• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.10B
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• pp.1360-1366
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• 2001

이 논문에서는, 잡음환경에서의 음성 향상을 위해서 일반적인 주파수 차감법에 기반한 새로운 형태의 방법을 제안한다. 기존의 방법들이 각각의 주파수 성분에 대해 잡음 및 음성스펙트럼을 추정하는데 비해, 본 논문에서는 주파수 대역을 여러 개의 서브밴드로 대역을 나누어 각각의 서브밴드에 대해서 잡음 및 음성의 스펙트럼을 추정한다. 본 논문에서는 잡음 스펙트럼을 추정하기 위하여 최소추적(Minima Tracking) 방법을 선택하였고, 필터링 방법으로는 스펙트럼 차감법에 기반한 Mel-Scaled 필터뱅크를 이용한 새로운 방법을 제안하였다. 모의실험결과, 기존의 방법들에 비해 음성구간에서의 SNR의 향상정도는 입력 SNR이 -10∼4dB의 범위에서 향상된 결과를 얻었다. 또한 전 구간에 대해서도 다른 알고리즘들 보다 향상된 결과를 얻었다.

• 한국지구과학회:학술대회논문집
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• 2005.09a
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• pp.400-407
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• 2005

웨이브렛 변환(Wavelet Transform)은 시간영역과 주파수영역에서 동시에 분석이 가능하고 불연속적인 자료를 분석하는데 유리하기 때문에 그동안 영상을 처리하고 분석하는데 널리 이용되어 왔다. Discrete Wavelet Transform(DWT)는 주어진 영상에서 특성 정보는 유지하면서 다른 여러 종류의 계수로 분해 할 수 있게 해주기 때문에, 계수에 임계치를 적용해 고주파 성분을 제거하면 잡음을 줄일 수 있다. Stationary Wavelet Transform(SWT)는 DWT에서 다운샘플링에 의해 발생하는 문제점을 해결하기 위한 변환방법으로 잡음제거에 DWT보다 효과적이라고 알려져 있다. 이 연구에서는 DWT와 SWT에 의한 필터링을 광학영상과 레이더 영상에 적용하여 보고, 기존의 필터링 기법과 그 결과를 비교하였다. 그 결과 SWT에 의한 방법이 경계성분은 보존하면서 잡음을 가장 효과적으로 줄일 수 있었다.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.9
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• pp.35-43
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• 2002

In this paper, microstrip ring with a narrow gap is characterized and used as a tunable microwave bandgap (MBG) structure. The center frequency of MBG is mainly determined by the mean circumference of the ring and coincides with odd mode resonance frequency of the ring resonator. The stop band formation by the proposed microstip MBG ring is due to the reflection of electromagnetic waves at the narrow gap introduced in the ring, and the reactive component mounted on the gap makes the stop band vary according to its value. The mounting of capacitor (inductor) is observed to decrease (increase) the center frequency of the stop band. The varactor-mounted microstrip MBG ring is expected to be useful in microwave switches and microwave amplifier circuits.

• Proceedings of the KIEE Conference
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• 2006.07b
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• pp.998-999
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• 2006

포항가속기 연구소의 선형가속기에는 22개의 솔레노이드 전자석, 16개의 사극전자석 그리고 전자빔의 궤도 조절을 위한 16개의 2극 전자석이 있다. 선형가속기의 빔을 저장링으로 공급해주는 빔 전송선에는 22개의 사극전자석과 13개의 빔 궤도 조절용 이극 전자석이 있다. 전자빔의 정밀 제어를 위하여 전자석의 전원장치는 출력전류 분해능은 16bit 이상이고 출력전류의 안정도는 최대출력에 대하여 ${\pm}50ppm$ 이하의 고정밀 고안정도가 요구된다. 이를 위하여 풀-브릿지 4상한 DC/DC 컨버터를 이용한 전자석전원장치를 개발 하였다. 전원장치의 입력전압 직류 40V이고 출력전류는 단방향 전원장치는 최대 50A/50V 이고 양방향 전원장치는 ${\pm}20A/20V$이다. 스위칭 주파수는 50 kHz이다. 전원장치의 출력부에 필터가 없으면 출력전류에는 스윗칭과 관련된 주파수 성분이 포함 되고 전자빔은 이들 주파수 성분에 대하여 영향을 받게 된다. 이러한 이유로 출력 필터의 cut-off 주파수는 5 kHz 이하가 되어야 한다. 본 논문에서는 고정밀 고안정 스윗칭 전자석전원장치를 위한 출력필터의 설계, 제작 그리고 이를 적용한 전자석전원장치에 대하여 논의 하고자 한다.

