• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.165-168
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• 2004

본 논문은 방송 및 통신 환경을 이용한 멀티채널 음향 재생 환경에서 다수 객체의 음상 정위를 보다 효과적이고 효율적으로 하기 위한 방법에 관한 것이다. 본 논문에서는 전송되는 정보의 양을 최소화 하면서도 재생되는 음향공간에서는 다수의 음향 객체들이 충실하게 재생되어 자연스러운 음향공간이 재현할 수 있는 방법을 제시하고가 한다. 기존 방법의 경우, 전송 선로를 통해 음원을 압축하여 전송하기 위해서는 먼저 멀티채널 신호를 합한 모노신호와 채널 신호사이의 음량차이(ICLD), 시간지연 차이(ICTD), 상관도(ICC)등을 전송하는 양귀단서 신호화 기술(Binaural Cue Coding)을 이용하고 있다. 본 논문에서는 멀티채널 음원을 분석하여, 음원의 가상 위치정보를 벡터적으로 표현하고, 이 위치벡터와 멀티채널 음원을 하나의 모노 음원으로 다운 믹스한 신호를 전송함으로써 전송 효율을 극대화 한 압축 재생 방법을 제시한다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2013.06a
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• pp.176-179
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• 2013

본 논문에서는 JCT-VC 에서 2013 년 1 월에 표준화가 완료된 High Efficiency Video Coding (HEVC)과 구글에서 2013 년 6 월에 개발 완료 예정인 VP9 의 압축 효율 비교를 수행한다. HEVC 는 UHD 등 고화질 방송 등에 대응하도록 디자인 되었으며, VP9 은 유튜브 (YouTube) 등과 같은 인터넷 비디오 스트리밍에 적합하도록 디자인되었다. VP9 의 경우 HEVC 와는 달리 로열티 프리 (royalty-free)를 지향하며 오픈소스 (open source) 방식으로 개발이 진행되고 있다. 본 논문에서는 HEVC 와 VP9 의 디자인 차별점을 소개하고, 랜덤 액세스 환경(Random Access, RA)과 저지연 환경 (Low Delay, LD)에서 HEVC 와 VP9 의 압축 효율을 비교한다. 실험 결과에 따르면, 방송 및 패키지 미디어 등에서 많이 사용될 랜덤 액세스 환경에서는 VP9 이 HEVC 대비 32.7% 열세를 보인다. 비디오 컨퍼런스등과 같은 저지연 환경에서는 VP9 이 HEVC 대비 26.7% 열세를 보인다. VP9 의 경우 개발이 완료된 것이 아니므로, 향후 압축 효율의 향상이 있을 것으로 기대된다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2008.02a
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• pp.159-162
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• 2008

이 연구는 Ka대역 위성방송 서비스를 제공할 때 발생하는 강우감쇠문제를 해결하기 위하여 MPEG-4 SVC 신호를 이용하는 방안에 대하여 고찰하였다. Ka대역 위성방송시스템은 DVB-S2 VCM 모드를 고려하였으며, JSCC (Joint Source Channel Coding) 기법을 이용하여, SVC신호를 Ka대역 위성방송시스템에 적용하였다. SVC신호는 Spatial Scalability, SNR Scalability 및 Temporal Scalability로 구분되어서, PSNR값의 변화에 따른 비트율 변화정도를 분석하였다. 비트율 변화율이 가장 큰 Spatial Scalability를 적용한 SVC신호가 Ka대역 위성방송서비스의 강우감쇠 문제를 해결하기 위한 방안으로 제안되었으며, 이에 대한 분석이 수행되었다.

• The Transactions of the Korea Information Processing Society
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• v.7 no.4
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• pp.1299-1304
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• 2000

In a speed coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech quality in case coexist with a voiced and an unvoiced consonants in a frame. So, I propose TSIUVC(Transition Segment Including UnVoiced Consonant) searching, extraction ad approximation-synthesis method in order to uncoexistent with a voiced and unvoiced consonants in a frame. This method based on a zerocrossing rate and pitch detector using FIR-STREAK Digital Filter. As a result, the extraction rates of TSIUVC are 84.8% (plosive), 94.9%(fricative), 92.3%(affricative) in female voice, and 88%(plosive), 94.9%(fricative), 92.3%(affricative) in male voice respectively, Also, I obain a high quality approximation-synthesis waveforms within TSIUVC by using frequency information of 0.547kHz below and 2.813kHz above. This method has the capability of being applied to speech coding of low bit rate, speech analysis and speech synthesis.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.7 no.12
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• pp.3149-3165
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• 2013

