• Journal of KIISE:Software and Applications
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• v.33 no.4
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• pp.402-410
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• 2006

This paper presents a method for feature based block motion estimation that uses multi -resolution image sequences to obtain the panoramic images in the continuous ultrasound image sequences. In the conventional block motion estimation method, the centers of motion estimation blocks are set at the predetermined and equally spaced locations. This requires the large blocks to include at least one feature, which inevitably requires long estimation time. In this paper, we propose an adaptive method which locates the center of the motion estimation blocks at the feature points. This make it possible to reduce the block size while keeping the motion estimation accuracy The Harris-Stephen corner detector is used to get the feature points. The comer points tend to group together, which cause the error in the global motion estimation. In order to distribute the feature points as evenly as Possible, the image is firstly divided into regular subregions, and a strongest corner point is selected as a feature in each subregion. The ultrasound Images contain speckle patterns and noise. In order to reduce the noise artifact and reduce the computational time, the proposed method use the multi-resolution image sequences. The first algorithm estimates the motion in the smoothed low resolution image, and the estimated motion is prolongated to the next higher resolution image. By this way the size of search region can be reduced in the higher resolution image. Experiments were performed on three types of ultrasound image sequences. These were shown that the proposed method reduces both the computational time (from 77ms to 44ms) and the displaced frame difference (from 66.02 to 58.08).

• Journal of KIISE:Software and Applications
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• v.30 no.7_8
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• pp.672-680
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• 2003

A shopbot is a software agent whose goal is to maximize buyer´s satisfaction through automatically gathering the price and quality information of goods as well as the services from on-line sellers. In the response to shopbots´ activities, sellers on the Internet need the agents called pricebots that can help them maximize their own profits. In this paper we adopts Q-learning, one of the model-free reinforcement learning methods as a price-setting algorithm of pricebots. A Q-learned agent increases profitability and eliminates the cyclic price wars when compared with the agents using the myoptimal (myopically optimal) pricing strategy Q-teaming needs to select a sequence of state-action fairs for the convergence of Q-teaming. When the uniform random method in selecting state-action pairs is used, the number of accesses to the Q-tables to obtain the optimal Q-values is quite large. Therefore, it is not appropriate for universal on-line learning in a real world environment. This phenomenon occurs because the uniform random selection reflects the uncertainty of exploitation for the optimal policy. In this paper, we propose a Mixed Nonstationary Policy (MNP), which consists of both the auxiliary Markov process and the original Markov process. MNP tries to keep balance of exploration and exploitation in reinforcement learning. Our experiment results show that the Q-learning agent using MNP converges to the optimal Q-values about 2.6 time faster than the uniform random selection on the average.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.13 no.4
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• pp.1830-1838
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• 2012

The intra prediction and the motion estimation of inter prediction occupy 70 ~ 80% of whole compression time in H.264/AVC. Compression efficiency has been higher, but complexity has increased and coding time has also increased much more. This paper proposes a block size decision method of the intra prediction and mode decision method which minimize the loss of video quality during the encoding and shorten the time spent. This paper, in addition, proposes an algorithm which determines the method of adaptive block mode for motion estimation of inter prediction. We investigated PSNR and the intra prediction and inter prediction of time-consuming calculations in order to measure video quality degradation and complexity through experiments. Consequently, when you use all three methods, these methods showed that average coding time is shortened as about 500 to 600ms in every frame in the case of all experimented videos, keeping video quality nearly similar, compared with existing methods of H.264.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.9A
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• pp.1276-1284
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• 1999

