Novel Multiplierless Wideband Comb Compensator with High Compensation Capability

This paper proposes a novel multiplierless comb compensation filter, which has the absolute passband deviation less than 0.1 dB in the wide passband. The compensator consists of a cascade of two simple filter sections, both operating at a low rate. The magnitude characteristics of the two-component filters are synthesized as sinewave functions, in which the main design parameters correspond to the amplitudes of sinewave functions. A systematic procedure is followed to select synthesis parameters, which depend only on the number of cascaded comb filters. In particular, they are independent of the decimation factor. Comparisons with comb compensators from the literature illustrate the benefits of the proposed design.Consejo Nacional de Ciencia y Tecnología 17958

DESIGN OF MULTIPLIERLESS COMB COMPENSATORS WITH MAGNITUDE RESPONSE SYNTHESIZED AS SINEWAVE FUNCTIONS

This paper presents a research on design of multiplierless comb compensators with magnitude response synthesized as sinewave functions. First, it is elaborated the importance of comb decimation filter and why we need its compensator. In continuation are presented some favorable characteristics of comb compensator. The compensators, with magnitude characteristic synthesized as sinewave functions fulfill those favorable characteristics. Next, are described some most important results on design of compensators with sinewave-based magnitude responses including single and cascaded sinewave-based functions. In all designs are presented the overall corresponding magnitude responses and the zooms in the passband. The parameters of design generally depend only on number of cascaded combs and generally do not depend on decimation factor. Design parameters are presented in tables along with the corresponding required number of adders

Processamento eficiente de arranjos de microfones modulados em densidade de pulso

Orientador: Bruno Sanches MasieroDissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Elétrica e de ComputaçãoResumo: Atualmente, os microfones digitais modulados por densidade de pulso (PDM) são amplamente utilizados em aplicações comerciais, já que esta é uma maneira eficiente de transmitir informação de áudio para processadores digitais em dispositivos móveis. No entanto, como o estado-da-arte em algoritmos de processamento digital de arranjos assume que todos os sinais recebidos dos microfones estão em uma representação em banda-base, estes microfones digitais requerem custosos filtros de decimação de alta ordem para converter o fluxo PDM para a modulação por código de pulso (PCM) em banda base. Assim, a implementação destes algoritmos em sistemas embarcados, onde os recursos de processamento são críticos, ou em circuitos integrados (VLSI), onde a energia consumida e área também são críticas, pode se tornar muito dispendiosa devido ao uso de dezenas de filtros de decimação para converter os sinais de PDM para PCM. Essa dissertação explora e propõe métodos eficientes em recursos para a implementação de arranjo de microfones. Com esse intuito, primeiro explora os atuais métodos de design de filtros de decimação e, baseado neles, propõe um algoritmo para fazer o seu design otimizando área e consumo de potência. Também são discutidas as vantagens e desvantagens de se realizar o processamento de arranjo de microfones diretamente nos sinais PDM ao invés dos sinais em PCM. Finalmente propõe um método eficiente para implementação de arranjos de microfones baseado em filtros maximamente planos (MAXFLAT). Como resultado, um novo método para o design de filtros de decimação que optimiza o número de somas por segundo é proposto, assim como demonstra-se que que um filtro espacial implementado no domínio PDM precisa de menos recursos que outras implementação no domínio do tempo. Conclui-se, portanto, que a implementação baseada em filtros MAXFLAT tem um melhor compromiso entre requisitos de armazenamento e poder de computação que o estado-da-arte e os métodos no domínio do PDMAbstract: Nowadays, pulse-density modulated (PDM) digital microphones are widely used on commercial applications as they have become a popular way to deliver audio to digital processors on mobile applications. However, as state-of-the-art array processing algorithms assume that all microphone signals are available in pulse-code modulated (PCM) representation, these digital microphones require costly high-order decimation filters to translate PDM bitstreams to baseband multi-bit PCM signals. In that manner, the implementation of microphone array algorithms in embedded systems, where processing resources are critical, or in very large-scale integration (VLSI) circuits, where power and area are critical, may become very expensive because of the use of the tens of decimation filters required to convert PDM bitstreams into PCM signals. This thesis explores and proposes resource-efficient methods to implement microphone array beamforming. For this purpose, it first reviews the state-of-the-art decimation filter design methods and proposes an algorithm to design decimation filters optimizing area and power consumption. Then it discusses the trade-offs of doing the beamforming calculations at the PDM bitstreams instead of PCM signals and proposes an architecture to implement beamformers without decimation filters. Finally it proposes an efficient approach to implement beamformers based on maximally flat (MAXFLAT) filters. As a result, a new generalized method to design decimation filters optimizing the number of addition per second is proposed, and it is shown that a beamformer implemented in PDM domain requires less resources for its implementation in time domain than other methods. It is concluded that the proposed MAXFLAT-based approach has better storage versus computation efficiency than state-of-the-art and PDM domain implementation approachesMestradoTelecomunicações e TelemáticaMestre em Engenharia Elétric

