• 한국통신학회논문지
• /
• 제22권11호
• /
• pp.2470-2476
• /
• 1997

We propose a simle scheme for detecting tone signals narrowly spaced in the frequency by employing a nonlinear device called Teager's energy tracker. When the input signal is a tone signal, the output of Teager's energy tracker is proportional to the energy and tone frequency of the input signal. When the envelope of the input tone is varying, this tracker can also track the energy of the envelope. By employing Teager's algorithm as a preprocessor, the detector for narrowly spaced tone signals can be implemented without the use of sharp notch filters. As an example, the proposed detector is applied to distinguishing ITU-T V.8 and V.25 tone signals. The detector perfodrmance is analyszed and is verified by computer simulation.

• 한국통신학회논문지
• /
• 제22권2호
• /
• pp.268-279
• /
• 1997

In this paper, we show the movable boundary (MB) method is the effective bandwidth allocation strategy for the ABR service among existing bandwidth allocation methods. The MB method can gurantee the QOS requirement according to the service priority and improve bandwidth utilization using the characteristics of the ABR service. The threshold values of the MB, which have an important effect upon connection blocking probability(CBP) performance of each service, are obtained by simulation. The MB method with the fixed thresholds, however, may not have good performance to variation of input traffic parameters. We suggest the dynamic MB method which changes the threshold values adaptively according to the required bandwidth of input traffic. We also show that the dynamic MB method can urantee the CBP performance according to the service priority regardless of variation of input traffic parameters.

• 디지털융복합연구
• /
• 제12권7호
• /
• pp.253-259
• /
• 2014

본 논문에서는 사용자 친화적이면서 좁은 영역에서도 효과적으로 입력을 수행할 수 있는 새로운 방식의 키보드를 제안한다. 기존의 키보드는 문자의 배열 형태나 사용 방식은 다르지만 대부분 키를 나열한 다음 선택하도록하는 방식이었다. 본 논문에서는 기존의 방식과 다르게 키보드 내용이 동적으로 변화하는 새로운 방식의 정보 입력장치인 Dynamic Keyboard(DynaKey)를 설계한다. Dynakey는 정확한 위치 선택이 어려운 장애인이나 화면이 작은 이동기기에서 사용하기 편리한 정보입력장치이다.

• Journal of information and communication convergence engineering
• /
• 제10권2호
• /
• pp.103-107
• /
• 2012

An end-launched adapter combining a coaxial loop within a rectangular waveguide has been analyzed and designed. The accurate electromagnetic modeling of input impedance for the adapter design has been developed using a modal analysis method. Also, design parameters to improve the end-launched adapter have been investigated. Numerical and experimental results of the proposed model have been compared with the results of previous works, and verified by an electromagnetic simulator of High Frequency Structure Simulator (HFSS). The input voltage standing wave ratios (VSWRs) are smaller than 2 over the wide frequency band from 7.5 GHz to 10.5 GHz.

• 한국정보통신학회:학술대회논문집
• /
• 한국정보통신학회 2015년도 춘계학술대회
• /
• pp.480-481
• /
• 2015

반파장 길이를 가지는 일반적인 마이크로스트립 안테나는 작은 대역폭 특성을 가진다. 패치 안테나의 모서리를 적절하게 잘라내면 입력 임피던스가 변하여 이에 상응하는 광대역 특성을 얻을 수 있다. 본 논문에서는 2.4GHz ISM 대역에서 동작하는 광대역 마이크로스트립 패치 안테나를 대상으로 패치 안테나의 모서리 단면의 모양에 따른 입력 임피던스의 변화를 통하여 광대역 특성을 얻을 수 있도록 설계하였다.

