• Proceedings of the IEEK Conference
• /
• 2001.09a
• /
• pp.561-564
• /
• 2001

잡음에 의해 훼손된 영상 신호를 복원할 때 쓰이는 Wiener filter는 국부영역의 잡음 분산과 신호 분산을 가지고 적응적으로 필터의 파라미터를 조절한다. 그러나 기존의 Wiener filter는 고정된 필터 마스크를 사용함으로써, 평탄 영역의 잡음을 크게 제거하면, 에지 부분의 잡음이 살고, 에지 부분의 잡음을 제거하면, 평탄영역의 잡음이 사는 특성이 있다. 본 논문은 Kirsh mask로 에지와 그 방향성을 판별한 후, 에지 부분의 잡음을 제거하면서 평탄 영역의 잡음도 동시에 제거하기 위해 가변적인 필터 마스크를 사용했으며, 잡음에 의해 훼손된 방향성 정보를 살러 주기위해 필터 마tm크와 훼손된 영상 이미지에 방향성 정보를 추가했다. 제안된 방법으로 실험한 결과 주관적 비교에서 에피 부분이 잡음을 제거하고 방향성을 살렸으며, PSNR을 이용한 객관적 비교에서도 기존알고리즘보다 개선된 성능을 보였다.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.29 no.2C
• /
• pp.222-228
• /
• 2004

This paper describes a new adaptive algorithm which can reduce the required computation quantities in the adaptive filter. The proposed adaptive algorithm uses only the signs of the normalized input signal rather than the input signals when coefficients of the filter are adapted. By doing so, there is no need for the multiplications and divisions which are mostly responsible for the computation quantities. To analyze the convergence characteristics of the proposed algorithm, the condition and speed of the convergence are derived mathematically. Also, we simulate an echo canceller adopting this algorithm and compare the performances of convergence for this algorithm with the ones for the other algorithm. As the results of simulations, it is proved that the echo canceller adopting this algorithm shows almost the same performances of convergence as the echo canceller adopting the SIA algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.27 no.7A
• /
• pp.651-658
• /
• 2002

Although least mean square(LMS) algorithm is known to one of the most popular algorithm in adaptive signal processing because of the simplicity and the small computation, the choice of the step size reflects a tradeoff between the misadjustment and the speed of adaptation. In this paper, we present a new variable step size LMS algorithm, so-called ECLMS(Estimated correlation LMS), using the correlation between reference input and error signal of adaptive filter. The proposed algorithm updates each weight of filter by different step size at same sample time. We applied this algorithm to adaptive multiple-notch filter. Simulation results are presented to compare the performance of the proposed algorithm with the usual LMS algorithm and another variable step algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.25 no.8A
• /
• pp.1198-1205
• /
• 2000

In this paper, we introduce a modified interference cancellation scheme to overcome MAI in DS-CDMA. Among ICs(Interference Cancellers), PIC(Parallel IC) requires the more complexity, and SIC(Successive IC) faces the problems of the long delay time. Most of all, the adaptive detector achieves the good BER performance using the adaptive Inter conducted iteration algorithm. So it requires many iterations. To resolve the problems of them, we propose an improved adaptive detector that the received signal removed MAI through the sorting scheme and the cancellation method are fed into the adaptive filter. Because the improved input signal is fed into the adaptive filter, it has the same BER performance only using smaller iterations than the conventional adaptive detector, and the proposed detector having adaptive filter requires less complexity than the other detectors.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.15 no.3
• /
• pp.99-105
• /
• 2015