We have evaluated the BER's and CLP's of Wireless ATM (WATM) cells employing the concatenated FEC code with pilot symbols for fading compensation through the simulation in indoor wireless channel modeled as a Rayleigh and a Rician fading channel, respectively. The results of the performance evaluation are compared with those obtained by employing the convolutional code in the same condition. In Rayleigh fading channel, considering the maximum tolerance BER ( $10^-3$) as a criterion of the voice service, it is blown that the performance improvement of about 4 dB is obtained in terms of $E_b/N_o$ by employing the concatenated FEC code with pilot symbols rather than the convolutional code with pilot symbols.When the values of K parameter which means the ratio of the direct signal to scattered signal power in Rician fading channel are 6 and 10, it is shown that the performance improvement of about 4 dB and 2 dB is obtained, respectively, in terms of $E_b/N_o$ by employing the concatenated FEC code with pilot symbols considering the maximum tolerance BER of the voice service. Also in Rician fading channel of K=6 and K= 10, considering CLP = $10^-3$ as a criterion, it is observed that the performance improvement of about 3.5 dB and1.5 dB is obtained, respectively, in terms of $E_b/N_o$ by employing the concatenated FEC code with pilot symbols.

• The Journal of the Acoustical Society of Korea
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• v.22 no.2
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• pp.121-128
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• 2003

Acoustic Echo Cancellation(m) is very active research topic having many applications like teleconference and hands-free communication and it employs Double Talk Detector(DTD) to indicate whether the near-end speaker is active or not. However. the DTD is very sensitive to the variation of acoustical environment and it sometimes provides wrong information about the near-end speaker. In this paper, we are focusing on the development of robust DTD algorithm which is a basic building block for reliable AEC system. The proposed AEC system consists of delayless subband AEC and narrow-band DTD. Delayless subband AEC has proven to have excellent performance of echo cancellation with a low complexity and high convergence speed. In addition, it solves the signal delay problem in the existing subband AEC. On the other hand, the proposed narrowband DTD is operating on low frequency subband. It can take most advantages from the narrow subband such as a low computational complexity due to the down-sampling and the reliable DTD decision making procedure because of the low-frequency nature of the subband signal. From the simulation results of the proposed narrowband DTD and wideband DTD, we confirm that the proposed DTD outperforms the wideband DTD in a sense of removing possible false decision making about the near-end speaker activity.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.37 no.1
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• pp.55-63
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• 2000

This paper investigated the sub-region compression effect of the three dimensional DCT(3D-DCT) using the difference component(DC) of inter-frame in images. The proposed algorithm are the method that obtain compression effect to divide the information into subband after 3D-DCT, the data appear the type of cubic block(8${\times}$8${\times}$8) in eight difference components per unit. In the frequence domain that transform the eight differential component frames into eight DCT frames with components of both spatial and temporal frequencies of inter-frame, the image data are divided into frame component(8${\times}$8 block) of time-axis direction into 4${\times}$4 sub block in order to effectively obtain compression data because image components are concentrate in corner region with low-frequency of cubic block. Here, using the weight of sub block, we progressed compression ratio as consider to adaptive sub-region of low frequency part. In simulation, we estimated compression ratio, reconstructed image resolution(PSNR) with the simpler image and the complex image contained the higher frequency component. In the result, we could obtain the high compression effect of 30.36dB(average value in the complex-image) and 34.75dB(average value in the simple-image) in compression range of 0.04~0.05bpp.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.4
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• pp.101-109
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• 2014

It is an essential element for the establishment of image processing related systems to find the exact contour from the image of an arbitrary object. In particular, if a vision system is established to inspect the products in the automated production process, it is very important to detect the contours for standardized shapes such lines and curves. In this paper, we propose a prototype adaptive dynamic contour model, p-Snake with improved contour extraction algorithms by adding the prototype energy. The proposed method is to find the initial contour by applying the existing Snake algorithm after Sobel operation is performed for prototype analysis. Next, the final contour of the object is detected by analyzing prototypes such as lines and circles, defining prototype energy and using it as an additional energy item in the existing Snake function on the basis of information on initial contour. We performed experiments on 340 images obtained by using an environment that duplicated the background of an industrial site. It was found that even if objects are not clearly distinguished from the background due to noise and lighting or the edges being insufficiently visible in the images, the contour can be extracted. In addition, in the case of similarity which is the measure representing how much it matches the prototype, the prototype similarity of contour extracted from the proposed p-ACM is superior to that of ACM by 9.85%.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.6B
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• pp.1082-1091
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• 2000

