• Journal of the Institute of Electronics Engineers of Korea SP
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• v.45 no.3
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• pp.93-102
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• 2008

In this paper, we propose a rate control algorithm for constant quality real time video coding. To achieve constant quality, previous algorithm exploit mean absolute of difference(MAD) as measure of frame complexity. However, if scene is abruptly changed or if quantization parameter is not constant, encoder produces various output bits with same MAD. Therefore we know that MAD does not appropriately reflect characteristic of frame. To solve this problem, we exploit model parameter as measure of frame complexity. Because model parameter means slope between output bits and MAD, it reflects correctly complexity of frame. And because previous model, R-MAD model, is not considered quantization parameter, as quantization parameter increases or decreases, model parameter of frame also vary. So model parameter obtained using previous model cannot reflect internal characteristic of video. We solve this problem using proposed model, which is considered quantization parameter. Experiment results show that our algorithm provide better performance, in terms of quality smoothness than previous algorithm. Especially, when scene is abruptly changed, our algorithm alleviates quality drop.

• The Journal of the Acoustical Society of Korea
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• v.17 no.4
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• pp.45-53
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• 1998

본 논문에서는 선스펙트럼 주파수(LSF) 파라미터를 위한 제한된 시간적 분해법을 제안한다. LSF 파라미터는 인접 차수에 대해 의존적이고, 차수간 순차성이 있으나, 기존의 시간적 분해법은 이러한 성질을 보존하지 못한다. 즉, 추정된 사건 벡터가 더 이상 LSF 파 라미터로서 해석되지 못하는 문제가 있다. 이를 해결하기 위하여, 본 논문에서는 사건 함수 간에 새로운 제약을 두어, 추정된 사건 벡터가 LSF 파라미터의 성질을 유지하도록 한다. 결 과적으로 제안된 방법을 이용하여 구해진 사건 벡터는 LSF 파라미터와 동일한 방법을 적용 하여 효과적으로 양자화될 수 있고, 실험 결과 평균 752bps의 전송률로 투명한 양자화를 수 행할 수 있었다.

• 한국HCI학회:학술대회논문집
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• 2009.02a
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• pp.443-448
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• 2009

We introduce an interactive image cartooning system based on personal subjectivity. To effectively tune various parameters needed to adjust image style, our system uses interactive genetic algorithm. By selecting several pre-stylized image samples using simple user interface, the user can easily achieve the desired result without having any signal-processing knowledge. Our system reduces the parameter tuning time drastically compared to the conventional system, which involves manual parameter setting.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.4
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• pp.73-80
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• 2013

In this paper, an experimental circuit model for an accurate high-frequency characterization of transmission line is proposed. Inherent resonance effects during measurements make it difficult to determine characteristic impedance and propagation constant at the resonance frequencies corresponding to the line length. Thus, resonance-effect-free transmission line parameter determination technique based on the physical insight and theory is proposed. Then, by using the parameters high-frequency circuit model is proposed for high-speed signal propagation characterization. The proposed frequency-variant transmission line model is verified with measurement and it can be usefully exploited in high-speed signal propagation characterization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.2C
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• pp.200-208
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• 2009

A macroblock layer rate control for H.264/AVC has the problem that allocated target bits for current frame occasionally are exhausted too fast due to inadequate quantization parameter assignment. In this case, the maximum permissible quantization parameter is used to encode for remaining macroblocks and it leads to degradation of the visual quality. In addition, the header bits length estimation algorithm used for quantization parameter assignment takes the average header bits length for the encoded macroblocks of the previous frame and the current frame. Therefore, it generates a big mismatch between the actually generated header bits length and the estimated header bits length. In this paper, we propose adaptive quantization parameter decision method to prevent early exhausting target bits during encoding the current frame by considering the number of macroblocks that have negative targets bits in previous frame and the improved header bits length estimation scheme for accurate quantization parameter decision.

• 한국항공운항학회:학술대회논문집
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• 2016.05a
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• pp.26-29
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• 2016

본 논문에서는 멀티로터 시스템의 모델리을 위한 방법으로 파라미터 추정법을 제시하였으며 이를 위해 실제 비행데이터를 이용한다. 파라미터 추정법으로는 예측오차 기법과 순화최소자승법이 사용되었고 그 결과를 나타내었다.

• Journal of KIISE:Software and Applications
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• v.34 no.9
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• pp.797-803
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• 2007

Adaptive binarization is very important for the camera-based document recognition. This paper proposes a binarization method which can effectively adapt to the variation of image Qualify. Firstly, it analyzes the effect of binarization parameters to the result and proposes a method to measure the image quality. Then, it statistically analyzes the relationship between the image quality and the binarization parameter. Finally, it proposes a binarization method that automatically adapts to the quality of the input image, using the analysis result. The experiment results show that there is a meaningful relationship between the image quality and the binarization parameter, and therefore, the proposed method can effectively adapt to the variation of image quality.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2002.12a
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• pp.391-394
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• 2002

진화 연산(Genetic Algorithm)은 최적화 분야에서 사용되는 강력하면서도 일반적인 방법이다. 이러한 진화 연산의 일반성은 진화 연산에서 사용되는 기본 연산자들이 문제에 대한 정보를 필요로 하지 않는 것에 기인하고 있기에, 실제 구현시에는 여러 파라미터들을 문제에 맞게 정해 줌으로써 성능 향상을 죄할 수 있다. 이러한 파라미터의 조절은 보통 시행착오를 거쳐 행해지나, 실행시에 동적으로 파라미터를 학습하는 적응적 진화 연산도 연구되어 왔다. 본 논문에서는 진화 연산에서의 파라미터 학습 과정을 강화 학습 과정으로 공식화하고 강화 학습을 사용한 적응적 진화 연산 구현을 제안한다.

• The Journal of the Acoustical Society of Korea
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• v.10 no.4
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• pp.5-11
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• 1991

This paper suggests a segmentation method of speech signal into phonemic units. The suggested segmentation system is speaker-independent and performed without anyprior information of speech signal. In segmentation process, we first divide input speech signal into purevoiced region and not pure voiced speech regions. After then we apply the second algorithm which segments each region into the detailed phonemic units by using the voiced detection parameters, i.e., the time variation of 0th LPC cepstrum coefficient parameter and the ZCR parameter. Types of speech, used to prove the availability of segmentation algorithm suggested in this paper, are the vocabulary composed of isolated words and continuous words. According to the experiments, the successful segmentation rate for 507 phonemic units involved in the total vocabulary is 91.7%.

• Journal of Broadcast Engineering
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• v.21 no.3
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• pp.369-379
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• 2016

This paper deals with an efficient method for skipping inter-channel phase differences (IPD) in the MPEG surround of the unified speech and audio coding (USAC). Based on the psycho-acoustic sensitivity on the IPD, we estimate a threshold on IPD, below which we can not notice degradation in spatial cue. We propose an IPD skip method, in which any IPDs within the threshold are set to zero and are not transmitted. The proposed IPD skip method gives about 38% savings in terms of bit amount for IPD. Nevertheless, in the MUSHRA test, the proposed method does not show any noticeable degradation in the decoded audio quality.