On Passband and Stopband Cascaded-Integrator-Comb Improvements Using a Second Order IIR Filter

This paper proposes an efficient second order IIR filter which considerably improves the passband as well as the stopband of the cascaded-integrator-comb (CIC) filter. Using the polyphase decomposition of the proposed filter, all filtering can be moved to a lower rate, which is D times less than the high input rate, where D is the decimation factor. The overall phase response of the compensated CIC is approximately linear in the passband. The design parameters are the number of cascaded CIC filter N, the decimator factor D, the passband frequency wp, and a weighted parameter a

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

Optimal Sharpening of Compensated Comb Decimation Filters: Analysis and Design

Comb filters are a class of low-complexity filters especially useful for multistage decimation processes. However, the magnitude response of comb filters presents a droop in the passband region and low stopband attenuation, which is undesirable in many applications. In this work, it is shown that, for stringent magnitude specifications, sharpening compensated comb filters requires a lower-degree sharpening polynomial compared to sharpening comb filters without compensation, resulting in a solution with lower computational complexity. Using a simple three-addition compensator and an optimization-based derivation of sharpening polynomials, we introduce an effective low-complexity filtering scheme. Design examples are presented in order to show the performance improvement in terms of passband distortion and selectivity compared to other methods based on the traditional Kaiser-Hamming sharpening and the Chebyshev sharpening techniques recently introduced in the literature

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