• 한국전자파학회:학술대회논문집
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• 한국전자파학회 2003년도 종합학술발표회 논문집 Vol.13 No.1
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• pp.108-112
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• 2003

A feedforward amplifier, which is composed of several components, is an open loop system. Therefore, feedforward amplifiers are apt to deteriorate the performance according to the environmental changes even though the cancellation performance and the linearization bandwidth of feedforward systems are superior to other linearization methods. A control method is needed for maintaining the original performance of feedforward amplifiers or to keep the performance within a little error bounds. In this paper, an adaptive control method, which has a good convergence characteristic and is easy to implement, is suggested. The characteristics of the suggested control method compare with the characteristics of other control methods and the simulation results are presented.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송공학회 2009년도 IWAIT
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• pp.668-672
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• 2009

In wireless ad-hoc network, knowing the available bandwidth of the time varying channel is imperative for live video streaming applications. This is because the available bandwidth is varying all the time and strictly limited against the large data size of video streaming. Additionally, adapting the encoding rate to the suitable bit-rate for the network, where an overlarge encoding rate induces congestion loss and playback delay, decreases the loss and delay. While some effective rate controlling methods have been proposed and simulated well like VTP (Video Transport Protocol) [1], implementing to cooperate with the encoder and tuning the parameters are still challenging works. In this paper, we show our result of the implementation experiment of VTP based encoding rate controlling method and then introduce some techniques of our parameter tuning for a video streaming application over wireless environment.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제9권5호
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• pp.1733-1751
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• 2015

The IEEE 802.15.4 standard not only provides a maximum of seven guaranteed time slots (GTSs) for allocation within a superframe to support time-critical traffic, but also achieves ultralow complexity, cost, and power in low-rate and short-distance wireless personal area networks (WPANs). Real-time wireless body area sensor networks (WBASNs), as a special purpose WPAN, can perfectly use the IEEE 802. 15. 4 standard for its wireless connection. In this paper, we propose an adaptive GTS allocation scheme for real-time WBASN data transmissions with different priorities in consideration of low latency, fairness, and bandwidth utilization. The proposed GTS allocation scheme combines a weight-based priority assignment algorithm with an innovative starvation avoidance scheme. Simulation results show that the proposed method significantly outperforms the existing GTS implementation for the traditional IEEE 802.15.4 in terms of average delay, contention free period bandwidth utilization, and fairness.

• Journal of electromagnetic engineering and science
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• 제4권1호
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• pp.23-29
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• 2004

In this paper we analyze convergence constraints and time constant of IT-LMS algorithm and derive a method of making it's time constant independent of signal power by using input variance estimation. The method for estimating the input variance is to use a single-pole low-pass filter(LPF) with common smoothing parameter value, θ. The estimator is with narrow bandwidth for large θ but with wide bandwidth for small θ. This small θ gives long term average estimation(low frequency) of the fluctuating input variance well as short term variations (high frequency) of the input power. In our simulations of multipath communication channel equalization environments, the method with large θ has shown not as much improved convergence speed as the speed of the original IT-LMS algorithm. The proposed method with small θ=0.01 reach its minimum MSE in 100 samples whereas the IT-LMS converges in 200 samples. This shows the proposed, tap-power normalized IT-LMS algorithm can be applied more effectively to digital wireless communication systems.

• Nuclear Engineering and Technology
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• 제39권6호
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• pp.725-730
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• 2007

This paper describes recently developed techniques for effective filtering of neutron detector signal noise. In this paper, three kinds of noise filters are proposed and their performance is demonstrated for the estimation of reactivity. The tested filters are based on the unilateral kernel filter, unilateral kernel filter with adaptive bandwidth and bilateral filter to show their effectiveness in edge preservation. Filtering performance is compared with conventional low-pass and wavelet filters. The bilateral filter shows a remarkable improvement compared with unilateral kernel and wavelet filters. The effectiveness and simplicity of the unilateral kernel filter with adaptive bandwidth is also demonstrated by applying it to the reactivity measurement performed during reactor start-up physics tests.

• 대한전자공학회논문지
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• 제15권5호
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• pp.1-6
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• 1978

본 논문은 음성신호의 디지탈화와 대역식 축소에 관한 일부1)에 이은 이부 논문이다. 몇가지 근래에 개발된 Vocoding 방법, 즉 linear predictive coding (LPC), formant vocoding, residual excited linear prediction (RELP) vocoding,그리고 adaptive predictive coding(APC)에 관하여 논하였다. 본 논문에서는 음성전송에 있어서의 대역 제한 방법 중 지금 가장 효과가 있는 LPC방법을 중점적으로 취급하였다. 또한 현재 처하고 있는 문제점들과 해결책을 토의하였다.

