• The Journal of the Acoustical Society of Korea
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• v.12 no.1E
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• pp.32-38
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• 1993

본 논문에서는 저 전송속도의 음성 부호화기를 제안하였고 컴퓨터 시뮬레이션을 통하여 성능분석과 유연성을 입증하였다. 제안된 부호화 방식은 입력 음성신호의 Stationarity에 따라 전송 프레임의 길이를 가변하고, 전송 프레임의 대표적인 특징 벡터를 Vector Quatization으로 부호화하였다. 제안된 부호화 방식에서 특징 벡터열은 입력 음성신호를 샘플단위로 Prewindowed RLS Lattice 알고리즘을 통해 구한 PARCOR 계수로 구성된다. 입력 음성신호는 Subsegment로 분할되고, 각 Subsegment에서 대표적인 PARCOR 계수를 구한다. Likelihood Ratio Distortion Measure를 사용하여 유사도에 따라 Subsegment를 병합함으로써 전송프레임을 결정한다. 컴퓨터 시뮬레이션 결과로부터 제안된 VTEL 음성 부호화 방식은 좋은 음질을 유지하면서 전체 전송속도를 크게 줄일 수 있다.

• ETRI Journal
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• v.13 no.3
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• pp.3-13
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• 1991

본 논문에서는 FDDI(Fiber Distributed Data Interface) 프로토콜의 성능분석을 위한 시뮬레이션 모델을 제안하며, 그 활용방안을 제시하였다. FDDI는 ANSI의 X3T9.5 위원회를 통해 표준화가 진행되어 온 프로토콜로서, 광섬유를 전송매체로 하여 100 Mbps의 전송속도에서 동작하는 고속 근거리망이다. FDDI는 IEEE 802.5 표준인 토큰 링 방식을 기초로 하였지만, 광섬유 링크상에서의 고속 데이터 전송을 가능케하기 위하여 기존의 토큰 링 방식에 존재하지 않는 TTR(Timed Token Rotation) 프로토콜을 사용한다. 본 논문에서는 동기 및 비동기 프레임의 전송 지연시간, 큐의 평균 길이, 전송된 프레임의 수(공정성), FDDI의 실시간 특성, 프레임 처리율,FDDI 토큰링의 효율 등을 측정하여 FDDI 토큰링의 성능을 분석하고 이에 따른 활용방안을 제시하였다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.6 no.2
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• pp.236-247
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• 2002

We propose a cell－multiplexing scheme for the real－time communication service in ATM network and a new service discipline guarantee end-to-end delay based on pseudo-isochronous cell switching. The proposed scheme consists of two level frame hierarchy, upper and lower frame, which is used to assign the bandwidth and to guarantee the requested delay bound, respectively. Since the Proposed algorithm employs two level frame hierarchy, it can overcome the coupling problem which is inherent to the framing strategy It can be comparatively reduce the complexity, and still guarantee the diverse delay bounds of end-to-end. Besides, it consists of two components, traffic controller and scheduller, as the imput traffic description model and regulates the input traffic specification. The function of the traffic controller is to shape real -time traffic to have the same input pattern at every switch along the path. The end-to-end delay is bounded by the scheduller which can limit the delay variation without using per-session jitter controllers, and therefore it can decrease the required buffer size. The proposed algorithm can support the QoS's of non-real time traffic as well as those of real time traffic.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.5C
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• pp.445-453
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• 2002

We propose a cell-multiplexing scheme for the real-time communication service in ATM network and a new service discipline guarantee end-to-end delay based on pseudo-isochronous cell switching. The proposed scheme consists of two level frame hierarchy, upper and lower frame, which is used to assign the bandwidth and to guarantee the requested delay bound, respectively. Since the proposed algorithm employs two level frame hierarchy, it can overcome the coupling problem which is inherent to the framing strategy. The proposed scheme consists of two components, traffic controller and scheduller, as the imput traffic description model and regulates the input traffic specification. The function of the traffic controller is to shape real-time traffic to have the same input pattern at every switch along the path. The end-to-end delay is bounded by the scheduller which can limit the delay variation without using per-session jitter controllers, and therefore it can decrease the required buffer size. The proposed algorithm can support the QoS's of non-real time traffic as well as those of real time traffic

• The Journal of the Acoustical Society of Korea
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• v.41 no.2
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• pp.122-129
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• 2022

In this paper, we propose a method to improve the performance of Transformer-based speech recognizers using data augmentation that locally adjusts the frame rate. First, the start time and length of the part to be augmented in the original voice data are randomly selected. Then, the frame rate of the selected part is changed to a new frame rate by using linear interpolation. Experimental results using the Wall Street Journal and LibriSpeech speech databases showed that the convergence time took longer than the baseline, but the recognition accuracy was improved in most cases. In order to further improve the performance, various parameters such as the length and the speed of the selected parts were optimized. The proposed method was shown to achieve relative performance improvement of 11.8 % and 14.9 % compared with the baseline in the Wall Street Journal and LibriSpeech speech databases, respectively.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.6
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• pp.1117-1119
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• 2015

In this letter, we propose a mathematical model of variable-length payloads transmitted in EDCA and transmitted using MU-MIMO. Assuming fixed-length or the use of mean value of payload length leads to discordant results while calculating the total payload length of variable-length frames transmitted within a fixed duration. Using the proposed model results in accurate results (less than 3% relative errors) for total payload length under variable-length traffic.

• Journal of the Computational Structural Engineering Institute of Korea
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• v.18 no.2
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• pp.169-179
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• 2005

An improved method for evaluating effective buckling lengths of beam-column members in plane frames is newly proposed based on system inelastic buckling analysis. To this end, the tangent stiffness matrix of be am-column elements is first calculated using stability functions and then the inelastic buckling analysis method is presented. The scheme for determining effective length of individual members is also addressed. Design examples and numerical results ?uc presented to show the validity of the proposed method.

• The KIPS Transactions:PartC
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• v.18C no.6
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• pp.433-440
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• 2011

In this paper, we propose a new scheduling method that can guarantee reliable jitter by minimizing the queue length variation in the streaming service provisioning such as IPTV and VoD. The amount of traffic to be delivered within a certain time is very fluid because MPEG-4 and H.264 encoders use VBR(Variable Bit Rate) for delivering video streaming traffic. This VBR characteristic increases the end-to-end propagation delay variation when existing scheduling methods are used for delivering video frames. Therefore, we propose the new scheduling method that can minimize change rate of queue length by adaptively controling service rate taking into account the size of bulk incoming packets and arrival rate for bulk streaming traffic. Video frames can be more reliably transmitted through the minimization of the queue length variation using the proposed method. We use the queueing model and also carry out OPNET simulation to validate the proposed method.

• The Journal of the Korea Contents Association
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• v.8 no.5
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• pp.52-60
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• 2008

A simple Time Division Multiplex(TDM) switching system which has been widely in satellite networks provides any size of bandwidth for a number of low bandwidth subscribers by allocating proper number of time slots in a frame. In this paper, we propose a new approach based on graph coloring model for efficient time slot assignment algorithm in contrast to network flow model in previous works. When the frame length of an initial matrix of time slot requests is 2's power, this matrix is divided into two matrices of time slot requests using binary divide and conquer method based on the graph coloring model. This process is continued until resulting matrices of time slot requests are of length one. While the most efficient algorithm proposed in the literature has time complexity of $O(N^{4.5})$, the time complexity of the proposed algorithm is $O(NLlog_2L)$, where N is the number of input/output links and L is the number of time slot alloted to each link in the frame.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1994.06a
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• pp.447-452
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• 1994