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An algorithm that is suitable for the computation of symmetrical components is presented in this paper. The structure of the algorithm consists of four decupled linear modules: the first for adaptive filtering of input signals, the second for frequency estimation, the third for symmetrical components estimation, and the fourth for the amplitudes of the component estimation. Filters are used to minimize the noise effect and to eliminate the presence of harmonics. A simple method for online designing of digital filters of sinusoidal signals is used in this paper. This method uses closed forms for calculation of filter coefficients and requires modest computations. The symmetrical components estimation through adaptive transformation matrix of phase shifters used in this paper makes it possible to get instantaneous symmetrical components independently on the frequency variation with modest-computation requirements. The technique used in this paper provides accurate frequency estimates with the maximum error of 0.002 Hz and symmetrical components estimation with the maximum error of 0.03% for SNR = 60 dB in about 25 ms. The algorithm is not sensitive to power-system frequency changes and to the harmonic distortion of the input signals. To demonstrate the performance of the developed algorithm, computer-simulated data records have been processed. It has been found that the proposed algorithm is suitable for real-time applications. [ABSTRACT FROM AUTHOR]

New algorithm for measurement of mean count rate of pulses from radiation detectors is described. The algorithm is based on the control of a bank of digital IIR filters governed by the state of mean count rate—stationary or transient. The algorithm solves successfully the compromise between the contradictory requirements for low statistical fluctuations in steady state and fast response to sudden changes of mean count rate. The design of the bank of HR filters is carried out by using the Chebyshev approximation of the second kind in order to achieve the high selectivity and high stopband attenuation applying as low as possible order of the constituent filters. All designed filters are of the second order. The static and dynamic characteristics of the new algorithm and the corresponding computational efficiency are tested and compared with those of the traditional ratemeter algorithms. The state of mean count rate, stationary or transient, is determined by a circuit for estimation of the gradient of mean count rate. The numbers of threshold levels within the adaptation logic circuit and their mutual positions have been optimized to achieve the best compromise. [ABSTRACT FROM AUTHOR]

In this paper, efficient architectures for realizing the recursive discrete cosine transform (DCT) and the recursive inverse DCT (IDCT) are proposed. By respectively folding the inputs of the DCT and the outputs of the IDCT, efficient formulations of the DCT and IDCT are derived to construct a transform kernels. The data throughout per transformation is twice that of the existing methods by spending only half of the computational cycles used by the single folding algorithms. To further improve efficiency, the double folding recursive architectures of the DCT and IDCT are developed. The computational cycles of the DCT are half of the single folding method, and the data throughout of the IDCT is twice that of the single folding method. The regular and modular properties of the proposed recursive archietectures are suitable for very scale integration (VLSI) implementation. With high throughout advantage, the proposed structures could be implemented with less power consumption, which could be applied to low rate video in mobile and portable information appliances. [ABSTRACT FROM AUTHOR]

It is known that linear phase FIR digital filters fall into four types considering the symmetry of impulse response. For example, filters of type 2 can be used at the time of partitioning and reconstruction of signals in multirate signal processing, and other types have similar special applications. Moreover, the linear phase FIR digital filters with flat passband and equiripple stopband characteristics are very important filters in practice because they have better time domain characteristics and less ringing than Chebyshev filters with equiripple stopband and passband characteristics. However, conventional design methods are restricted to type 1. In this paper, we therefore propose a method of design of linear phase FIR digital filters with flat passband and equiripple stopband characteristics for all four types. The proposed function is composed of two functions, one with a flat characteristic in the passband and the second with equiripple characteristics in the stopband. The latter function is obtained by using the well-known Remez algorithm only for the stopband. © 2002 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 86(4): 89–95, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.1148 [ABSTRACT FROM AUTHOR]

This paper proposes the parallel construction of the second-order infinite impulse response (IIR) adaptive digital filter (ADF) as an adaptive digital filter for eliminating multiple narrowband signals mixed in the broadband signal. Then the pole-control method for the filter is proposed. In the proposed pole-control, the pole radius and the pole argument are controlled separately by different algorithms. In this study, the pole argument is corrected by estimating the deviation of the central frequency of the signal to be eliminated from the convergent value of the multiplier coefficient that determines the zero. The pole radius is controlled by the method that determines the direction of correction, using the parallel-connected three ADFs with the same pole argument and a slightly different pole radius. It is verified by a simulation that those pole-control methods work effectively. It is demonstrated that the pole radius and the pole argument can be controlled simultaneously. [ABSTRACT FROM AUTHOR]