• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.3
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• pp.600-607
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• 2007

In the next generation mobile multimedia communications, the broad band shot-burst transmissions are used to reduce end-to-end transmission delay, and to limit the time variation of wireless channels over a burst. However, training overhead is very significant for such short burst formats. So, the availability of the short training sequence and the fast converging adaptive algorithm is essential in the system adopting the symbol-by-symbol adaptive equalizer. In this paper, we propose an adaptive equalizer using the DWMT (discrete multi-wavelet transform) and LMS (least mean square) adaptation. The proposed equalizer has a faster convergence rate than that of the existing transform-domain equalizers, while the increase of computational complexity is very small.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.1
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• pp.164-171
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• 2008

Adaptive equalizers are inevitable schemes in digital communication systems for compensating the transmission channel distortion. Additionally, to obtain the required BER(Bit Error Rate), the adaptive algorithms appropriate to the mobile communication channels are required. In this paper, we propose the NPDFE (Noise-Predictive Decision Feedback Equalizer) for communication systems performance improvement in mobile communication channels. The performance of the proposed NPDFE with QPSK (Quadrature Phase Shift Keying) is simulated under AWGN (Additive White Gaussian Noise), Ricean fading, ETSI (European Telecommunications Standards Institute) fading, and Rayleigh fading channels. The equalizers used in simulations are a LE (Linear Equalizer), a DFE (Decision Feedback Equalizer), and a NPDFE. Moreover, the equalizer performance criterion of the QPSK is the BER.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.6B
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• pp.1082-1091
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• 2000

In this paper, the efficient equalization method for OFDM(Orthogonal Frequency Division Multiflexing) System using the QAM(Quadrature Amplitude Modulation) in multipath fading channel is proposed in order to faster and more efficiently equalize the received signals that are sent over real channel. In generally, the one-tap linear equalizers have been used in the frequency-domain as the existing equalization method for OFDM system. In this technique, if characteristics of the channel are changed fast, the one-tap linear equalizers cannot compensate for the distortion due to time variant multipath channels. Therefore, in this paper, we use one-tap non-linear equalizers instead of using one-tap linear equalizers in the frequency-domain, and also use the linear equalizer in the time-domain to compensate the rapid performance reduction at the low SNR(Signal-to-Noise Ratio) that is the disadvantage of the non-linear equalizer. In the frequency-domain, when QAM signals, consisting of in-phase components and quadrature (out-phase) components, are sent over the complex channel, the only in-phase and quadrature components of signals distorted by the multipath fading are changed the same as signals distorted by the noise. So the cross components are canceled in the frequency-domain equalizer. The time-domain equalizer and the adaptive algorithm that has lower-error probability and fast convergence speed are applied to compensate for the error that is caused by canceling the cross components in the frequency-domain equalizer. In the time-domain, To compensate for the performance of frequency-domain equalizer the time-domain equalizes the distorted signals at a frame by using the Gold-code as a training sequence in the receiver after the Gold-codes are inserted into the guard signal in the transmitter. By using the proposed equalization method, we can achieve faster and more efficient equalization method that has the reduced computational complexity and improved performance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.2B
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• pp.197-209
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• 2010

In this paper, We have designed and analyzed a characteristics of backplane channel having 40 inch strip line length of four lanes and Flame Retardant four (PR-4) material, and have designed 40 Gb/s Receive and adaptive equalizer and its high-speed equalization algorithm using the backplane channel characteristics. For 40 Gb/s high-speed data communications pass through the backplane, a 10Gb/s 4 channel receive & equalizer with DFE except for FFE was proposed. This receive and equalizer meets the requirements of the IEEE Std P802.3ba standard-based receive equalizer to implement equalizers on the receive end of a 46 inch length's backplane channel.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.32A no.1
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• pp.14-25
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• 1995

The main contribution of this paper can be summarized in two items as follws. Firstly, a modelling of mobile communication channel with Rayleigh fading characteristics is presented. Actual signal environments can be approximated as being real measurements by a proper mathematical representation of fluctuation of channel parameters due to Doppler effect, that is determined by the relative speed between transmitter and receiver, and noises, that vary at each sampling time. Secondly, an alternative procedure of synthesizing an adaptive equalizers is presented for recovering original signals that have been corrupted through the modelled channel. In order to compute the optimal tap coefficients for a high speed data(512 k symbol/sec) on a real-time basis, the CGM that guarantees fast and stable convergency is adopted during the training period of each frame. The coefficients obtained by the CGM are used as initial values for the LMS algorithm to trace the optimal coefficients during the data period that vary at each sampling time due to the mobility and noise at the receiver. Using the modelling presented in this paper, distributions of received signal power in various signal environments are demonstrated. The performance of the eqalizer proposed in this paper is shown as a function of BER under the various signal circumstances of mobile communications.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.7
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• pp.1364-1373
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• 1994

In this thesis, an algorithm for fast convergence to the steady state and for achieving an improved MSE in blind adaptive equalizers is proposed. The conventional radius-directed algorithm can be transformed into an algorithm that provides effective blind convergence in the aspect of the MSE as the convergence speed. This can be achieved through altering the stop and go algorithm. The performance of the new algorithm is analyzed and compared with the two conventional algorithms, such as the CMA and the stop and go algorithm. The experimental results show the superiority of the new algorithm.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.22 no.3
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• pp.627-632
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• 2021

ITL (information-theoretic learning) has been applied successfully to adaptive signal processing and machine learning applications, but there are difficulties in deciding the kernel size, which has a great impact on the system performance. The correntropy algorithm, one of the ITL methods, has superior properties of impulsive-noise robustness and channel-distortion compensation. On the other hand, it is also sensitive to the kernel sizes that can lead to system instability. In this paper, considering the sensitivity of the kernel size cubed in the denominator of the cost function slope, a new adaptive kernel estimation method using the rate of change in error power in respect to the kernel size variation is proposed for the correntropy algorithm. In a distortion-compensation experiment for impulsive-noise and multipath-distorted channel, the performance of the proposed kernel-adjusted correntropy algorithm was examined. The proposed method shows a two times faster convergence speed than the conventional algorithm with a fixed kernel size. In addition, the proposed algorithm converged appropriately for kernel sizes ranging from 2.0 to 6.0. Hence, the proposed method has a wide acceptable margin of initial kernel sizes.

• Journal of Communications and Networks
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• v.12 no.5
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• pp.399-405
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• 2010

Blind equalization techniques have been used in broadcast and multipoint communications. In this paper, two criteria of minimizing Euclidian distance between two probability density functions (PDFs) for adaptive blind equalizers are presented. For PDF calculation, Parzen window estimator is used. One criterion is to use a set of randomly generated desired symbols at the receiver so that PDF of the generated symbols matches that of the transmitted symbols. The second method is to use a set of Dirac delta functions in place of the PDF of the transmitted symbols. From the simulation results, the proposed methods significantly outperform the constant modulus algorithm in multipath channel environments.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.9C
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• pp.876-881
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• 2006

In this paper, we propose a complex-valued IIR filter for digital VSB signals based on CMA in order to efficiently mitigate multipath distortions, especially the leakage from the quadrature component. The proposed equalizer overcomes the drawback of the conventional real-valued IIR equalizers that it attempts to equalize Hilbert transform of quadrature component. We demonstrate via simulation that the proposed complex IIR filter successfully mitigates the leakages from the quadrature component, while the conventional real IIR filter requires a longer IIR filter to achieve the same performance. We present cost function analysis for a simple two-tap case showing that the proposed IIR equalizer with CMA for VSB signals has a global minimum at the desired location.

• 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.