• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.9B
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• pp.908-913
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• 2009

Mobile phones became important household appliance that cannot be without in our daily lives. And the short messaging service (SMS) in these mobile phones is 1.5 to 2 times more than the voice service. However, the spam filtering functions installed in mobile phones take a method to receive specific number patterns or words and recognize spam messages when those numbers or words are present. However, this method cannot properly filters various types of spam messages currently dispatched. This paper proposes a more powerful and more adaptive spam filtering system using SVM and thesaurus. The system went through a process of isolating words from sample data through pro-processing device and integrating meanings of isolated words using a thesaurus. Then it generated characteristics of integrated words through the chi-square statistics and studied the characteristics. The proposed system is realized in a Window environment and the performance is confirmed through experiments.

• The Journal of the Acoustical Society of Korea
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• v.14 no.3
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• pp.28-36
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• 1995

In this paper, a double loop control model using leaky delay LMS algorithm are proposed for active noise control. The proposed double loop control model estimates the loudspeaker characteristic and the error path transfer function with on-line using only gain and acoustic time delay to reduce computation burden. The control of error signal through double loop control scheme makes the more robust cntrol system. The input signal of filter to estimate acoustic time delay is used difference between input signal of input microphone and adaptive filter output. And also, in nonstationary environments, the leaky delay LMS algorithm is employed to counteract parameter drift of delay LMS algorithm. For practical noise signal, the proposed double loop control model reduces noise level about 12.9 dB.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.9A
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• pp.1329-1338
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• 1999

In this paper, an efficient signal interference control technique to improve the convergence speed of LMS algorithm is introduced. The convergence characteristics of the proposed algorithm, whose coefficients are multiply adapted in a symbol time period by recycling the received data, are analyzed to prove theoretically the improvement of convergence speed. According as the step-size parameter $\mu$ is increased, the rate of convergence of the algorithm is controlled. Also, a increase in the step-size parameter $\mu$ has the effect of reducing the variation in the experimentally computed learning curve. Increasing the eigenvalue spread has the effect of controlling down the rate of convergence of the adaptive equalizer and also increasing the steady-state value of the mean squared error and also demonstrate the superiority of signal interference control to the filter algorithm increasing convergence speed by (B+1) times due to the data-recycling LMS technique.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.05a
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• pp.712-716
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• 2003

본 논문은 신경망을 이용한 간섭 신호 제어로써 복합 다중 퍼셉트론에서 DS/SS 이동 통신에서의 수신된 신호들을 역전파 학습알고리즘을 이용하여 검출하는 것에 대하여 연구한다. 수신 신호가 일정한 비트율을 갖는 채널에 전송하기 위하여 신경망을 이용한 새로운 탭 가중치 갱신 제어 방법을 제안한다. 적응 횡단선 필터는 심볼간의 채널에 발생하는 상호 심볼간 간섭을 억압하기 위해 LMS 알고리즘 사용한다. 이 알고리즘은 원하는 응답과 실제 출력간의 차인 에러를 이용하여 탭 가중치 조절 메카니즘을 통해 탭 가중치를 갱신함으로서 효과적으로 간섭을 제거하였다. 본 논문은 상호 심볼간 간섭을 효율적으로 억압해온 기존의 LMS 알고리즘에 다계층 퍼셉트론 신경망을 조합한 새로운 BISP 알고리즘을 제안하였으며, 제안된 알고리즘을 통해 탭 가중치 갱신이 보다 효율적으로 이루어짐을 알 수 있다. 시뮬레이션 결과를 통해 제안된 알고리즘의 평균 자승 에러의 수렴 특성이 기존 LMS 알고리즘을 이용한 수렴특성보다 우월하다는 것을 나타내었다.

