• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.1
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• pp.22-33
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• 2005

In this paper, an efficient signal interference control technique to improve the high convergence speed of LMS algorithms is introduced in the adaptive transversal filter of DS/SS. The convergence characteristics of the proposed algorithm, whose coefficients are multiply adapted in a symbol time period by recycling the received data, is analyzed to prove theoretically the improvement of high convergence speed. According as the step-size parameter ${\mu}$ is increased, the rate of convergence of the algorithm is controlled. Also, an increase in the stop-size parameter ${\mu}$ has the effect of reducing the variation in the experimentally computed learning curve. Increasing the eigenvalue spread has the effect of controlling which is downed the rate of convergence of the adaptive equalizer. Increasing the steady-state value of the average squared error, proposed algorithm also demonstrate the superiority of signal interference control to the filter algorithm increasing convergence speed by (B+1) times due to the data-recycling LMS technique.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.1189-1191
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• 2005

This paper introduces the least-squares order-recursive lattice (LSORL) Laguerre smoother that has order-recursive smoothing structure based on the Laguerre signal representation. The LSORL Laguerre smoother gives excellent performance for a channel equalization problem with smaller order of tap-weights than its counterpart algorithm based on the transversal filter structure. Simulation results show that the LSORL Laguerre smoother gives better performance than the LSORL transversal smoother.

• 제어로봇시스템학회:학술대회논문집
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• 2000.10a
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• pp.185-185
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• 2000

This paper designs an interference mitigation filter for GPS receiver using a time-domain signal processing techniques (Adaptive Transversal filter). ATF is designed using verilog and simulated using COSSAP for evaluating its performance.

• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.1577-1580
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• 2004

This paper proposes a design and implementation of transversal adaptive digital filter using LMS (Least Mean Squares) adaptive algorithm. The filter structure is based on Distributed Arithmetic (DA) which is able to calculate the inner product by shifting and accumulating of partial products and storing in look-up table, also the desired adaptive digital filter will be multiplierless filter. In addition, the hardware implementation uses VHDL (Very high speed integrated circuit Hardware Description Language) and synthesis using FLEX10K Altera FPGA (Field Programmable Gate Array) as target technology and uses Leonardo Spectrum and MAX+plusII program for overall development. The results of this design are shown that the speed performance and used area of FPGA. The experimental results are presented to demonstrate the feasibility of the desired adaptive digital filter.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.62 no.1
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• pp.133-139
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• 2013

Convex combination of two adaptive filters is an efficient method to improve adaptive filter performances. In this paper, a subband convex combination method of two adaptive filters for fast convergence rate in the transient state and low steady state error is presented. The cost function of mixing parameter for a subband convex combination is defined, and from this, the coefficient update equation is derived. Steady state analysis is used to prove the stability of the subband convex combination. Some simulation examples in system identification scenario show the validity of the subband convex combination schemes.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1983.10a
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• pp.85-87
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• 1983

In this paper, a improved whitening filter for narrowband interference rejetion in PN spread spectrum communication systems is proposed. This whitening filter has the feedback from the output to the input of adaptive transversal filter and is superior to conventional LMS whitening filter in respect to SNR and power spectrum since the calculation of tap weight is based on maximum likelihood estimation

• The Journal of the Acoustical Society of Korea
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• v.15 no.2
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• pp.79-84
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• 1996

In this paper, adaptive correlator model using backpropagation neural network based on complex multilayer perceptron is presented for suppressing interference of narrow-band of direct sequence spread spectrum receiver in CDMA mobile communication systems. Recursive least square backpropagation algorithm with backpropagation error is used for fast convergence and better performance in adaptive correlator scheme. According to signal noise ratio and transmission power ratio, computer simulation results show that bit error ratio of adaptive correlator uswing backpropagation neural network improved than that of adaptive transversal filter of direct sequence spread spectrum considering of co-channel and narrow-band interference. Bit error ratio of adaptive correlator using backpropagation neural network is reduced about $10^{-1}$ than that of adaptive transversal filter where interference versus signal ratio is 5 dB.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.9A
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• pp.830-837
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• 2005

ISI is caused by delay spread in the multipath channel environment. There are two kinds of channel equalizer: Linear and Non-Linear type according to the structures. In this paper, we propose an improved adaptive linear equalizer to mitigate ISI. The proposed adaptive equalizer is constructed by using asymmetrical Dsmvenu filter based on USE sub-optimal receiver. Asymmetrical structure of the transversal filter is realized by moving the main tap position from center to side. If this structure is used, we can divide ISI to precusor and postcusor. As a result the proposed equalizer has a larger extended compensation range than conventional adaptive linear equalizer. In computer simulation, we compare the bit error rate performance of the proposed linear equalizer with the conventional one on the S-V channel which is modeled for WB systems.

• Proceedings of the Acoustical Society of Korea Conference
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• 1984.12a
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• pp.70-72
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• 1984

본 논문에서는 음성신호에 섞인 잡음을 효과적으로 제거하기 위해 Two-sided Transversal Filter를 이용하는 방법을 제시하였다. 이 방법은 기존의 방법과는 달리 신호의 단기자기상관관계(short-term autocorrelation)을 충분히 이용하기 위해 적응 필터의 입력으로, 지연된 신호뿐만 아니라 미래의 신호도 이용하도록 하였다. 이 결과 음성신호에 섞인 잡음(백색잡음, 유색잡음)을 아주 효과적으로 제거함을 알 수 있다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.6
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• pp.881-888
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• 1993

A new stabilizing algorithm for the fast transversal filter (FTF) which is known to exhibit a very unstable behavior, due basically to the accumulation in two different ways, we define a emasure counting these at each step of the algorithm in order to reduce the roundoff error. It has been shown that required amount of computation for the proposed algorithm is linearly increased as the order of equalizer and that stability is improved in comparison with the other existing algorithm. It has been shown that the proposed algorithm exhibits a good performance using computer simulation of adaptive equalization for multipath fading channel in urban mobile radio environment.