• 한국통신학회논문지
• /
• 제21권5호
• /
• pp.1172-1179
• /
• 1996

General adaptive noise canceller(ANC) suffers from the misadjustment of adaptive filter weights, because of the gradient-estimate noise at steady state. In this paper, an adaptive noise cancellation algorithm with speech detector which is distinguishing speech from silence and adaptation-transient region is proposed. The speech detector uses property of adaptive prediction-error filter which can filter the highly correlated speech. To detect speech region, estimation error which is the output of the adaptive filter is applied to the adaptive prediction-error filter. When speech signal apears at the input of the adaptive prediction-error filter. The ratio of input and output energy of adaptive prediction-error filter becomes relatively lower. The ratio becomes large when the white noise appears at the input. So the region of speech is detected by the ratio. Sign algorithm is applied at speech region to prevent the weights from perturbing by output speech of ANC. As results of computer simulation, the proposed algorithm improves segmental SNR and SNR up to about 4 dBand 11 dB, respectively.

• 한국통신학회논문지
• /
• 제21권11호
• /
• pp.3001-3011
• /
• 1996

In this paper, using DBNN algorithm which is used in the binary pattern classification or speech signal processing the digital DBNN circuit is designed having the variable expansion depending the size of input data and pattern type. The processing elemen(PE) of the proposed network consists of the synapse and MAXNET circuits for the similarity measurement between reference and input pattern. Global MAXNET selects the global winner among the local winners which is selected in each PE. Through the several simultions, and thus each PE and global MAXNET search the reference pattern that was the most simlar to input pattern for the discord of the pattern.

• Journal of the Korean Data and Information Science Society
• /
• 제27권3호
• /
• pp.827-833
• /
• 2016

In this paper, we propose a deep least squares support vector machine (LS-SVM) for regression problems, which consists of the input layer and the hidden layer. In the hidden layer, LS-SVMs are trained with the original input variables and the perturbed responses. For the final output, the main LS-SVM is trained with the outputs from LS-SVMs of the hidden layer as input variables and the original responses. In contrast to the multilayer neural network (MNN), LS-SVMs in the deep LS-SVM are trained to minimize the penalized objective function. Thus, the learning dynamics of the deep LS-SVM are entirely different from MNN in which all weights and biases are trained to minimize one final error function. When compared to MNN approaches, the deep LS-SVM does not make use of any combination weights, but trains all LS-SVMs in the architecture. Experimental results from real datasets illustrate that the deep LS-SVM significantly outperforms state of the art machine learning methods on regression problems.

• ETRI Journal
• /
• 제22권3호
• /
• pp.1-9
• /
• 2000

Self-Clocked Fair Queueing (SCFQ) algorithm has been considered as an attractive packet scheduling algorithm because of its implementation simplicity, but it has unbounded delay property in some input traffic conditions. In this paper, we propose a Rate Proportional SCFQ (RP-SCFQ) algorithm which is a rate proportional version of SCFQ. If any fair queueing algorithm can be categorized into the rate proportional class and input is constrained by a leaky bucket, its delay is bounded and the same as that of Weighted Fair Queueing (WFQ) which is known as an optimal fair queueing algorithm. RP-SCFQ calculates the timestamps of packets arriving during the transmission of a packet using the current value of system potential updated at every packet departing instant and uses a starting potential when it updates the system potential. By doing so, RP-SCFQ can have the rate proportional property. RP-SCFQ is appropriate for high-speed packet-switched networks since its implementation complexity is low while it guarantees the bounded delay even in the worst-case input traffic conditions.

• 전자공학회논문지SC
• /
• 제48권2호
• /
• pp.56-63
• /
• 2011

본 논문에서는 마이크로프로세서의 디지털 입력포트에 히스테리시스 특성을 부여하는 방법 및 설계 순서를 제시하고 이를 실험해 그 효과를 확인하였다. 프로세서의 잉여포트가 있을 때 제시한 방법을 이용하면 저항 2개의 추가만으로 히스테리시스 특성을 얻을 수 있으며, 더구나 기존의 TTL과 CMOS의 슈미트 트리거 게이트에 비해 큰 히스테리시스 폭을 얻을 수 있다.