• Journal of IKEEE
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• v.21 no.3
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• pp.202-210
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• 2017

Recently, some variable length adaptive filters which employ variable lengths taps for the input signal statistics are proposed [1-5]. In this paper, a new subband adaptive filter with variable filter tap length is proposed. The proposed subband variable length adaptive filters can optimize filter length for each subband which can result less computational complexities with respect to the conventional full band adaptive filters. When the signal in the full band has narrow spectrum, the conventional full band adaptive requires very long filter taps, whereas the proposed subband variable filter requires less taps with the spectrum split in subband. The computer simulation results reveals that in many case, in system identification with narrow band system estimation, the proposed adaptive filter has less computational complexities with faster convergence.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.21 no.3
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• pp.301-306
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• 2010

In this paper, a SIT(Source Independent Tree) method for ray tracing is proposed to enhance the efficiency of tree construction with reuse of sub tree in urban environment, As the SIT method is applied, the decrease of the number of nodes for picocell and microcell simulations is 100 times. And 88~98 % of the total nodes are reused with change of location of signal source from an analysis of node reuse efficiency. Therefore the proposed SIT method is useful in performance enhancement of ray tracing, especially, for multiple antenna simulation like as MIMO system and cell planning.

• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.1
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• pp.27-31
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• 2012

In this paper, we constructed an array of speaker system with directivity by using FXLMS(Filtered-X LMS) algorithm and confirmed its directivity. The front $0^{\circ}$ characteristics of the controlled speaker was suppressed by interfering it with the control signal produced with filter coefficients optimized with respect to the $180^{\circ}$ characteristics of the rear speakers. The directivity of the array of rear speakers was measured and the damping effect of the signal from the front speaker array was confirmed. The frequency characteristics and directivity was investigated by using the adaptive filter coefficients on damping, the damping on the control point was verified in all the frequency range. In 100Hz, 200Hz, 1000Hz regime, the damping effect was observed in the range of front $60^{\circ}{\sim}100^{\circ}$.

• Journal of Digital Convergence
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• v.17 no.10
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• pp.251-258
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• 2019

In the future, various products are created in various fields using artificial intelligence. In this age, it is a very important problem to know the operation principle of artificial intelligence learning method and to use it correctly. This paper introduces artificial intelligence learning methods that have been known so far. Learning of artificial intelligence is based on the fixed point iteration method of mathematics. The GD(Gradient Descent) method, which adjusts the convergence speed based on the fixed point iteration method, the Momentum method to summate the amount of gradient, and finally, the Adam method that mixed these methods. This paper describes the advantages and disadvantages of each method. In particularly, the Adam method having adaptivity controls learning ability of machine learning. And we analyze how these methods affect digital signals. The changes in the learning process of digital signals are the basis of accurate application and accurate judgment in the future work and research using artificial intelligence.

• Journal of the Korea Society of Computer and Information
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• v.15 no.9
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• pp.129-136
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• 2010

With mobile computing, wireless sensor network and sensor technologies, ubiquitous computing services are being realized and could satisfy the feasibility of ubiquitous healthcare to everyone. This u-Healthcare service can improve life quality of human since medical service can be provided to anyone, anytime, and anywhere. To confirm the vision of u-Healthcare service, we've implemented a healthcare system for heart disease patient which is composed of two components. Front-end collects various signals such as temperature, blood pressure, SpO2, and electrocardiogram, etc. As a backend, medical information server accumulates sensing data and performs back-end processing. To simply transfer these sensing values to a medical team may be too trivial. So, we've designed a model based on context awareness for more improved medical service which is based on artificial neural network. Through rigid experiments, we could confirm that the proposed system can provide improved medical service.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.24 no.3
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• pp.398-404
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• 2020

In this paper, we propose a new design method of signal constellation of the spiral quadrature amplitude modulation (QAM) exploiting a modified gradient descent search algorithm and its binary mapping rule. Unlike the conventional method, the new method, which uses and the constellation optimization algorithm and the maximum number of iterations as a parameter for the iterative design, is more robust to phase noise. And the proposed binary mapping rule significantly reduces the average Hamming distance of the spiral constellation. As a result, the proposed spiral QAM constellation has much improved error performance compared to the conventional ones even in a very severe phase noise environment. It is, therefore, considered that the proposed QAM may be a useful modulation format for coherent optical communication systems and orthogonal frequency division multiplexing (OFDM) systems.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.1
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• pp.24-36
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• 2003

