• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.1
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• pp.150-158
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• 2010

Filtering is necessary for the SRC(sample rate converter), that is used to change the sampling rate of a digital signal. The larger the conversion ratio of the sampling rate becomes, the more signal processing is needed for the filter, which means more complexity on realization. Thus it is important to reduce the amount of signal processing for the case of substantial conversion ratios. In this paper it is presented an efficient design method of a multistage FIR(finite impulse response) filter, with which the rate conversion occurs in stages rather than in one step. In this method, filter searching is performed exhaustively over all possible factorization of the conversion ratio, and also the filter complexity is measured based on direct realization rather than on estimation. It has been shown a designed multistage filter to have a less number of multiplications for filtering operation in comparison with a conventionally designed one. It has also been found that by allowing some variations of the filter architecture such as a halfband filter or a filter with multiple transition bands, the number of multiplications can be reduced further.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.1
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• pp.35-44
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• 2004

We Propose the methods to design the finite impulse response (FIR) and the infinite impulse response (IIR) lattice filters using Schur algorithm through the spectral factorization of the covariance matrix by circulant matrix factorization (CMF). Circulant matrix factorization is also very powerful tool used for spectral factorization of the covariance polynomial in matrix domain to obtain the minimum phase polynomial without the polynomial root finding problem. Schur algorithm is the method for a fast Cholesky factorization of Toeplitz matrix, which easily determines the lattice filter parameters. Examples for the case of the FIR filter and for the case of the In filter are included, and performance of our method check by comparing of our method and another methods (polynomial root finding and cepstral deconvolution).

• Journal of the Korean Institute of Telematics and Electronics
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• v.22 no.3
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• pp.16-22
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• 1985

In this paper a new algorithm named modified delta modulation (MDM) for encoding filter coefficients is proposed. And this paper presents the designing method of multiplier less FIR filters rosin영 Proposed MDM a19orithm. In the delta modulation (DM) system the quantiaation levels consist of two levels $\pm$1, but in newly proposed MDM algorithm quantization levels are extended to many levels 0, $\pm$2$^n$, n=0, 1, 2... It is recognized by the result of computer simulations that frequency response of multi-plierless FIR filters designed by MDM algorithm is relatively good. And comparing with con-ventional FIR filters on the number of hardware devices, this filter needs a little increased memory, but regardless of filter order it needs only one multiplier which is used for signal scaling.

• The Transactions of the Korean Institute of Electrical Engineers
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• v.32 no.12
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• pp.427-434
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• 1983

Singular Value Decomposition of 2-D Finite Impulse Response was used to design a variable cutoff 2-D FIR filters. Basically this approach allows to use 1-D techniques while dealing with 2-D filters. In this paper, our aim is to provide away of easily tuning 2-D FIR filters. For example, one would like to adjust interactively the cutoff frequency of a 2-D Circular Symmetric low-pass filter on an image processing system. The decomposition of 2-D FIR filters along with techniques available for tuning 1-D FIR filters allows such adjustment. Several design examples are shown.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1983.10a
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• pp.61-64
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• 1983

It is shown that the new window function of Exponential type can be derived from Cheby-chev Identity for the design of FIR filter, and the various kinds of windows which have different characterristics can also be made by changing the coefficients of them oasily

• The Journal of the Korea institute of electronic communication sciences
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• v.11 no.7
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• pp.659-664
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• 2016

In this paper, we suggest operational trans-conductance amplifier(OTA) for current-mode FIR filter that can be used in a digital circuit system requiring high operating frequency and low power consumption. The current-mode signal processing is one of the very innovative design method for a low power consumption system with high operating frequency because it shows a constant power regardless of frequency. From the simulation result using 0.35um CMOS process, when Vdd is 2V, it is confirmed that the proposed circuit showed the dynamic range of the about 1V, about 50% of supply voltage and output current swing of about 0~200uA. Also, the power consumption was evaluated with about 21uW and the active size for an integration was measured with $71um{\times}166um$.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.10A
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• pp.1598-1603
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• 2000

In this paper, an efficient pulse shaping filter architecture for SSB/BPSK-DS/CDMA is proposed. The filter satisfies the specifications in IS-95. The proposed architecture is based on polyphase decomposition and look-up table method. By exploiting the linear phase property of the decomposed filter coefficients, the chip area required for look-up table can be reduced by half compared with the conventional methods. By Synopsys simulations, it is shown that the use of the proposed method can result in reduction in the number of gates by 40%.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.10C
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• pp.1008-1014
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• 2002

This paper is concerned about multiplierless interpolation FIR filters. In this paper, we propose a filter that performs T tap 1:N interpolation FIR filter operation with B-bit inputs without using multipliers. This is done by applying a method which converts a 2s complement multi-bits input to multiple single-bit inputs and a lookup table minimization method which reduces the size of lookup tables by use of the symmetry of filter coefficients and the symmetry of each lookup table. Two FIR filters are implemented using the methods proposed in this paper. Each of the two filters respectively follows the two design parameters in the specification of IMT-2000. Those two FIR filters have an advantage that the number of required gates is reduced up to 70% comparing to that of a conventional transversal FIR filter.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.2532-2537
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• 2005

FIR(Finite Impulse Response) filter is well known to be ideal for the finite time state-space model, but it requires much computation due to its inherent non-recursive structure especially when the measurement interval grows to a large extent. And often a fixed-lag smoother based on the finite time interval is needed to monitor the soundness of the system model and the measurement model, but the computation burden of FIR-type smoother imposes much restriction of its usage for real-time application. Conventional recursive forms of FIR estimator[1]-[4] could not be used for real time applications, since they are numerically unstable in their recursive equations. To cope with this problem, we suggest a stable recursive form FIR estimator(SRFIR) and its usefulness is demonstrated for designing the real-time fixed-lag smoother on the finite time window through an example of detection of rate bias in the anti-aircraft gun fire control system.

• 제어로봇시스템학회:학술대회논문집
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• 1996.10b
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• pp.445-448
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• 1996

For a lightly damped system, IIR based filter can have better performance than FIR filter as an adaptive filter in ANC algorithm. IIR based filter which has an infinite impulse response can model a lightly damped acoustic system with a small number of weights compared that of FIR filter and has no nonlinearity and instability problems on weight updating process which are associated to the conventional IIR filters. There are, however, some drawbacks such as design parameters to be determined earlier to get better performance and much increased computational power especially in the presence of error path. In this study, base filter parameters are determined in systematic manner, with the knowledge of the nominal impulse response of the system which should be identified, by Prony's method. Three methods reducing the computational load are proposed and their performance and application limits are discussed. Simulation and experimental results demonstrate the feasibility of the proposed method.