• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.6B
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• pp.1074-1081
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• 2000

In this paper, in order to speed up the convergence process and improve the steady mean square error simultaneously, we propose the Stop-and-Go Dual Mode Modified Constant Modulus Algorithm(SAG DM MCMA) for adaptive blind channel equalization of high order QAM. The proposed algorithm is a hybrid scheme of the Modified CMA that treat error signals with real and imaginary components of the equalizer output, the concept of dual mode CMA, and Stop-and-Go algorithm. As a result it can prevent blind equalization from converging to incorrect direction and simultaneously operates reliably for tap weight adaptation. We demonstrate via simulation that the proposed algorithm achieves lower steady state mean square error and residual ISI than the conventional algorithms under high order QAM signals and severe channel environment.

• 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.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.17 no.1
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• pp.19-24
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• 2017

Nowadays, based on the advancement of IT techniques, innovative products combining IT techniques to PA system are developing. In this paper, we presented the hybrid mixer controller for digital PA system. We develop the integrated mixer controller which includes the digital mixer composing an existing digital PA system and function of digital integrated controller. Developed integrated mixer controller consists of multichannel mixer function with 16 audio input channels, 8 output channels. And, it has an equalizer for processing digital audio signal, matrix and limiter. Also, the developed controller has some features such as internet connection for controlling of overall PA system and remote monitoring of mixer process condition.

• 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.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.4
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• pp.826-832
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• 2007

A recurrent neural network has been frequently used in equalizing the channel for fast communication systems. The existing techniques, however, have mainly dealt with time-invariant chamois. The modern environments of communication systems such as mobile ones have the time-varying feature due to fading. In this paper, powerful decision feedback - recurrent neural network is used as channel equalizer for nonlinear and time-varying system, and two kinds of algorithms, such as extended Kalman filter (EKF) and sigma-point Kalman filter (SPKF), are proposed; EKF is for fast convergence and good tracing function, and SPKF for overcoming the problems which can be developed during the process of first linearization for nonlinear system EKF.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.6
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• pp.115-121
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• 2013

This paper related with the DSE-MMA (Dithered Sign-Error MMA) that is the simplification of computational arithmetic number in blind equalization algorithm in order to compensates the intersymbol interference which occurs the passing the nonlinear communication channel in the presence of the band limit and phase distortion. The SE-MMA algorithm has a merit of H/W implementation for the possible to reduction of computational arithmetic number using the 1 bit quantizer in stead of multiplication in the updating the equalizer tap weight. But it degradates the overall blind equalization algorithm performance by the information loss at the quantization process compare to the MMA. The DSE-MMA which implements the dithered signed-error concepts by using the dither signal before qualtization are added to MMA, then the improved SNR performance which represents the roburstness of equalization algorithm are obtained. It has a concurrently compensation capability of the amplitude and phase distortion due to intersymbol interference like as the SE-MMA and MMA algorithm. The paper uses the equalizer output signal, residual isi, MD, MSE learning curve and SER curve for the performance index of blind equalization algorithm, and the computer simulation were performed in order to compare the SE-MMA and DSE-MMA applying the same performance index. As a result of simulation, the DSE-MMA can improving the roburstness and the value of every performance index after steady state than the SE-MMA, and confirmed that the DSE-MMA has slow convergence speed which meaning the reaching the seady state from initial state of adaptive equalization filter.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.2
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• pp.101-106
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• 2016

This paper proposes the SU-SE-MMA algorithm which applying the concept of selective updaing to the SE-MMA that is possible to reduce the intersymbol interference due to distortion occurred at the channel when transmit the nonconstant modulus 16-QAM signal. The SE-MMA emerged for the simplifying the computational operation from the current MMA adaptation algorithm, then it's has the fast convergence speed and has a problem of increase the residual component in the steady state. The SU-SE-MMA performs the selectively tap updating when the distance of equalizer output and specified transmit signal point is greater than the given threshold value and tap updaing does not occurred in the small distance. By this selective updating process, it is possible to more reduction in the computational operation in the propose algorithm. The improved adaptive equalization performance of SU-SE-MMA like as the equalizer output signal constellation, residual isi, MD, SER were confirmed by computer simulation compared to SE-MMA. As a result of simulation, the AV-SE-MMA has better performance in output signal constellation, residual isi and MD compared to the SE-MMA, but it was confirmed that the AV-SE-MMA has similar in the SER performance that means the robustness to the noise.

• Journal of Internet Computing and Services
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• v.15 no.3
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• pp.71-77
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• 2014

The equalization algorithm based on the cross-information potential concept and Dirac-delta functions (CIPD) has outstanding ISI elimination performance even under impulsive noise environments. The main drawback of the CIPD algorithm is a heavy computational burden caused by the use of a block processing method for its weight update process. In this paper, for the purpose of reducing the computational complexity, a new method of the gradient calculation is proposed that can replace the double summation with a single summation for the weight update of the CIPD algorithm. In the simulation results, the proposed method produces the same gradient learning curves as the CIPD algorithm. Even under strong impulsive noise, the proposed method yields the same results while having significantly reduced computational complexity regardless of the number of block data, to which that of the e conventional algorithm is proportional.

• Journal of Korea Multimedia Society
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• v.12 no.3
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• pp.329-334
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• 2009

When an optical disc is being played. the pick-up converts light to analog signal at first. The analog signal is equalized for removing the inter-symbol interference and then the equalized analog signal is converted into the digital signal for extracting the synchronized data and clock signals. There are a lot of algorithms that minimize the BER in extracting the synchronized data and clock when high. density optical disc like BD is being played in low speed. But if the high-density optical disc is played in high speed, it is difficult to adopt the same extraction algorithm to data PLL and PRML architecture used in low speed application. It is because the signal with more than 800MHz should be processed in those architectures. Generally, in the 0.13-${\mu}m$ CMOS technology, it is necessary to have the high speed analog cores and lots of efforts to layout. In this paper, the parallel data PLL and PRML architecture, which enable to process in BD 8x speed of the maximum speed of the high-density optical disc as the extracting data and clock circuit, is proposed. Test results show that the proposed architecture is well operated without processing error at BD 8x speed.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.2A
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• pp.157-165
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• 2004

The synchronization algorithm of IEEE 802.11a WLAN(Wireless Local Area Network) has three consecutive processes, which use a short code training symbol, a long code training symbol and a pilot symbol respectively. But in using this synchronization processes, the actual embodiment has two problems. First, the synchronization process has the complex structure using a long code training symbol and a pilot symbol. Second, since the long training symbol is only compensated with the offset correction coefficient, it can not be trusted perfectly. If the equalizer coefficient is obtained in this unstable period, the system performance is degraded. In particular, the system performance becomes worst in case of the 54 Mbps transmission system using the maximum length of data. In this paper, the new algorithm is proposed which can resolve the embodiment complexity of synchronization processes and structural defect, and also it is confirmed by simulation.