• Journal of the Institute of Electronics Engineers of Korea SP
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• v.38 no.4
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• pp.57-57
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• 2001

본 논문은 영상 시퀸스로부터 이미지 모자이킹의 구성을 위한 효율적인 알고리즘을 기술한다. 영상의 기하학적인 특징을 이용하거나 비선형 방정식을 풀었던 기존의 알고리즘과는 달리, 제안한 알고리즘은 4개의 유사특징점을 이용해 영상간 사영 변환식의 8개 파라미터를 직접 계산한다. 본 논문에서 정의된 유사특징점은 영상의 그레이레벨의 분산을 기반으로 하고, 두 영상의 중첩 영역에서만 결정된다. 또한 선택된 4개의 유사특징점에 대한 대응점 검출을 위해 카메라 이동 및 조명 변화에 의한 영상의 변화를 고려한 블록 정합 알고리즘을 적용한다. 제안된 알고리즘은 다양한 영상에 적용하여 그 성능을 평가하였다. 모의 실험 결과는 제안된 알고리즘이 기존의 알고리즘에 비해 계산량을 감소시키면서, 정확한 사영 변환식을 유도하여 모자이킹 영상을 구성하는 것을 보여주고 있다.

• Proceedings of the Korean Information Science Society Conference
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• 2000.04b
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• pp.39-41
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• 2000

공간조인 연산은 지리정보시스템의 연산 중 매우 높은 처리비용을 요구하는 연산이다. DOT 공간 색인 기법은 전통적인 데이터베이스 시스템의 주색인 기법을 적용할 수 있으며, 공간객체의 상호 인접성이 유지되도록 Hilbert 값으로 정렬되어 클러스터링 된다. 이러한 특징을 이용한 DOT공간 조인 알고리즘은 적정한 버퍼크기를 유지하는 경우 잘 알려진 R-tree를 이용한 공간조인 알고리즘에 비해 디스크 액세스면에서 유리한 장점이 있으나, 조인가능영역 산출시 많은 양의 공간변환 연산을 필요로 하므로 전체적인 성능이 만족스럽지 못하다. 본 논문은 DOT 공간조인 알고리즘의 성능을 향상시키기 위하여 이러한 공간변환 연사의 횟수를 최소화시킨 효율적인 여과단계처리 방법을 제시하며, 이를 적용한 DOT공간조인 알고리즘과 R-tree 공간조인 알고리즘의 실행시간을 비교 분석하여 DOT 공간조인 알고리즘이 최대 약 2배까지 우수한 성능을 가지고 있음을 보인다.

• Journal of KIISE:Databases
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• v.33 no.6
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• pp.600-619
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• 2006

According to recent technical advances on sensors and mobile devices, processing of data streams generated by the devices is becoming an important research issue. The data stream of real values obtained at continuous time points is called streaming time-series. Due to the unique features of streaming time-series that are different from those of traditional time-series, similarity matching problem on the streaming time-series should be solved in a new way. In this paper, we propose an efficient algorithm for streaming time- series matching problem that supports normalization transform. While the existing algorithms compare streaming time-series without any transform, the algorithm proposed in the paper compares them after they are normalization-transformed. The normalization transform is useful for finding time-series that have similar fluctuation trends even though they consist of distant element values. The major contributions of this paper are as follows. (1) By using a theorem presented in the context of subsequence matching that supports normalization transform[4], we propose a simple algorithm for solving the problem. (2) For improving search performance, we extend the simple algorithm to use $k\;({\geq}\;1)$ indexes. (3) For a given k, for achieving optimal search performance of the extended algorithm, we present an approximation method for choosing k window sizes to construct k indexes. (4) Based on the notion of continuity[8] on streaming time-series, we further extend our algorithm so that it can simultaneously obtain the search results for $m\;({\geq}\;1)$ time points from present $t_0$ to a time point $(t_0+m-1)$ in the near future by retrieving the index only once. (5) Through a series of experiments, we compare search performances of the algorithms proposed in this paper, and show their performance trends according to k and m values. To the best of our knowledge, since there has been no algorithm that solves the same problem presented in this paper, we compare search performances of our algorithms with the sequential scan algorithm. The experiment result showed that our algorithms outperformed the sequential scan algorithm by up to 13.2 times. The performances of our algorithms should be more improved, as k is increased.

• Journal of KIISE:Computing Practices and Letters
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• v.8 no.3
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• pp.289-294
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• 2002

A valid speech-sound block can be classified to provide important information for speech recognition. The classification of the speech-sound block comes from the MRA(multi-resolution analysis) property of the DWT(discrete wavelet transform), which is used to reduce the computational time for the pre-processing of speech recognition. The merging algorithm is proposed to extract valid speech-sounds in terms of position and frequency range. It needs some numerical methods for an adaptive DWT implementation and performs unvoiced/voiced classification and denoising. Since the merging algorithm can decide the processing parameters relating to voices only and is independent of system noises, it is useful for extracting valid speech-sounds. The merging algorithm has an adaptive feature for arbitrary system noises and an excellent denoising SNR(signal-to-nolle ratio).

