• Journal of the Institute of Convergence Signal Processing
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• v.9 no.1
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• pp.1-9
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• 2008

One of the problems for implementing the spot detection phase in the 2-DE gel image analysis program is the eliminating noises in the image. Remained noises after the preprocessing phase cause the over-segmented regions by the segmentation phase. To identify and exclude the over-segmented background regions, if we use the fixed thresholding method that is choosing an intensity value for the threshold, the spots that is invisible by the eyes but mean a very small amount proteins which have important role in the biological samples could be eliminated. This paper propose an adaptive thresholding method that come from an idea that is got on statistical analysing for the prominences of the peaks. The adaptive thresholding method works as following. Firstly we calculate an average prominence value curve and fit it to exponential function curve, as a result we get parameters for the exponential function. And then we calculate a threshold value by using the parameters and probability distribution of errors. Lastly we apply the threshold value to the region for determining the region is a noise or not. According to the probability distribution of errors, the reliability is 99.85% and we show the correctness of the proposed method by representing experiment results.

• The Transactions of the Korea Information Processing Society
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• v.4 no.10
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• pp.2562-2570
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• 1997

This paper deals with the design and performance of the adaptive equalizer using high order statistics in order to improve the transmission characteristics of multipath fading channel. The multipath propagational phenomenon occurred in digital radio transmission causes the distortion and ISI of receiving signal. These are main reasons to increase the bit error rate and degrade the performance of receivers. In this paper, the adaptive equalization algorithm using high order statistics of received signal is used instead of CMA algorithm, Bussgang and Godard which are known widely. The performance of this algorithm (residualisi, recovered constellation, calculation) is presented varing SNR. As the result of the computer simulation, equalizer algorithm using high order statistics is better than CMA in the range of low SNR, $10{\sim}20dB$. Therefore, considering the actual communication systems which use the range of $14{\sim}20$ SNR, the adaptive equalizer using high order statistics can be used in the real multipath fading environment.

• Journal of Information Processing Systems
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• v.9 no.2
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• pp.333-348
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• 2013

Mobile authentication/identification has grown into a priority issue nowadays because of its existing outdated mechanisms, such as PINs or passwords. In this paper, we introduce gait recognition by using a mobile accelerometer as not only effective but also as an implicit identification model. Unlike previous works, the gait recognition only performs well with a particular mobile specification (e.g., a fixed sampling rate). Our work focuses on constructing a unique adaptive mechanism that could be independently deployed with the specification of mobile devices. To do this, the impact of the sampling rate on the preprocessing steps, such as noise elimination, data segmentation, and feature extraction, is examined in depth. Moreover, the degrees of agreement between the gait features that were extracted from two different mobiles, including both the Average Error Rate (AER) and Intra-class Correlation Coefficients (ICC), are assessed to evaluate the possibility of constructing a device-independent mechanism. We achieved the classification accuracy approximately $91.33{\pm}0.67%$ for both devices, which showed that it is feasible and reliable to construct adaptive cross-device gait recognition on a mobile phone.

• Journal of the Institute of Convergence Signal Processing
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• v.4 no.2
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• pp.46-52
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• 2003

In this paper, the wavelet watermarking algorithm based on adaptive weight MASK processing as a watermark embedded-method for Copyright Protection of Digital contents is Proposed. Because watermark induce as a noise of original image, the watermark size should be limited for preventing quality losses and embedding watermark into images. Therefore, it should be preserve the best condition of the factors, robustness, capacity and visual quality. Tn order to solve this problem, we propose watermarking embedded method by applying adaptive weight MASK to the algorithm and optimize its efficiency. In that result, the watermarked images are improved about external attack. Specifically, correlation coefficient has over 0.8 on both modifications of brightness and contrast. Also, correlation coefficient of wavelet compression of embedded watermark last by over 0.65.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.49 no.4
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• pp.120-128
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• 2012

In this paper, we propose a modified iterative pre-whitening projection statistics (MIPPS) scheme for improving multi-target detection performance in non-homogeneous clutter environments. As a non-homogeneity detection (NHD) technique of space-time adaptive processing algorithm for airborne radar, the MIPPS scheme improves the average detection probability of weak target when multiple targets with different reflection signal intensities are located in close range. Numerical results show that the conventional NHD schemes suffers from the masking effect by strong targets and clutters and the proposed MIPPS scheme outperforms the conventional schemes with respect to the average detection probability of the weak target at low signal-to-clutter ratio.

