• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.8B
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• pp.745-751
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• 2002

The performance of existing CDMA-type multiple access data transmission systems is limited by interferences due to adverse mobile communication channel characteristics. Among them, the multi-user interference becomes one of the major performance degradation factors because the signal-to-signal orthogonality can be easily perturbed when numerous user signals are distorted by multipath fading channels and mixed together. In order to enhance the performance of CDMA-type systems by suppressing the multi-user interference, we have adopted chip-based cyclic prefix insertion along with adaptive one-tap DFE equalization into MC-DS-CDMA, which is known for its robustness in the frequency selective multipath fading channel environment. In order to assess the performance of the proposed system, a set of computer simulations is performed in the reverse link in which each user signal undergoes different multipath Rayleigh fading. The results show us a superior performance of our system over other CDMA systems in terms of SNR to BER measurements.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.5C
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• pp.295-301
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• 2011

In [1], it has been shown that the energy spreading transform (EST) based iterative equalizer (IE) could enhance its performance by improving the reliability of the decision feedback symbols without the help of the complicated channel decoder. In the matched filter (MF) based IE proposed in [1], however, its feedforward filter (FFF) has been designed in the frequency domain while its feedback filter (FBF) in the time domain. So its complexity increases proportional to the channel memory length. To solve this problem, in this paper, both FFF and FBF are designed in the frequency domain. This enables the proposed frequency domain IE (FD-IE) to achieve the lower complexity over the conventional method in the highly dispersive channel. In addition, simulation results demonstrate that the BER performance of the proposed method is the same as the conventional.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.9
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• pp.2131-2139
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• 2014

In this paper, a binary-coded physical-layer network coding (PNC) is considered when high-order modulation techniques are used at source nodes in wireless communication environments. In the conventional PNC schemes, tight power control and phase compensation are required at a relay node. However, they may not be feasible in practical wireless communication environments. Thus, we do not assume the pre-equalization in this paper, and we only utilize the channel state information at receiver (CSIR). We propose a signal detection method for the binary-coded PNC with high-order modulation, such as QPSK and 16QAM, at the source nodes, while the conventional scheme only consider the BPSK at source nodes. We also analyze the bit-error performance of the proposed technique in both uncoded and coded cases.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5C
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• pp.532-538
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• 2003

In this paper we have investigated a novel approach applying low-density parity-check coding to a COFDM-CDMA system, which operates in a multi-path fading mobile channel. Developed as a linear-block channel coder, the LDPC code is known for a superior signal reception capability in AWGN and/or flat fading channels with respect to increased encoding rates, however, its performance degrades when the communication channel becomes multi-path fading. For a typical multi-path fading mobile channel with a SNR of 16㏈ or lower. in order to obtain a BER lower than 1 out of 10000, the LDPC code with encoding rates below 1:3 requires not only the inherent parity check information but also the piloting information for refreshing front-end equalizer taps of COFDM-CDMA, periodically. For instance, while the 1:3-rate LDPC coded transmission symbol is consisted of data bits and parity-check bits in 1 to 3 proportion, on the other hand, in the proposed method the same rate LDPC transmission symbol contains data bits, parity check bits, and pilot bits in 1 to 2 to 1 proportion, respectively. The included pilot bits are effective not only for channel estimation and channel equalization but for symbol decoding by assisting the parity-check bits, hence, improving SNR vs BER performance over the conventional 1:3-rate LDPC code. The proposed system performance has been verified using computer simulations in multi-path, Rayleigh fading channels, and the results show us that the proposed method out-performs the general LDPC channel coding methods in terms of SNR vs BER measurements.

• 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.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.17 no.1
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• pp.77-82
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• 2017

This paper related with the VSS_SE-MMA (Varying Step Size_Signed Error-MMA) which possible to improving the equalization performance that employing the varying adaptive step size based on the nonlinearities of error signal of SE-MMA (Signed Error-MMA), compensates the intersymbol interference by distortion occurs at the communication channel, in the transmitting the spectral efficient nonconstant modulus signal such as 16-QAM. The SE-MMA appeared to the reducing the computational arithematic operation using the polarity of error signal in the updating the tap coefficient of present MMA adaptive equalizer, but have a problem of equalization performance degradation. The VSS_SE-MMA improves the problem of such SE-MMA, using the varying step size consider the error signal in the update the adaptive equalizer tap coefficient, and its improved performance were confirmed by simulation. For this, the output signal constellation of equalizer, the residual isi and maximum distortion, MSE and SER were applied. As a result of computer simulation, it was confirmed that the VSS_SE-MMA algorithm has nearly same in convergence speed and has more good performance in every performance index at the steady state.

