• Communications for Statistical Applications and Methods
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• v.19 no.2
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• pp.225-235
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• 2012

Noise reduction is an important issue in the field of image processing because image noise worsens the quality of the input image. The basic difficulty is that the noise and the signal are not easy to distinguish. Simple moothing is one of the most basic and important procedures to remove the noise, however, it does not consider the level of noise. This method effectively reduces the noise but the feature area is simultaneously blurred. This paper considers the block approach to detect noise and image features of the input image so that noise reduction could be adaptively applied. Simulation results show that the proposed algorithm improves the overall quality of the image by removing the noise according to the noise level.

• Journal of the Institute of Convergence Signal Processing
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• v.5 no.2
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• pp.85-90
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• 2004

This paper proposes the wavelet image coding using the significant cluster extraction by morphology and the adaptive quantization. In the conventional MRWD method, the additional seed data takes large potion of the total data bits. The proposed method extracts the significant cluster using morphology to improve the coding efficiency. In addition, the adaptive quantization is proposed to reduce the number of redundant comparative operations which are indispensably occurred in the MRWD quantization. The experimental result shows that the proposed algorithm has the improved coding efficiency and computational cost while preserving superior PSNR

• Journal of the Institute of Convergence Signal Processing
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• v.10 no.3
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• pp.158-163
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• 2009

LMS(Least Mean Square) algorithm requires simple equation and is used widely because of the low complexity. If the convergence speed increase, LMS algorithm has a divergence in case of sharp environment changes. And if a stability increase, the convergence speed becomes slow. This algorithm based on a trade off between fast convergence and system stability. To improve this problem, VSSLMS (Variable Step Size LMS) algorithm was developed. The VSSLMS algorithm improved the convergence speed and performance as adjusting step size using error signal. In this paper, I-VSSLMS algorithm is proposed tor improve the performance of adaptive noise canceller in real-time environments. The proposed algorithm is applied to adaptive noise canceller using TMS320C6713 DSP board and we did simulation by real time. Then we compared performance of each algorithm and demonstrated that proposed algorithm has superior performance.

• Journal of the Institute of Convergence Signal Processing
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• v.3 no.3
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• pp.21-26
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• 2002

This paper presents the new approach of identifying human emotional states such as anger, happiness, normal, sadness, or surprise. This is accomplished by using discrete duration continuous hidden Markov models(DDCHMM). For this, the emotional feature parameters are first defined from input speech signals. In this study, we used prosodic parameters such as pitch signals, energy, and their each derivative, which were then trained by HMM for recognition. Speaker adapted emotional models based on maximum a posteriori(MAP) estimation were also considered for speaker adaptation. As results, the simulation performance showed that the recognition rates of vocal emotion gradually increased with an increase of adaptation sample number.

• Journal of Korea Multimedia Society
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• v.6 no.7
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• pp.1131-1138
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• 2003

Active noise control(ANC) is an approach to noise reduction in which a secondary noise source destructively interferes with the unwanted noise is introduced. Generally, the performance of ANC is determined how well a secondary noise tracks noises. A secondary noise is generated from the cancelling speaker and a error sensor pick up error signal. The transfer function between the cancelling speaker and the error sensor is not flat and distorts secondary noises. Consequently, the performance of ANC is degraded by the transfer function. In this paper, a single sensor ANC which considers the characteristics of the speaker and the error sensor is proposed. To reduce distortion of secondary noises, the transfer function is estimated by adaptive inverse modelling and the primary noises are estimated by Kalman filter. Experimental results show that the proposed single sensor ANC effectively attenuates noises.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.483-486
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• 2004

수중음향채널의 다중경로에 의한 수신신호의 시간확산(Time spread)은 ISI를 유발하여 수중통신시스템의 성능을 저하시킨다. 시간확산에 의한 ISI를 줄이고 전송율이 높은 코히어런트(coherent) 시스템을 적용하기 위해서는 다중경로 영향을 제거해야 한다. 본 연구에서 적용하는 수동 페이저 컨쥬게이션(passive phase conjugation) 기법은 시역전 기술로 프로브 신호에 의한 시역전을 수행하여 다중경로에 의한 ISI가 감소하여 비트 오류율이 적어진다. 아울러 수신신호의 처리가 간단하여 다중경로에 의한 시간확산에 비례하는 탭수의 증가로 수신신호 처리시간이 과다한 적응등화기 기법에 비해 실시간 시스템 구현에 유리한 기법이다. 수치모의 실험으로 제안하는 기법의 성능을 해석하였다.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.15 no.4
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• pp.360-367
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• 2004

In this paper, we have implemented the digital signal processing system for intruder detection using speckle pattern variation in multi-me optical fiber with hypersensitive and high fidelity. The performance of the implemented system was evaluated by experiments. In order to improve the system performances we applied the adaptive digital filter. In experimental results we could see 96 % intruder detection and 90 % man/car discrimination probability.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.3 no.3
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• pp.501-507
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• 1999

At the mobile unit for adaptation the propagation environment, it is necessity to adapt very fast the weight coefficient vector of adaptive array antenna In this paper, for the BPSK and BFSK signals with S/I=2, S/N=10 subband adaptive array signal processing method to the linear array antenna using the LMS & the Kalman filter algorithm is proposed. For the 4 elements equidistance linear array antenna systems LMS and Kalman algorithms with subband adaptive instruction principles using the subband signal processing method are adopted and the computer simulation results to the constant amplitude envelope signals such as BPSK or BFSK can be seen that the convergence characteristics of directional patterns and the signal following characteristics are more fast and stable.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.14 no.5
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• pp.549-560
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• 1989

In this paper, the method of determining the initial weighting vlaues fuctioning as a filter under the Directional Constrained Minimization of Power(DCMP) algorithm is presented. By analyzing the sideband beamformer with the Finite Impulse Response (FIR) filter concepts, the constraints of any desired directions are obtained and the initial weighing values with fast adaptation time are formulated from those constraints. By applying this proposed initial weighting values to the DCMP and the spatial averaging processor, the interference of a desired direction and the coherent noises are eliminated at the same time. The improvement of this method compared with the existing algorithm is confirmed by computer simulation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.8
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• pp.1774-1781
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• 1997

In this paper, we propose a full digital frequency and phase locked loop for CATV and HDTV receivers adopting VSB modulation. The CATV and HDTV receivers proposed by the Grand-Alliance in USA are ultilizing analog signal processing technology for carrier recovery. By the way, it is not a good architecture for the development of single chip ASIC operating in digital domain. To solve this problem while improving the performance, we first down convert the received r.f. signal to a near baseband signal for a low-rate AD converter and then we use digital signal processing techniques. The proposed system has the frequency pull-in range of -200 KHz +2.50 KHz. Moreover, it has the ability of adaptive loop bandwidth control according to the amount of frequency offset to improve the acquisition time while reducing the phase noise.