Applications of MATLAB in Science and Engineering

The book consists of 24 chapters illustrating a wide range of areas where MATLAB tools are applied. These areas include mathematics, physics, chemistry and chemical engineering, mechanical engineering, biological (molecular biology) and medical sciences, communication and control systems, digital signal, image and video processing, system modeling and simulation. Many interesting problems have been included throughout the book, and its contents will be beneficial for students and professionals in wide areas of interest

Equalização digital para sistemas de transmissão ópticos coerentes

This thesis focus on the digital equalization of fiber impairments for coherent optical transmission systems. New efficient and low-complexity equalization and mitigation techniques that counteract fiber nonlinear impairments are proposed and the tradeoff between performance and complexity is numerically assessed and experimentally demonstrated in metro and long-haul 400G superchannels-based transmission systems. Digital backpropagation (DBP) based on low-complexity split-step Fourier method and Volterra series nonlinear equalizers are experimentally assessed in an uniform superchannel system. In contrast with standard DBP methods, these techniques prove to be able to be implemented with larger step-sizes, consequently requiring a reduced number of multiplications, and still achieve a significant reach extension over linear equalization techniques. Moreover, given its structure, the complexity of the proposed Volterra-based DBP approach can be easily adjusted by changing the nonlinear filter dimension according to the system requirements, thus providing a higher flexibility to the nonlinear equalization block. A frequency-hybrid superchannel envisioning near-future flexible networks is then proposed as a way to increase the system bit-rate granularity. The problematic of the power-ratio between superchannel carriers is addressed and optimized for linear and nonlinear operation regimes using three distinct FEC paradigms. Applying a single FEC to the entire superchannel has a simpler implementation and is found to be a more robust approach, tolerating larger uncertainties on the system parameters optimization. We also investigate the performance gain provided by the application of different DBP techniques in frequency-hybrid superchannel systems, and its implications on the optimum power-ratio. It is shown that the application of DBP can be restricted to the carrier transporting the higher cardinality QAM format, since the DBP benefit on the other carriers is negligible, which might bring a substantially complexity reduction of the DBP technique applied to the superchannel.A presente tese foca-se na equalização digital das distorções da fibra para sistemas óticos de transmissão coerente. São propostas novas técnicas eficientes e de baixa complexidade para a equalização e mitigação das distorções não lineares da fibra, e o compromisso entre desempenho e complexidade é testado numericamente e demonstrado experimental em sistemas de transmissão metro e longa distância baseados em supercanais 400G. A propagação digital inversa baseada no método de split-step Fourier e equalizadores não lineares de séries de Volterra de baixa complexidade são testadas experimentalmente num sistema baseado em supercanais uniformes. Ao contrário dos métodos convencionais utilizados, estas técnicas podem ser implementadas utilizando menos interações e ainda extender o alcance do sistema face às técnicas de equalização linear. Para além disso, a complexidade do método baseado em Volterra pode ser facilmente ajustada alterando a dimensão do filtro não linear de acordo com os requisitos do sistema, concedendo assim maior flexibilidade ao bloco de equalização não linear. Tendo em vista as futuras redes flexı́veis, um supercanal hı́brido na frequência é proposto de modo a aumentar a granularidade da taxa de transmissão do sistema. A problemática da relação de potência entre as portadoras do supercanal é abordada e optimizada em regimes de operação linear e não linear utilizando paradigmas diferentes de códigos correctores de erros. A aplicação de um único código corrector de erros à totalidade do supercanal mostra ser a abordagem mais robusta, tolerando maiores incertezas na optimização dos parâmetros do sistema. O ganho de desempenho dado pela aplicação de diferentes técnicas de propagação digital inversa em sistemas de supercanais hı́bridos na frequência é tamém analizado, assim como as suas implicações na relação óptima de potência. Mostra-se que esta pode ser restringida à portadora que transporta o formato de modulação de ordem mais elevada, uma vez que o benefı́cio trazido pelas restantes portadotas é negligenciável, permitindo reduzir significativamente a complexidade do algoritmo aplicado.Programa Doutoral em Telecomunicaçõe

Design of discrete-time filters for efficient implementation

Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2011.Cataloged from PDF version of thesis.Includes bibliographical references (p. 325-333).The cost of implementation of discrete-time filters is often strongly dependent on the number of non-zero filter coefficients or the precision with which the coefficients are represented. This thesis addresses the design of sparse and bit-efficient filters under different constraints on filter performance in the context of frequency response approximation, signal estimation, and signal detection. The results have applications in several areas, including the equalization of communication channels, frequency-selective and frequency-shaping filtering, and minimum-variance distortionless-response beamforming. The design problems considered admit efficient and exact solutions in special cases. For the more difficult general case, two approaches are pursued. The first develops low-complexity algorithms that are shown to yield optimal or near-optimal designs in many instances, but without guarantees. The second focuses on optimal algorithms based on the branch-and-bound procedure. The complexity of branch-and-bound is reduced through the use of bounds that are good approximations to the true optimal cost. Several bounding methods are developed, many involving relaxations of the original problem. The approximation quality of the bounds is characterized and efficient computational methods are discussed. Numerical experiments show that the bounds can result in substantial reductions in computational complexity.by Dennis Wei.Ph.D