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).

Orientadora: Profa. Dra. Aline de Oliveira Neves Panazio / Dissertação (mestrado) - Universidade Federal do ABC, Programa de Pós-Graduação em Engenharia da Informação, Santo André, 2018. / Este trabalho dedica-se ao estudo de algoritmos de filtragem adaptativa autodidata no modo difusão, com aplicações em redes de sensores sem fio (RSSF). No modo difusão, os nós sensores da rede possuem poder de processamento local e trocam informações com seus vizinhos. Neste trabalho, propomos dois algoritmos utilizando como base o algoritmo CMA no modo Difusão (CMAD), com duas abordagens distintas da técnica Set-Membership. O primeiro baseia-se no algoritmo Set-Membership Least Mean Squares (SM-LMS), desenvolvido também no modo difusão. Estendemos o algoritmo para o contexto não supervisionado, denotando por Algoritmo Set-Membership CMA no modo Difusão (SM-CMAD). Mostramos que este algoritmo apresenta desempenho melhor ou similar ao CMAD, em termos de velocidade de convergência, patamar de interferência intersimbólica (IIS) e possuindo a importante vantagem de reduzir as trocas de informações entre os nós, economizando energia e recursos da rede. O segundo algoritmo proposto se baseia no Set-Membership do Módulo Constante (SM-CM), o qual estendemos para o contexto de redes de sensores sem fio no modo difusão. Tal algoritmo é denotado por Algoritmo Set-membership CMA no modo Difusão Square-root Gamma (SM-CMAD-SG). Novamente o algoritmo apresenta um bom desempenho quando comparado com o CMAD e, quando comparado ao SM-CMAD, vemos que sua principal vantagem está na economia em termos de atualizações dos coeficientes do filtro, que chega a valores acima de 70% em diversos cenários de simulação, sem grandes perdas de desempenho economizando energia. / This work is devoted to the study of unsupervised adaptive filtering algorithms in diffusion mode, with applications in wireless sensor networks (WSNs). In diffusion mode, network sensing nodes have local processing power and exchange information with their neighbors. In this work, we propose two algorithms based on the CMA algorithm in Diffusion mode (CMAD), with two different approaches to the Set-Membership technique. The first one is based on the Set-Membership Least Mean Squares (SM-LMS) algorithm, also developed in the diffusion mode. We extend the algorithm to the unsupervised context, denoting by Set-Membership CMA in Diffusion mode (SM-CMAD). We show that this algorithm presents better or similar performance to CMAD, in terms of convergence speed, intersymbol interference threshold (IIS), and has the important advantage of reducing the exchange of information between nodes, saving energy and network resources. The second proposed algorithm is based on the Set-Membership of the Constant Modulus (SM-CM), which we extend to the context of wireless sensor networks in the diffusion mode. This algorithm is denoted by the Set-membership CMA in Diffusion mode Square-root Gamma (SM-CMAD-SG). This algorithm performs well when compared to CMAD and, when compared to SM-CMAD, we see that its main advantage lies in the economy in terms of the update of the filter coefficients, which reaches values above 70% in several scenarios without loss of performance, saving energy.