• Journal of the Institute of Convergence Signal Processing
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• v.20 no.1
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• pp.45-50
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• 2019

When PPG signal is used in mobile healthcare devices, the accuracy of the measured heartbeat decreases from the influence by the movement of the user. The reason is that the frequency band of the noise overlaps the frequency band of the PPG signal. In order to remove these same noises, the methods using frequency analysis method or application of acceleration sensor have been investigated and showed excellent performance. However, in applying these methods to low-cost healthcare devices, it is difficult to apply these methods because of much processing time and sensor's cost. In order to solve these problems, this study proposed the filter design method using an adaptive notch filter and the fuzzy inference system to extract more accurate heart rate in real time and evaluated its performance. As results, it showed better results than the other methods. Based on the results, when applying the proposed method to design the mobile healthcare device, it is possible to measure the heartbeat more accurately in real time.

• Journal of Korea Multimedia Society
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• v.10 no.8
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• pp.1004-1016
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• 2007

In this paper we proposed a Weight-Update-Control Adaptive Noise Canceller which improves speech when environmental noise is stationary and it is hard to acquire a reference signal. Adaptive Noise Canceller(ANC) needs a reference signal, but it is not easy to measure pure noise without voice for reference in factory. Because there are mixed various mechanical noise and workers' voice. Therefore ANC is not suitable to reduce background noise. So we proposed the method that uses an arbitrary constant as an input signal and inputs microphone signal to the reference signal. The noise is eliminated using updated weights in non-speech range. In speech range the weight is fixed and the modified voice is acquired then voice is restored through transversal filter. The proposed method is based on facts that the factory noise is stationary and the noise is not changed in short conversation range. As a result of simulation using MATLAB, we confirmed that the proposed method is effective for reducing factory noise and has high signal to noise ratio(SNR).

• The Journal of Korean Institute of Communications and Information Sciences
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• v.42 no.4
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• pp.819-830
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• 2017

An FM radio based PCL system is a passive radar technique for detecting the multiple moving targets from FM radio signals and tracking the trajectories of the targets by calculating the cross-correlation function of direct-path signal and target echo signals. However, the interference signals are received from a surveillance channel, which is designed to receive the target echo signals. Because of this problem, the target echo signals are masked by the strong interference signals and this makes it difficult to detect the true targets from the cross-correlation function. Adaptive filters are known as effective methods for suppressing the interference signals but there is a problem to present their accurate performances in the PCL system because many literatures used the cross-correlation function and the ratio of input and output power as a measure of the performance analysis. In this paper, a performance analysis method is proposed to evaluate the performance of interference cancellation algorithms. By using the property that each component of the filter weight vector is adjusted to suppress the specific interference signal, a performance measure of the interference signal suppression is defined by a function of adaptive filter weights. Based on the proposed method, we compare the performance of the adaptive filters used in the PCL system. Simulation results show that the proposed method can be very effective for evaluating the performance of interference cancellation algorithms.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2016.11a
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• pp.163-164
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• 2016

이전까지 많은 연구자들은 적응 신호처리(Adaptive Signal Process)를 이용한 잡음 제거 방법을 연구해 왔다. 그러나, 최근 발전하고 있는 멀티콥터는 프로펠러 모터의 RPM(Revolution Per Minute)이 실시간으로 변하기 때문에 적응 신호처리를 이용하여도 깔끔한 결과를 얻어 내기가 어렵다는 한계가 존재한다. 또한, 특정 주파수를 기준으로 형성되는 고조파(Harmonics)는 적응 알고리즘인 (N)LMS 를 이용한 예측에서 오차를 발생시키는 문제를 발생시킨다. 따라서, 본 논문에서는 멀티콥터를 이용한 음향 취득에 대한 소음 저감 방법으로 회전 속도계(Tachometer), 콤 필터(Comb Filter), NLMS 알고리즘(Normalized Least Mean Square Algorithm)을 이용한 방법을 제안한다.

