• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.4C
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• pp.505-513
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• 2004

Most video codecs including the international standards use the block-based hybrid structure for efficient compression. But for low bit-rate applications such as video transmission through wireless channels, the blocking artifacts degrade image qualify seriously. In this paper, we propose an adaptive de-blocking algorithm using characteristics of the block boundaries. Blocking artifacts contain the high frequency components near the block boundaries, therefore the lowpass filtering can remove them. However, simple lowpass filtering results into blurring by removing important information such as edges. To overcome this problem, we determine the modes depending upon the characteristics of pixels adjacent to block boundary then proper filter is applied to each area. Simulation results show that proposed method improves de-blocking performance compared to that of MPEG-4.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.1
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• pp.65-70
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• 2005

This paper deals with the problem of double-talk (DT) detection in anacoustic echo canceller (AEC). In the DT detection algorithm with correlation coefficient, detection errors occasionally occur because it is hard to set the threshold to distinguish DT from echo path change (EPC). Adaptive filter falls into the situation that it stops updating its filter coefficients when EPC is erroneously considered as DT at the starting-point of EPC. In addition, in case of echo path changing during the DT period, the end-point detection of DT period fails so that the AEC cannot update its filter coefficients for a while even after the DT period ends. To solve these problems, in this paper, we propose a novel AEC that employs an auxiliary filter. We use the idea that though the error signal cannot be estimated using reference signal in case or DT situation but it can be in case or EPC situation. The experimental result verifies that the proposed method could solve the problems caused by DT detection error or echo path change during the DT period.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.7 s.361
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• pp.35-40
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• 2007

As the mobile communication service is widely used and the demand for wireless repeaters is rapidly increasing because of the easiness of extending service areas. But a wireless repeater has a problem that the output of the transmit antenna is partially fed back to the receive antenna, which results in feedback interference. In this paper, we propose a new varable step-size LMS algorithm which utilizes correlation between reference and error signals to adjust the step sizes, for cancelling the feedback interference signals in the WCDMA repeater under time-varying multi-path channels. The proposed algorithm was evaluated through computer simualation by being applied to the feedback canceling filter of the WCDMA repeater. The simulation results demonstrated that the proposed one is superior to the conventional ones in terms of the cancelation perormance.

• Journal of the Institute of Convergence Signal Processing
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• v.1 no.1
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• pp.82-88
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• 2000

In this paper, We compare the Harr wavelet of the S+P transform with various integer coefficient filter banks and apply 9/7 ICFB to the wavelet transform. In addition, we propose a entropy-coding method that exploits the multiresolution structure and the feedback of the prediction error, and can efficiently compress the transformed image for progressive transmission. Simulation results are included to compare to the compression ratio using the S+P transform with different types of images.

• The Journal of the Acoustical Society of Korea
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• v.40 no.5
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• pp.452-459
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• 2021

Estimating the mutual time delay between two acoustic sensors is used in various fields such as tracking and estimating the location of a target in room acoustics and sonar. In the time delay estimation methods, there are a non-parametric method, such as Generalized Cross Correlation (GCC), and a parametric method based on system identification. In this paper, we propose a time delay estimation method based on the parametric method. In particular, we propose a method that considers the noise in each receiving acoustic sensor. Simulation confirms that the proposed algorithm is superior to the existing generalized cross-correlation and adaptive eigenvalue analysis methods in white noise and reverberation environments.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.48 no.10
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• pp.53-59
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• 2011

In recent years, mobile communication service is used extensively as a larger service area for the maintenance of quality of service required by the expansion of service areas and As the ever-increasing role in relays, and the location is relatively easy to install and less constrained costs, operating cost savings in terms of ICS(Interference Cancellation System) repeaters are required. However, an adaptive algorithm that is applied when updating the filter due to the increase in volume of operations increase the complexity of hardware implementation is fraught with many difficulties. In this paper, if there is a path that feedback. ICS repeater utilizing baseband signal processing for the removal of interfering signals from the feedback operation, significantly reducing the amount of reducing hardware complexity Complex Singed Signed LMS adaption algorithm is proposed. Proposed algorithm for evaluating the performance of Static channel WCDMA signal environment for the ICS, the results of the simulation algorithm, convergence speed and better performance in therms of convergence errors that are required through the implementation of the operation greatly reduces the amount of hardware complexity able to reduce the effect was visible.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.10
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• pp.166-174
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• 1998