The digital mobile communication will be developed till getting multimedia service in anyone, any where, any time. Theses requiring items are going to be come true via IMT-2000 system. Transmitting signal bandwidth of IMT-2000 system is 3 times as large as IS-95 system. That is mean peak to average of signal is higher than IS-95A system. So we have to design it carefully not to effect in adjacent channel. HPA(High Power Amplifier) located in the end point of system is operated in 1-㏈ compression point(Pl㏈), then it generates 3rd and 5th inter modulation signals. Theses signals affect at adjacent channel and RF signal is distorted by compressed signal which is operated near by Pl㏈ point. Then the most important design factor is how we make HPA having high linearity. Feedback, Pre-distorter and Feed-forward methods are presented to solve theses problems. Feed-forward of these methods is having excellent improving capacity, but composed with complex structure. Generally, Linearity and Efficiency in power amplifier operate in the contrary, then it is difficult for us to find optimal operating point. In this paper we applied algorithm which searches optimal point of linear characteristics, which is key in Power Amplifier, using minimum current point of error amplifier in 1st loop. And we made 2nd loop compose with new structure. We confirmed fabricated LPA is operated by having high linearity and minimum current condition with ACPR of -26 ㏈m max. @ 30㎑ BW in 3.515㎒ and ACLR of 48 ㏈c max@${\pm}$㎒ from 1W to 40W.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.43 no.10 s.352
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• pp.80-89
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• 2006

The rapid development of process technology and the introduction of new materials not only make it difficult for process control but also as a result increase process variations. These process variations are barriers to successful implementation of design circuits because there are disparities between data on layout and that on wafer. This paper proposes a new design environment to determine the interconnect worstcase with accuracy and speed so that the interconnect effects due to process-induced variations can be applied to designs of $0.13{\mu}m$ and below. Common Geometry and Maximum Probability methods have been developed and integrated into the new worstcase optimization algorithm. The delay time of the 31-stage Ring Oscillator, manufactured in UMC $0.13{\mu}m$ Logic, was measured, and the results proved the accuracy of the algorithm. When the algorithm was used to optimize worstcase determination, the relative error was less than 1.00%, two times more accurate than the conventional methods. Furthermore, the new worstcase design environment improved optimization speed by 32.01% compared to that of conventional worstcase optimizers. Moreover, the new worstcitse design environment accurately predicted the worstcase of non-normal distribution which conventional methods cannot do well.

• Journal of Broadcast Engineering
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• v.13 no.4
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• pp.452-462
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• 2008

In this paper, we address issues associated with the real-time implementation of the MPEG-1/2 Layer-II (or MUSICAM) and MPEG-4 ER-BSAC decoders for Digital Multimedia Broadcasting (DMB) on TMS320C64x+ that is a fixed-point DSP processor with a clock speed of 330 MHz. To achieve the real-time requirement, they should be optimized in different steps as follows. First of all, a C-code level optimization is performed by sharing the memory, adjusting data types, and unrolling loops. Next, an algorithm level optimization is carried out such as the reconfiguration of bitstream reading, the modification of synthesis filtering, and the rearrangement of the window coefficients for synthesis filtering. In addition, the C-code of a synthesis filtering module of the MPEG-1/2 Layer-II decoder is rewritten by using the linear assembly programming technique. This is because the synthesis filtering module requires the most processing time among all processing modules of the decoder. In order to show how the real-time implementation works, we obtain the percentage of the processing time for decoding and calculate a RMS value between the decoded audio signals by the reference MPEG decoder and its DSP version implemented in this paper. As a result, it is shown that the percentages of the processing time for the MPEG-1/2 Layer-II and MPEG-4 ER-BSAC decoders occupy less than 3% and 11% of the DSP clock cycles, respectively, and the RMS values of the MPEG-1/2 Layer-II and MPEG-4 ER-BSAC decoders implemented in this paper all satisfy the criterion of -77.01 dB which is defined by the MPEG standards.

• Journal of the Korean Institute of Telematics and Electronics T
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• v.36T no.1
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• pp.1-7
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• 1999

We present VLSI architecture for a high performance 2-D DCT processor which is used compressing system of real time image processing or HDTV using fast computational algorithm of the Optimized Integer Cosine Transform(OICT). The coefficients of the OICT are integer, so the OICT performs only the integer operations for both forward and inverse transform. Therefore the proposed architecture could be greatly enhanced in improving the speed, reduced the hardware cost considerably by replacing the multiplication operations with shift and addition operations compared with DCT which performs floating-point operations.