• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.4
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• pp.80-87
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• 2007

In this paper, link-level performance of a OFDMA technique, which has been considered as a standard for 802.16, 802.20, 3G-LTE, WiMax, and WiBro, is evaluated for next-generation wireless communication systems. While many researches have focused on uncoded forward-link performances, this paper provides a coded performance of a reverse-link OFDMA system. Performance degradation due to time offset among reverse-link users and frequency offset during FFT process is investigated. Transmitter and receiver antenna diversity techniques are used to overcome performance degradation. Performance of a OFDMA system is compared with a CDMA system in terms of FER and throughput to emphasize the advantage of OFDMA system for a reverse-link. Finally, under given specification, Eb/No required to achieve the maximum throughput of a reverse-link is proposed considering various coded rates and antenna permutations.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.81-84
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• 2004

패킷망을 통한 음성 통신이 발달됨에 따라 패킷 스위칭 채널 환경에서 계층적 구조를 가지는 광대역 음성 부호화기의 개발에 대한 요구가 늘어나고 있다. 본 논문에서는 이러한 차세대 패킷 전화용 광대역 음성 부호화기의 상위 대역에 대해서 효율적인 양자화 방법을 제안한다. 먼저 전체 프레임을 다수의 짧은 부프레임으로 구분하고, 각각의 부프레임에 MLT(Modulated Lapped Transform)변환을 적용하여 주파수 영역으로 변환하여 2차원 구조의 데이터 행렬을 생성한다. 이러한 2차원 구조의 데이터를 크기와 부호로 분리하고, 크기는 2차원 DCT를 사용하여 시간과 주파수 영역에서의 신호 압축을 동시에 얻을 수 있게 하였다. 이와 같은 새로운 구조를 활용하여 기존의 방법보다 Energy Compaction 효과를 높이고 양자화 성능을 향상시킬 수 있었다. 또한 Core Layer의 부호화된 파라미터를 상위 대역의 양자화에 이용함으로써 그 성능을 향상시킬 수 있는 방법을 제안한다.

• Journal of Broadcast Engineering
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• v.7 no.1
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• pp.45-54
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• 2002

This paper proposes a video compression algorithm using characteristics of wavelet coefficients. The proposed algorithm can provide lowed bit rate and faster running time while guaranteeing the reconstructed image qualify by the human virtual system. In this approach, each video sequence is decomposed into a pyramid structure of subimages with various resolution to use multiresolution capability of discrete wavelet transform. Then similarities between two neighboring frames are obtained from a low-frequency subband which Includes an important information of an image and motion informations are extracted from the similarity criteria. Four legion selection filters are designed according to the similarity criteria and compression processes are carried out by encoding the coefficients In preservation legions and replacement regions of high-frequency subbands. Region selection filters classify the high-frequency subbands Into preservation regions and replacement regions based on the similarity criteria and the coefficients In replacement regions are replaced by that of a reference frame or reduced to zero according to block-based similarities between a reference frame and successive frames. Encoding is carried out by quantizing and arithmetic encoding the wavelet coefficients in preservation regions and replacement regions separately. A reference frame is updated at the bottom point If the curve of similarity rates looks like concave pattern. Simulation results show that the proposed algorithm provides high compression ratio with proper Image quality. It also outperforms the previous Milton's algorithm in an Image quality, compression ratio and running time, leading to compression ratio less than 0.2bpp. PSNR of 32 dB and running tome of 10ms for a standard video image of size 352${\times}$240 pixels.

• Journal of the Institute of Convergence Signal Processing
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• v.2 no.2
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• pp.20-27
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• 2001

3D image coding and transmission technologies are essential for transmitting 3D image information through the limited transmission channel, and very important in the area of information and telecommunication services using 3D image media. In this paper, we propose the 3D image coding and transmission methods welch use disparity information and MPEG-2's scalability. Left and Fight image sequences are coded in base layer and enhancement layer, respectively. The enhancement layer contains the disparity and prediction informations. In order to reduce searching time for disparity information, we propose a method in which the disparity information of previous image paid is used in the next image pair, and we adopt the bit rate control method proposed in MPEG-2 TM6. An experimental result showed that the proposed searching method considerably reduced the entire encoding time and reached the desired bit rates.

