In this thesis we present new approaches in the domains of soft fault detection and location in complex wire networks, based on the properties of time reversal. When addressing the detection of soft faults, the idea is to adapt the testing signal to the network under test, instead of being predefined for all the tested networks, as opposed to standard reflectometry techniques. We prove that this approach, which we name the Matched Pulse approach (MP), is beneficial whenever the system is more complex, i.e., its response is richer in echoes, which is opposed to common understanding. The MP analysis is conducted via a formal mathematical analysis, followed by simulation and experimental results validating the proposed approach. In the domain of soft fault location, and based on the DORT (Décomposition de l'Opérateur de Retournement Temporel) properties, we derive a distributive non-iterative method able to synthesize signals that focus on the fault position. Through a statistical study we analyze some of the influencing parameters on the performance of the method, and then simulation and experimental results show that the method is able to synthesize signals directly focalizing on the soft fault position, without the need for iterations.

Direct sequence code division multiple access (DS-CDMA) technique is one of the significant multiplexing technologies used in wireless communication services. In the DS-CDMA framework, all users have been assigned distinct signature code sequence to achieve multiple accesses within the same frequency band, and allow signal separating at the receiver. Under multipath fading environment with near-far effect, the current CDMA systems employed the RAKE receiver, to enhance the system performance. It is known that if training data is available the minimum mean squares error (MMSE) multiuser receiver, in which the average power of the receiver output is minimized subject to appropriate constraints, could be obtained by solving directly by the constrained Wiener estimation solution. However, if this is not the case, the blind multiuser receiver is an alternative approach to achieve desired performance closed to the one with the MMSE approach. In this thesis, based on the max/min criterion, the blind multiuser receiver, with linear constraints, is devised. Here constraint equations are written in parametric forms, which depend on the multipath structure of the signal of interest. Constraint parameters are jointly optimized with the parameters of the linear receiver to obtain the optimal parameters. In consequence, the sliding window linearly constrained RLS (SW-LC-RLS) algorithm is employed to implement the optimal blind receiver, with max/min approach. This new proposed scheme can be used to deal with multiple access interference (MAI) suppression for the environments, in which the narrow band interference (NBI) due to other systems is joined suddenly to the DS-CDMA systems, and having serious near-far effect. Under such circumstance, the channel character due to the NBI and near-far effect will become violent time varying, such that the conventional LC-RLS algorithm as well as LC-LMS algorithms could not perform well. Via computer simulation it confirms that our proposed scheme has better capability for MAI suppression in DS-CDMA systems than other existing schemes, and is more robust against the NBI and near-far problems.

Made available in DSpace on 2015-05-08T14:59:39Z (GMT). No. of bitstreams: 1 parte3.pdf: 17677016 bytes, checksum: 0da8175d15bd36a40467f7f50abd9987 (MD5) Previous issue date: 2012-03-28 / Coordenação de Aperfeiçoamento de Pessoal de Nível Superior - CAPES / This work presents a study on the application of an ADALINE neural network acting as a notch filter applied to the estimation of the stator flux, in order to obtain the resulting electromagnetic torque of three-phase induction motors (MIT). The estimation of the stator flux was performed by means of the voltage model of the induction machine, in which a integrator is directly applied over the stator counter electromotive force. The ADALINE neural network adaptive filter is employed in this research with the purpose of eliminating existing CC levels, which are present due to the problem of the initial values of the integrator and in the voltage and current measurements. Simulated and experimental results are presented to validate the proposed strategy. The algorithm used in the ADALINE adaptive neural filter simulations was created on the MATLABTM language, and the algorithm used for both the simulations and the laboratory experiments to estimate the flux and the torque was created in the C/C++ language. The hardware used to confirm the effectiveness of the proposed method is based on the Texas Instruments DSP TMS320F28335 platform, along with and induction motor manufactured by WEG, model W21 High Efficiency / Apresenta-se neste trabalho um estudo do emprego de rede neural ADALINE funcionando como notch filter , aplicada na estimação do fluxo estatórico para consequente obtenção do conjugado eletromagnético de Motores de Indução Trifásicos (MIT). A estimação do fluxo do estator foi feita por meio do modelo de tensão da máquina de indução, em que é aplicado um integrador diretamente sobre a força contra eletromotriz do estator. O filtro neural adaptativo ADALINE é empregado nesta pesquisa com o objetivo de eliminar os níveis de cc presentes devido ao problema de valores iniciais do integrador nas medições de tensão e corrente. São apresentados resultados simulados e experimentais para validação da estratégia proposta. O algoritmo utilizado nas simulações do filtro neural adaptativo ADALINE foi elaborado na linguagem computacional MATLABTM, e o algoritmo utilizado tanto nas simulações, como nas experiências em laborátorio para estimação do fluxo e do conjugado foram elaboradas na linguagem computacional C/C++. O hardware utilizado para comprovar a eficácia do método proposto neste trabalho tem como base a plataforma DSP TMS320F28335 da Texas Instruments , juntamente com o um motor de indução fabricado pela WEG modelo W21 Alto Rendimento.

