um algoritmo de filtragem adaptativa supervisionada sem o conhecimento prévio da imagem original. O desempenho da combinação é ilustrado por meio de simulações, que comprovam a eficiência desse esquema quando comparado a outras soluções da literatura. / The goal of blind image deconvolution is to restore a degraded image without using information from the actual image or from the point spread function. The mapping of the gray levels of an image into a communication signal enables the use of blind equalization techniques for image restoration. In this work, we use a blind image deconvolution scheme based on the convex combination of a blind equalizer with an equalizer in the decision-directed mode. The combination is also blindly adapted, which enables automatic switching between the component filters. Thus, the proposed scheme is able to achieve the performance of a supervised adaptive filtering algorithm without prior knowledge of the original image. The performance of the combination is illustrated by simulations, which show the efficiency of this scheme when compared to other solutions in the literature.

Neste trabalho, são propostos e analisados algoritmos autodidatas eficientes para a equalização de canais de comunicação, considerando a transmissão de sinais QAM (quadrature amplitude modulation). Suas funções de erro são construídas de forma a fazer com que o erro de estimação seja igual a zero nas coordenadas dos símbolos da constelação. Essa característica os possibilita ter um desempenho similar ao de um algoritmo de equalização supervisionada como o NLMS (normalized least mean-square), independentemente da ordem da constelação QAM. Verifica-se analiticamente que, sob certas condições favoráveis para a equalização, os vetores de coeficientes dos algoritmos propostos e a correspondente solução de Wiener são colineares. Além disso, usando a informação da estimativa do símbolo transmitido e de seus símbolos vizinhos, esquemas de baixo custo computacional são propostos para aumentar a velocidade de convergência dos algoritmos. No caso do algoritmo baseado no critério do módulo constante, evita-se sua divergência através de um mecanismo que descarta estimativas inconsistentes dos símbolos transmitidos. Adicionalmente, apresenta-se uma análise de rastreio (tracking), que permite obter expressões analíticas para o erro quadrático médio em excesso dos algoritmos propostos em ambientes estacionários e não-estacionários. Através dessas expressões, verifica-se que com sobreamostragem, ausência de ruído e ambiente estacionário, os algoritmos propostos podem alcançar a equalização perfeita, independentemente da ordem da constelação QAM. Os algoritmos são estendidos para a adaptação conjunta dos filtros direto e de realimentação do equalizador de decisão realimentada, levando-se em conta um mecanismo que evita soluções degeneradas. Resultados de simulação sugerem que a utilização dos esquemas aqui propostos pode ser vantajosa na recuperação de sinais QAM, fazendo com que seja desnecessário o chaveamento para o algoritmo de decisão direta. / In this work, we propose efficient blind algorithms for equalization of communication channels, considering the transmission of QAM (quadrature amplitude modulation) signals. Their error functions are constructed in order to make the estimation error equal to zero at the coordinates of the constellation symbols. This characteristic enables the proposed algorithms to have a similar performance to that of a supervised equalization algorithm as the NLMS (normalized least mean-square), independently of the QAM order. Under some favorable conditions, we verify analytically that the coefficient vector of the proposed algorithms are collinear with the Wiener solution. Furthermore, using the information of the symbol estimate in conjunction with its neighborhood, we propose schemes of low computational cost in order to improve their convergence rate. The divergence of the constant-modulus based algorithm is avoided by using a mechanism, which disregards nonconsistent estimates of the transmitted symbols. Additionally, we present a tracking analysis in which we obtain analytical expressions for the excess mean-square error in stationary and nonstationary environments. From these expressions, we verify that using a fractionally-spaced equalizer in a noiseless stationary environment, the proposed algorithms can achieve perfect equalization, independently of the QAM order. The algorithms are extended to jointly adapt the feedforward and feedback filters of the decision feedback equalizer, taking into account a mechanism to avoid degenerative solutions. Simulation results suggest that the proposed schemes may be advantageously used to recover QAM signals and make the switching to the decision direct mode unnecessary.

This thesis attempts to address the problem of how best to remedy different types of channel distortions on speech when that speech is to be used in automatic speaker recognition and verification systems. Automatic speaker recognition is when a person's voice is analysed by a machine and the person's identity is worked out by the comparison of speech features to a known set of speech features. Automatic speaker verification is when a person claims an identity and the machine determines if that claimed identity is correct or whether that person is an impostor. Channel distortion occurs whenever information is sent electronically through any type of channel whether that channel is a basic wired telephone channel or a wireless channel. The types of distortion that can corrupt the information include time-variant or time-invariant filtering of the information or the addition of 'thermal noise' to the information, both of these types of distortion can cause varying degrees of error in information being received and analysed. The experiments presented in this thesis investigate the effects of channel distortion on the average speaker recognition rates and testing the effectiveness of various channel compensation algorithms designed to mitigate the effects of channel distortion. The speaker recognition system was represented by a basic recognition algorithm consisting of: speech analysis, extraction of feature vectors in the form of the Mel-Cepstral Coefficients, and a classification part based on the minimum distance rule. Two types of channel distortion were investigated: • Convolutional (or lowpass filtering) effects • Addition of white Gaussian noise Three different methods of channel compensation were tested: • Cepstral Mean Subtraction (CMS) • RelAtive SpecTrAl (RASTA) Processing • Constant Modulus Algorithm (CMA) The results from the experiments showed that for both CMS and RASTA processing that filtering at low cutoff frequencies, (3 or 4 kHz), produced improvements in the average speaker recognition rates compared to speech with no compensation. The levels of improvement due to RASTA processing were higher than the levels achieved due to the CMS method. Neither the CMS or RASTA methods were able to improve accuracy of the speaker recognition system for cutoff frequencies of 5 kHz, 6 kHz or 7 kHz. In the case of noisy speech all methods analysed were able to compensate for high SNR of 40 dB and 30 dB and only RASTA processing was able to compensate and improve the average recognition rate for speech corrupted with a high level of noise (SNR of 20 dB and 10 dB).