• The Journal of the Acoustical Society of Korea
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• v.19 no.8
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• pp.54-62
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• 2000

Adaptive filtering method is one of signal processing area which is frequently used in the case of statistical characteristic change in time-varing situation. The performance of adaptive filter is usually evaluated with complexity of its structure, convergence speed and misadjustment. The structure of adaptive filter must be simple and its speed of adaptation must be fast for real-time implementation. In this paper, we propose chirp discrete cosine transform (CDCT), which has the characteristics of CZT (chrip z-transform) and DCT (discrete cosine transform), and then CDCTLMS (chirp discrete cosine transform LMS) using the above mentioned algorithm for the improvement of its speed of adaptation. Using loaming curve, we prove that the proposed method is superior to the conventional US (normalized LMS) algorithm and DCTLMS (discrete cosine transform LMS) algorithm. Also, we show the real application for the ultrasonic signal processing.

• The Journal of the Acoustical Society of Korea
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• v.7 no.2
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• pp.39-50
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• 1988

It is proposed to use a pair of frequency-domain adaptive digital filters to estimate the magnitude squared coherence (MSC) functions of two signals. Such a method requires less computations than the LMS-MSC algorithm in which the least mean square (LMS) algorithm is applied in the time domain to compute the coefficients of a pair of adaptive digital filters. The frequency-domain adaptive digital filtering algorithms considered in this paper include the constrained frequency domain LMS (CFLMS) and the unconstrained frequency domain LMS (UFLMS) algorithms. The performance of the proposed methods are compared with those of the LMS-MSC algorithm.

• The Journal of the Acoustical Society of Korea
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• v.15 no.1
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• pp.46-53
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• 1996

Adaptive LMS algorithm has been used in many application areas due to its low complexity. In this paper input signal is transformed into the subbands with arbitrary bandwidth. In each subbands the dynamic range can be reduced, so that the independent filtering in each subbands has faster convergence rate than the full band system. The DCT transform domain LMS adaptive filtering has the whitening effect of input signal at each bands. This leads the convergence rate to very high speed owing to the decrease of eigen value spread Finally, the filtered signals in each subbands are synthesized for the output signal to have full frequency components. In this procedure wavelet filter bank guarantees the perfect reconstruction of signal without any interspectra interference. In simulation for the case of speech signal added additive white gaussian noise, the suggested algorithm shows better performance than that of conventional NLMS algorithm at high SNR.

• Korean Journal of Optics and Photonics
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• v.16 no.5
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• pp.469-475
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• 2005

An integrated photonic microwave bandpass filter has been proposed and demonstrated incorporating a coherently coupled microring resonator in low-loss polymers. The proposed device may feature compact site, simple structure, tuning via the thermooptic and electrooptic effect, and flexible integration with other electrical and optical devices. The resonator was designed to have an extremely small bandwidth so that it could be used to selectively pass the optical signal carrying the microwave signal to attain efficient bandpass filtering. We made and tested two resonators with a single ring and double rings, and performed a theoretical fitting of their measured transfer curves to predict the performance of the microwave filters based on them precisely. It was found that as the number of the rings used for the resonator increases, the bandwidth gets smaller, the rolloff sharper, and the band rejection higher. Finally our filter exhibited the center frequency of 10GHz, the 3-dB bandwidth of 1.0GHz, the corresponding quality (Q) factor of 10, and the rejection outside of the passband of more than 25dB.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.12
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• pp.2671-2676
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• 2009

Empirical mode decomposition (EMD) is new time-frequency analysis method to decompose the signal adaptively and efficiently. The key idea of EMD is to decompose the signal into a set of functions defined by the signal itself, named Intrinsic Mode Functions (IMFs), which preserve the inherent properties of the original signal. Since the decomposition is based on the local time scale of the signal, it is not only applicable to nonlinear and non-stationary processes but also useful in biomedical signals like electrocardiogram (ECG). Traditional low-pass filter uses fourier transform to analysis signal in frequency domain, but EMD is filtered to maintain signal properties in time domain. This paper performed signal decomposition and filtering for noisy ECGs using EMD method. The proposed method is presented and compared with traditional low-pass filter by two performance indices. Our results show effectiveness for enhancement of the noisy ECG waveforms.

• Proceedings of the Korea Information Processing Society Conference
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• 2017.04a
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• pp.960-962
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• 2017

This thesis suggests method that How sound sources used to simulation that can be used to military training and education divide each frequency and each bandwidth filtering method. method for frequency dividing and denoising are suggested into Wavelet analysis. And We materialize authoring tool about filtering that design for wavelet job.