• The Journal of the Acoustical Society of Korea
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• 제19권2E호
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• pp.8-13
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• 2000

In this paper, we present an adaptive formant tracking algorithm for speech using closed phase WRLS-VFF-VT method. The pitch synchronous closed phase methods is known to give more accurate estimates of the vocal tract parameters than the pitch asynchronous method. However the use of a pitch-synchronous closed phase analysis method has been limited due to difficulties associated with the task of accurately isolating the closed phase region in successive periods of speech. Therefore we have implemented the pitch synchronous closed phase WRLS-VFF-VT algorithm for speech analysis, especially for formant tracking. The proposed algorithm with the variable threshold(VT) can provide a superior performance in the boundary of phone and voiced/unvoiced sound. The proposed method is experimentally compared with the other method such as two channel CPC method by using synthetic waveform and real speech data. From the experimental results, we found that the block data processing techniques, such as the two-channel CPC, gave reasonable estimates of the formant/antiformant. However, the data windows used by these methods included the effects of the periodic excitation pulses, which affected the accuracy of the estimated formants. On the other hand the proposed WRLS-VFF-VT method, which eliminated the influence of the pulse excitation by using an input estimation as part of the algorithm, gave very accurate formant/bandwidth estimates and good spectral matching.

• JSTS:Journal of Semiconductor Technology and Science
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• 제13권4호
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• pp.318-330
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• 2013

This paper presents a high-linearity low-noise small-size programmable gain amplifier (PGA) based on a new low-noise low-distortion differential amplifier and a proposed reconfiguration technique. The proposed differential amplifier combines an inverter-based differential pair with an adaptive biasing circuit to reduce noise and distortion. The reconfiguration technique saves the chip size by half by utilizing the same differential pair for the input transconductance and load-stage, interchangeably. Fabricated in $0.18-{\mu}m$ CMOS, the proposed PGA shows a dB-linear control range of 21dB in 16 steps from -11 dB to 10 dB with a gain error of less than ${\pm}0.33$ dB, an IIP3 of 7.4~14.5 dBm, a P1dB of -7~1.2 dBm, a noise figure of 13dB, and a 3-dB bandwidth of 270MHz at the maximum gain, respectively. The PGA occupies a chip area of $0.04mm^2$ and consumes only 1.3 mA from the 1.8 V supply.

• 한국전자파학회논문지
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• 제26권11호
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• pp.978-984
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• 2015

본 논문에서는 유무인항공기에서 사용될 통달거리 200 km, 전송속도 45 Mbps급 데이터 전송장비의 RF 송수신기에 대해 다룬다. RF 송수신기는 Ku 대역에서 동작하도록 설계되었으며, 상/하향 변환모듈, 고출력 증폭 모듈, 전단 처리 모듈로 구성된다. 제안된 RF 송수신기의 성능을 만족시키기 위해 출력신호 레벨이 10 W급인 전력 증폭기가 요구되었으며, 이는 GaN 소자기반의 전력증폭기로 구현되었다. 또한, 적응형 전송속도 변환기능이 요구되었고, 이를 위해 다양한 대역폭의 신호를 수신할 때의 상호 주파수 간섭을 최소화하기 위하여 6종의 SAW 필터로 구성된 수신기 구조가 적용되었다. 시스템 요구사항의 만족 여부를 확인하기 위해 AWR 시뮬레이션 툴을 이용하였다.

• Journal of Electrical Engineering and Technology
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• 제10권3호
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• pp.1319-1327
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• 2015

As wind power farm (WPF) installations continue to grow, monitoring and controlling large-scale WPFs presents new challenges. In this paper, a hierarchical network architecture is proposed in order to provide remote monitoring and control of large-scale WPFs. The network architecture consists of three levels, including the WPF comprised of wind turbines and meteorological towers, local control center (LCC) responsible for remote monitoring and control of wind turbines, and a central control center (CCC) that offers data collection and aggregation of many WPFs. Different scenarios are considered in order to evaluate the performance of the WPF communications network with its hierarchical architecture. The communications network within the WPF is regarded as the local area network (LAN) while the communication among the LCCs and the CCC happens through a wide area network (WAN). We develop a communications network model based on an OPNET modeler, and the network performance is evaluated with respect to the link bandwidth and the end-to-end delay measured for various applications. As a result, this work contributes to the design of communications networks for large-scale WPFs.

• 한국정보처리학회논문지
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• 제7권4호
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• pp.1305-1315
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• 2000

It is known that a digital image enlargement technique can increase the size of he image but the practical enhancement of resolution is trifle because the frequency bandwidth of the original image is basically limited. To solve this problem, this paper proposes the digital image enlargement technique which interpolates the interpolation points of horizontal and vertical direction by weighting according to the direction of edge information with the component of FOI(First Order Interpolation)and output of the pseudomedian filter for image enlargement and interpolates the interpolation points of diagonal direction by selectively transposing the direction of the subwindows of the pseudomedian filter according to the distribution of neighbored pixels thereto in the extended image. According to the proposed methods, the digital image enlargement which preserves the characteristic of the pseudomedian filter capable of keeping the reconstruction of edge information and reflects the advantage of FOI can be performed. Therefore, visual artifacts could be effectively suppressed, and most characteristics and shape of the original image can be reconstructed as well.

