• Journal of Digital Contents Society
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• v.16 no.3
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• pp.365-371
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• 2015

In wireless network, the amount of video streaming packet information in receiver replay buffer can be varied according tothe wireless network condition. By the effect, unforeseeable delay and jitter are generated and then busty video traffics can be made. If the amount of buffer information coming in receiver replay buffer is larger than the amount of a specific buffer information, buffer overflow is generated. Such a problem makes the image skip effect and packet loss, and then causes the quality degradation and replay discontinuity of the video streaming service in destination receiver. To solve the buffer overflow problem, this paper applies the token bucket for the busty traffic to the receiver terminal and analyzes the effect of the token bucket. The simulation result using NS-2 and JSVM shows that the proposed scheme with the token bucket has significantly better performance than the conventional scheme without the token bucket in terms of overflow generation number, packet loss rate and PSNR.

• Journal of Korea Multimedia Society
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• v.14 no.1
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• pp.44-53
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• 2011

H.264/SVC provides scalable video streams which consist of a base layer and one or more enhancement layers. Thus, it can efficiently adapt encoded streams to individual network conditions by dropping some layers of bit streams. However, on a dynamic environment such as the Internet, random packet losses due to network congestion can cause drastic effect on SVC quality. To avoid network congestion, the rate of video streams should be adjusted by carefully selecting a layer of each stream. In this paper, we propose three layer selection algorithms which can avoid network congestion by using the rate-distortion characteristics of streams. Simulation results show that FS(Far-Sighted) algorithm can maximize the overall PSNR value of streams by efficiently using the characteristics of video streams.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.118-123
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• 2009

In this paper, we propose packet selection and significance based interval allocation algorithm for real-time streaming service. In real-time streaming of inter-frame (and layer) coded video, minimizing packet loss does not imply maximizing QoS. It is true that packet loss adversely affects the QoS but one single packet can have more impact than several other packets. We exploit the fact that the significance of each packet loss is different from the frame type it belongs to and its position within GoP. Using packet dependency and PSNR degradation value imposed on the video from the corresponding packet loss, we find each packet's significance value. Based on the packet significance, the proposed algorithm determines which packets to send and when to send them. The proposed algorithm is tested using publicly available MPEG-4 video traces. Our scheduling algorithm brings significant improvement on user perceivable QoS. We foresee that the proposed algorithm manifests itself in last mile connection of the network where intervals between successive packets from the source and to the destination are well preserved.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.23 no.1
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• pp.37-43
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• 2023

Compressed still image or video bitstreams require protection from channel errors in a wireless channel. Embedded Zerotree Coding(EZW), SPIHT could have provided unprecedented high performance in image compression with low complexity. If bit error is generated by dint of wireless channel transmission problem, the loss of synchronization on between encoder and decoder causes serious performance degradation. But wavelet zerotree coding algorithms are producing variable-length codewords, extremely sensitive to bit errors. The idea is to partition the lifting coefficients. A many partition of lifting transform coefficients distributes channel error from wireless channel to each partition. Therefore synchronization problem that caused quality deterioration in still image and video stream was improved.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.48 no.3
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• pp.79-89
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• 2011

Recent development of embedded processors makes it possible to play back video contents in real-time on portable devices. Because of their limited battery capacity and low computational performance, however, portable devices still have significant problems in real-time decoding of high quality or high resolution compressed video. Although previous approaches are successful in achieving complexity-scalable decoder by controlling computational complexity of decoding elements, they cause significant objective quality loss coming from mismatch between encoder and decoder. In this paper, we propose a selective B slice skip-decoding method to implement a low complexity video decoder. The proposed method performs selective skip decoding process of B slice which satisfies the proposed conditions. The skipped slices are reconstructed by simple reconstruction method utilizing adjacent reconstructed pictures. Experimental result shows that proposed method not only reduces computational complexity but also maintains subjective visual quality.