Este trabalho apresenta um método para a estimação da freqüência em sistemas elétricos de potência utilizando filtros adaptativos baseados no algoritmo dos mínimos quadrados (LMS - least mean square). A análise do sistema de potência é realizada através da conversão das tensões trifásicas em um sinal complexo pela aplicação da transformada \'alfa\'\'beta\', cuja forma complexa foi direcionada ao algoritmo de filtragem adaptativa. O método é baseado na aplicação da filtragem adaptativa para a realização de rastreio do sinal de entrada, o que permite verificar o seu comportamento variante no tempo. O algoritmo proposto foi testado através de formas de ondas geradas com o software Matlab e de simulações realizadas através do software Alternative Transients Program (ATP). É importante salientar que nas simulações do ATP foram modelados diversos equipamentos que constituem o sistema elétrico de potência, incluindo um gerador síncrono com regulação de velocidade, linhas de transmissão com variação em freqüência e transformadores de potência com suas respectivas curvas de saturação. Estas modelagens tiveram por objetivo gerar dados das mais diversas e distintas situações para a verificação e análise da metodologia proposta. Os resultados da pesquisa mostram a excelência na aplicabilidade do algoritmo proposto na estimação da freqüência de um sistema elétrico, mesmo com sinais ruidosos, além do rastreio fiel da freqüência em situações de manobra e operação. Alguns dos resultados apresentados comparam as estimações obtidas pela técnica proposta em relação às estimações de um determinado relé comercial, habilitado à supervisão da freqüência. / This work presents a method for frequency estimation in power systems using adaptive filters based in the algorithm of least mean square (LMS). The analysis of the power system is made through the conversion of the three-phase voltages in a complex signal with the application of \'alfa\'\'beta\' transform, whose complex form was directed to the algorithm of adaptive filtering. The method is based on the application of the adaptive filtering for tracking the input signal, and it allows verifying its variant behavior in time. The algorithm was tested through waveforms generated by Matlab software and simulations carried out through Alternative Transients Program (ATP) software. It is important to point out that in the simulations using ATP many diferent power system equipments had been modeled, including a synchronous generator with speed regulation, transmission lines with variation in frequency and power transformers with their saturation curves. The objective of these tests was to generate data for diverse and distinct situations for the verification and the analysis of the proposed methodology. The results of the research show the excellence in the applicability of the algorithm considered in frequency estimation of an electrical system, even with noisy signals, as well as the tracking of the frequency during operation. Some of the results are compared to the ones presented by a commercial relay set to track frequency.

Este trabalho apresenta um método para a estimação da freqüência em sistemas elétricos de potência utilizando filtros adaptativos baseados no algoritmo dos mínimos quadrados (LMS - least mean square). A análise do sistema de potência é realizada através da conversão das tensões trifásicas em um sinal complexo pela aplicação da transformada \'alfa\'\'beta\', cuja forma complexa foi direcionada ao algoritmo de filtragem adaptativa. O método é baseado na aplicação da filtragem adaptativa para a realização de rastreio do sinal de entrada, o que permite verificar o seu comportamento variante no tempo. O algoritmo proposto foi testado através de formas de ondas geradas com o software Matlab e de simulações realizadas através do software Alternative Transients Program (ATP). É importante salientar que nas simulações do ATP foram modelados diversos equipamentos que constituem o sistema elétrico de potência, incluindo um gerador síncrono com regulação de velocidade, linhas de transmissão com variação em freqüência e transformadores de potência com suas respectivas curvas de saturação. Estas modelagens tiveram por objetivo gerar dados das mais diversas e distintas situações para a verificação e análise da metodologia proposta. Os resultados da pesquisa mostram a excelência na aplicabilidade do algoritmo proposto na estimação da freqüência de um sistema elétrico, mesmo com sinais ruidosos, além do rastreio fiel da freqüência em situações de manobra e operação. Alguns dos resultados apresentados comparam as estimações obtidas pela técnica proposta em relação às estimações de um determinado relé comercial, habilitado à supervisão da freqüência. / This work presents a method for frequency estimation in power systems using adaptive filters based in the algorithm of least mean square (LMS). The analysis of the power system is made through the conversion of the three-phase voltages in a complex signal with the application of \'alfa\'\'beta\' transform, whose complex form was directed to the algorithm of adaptive filtering. The method is based on the application of the adaptive filtering for tracking the input signal, and it allows verifying its variant behavior in time. The algorithm was tested through waveforms generated by Matlab software and simulations carried out through Alternative Transients Program (ATP) software. It is important to point out that in the simulations using ATP many diferent power system equipments had been modeled, including a synchronous generator with speed regulation, transmission lines with variation in frequency and power transformers with their saturation curves. The objective of these tests was to generate data for diverse and distinct situations for the verification and the analysis of the proposed methodology. The results of the research show the excellence in the applicability of the algorithm considered in frequency estimation of an electrical system, even with noisy signals, as well as the tracking of the frequency during operation. Some of the results are compared to the ones presented by a commercial relay set to track frequency.

