• Journal of Digital Contents Society
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• v.5 no.4
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• pp.295-299
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• 2004

In this paper, we compare and analyze the LMS ard RLS algorithm of IEEE802.15.3(HDR-WPAN) system. The LMS algorithm have two merits that easily embody and not complex, but convergence speed is slow. The RLS algorithm have fast convergence speed, but very complex. When equalization using LMS algorithm, it can achieve adaptive equalization after 250 sample in fading environment, but case of RLS algorithm can achieve adaptive equalization after just 50 sampls. The computer simulation proved that adaptive equalizer to fast equalization and stability of HDR-WPAN system is more effective using RLS algorithm then LMS algorithm.

• The Transactions of the Korean Institute of Electrical Engineers P
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• v.57 no.1
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• pp.14-21
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• 2008

In this paper, we propose an adaptive contrast enhancement method using dynamic range segmentation. Histogram Equalization (HE) method is widely used for contrast enhancement. However, histogram equalization method is not suitable for commercial display because it may cause undesirable artifacts due to the significant change in brightness. The proposed algorithm segments the dynamic range of the histogram and redistributes the pixel intensities by the segment area ratio. The proposed method may cause over compressed effect when intensity distribution of an original image is concentrated in specific narrow region. In order to overcome this problem, we introduce an adaptive scale factor. The experimental results show that the proposed algorithm suppresses the significant change in brightness and provides wide histogram distribution compared with histogram equalization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.4C
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• pp.384-391
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• 2003

Blind channel identification and equalization attempt to identify the communication channel and to remove the inter-symbol interference caused by a communication channel without using any known trainning sequences. In this paper, we propose a blind adaptive channel identification and equalization algorithm with phase offset compensation for single-input multiple-output (SIMO) channel. It is based on the one-step forward multichannel linear prediction error method and can be implemented by an RLS algorithm. Phase offset problem, we use a blind adaptive algorithm called the constant modulus derotator (CMD) algorithm based on condtant modulus algorithm (CMA). Moreover, unlike many known subspace (SS) methods or cross relation (CR) methods, our proposed algorithms do not require channel order estimation. Therefore, our algorithms are robust to channel order mismatch.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.3C
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• pp.141-146
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• 2005

The authors investigate the convergence speed problem of nonlinear adaptive equalization. Convergence constraints and time constant of radial basis function network using stochastic gradient (RBF-SG) algorithm is analyzed and a method of making time constant independent of hidden-node output power by using sample-by-sample node output power estimation is derived. The method for estimating the node power is to use a single-pole low-pass filter. It is shown by simulation that the proposed algorithm gives faster convergence and lower minimum MSE than the RBF-SG algorithm.

• 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.31 no.3A
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• pp.214-220
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• 2006

In this paper, we propose the transmit power controlled adaptive modulated OFDM with frequency symbol spreading and equalization(TPC-AMS/FSS-OFDM) system. In the transmitter of the TPC-AMS/FSS-OFDM, each SP transformed signal is spread by orthogonal spreading codes and combined, so the detected signals obtain the same SINR(signal interference to noise ration) for each frequency symbol spreading block. In this case, we can assign the same modulation level and transmit power for each frequency symbol spreading block. Thus, the proposed system provides the increased throughput performance with reducing the total transmit power, FBI and MLI.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.1
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• pp.115-120
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• 2016

This paper deals with AM-SMMA (Adaptive Modulus-SMMA) adaptive equalization algorithm that is possible to improves the equalization performance using adaptive modulus which is propotional to the output signal power of equalizer in the current SMMA algorithm. The fixed statistic modulus of transmitted QAM signal is used in current SMMA algorithm in order to generate the error signal for updating the tap coefficient of equalizer. The adaptive modulus which is propotion to the equalizer output signal power is used in the proposed AM-SMMA. The computer simulation was performed in order to confirm the improved equalization performance of proposed algorithm. The performance index includes the output signal constellation, the residual isi and maximum distortion that is for the convergence characteristics, the SER. As a result of computer simulation, the residual isi, maximum distortion are better in the small weighting values. But in SER performance is better in the large weighting values.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.9
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• pp.2005-2013
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• 1997

The MCMA adaptive blind equalization algorithm has a excellent phase correction capabilities in addition to channel amplitude equalization, but has an inevitable error by mismatching between the original constellation points in arriving at the perfect equalization since unique new type constellation points are used as desired response instead of original constellation points and follows the slow convergence speed of CMA. In this paper, We propose an adaptive blind equalization algorithm with dual-mode, which has decision regions. Inside the decision regions, it operates as considering the moudlus of original data symbol point and outside the decision region, it operates as considerin gthe modulus of new constellation points. The proposed algorithm has a lower error in the steady state and rapid convergence speed toward steady state using the original data symbol points instead of new constellation points in the decision regions. From computer simulation, we confirm that the propposed algorithm has the performance superiority in residual ISI, convergence speed compared with the cnventional adaptive blind equalization algorithms, CMA, MCMA, Stop-and-Go algorithm.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.15 no.1
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• pp.131-137
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• 2015

This paper relates with the VSS-MMA (Variable Step Size-Multiple Modulus Algorithm) adaptive equalization algorithm which is possible to improving the equalization performance by use the nonlinear fuction of error signal in the MMA adaptive equalization algorithm that are used for the minimization of the intersymbol interference due to the distortion which occurs in the time dispersive channel for the transmission of QAM signal in the system.. In the conventional MMA, we obtains the tap coefficient of adaptive equalizer using the fixed step size, but in the VSS-MMA, we obtains the tap coefficient of adaptive equalizer using the variable step size based on a nonlinear function of error signal. By adapting the variable step size, it was confirmed that the improved equalization performance were obtained by computer simulation. For this, the equalizer output signal constellation, residual isi, maximum distortion, MSE and SER were used in the performace index.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.1901-1904
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• 2002

A good performance on communication systems is obtained by decreasing the length of training sequence In the initial stage of adaptive equalization. This paper presents a new approach to accomplish this, with the use of a training adaptive equalizer. The approach is based on combining the training and tracking modes, in which the training equalizer is updated by the LMS algorithm with the training sequence and then updated by a blind algorithm. By computer simulations, it is shown that a class of the proposed equalizers provides better performance than the conventional training equalizer.