• Journal of the Korean Magnetic Resonance Society
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• v.2 no.1
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• pp.59-65
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• 1998

The correction of image artifacts due to misadjustment in tuning of RF coils (tip angle) and in the RF single sideband spectrometer was investigated using phase cycling of the $\pi$/2 and $\pi$ pulses in spin-echo sequences. A general procedure was developed for the systematic design of phase cycles that select desirable coherence transfer pathways. To analyze a phase cycling sequence, changes in the coherence level and phase factor for each RF pulse in the spin-echo cycle must be determined. Four different phase cycling schemes (FIXED, ALTERNATE, FORWARD, REVERSED) to suppress unwanted signal components such as mirror and ghost images were evaluated using two signal acquisitions. When the receiver phase factor is cycled counter-clockwise (REVERSED), these artifacts are completely removed.

• The Journal of the Acoustical Society of Korea
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• v.12 no.1E
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• pp.20-31
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• 1993

Berman과 Feuer의 SLMS(smoothed least mean square)알고리즘의 근사적 분석을 행하여 보다 유용한 분석결과를 얻었다. 수렴범위와 misadjustment에 대한 분석에서는 기존의 알고리즘의 분석결과들과 비교할 수 있는 형태로 얻었을뿐만아니라 여러 변수들이 이 알고리즘의 성능에 미치는 영향을 명확히 알 수 있는 형태로 얻었다. 둘째로 몇몇 서로 유사한 알고리즘들을 비교검토함으로써 서로간의 관계를 밝히고 이 결과들을 해석하였다. 이어서 위의 분석결과들이 유효함을 실험을 통하여 밝혔다. 수렴한계 근처에서 LMS알고리즘보다 안정됨을 보였다. 이들 아고리즘의 비정상특성(nonstationary characteristics)에 대하여서도 살펴보았는데, SLMS알고리즘의 경우 추적능력의 별다른 희생 없이도 가중계수(weight)의 잡음을 줄일 수 있음을 보였다.

• Proceedings of the Korea Multimedia Society Conference
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• 2004.05a
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• pp.179-182
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• 2004

무선통신분야에서 LMS5(Least Mean Square) 알고리즘은 식이 간단하고 계산량이 비교적 적기 때문에 널리 사용되고 있다. 그러나 시간영역에서 처리할 경우 입력신호의 고유치 변동폭이 넓게 분포되어 수렴속도가 저하하는 문제점이 있다. 이를 해결하기 위하여 신호를 FFT(Fast Fourier Trasnform)나 DCT(Discrete Cosine Transform)로 변환하여 신호간의 상관도를 제거함으로써 시간영역에서 LMS알고리즘을 적용할 때 보다 수렴속도를 크게 향강시킬 수 있다. 본 논문에서는 수렴속도 향상을 위해 시간영역의 적응 알고리즘을 직교변환인 고속웨이브렛(wavelet)변환을 이용하여 변환영역에서 수행하며, 짧은 필터계수를 가지는 DWT(Discrete Wavelet Transform)특성에 맞는 Fast running FIR 알고리즘을 이용하여 WTLMS(Wavelet Transform LMS)적응알고리즘을 통신시스템에 적용한다. 적응 알고리즘의 성능향상을 위하여 시간에 따라 적응상수의 크기를 가변시켜 수렴 초기에는 큰 적응상수로 따른 수렴이 가능하도록 하고 점차 적응상수의 크기를 줄여서 misadjustment도 줄이는 방법의 적응 알고리즘을 제안하였다. 제안한 알고리즘을 실제로 적응잡음제거기(adaptive noise canceler)에 적용하여 컴퓨터 시뮬레이션을 하였으며, 각 알고리즘들의 계산량, 수렴속도를 이용하여 각각 비교, 분서하여 그 성능이 우수함을 입증하였다.

• Proceedings of the Korea Multimedia Society Conference
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• 2003.05b
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• pp.446-449
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• 2003

The conventional GSC is inappropriate in real situation when the target signal is present. The steering vector error cancels the target signal and the target signal misadjusts the weight of the adaptive filter. To prevent the target signal cancellation, the robust GSC using the constrained adaptive filters was already proposed. However, the adaptive weight misadjustment is not settled in robust GSC. This Paper proposes a revised robust sidelobe canceller with adaptive compensator. To compensate the influence of target signal, the adaptive compensator is used in cascade. In computer simulation, we show the performance improvement by comparing the robust GSC with the proposed GSC.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.1
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• pp.27-34
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• 2013

