• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.3
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• pp.55-61
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• 2013

This paper compare the performance of S-MMA(Sliced-MultiModulus Algorithm) adaptive equalization algorithm by effect of slice weighting value for the minimization of the distortion and noise in the communication channel.. In the traditional MMA algorithm, the output signal of equalizer and the dispersion constant of transmitting signal is used for calculating the equalizer coefficient, but in S-MMA, the output of equalizer and dispersion constant and the considering the output of decision device by the power of slice constant are used in order to simultaneously compensate the distortion of amplitude and phase distortion. It is confirmed by computer simulation that the slice weighting value affects the performance of adaptive equalization algorithm. The performance index includes the output signal constellation, the residual isi and maximum distortion and MSE that is for the convergence characteristics, the SER according to the signal and noise power ratio at the channel is used. As a result of simulation, the residual isi, maximum distortion and MSE performances are better in the small weighting values. But in SER performance is better in the large weighting values.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.7
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• pp.1321-1327
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• 2009

In this paper, as the mobile communication service is widely used and the demand for wireless repeaters is rapidly increasing because of the easiness of extending service areas. But a wireless repeater has a problem the oscillation due to feedback signal. We proposed a new hybrid interference canceller using the adaptive filter with CMA(Constant Modulus Algorithm)-Grouped LMS(Least Mean Square) algorithm in the adaptive interference canceller. The proposed interference canceller has better channel adaptive performance and a lower MSE(Mean Square Error) than conventional structure because it uses the cancellation method of Grouped LMS algorithm. The proposed detector uses the LMS algorithms with two different step size to reduce mean square error and to obtain fast convergence. This structure reduces the number of iterations for the same MSE performance and hardware complexity compared to conventional nonlinear interference canceller.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.19 no.5
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• pp.115-121
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• 2019

In this paper, we propose MSAG-SCS-MMA-I adaptive blind equalization with double step size with very small residual ISI and MSE at steady-state while significantly improving the convergence speed of the traditional SCS-MMA-I algorithm in 256-QAM system. And we evaluate the equalization performance for this algorithm. Different step sizes according to the absolute value of decision-directed error instead of a fixed step-size are applied to the tap update equation of MSAG-SCS-MMA-I, which is controlled by binary flags of '1' or '0' obtained from SCS-MMA-I and decision-directed algorithms. This makes for excellent equalization performance. As a result of computer simulation, we confirmed that the proposed algorithm has more better performance than the MMA, SCS-MMA-I, and MSAG-SCS-MMA-I algorithms in terms of the performance index such as residual ISI, MSE, and MD.

• Journal of information and communication convergence engineering
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• v.8 no.2
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• pp.155-158
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• 2010

In this article, we proposed a new interference canceller using the adaptive algorithm. We designed constant modulus algorithm-correlation normailized least mean square (CMA-CNLMS) for wireless system. This structure is normalized LMS algorithm using correlation between the desired and input signal for cancelling the interference signals in the wideband code division multiple access (WCDMA) system. We showed that the proposed algorithm could improve the Mean Square Error (MSE) performance of LMS algorithm. MATLAB (Matrix Laboratory) is employed to analyze the proposed algorithm and to compare it with the experimental results. The MSE value of the LMS with mu=0.0001 was measured as - 12.5 dB, and that of the proposed algorithm was -19.5 dB which showed an improvement of 7dB.

• Journal of information and communication convergence engineering
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• v.9 no.6
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• pp.671-675
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• 2011

DVC (Distributed Video Coding) and FRUC (Frame Rate Up Conversion) techniques need to have an efficient motion compensated frame interpolation algorithms. Conventional works of these applications have mainly focused on the performance improvement of overall system. But, in some applications, it is necessary to evaluate how well the MCI (Motion Compensated Interpolation) frame matches the original frame. For this aim, this paper deals with the modeling methods for evaluating the block-based matching cost. First, several matching criteria, which have already been dealt with the motion compensated frame interpolation, are introduced and then combined to make estimate models for the size of MSE (Mean Square Error) noise of the MCI frame to original one. Through computer simulations, it is shown that the block-based matching criteria are evaluated and the proposed model can be effectively used for estimating the MSE noise.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.193-196
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• 1999