In this paper, the efficient equalization method for OFDM(Orthogonal Frequency Division Multiflexing) System using the QAM(Quadrature Amplitude Modulation) in multipath fading channel is proposed in order to faster and more efficiently equalize the received signals that are sent over real channel. In generally, the one-tap linear equalizers have been used in the frequency-domain as the existing equalization method for OFDM system. In this technique, if characteristics of the channel are changed fast, the one-tap linear equalizers cannot compensate for the distortion due to time variant multipath channels. Therefore, in this paper, we use one-tap non-linear equalizers instead of using one-tap linear equalizers in the frequency-domain, and also use the linear equalizer in the time-domain to compensate the rapid performance reduction at the low SNR(Signal-to-Noise Ratio) that is the disadvantage of the non-linear equalizer. In the frequency-domain, when QAM signals, consisting of in-phase components and quadrature (out-phase) components, are sent over the complex channel, the only in-phase and quadrature components of signals distorted by the multipath fading are changed the same as signals distorted by the noise. So the cross components are canceled in the frequency-domain equalizer. The time-domain equalizer and the adaptive algorithm that has lower-error probability and fast convergence speed are applied to compensate for the error that is caused by canceling the cross components in the frequency-domain equalizer. In the time-domain, To compensate for the performance of frequency-domain equalizer the time-domain equalizes the distorted signals at a frame by using the Gold-code as a training sequence in the receiver after the Gold-codes are inserted into the guard signal in the transmitter. By using the proposed equalization method, we can achieve faster and more efficient equalization method that has the reduced computational complexity and improved performance.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.6
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• pp.209-220
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• 2004

The main feature of H.264 standard against conventional video standards is the high coding efficiency and the network friendliness. In spite of these outstanding features, it is not easy to implement H.264 codec as a real-time system due to its high requirement of memory bandwidth and intensive computation. Although the variable block size motion compensation using multiple reference frames is one of the key coding tools to bring about its main performance gain, it demands substantial computational complexity due to SAD (Sum of Absolute Difference) calculation among all possible combinations of coding modes to find the best motion vector. For speedup of motion estimation process, therefore, this paper proposes fast algorithms for both integer-pel and fractional-pel motion search. Since many conventional fast integer-pel motion estimation algorithms are not suitable for H.264 having variable motion block sizes, we propose the motion field adaptive search using the hierarchical block structure based on the diamond search applicable to variable motion block sizes. Besides, we also propose fast fractional-pel motion search using small diamond search centered by predictive motion vector based on statistical characteristic of motion vector.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.19 no.9
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• pp.1028-1036
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• 2008

In the OFDM system, IQ imbalance problem happens at the RF front-end of transceiver, which degrades the BER(bit error rate) performance because it affects the constellation in the received signal. Also, phase noise is generated in the local oscillator of transceivers and it destroys the orthogonality between the subcarriers. Conventional PNS algorithm is effective for phase noise suppression, but it is not useful anymore when there are jointly IQ(In-phase and Quadrature) imbalance and phase noise. Therefore, in this paper, we analyze the effect of IQ imbalance and phase noise generated in the down-conversion of the receiver. Then, we estimate and compensate the IQ imbalance and phase noise at the same time. Compared with the conventional method that IQ imbalance after IFFT is estimated and compensated in front of FFT via the feedback, this proposed method extracts and compensates effect of IQ imbalance after FFT stage. In case IQ imbalance and phase noise exist at the same time, we can decrease complexity because it is needless to use elimination of IQ imbalance in time domain and training sequences and preambles. Also, this method shows that it reduces the ICI and CPE component using adaptive forgetting factor of MMSE after FFT.