• Journal of the Institute of Convergence Signal Processing
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• v.14 no.2
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• pp.95-98
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• 2013

The LMS(Least Mean Square) algorithm, one of the most famous, is generally used because of tenacity and high mating spots and simplicity of realization, But it has trade-off between nonuniform collection and EMSE(Excess mean square error). To overcome this weakness, a variable step size is used widely, but it needs a lot of calculation loads. In this paper, we suggest changed algorithm in case of environment changes of cars and reduce amount of calculation as it uses original signal and noise signal of IGC(Instantaneous Gain Control) algorithm. In this paper, logarithmic function is removed because of real-time processing IGC. The performance of proposed algorithm is tested to adaptive noise canceller in automobile.

• The Journal of the Acoustical Society of Korea
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• v.23 no.6
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• pp.446-453
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• 2004

In this paper, a new adaptive perceptual filter using variable threshold to enhance audio signals degraded by additively nonstationary noise is proposed. The adaptive perceptual filter updates variable threshold each time according to the power of signal and the effect of noise variation. So the noisy audio signal is enhanced by the method which controls a residual noise effectively. The proposed algorithm uses the perceptual filter which transforms a time domain signal into frequency domain and calculates an intensity energy and an excitation energy in bark domain. In this method. the stage updated the response of filter is decided by threshold. The proposed algorithm using vairable threshold effectively controls a residual noise using the energy difference of audio signals degraded by the additive nonstationary noise. The proposed method is tested with the noisy audio signals degraded by nonstationary noise at various signal -to-noise ratios (SNR). We carry out NMR and MOS test when the input SNR is 15dB. 20dB. 25dB and 30dB. An approximate improvement of 17.4dB. 15.3dB, 12.8dB. 9.8dB in NMR and enhancement of 2.9, 2.5, 2.3, 1.7 in MOS test is achieved with the input signals. respectively.

• The Transactions of the Korea Information Processing Society
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• v.7 no.2S
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• pp.701-710
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• 2000

This paper presents a new method for an efficient coding of very low bit-rate color video based on adaptive wavelet transform. Our approach reveals that the coding process works more efficiently if the quantized wavelet coefficients are preprocessed by a mechanism exploiting the redundancies in the wavelet subband structure. Thus, we focuses optimized activity of coding part, and exhaustive overlapped block motion compensation is utilized to ensure coherency in motion compensated error frames, and raised cosine window is applied. The horizontal and vertical components of motion vectors are encoded separately using adaptive arithmetic coding while significant wavelet coefficients are encoded in bit-plane order using adaptive arithmetic coding. On average the proposed codec exceeds H.263 and ZTE in peak signal-to-noise ratio by as much as 2.07 and 1.38dB at 28 kbits, respectively. Fore entire sequence coding, 3DWCVC method is superior to H.263 and ZTE by 0.35 and 0.71dB on average, respectively.

• MALSORI
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• no.52
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• pp.121-132
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• 2004

This paper is aiming at real-time implementation of adaptive noise cancellation system using DSP processor. ACHARF algorithm, which guarantees stability and fast convergence by adaptive compensator, is used on this DSP system. For the experiments, TLV320AIC23 stereo CODEC of TI Inc. is used with TMS320C6413 DSP processor. Signals of primary input and reference input are obtained by two microphones. The primary input is the voice plus noise signal and the reference input is white noise or real noise. The experimental results show that ANC system using DSP processor with ACHARF is verified to be an effective speech enhancement method for various speech processing units.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2001.10a
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• pp.323-326
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• 2001

We propose a new MUD adaptive beamforming algorithm. It requires no reference signal and consists of two Processing stages. The first stage performs a scan function, and the second stage performs an adaptive beamforming algorithm. Computer simulations, considering multi-path Rayleigh Fading Channel in CUMA, are presented to verify the performance.