This paper deals with the performance anlysis of CCA adaptive equalization algorithm, that is used for reduction of intersymbol interference at the receiving side which occurs in the time dispersive communication channel. Basically, this algorithm is borned for the solving phase unrecovery problem in the CMA equalizer, and the comines the concept of DDA (Decision Directed Algorithm) and RCA (Reduce Constellation Algorithm). The DDA has a stable convergence characteristics in unilevel signal, but not in the number of levels in multilevel signal such as QAM, so it has unstable problem. The RCA does not provide reliable initial convergence. And even after convergence, the equalization noise due to the steady state misadjustment exhibited by it is very high as compared to DDA. For the solving the abovemensioned point, the CCA adaptive eualization alogorithm has borned. In order to performance analysis of CCA algorithm, the recovered signal constellation that is the output of the equalizer, the convergence characteristic by the residual isi and MD (maximum distortion), the SER characteristic are used by computer simulation and it was compared with the DDA, RCA respectively. As a result of simulation, the DDA has superior performance than other algoithm, but it has a convergence unguarantee and unstability in the multilevel signal. In order to solving this problem, the CCA has more good performance than RCA in every performance index.

• Journal of Korea Society of Industrial Information Systems
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• v.4 no.3
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• pp.79-88
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• 1999

One of the most popular algorithm in adaptive signal processing is the least mean square(LMS) algorithm. The majority of these papers examine the LMS algorithm with a constant step size. The choice of the step size reflects a tradeoff between misadjustment and the speed of adaptation. Subsequent works have discussed the issue of optimization of the step size or methods of varying the step size to improve performance. However there is as yet no detailed analysis of a variable step size algorithm that is capable of giving both the speed of adaptation and convergence. In this paper we propose a new variable step size algorithm where the step size adjustment is controlled by square of the prediction error. The simulation results obtained using the new algorithm about noise canceller system and system identification are described. They are compared to the results obtained for other variable step size algorithm. function.

• Journal of The Institute of Information and Telecommunication Facilities Engineering
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• v.5 no.1
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• pp.25-36
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• 2006

This paper discusses the implementation and performance evaluation of a DSP-based digital beamformer using the Texas Instrument TMSC6711 DSP processor for smart antenna applications. Two adaptive beamforming algorithms which served as the brain for the beamformer, the Normalized Least-Mean-Square (NLMS) and the Constant Modulus Algorithms (CMA) were embedded into the processor and evaluated. Result shows that the NLMS-based digital beamformer outperforms the CMA-based digital beamformer: 1)For NLMS algorithm, the antenna steers to the direction of the desired user even at low iteration value and the suppression level towards the interferer increases as the number of iteration increase. For CMA algorithm, the beam radiation pattern slowly steers to the desired user as the number of iteration increased, but at arate slower than NLMS algorithm and the sidelobe level is shown to increases as the number of iteration increase. 2) The NLMS algorithm has faster convergence than CMA algorithm and the error convergence for CMA algorithm sometimes is subject to misadjustment.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.15 no.2
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• pp.99-104
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• 2015

This paper proposes the RMMA (Region based Multiple Modulus Algorithm) algorithm that is possible to improving the performance of MMA adaptive equalization algorithm in order to the reduction of intersymbol interference occurs at the communication channel. In RMMA algorithm, the output constellation of equalizer are divided by 4 different regions in order to get the error signal for adapting the channel characteristic, and the small error signal is obtained by mapping each region to 4-QAM signal. The conversion effect of constant modulus from nonconstant modulus signal was obtained. In this paper, the adaptive equalization performance of proposed RMMA were evaluated comared to the present MMA. As a result of computer simulation, the convergence speed and residual quantity were improved in residual isi and MD. Especially the superiorities of robustness was confirm in SER performance compared to present MMA.

• Journal of the Korean Institute of Telematics and Electronics
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• v.25 no.10
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• pp.1263-1273
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• 1988

An adaptive nulling technique is presented to synthetically overcome the integrated problems associated with the conventional LMS array in the performances of jammer rejection, convergence rate, misadjustment, and reference signal generation. The proposed method is to remove the target signal from the array input and to eliminate the reference signal prior to minimization processing. The algorithm is constrained to the residue noise level in adaptive processor. Analysis shows effectiveness of the algorithm for coherent and/or incoherent interference rejection, wide dynamic range of convergence factor, rapid adaptation rate, and small mean square error. Simulation results confirm the theoretical prediction.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.7A
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• pp.651-658
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• 2002

Although least mean square(LMS) algorithm is known to one of the most popular algorithm in adaptive signal processing because of the simplicity and the small computation, the choice of the step size reflects a tradeoff between the misadjustment and the speed of adaptation. In this paper, we present a new variable step size LMS algorithm, so-called ECLMS(Estimated correlation LMS), using the correlation between reference input and error signal of adaptive filter. The proposed algorithm updates each weight of filter by different step size at same sample time. We applied this algorithm to adaptive multiple-notch filter. Simulation results are presented to compare the performance of the proposed algorithm with the usual LMS algorithm and another variable step algorithm.