• The Journal of the Korea Contents Association
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• v.9 no.11
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• pp.309-315
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• 2009

Image reconstruction of Inverse Fourier Transform after Frequency Domain Data is filtered applies to Image signal acquired from MR. There are various kinds of image processing techniques; image preprocessing, image reconstruction, image compression, image restoration image mixture, noise and artifact elimination, and image quality improvement. In this paper, optimum filter applicable to diagnosis in clinic by comparing and analyzing the characteristics of the filter will be explained. Fermi-Dirac filter will improve the image quality better than the previous MR image.

• Annual Conference on Human and Language Technology
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• 2015.10a
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• pp.18-22
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• 2015

기계학습 기반의 의미역 인식에서 주로 어휘, 구문 정보가 자질로 주로 쓰이지만, 의미 정보를 분석하는 의미역 인식은 단어의 의미 정보 또한 매우 주요한 정보이다. 그러나, 기존 연구에서는 의미 정보를 활용할 수 있는 방법이 제한되어 있기 때문에, 소수의 연구만 진행되었다. 본 논문에서는 동형이의어 수준의 의미 애매성 해소 기술, 고유 명사에 대한 개체명 인식 기술, 의미 정보에 기반한 필터링, 유의어 사전을 이용한 클러스터 및 기존 프레임 정보를 확장하는 방법을 제안한다. 제안하는 방법은 기존 연구 대비 뉴스 도메인인 Korean Propbank는 3.14, 위키피디아 문서 기반의 WiseQA 평가셋인 GS 3.0에서는 6.57의 성능 향상을 보였다.

• Proceedings of the Korea Information Processing Society Conference
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• 2013.05a
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• pp.285-288
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• 2013

의료영상에서의 노이즈는 환자 진단에 있어서 막대한 영향을 미치는 영상의 화질을 떨어트림으로써, 진단에 대한 유효성을 낮추게 된다. 특히, 현재 이슈화 되고 있는 저선량 의료영상은 기존의 고선량 의료영상보다 노이즈 레벨이 높으며, 이에 따라서 의료영상에서의 노이즈 제거 기술은 매우 중요한 사안으로 부각되고 있다. 본 논문에서 제시하는 노이즈 제거 기술은 각각의 투영 영상을 여러개의 부대역(sub-band)으로 분해하는 것으로부터 시작한다. 분해된 각각의 부대역 영상은 엣지 검출기를 통하여 엣지 부분과 평탄한 영역으로 구별되어 진다. 검출된 엣지는 0 ~ 1 사이의 값으로 정규화 되며, 퍼지기반의 연산을 통하여 엣지의 확실성을 나타내는 엣지맵으로 변환하게 된다. 이 엣지맵을 통하여 각 부대역 영상의 필터링 정도를 제어하고, 분해된 각 부대역을 결합하는 방식을 취함으로써 영상의 엣지 부분을 최대한 보존하면서 노이즈는 효과적으로 제거하도록 하였다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.113-117
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• 2003

In this paper, have necessity of PACS (Picture Archiving and Communication System) at hospital but hereafter by economical problem PACS apply this to medical treatment image enhancing image quality applying histogram equalization for improvement of light and darkness after reconstruct because make image that pretreatment filtering has wild picture and is processed in wave lets dissolution and wave lets area using weight median filter because could not buy expensive equipment at hospital which introduction is difficulty do inversion and extracted characteristic.

• Journal of the Korean Institute of Telematics and Electronics T
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• v.35T no.2
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• pp.48-55
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• 1998

The main drawback of filtered-X LMS(FXLMS) algorithm for the ANC of broadband noises is its low convergence speed when the filtered reference signals are strongly correlated, producing a large eigenvalue ratio in correlation matrix. This correlation can be caused either by autocorrelation of the signals of the reference sensors, or by coupling between the error path which introduces intercorrelation in the filtered reference signals. In this paper, we introduce a transform domain FXLMS(TD-FXLMS) algorithm that has a high convergence speed by orthogonal transform's decorrelation properties.

