Papers by Paulo S. R. Diniz
Proportionate adaptive filters can improve the convergence speed for the identification of sparse... more Proportionate adaptive filters can improve the convergence speed for the identification of sparse systems as compared to their conventional counterparts. In this paper, the idea of proportionate adaptation is combined with the framework of set-membership filtering (SMF) in an attempt to derive novel computationally efficient algorithms. The resulting algorithms attain an attractive faster converge for both situations of sparse and dispersive channels while decreasing the average computational complexity due to the data discerning feature of the SMF approach. In addition, we propose a rule that allows us to automatically adjust the number of past data pairs employed in the update. This leads to a set-membership proportionate affine projection algorithm (SM-PAPA) having a variable data-reuse factor allowing a significant reduction in the overall complexity when compared with a fixed data-reuse factor. Reduced-complexity implementations of the proposed algorithms are also considered th...
ICASSP 2019 - 2019 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), 2019
Set-membership affine projection (SM-AP) adaptive filters have been increasingly employed in the ... more Set-membership affine projection (SM-AP) adaptive filters have been increasingly employed in the context of online data-selective learning. A key aspect for their good performance in terms of both convergence speed and steady-state mean-squared error is the choice of the so-called constraint vector. Optimal constraint vectors were recently proposed relying on convex optimization tools, which might sometimes lead to prohibitive computational burden. This paper proposes a convex combination of simpler constraint vectors whose performance approaches the optimal solution closely, utilizing much fewer computations. Some illustrative examples confirm that the sub-optimal solution follows the accomplishments of the optimal one.
In this paper, a new structure, called the channel split-and-add method, for de- signing oversamp... more In this paper, a new structure, called the channel split-and-add method, for de- signing oversampled transmultiplexers and filter banks is presented. The proposed method is based on an initial design with an additional number of bands. The band number is then reduced to the desired value by the proper combination of adjacent and/or nonadjacent bands (subchannels). With the proposed approach it is always possible to perform the filtering tasks at the lowest data rate of the system. An example illustrates the design flexibility achieved with the proposed structure. Ke yw ords: Oversampled filter banks, transmultiplexers, digital filtering, multicarrier sys- tems.
IEEE Transactions on Signal Processing, 2004
Due to the growing importance of multichannel modulation, there has been great interest in the de... more Due to the growing importance of multichannel modulation, there has been great interest in the design of high-performance transmultiplex systems. In this paper, a new cosine-modulated transmultiplex structure is proposed based on a prototype filter designed with the frequency-response masking (FRM) approach. This new structure leads to substantial reduction in the computational complexity (number of multiplications per output sample) of the prototype filters having sharp transition band and equivalently small roll-off values. The relation between the interpolation factor used in the FRM prototype filter and the decimation factor in the subbands leads to distinct structures. Examples included indicate that the reduction in computational complexity can be higher than 50% of the current state-of-art designs, whereas the reduction on the number of distinct coefficients of the prototype filter can be reduced even further (over 75%). As a result, the proposed approach allows the design of very selective subfilters for transmultiplexes with a very large number of subchannels.
System Analysis and Design
In this paper, rational orthogonal basis functions are introduced to realize cascaded-parallel st... more In this paper, rational orthogonal basis functions are introduced to realize cascaded-parallel structure of adaptive notch filters (ANF) in the case of identifying and separating multiple sinusoids from wideband signals. The structural orthogonality provides good tradeoff between convergence speed and computational complexity. A simplified mean square output error (MSOE) updating algorithm is derived for the cascade-parallel realization. Simulation results confirm that the proposed structure converges faster compared with the cascade realizations with the same computational complexity.
1998 IEEE International Conference on Electronics, Circuits and Systems. Surfing the Waves of Science and Technology (Cat. No.98EX196)
ABSTRACT Orthogonal basis functions are a powerful tool for efficient system representation. Exce... more ABSTRACT Orthogonal basis functions are a powerful tool for efficient system representation. Except for the lattice realization, that is based on the nice properties of Szego orthonormal polynomials, no other adaptive IIR filter realization using orthonormal characteristics seems to be extensively studied in the literature. However, many orthogonal realizations for adaptive FIR filters, that are particularly suitable for rational modeling, have been proposed in past years. In this paper, we present some theoretical results related to the properties of a generalized orthonormal realization when used for mean square output error minimization in a system identification application. One result is related to the low computational complexity of the updating gradient algorithm when some properties of the orthonormal realization are used. An additional result establishes conditions for the stationary points of the proposed updating algorithm
Academic Press Library in Signal Processing, 2014
The present chapter presents a brief overview of some key aspects of the theory of signal process... more The present chapter presents a brief overview of some key aspects of the theory of signal processing trying as much as possible to interconnect them. We start by discussing the basic concepts related to the continuous- and discrete-time signals and systems with emphasis on the latter. Then several basic concepts of discrete-signal representation, signal quantization, filter design and implementation are discussed in order to review the basic fundamentals of the classical digital signal processing theory. This chapter proceeds with a discussion on multirate signal processing along with a summary of the aspects and benefits of employing filter banks. From this knowledge we can appreciate the importance of the tools available for signal representation such as discrete transforms multiscale representations and frames. The last part of the chapter briefly reviews the basic aspects of signal modeling and adaptive filtering which are basic tools in learning the properties of random signal acquired in many practical applications. All concepts discussed in the present chapter are illustrated through simple toy examples. It is also available in the book Web-page a Matlab© code illustrating the filtering and transform concepts. The aim of this first chapter is to revisit the basic concepts of signal processing that will be discussed in details in the chapters to follow.
