• Journal of Korea Multimedia Society
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• v.11 no.12
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• pp.1625-1634
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• 2008

In this study, a modified Fuzzy C-Means algorithm with Gaussian weights (MFCM_GW) is presented for the problem of nonlinear blind channel equalization. The proposed algorithm searches for the optimal channel output states of a nonlinear channel based on received symbols. In contrast to conventional Euclidean distance in Fuzzy C-Means (FCM), the use of the Bayesian likelihood fitness function and the Gaussian weighted partition matrix is exploited in this method. In the search procedure, all possible sets of desired channel states are constructed by considering the combinations of estimated channel output states. The set of desired states characterized by the maxima] value of the Bayesian fitness is selected and updated by using the Gaussian weights. After this procedure, the Bayesian equalizer with the final desired states is implemented to reconstruct transmitted symbols. The performance of the proposed method is compared with those of a simplex genetic algorithm (GA), a hybrid genetic algorithm (GA merged with simulated annealing (SA):GASA), and a previously developed version of MFCM. In particular, a relative]y high accuracy and a fast search speed have been observed.

• Journal of Digital Convergence
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• v.12 no.12
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• pp.329-335
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• 2014

In this paper, to improve the performance of PB/MC-CDMA system, we have researched about the decision feedback channel estimation method using the pilot symbol with data symbol. The PB/MC-CDMA system is able to obtain the improved BER by frequency diversity gain and frequency domain equalizer in the frequency selective fading channel. However, when it is not the estimating of channel exactly, it is degrading the performance of BER in the system for occurred the interference among users. To improve the performane system in the multi-user environment the proposed system is using the decision feedback to estimate channel using the channel estimated value of the first stage and second stage. The proposed system is evaluated by computer simulation. The proposed system is not only able to improve the performance of BER by decreasing the interference to each user, but also the proposed system is possible to reduce number of pilot symbol to estimate the channel. Therefore, it confirmed the proposed system improves the performance than the conventional system.

• Journal of Broadcast Engineering
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• v.9 no.4 s.25
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• pp.371-383
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• 2004

In this paper we consider technological requirements to broadcast digital television signals using single frequency networks (SFN) in the Advanced Television Systems Committee (ATSC) transmission systems and propose equalization digital on-channel repeater (EDOCR) that overcomes the limitations of conventional digital on-channel repeaters (DOCRs). Since there are no forward error correction (FEC) decoder and encoder, the EDOCR does not have an ambiguity problem. In addition, since an adaptive equalizer in the EDOCR removes multi-path signals, additive white Gaussian noise (A WGN), and feedback signal due to low antenna isolation, the EDOCR may have good output signal quality with high power.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.30B no.12
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• pp.31-39
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• 1993

In the playback signal processing of a digital magnetic recording system, the major signal processing processes consist of pulse equalization. pulse detection, clock recovery, and data recovery. Equalizer which compensates interference occurrde between pulses recorded in high density on a magnetic media is realized by pulse slimming method, and pulse detection by a integrating detector. Clock recovery from the detector output was accomplished by using PLL. and data recovery to reduce noise effects was carried out by utilizing the three sampling clocks recovered in clock recovery process. In this paper these processes are implemented in hardware and its performance is evaluated by experimenting with a commercial DAT. It was found that the playback signal processor proposed is suitable to the practical high density magnetic recording system.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1E
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• pp.18-24
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• 2002

This paper discusses the development of an acoustic vector channel simulator for the performance analysis of an acoustic digital communication system. The channel simulator consists of transmission module, acoustic channel model, receiver, beamformer, and adaptive equalizer. The source signal (QPSK) is generated by the specified parameters. The transmitted signal generates multipath signals which have a different delay, amplitude and doppler frequency. The paper presents in details the approach to the performance analysis of an acoustic digital communication system according to the antenna structure and the various baseband signal processing techniques.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.9B
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• pp.1226-1235
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• 2001