본 논문은 음향 임펄스를 모델링하는데 코사인 변조된 필터 뱅크와 Decimation을 이용하여 수렴 속도를 개선하는 방법을 제안하고 이를 잡음제거에 응용하였다. 제안된 구조는 입력신호를 필터뱅크를 이용하여 각 서브밴드로 분할한 후 필터 입력신호의 고유벡터의 최대값과 최소값의 비를 줄이고 필터의 탭수를 줄이기 위해서 decimation을 행한다. 그리고 서브밴드대역의 샘플링 주파수를 낮추어 신호 스펙트럼을 확장시켜 이를 적응필터에 입력하여 수렴속도를 향상시켰다. 실험 결과, Colored잡음의 경우 LMS 알고리즘보다 제안된 방법이 MSE(Mean Square Error)는 좋지는 않았다. 실제 음향시스템의 모델링에는 거의 같은 MSE을 갖으며, 수렴 속도에는 모두 빠른 성능을 보였으며, 이를 음질향상에 적용하여 향상된 음질을 얻을 수 있었다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.8A
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• pp.760-767
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• 2010

For complex channel blind equalization, this study presents the performance and characteristics of two complex blind information theoretic learning algorithms (ITL) which are based on minimization of Euclidian distance (ED) between probability density functions compared to constant modulus algorithm which is based on mean squared error (MSE) criterion. The complex-valued ED algorithm employing constant modulus error and the complex-valued ED algorithm using a self-generated symbol set are analyzed to have the fact that the cost function of the latter forces the output signal to have correct symbol values and compensate amplitude and phase distortion simultaneously without any phase compensation process. Simulation results through MSE convergence and constellation comparison for severely distorted complex channels show significantly enhanced performance of symbol-point concentration with no phase rotation.

• The Journal of the Acoustical Society of Korea
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• v.14 no.5
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• pp.20-28
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• 1995

This paper studied adaptive blind equalizer which belong to Bussgang type. It is well known that blind equalizer performs equalization without using a training sequence. Especially, this paper concentrated on real time processing of them. The channel characteristic was obtained from measurements taken in a real urban multipath environment. A T/2 fractionally-spaced equalizer was used at the receiving end. Our computer simulations demonstrated that Stop and Go, Benveniste-Goursat, and optimal Bussgang algorithms have relatively low MSE property. CMA shows faster convergence property than any other of Bussgang type algorithm.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.478-481
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• 1999

In this paper, we could analyze and compare with the generalized Recurrent neural networks and the Recurrent neural networks applying the fuzzy. The total system is digitalized in order to be filtering the image, and the fuzzy is applied to the generalized Recurrent in order to be fast the operation speed. So the fuzzified Recurrent neural networks are completely removed to the included noise in the image, and could converge on a certain value as controlling the weight and iteration frequency corresponding to the desired target value. Also, that values are compared and analysed using MSE and PSNR. When applying to the image which is included to the noise in the generalized Recurrent and the Recurrent applying the fuzzy, the Recurrent applying the fuzzy is shown the superiority at the noise and the fixed convergence part through MSE and PSNR in the computer simulations.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.34S no.9
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• pp.23-30
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• 1997

In this paper, a decision feedback recurrent neural equalization (DFRNE) scheme is proposed for adaptive equalization problems. The proposed equalizer models a nonlinear infinite impulse response (IIR) filter. The modified Real-Time recurrent Learning Algorithm (RTRL) is used to train the DFRNE. The DFRNE is applied to both linear channels with only intersymbol interference and nonlinear channels for digital video cassette recording (DVCR) system. And the performance of the DFRNE is compared to those of the conventional equalizaion schemes, such as a linear equalizer, a decision feedback equalizer, and neural equalizers based on multi-layer perceptron (MLP), in view of both bit error rate performance and mean squared error (MSE) convergence. It is shown that the DFRNE with a reasonable size not only gives improvement of compensating for the channel introduced distortions, but also makes the MSE converge fast and stable.