• Proceedings of the Korea Contents Association Conference
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• 2003.05a
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• pp.214-217
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• 2003

In this paper, we proposed an efficient blind watermarking algorithm using wavelet transform. The proposed algorithm inserts watermark into the high frequency subbands that were transformed by 1-level wavelet transform of original image. Watermark insertion is achieved by exchanging each the corresponding coefficients in the HL, LH, HH subbands according to be inserted watermark value. And watermark is extracted by the relation of wavelet coefficients without original image. Experimental results demonstrate that watermarked image has a good quality not to be able to be perceptible and is robust various attacks such as JEPG lossy compression, clipping and sharpening.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2018.10a
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• pp.180-181
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• 2018

JPEG is a most widely used standard image compression technology. This research introduces the JPEG image compression algorithm and describes each step in the compression and decompression. Image compression is the application of data compression on digital images. The DCT (discrete cosine transform) is a technique for converting a time domain to a frequency domain. First, the image is divided into 8 by 8 pixel blocks. Second, working from top to bottom left to right, the DCT is applied to each block. Third, each block is compressed through quantization. Fourth, the array of compressed blocks that make up the image is stored in a greatly reduced amount of space. Finally if desired, the image is reconstructed through decompression, a process using IDCT (inverse discrete cosine transform).

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.9
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• pp.2203-2208
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• 2014

The Hough transform (HT), that is a typical algorithm for detecting lines in images, needs considerable computational costs and easily detects pseudo-lines on the real world images, because of the large amount of features generated by their complex background or noise. This paper proposes an improved HT that add a preprocessing to estimate the validity of features to the conventional HT. The feature estimation can remove a lot of inessential features for the line detection using a pattern of $3{\times}3$ block features. Experiments using various images show that the proposed algorithm saves computational costs by removing 14%~58% of features depending on images and besides it is superior to the conventional HT in valid line detection.

• The KIPS Transactions:PartB
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• v.17B no.2
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• pp.169-176
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• 2010

This paper proposes color transformation of images based on user's preference. Traditional color transformation method transforms only hue based on existing templates that define range of harmonious hue. It does not change saturation and intensity. Users would appreciate the resulting images that adjusted unnatural hue of images. Since color is closely related to peoples' emotion, we can enhance interaction of emotion-based contents and technologies. Therefore, in this paper, we define the range of color of each emotion for the transformation of color and perform the transformation of hue, saturation and intensity. However, the relationship of color and emotion depends on the culture and environment. To reflect these characteristics in color transformation, we propose the transformation of color that is based on user's preference and as a result, people would be more satisfied. We adopt interactive genetic algorithm to learn about user's preference. We surveyed the subject to analyze user's satisfaction about transformed images that are based on preference, and we found that people prefer transformed images to original images. Therefore, we conclude that people are more satisfied with the transformation of the templates which reflected user's preference than the one that did not.

• The Journal of the Acoustical Society of Korea
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• v.15 no.2
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• pp.5-11
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• 1996

The purpose of this paper is to propose a method improving the performance of a semi-continuous hidden Markov model(SCHMM) speaker adaptation system which uses Bayesian Parameter reestimation approach. The performance of Bayesian speaker adaptation could be degraded in case that the features of a new speaker are severely different from those of a reference codebook. The excessive codewords of the reference codebook still remain after adaptation proess. which cause confusion in recognition process. To solve such problems, the proposed method uses formant information which is extracted from the cepstral coefficients of the reference codebook and adaptation data. The reference codebook is adapted to represent the formant distribution of a new speaker and it is used for Bayesian speaker adaptation as an initial codebook. The proposed method provides accurate correspondence between reference codebook and adaptation data. It was observed that the excessive codewords were not selected during recognition process. The experimental results showed that the proposed method improved the recognition performance.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.305-308
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• 2000

IP 네트웍을 통해 실시간 비디오를 전송할 때 네트윅 특성을 고려하여 비디오 데이터를 부호하면 패킷 손실로 인한 품질 열화를 최소화하여 더 좋은 품질의 비디오를 얻을 수 있다. 이를 위해 현재 네트윅의 정보를 빠르고 정확하게 얻어내는 메커니즘과 부호화 변수를 네트윅 상황에 적응적으로 조절하여 패킷 손실에 강인한 압축 및 전송 메커니즘의 두 가지 기능이 요구된다. 첫번째 메커니즘은 RTP(Real Time Transport Protocol)을 통해 구현될 수 있으며, 두 번째 메커니즘을 위해 본 연구에서는 다중기술 변환부호화(Multiple Description Transform Coding) 기법을 적용한 비디오 부호화 알고리즘을 제안한다. RTP에서 제공하는 RTCP(Real Time Control Protocol) 정보를 이용하여 현재 네트웍 정보를 얻을 수 있으며, 다중기술 변환부호화 기법을 이용하여 현재의 패킷 손실률에서 최적의 품질을 보장하도록 부호화 변수를 조절할 수 있다. 본 논문에서는 다중기술 변환부호화 기법을 비디오 부호화에 적용하여 순수 비디오 정보에 추가되는 잉여 정보량과 패킷 손실에 대한 강인성 사이의 관계를 도출함으로써 다중기술 변환부호화 기법이 네트웍 적응적 부호화 방식에 적합한 방식임을 제시한다.