Proceedings of the 1998 IEEE International Conference on Acoustics, Speech and Signal Processing, ICASSP '98 (Cat. No.98CH36181), 1998
Providing a quantitative mean-squared-error analysis of adaptation algorithms is of great importa... more Providing a quantitative mean-squared-error analysis of adaptation algorithms is of great importance for determining their usefulness and for comparison with other algorithms. However, when the algorithm reutilizes previous data, such analysis becomes very involved as the independence assumption cannot be used. In this paper, a thorough mean-squared-error analysis of the binormalized data-reusing LMS algorithm is carried out. The analysis is based on a simplified model for the input-signal vector, assuming independence between the continuous radial probability distribution and the discrete angular probability distribution. Throughout the analysis only parallel and orthogonal input-signal vectors are used in order to obtain a closed-form formula for the excess meansquared error. The formula agrees closely with simulation results even when the input-signal vector is a delay line. Furthermore, the analysis can be readily extended to other algorithms with expected similar accuracy.
APCCAS 2006 - 2006 IEEE Asia Pacific Conference on Circuits and Systems, 2006
Abstract— In recent years, several approaches have been proposed,aiming,the,optimal,joint design,... more Abstract— In recent years, several approaches have been proposed,aiming,the,optimal,joint design,of finite impulse response,(FIR) multiple-input multiple-output,(MIMO) trans- mitter and,receiver. Assuming,that the channel,input is power constrained,and,that the channel,model,is a known,FIR MIMO system, it is possible to design high performance transceivers in the presence of additive noise. Unfortunately, most optimal transceivers available so far, do not have fast implementation and,simple,design approaches. This work,proposes,a
Acoustical Science and Technology, 2003
2017 25th European Signal Processing Conference (EUSIPCO), 2017
The progressive increase of data rates in wireless communication systems has induced channel mode... more The progressive increase of data rates in wireless communication systems has induced channel models with sampled impulse responses which are mostly sparse. This paper presents a unified derivation of adaptive filters exploiting sparsity in the complex domain, and compares the performance of classic and state-of-the-art adaptive algorithms for estimating sparse wireless channels as well as their tracking ability in this inherently time-varying environment. Simulation results confirm the efficiency of the sparsity-aware algorithms.
Anais do XXI Simpósio Brasileiro de Telecomunicações
Resumo-Neste trabalho avaliamos a possibilidade de construção ou geração de frames de exponenciai... more Resumo-Neste trabalho avaliamos a possibilidade de construção ou geração de frames de exponenciais decrescentes. Istoé feito tanto no contexto de sistemas de Gabor gerados a partir de deslocamentos e modulações de uma exponencial decrescente como no caso de frames de wavelets gerados a partir de dilatações e deslocamentos de uma exponencial decrescente. Mostramos que com a primeira abordagemé possível gerar frames enquanto com a segunda não. Entretanto, mostramos formas de combinar dilatação, deslocamento e modulação para gerar frames de exponenciais decrescentes.
2003 IEEE International Conference on Acoustics, Speech, and Signal Processing, 2003. Proceedings. (ICASSP '03).
The frequency-response masking (FRM) method allows the design of selective prototype filters for ... more The frequency-response masking (FRM) method allows the design of selective prototype filters for cosine-modulated filter banks (CMFBs) with a reduced number of distinct coefficients. Such methodology may result in filter banks with large number of bands (e.g. 1024 or more) and a simplified optimization procedure, as there are less parameters to adjust. This work introduces a numerically efficient optimization procedure, based on a quasi-Newton algorithm, for designing selective FRM-based CMFBs. The proposed method uses a perfect-reconstruction FRM prototype filter as a starting point and updates the number of bands of the filter bank during the optimization procedure. Examples provided indicate that figures-of-merit, such as intersymbol and intercarrier interferences, for the optimized FRM-CMFB structure are significantly improved without increasing the complexity of the resulting structure.
2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221)
In this paper, we use the frequency-response masking (FRM) approach to design prototype filters f... more In this paper, we use the frequency-response masking (FRM) approach to design prototype filters for cosine-modulated filter banks in the nearly-perfect reconstruction case. With such approach, it is possible to design a FRM filter with overall order almost equal to the direct-form FIR design, with only slight changes in the values of the inter-carrier and inter-symbol interferences and the attenuation of the bank filters. The result is an efficient design with reduced number of multipliers for the overall structure.
Proceedings of 1997 IEEE International Symposium on Circuits and Systems. Circuits and Systems in the Information Age ISCAS '97
ABSTRACT A family of continuous-time and switched-current (SI) four-quadrant current-multipliers ... more ABSTRACT A family of continuous-time and switched-current (SI) four-quadrant current-multipliers are proposed, based on a current-gain cell linearly controlled by a current, as well as an SI version of that cell. The use of a subtractor is also proposed, which eliminates current offsets. Based on this property an offset-free four-quadrant multiplier-integrator cell is also presented. The new circuits are meant to implement adaptive filters
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Papers by Paulo S. R. Diniz