• Proceedings of the Korea Society of Information Technology Applications Conference
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• 2002.11a
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• pp.407-411
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• 2002

Ternary sequences are digital codes consisting of discrete values -1, 0, and 1 only. They are advantageous in that the correlation can be carried out using additions only. Also, they feature an ideal circular autocorrelation function, but in channel characterization tasks, the usual requirement is that the linear autocorrelation function be ideal, i.e., a Kronecker delta function. In this article, we consider two approaches to improving their linear autocorrelation or crosscorrelation properties: one is an inverse filtering method with thresholding, and the other is a singular value decomposition (SVD) method. Both methods are simulated under noisy circumstances. The inverse filtering method resulted in an improvement in peak sidelobe level of about 11 dB at an SNR of 30 dB, and the SVD method showed similar performances, albeit more sensitive to noise depending on the singular value selection strategy.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.12B
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• pp.2407-2414
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• 1999

This paper introduces the concept of warped discrete cosine transform (WDCT) and an image compression algorithm based on the WDCT. The proposed WDCT is a cascade connection of a conventional DCT and all-pass filters whose parameters can be adjusted to provide frequency warping. In the proposed image compression scheme, the frequency response of the all-pass filter is controlled by a set of parameters with each parameter for a specified frequency range. For each image block, the best parameter is chosen from the set and is sent to the decoder as a side information along with the result of corresponding WDCT computation. For actual implementation, the combination of the all-pass IIR filters and the DCT can be viewed as a cascade of a warping matrix and the DCT matrix, or as a filter bank which is obtained by warping the frequency response of the DCT filter bank. Hence, the WDCT can be implemented by a single matrix computation like the DCT. The WDCT based compression, outperforms the DCT based compression, for high bit rate applications and for images with high frequency components.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.3
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• pp.25-31
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• 2003

In this paper, we proposed a novel compact microstrip lowpass filter with an ultra-wide stopband and a sharp cutoff frequency response. To improve the cutoff response of the conventional filter, the coupling effect between two open stubs is used. The slot on the ground with conventional shunt open stubs plays an important role of increasing the stopband at a high frequency. The combination of these structures gives the filter the characteristics of an ultra-wide stopband with a prominent cutoff sharpness. The fabricated microstrip lowpass filter with the size of 19.2 mm $\times$ 14.9 mm has -3 dB cutoff frequency at 1.1 GHz and -20 dB stopband from 1.5 GHz to over 30 GHz. The insertion loss is less than -0.15 dB in the passband.

• The Journal of the Acoustical Society of Korea
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• v.19 no.4
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• pp.52-57
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• 2000

In this paper, we designed the 16 Kbps speech coder that has compatibility with CS-ACELP algorithm(G.729). The speech signal is sampled at rate of 16 KHz, divided into two narrowband signal by QMF filterbank, and decimated to rate of 8 KHz. The lower-band signal is encoded by CS-ACELP and the upper-band signal is encoded by Adaptive Transform Coding(ATC) algorithm. At the receiver, two band signals are synthesized by decoder of CS-ACELP and ATC, respectively. The reconstructed output is obtained by passing the QMF synthesis bank. The proposed wideband coder is evaluated with ITU-T G.722 coder through the Mean Opinion Score(MOS) test.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.185-188
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• 2000

본 논문은 인간이 양귀로 3차원 공간상의 음의 위치를 파악하는 원리를 이용하여 만들어진 입체음향을 청취자의 안귀에 가장 효과적으로 재생시킴으로써 완벽한 입체음향을 재현하는 장치로 3차원 게임, 가상현실 시뮬레이터, 놀이공원에서 사용될 수 있는 A/V용 입체음향 재생기 구현 기법이다. 본 개발 품은 전방 30도에 스피커를 배치하는 기존 제품에 비해 안정된 정면 음상 정위와 후면 음상 정위가 가능하고, 청취자의 머리 움직임에 따른 입체음향 효과의 저하를 막을 수 있으며, 인지적 특성을 고려한 역 필터링으로 과도한 신호처리로 인한 음질 저하를 개선할 수 있고, 다중 사용자에게 동일한 음질을 골고루 전달할 수 있어 카 오디오나 영화관에서도 사용되어질 수 있다.