Induction motor diagnosis by advanced notch FIR filters and the wigner-ville distribution

During the last years, several time-frequency decomposition tools have been applied for the diagnosis of induction motors, for those cases in which the traditional procedures, such as motor current signature analysis, cannot yield the necessary response. Among them, the Cohen distributions have been widely selected to study transient and even stationary operation due to their high-resolution and detailed information provided at all frequencies. Their main drawback, the cross-terms, has been tackled either modifying the distribution, or carrying out a pretreatment of the signal before computing its time-frequency decomposition. In this paper, a filtering process is proposed that uses advanced notch filters in order to remove constant frequency components present in the current of an induction motor, prior to the computation of its distribution, to study rotor asymmetries and mixed eccentricities. In transient operation of machines directly connected to the grid, this procedure effectively eliminates most of the artifacts that have prevented the use of these tools, allowing a wideband analysis and the definition of a precise quantification parameter able to follow the evolution of their state. © 1982-2012 IEEE

The effect of coefficient quantization optimization on filtering performance and gate count

Abstract. Digital filters are an essential component of Digital Signal Processing (DSP) applications and play a crucial role in removing unwanted signal components from a desired signal. However, digital filters are known to be resource-intensive and consume a large amount of power, making it important to optimize their design in order to minimize hardware requirements such as multipliers, adders, and registers. This trade-off between filter performance and hardware consumption can be influenced by the quantization of filter coefficients. Therefore, this thesis investigates the quantization of Finite Impulse Response (FIR) filter coefficients and analyzes its impact on filter performance and hardware resource consumption. A method called dynamic quantization is introduced and an algorithm for step-by-step dynamic quantization is provided to improve upon the results obtained with the classical fixed point quantization method. To demonstrate the effectiveness of this approach, the dynamic quantization of filter coefficients for a Low-pass Equiripple FIR filter is examined and a comparative study of the magnitude response and hardware consumption of the generated filter using both the classical and dynamic quantization methods is presented. By understanding the trade-offs and benefits of each quantization method, engineers can make informed decisions about the most appropriate approach for their specific application

A robust and scalable implementation of the Parks-McClellan algorithm for designing FIR filters

Preliminary version accepted for publicationInternational audienceWith a long history dating back to the beginning of the 1970s, the Parks-McClellan algorithm is probably the most well-known approach for designing finite impulse response filters. Despite being a standard routine in many signal processing packages, it is possible to find practical design specifications where existing codes fail to work. Our goal is twofold. We first examine and present solutions for the practical difficulties related to weighted minimax polynomial approximation problems on multi-interval domains (i.e., the general setting under which the Parks-McClellan algorithm operates). Using these ideas, we then describe a robust implementation of this algorithm. It routinely outperforms existing minimax filter design routines

Design And Implementation Of Low Passband Ripple Digital Down Converter Filter For Software Defined Radio Transceiver

The main aim of this research is the design and implementation of the Digital Down Converter (DDC) filter with low passband ripple and high attenuation in the adjacent rejection and blocker requirements in the filter response for Software Defined Radio (SDR) transceiver to decrease the power consumption and avoid the interference in the channel. The proposed DDC filters incorporate of Remez algorithm and Mini-max algorithm to reduce the error rate in the filter response. The DDC filter is acombination of 5-stages Cascaded Integrated Comb (CIC) filter and two linear phase Equiripple FIR filter (CFIR and PFIR). The passband ripple, adjacent rejection and blocker band is developed by controlling the transition width, filter order and weight function of the FIR filter using MATLAB and Xilinx System Generator environment

Design of interpolative sigma delta modulators via a semi- infinite programming approach

This paper considers the design of interpolative sigma delta modulators (SDMs). The design problem is formulated as two different optimization problems. The first optimization problem is to determine the denominator coefficients. The objective of the optimization problem is to minimize the energy of the error function in the passband of the loop filter in which the error function reflects the noise output transfer function and the ripple of the input output transfer function. The constraint of the optimization problem refers to the specification of the error function defined in the frequency domain. The second optimization problem is to determine the numerator coefficients in which the cost function is to minimize the stopband ripple energy of the loop filter subject to the stability condition of the noise output and input output transfer functions. These two optimization problems are actually quadratic semi-infinite programming (SIP) problems. By employing our recently proposed dual parameterization method for solving the problems, global optimal solutions that satisfy the corresponding continuous constraint are guaranteed if the solutions exist. The advantages of this formulation are the guarantee of the stability of the noise output and input output transfer functions, applicability to design rational IIR filters without imposing specific filter structures such as Laguerre filter and Butterworth filter structures, and the avoidance of the iterative design of numerator and the denominator coefficients because the convergence of the iterative design is not guaranteed. Our simulation results show that this proposed design yields a significant improvement in the signal-to-noise ratio (SNR) compared to the existing designs