This paper deals with the mMMA (modified MMA) which possible to improving the convergence speed that employing the serial connecting form of two stage digital filter instead of signal filter of MMA adaptive equalizer without applying the variable step size for compensates the intersymbol interference by channel distortion in the nonconstant modulus signal. The adaptive equalizer can be implemented by signal digital filter using the finite order tap delay line. In this paper, the equalizer is implemented by the two stage serial form and the filter coefficient are updated by the error signal using the same algorithm of MMA in each stage. The fast convergence speed is determined in the first stage, and the residual isi left at the output of first stage output is minimized in the second stage filter. The same digital filter length was considered in single stage and two stage system and the performance of these systems were compared. The performance index includes the output signal constellation, the residual isi and maximum distortion, MSE that is measure of the convergence characteristics, the SER. As a result of computer simulation, mMMA that has a FIR structure of two stage, has more good performance in every performance index except the constellation diagram due to equalization noise and improves the convergence speed about 1.5~1.8 time than the present MMA that has a FIR structure of single stage.

• Proceedings of the KIEE Conference
• /
• 2008.04a
• /
• pp.131-132
• /
• 2008

오실로메트릭 방법으로 휴대용 혈압 측정 시 빈번하게 발생하는 잡음에 의한 오실레이션 신호의 왜곡을 줄이기 위해 새로운 잡음 제거 알고리즘을 제안하였다. 제안된 잡음 제거 알고리즘은 선형 예측기 구조 기반의 적응 필터를 이용한다. 제안된 잡음 제거알고리즘의 성능을 평가하기 위해 왜곡된 오실레이션 신호에 선형보간법을 사용하는 기존의 방법과 적응 필터를 사용하는 제안된 방법을 적용하여 잡음 제거 성능을 비교하였다. 제안된 방법은 잡음이 오실레이션과 중첩되어 나타난 경우에 기존의 방법과 달리 잡음에 강인한 특징을 보여주었다.

• Proceedings of the IEEK Conference
• /
• 2005.11a
• /
• pp.465-468
• /
• 2005

심전도 신호의 잡음 신호는 일정한 주파수대역에 존재하지 않고 측정자의 신체 및 환경조건에 따라서 잡음의 종류와 정도가 다르다. 따라서 기존의 고정 주파수 특성을 갖고 있는 필터로는 효율적인 잡음 제거가 불가능하다. 그래서 본 논문에서는 상황인식을 통해 잡음의 형태를 파악하여 적응적으로 필터를 재구성하는 적응적 잡음제거기술을 제안한다.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
• /
• v.2 no.3
• /
• pp.71-78
• /
• 2009

In this study, estimation of heart rate from measured PPG signal is proposed. PPG signal is to be measured blood flow in a blood vessel effected by systole and diastole. PPG sianl has single frequency so that PPG frequency can be tracked by 2nd IIR adaptive notch filter. PPG frequency is obtained continually from updating filter coefficient throughout adaptive algorithm and then the heart rate of human is approximately estimated.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.12 no.5
• /
• pp.887-892
• /
• 2008

This paper is on a new adaptive algorithm which can cancel the acoustic feedback signals in the digital hearing aids. The proposed algorithm uses the normalized LMS algorithm with decorrelators. By doing so, it can be reduced the autocorrelation for the voice signals. To analyze the convergence characteristics of the proposed algorithm, the simulations were carried out about various input signals. And we had compared the performances of convergence for this algorithm with the ones for the NLMS algorithm. As the results of simulations, it is proved that the feedback canceller adopting this algorithm shows about 5-10 dB more high SNR than the NLMS algorithm for the colored inputs.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.17 no.3
• /
• pp.691-697
• /
• 2013

Recently, u-healthcare device is developed and commercialized for healthcare management and emergency medical. The kinds of the measurable biomedical signals on the device are electrocardiogram, skin temperature, pulse oxygen, heart rate, respiration, etc. Specially, the photoplethysmograph(PPG) signal of these signals is the important signal in measuring oxygen, heart rate and peripheral vascular compliance. The accuracy of PPG signal reduce from influence of the motion artifacts that generated from the movements of user or patient. Therefore, this study suggests a new method to remove the motion artifact that is using optical power modulation and ICA(Independent Component Analysis). For analyzing the proposed method, we used variety of noises made by artificially. In the results of experiments, the proposed method showed good performances than an adaptive filter.