• The Journal of the Korea Contents Association
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• v.8 no.6
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• pp.34-42
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• 2008

MAC(Medium Access Control) is demanded to provide end-to-end QoS(Quality of Service) for a variety of traffic in the wireless networks. When all the traffic is integrated in the channel, the main difficulty of the MAC protocol is how to efficiently support multi-class traffic in the limited bandwidth wireless channel. In this paper, we proposed the dynamic bandwidth slot method for improving QoS of the real time traffics. In this paper, we used in-band scheme to send dynamic parameter and considering buffer size and delay variation, we enabled 2 state bits to send to base station in mobile station. The proposed algorithm is to guarantee QoS of real time traffic and maximize transfer efficiency in wireless networks.

• Journal of Korea Multimedia Society
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• v.8 no.9
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• pp.1216-1226
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• 2005

In this paper, a preventive transmission rate decision algorithm, called PTRD, is proposed for the transmission of live video over networks with dynamic bandwidth allocation capability. Frame analyzer predicts the bit-rates of future frames before encoding by analyzing the source information such as spatial variances and the degree of scene changes. By using the predicted bit-rates, transmission rate bounds are derived from the constraints of encoder and decoder buffers. To resolve the problem of renegotiation cost increment due to frequent renegotiations, the PTRD algorithm is presented to decide transmission rates considering the elapsed time after the recent renegotiation and the perceived video quality. From the simulation results, compared to the normalized LMS based method, PTRD is shown to achieve high channel utilization with low renegotiation cost and no delay violation.

• The Journal of Korean Association of Computer Education
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• v.12 no.2
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• pp.77-86
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• 2009

Large-scale applications, that needs large-capacity R&D resources and realtime data transmission, have demanded more stable and high-performance network environment than current Internet environments. Recently, global R&D networks have focuses on utilizing Lambda networking technologies and resource reservation systems to be satisfied with various applications' requirements. In this paper, we modify the existing DRS (Dynamic Right-Sizing) model to reflect various advantages in terms of the stability and high-capacity of Lambda network. In addition, we suggest the design methodology of high-performance Lambda network, which can integrate NRPS (Network Resource Provisioning System) into our modified DRS model.

• Journal of the Institute of Convergence Signal Processing
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• v.3 no.1
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• pp.21-26
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• 2002

Video stream invoke a variety of traffic with the structure of compression algorithm and image complexity. Thus, it is difficult to allocate the resource on the both sides of sender and receiver, and playout on the Internet such as a packet switched network. Thus, in this paper we proposed video stream smoothing using multistream for the effective transmission of video stream. This method specifies the type of LDU(logical data unit) according to the type of original stream, and then makes a large number of streams as a fixed size, and transfers them. So, the proposed method can reduce the buffering time which occurs during the process of the smoothing and prefetch be robust to the jitter on network, as well. Consequently, it has the effective transmission characteristics of fully utilizing the clients bandwidth.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.3
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• pp.68-74
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• 2009

In this paper, we propose an effective memory reduction algorithm to reduce the amount of reference frame buffer and memory bandwidth in video encoder and decoder. In general video codecs, decoded previous frames should be stored and referred to reduce temporal redundancy. Recently, reference frames are recompressed for memory efficiency and bandwidth reduction between a main processor and external memory. However, these algorithms could hurt coding efficiency. Several algorithms have been proposed to reduce the amount of reference memory with minimum quality degradation. They still suffer from quality degradation with fixed-bit allocation. In this paper, we propose an adaptive block-based min-max quantization that considers local characteristics of image. In the proposed algorithm, basic process unit is $8{\times}8$ for memory alignment and apply an adaptive quantization to each $4{\times}4$ block for minimizing quality degradation. We found that the proposed algorithm can obtain around 1.7% BD-bitrate gain and 0.03dB BD-PSNR gain, compared with the conventional fixed-bit min-max algorithm with 37.5% memory saving.