Neural networks, having highly nonlinear dynamics by virtue of the distributed nonlinearities and the learing ability, have the potential for the adaptive prediction of nonstationary signals. This paper describes the nonlinear prediction of these signals in two ways; using a nonlinear module and the cascade combination of nonlinear and linear modules. Fully-connected recurrent neural networks (RNNs) and a conventional tapped-delay-line (TDL) filter are used as the nonlinear and linear modules respectively. The dynamic behavior of the proposed predictors is demonstrated for chaotic time series adn speech signals. For the relative comparison of prediction performance, the proposed predictors are compared with a conventional ARMA linear prediction model. Experimental results show that the neural networks based adaptive predictor ourperforms the traditional linear scheme significantly. We also find that the cascade combination predictor is well suitable for the prediction of the time series which contain large variations of signal amplitude.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.10A
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• pp.922-929
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• 2005

This paper proposes an adaptive pilot beacon equipment for mobile communication systems based on direct spread spectrum technology which generates the pilot channel for handoff between base stations by using the information acquired from the downstream wireless signal regarding the overhead channel information. Such an adaptive pilot beacon equipment will enable low power operation since among the wireless signals, only the pilot channel will be generated and transmitted. The pilot channel in the downstream link of the CDMA receiver is used to acquire time and frequency synchronization and this is used to calibrate the offset for the beacon, which implies that time synchronization using GPS is not required and any location where forward receive signal can be received can be used as the installation site. The downstream link pilot signal searching within the CDMA receiver is performed by FPGA and DSP. The FPGA is used to perform the initial synchronization for the pilot searcher and DSP is used to perform the offset correction between beacon clock and base station clock. The CDMA transmitter the adaptive pilot beacon equipment will use the timing offset information in the pilot channel acquired from the CDMA receiver and generate the downstream link pilot signal synchronized to the base station. The intermediate frequency signal is passed through the FIR filter and subsequently upconverted and amplified before being radiated through the antenna.

• Proceedings of the KIEE Conference
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• 2003.04a
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• pp.212-216
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• 2003

The performance of active power filter depends on the inverter characteristic, the control method, and the accuracy of reference signal generator. The accuracy of reference generator is the most critical item to determine the performance of active power filter. This paper introduces a novel reference signal generator composed of improved adaptive predictive filter. The performance of proposed reference signal generator was verified by means of simulation with MATLAB. The simulation result confirm that the proposed reference signal generator can be utilized for the active power filter.

• Journal of Biomedical Engineering Research
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• v.20 no.6
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• pp.601-607
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• 1999

상악(maxilla)과 하악(mandibular)의 최대 교합상태에서 하악을 기계적으로 자극(jaw jerk)할 경우 교근(masseter muscle)의 근전도 (electromyography)에서는 근신호가 일시적으로 침묵하는 형태의 휴지기(silent period) 현상이 발생한다. 턱관절 질환(temporo-mandibular joint dysfuntion)이 없는 정상인의 경우 24ms 정도의 휴지기가 나타나지만, 턱관절 질환 환자의 경우 평균 60ms 정도임을 볼 때 휴지기는 턱관절 질환을 진단하는 중요한 요소라 할 수있다. 본 논문에서는 이러한 휴지기 기간을 자동적으로 결정하기 위해 mCFAR 알고리즘을 제안하고 CLMS 적응 필터를 사용하여 근전도 신호의 왜곡을 가져오는 전원 잡음의 영향을 효과적으로 제거하였다. 실험 결과 전원 잡음에 대해 강건하며, 정확한 휴지기 기간을 결정할 수 있다.