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

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

COEFFICIENT QUANTIZATION EFFECTS ON NEW FILTERS BASED ON CHEBYSHEV FOURTH-KIND POLYNOMIALS

The aim of this paper is to construct non-recursive filters, extensively used type of digital filters in digital signal processing applications, based on Chebyshev orthogonal polynomials. The paper proposes the use of the fourth-kind Chebyshev polynomials as functions in generating new filters. In this kind, low-pass filters with linear phase responses are obtained. Comprenhansive study of the frequency response characteristics of the generated filter functions is presented. The effects of coefficient quantization as one type of quantization that influences a filter characteristic are investigated here also. The quantized-coefficient errors are considered based on the number of bits and the implementation algorithm

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

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

Design and Implementation of Complexity Reduced Digital Signal Processors for Low Power Biomedical Applications

Wearable health monitoring systems can provide remote care with supervised, inde-pendent living which are capable of signal sensing, acquisition, local processing and transmission. A generic biopotential signal (such as Electrocardiogram (ECG), and Electroencephalogram (EEG)) processing platform consists of four main functional components. The signals acquired by the electrodes are amplified and preconditioned by the (1) Analog-Front-End (AFE) which are then digitized via the (2) Analog-to-Digital Converter (ADC) for further processing. The local digital signal processing is usually handled by a custom designed (3) Digital Signal Processor (DSP) which is responsible for either anyone or combination of signal processing algorithms such as noise detection, noise/artefact removal, feature extraction, classification and compres-sion. The digitally processed data is then transmitted via the (4) transmitter which is renown as the most power hungry block in the complete platform. All the afore-mentioned components of the wearable systems are required to be designed and fitted into an integrated system where the area and the power requirements are stringent. Therefore, hardware complexity and power dissipation of each functional component are crucial aspects while designing and implementing a wearable monitoring platform. The work undertaken focuses on reducing the hardware complexity of a biosignal DSP and presents low hardware complexity solutions that can be employed in the aforemen-tioned wearable platforms. A typical state-of-the-art system utilizes Sigma Delta (Σ∆) ADCs incorporating a Σ∆ modulator and a decimation filter whereas the state-of-the-art decimation filters employ linear phase Finite-Impulse-Response (FIR) filters with high orders that in-crease the hardware complexity [1–5]. In this thesis, the novel use of minimum phase Infinite-Impulse-Response (IIR) decimators is proposed where the hardware complexity is massively reduced compared to the conventional FIR decimators. In addition, the non-linear phase effects of these filters are also investigated since phase non-linearity may distort the time domain representation of the signal being filtered which is un-desirable effect for biopotential signals especially when the fiducial characteristics carry diagnostic importance. In the case of ECG monitoring systems the effect of the IIR filter phase non-linearity is minimal which does not affect the diagnostic accuracy of the signals. The work undertaken also proposes two methods for reducing the hardware complexity of the popular biosignal processing tool, Discrete Wavelet Transform (DWT). General purpose multipliers are known to be hardware and power hungry in terms of the number of addition operations or their underlying building blocks like full adders or half adders required. Higher number of adders leads to an increase in the power consumption which is directly proportional to the clock frequency, supply voltage, switching activity and the resources utilized. A typical Field-Programmable-Gate-Array’s (FPGA) resources are Look-up Tables (LUTs) whereas a custom Digital Signal Processor’s (DSP) are gate-level cells of standard cell libraries that are used to build adders [6]. One of the proposed methods is the replacement of the hardware and power hungry general pur-pose multipliers and the coefficient memories with reconfigurable multiplier blocks that are composed of simple shift-add networks and multiplexers. This method substantially reduces the resource utilization as well as the power consumption of the system. The second proposed method is the design and implementation of the DWT filter banks using IIR filters which employ less number of arithmetic operations compared to the state-of-the-art FIR wavelets. This reduces the hardware complexity of the analysis filter bank of the DWT and can be employed in applications where the reconstruction is not required. However, the synthesis filter bank for the IIR wavelet transform has a higher computational complexity compared to the conventional FIR wavelet synthesis filter banks since re-indexing of the filtered data sequence is required that can only be achieved via the use of extra registers. Therefore, this led to the proposal of a novel design which replaces the complex IIR based synthesis filter banks with FIR fil-ters which are the approximations of the associated IIR filters. Finally, a comparative study is presented where the hybrid IIR/FIR and FIR/FIR wavelet filter banks are de-ployed in a typical noise reduction scenario using the wavelet thresholding techniques. It is concluded that the proposed hybrid IIR/FIR wavelet filter banks provide better denoising performance, reduced computational complexity and power consumption in comparison to their IIR/IIR and FIR/FIR counterparts