The convergence analysis of the improved multiplication free adaptive digital filter (IMADF) with a fractionally-spaced equalizer (FSE) to remove the intersymbol interference (ISI) in fast data transmission is presented. The IMADF structure use the one-step predicted filter in the multiplication-free adaptive digital filter (MADF) structure using the DPCM and Sign algorithm. In the experimental results, the IMADF algorithm has reduced the computational complexity by use of only the addition operation without a multiplier. Also, under the condition of identical stationary-state error, it could obtain the stabled convergence characteristic that the IMADF algorithm is almost same as the sign algorithm, but is better than the MADF algorithm. Here, this algorithm has effective characteristics when the correlation of the input signal is highly.

• Korean Journal of Optics and Photonics
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• v.13 no.6
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• pp.486-492
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• 2002

In this paper. we presented 1 & 2 dimensional minimum mean-squared-error (MMSE) equalization scheme in a digital holographic data storage system to improve bit-error-rate (BER) and to mitigate inter-symbol interference (ISI) which were generated during the data storage and retrieval processes. We showed experimentally for ten data pages retrieved from the holographic storage system that BER and signal-to-noise ratio (SNR) were improved by adopting MMSE equalization.

• Journal of Broadcast Engineering
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• v.5 no.2
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• pp.159-166
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• 2000

This paper analyzed the transmission performance when we transmited digital TV signal through analog CATV network. We considered random noise, microreflection, and composite tripple beats as channel impairment factors, and random noise and CTB( Composite Tripple Beat) are modelled as Gaussian distribution and Weibull distribution respectively. Channel modellings for microreflections are classified as three cases, and we exploited carrier recovery and blind equalizer to minimize their effects. We developed a computer simulator for the digital transmission system using DVB-C specification to analyze digital transmission performance, and found BER values according to $E_{b}$/$N_{o}$ for each configuration.

• Journal of the Institute of Electronics and Information Engineers
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• v.49 no.12
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• pp.184-193
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• 2012

In this paper, a 5-bit 2-GS/s 2-way time interleaved pipeline ADC for high-speed serial link receiver is demonstrated. Implemented as a current-mode amplifier, the stage ADC simultaneously processes the tracking and residue amplification to achieve higher sampling rate. In addition, each stage incorporates a built-in 1-tap FIR equalizer, reducing inter-symbol-interference (ISI）without an extra digital post-processing. The ADC is designed in a 110nm CMOS technology. It comsumes 91mW from a 1.2-V supply. The area excluding the memory block is $0.58{\times}0.42mm^2$. Simulation results show that when equalizer is enabled, the ADC achieves SNDR of 25.2dB and ENOB of 3.9bits at 2.0GS/s sample rate for a Nyquist input signal. When the equalizer is disengaged, SNDR is 26.0dB for 20MHz-1.0GHz input signal, and the ENOB of 4.0bits.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.6
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• pp.815-823
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• 1993

In this paper, the structure of modified multiplication-free adaptive filter(M-MADF) and convergence analysis are presented. To evaluate the performance of proposed M-MADF algorithm, fractionally spaced equalizer (FSE) is used. The input signals are quantized using DPCM and the reference signals is processed using a first-order linear prediction filter, and the outputs are processed by a conventional adaptive filter. The filter coefficients are updated using the Sign algorithm. Under the assumption that the primary and reference signals are zero mean, wide-sense stationary and Gaussian, theoretical results for the coefficient misalignment vector and its autocorrelation matrix of the filter are driven. The convergence properties of Sign. MADF and M-MADF algorithm for updating of the coefficients of a digital filter of the fractionally spaced equalizer (FSE) are investigated and compared with one another. The convergence properties are characterized by the steady state error and the convergence speed. It is shown that the convergence speed of M-MADF is almost same as Sign algorithm and is faster that MADF in the condition of same steady error. Especially it is very useful for high correlated signals.