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A binary logic synthesis approach to the bit-level implementation of generalized rank-order filters

A binary logic synthesis approach to the bit-level implement...

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Q. Gu M.N.S. Swamy Center for Signal Processing and Communications Department of Electrical & Computer Engineering Concordia University Montreal QUE Canada

A binary logic synthesis approach is presented for the bit-level implementation of generalized rank-order filters. It is shown that rank-order filters can be realized directly from the binary weighted signal by using combinational logic circuits. A pipelined architecture based on the algorithm is also introduced. The realization is simple and modular in structure and suitable for VLSI implementation, and no threshold decomposition and reconstruction circuits are needed. Only one type of basic cell is employed. K basic cells connected together will perform K-bit rank-order filtering. The proposed filter has flexibility for various applications. Considering the proposed architecture as a building block, recursive rank-order filters, 2-D rank-order filters, and adaptive rank-order filters can easily be realized.< >

关键词： Logic Nonlinear filters signal processing signal processing algorithms Very large scale integration Adaptive filters signal synthesis Combinational circuits Digital filters Circuit synthesis

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Limit cycles in floating-point implementations of recursive filters-a review

Limit cycles in floating-point implementations of recursive ...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： T.I. Laakso B. Zeng Laboratory of Signal Processing and Computer Technology Helsinki University of Technology Espoo Finland Department of Electrical Engineering University of Toronto Toronto ONT Canada

Results on limit cycle oscillations in floating-point implementations of recursive filters are reviewed. Special attention is paid to structures with a single floating-point nonlinearity in the recursive filter loop, for which efficient criteria are proposed to guarantee the absence of limit cycles. Both unlimited exponent range limit cycles and underflow limit cycles are considered. Also results on wave digital filters and related structures are discussed.< >

关键词： Limit-cycles Quantization Laboratories signal processing Digital filters Feedback loop Information analysis Information filtering Information filters Hardware

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High-speed VLSI pipelined processor design for lossless image data compression 6

High-speed VLSI pipelined processor design for lossless imag...

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6th IEEE Workshop on VLSI signal processing

作者： Fang, Wai-Chi Sheu, Bing J. Radar Science and Engineering Jet Propulsion Laboratory California Institute of Technology PasadenaCA91109-8099 United States Signal and Image Processing Institute Department of Electrical Engineering University of Southern California Los AngelesCA90089-0271 United States

ISBN: (纸本)0780308115

An efficient VLSI pipelined processor design for high-speed lossless compression based on "Rice algorithm" has been developed to meet the increasing strong demands on high-volume/high-speed image data communication and storage. The Rice algorithm is an adaptive lossless coding scheme that provides near-optimal performance over a broad range of data entropies. The Rice algorithm is also an efficiently implementable scheme for VLSI realization. A VLSI pipelined architecture was developed to allow compact implementation of a single-chip VLSI compressor. This lossless compressor is named PSI14,K+ since it implements an advanced version of the Rice's universal noiseless coding method called PSI14,K+. The chip layout was generated for a 1.0- micron CMOS technology. It occupies a compact chip area of 5.1 × 5.3 mm2, with 49,000 transistors, 57 input/output pads, and 6 power/ground pads. The total power dissipation is 0.4 watts at the 40 MHz system clock with a 50% switching duty cycle. This compressor chip is mounted in a 68-pin pin-grid-array package. It can operate up to 40 Mpixels/sec. The potential applications of the proposed lossless compressor include database management systems, scientific instruments, CAE workstations, desktop computing machines, and the data systems that require highspeed compression without fidelity loss. © 1992 IEEE.

关键词： Compressors

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Optimal multi-stage stack filtering for image restoration

Optimal multi-stage stack filtering for image restoration

引用

1992 IEEE International Symposium on Circuits and Systems, ISCAS 1992

作者： Zeng, Bing Nevvo, Yrjö Venetsanopoulos, Anastasios N. Department of Electrical Engineering University of Toronto TorontoONM5S 1A4 Canada Signal Processing Laboratory Tampere University of Technology P.O. Box 553 TampereSF-33101 Finland

ISBN: (纸本)0780305930

In this paper, the optimal multi-stage (MS) stack filtering under the mean absolute error (MAE) criterion is discussed, with image restoration in mind. By using the recently developed techniques [5,6], the optimal MS stack filters can be designed efficiently. Compared to the single-stage (SS) stack filtering, an MS filter provides an additional design flexibility by choosing the window masks at different stages in order to provide a good compromise between the noise removing and the image detail preserving. To achieve a comparable performance, MS filters can select considerably smaller window masks, whereas SS stack filters have to use larger masks. Due to the relationship between the filtering complexity/memory requirement and the window size, this suggests that the MS stack filtering is much faster and requires much less memories than the SS stack filtering. © 1992 IEEE.