With the development of VLSI and wireless communication technology, portable devices such as personal digital assistants (PDAs), pocket PCs, and mobile phones have gained a lot of popularity. Many such devices incorporate a speech recognition engine, enabling users to interact with the devices using voice-driven commands and text-to-speech synthesis. The power consumption of DSP microprocessors has been consistently decreasing by half about every 18 months, following Gene's law. The capacity of signal processing, however, is still significantly constrained by the limited power budget of these portable devices. In addition, analog-to-digital (A/D) converters can also limit the signal processing of portable devices. Many systems require very high-resolution and high-performance A/D converters, which often consume a large fraction of the limited power budget of portable devices. The proposed research develops a low-power audio signal enhancement system that combines programmable analog signal processing and traditional digital signal processing. By utilizing analog signal processing based on floating-gate transistor technology, the power consumption of the overall system as well as the complexity of the A/D converters can be reduced significantly. The system can be used as a front end of portable devices in which enhancement of audio signal quality plays a critical role in automatic speech recognition systems on portable devices. The proposed system performs background audio noise suppression in a continuous-time domain using analog computing elements and acoustic echo cancellation in a discrete-time domain using an FPGA.

Orthogonal frequency division multiplexing (OFDM) systems with cyclic prefix (CP) can be used to protect signal from the time-variant multipath channel induced distortions. However, the presence of CP could greatly decrease the effective data rate, thus many recent research works have been focused on the multiple-input multiple-output (MIMO) OFDM systems without CP (CP-free), equipped with the space-time block codes (ST-BC). The constraint of the conventional MIMO-OFDM (without using the ST-BC) system is that the number of receive-antenna has to be greater than the transmit-antenna. In this thesis, we first consider the ST-BC MIMO-OFDM system and show that the above-mentioned constraint can be removed, such that the condition become that the receive antenna should be greater than one, that is the basic requirement for MIMO system. It is particular useful and confirm to the recently specification, e.g., 3GPP LTE (Long Term Evolution) where the system deploy the 2¡Ñ2 or 4¡Ñ4 antennas systems. This thesis also considers the effects of peak-to-average power ratio (PAPR) in the transmitter and In-phase/ Quadrature-phase (IQ) imbalance in the receiver, and solves them by using the adaptive Volterra predistorter and blind adaptive filtering approach of the nonlinear parameters estimation and compensation, along with the power measurement, respectively. After the compensator of IQ imbalance in the receiver, an equalizer under the framework of generalized sidelobe canceller (GSC) is derived for interference suppression. To further reduce the complexity of receiver implementation, the partially adaptive (PA) scheme is applied by exploiting the structural information of the signal and interference signature matrices. As demonstrated from computer simulation results, the performance of the proposed CP-free ST-BC MIMO-OFDM receiver is very similar to that obtained by the conventional CP-based ST-BC MIMO-OFDM system under either the predistortion or compensation scenario.