• 한국통신학회논문지
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• 제29권2A호
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• pp.140-146
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• 2004

본 논문은 다중사용자를 위한 OFDMA/FDD(Orthogonal Frequency Division Multiple Access/Frequency Division Duplex) 시스템에서 가변적인 전송율로 전송하기위한 무선 채널의 피드백 정보를 감소하는 방안을 제시한다. 제한된 상향링크 제어채널을 통해 기지국으로 무선 채널 정보를 전송하기 위해서, 제안된 알고리즘은 인근 클러스터간의 무선 채널의 유사 정도를 나타내는 정보로 채널 변화 수준(channel variation level)을 사용하며, 모든 클러스터에 대해서 하나의 MCS(Modulation & Coding Scheme)을 사용한다. 시뮬레이션은 하나의 셀 내에서 수행되었으며, 기존의 알고리즘의 피드백 정보와 유사한 오버헤드로 대역 효율 및 outage probability에서 더 좋은 성능을 나타내었다.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제9권11호
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• pp.4419-4435
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• 2015

The use of an efficient packet scheduling scheme for a forward link in satellite communication networks is very important to support fairness for each return channel satellite terminal (RCST) and the service differentiations for user traffics. To support fairness and QoS for each RCST with service-level agreement (SLA), the Adaptive Coding and Modulation (ACM) system in a satellite hub has to process packets with considering modulation and coding (MODCOD) and packet types. Although a DVB-S2 system with ACM scheme has higher transmission efficiency, it cannot offer fairness or quality of service (QoS) to RCSTs. Because the data are transmitted with high MODCOD in regions with clear skies, while data are transmitted using low MODCOD in regions experiencing rain events. In this paper, we propose a two-step scheduling scheme offering fairness and QoS to RCSTs, while minimizing a decrease in throughput. The proposed scheme is carried out performance evaluations using a computer simulation. As results of this simulation, the proposed scheduler was shown to support bandwidth fairness to an individual RCST, and provide a level of QoS differentiation for user traffics.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제9권8호
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• pp.3121-3135
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• 2015

Owing to the increase in consumption of multimedia content and the improvement of device capacity, user demand for high-quality content has increased. However, it is difficult to transport such large amounts of content over the existing broadcasting network with limited bandwidth. To provide high-definition broadcasting, some studies suggest methods of transporting multimedia over heterogeneous networks after encoding content hierarchically. MPEG Media Transport (MMT), standardized by Moving Picture Experts Group (MPEG), is a solution that enables large-volume media transport over heterogeneous networks such as digital broadcasting networks and packet-switched networks. In the case of delivering a scalable encoded video over different networks, synchronization of each stream is an important issue. MMT defines a synchronization scheme, but does not contain sufficient functions to implement it. In this paper, we propose a synchronization scheme for media streams that are encoded hierarchically, divided into layers, and transported over heterogeneous networks. We implement our scheme using MMT and HTTP, and experimented using three encapsulated video streams with different durations. As a result, we show that the proposed scheme can reduce the waiting time to display high-quality video, relative to Dynamic Adaptive Streaming over HTTP-Scalable Video Coding (DASH-SVC) by requesting segments of enhanced layers after calculating the transmission time. Additionally, we find out that the selection of durations have a relation to the characteristics of the video.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제10권4호
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• pp.1768-1789
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• 2016

In contemporary video streaming systems based on HTTP protocol, video players at the client side are responsible for adjusting video quality to network conditions and user expectations. However, when multiple video clips are streamed simultaneously, an intricate application logic implemented in the video players overlays the TCP mechanism which is responsible for a balanced access to a shared network link. As a result, some video players may not obtain a fair share of network throughput and may be vulnerable to an unstable video bit-rate. Therefore, we propose to simplify the algorithms implemented in the video players, which are responsible for the adjustment of video quality and constrain their functionality only to sending feedback to a server about a state of the player buffer. The main logic of the system is shifted to the server, which is now responsible for bit-rate selection and prioritisation of the video streams transmitted to multiple clients. To verify our proposition, we performed several experiments in a laboratory environment which show that when the server cooperates with the clients, the video players experience fewer quality switches and the system achieves better fairness when allocating network throughput among the video players. However, this comes at the cost of worse utilisation of network bandwidth.

• 정보과학회 컴퓨팅의 실제 논문지
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• 제21권1호
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• pp.82-87
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• 2015

HTTP 적응적 스트리밍 서비스의 품질을 향상시키기 위한 연구가 활발히 진행되고 있다. 그러나 기존 HTTP 적응적 스트리밍 기술은 VBR 비디오 전송에 대한 고려가 부족하여, 스트리밍 서비스의 QoE가 저하되는 문제점이 있다. 본 논문에서는 이러한 문제점을 해결하기 위해, CB-DASH (Content and Buffer-aware DASH)라는 DASH 기반의 새로운 품질 적응 기법을 제안한다. 제안하는 기법은 비디오의 VBR 특성과 클라이언트 버퍼 상태를 고려하여 비디오 품질을 조절한다. 시뮬레이션 결과를 통해 제안한 기법의 정확한 대역폭 예측 및 QoE 향상을 확인하였다.