• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1991.10a
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• pp.256-258
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• 1991

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2015.11a
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• pp.13-15
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• 2015

본 논문에서는 하나의 중계기가 하나의 송신기로부터 신호를 받아 증폭한 후에 수신기로 재전송하는 방식으로 송신기와 수신기 사이의 통신을 돕는 네트워크를 고려하였다. 중계기가 독자적인 에너지원이 없는 경우 일정한 양을 에너지를 확보하여 중계에 사용하기 위해서 송신기로부터의 신호를 에너지로 하베스팅하는 모델을 생각하였다. 또한, 나아가 현재의 다양한 무선통신 네트워크에서 사용중인 적응변조를 적용하여 항상 일정이상의 비트오율을 만족할 수 있는 더욱 현실적인 모델이 되도록 하였다. 이러한 모델에서 정해진 만큼의 시간을 하베스팅에 사용했을 경우 처리율을 구하였으며, 나아가 그 시간을 최적화하여 유도한 처리율을 최대화하는 문제를 만들었다.

• The KIPS Transactions:PartB
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• v.11B no.2
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• pp.199-206
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• 2004

This paper proposes a hybrid independent component analysis(ICA) of fixed-point(FP) algorithm and robust algorithm. The FP algorithm is applied for improving the analysis speed and performance, and the robust algorithm is applied for preventing performance degradations by means of very small kurtosis and temporal correlations between components. And the adaptive adaptation of temporal correlations has been proposed for solving limits of the conventional robust algorithm dependent on the maximum time delay. The proposed ICA has been applied to the problems for separating the 4-mixed signals of 500 samples and 10-mixed images of $512\times512$pixels, respectively. The experimental results show that the proposed ICA has a characteristics of adaptively adapting the maximum time delay, and has a superior separation performances(speed, rate) to conventional FP-ICA and hybrid ICA of heuristic correlation. Especially, the proposed ICA gives the larger degree of improvement as the problem size increases.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.407-410
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• 2000

본 논문에서는 3차원 웨이블릿 부호화 방식으로 압축된 영상 시퀸스를 정칙화 기반 영상복원 방법으로 후처리하는 알고리듬을 제안한다. 우선, 웨이블릿 압축 시스템을 적절한 영상 열화 시스템으로 모델화한다. 그리고, 시간축에 관하여 프레임 간의 같은 위치에 있는 각 픽셀에 대하여 복원을 수행한다. 그 다음으로 2차원 영상 신호에 대하여 복원을 수행하는데. 즉 웨이블릿 변환 계수 정보를 이용하여 영상 및 시간 정보를 여러 스케일의 에지로 분류한 다음, 에지의 방향에 따른 적응적인 제약조건을 사용한다. 이는 각각의 에지 방향에 적합한 고주파 성분을 유지하고, 신호의 각 특성에 적합한 적응적인 정칙화 매개변수를 적용한다. 마지막으로 시간 축에서의 복원과 그것에 이어지는 적응적인 공간 복원에 대한 실험 결과를 보여준다.

• Journal of Broadcast Engineering
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• v.11 no.2 s.31
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• pp.200-206
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• 2006

This paper presents a preprocessing method to modify the input audio signals of a speech coder to obtain the finally enhanced signals at the decoder. For the purpose, we introduce the noise suppression (NS) scheme and the adaptive gain control (AGC) where an audio input and its coding error are considered as a noisy signal and a noise, respectively. The coding error is suppressed from the input and then the suppressed input is level aligned to the original input by the following AGC operation. Consequently, this preprocessing method makes the spectral energy of the music input redistributed all over the spectral domain so that the preprocessed music can be coded more effectively by the following coder. As an artifact, this procedure needs an additional encoding pass to calculate the coding error. However, it provides a generalized formulation applicable to a lot of existing speech coders. By preference listening tests, it was indicated that the proposed approach produces significant enhancements in the perceived music qualities.

