• Journal of the Institute of Convergence Signal Processing
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• v.8 no.1
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• pp.73-77
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• 2007

In this paper, we propose a VLSI architecture of an LMS filter based on a Cyclo-static scheduler for fast computation of LMS filteing algorithm which is widely used in adptive filtering area. This process is composed of two steps: scheduling and circuit synthesis. The scheduling step accepts a fully specified flow graph(FSFG) as an input, and generates an optimal Cyclo-static schedule in the sense of the sampling rate, the number of processors, and the input-output delay. Then the generated schedule is transformed so that the number of communication edges between the processors. The circuit synthesis part translates the modified schedule into a complete circuit diagram by performing resource allocations. The VLSI layout generation can be performed easily by an existing silicon compiler.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.37 no.6
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• pp.70-75
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• 2000

This paper examines the performance of adaptive filtering algorithms in relation to the asymptotic relative efficiency (ARE) of estimators. The adaptive filtering algorithms are Hybrid II and modified zero forcing (MZF) algorithms. The Hybrid II and MZF algorithms are simplified forms of the LMS algorithm, which use the polarity of the input signal, and polarities of the error and input signals, respectively. The ARE of estimators for each algorithm is analyzed under the condition of the same convergence speed. Computer simulations for adaptive equalization are performed to check the validity of the theory. The explicit expressions for the ARE values of the Hybrid II and MZF algorithms are derived, and its results have similar values to the results of computer simulation. It also revealed that the ARE values depend on the correlation coefficients between input signal and error signal.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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• pp.201-204
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• 1998

This paper study on blind adaptive interference suppression algorithm without training sequence to solve Near-Far problem due to multi access interference. And the performance of each algorithm in the presence of the multipath fading channels over DS-CDMA is evaluated. Simulation results showed that Modified LMS-CMA algorithm has a higher capacity than MOE in SIR/SNR.

• The Journal of the Acoustical Society of Korea
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• v.34 no.2
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• pp.130-137
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• 2015

In this paper, we propose a new Modified Filtered-x Affine Projection Sign Algorithm(MFxAPSA) to improve the convergence speed of the conventional MFxAPSA which has been proposed for active control of impulsive noise. Under the impulsive noise environment, the adaptive algorithms based on the second order moment such as the Filtered-x Least Mean Square(FxLMS) show slow convergence speed or diverge because the noise source tends to have infinite variance. The MFxAPSA is the algorithm derived by applying the Affine Projection Sign Algorithm(APSA) to active noise control. The APSA has an advantage that it does not need the calculation for the inverse matrix, so it may be suitable for the active noise control that requires low computational burden. The proposed MFxAPSA also has APSA's advantage and furthermore, better performance than the conventional MFxAPSA. We carried out a performance comparison of the proposed MFxAPSA with the conventional MFxAPSA. It is shown that the proposed MFxAPSA has the faster convergence speed than the conventional MFxAPSA.

• Journal of the Korean Institute of Intelligent Systems
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• v.6 no.3
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• pp.25-36
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• 1996

Conventional radial-basis-function networks and fuzzy systems have serious problems in dealing with the non1inea:r approximations on high-dimensional spaces due to the explosive increase of the number of hidden nodes or fuzzy IF-THEN rules. In order to avoid such problems, this paper proposes a tree-structured intelligent system in which semi-local basis functions form its basic elements, and develops a training algorithm for the proposed system based on the modified genetic algorithm and LMS rule. Theoretical analysis is performed on the approximation capability of the proposed system, together with experimental studies which demonstrate the effectiveness of the developed methodology.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.1 no.2
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• pp.199-207
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• 1997

The Maximum Likelihood Sequence Estimation scheme is modified to improve the error performance of the correlative coding in the Tamed FM. To remove intersymbol interference, the Decision Feedback Equalization scheme with the delayed LMS algorithm and the Viterbi algorithm(10-symbol delay) in the delayed adaptive equalization are proposed for the performance of decision-directed adaptive equalization under the High Frequency channels, and the condition of convergence is analyzed.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.10
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• pp.2339-2347
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• 1997

The performance of the adaptive noise canceller using LMS algorithm is degraded by the gradient noise due to target speech signals. An adaptive noise canceller with speech detector was proposed to reduce this performande degradation. The speech detector utilized the adaptive prediction-error filter adapted by the NLMS algorithm. This paper discusses to enhance the performance of the adaptive noise canceller forthecorlored noise. The affine projection algorithm, which is known as faster than NLMS algorithm for correlated signals, is used to adapt the adaptive filter and the adaptive prediction error filter. When the voice signals are detected by the speech detector, coefficients of adaptive filter are adapted by the sign-error afine projection algorithm which is modified to reduce the miaslignment of adaptive filter coefficients. Otherwirse, they are adapted by affine projection algorithm. To obtain better performance, the proper step size of sign-error affine projection algorithm is discussed. As resutls of computer simulation, it is shown that the performance of the proposed ANC is better than that of conventional one.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.3
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• pp.231-238
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• 2004

This paper presents the high speed noise reduction processing system using the modified discrete fourier transform(MDFT) on the frequency domain. The proposed filter uses the linear prediction coefficients of the adaptive line enhance(ALE) method based on the Sign algorithm The signals with a random noise tracking performance are examined through computer simulations. It is confirmed that the fast adaptive digital filter is realized by the high speed adaptive noise reduction(HANR) algorithm with rapid convergence on the frequency domain(FD).

• Proceedings of the KIEE Conference
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• 1999.11c
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• pp.760-762
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• 1999

본 논문에서는 기존의 적응필터인 LMS(Least Mean Square)와 RLS(Recursive Least Square)의 수렴속도의 향상과 안정성을 개선하기 위한 방안을 제안하였다. 제안된 알고리즘은 기존의 시간영역 LMS 알고리즘보다 상당히 빠른 수렴속도를 보일 수 있도록 설계하였다. RLS 알고리즘는 역행렬연산으로 인한 연산량이 많고 자기상관행렬이 positive definite 특성을 잃어버릴 경우 시스템이 수치적으로 불안정하게 되어 발산하는 단점이 있다. 이런한 단점을 보완하기 위해 제안된 알고리즘을 사용하였다. 기존의 알고리즘은 전력 정규화 과정에서 입력신호의 변환이 백색화가 완전히 이루어지지 않게 되어 자기상관행렬이 순수한 대각행렬이 되지 않는 단점을 지니고 있으나, 본 연구에서는 이러한 대각화 과정에서 좀더 많은 정보를 포함하도록 설계하였다. 아울러 제안된 알고리즘을 적응 등화기에 적용하여 수렴속도가 개선됨을 검증하였다.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.33B no.6
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• pp.176-185
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• 1996

In this paper, we present the modified CMA (constant modulus algorithm) and LMS (least mean square) algorithms for digiCipher system with reduced hardware cost, in which the pipelined architecture is employed. They yield the performance comparable to that using floating-point operations. We show the effecstiveness of the proposed architecture through the implementation results using VHDL.