• Journal of Broadcast Engineering
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• v.5 no.2
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• pp.227-238
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• 2000

This paper, at first, provides analysis on loss pattern of Internet based on real experiments of the current Internet. Then, we propose an effective adaptive QoS management technique, in which measured loss pattern as well as PLR (Packet Loss Ratio) are used to select titrate of temporal scalability. level of FEC and retransmission This selection is also incorporated to the MPEG-4 error resilience tools and error concealment techniques. In order to minimize effect of packet loss, multimedia stream is segmented in the unit of group of Pictures (GOP) and interleaving and FEC are applied to the segment. Proposed algorithms are applied to build a VOD system.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.8A
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• pp.859-867
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• 2004

To provide high-quality media streaming service over the best-effort Internet, a streaming methodology is required to response to the dynamic fluctuation of underlying networks. In this paper, we implement the MPEG-4 streaming system with adaptive transport based on priorities of media packets. The implemented system is composed of the common MPEG-4 streaming components such as elementary stream provider, sync and DMIF layer, and adaptive transport module including data prioritization and FEC control. More specifically, the prioritized sync layer packets (based on object level) are delivered to the transport module and then are encoded by an adaptive FEC encoder to help reliable transport. The FEC combination is dynamically adjusted by the feedback information from the receiver. In addition, low priority packets are selectively dropped to meet the limitation of available bandwidth. The experimental results over the emulation-based testbed show that the Proposed system can mitigate the impact of network fluctuation and thus improve the quality of streaming.

• Journal of Broadcast Engineering
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• v.2 no.2
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• pp.169-181
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• 1997

In this paper, we propose and discuss about image enhancement techniques for MPEG-4. which represents very low bit-rate, content-based. and object-based hierarchical audio-visual coding standard. The proposed enhancement technique removes undesired artifacts arising in the compression procedure and increase resolution in both spatial and temporal domains. In order to remove undesired artifacts. we divide the MPEG-4 video algorithm in two parts: MPEG-2 like part and the new part. For removing artifacts caused by the first part. we adopt the conventional blocking artifacts algorithm developed for MPEG-2. On the other hand for removing artifacts caused by the second part. we provide a new degradation model. and propose the corresponding image restoration method. For increasing resolution of the MPEG-4 images, we propose a general framework of multichannel image interpolation process. which includes both spatial and temporal interpolations. As the MPEG-4 standard is under development. various sophisticated techniques are considered. but research on image enhancement techniques is relatively underestimated. By this reason. additional image enhancement techniques will become very important issue in realization phase of MPEG-4.

• Journal of Broadcast Engineering
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• v.12 no.5
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• pp.491-502
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• 2007

In this parer, we propose an efficient intra prediction mode decision scheme in Scalable Video Coding(SVC) which is an emerging video coding standard as an extension of H.264/MPEG-4 AVC(Advanced Video Coding). The proposed method in base on the characteristic of macroblock smoothness follows the statistical analysis of intra prediction mode in an enhancement layer and it decides a candidate intra prediction mode. We also propose an early termination scheme for Intra_BL mode decision where the RD cost value of Intra_BL is utilized. Simulation results show that the proposed method reduces 54.67% of the computation complexity of intra prediction coding, while the degradation in video quality is negligible; for low QP values, the average PSNR loss is very negligible, equivalently the bit rate increases by 0.011%. For high QP values, the average PSNR loss is less than 0.01dB, which equals to 0.249% increase in bitrate.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39C no.5
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• pp.435-446
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• 2014

Even if it is important to transmit high rate multimedia information without any transmission errors for surveillance systems, it is difficult to achieve error-free transmission due to infra-less adhoc networks. In order to reduce the transmission errors furthermore, additional signal overheads or retransmission of signals should be required, but they may lead to transmission delay. This paper represents a study on low delay and adaptive video transmission for the unmanned surveillance systems by developing system protocols. In addition, we introduce an efficient and adaptive control algorithm using system parameters for exploiting unmanned surveillance system properly over multi-channels.

• Journal of Broadcast Engineering
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• v.14 no.6
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• pp.796-806
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• 2009

SVC (Scalable Video Coding) is a new video coding standard to provide convergence media service in heterogeneous environments with different networks and diverse terminals through spatial-temporal-quality combined flexible scalabilities. This paper presents the performance analysis on packet loss in the delivery of SVC over IP networks and an efficient adaptation method to packet loss caused by buffer overflow. In particular, SVC with MGS (Medium Grained Scalability) as well as spatial and temporal scalabilities is addressed in the consideration of packet-based adaptation since finer adaptation is possible with a sufficient numbers of quality layers in MGS. The effect on spatio-temporal quality due to the packet loss of SVC with MGS is evaluated. In order to minimize quality degradation resulted by packet loss, the proposed adaptation of MGS based SVC first sets adaptation unit of AU (Access Unit) or GOP corresponding to allowed delay and then selectively discards packets in order of importance in terms of layer dependency. In the experiment, the effects of packet loss on quantitative qualities are analyzed and the effectiveness of the proposed adaptation to packet loss is shown.