• Journal of the Korea Society of Computer and Information
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• v.19 no.3
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• pp.63-71
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• 2014

We analyze the execution time of Reed-Solomon coding, which is the MAC-layer forward error correction scheme used in CDMA2000 1xEV-DO broadcast services, under different air channel conditions. The results show that the time constraints of MPEG-4 cannot be guaranteed by Reed-Solomon decoding when the packet loss rate (PLR) is high, due to its long computation time on current hardware. To alleviate this problem, we propose three error control schemes. Our static scheme bypasses Reed-Solomon decoding at the mobile node to satisfy the MPEG-4 time constraint when the PLR exceeds a given boundary. Second, dynamic scheme corrects errors in a best-effort manner within the time constraint, instead of giving up altogether when the PLR is high; this achieves a further quality improvement. The third, video-aware dynamic scheme fixes errors in a similar way to the dynamic scheme, but in a priority-driven manner which makes the video appear smoother. Extensive simulation results show the effectiveness of our schemes compared to the original FEC scheme.

• Journal of KIISE:Information Networking
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• v.36 no.5
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• pp.425-436
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• 2009

In recent years, the user demands have increased for multimedia service of high quality over the broadband convergence network. These rising demands for high quality multimedia service led the popularization of various user terminals and large scale display equipments, which needs a variety type of QoS (Quality of Service). In order to support demands for QoS, numerous research projects are in progress both from the perspective of network as well as end system; For example, at the network perspective, QoS guaranteeing by improving of internet performance such as Active Queue Management, while at the end system perspective, SVC (Scalable Video Coding) encoding scheme to guarantee media quality. However, existing AQM algorithms have problems which do not guarantee QoS, because they did not consider the essential characteristics of video encoding schemes. In this paper, it is proposed to solve this problem by deploying the TS- AQM (Temporal Scalability Active Queue Management) which employs the differentiated packet dropping for dependency of the temporal level among the frames, based on SVC encoding characteristics by exploiting the TID (Temporal ID) field of the SVC NAL unit header. The proposed TS-AQM guarantees multimedia service quality through video decoding reliability for SVC streaming service, by differentiated packet dropping when congestion exists.

• Journal of Digital Convergence
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• v.12 no.8
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• pp.235-242
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• 2014

Internet telephony is an Internet service which supports voice telephone using VoIP technology on the IP-based Internet. It has some advantages in that voice telephone services can be accompanied with multimedia services such as video communication and messaging services. In this paper we suggested an Asterisk-based Internet telephony system which can be easily scalable. Most current systems use text files to manage their configuration: SIP users, dialplans, IVR service and etc. But we designed the management system which introduces database tables for efficiency and scalability. It also supports web-based functions developed by using Asterisk, Apache, MySQL, jQuery, PHP and open source softwares.

• Journal of KIISE:Computing Practices and Letters
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• v.14 no.8
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• pp.793-797
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• 2008

To achieve flexible visual contents adaptation for multimedia communications, the ISO/IEC MPEG & ITU-T VCEG form the JVT to develop an SVC amendment for the H.264/AVC standard. JVT uses inter-layer prediction that can improve the rate-distortion efficiency of the enhancement layer. But inter-layer prediction causes computational complexity to be increased. In this paper, we propose a fast mode decision for inter frame coding. It makes use of the correlation between optimized prediction mode and its RD cost. Experimental results show that the proposed schemes save up to 38% of encoding time with a negligible coding loss and bit-rate increase.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2016.05a
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• pp.683-684
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• 2016

The up-to-date high-performance DSP or FPGA employs SRIO(Serial Rapid IO) as a high-speed serial communications. SRIO is an industry standard regulated upto Ver 3.1. In this study we developed a backplane having a transmission rate to 15Gbps based on a TI DSP. The back plane icould be used to High-speed video transmission, and will be adopted to connecting multiple DSPs for scalable architecture. This paper will discuss the design constraints for a high-speed communication and multiple-core operation.