Decimacijski filtri bez množila temeljeni na izoštravanju i kompenzaciji amplitude

The simplest multiplierless decimation filter is the cascaded-integrator-comb (CIC) filter. However, CIC filters introduce a passband droop, which is intolerable in many applications. The droop can be reduced by connecting a linear-phase finite-impulse-response filter called compensator in cascade with CIC filter. Since CIC filters are multiplierless, CIC compensators with multiplierless structures are preferable. In the thesis, two methods for the design of multiplierless CIC compensators have been proposed. Both methods are based on minimization of the maximum passband deviation. However, the first method provides an efficient compensation by using coefficients expressed as sums of powers of two (SPT), whereas the second method brings simple compensator's structures by representing each coefficient as signed power of two. In both approaches, the optimum coefficients are found by using global optimization. In processing of wideband signals, CIC filter is often incapable of meeting the requirement for high folding-band attenuations. To improve CIC filter folding-band response, various structures have been developed. An efficient structure arises from polynomial sharpening of the folding-band response. This structure implements a so-called sharpened CIC (SCIC) filter. To obtain very high folding-band attenuations of SCIC filters, the minimax sharpening of the folding bands is proposed. In addition, to obtain multiplierless SCIC structures, polynomials with SPT coefficients are used. However, the SCIC response also introduces a high passband droop. The droop can be reduced by connecting a compensator in cascade with the SCIC filter. For the multiplierless SCIC filters, multiplierless compensators are preferable. In the thesis, two approaches for design of multiplierless SCIC compensators are proposed. The first approach brings a closed-form method based on maximally flat approximation. Such an approximation is suitable for narrowband SCIC filters. The second approach results in a global method based on the minimization of the maximum passband deviation. This method is preferable for wideband SCIC filters.Najjednostavniji decimacijski filtar bez množila sastoji se od kaskade integratorskih i češljastih sekcija (Cascaded Integrator-Comb, CIC). Međutim, CIC filtri uvode veliki pad amplitudne karakteristike u području propuštanja, koji nije prihvatljiv u mnogim primjenama. Ovaj pad može se smanjiti spajanjem filtra s konačnim impulsnim odzivom i linearnom fazom, zvanog CIC kompenzator, u kaskadu sa CIC filtrom. S obzirom da CIC filtri ne sadrže množila, preferiraju se i CIC kompenzatori bez množila. U radu su predložene dvije metode za projektiranje CIC kompenzatora koji ne sadrže množila. Obje metode su temeljene na minimizaciji maksimalnog odstupanja u području propuštanja. Prva metoda osigurava učinkovitu kompenzaciju korištenjem koeficijenata izraženih sumama potencija broja dva (Sum of Power of Two, SPT), dok druga metoda donosi jednostavne strukture kompenzatora izražavajući svaki koeficijent kao predznačenu potenciju broja dva. U oba pristupa, optimalni koeficijenti se dobivaju primjenom globalnih optimizacijskih postupaka. U obradi širokopojasnih signala, CIC filtar često ne može osigurati dovoljno velika gušenja u područjima preklapanja spektra. Za poboljšanje amplitude CIC filtra u ovim područjima razvijene su razne strukture. Jedna od učinkovitih struktura proizlazi iz polinomnog izoštravanja amplitude. Ova struktura implementira takozvani izoštreni CIC (Sharpened CIC, SCIC) filtar. Kako bi se postiglo veliko gušenje preklopljenih signala kod SCIC filtera, predloženo je izoštravanje područja preklapanja spektra temeljeno na minimax aproksimaciji. Osim toga, da bi se dobile SCIC strukture bez množila, korišteni su polinomi s SPT koeficijentima. Međutim, odziv SCIC filtra ima veliki propad u području propuštanja. Propad se može smanjiti spajanjem kompenzatora u kaskadi s SCIC filtrom. Za SCIC filtre bez množila, također su poželjni kompenzatori bez množila. U radu se predlažu dva pristupa za projektiranje ovakvih kompenzatora. Prvi pristup rezultira eksplicitnom metodom temeljenom na maksimalno glatkoj aproksimaciji. Takva aproksimacija je prikladna za uskopojasne SCIC filtre. Drugi pristup rezultira globalnom metodom koja se temelji na minimiziranju maksimalne devijacije u području propuštanja. Ova metoda je prikladna za širokopojasne SCIC filtre