Content based image retrieval is increasingly gaining popularity among image repository systems as images are a big source of digital communication and information sharing. Identification of image content is done through feature extraction which is the key operation for a successful content based image retrieval system. In this paper content based image retrieval system has been developed by adopting a strategy of combining multiple features of shape, color and relevance feedback. Shape is served as a primary operation to identify images whereas color and relevance feedback have been used as supporting features to make the system more efficient and accurate. Shape features are estimated through second derivative, least square polynomial and shapes coding methods. Color is estimated through max-min mean of neighborhood intensities. A new technique has been introduced for relevance feedback without bothering the user.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.32B no.11
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• pp.1383-1392
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• 1995

In this paper, the method to incorporate TCVQ(Trellis Copded Vector Quantizer) into the encoding of the wavelet trans formed(WT) image followed by a variable length coding(VLC) or an entropy coding(EC) is considered. By WT, an original image is separated into 10 bands with various resolutions and directional components. TCVQ used to compress these WT coefficients is a finite state machine that encodes the input source on the basis of the current input and the current state. Wavelet basis used in this paper is designed by orthogonal spline function. A modified set partitioning algorithm to Wang's is also presented. A simple modification to Wang's algorithm gives a highly time-efficient result. Proposed WT-TCVQ encoder shows a very competitive result, giving 37.46dB in PSNR at 1.002bpp when encoding 512$\times$512 LENA.

• Journal of Biomedical Engineering Research
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• v.15 no.1
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• pp.83-88
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• 1994

The transform trellis code is an optimal source code as a block size and the constraint length of a shift register go to infinite for stationary Gaussian sources with the squared-error distortion measure. However to implement this code, we have to choose the finite block size and constraint length. Moreover real-world sources are inherently non stationary. To overcome these difficulties, we developed a training algorithm for the transform trellis code. The trained transform trellis code which uses the same rates to each block led to a variation in the resulting distortion from one block to another. To alleviate this non-uniformity in the encoded image, we constructed clusters from the variance of the training data and assigned different rates for each cluster.

• Journal of Korea Multimedia Society
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• v.3 no.2
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• pp.164-173
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• 2000

This paper presents an efficient bandwidth allocation method for multiple source in the two-layer video coding of different spatial resolution. We first investigate the model of bitrate distortion in the MPEG-2 spacial scalable coding,. By using approximated model parameters, than we propose an efficient bitrate control method in order to keep the same distortion level among coders and the constant quality ratio between layers. Simulation results show that the proposed method can satify the user requirement in comparison to the conventional method.

• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.4
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• pp.753-760
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• 2012

This paper is to show the character coding rule in computer and information equipment etc to improve the transmission efficiency in telecommunications. In the transmission system, the transmission efficiency can be increased by applying the proper character coding method. In datalink layer, HDSL ptotocol use FLAG byte to identify the frame to frame which consists of data bit stream and other control bytes. FLAG byte constits of "01111110". When data bit stream consist of the consecutive 5-bit "1" after "0", the decoder can not distinguish whether the data bit sequence is flag bit stream or data bit stream. To solve the problem, when the line coder in transmitter detects the consecutive 5-bits "1" after "0" in the input data stream, inserts violently the "0" after 5th "1" of the consecutive 5-bit "1" after "0". As a result, when the characters are decoded with the above procedure, the efficiency of system should be decreased. This paper shows the character code rule to minimize the consecutive 5-bits "1" after "0" when the code is given to each characters.

• Journal of Korea Multimedia Society
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• v.7 no.9
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• pp.1233-1239
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• 2004

In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech qualify in case coexist with a voiced and an unvoiced consonants in a frame. This paper present a new multi-pulse coding method by using V/S/TSIUVC switching, individual pitch pulses and TSIUVC approximation-synthesis method in order to restrict a distortion of speech quality. The TSIUVC is extracted by using the zero crossing rate and individual pitch pulse. And the TSIUVC extraction rate was 91% for female voice and 96.2% for male voice respectively. The important thing is that the frequency information of 0.347kHz below and 2.813kHz above can be made with high quality synthesis waveform within TSIUVC. I evaluate the MPC use V/UV and the FBD-MPC use V/S/TSIUVC. As a result, I knew that synthesis speech of the FBD-MPC was better in speech quality than synthesis speech of the MPC.