• KSII Transactions on Internet and Information Systems (TIIS)
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• v.6 no.4
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• pp.1128-1139
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• 2012

With the development of three-dimensional display and related technologies, depth video coding becomes a new topic and attracts great attention from industries and research institutes. Because (1) the depth video is not a sequence of images for final viewing by end users but an aid for rendering, and (2) depth video is simpler than the corresponding color video, fast algorithm for depth video is necessary and possible to reduce the computational burden of the encoder. This paper proposes a fast mode decision algorithm for depth video coding based on depth segmentation. Firstly, based on depth perception, the depth video is segmented into three regions: edge, foreground and background. Then, different mode candidates are searched to decide the encoding macroblock mode. Finally, encoding time, bit rate and video quality of virtual view of the proposed algorithm are tested. Experimental results show that the proposed algorithm save encoding time ranging from 82.49% to 93.21% with negligible quality degradation of rendered virtual view image and bit rate increment.

• Journal of Communications and Networks
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• v.7 no.2
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• pp.144-156
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• 2005

Video streaming over wireless networks is challenging due to node mobility and high channel error rate. In this paper, we propose a multi-source video streaming (MUVIS) system to support high quality video streaming service over IEEE 802.1l-based wireless networks. We begin by collocating a streaming proxy with the wireless access point to help leverage both the media server and peers in the WLAN. By tracking the peer mobility patterns and performing content discovery among peers, we construct a multi-source sender group and stream video using a rate-distortion optimized scheme. We formulate such a multi-source streaming scenario as a combinatorial packet scheduling problem and introduce the concept of asynchronous clocks to decouple the problem into three steps. First, we decide the membership of the multisource sender group based on the mobility pattern tracking, available video content in each peer and the bandwidth each peer allocates to the multi-source streaming service. Then, we select one sender from the sender group in each optimization instance using asynchronous clocks. Finally, we apply the point-to-point rate-distortion optimization framework between the selected sender-receiver pair. In addition, we implement two different caching strategies, simple caching simple fetching (SCSF) and distortion minimized smart caching (DMSC), in the proxy to investigate the effect of caching on the streaming performance. To design more realistic simulation models, we use the empirical results from corporate wireless networks to generate node mobility. Simulation results show that our proposed multi-source streaming scheme has better performance than the traditional server-only streaming scheme and that proxy-based caching can potentially improve video streaming performance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.8
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• pp.2010-2024
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• 1998

In video coding, it is important to improve the average picture quality as well as to maintain cosistent picture quality between consecutive pictures. In this paper, we propose a distortion-rate estimation method for MPEG-2 video and a forward rate control method, using the proposed estimation result, to be able to obtain the improved and consistent picture quality of CBR (Constant Bit Rate) encoded MPEG-2 video. The proposed distortion-rate estimation enable us to predict the distortion and the bits generated from an encoded picture at a given quantization step size and vice versa. The most attactive features of proposed distortion-rate estimation are its accuracy and low computational complexity enough to be applied to the practical video coding. In addition, the proposed rate control first determined a quantization parameter per frame by following procedure: distortion-rate estimation, target bit allocation, distortion constraint and VBV(Video Buffer Verification) constraint. And then this quantization parameter is applied to the encoding so that improved and consisten picture quality can be obtained. Furthermore the proposed rate control method can solve the error propagation problem caused by scene change or anchor picture degradation by using the B-picture skipping and the guarantee of the minimum bit allocation for the anchor picture. Experimental results, comparing the proposed forward rate control method with TM5 method, show that the proposed method makes more improed and consistent picture quality than TM5.

• The Transactions of the Korea Information Processing Society
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• v.4 no.10
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• pp.2629-2639
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• 1997

The video coding for very low bit-rate has recently received considerable attention, but the conventional coding schemes with block based transform suffer from the blocky effect for the constraints of limited bit-rate. In this paper, we present a video coding system based on wavelet transform and multiresolution motion estimation/compensation for very low bit-rate video. The proposed scheme uses the wavelet transform which is flexible to represent non-stationary image signals and adaptable to the human visual characteristics. The wavelet transformed coefficients are coded by various coding modes in accordance with the sum of absolute error after motion estimation/compensation in wavelet decomposed domain. And simple buffer control technique is applied to handle constant image quality. It is shown that the presented scheme has more acceptable image quality without blocky effects than conventional block based transform video coding.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.3
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• pp.637-644
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• 2004