• Proceedings of the KSMRM Conference
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• 2003.10a
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• pp.76-76
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• 2003

목적： fMRI와 뇌파의 동시측정 방법은 높은 시간적-공간적 해상도를 동시에 달성할 수 있는 뇌기능 측정을 가능케 하고, 각각의 방법이 제공하는 정보를 상호보완적으로 동시에 이용할 수 있도록 하는 유용한 방법이다. 그러나 자기공명영상을 얻기 위한 경사자계가 유발하는 잡음과 높은 자기장 내에서의 심장박동으로 인한 미세한 움직임 때문에 생기는 잡음이 뇌파신호의 질을 크게 저하하는 요소로 작용하여 이를 해결하기 위한 계측시스템의 최적화와 신호처리 방법의 개발이 요구된다. 본 논문에서는 평균차감법과 recursive least square 적응 필터링 방법에 기반한 심장 박동잡음의 감소를 위한 신호처리 방법의 개발 결과에 대한 연구 결과를 제시한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.1
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• pp.193-204
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• 1997

In this research we propose the design of the Order Statistic-Adaptive Weighted Average Hybrid(OS-AWAH) filter which can suppress noise from the corrupted image sequence effectively while preserving the image structure. The proposed filter combines the desirable properties of the order static based spatial filter which can preserve the image structure while reducing noise and the adaptive weighted average based temporal filter which can adapt the filtering weights according to the amount of motion without motion estimation. Performance characteristics of the OS-AWAH filter in noisy sequences containing moving step edges are investigated throuth computer simulations and compared with the median based filters such as 3-D WM(weighted median) filter, MMF (multistage median filter), ADCWM(adaptive directional center weighted median) filter. The visual evaluations are also carried out by applyin gthe filters to the real images. The statistical analysis and experimental reslts show that the OS-AWAH filter is effective in preserving image structures while suppressing noise effectively without motion compensation preprocessing.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.15 no.4
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• pp.360-367
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• 2004

In this paper, we have implemented the digital signal processing system for intruder detection using speckle pattern variation in multi-me optical fiber with hypersensitive and high fidelity. The performance of the implemented system was evaluated by experiments. In order to improve the system performances we applied the adaptive digital filter. In experimental results we could see 96 % intruder detection and 90 % man/car discrimination probability.

• Proceedings of the KAIS Fall Conference
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• 2010.05a
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• pp.249-252
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• 2010

LPC 계수를 LSP 변환하는 방법에는 복소근, 실근, 비율 필터, 체비셰프 급수, 적응적 순차형 최소제곱 평균 방법(adaptive sequential LMS) 등이 있다. 이 방법들 중 음성 부호화기에서 주로 사용하는 실근 방법은 근을 구하기 위해 주파수 영역을 순차적으로 검색하기 때문에 계산시간이 많이 소요되는 단점을 갖는다. 본 논문에서는 LPC에서 LSP로 변환하는 4가지 고속 알고리즘을 제안한다. 첫 번째 방식에서는 검색간격에 멜 스케일을 적용하였고, 두 번째는 홀수번째 LSP 파라미터의 분포도를 이용하여 검색순서를 조정한 방법이다. 세 번째 방식과 네 번째 방식에서는 각각, 모음 특성, LSP 분포특성과 해상도를 이용하여 계산시간을 단축하였다. LSP 변환시간은 4가지 방법 모두 35~50% 단축되었다. 또한 실험결과에서는 각 알고리즘의 고유한 특성에 대하여 분석한다.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.10 s.340
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• pp.1-8
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• 2005

This Paper Proposes the algorithm for improving the Performance or symbol timing synchronization in hoc terrestrial DTV system. The Gardner algerian is used for symbol timing synchronization has good performance in multipath fading environment but degradation of performance is caused by jitter. Though the amount of jitter becomes more little as narrow bandwidth of loop Inter, convergence speed becomes slower. This paper propose the algorithm that averages out values of loop filter every certain time and gradually reduces the bandwidth of loop filter after estimating offset using this average for the high speed of convergence and reducing the met of jitter. The proposed algorithm has better performance with high speed of convergence and the amount of jitter than conventional method.

• Journal of the Korea Society of Computer and Information
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• v.19 no.2
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• pp.101-107
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• 2014

This study is proposed the novel PPG sensor device and the signal processing method to replace the acceleration sensor that is used to reject motion artifacts contained in photoplethysmography(PPG). The proposed method is to reject motion artifacts by an adaptive filter based on the estimated motion artifact by using a blue LED light. To evaluate the performance of the proposed method experimentally, We did design a novel sensor consisted of blue/red LEDs and photo-sensor and implemented, and then rejected the motion artifacts by using an adaptive filter and the implemented sensor. In the results of the experiments, it is shown that the proposed sensor device and signal processing can reconstruct the PPG signal despite the occurrence of motion artifacts, and also that the SNR was 4.5 times of moving average filter. According to the experimental results, the proposed method can be applied to design a low-cost device.