• JSTS:Journal of Semiconductor Technology and Science
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• 제11권1호
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• pp.33-39
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• 2011

H.264 decoders usually have pipeline architecture by a macroblock or a 4 ${\times}$ 4 sub-block. The period of the pipeline is usually fixed to guarantee the operation in the worst case which results in many idle cycles and higher data bandwidth. Adaptive pipeline architecture for H.264 decoders has been proposed for efficient decoding and lower the requirement of the bandwidth for the memory bus. However, it requires a controller for the adaptive priority control to utilize the advantage. We propose a smart bus arbiter that replaces the controller. It is introduced to adjust the priority adaptively the QoS (Quality of Service) control of the decoding process. The smart arbiter can be integrated the arbiter of bus systems and it works when certain conditions are met so that it does not affect the original functions of the arbiter. An H.264 decoder using the proposed architecture is designed and implemented to verify the operation using an FPGA.

• 한국정보과학회논문지:소프트웨어및응용
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• 제31권7호
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• pp.901-910
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• 2004

본 연구에서는 DCT(Discrete Cosine Transform) 영역에서 불연속 특징을 검출하여 내용기반의 동적 비디오 전송을 위한 트랜스코딩 방법을 제안한다. 이 방법에 의하여 동영상 스트림 각각의 DCT블록은 내부의 대표적인 불연속의 크기에 따라 다르게 트랜스코딩되어 전송되며, 실험에서 같은 대역폭을 유지하면서 기존의 저주파 통과 필터에 의한 방법보다 내용기반 방법이 더 좋은 비디오 화질을 나타내며 픽셀 영역에서의 방법보다 처리 시간이 빠름을 나타낸다.

• 한국정보통신학회논문지
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• 제21권1호
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• pp.42-52
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• 2017

DASH는 네트워크 상황에 따라 클라이언트가 서버에게 멀티미디어 콘텐츠를 요청하고 이를 이용하여 멀티미디어 콘텐츠를 전송하는 적응형 스트리밍 기술이다. 이러한 구조에서 멀티미디어 콘텐츠 사용자에게 최선의 품질을 보장하기 위해서는 정확한 가용 대역폭 측정이 필요하다. 그런데 TCP의 전송 특성을 고려하지 않는 DASH는 이전 미디어 세그먼트 크기에 따라 측정된 가용 대역폭이 다르고 때문에 사용자에게 QoE를 보장하기 어렵다. 본 논문은 TCP Slow start구간을 가용 대역폭 측정에서 배제하여 가용대역폭 측정 오류를 줄이는 새로운 dash대역폭 측정 방법을 제안한다. 제안 방법은 이전 세그먼트 크기에 따라 가용 대역폭 측정의 결과가 달라지는 문제를 해결할 수 있다. 우리가 제안하는 가용 대역폭 측정 방법을 오픈 소스 기반 DASH시스템에서 구현하여 기존 대역폭 측정 방법과 성능을 비교 평가하였다. 성능 평가 결과 제안 방법은 기존 대역폭 측정 방법에 비해 정확도가 20% 향상되었다. 또 평균 세그먼트 서비스 품질, 세그먼트 품질 변경 횟수 등의 측면에서 사용자 QoE가 개선됨을 확인하였다.

• 한국위성정보통신학회논문지
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• 제10권4호
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• pp.136-140
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• 2015

본 논문에서는 적응형 배열 안테나에 적용되는 대표적 항재밍 기술로 알려진 STAP 및 SFAP 신호처리 방식의 시뮬레이션 성능을 비교 분석하였다. 시뮬레이션을 위해 두 방식의 가중 벡터(weighting vector)를 추정하는 방법으로 공통적으로 MMSE(Minimum Mean Square Error) 알고리즘을 사용하여 다양한 재밍환경에서 시뮬레이션을 통한 성능을 제시하였다. 특히, 재밍 전력 J/S(Jamming to Signal Power Ratio)에 따른 STAP 및 SFAP 성능 비교 분석, 신호대역에 대한 재밍 대역의 비율에 따른 성능 비교 분석 및 두 방식간 BER 성능을 비교하여 재밍 환경에 따른 항재밍 성능을 분석하였다.