This paper proposed and implemented the adaptive transmission middleware for video streaming, which is able to support the adaptive transmission of video data to the fluctuating changes of network environment in the packet-based network and the properties of transmitted video data. The adaptive transmission middleware is made up SR-RTP-based transfer module and TFRC(TCP Friendly Rate Control)-based transfer-rate control module. The SR-RTP-based transfer module supports RTP-based real-time transfer of video data and packet retransmission scheme retransmitting the high-priority packets selectively in the damaged video data to reduce the error induced by the packet loss. Sharing the transmission bandwidth of network with the TCP-based data transfer, the TFRC-based transfer-rate control module controls the transfer rate of video data according to the most allowable transmission bandwidth in the network, so that the transfer rate is controlled adaptively to the fluctuating changes of transmission bandwidth. This paper, for the experiment, applied the adaptive transmission middleware to video streaming in the external Internet environment, and analyzed the effective frame transfer rate and the degree of the streaming jitter to evaluate the performance of packet-loss recovery and adaptive transfer rate control. In the external Internet environment where the packet-loss rate is high a bit, the relatively high streaming performance was showed compared with the case that didn't apply the adaptive transmission middleware.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.7 no.6
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• pp.1129-1133
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• 2006

In this thesis, we propose a semantics-based video retrieval system which supports semantics-retrieval for various users of massive video data. Proposed system automatically processes the extraction of contents information which video data has and retrieval process using agent which integrate annotation-based retrieval and feature-based retrieval. From experiment, the designed and implemented system shows increase of recall rate and precision rate for video data scene retrieval in performance assessment.

• The Transactions of the Korea Information Processing Society
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• v.3 no.3
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• pp.608-616
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• 1996

In this paper, we propose a multimedia hardware architecture for desktop video-conferencing in view of the multimedia data flow and analyze system operation using queueing model. We analyze the bottleneck of multimedia data flow for video- conferencing by simulation, varying the video size, frame rate, the number of participants, and video data compression rate, And we also implement and test the archiecture, that almost includes the analyzed requirements for video-conferencing, to confirm the simulation results. This paper describes the considerations in designing a multimedia I/O system.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.5
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• pp.1132-1140
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• 1996

In video coding systems, an effective rate control method is one of the most improtant issues for the good video quality. This paper presents an adaptive rate control scheme based on the buffer fullness, quantization, and buffer utilization for very low bit rate communication lines, such as 16kbit/s, 24bit/s, and so on. The strategy is implemented on H.263, whichis a vide coding algorithm for narrow band telecommunication channels up to 64kbit/s recommended by ITU-T SG15, to show the effectiveness. The simulation result shows that the suggested rate control scheme has better SNR performance and buffer utilization of source coder than those of linear and non-linear[9] buffer control strategies.

• Journal of the Institute of Convergence Signal Processing
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• v.2 no.3
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• pp.9-15
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• 2001

Very low bit rate video coding uses the inter-picture video coding method for high compression. The inter-picture video coding is coded based on the information of the previous frames so any packet loss can lead to reduce the image quality on the transmission. In this paper, we proposed the multi-stream generation method for inter-media synchronization of very low bit rate video based on TCP for reliable transmission. The proposed approach performs a reliable transmission via a TCP based protocol. This method incorporates multi-streams in order to enhance the robustness of delivery and can withstand against network jitter. Moreover, the client bandwidths are fully utilized in a highly efficient way.

• Journal of Korea Society of Industrial Information Systems
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• v.4 no.4
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• pp.32-39
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• 1999

Video-On-Demand servers have to provide timely processing guarantees and reduce the storage and reduce the storage and bandwidth requirements for continuous media However, compression techniques used in Video-On-Demand servers make the bit rates of compressed video data significantly variable from frame to frame Consequently, most pervious Video-On-Demand servers which use constant bit rate retrieval to guarantee deterministic service under-utilize the system resources and restrict the number of clients. In this paper, I propose a buffer management scheme called CAP(Caching And Prefetching) for Video-On-Demand server using variable bit rate continuous media. By caching and prefetching the data CAP reduces the variation of the compressed data and increases the number of clients simultaneously served and maximizes the utilization of system resources. Results of trace-driven simulations show the effectiveness of the scheme.