• Journal of KIISE:Computing Practices and Letters
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• v.14 no.7
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• pp.625-635
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• 2008

Portable multimedia players have some different properties compared to general multimedia file server. Some of those properties are single user ownership, relatively low hardware performance, I/O burst by user special functions, and short software development cycles. Though suitable for processing multiple user requests at a time, the general multimedia file systems are not efficient for special user functions such as fast forwards/backwards. Soml' methods has been proposed to improve the performance and functionality, which the application programs give prediction hints to the file system. Unfortunately, they require the modification of all applications and recompilation. In this paper, we present a file system that efficiently supports user special functions in embedded multimedia systems using file block allocation, buffer-cache, and prefetch. A prefetch algorithm, SPRA (SPeed-based PRefetch Algorithm) predicts the next block using I/O patterns instead of hints from applications and it is resident in the file system, so doesn't affect application development process. From the experimental file system implementation and comparison with Linux readahead-based algorithms, the proposed system shows $4.29%{\sim}52.63%$ turnaround time and 1.01 to 3,09 times throughput in average.

• Journal of KIISE:Information Networking
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• v.36 no.6
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• pp.492-504
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• 2009

The delivery of multimedia that efficiently adapts its bit-rate to changing network characteristics and conditions is one of the important challenging tasks in the design of today's real-time multimedia streaming systems such as IPTV, Mobile IPTV and so on. In these work, the primary focus is on network congestion, to improve network stability and inter-protocol fairness. However, these existing works have problems which do not support QoE (Quality of Experience), because they did not consider essential characteristics of contents playback such as the media continuity. In this paper, we propose a novel rate control scheme for improving the QoE of multimedia streaming service in the Internet congestion, called NCAR (Network and Client-Aware Rate control), which is based on network-aware congestion control and client-aware flow control scheme. Network-aware congestion control of the NCAR offers an improving reliability and fairness of multimedia streaming, and reduces the rate oscillation as well as keeping high link utilization. Client-aware flow control of NCAR offers a removing the media discontinuity and a suitable receiver buffer allocation, and provides a good combination of low playback delay. Simulation results demonstrate the effectiveness of our proposed schemes.

• Journal of the Institute of Electronics and Information Engineers
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• v.49 no.12
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• pp.201-208
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• 2012

In this paper, the hardware design of rate control for real-time video encoded is proposed. In the proposed method, a quadratic rate distortion model with high-computational complexity is not used when quantization parameter values are being decided. Instead, for low-computational complexity, average complexity weight values of frames are used to calculate QP. For high speed and low computational prediction, the MAD is predicted based on the coded basic unit, using spacial and temporal correlation in sequences. The rate control is designed with the hardware for fast QP decision. In the proposed method, a quadratic rate distortion model with high-computational complexity is not used when quantization parameter values are being decided. Instead, for low-computational complexity, average complexity weight values of frames are used to calculate QP. In addition, the rate control is designed with the hardware for fast QP decision. The execution cycle and gate count of the proposed architecture were reduced about 65% and 85% respectively compared with those of previous architecture. The proposed RC was implemented using Verilog HDL and synthesized with UMC $0.18{\mu}m$ standard cell library. The synthesis result shows that the gate count of the architecture is about 19.1k with 108MHz clock frequency.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.38 no.2
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• pp.218-226
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• 2001

The MPEG-2 Test Model 5 (TM5) algorithm is widely used for bit rate control. In TM5, however, the target number of bits and the number of actual coding bits for each picture do not match well. Therefore, buffer overflow and picture quality degradation may occur at the end of the GOP. In this paper, we propose a new bit rate control algorithm for matching the target and the actual coding bits based on accurate bit allocation. The key idea of the proposed algorithm is to determine quantization Parameters which enable us to generate the number of actual coding bits close to the target number of bits for each picture, while maintaining uniform picture quality and supporting real-time processing. The proposed algorithm exploits the relationship between the number of actual coding bits and the number of estimated bits of the previous macroblock.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.8
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• pp.1505-1512
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• 2009

This paper propose a new frame level rate control algorithm for improving video quality and decreasing quality variation of an entire video sequence in a very low bit rate environment. In the proposed scheme, the allocated bits to a GOP are distributed to each frame properly according to the frame characteristics as well as the buffer status and the channel bandwidth. The H.264 standard uses various coding modes and optimization methods to improve the compression performance, which makes it difficult to control the generated traffic accurately. In this paper, proper prediction models for low bit rate environments are lust proposed, and a target distortion is determined using the models. According to the target distortion, the bit budget is allocated to each frame. It is shown by experimental results that the new algorithm can generate the PSNR performance better than that of the existing rate control algorithm.