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

Digital quadrature demodulation of Doppler signals

Dissertação de mest., Engenharia Electrónica e Telecomunicações, Faculdade de Ciências e Tecnologia, Universidade do Algarve, 2009Ultrasound has for many years been an important tool in the detection and quantification of various health problems. In vascular diseases, for example, the ultrasound can be applied with different techniques such as Transit-Time Flow Measurements (TTFM) [36], Doppler [28][40][41][42] and elastography [37] [38]. Research has been developed focusing the signal processing of Doppler ultrasound signals. In an ongoing project, named Desarrollo de Sistemas Ultras´onicos y Computacionales para Diagn´ostico Cardiovascular (SUCoDiC), Doppler ultrasound signals are processed by an analog signal processing unit, in order to obtain the inphase (I) and quadrature (Q) components of the Doppler ultrasound signals, to allow directional blood flow separation. Problems associated with unbalanced channels’ gain of the employed analog system have been detected, resulting in an inapropriate directional blood flow separation. This thesis reports the research performed to eliminate such problems by substituting the analog system’s demodulator by digital signal processing approaches aiming at the achievement of the same goals, i.e., obtaining the Doppler ultrasound signal’s inphase (I) and quadrature (Q) components, for efficient directional blood flow separation. Five digital quadrature techniques have been studied to achieve such goal. Also, given technical constraints imposed by the nature of the Doppler ultrasound signals to be used, and limitations of the sampling rate of the Analog-to-Digital Converter (ADC) used, two strategies to acquire the Doppler ultrasound signals were studied. Such strategies involved the sampling of a downconversion version of the Doppler ultrasound signals (by application of the heterodyne function) and direct sampling of the Doppler ultrasound signals using uniform bandpass sampling. From the results obtained, three approaches are selected and proposed for real time implementation. Comparison between both signal sampling strategies employed are also presente

Design of interpolative sigma-delta modulators via semi-indefinite programming

This correspondence considers the optimized design of interpolative sigma delta modulators (SDMs). The first optimization problem is to determine the denominator coefficients. The objective of the optimization problem is to minimize the passband energy of the denominator of the loop filter transfer function (excluding the dc poles) subject to the continuous constraint of this function defined in the frequency domain. The second optimization problem is to determine the numerator coefficients in which the cost function is to minimize the stopband ripple energy of the loop filter subject to the stability condition of the noise transfer function (NTF) and signal transfer function (STF). These two optimization problems are actually quadratic semi-infinite programming (SIP) problems. By employing the dual-parameterization method, global optimal solutions that satisfy the corresponding continuous constraints are guaranteed if the filter length is long enough. The advantages of this formulation are the guarantee of the stability of the transfer functions, applicability to design of rational infinite-impulse-response (IIR) filters without imposing specific filter structures, and the avoidance of iterative design of numerator and denominator coefficients. Our simulation results show that this design yields a significant improvement in the signal-to-noise ratio (SNR) and have a larger stability range, compared with the existing designs

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

Optimal design of all-pass variable fractional-delay digital filters

This paper presents a computational method for the optimal design of all-pass variable fractional-delay (VFD) filters aiming to minimize the squared error of the fractional group delay subject to a low level of squared error in the phase response. The constrained optimization problem thus formulated is converted to an unconstrained least-squares (LS) optimization problem which is highly nonlinear. However, it can be approximated by a linear LS optimization problem which in turn simply requires the solution of a linear system. The proposed method can efficiently minimize the total error energy of the fractional group delay while maintaining constraints on the level of the error energy of the phase response. To make the error distribution as flat as possible, a weighted LS (WLS) design method is also developed. An error weighting function is obtained according to the solution of the previous constrained LS design. The maximum peak error is then further reduced by an iterative updating of the error weighting function. Numerical examples are included in order to compare the performance of the filters designed using the proposed methods with those designed by several existing methods