• Journal of Navigation and Port Research
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• v.39 no.4
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• pp.293-297
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• 2015

Underwater acoustic communications has been limited use for military purposes in the past. However, the fields of underwater applications expend to detection, submarine and communication in recent. The excessive multipath encountered in underwater acoustic communication channel is creating inter symbol interference, which is limiting factor to achieve a high data rate and bit error rate performance. To improve the performance of a received signal in underwater communication, many researchers have been studied for channel coding scheme with excellent performance at low SNR. In this paper, we applied BCJR decoder based ( 2,1,7 ) convolution codes and to compensate for the distorted data induced by the multipath, we applying the turbo equalization method. Through the underwater experiment on the Gyeungcheun lake located in Mungyeng city, we confirmed that turbo equalization structure of BCJR has better performance than hard decision and soft decision of Viterbi decoding. We also confirmed that the error rate of decoder input is less than error rate of $10^{-1}$, all the data is decoded. We achieved sucess rate of 83% through the experiment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.9A
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• pp.1290-1298
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• 2000

Turbo Code decoder is an iterate decoding technology, which extracts extrinsic information from the bit to be decoded by calculating both forward and backward metrics, and uses the information to the next decoding step Turbo Code shows excellent performance, approaching Shannon Limit at the view of BER, when the size of Interleaver is big and iterate decoding is run enough. But it has the problems which are increased complexity and delay and difficulty of real-time processing due to Interleaver and iterate decoding. In this paper, it is analyzed that MAP(maximum a posteriori) algorithm which is used as one of Turbo Code decoding, and the factor which determines its performance. MAP algorithm proceeds iterate decoding by determining soft decision value through the environment and transition probability between all adjacent bits and received symbols. Therefore, to improve the performance of MAP algorithm, the trust between adjacent received symbols must be ensured. However, MAP algorithm itself, can not do any action for ensuring so the conclusion is that it is needed more algorithm, so to decrease iterate decoding. Consequently, MAP algorithm and Turbo Code performance are analyzed in the nongaussian channel applying Robust equalization technique in order to input more trusted information into MAP algorithm for the received symbols.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.1
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• pp.121-127
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• 2010

This paper relates with the improved CMA blind adaptive equalization algorithm which uses the constellation matching method that improve the inverse modelling efficiency of a communication channel compared to the present CMA blind adaptive equalizer. The amplitude distortion can be compensated in the present CMA blind adaptive equalizer which is used for the reduction of intersymbol interference by distortion that generate such as a band limited wireless mobile channel, but in the improved adaptive alogorithm operates with the minimize the amplitude phase distortion in the output of equalizer by applying the cost function that is composition of additional signal constellation matching error terms. In order to evaluation of the inverse modeling efficiency of improved algorithm, the residual intersymbol interference and recovered signal constellation were compared by computer simulation. As a result of comparion of computer simulation, the improved algorithm has a good stability in the residual intersymbol interference in the steady state, but it has a slow convergence rate in the adaptation state in initial state.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.8 s.338
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• pp.17-24
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• 2005

This paper proposes a Decision Feedback Equalizer (DFT) with an error feedback filter for blind channel equalization. The proposed equalizer uses Least Mean Square(LMS) Algorithm and Multi-Modulus Algorithm (MMA), and has been designed for 64/256 QAM constellations. The existing MMA equalizer uses either two transversal filters or feedforward and feedback filers, while the proposed equalizer uses feedforward, feedback and error feedback filters to improve the channel adaptive performance and to reduce the number of taps. The proposed equalizer has been simulated using the $SPW^{TM}$ tool and it shows performance improvement. It has been modeled by VHDL and logic synthesis has been performed using the $0.25\;\mu m$ Faraday CMOS standard cell library. The total number of gates is about 190,000 gates. The proposed equalizer operates at 15 MHz. In addition, FPGA vertification has been performed using FPGA emulation board.