• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2000.10a
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• pp.468-472
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• 2000

This paper discusses the development of a acoustic vector channel simulator for the performance analysis of a acoustic digital communication system. The channel simulator consists of transmission module, acoustic channel model, receiver, beamformer, and adaptive equalizer. QPSK source signal is generated by the parameters specified by a user and transmitted. The transmitted signal generate multipath signals which have a different delay, amplitude, and dopper Sequency. The multipath signals with the acoustic noises is the received signal. We can analysis the communication system performance according to the antenna structure, beamforming algorithm, and equalization algorithm.

• Journal of information and communication convergence engineering
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• v.8 no.6
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• pp.655-659
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• 2010

In the conventional PNC scheme, the relay node requires simultaneous transmission of two source nodes with strict power control and carrier-phase matching between two received symbols. However, this pre-equalization process at source nodes is not practical in fading channels. In this letter, we propose a novel physical-layer network coding (PNC) scheme with log-likelihood ratio (LLR) conversion for fading channels, which utilizes not pre-equalizer at transmitters (source nodes) but joint detector at receiver (relay node). The proposed PNC requires only channel side information at the receiver (CSIR), which is far more practical assumption in fading channels. In addition, the proposed PNC scheme can use the conventional modulation scheme like M-QAM regardless of modulation order, while the conventional PNC scheme requires reconfiguration of modulation scheme at the source nodes for detection of the received signal at relay node. We consider the combination of the proposed PNC and channel coding, and find that the proposed PNC scheme is easily combined the linear channel codes such as turbo codes, LDPC, and convolutional codes.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.4
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• pp.884-892
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• 1996

An adaptive blind equalizer is used to combat the distortions caused by a nonideal channel without resorting to a training sequence, given the received signal and statistical information of the transmitted signal. Incidentally, a multipath channel may result in a fade which produces intersymbol interference in the received signal. Therefore, a new type of algorithm which can compenste the effects of this fade is required in the multipath channel environment. In this paper, a modified form of adaptive blind equalization algorithm using stop-and-go algorithm for multichannel system is proposed. It is demonstrated via computer simulations that the performance of the proposed multichannel stop-and-go algorithm is much better than that of the conventional multichannel algorithms.

• The Transactions of the Korean Institute of Electrical Engineers A
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• v.48 no.8
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• pp.1037-1045
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• 1999

We propose two new classes of robust blind equalization algorithms against abrupt changes of channel conditions, which we call a triple-mode algorithm(TMA) and an automatic switch-over algorithm(ASA). The conventional DMGSA exhibits slow convergence rates due to the incorrect equalizer tap-updating process under the severe channel conditions. In order to speed up the convergence process, the TMA operates in triple-mode that is based on the dual-mode of the DMGSA incorporated with the tap-updating control modes of the SGA as well as the MSGA. Without resorting to the decision region for selecting the operation mode in the TMA, the ASA automatically switches the blind mode to the smoother conventional decision-directed mode. The ASA uses the error functional that is the weighted sum of the Generalized Sato error and the decision-directed error, where the weights correspond to the channel conditions. Test results on 16-QAM and 8-VSB datas confirm that the TMA and the ASA perform well under the sudden changes of channel conditions.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.4 no.4
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• pp.861-868
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• 2000

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. QPSK source signal is generated by the parameters specified by a user. The transmitted signal generates multipath signals which have a different delay, amplitude, and dopper frequency. The multipath singnals with the acoustic noises are the received signal. This paper presents the performance analysis of an acoustic digital communication system according to the antenna structure and the various baseband signal processing techniques.

• Journal of IKEEE
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• v.23 no.3
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• pp.1038-1045
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• 2019

This paper describes high-speed duobinary transceiver design techniques which are widely used to increase data-rate despite limited channel bandwidth. At high data-rate, signal level is severely degraded as signal frequency becomes larger than the channel bandwidth. Mathematically, a duobinary signal has lower frequency components compared to a Non-Return-to-Zero signal for the same data-rate. Therefore, by using the duobinary signaling, the signal loss can be effectively reduced in physical channel environment as compared to the Non-Return-to-Zero signaling. The mathematical basis of duobinary signaling, and its applications to high-speed transceiver design are investigated in this paper.

• ETRI Journal
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• v.44 no.5
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• pp.715-732
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• 2022

Despite the successful launch of Advanced Television Systems Committee (ATSC) 3.0 broadcasting worldwide, broadcasters are facing obstacles in constructing void-less large-scale single-frequency networks (SFNs). The bottleneck is the absence of decent on-channel repeater (OCR) solutions necessary for SFNs. In the real world, OCRs suffer from the maleficent feedback interference (FI) problem, which overwhelms the desired input signal. Moreover, the undesired multipaths between studio-linked transmitters and the OCR deteriorate the forward signals' quality as well. These problems crucially restrict the feasibility of conventional OCR systems, arousing the strong need for cost-worthy advanced OCR solutions. This paper presents an ATSC 3.0-specific solution of advanced OCR that solves the FI problem and refines the input signal. To this end, the FI canceler and channel equalizer functionalities are carefully implemented into the OCR system. The presented OCR system is designed to be fully compliant with the ATSC 3.0 specifications and performs a fast and efficient signal processing by exploiting the specific frame structure. The real product of ATSC 3.0 OCR is fabricated as well, and its feasibility is verified via field and laboratory experiments. The implemented solution is installed at a commercial on-air site and shown to provide substantial coverage gain in practice.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.12 no.2
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• pp.268-277
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• 2001

In this paper, the system performance with the convolution code using a Viterbi decoding and the one tap LMS equalizer applied to the OFDM system, which is suitable for IEEE 802.1la wireless LAN in indoor wireless channel, is analyzed through computer simulation. Indoor wireless channel is modeled as Rician fading channel, and QPSK and 16QAM scheme are used for subchannel modulation. In Rician fading channel with the power ratio of the direct path signal to the scattered signals, K=5 dB, BER of $10^{-4}$ is satisfied if the SNRs of the QPSK/OFDM and the 16QAM/OFDM are 8.6 dB and 19.2 dB in hard decision and 5.3 dB and 9.8 dB in soft decision, respectively. Compared with convolution code scheme, it is observed that 16QAM/OFDM system with the one tap LMS equalizer has the performance improvement of 8.6 dB and 2 dB in hard decision and soft decision, respectively.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.3
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• pp.115-120
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• 2011

This paper is concerned with the performance comparison of CR-CMA (Coordinate Reduction-CMA) and CM-CMA (Constellation Matching-Constant Modulus Algorithm) that is used for improving the convergence characteristic and residual intersymbol interference which are used as the performance index for an adaptive equalizer. The equalizer is used to reduce the distortion caused by the intersymbol interference on the wireless and the wired band-limited channel, and the blind method which does not need for extra bandwidth by the training sequence of digital code are researched. Recently, by using the merit of simple operation in the CMA, the performance improvement is obtained by the modifying the cost function of it. In this paper, the new algorithm, CR-CMA and CM-CMA, the performance analysis are performed and compared by computer simulation. The CR-CMA has a superior equalization characteristics in the recovered constellation, convergence speed and residual intersymbol interference than the CM-CMA by computer simulation.

• 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.