• Journal of the Institute of Convergence Signal Processing
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• v.9 no.4
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• pp.295-303
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• 2008

In general, transform domain adaptive filters show faster convergence speed than the time domain adaptive filters, but the amount of calculation increases dramatically as the filter order increases. This problem can be solved by making use of the subband structure in transform domain adaptive filters. In this paper, to increase the convergence speed on the generalized subband decomposition FIR adaptive filters, a structure of the adaptive filter with subfilter of dyadic sparsity factor in wavelet transform domain is designed. And, in this adaptive filter, the equivalent input in transform domain is derived and, by using the input, the convergence properties for the LMS algorithm is analyzed and evaluated. By using this sub band adaptive filter, the inverse system modeling and the periodic noise canceller were designed, and, by computer simulation, the convergence speeds of the systems on LMS algorithm were compared with that of the subband adaptive filter using DFT(discrete Fourier transform).

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2003.06a
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• pp.6-9
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• 2003

In this paper, we studied on the difference of speaker adaptation according to the phoneme classification for Korean Speech recognition. In order to study of speech adaptation according to the weight of difference of phoneme as recognition unit, we used SCHMM as recognition system. And Speaker adaptation method used in this paper was MAPE(Maximum A Posteriori Probability Estimation), Linear Spectral Estimation. In order to evaluate the performance of these methods, we used 10 Korean isolated numbers as the experimental data. It is possible for the first and the second methods to be carried out unsupervised learning and used in on-line system. And the first method was shown performance improvement over the second method, and hybrid adaptation showed the better recognition results than those which performed each method. And the result of Speaker adaptation using the variable weight according to the phoneme had better than the result using fixed weight.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.9 no.4
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• pp.17-24
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• 2009

This paper deals with the cancellation performance of loop back interference signal in the case of multihop relay of 16-QAM received signal at the USN radio network. For this, it is necessary to the exchange of information with long distance located station by means of the relay function between the node in the USN environment. In the relay node, the loop-back interference signal which the retransmitting signal is feedback to the receiver side due to the antenna of transmitter and receiver are co-used or very colsely located or using the nonlinear device. Due to this signal, the performance of USN system are degraded which are using the limited resource of frequency and power. For improve this, it is necessary to applying the adaptive signal processing algorithm in order to cancellating the unwanted loop-back interference signal at the frontend of receiver in relaying node, we can get the better system and multi hop performance. In the adaptive signal processing, we considered the 16-QAM signal which has a good spectral efficiency, firstly, than, the QR-Array RLS algorithm was used that has a fairly good convergence property and the solving the finite length problem in the H/W implementation. Finaly, we confirmed that the good elimination performanc was confirmed by computer simulation in the learing cuved and received signal constellation compared to the conventional RLS.

• Journal of Korea Society of Industrial Information Systems
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• v.12 no.1
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• pp.39-45
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• 2007

In this paper, we present a new postprocessing algorithm using interpolation and signal adaptive filter according to the each block characteristic which is acquired in block classification process. We applied blocking artifact reduction algorithm for four neighbor low frequency block and ringing artifacts is removed with preserving edges by applying a signal adaptive filter in high frequency block based on edge map. The computer simulation results confirmed a better performance by the proposed method in both the subjective and objective image qualities.

• Journal of the Institute of Convergence Signal Processing
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• v.11 no.2
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• pp.104-111
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• 2010

This paper presents an adaptive image enhancement method using the spatial frequency property in the DCT(discrete cosine transform) compressed domain. The dc coefficients, the illumination components of image, are adjusted to compress the dynamic range of image, and the ac coefficients are modified to enhance the contrast by using the human visual system(HVS) and the spatial frequency property. The ac coefficients are separated into vertical direction, horizontal direction, and mixed spatial frequency components, and adaptively modified to minimize the block artifacts that possibly occur in the image enhancement. The proposed method using dynamic range compression and adaptive contrast enhancement shows the advanced performance without the block artifact compared with existing method.