• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.12
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• pp.2724-2729
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• 2010

This paper presents a method of constructing the sequential digital systems over finite fields. We assign all elements in finite fields to digit codes using mathematical properties of finite fields. Also, we discuss the operational characteristics and properties of the building block T-gate which is used to implement the sequential digital systems over finite fields. Then, we implemented sequential digital systems without feed-back. The sequential digital systems without feed-back is constructed as following steps. First, we assign the states in state-transition diagram to state digit codes, then obtain the state function and predecessor table which is explaining the relationship between present states and previous states. Next, we obtained the next-state function from state function and predecessor table. Finally we realize the circuit using T-gate and decoder. The proposed method is more efficiency and systematic than previous method.

• Journal of Digital Contents Society
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• v.13 no.4
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• pp.491-500
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• 2012

In this paper, a new loopback BIST structure which is effective to test RF transceiver chip and LNA(Low Noise Amplifier) in the chip is presented. Because the presented BIST structure uses a baseband processor in the chip as a tester while the system is under testing mode, the developed test technique has an advantage of performing test application and test evaluation in effectiveness. The presented BIST structure can change high frequency test output signals to a low frequency signals which can make the CUT(circuits under test) tested easily. By using this technique, the necessity of RF test equipment can be mostly reduced. The test time and test cost of RF circuits can be cut down by using proposed BIST structure, and finally the total chip manufacturing costs can be reduced.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.6 no.5
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• pp.724-730
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• 2002

Digital communication systems are required to cause the minimum interference to adjacent channels, they must therefore employ the linear power amplifiers. In respect to linear power amplifiers, there are many linearization techniques. Feedforward power amplifier represent very wide bandwidth and high linearization capability. In the feedforward systems, overall efficiency is reduced due to the loss of delay line. In this paper, delay filter instead of transmission delay line adapted to get more high efficiency. Experimental results showed that ACLR (Adjacent Channel Leakage Ratio) has improved 17.43(dB), which is added 3.44(dB) by using the delay filter.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.9 no.2
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• pp.205-212
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• 2016

In this paper, we implement the development platform of multichannel digital hearing aid algorithm using Simulink provided by Matlab. The digital hearing aids are considered medical devices designed to compensate for hearing loss, they need to be correctly selected, to help a person who has difficulty in hearing. The development platform that implemented in this paper, includes WOLA filterbank for analysis/synthesis of input signal, Wide dynamic range compression for hearing loss compensation and adaptive filter for feedback cancellation. Using the development platform, algorithm parameters for each block can be set depending on the hearing aid user. Thus it is possible to test the algorithm before the machine language. As a result, the time for algorithm development can be saved and performance and computational complexity can be optimized.

• Journal of IKEEE
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• v.17 no.2
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• pp.145-150
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• 2013

In the paper, we propose a new CSS(Complex Signed-Signed) CMA(Constant Modulus Algorithm) algorithm for ICS(Interference Cancellation System). When the repeater get the feedback signal, the CSS CMA algorithm is proposed at the ICS repeater using DSP(Digital Signal Processing) for the removal of interfering signals from the feedback paths. The proposed CSS CMA algorithm improved performances and hardware complexity by adjusting step size values. the steady state MSE(Mean Square Error) performance of the proposed CSS CMA algorithm with step size of 0.00043 is about 4dB better than the conventional CMA algorithm. And the proposed Complex Signed Signed CMA algorithm requires 1950 ~ 2150 less iterations than the LMS(Least Mean Square) and Signed LMS(Normalized Least Mean Square) algorithms at MSE of -25dB.

• Speech Sciences
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• v.12 no.3
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• pp.79-89
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• 2005

Acoustic feedback between microphone and receiver can be effectively cancelled adaptive feedback cancellation algorithm. Although many speech sounds have non-Gaussian distribution, most algorithms were tested with speech like sounds whose distribution were Guassian type. In this paper, we proposed an adaptive feedback cancellation algorithm based on independent component analysis (ICA) for digital hearing aid. The algorithm was tested with not only Gaussian distribution but also Laplacian distribution. We verified that the proposed algorithm has better acoustic feedback cancelling performance than conventional normalized root mean square (NLMS) algorithm, especially speech like sounds with Laplacian distribution.

• Journal of the Korean Institute of Telematics and Electronics
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• v.27 no.3
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• pp.137-144
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• 1990

We describe how single layer perceptrons and new nonsymmetry feedback type neural networks can be implemented by VLSI CMOS technology. The network described provides a flexible tool for evaluation of boolean expressions and arithmetic equations. About 50 CMOS VLSI chips with an architecture based on two neural networks have been designed and me being fabricated by 2-micrometer double metal design rules. These chips have been developed to study the potential of neural network models for the use in character recognition and for a neural compute.

• The Journal of the Acoustical Society of Korea
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• v.12 no.1E
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• pp.5-11
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• 1993

본 논문에서는 페이딩 채널에서 백색 가우시안 잡음과 임펄스성 잡음이 부가된 디지털 신호를 복원하기 위하여 적응 LMS알고리즘과 RLS 알고리즘을 사용하여 TDL 등화기, 결정 궤환 등화기, 그리고 격자 등화기의 성능을 평가하고 비교하였다. 오차 성능 분석 결과, 페이딩이 존재하고 임펄스성과 가우스성 잡음이 존재하는 채널에서 10-3BER을 얻기 위해서, 격자 등화기는 LMS이 등화기보다 3.0dB, RLS TDL 등화기보다 3.9dB의 신호대 잡음비(SNR)여유를, 그리고 LMS DFE 등화기보다 0.5dB, RLS DFE등화기보다 -0.5dB의 SNR 여유를 갖음을 확인하였다.

• The Transactions of the Korean Institute of Electrical Engineers
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• v.43 no.2
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• pp.259-267
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• 1994

This paper describes a current control of a single-phase PWM inverter. The proposed PWM inverter utilizes the instantaneous control method which is based on the real-time digital feedback control and the microprocessor-based deadbeat control. The deadbeat current controller is proposed to control the output current regardless of load component variations by the same method as voltage control. That is, in current control, with a very short sampling time and the successive feedback of the output current, the load current is mainly effected by the magnitude of load impedance rather than load component, the load current is mainly effected by the magnitude of load impedance rather than load component. Therefore, by treating the load as an impedance, the system's order is reduced and the instantaneous current control using the proposed deadeat controller is simplified.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 1998.05a
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• pp.115-118
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• 1998

본 연구에서는 TMS320C40을 이용하여 자연스러운 실시간 잔향기를 구현하기 위하여 주파수특성의 보상 및 coloeration의 제거, 그리고 실행코드를 최적화하는 방법을 연구분석하였다. 주파수특성이 평탄한 문제를 해결하기 위하여 잔향기를 구성하는 comb필터의 궤환부에 저역통과필터를 추가하는 방법을 제안하였으며 coloration을 제거하기 위하여 comb 필터의 지연과 allpass 필터의 이득 및 지연을 변화시켜가면서 최적의 값을 결정하였다. 임펄스응답을 이용하여 구현된 잔향기의 특성을 실험한 결과, 실제 공간에서 측정된 임펄스응답의 잔향 시간에 따른 주파수특성과 유사한 결과를 얻었으며 CD음악을 이용하여 44.1kHz에서 실시간 동작시킨 청취실험에서도 자연스러운 특성을 나타내었다.