Decimacijski filtri bez množila temeljeni na izoštravanju i kompenzaciji amplitude

The simplest multiplierless decimation filter is the cascaded-integrator-comb (CIC) filter. However, CIC filters introduce a passband droop, which is intolerable in many applications. The droop can be reduced by connecting a linear-phase finite-impulse-response filter called compensator in cascade with CIC filter. Since CIC filters are multiplierless, CIC compensators with multiplierless structures are preferable. In the thesis, two methods for the design of multiplierless CIC compensators have been proposed. Both methods are based on minimization of the maximum passband deviation. However, the first method provides an efficient compensation by using coefficients expressed as sums of powers of two (SPT), whereas the second method brings simple compensator's structures by representing each coefficient as signed power of two. In both approaches, the optimum coefficients are found by using global optimization. In processing of wideband signals, CIC filter is often incapable of meeting the requirement for high folding-band attenuations. To improve CIC filter folding-band response, various structures have been developed. An efficient structure arises from polynomial sharpening of the folding-band response. This structure implements a so-called sharpened CIC (SCIC) filter. To obtain very high folding-band attenuations of SCIC filters, the minimax sharpening of the folding bands is proposed. In addition, to obtain multiplierless SCIC structures, polynomials with SPT coefficients are used. However, the SCIC response also introduces a high passband droop. The droop can be reduced by connecting a compensator in cascade with the SCIC filter. For the multiplierless SCIC filters, multiplierless compensators are preferable. In the thesis, two approaches for design of multiplierless SCIC compensators are proposed. The first approach brings a closed-form method based on maximally flat approximation. Such an approximation is suitable for narrowband SCIC filters. The second approach results in a global method based on the minimization of the maximum passband deviation. This method is preferable for wideband SCIC filters.Najjednostavniji decimacijski filtar bez množila sastoji se od kaskade integratorskih i češljastih sekcija (Cascaded Integrator-Comb, CIC). Međutim, CIC filtri uvode veliki pad amplitudne karakteristike u području propuštanja, koji nije prihvatljiv u mnogim primjenama. Ovaj pad može se smanjiti spajanjem filtra s konačnim impulsnim odzivom i linearnom fazom, zvanog CIC kompenzator, u kaskadu sa CIC filtrom. S obzirom da CIC filtri ne sadrže množila, preferiraju se i CIC kompenzatori bez množila. U radu su predložene dvije metode za projektiranje CIC kompenzatora koji ne sadrže množila. Obje metode su temeljene na minimizaciji maksimalnog odstupanja u području propuštanja. Prva metoda osigurava učinkovitu kompenzaciju korištenjem koeficijenata izraženih sumama potencija broja dva (Sum of Power of Two, SPT), dok druga metoda donosi jednostavne strukture kompenzatora izražavajući svaki koeficijent kao predznačenu potenciju broja dva. U oba pristupa, optimalni koeficijenti se dobivaju primjenom globalnih optimizacijskih postupaka. U obradi širokopojasnih signala, CIC filtar često ne može osigurati dovoljno velika gušenja u područjima preklapanja spektra. Za poboljšanje amplitude CIC filtra u ovim područjima razvijene su razne strukture. Jedna od učinkovitih struktura proizlazi iz polinomnog izoštravanja amplitude. Ova struktura implementira takozvani izoštreni CIC (Sharpened CIC, SCIC) filtar. Kako bi se postiglo veliko gušenje preklopljenih signala kod SCIC filtera, predloženo je izoštravanje područja preklapanja spektra temeljeno na minimax aproksimaciji. Osim toga, da bi se dobile SCIC strukture bez množila, korišteni su polinomi s SPT koeficijentima. Međutim, odziv SCIC filtra ima veliki propad u području propuštanja. Propad se može smanjiti spajanjem kompenzatora u kaskadi s SCIC filtrom. Za SCIC filtre bez množila, također su poželjni kompenzatori bez množila. U radu se predlažu dva pristupa za projektiranje ovakvih kompenzatora. Prvi pristup rezultira eksplicitnom metodom temeljenom na maksimalno glatkoj aproksimaciji. Takva aproksimacija je prikladna za uskopojasne SCIC filtre. Drugi pristup rezultira globalnom metodom koja se temelji na minimiziranju maksimalne devijacije u području propuštanja. Ova metoda je prikladna za širokopojasne SCIC filtre