¡@¡@Pseudo-noise (PN) code synchronizer is an essential element of direct-sequence code division multiple access (DS-CDMA) system because data transmission is possible only after the receiver accurately synchronizes the locally generated PN code with the incoming PN code. The code synchronization is processed in two steps, acquisition and tracking, to estimate the delay offset between the two codes. Recently, the adaptive LMS filtering scheme has been proposed for performing both code acquisition and tracking with the identical structure, where the LMS algorithm is used to adjust the FIR filter taps to search for the value of delay-offset adaptively. A decision device is employed in the adaptive LMS filtering scheme as a decision variable to indicate code synchronization, hence it plays an important role for the performance of mean acquisition time (MAT). In this thesis, only code acquisition is considered. ¡@¡@In this thesis, a new decision device, referred to as the weight vector square norm (WVSN) test method, is devised associated with the adaptive LMS filtering scheme for code acquisition in DS-CDMA system. The system probabilities of the proposed scheme are derived for evaluating MAT. Numerical analyses and simulation results verify that the performance of the proposed scheme, in terms of detection probability and MAT, is superior to the conventional scheme with mean-squared error (MSE) test method, especially when the signal-to-interference-plus-noise ratio (SINR) is relatively low. ¡@¡@Furthermore, an efficient and joint-adaptation code acquisition scheme, i.e., a smart antenna coupled with the proposed adaptive LMS filtering scheme with the WVSN test method, is devised for applying to a base station, where all antenna elements are employed during PN code acquisition. This new scheme is a process of PN code acquisition and the weight coefficients of smart antenna jointly and adaptively. Numerical analyses and simulation results demonstrate that the performance of the proposed scheme with five antenna elements, in terms of the output SINR, the detection probability and the MAT, can be improved by around 7 dB, compared to the one with single antenna case.

A novel and efficient methodology is presented for the conversion of stereo to multi-view video in order to address the 3D content requirements for the next generation 3D-TVs and auto-stereoscopic multi-view displays. There are two main algorithmic blocks in such a conversion system / stereo matching and virtual view rendering that enable extraction of 3D information from stereo video and synthesis of inexistent virtual views, respectively. In the intermediate steps of these functional blocks, a novel edge-preserving filter is proposed that recursively constructs connected support regions for each pixel among color-wise similar neighboring pixels. The proposed recursive update structure eliminates pre-defined window dependency of the conventional approaches, providing complete content adaptibility with quite low computational complexity. Based on extensive tests, it is observed that the proposed filtering technique yields better or competitive results against some leading techniques in the literature. The proposed filter is mainly applied for stereo matching to aggregate cost functions and also handles occlusions that enable high quality disparity maps for the stereo pairs. Similar to box filter paradigm, this novel technique yields matching of arbitrary-shaped regions in constant time. Based on Middlebury benchmarking, the proposed technique is currently the best local matching technique in the literature in terms of both precision and complexity. Next, virtual view synthesis is conducted through depth image based rendering, in which reference color views of left and right pairs are warped to the desired virtual view using the estimated disparity maps. A feedback mechanism based on disparity error is introduced at this step to remove salient distortions for the sake of visual quality. Furthermore, the proposed edge-aware filter is re-utilized to assign proper texture for holes and occluded regions during view synthesis. Efficiency of the proposed scheme is validated by the real-time implementation on a special graphics card that enables parallel computing. Based on extensive experiments on stereo matching and virtual view rendering, proposed method yields fast execution, low memory requirement and high quality outputs with superior performance compared to most of the state-of-the-art techniques.

The objective of this research is to presents new acoustic echo cancellation design methods that can effectively work in the nonlinear environment. Acoustic echo is an annoying issue for voice communication systems. Because of room acoustics and delay in the transmission path, echoes affect the sound quality and may hamper communications. Acoustic echo cancellers (AECs) are employed to remove the acoustic echo while keeping full-duplex communications. AEC designs face a variety of challenges, including long room impulse response, acoustic path nonlinearity, ambient noise, and double-talk situation. We investigate two parts of echo canceller design: echo cancellation algorithm design and control logic algorithm design. In the first part, our work focuses on the nonlinear adaptive and fast-convergence algorithms. We investigate three different structures: predistortion linearization, cascade structure, and nonlinear residual echo suppressor. Specifically, we are interested in the coherence function, since it provides a means for quantifying linear association between two stationary random processes. By using the coherence as a criterion to design the nonlinear echo canceller in the system, our method guarantees the algorithm stability and leads to a faster convergence rate. In the second part, our work focuses on the robustness of AECs in the presence of interference. With regard to the near-end speech, we investigate the double-talk detector (DTD) design in conjunction with nonlinear AECs. Specifically, we propose to design a DTD based on the mutual information (MI). We show that the advantage of the MI-based method, when compared with the existing methods, is that it is applicable to both the linear and nonlinear scenarios. With respect to the background noise, we propose a variable step-size and variable tap-length least mean square (LMS) algorithm. Based on the fact that the room impulse response usually exhibits an exponential decay power profile in acoustic echo cancellation applications, the proposed method finds optimal step size and tap length at each iteration. Thus, it achieves faster convergence rate and better steady-state performance. We show a number of experimental results to illustrate the performance of the proposed algorithms.