• The Transactions of the Korean Institute of Power Electronics
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• v.9 no.6
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• pp.592-601
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• 2004

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 signal 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 application feasibility was evaluated by building and experimenting a single-phase active power filter based on the proposed reference generator, which was implemented in the DSP(digital signal processor) TMS320C31. Both simulation and experimental results confirm that the proposed reference signal generator can be utilized for the active power filter.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.179-182
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• 2004

수중음향통신을 위한 빔형성 알고리듬은 수중음향통신 신호가 상대대역폭(대역폭에 대한 반송주파수 비)이 큰 광대역 신호 특성을 고려해야한다. 우리는 본 논문에서 위상동기식 디지털 수중음향통신을 위한 협대역 신호 가정을 하지 않은 기저대역 배열 신호 모델을 유도하고 RLS 알고리듬을 이용하여 가중치를 갱신하는 FTR 필터 구조의 광대역 적응 빔형성 알고리듬에 대해서 전산 모의실험을 통하여 FIR 필터 길이, tap간 간격 등과 같은 최적의 FIR 필터 파라미터를 제시하였다.

• The Journal of the Acoustical Society of Korea
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• v.19 no.8
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• pp.35-46
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• 2000

In this paper, we propose a subband acoustic echo canceller using the weighted Overlap-add adaptive filter bank to prevent the decrease of convergence speed in full-band US processing, and make it possible to realize the adaptive filter in block-parallel processing, this paper introduces the weighted overlap-add technique for subband echo canceller. Moreover, we propose a new double-talk detector which employs dedicated filter in addition to the energy comparison method simultaneously. The computer simulation results show that the performance of the proposed subband adaptive echo canceller double-talk detection

• The Journal of the Korea institute of electronic communication sciences
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• v.4 no.1
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• pp.66-69
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• 2009

An adaptive echo canceller is necessary for an application such as a speakerphone, 3G image telephony and VoIP service system. These echo cancellers need to have many taps for filtering echo signals. Many taps cause computation data to increase and convergence speed to be low. To overcome these problems, An adaptive echo canceller with the advanced convergence speed is proposed in this paper. To improve the speed, we divide an echo band into subbands and place a subband filter to be adaptive for each subband. Each subband filter recognizes the echo signal as subband echo signals. So, dynamic range of subband is small, the convergence speed is fast. Moreover, as the number of Tap and weight update are estimated in each subband, the implementation complex of a adaptive filter is low.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.6B
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• pp.1203-1210
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• 1999

Coefficients of the adaptive filter are misadjusted by primary signals which are uncorrelated with reference signals of the adaptive filter. In this paper, the normalized sign algorithm is analyzed and compared with the NLMS algorithm by the steady state performance and the transient characteristics when target signals are included in primary signals. The excess mean square error of the NLMS algorithm is proportional to the power of target signals. That of normalized sign algorithm is proportional to the square root of the target signal power. However, the convergence speed of the normalized sign algorithm is slower than that of NLMS algorithm. In this paper, it is shown that theoretical analysis of the steady state performance and the transient characteristics are well consisted with the results of computer simulation.

• Investigative Magnetic Resonance Imaging
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• v.4 no.2
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• pp.120-127
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• 2000

Adaptive template filtering has been proposed recently for enhancement of signal-to-noise ratio without loss of resolution. In the adaptive template filtering, an optimal template among multiple templates is selected, then linear least square error filtering based on the template is applied in vowel by vowel basis. In some magnetic resonance imaging, where the distribution of gray level has relatively small dynamic range, e.g., $T_1$ imaging, however, artificial stair-like artifact is observed at near edges. This is partially due to the edge enhancement effect in such yokels that contain multiple compounds at the boundaries of tissues. The gray levels of these yokels become similar gray levels of near dominant vowels that contain single compound by the adaptive filtering, which enlarges edge discontinuities. In this paper, we propose a technique to eliminate such artifact by identifying those yokels that contain multiple compounds and assigning the largest template for them. Filtered images with the proposed technique show substantial visual enhancement at the edges without degradation of peak signal-to-noise ratio compared to the original adaptive template filtering for both magnetic resonance images and phantom images.