关键词： Image reconstruction

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Multiple branch FIR filters for sampling rate conversion

Multiple branch FIR filters for sampling rate conversion

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： T. Saramaki S.K. Mitra Signal Processing Laboratory Tampere University of Technology Tampere Finland Department of Electrical and Computer Engineering University of California Santa Barbara CA USA

A class of linear-phase finite impulse response (FIR) digital filters for sampling rate conversion is introduced. These filters consist of two cascaded parts. The first part is a parallel connection of two or three branches with transfer function of the form A/sub k/(z)B/sub k/) (z/sup D/), where D is the sampling rate conversion ratio. The second part has a transfer function of the form C(z/sup D/). The advantage of these designs is that they provide a significant reduction in the multiplication and addition rates as well as in the number of required multipliers over the equivalent conventional direct-form designs also in cases where D is a prime number. In these cases, the sampling rate conversion cannot be performed in several stages and many of the other existing designs lose their advantages. Special multiplier-free solutions are generated for implementing the filters A/sub k/(z), resulting in dramatic savings. Several examples are included showing the efficiency of the proposed filters compared to other existing designs.< >

关键词： Finite impulse response filter Sampling methods Delay Passband Niobium USA Councils Baseband Shape Transfer functions Interpolation

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Impact of Noise and Target Fluctuation on the Performance of Frank Polyphase Coded Radar signals

Impact of Noise and Target Fluctuation on the Performance of...

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Southeastern Symposium on System Theory

作者： A.K. Ojha D.B. Koch Communications Information and Signal Processing Group Department of Electrical and Computer Engineering University of Tennessee Knoxville TN USA

Power-limited radar systems generally employ some form of pulse compression or waveform coding to improve range resolution. Frank polyphase coded waveforms have been used in the past for pulse compression in high resolution radars. This paper evaluates the impact of Swerling I and Swerling II target scintillation models in the presence of additive white Gaussian noise (AWGN) on the resolution properties of Frank coded waveforms. The results presented in this paper bear out the interesting fact that the use of an appropriate model is highly desirable to avoid misleading results.

关键词： Fluctuations Autocorrelation Pulse compression methods Space vector pulse width modulation Radar signal processing signal processing signal resolution Additive white noise Radar detection Matched filters

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A theoretical framework for a class of frequency estimation algorithms

A theoretical framework for a class of frequency estimation ...

引用

IEEE Workshop on Statistical signal and Array processing

作者： R.M. Todd J.R. Cruz Communications and Signal Processing Laboratory (CMP Lab) School of Electrical Engineering and Computer Science University of Oklahama Norman OK USA

This paper discusses a general class of algorithms for estimating the frequencies of a set of complex exponentials, and presents a corrected proof of the validity of the algorithms when applied to either real or complex data. The linear-prediction least-squares algorithms, involve the formulation of the estimation problem in terms of finding the roots of a polynomial in C(x) (the vector space of polynomials over the complex numbers C) that has minimum norm with respect to some inner product defined over C(x).< >

关键词： Frequency estimation Polynomials signal processing algorithms Kernel Laboratories computer science Vectors Symmetric matrices

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Complex system identification methods for fast echo canceler initialization

Complex system identification methods for fast echo canceler...

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International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： S.A. Ahmed J.R. Cruz Communications and Signal Processing Laboratory (CMP Lab) School of Electrical Engineering and Computer Science University of Oklahama Norman OK USA

The authors propose a new recursive version of a technique proposed by G. Long and F. Ling (1990) for the initialization of a data-driven echo canceler (DDEC). They prove that the Long and Ling algorithm yields a least-squares solution, and then a new technique is presented which is comparable to the recursive-least-squares (RLS) algorithm. However, the use of a unique training sequence reduces the complexity of the RLS algorithm to that of the least-mean-square (LMS) algorithm. An analysis of the covariance of the estimated weight vector is presented, and simulation results show a remarkable improvement in both convergence speed and steady-state error.< >

关键词： System identification signal processing algorithms Least squares approximation Resonance light scattering Least squares methods Steady-state Computational efficiency signal processing Laboratories computer science

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Design of centrally symmetric 2-D FIR fan filters using the McClellan transform

Design of centrally symmetric 2-D FIR fan filters using the ...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： W.P. Zhu Z.Y. He M.O. Ahmad M.N.S. Swamy Center for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada Division of Digital Signal Processing Department of Radio Engineering South-East University Nanjing Jiangsu China

The authors present a novel design method for 2-D zero-phase finite impulse response (FIR) fan filters with centrally symmetric frequency responses, via the generalized McClellan transform. The formulas are derived from the prescribed frequency response specifications of the fan filter and are optimal in the mean square sense. The efficiency of the method is shown through design examples.< >

关键词： Finite impulse response filter Frequency response IIR filters Lubricating oils Digital signal processing Design engineering Design methodology Algorithm design and analysis Digital filters Frequency conversion

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2-D system identification from bispectrum samples

2-D system identification from bispectrum samples

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International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： A.D. Salvia H.M. Valenzuela Spatial and Temporal Signal Processing Center Department of Electrical and Computer Engineering Pennsylvania State University University Park USA

The problem of modeling 2-D system transfer functions given only bispectrum samples of the noisy output is considered. This identification procedure is a three-step process. First, the output bispectrum samples are calculated along a 2-D slice in the 4-D bispectrum frequency region. Next, these samples are used to obtain a rational model of the bispectrum slice via a 2-D divided difference approach. Finally, two methods for finding the underlying system transfer function parameters from this rational model are developed. This system identification procedure is general in that it does not impose causality, recursibility, or minimum phase restrictions on the system to be identified.< >

关键词： System identification Transfer functions Fourier transforms Multidimensional signal processing Frequency domain analysis Multidimensional systems Frequency conversion Additive noise Gaussian noise Rail to rail outputs

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