• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.25-28
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• 2002

This paper presents variable bit rate(VBR) rate control scheme based on MPEG-4 VM8 rate control scheme. An initial Q searching method provides more accurate bit allocation for the first frame. A frame skipping and RD Model update scheme when coded frame quality is too low or high prevents image quality fluctuation.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.5
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• pp.868-874
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• 2007

A real-time frame-layer rate control algorithm with a token bucket traffic shaper is proposed for minimizing video duality variation. The proposed rate control method uses a non-iterative optimization method for low computational complexity, and performs bit allocation at the frame level to minimize variation in distortion between frames. In order to reduce the quality fluctuation, we use a sliding window scheme which does not require the pre-analysis process. Therefore, the proposed algorithm does not produce the delay from encoding, and is suitable for real-time low-complexity video encoder. Experimental results indicate that the proposed control method provides better PSNR performance than the existing rate control method.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.413-417
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• 2009

The use of scalable video coding scheme has been regarded as a promising solution for guaranteeing the quality of service of the video streaming over the Internet because it is a capable coding scheme to perform quality adaptation depending on network conditions. In this paper, we use a streaming model that transmits base layer using TCP and enhancement layers using DCCP, which try to provide transmission reliability of the BL and TCP friendliness. Unlike pervious works, the proposed algorithm performs rate adaptation based on playout buffer status. The PoB status of the client is sent back periodically to the server and serves as a network congestion indicator. Experimental results show that our scheme improves streaming quality comparing with pervious scheme in the case of not only constant/dynamic background flows but also VBR-encoded video sequence.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.7A
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• pp.711-723
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• 2002

A finer granular scalable (FGS) version of ISO/IEC MPEG-4 video streaming is investigated in this work with the prioritized stream delivery over loss-rate differentiated networks. Our proposed system is focused on the seamless integration of rate adaptation, prioritized packetization, and simplified differentiation for the MPEG-4 FGS video streaming. The proposed system consists of three key components: 1) rate adaptation with scalable source encoding, 2) content-aware prioritized packetization, and 3) loss-based differential forwarding. More specifically, a constant-quality rate adaptation is first achieved by optimally truncating the over-coded FGS stream based on the embedding rate-distortion (R-D) information (obtained from a piecewise linear R-D model). The rate-controlled video stream is then packetized and prioritized according to the loss impact of each packet. Prioritized packets are transmitted over the underlying network, where packets are subject to differentiated dropping and forwarding. By focusing on the end-to-end quality, we establish an effective working conditions for the proposed video streaming and the superior performance is verified by simulated MPEG-4 FGS video streaming.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.11
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• pp.4068-4086
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• 2014

Due to the prevalence of powerful mobile terminals and the rapid advancements in wireless communication technologies, the wireless video streaming service has become increasingly more popular. Recent studies show that video streaming services via Transmission Control Protocol (TCP) are becoming more practical. TCP has more advantages than User Diagram Protocol (UDP), including firewall traversal, bandwidth fairness, and reliability. However, each video service shares an equal portion of the limited bandwidth because of the fair sharing characteristics inherent in TCP and this bandwidth fair sharing cannot always guarantee the video quality for each user. To solve this challenging problem, an Adaptive Multiple TCP (AM-TCP) scheme is proposed in this paper to guarantee the video quality for mobile devices in wireless networks. AM-TCP adaptively controls the number of TCP connections according to the video Rate Distortion (RD) characteristics of each stream and network status. The proposed scheme can minimize the total distortion of all participating video streams and maximize the service quality by guaranteeing the quality of each video streaming session. The simulation results show that the proposed scheme can significantly improve the quality of video streaming in wireless networks.

• The KIPS Transactions:PartA
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• v.12A no.3 s.93
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• pp.197-204
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• 2005

In this paper, we consider the transmission or variable bit rate (VBR) stored video for the distributed video streaming service over Internet. In streaming service, users often suffer from the discontinuity in playback due to the decrease in bandwidth during transmission according to bandwidth renegotiation protocol. We propose a novel transmission technique to overcome this problem for stored variable bit rate video. This scheme uses a priori information of stored VBR video to continue streaming without playback discontinuity. In addition, an approximation scheme for the buffer-bandwidth relation is proposed in order to facilitate the admission control under the proposed scheme.

• Journal of information and communication convergence engineering
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• v.14 no.4
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• pp.240-245
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• 2016

Scalable video streaming over wireless networks has many challenges. The most significant challenge is related to packet loss. To overcome this problem, in this paper, we propose an unequal loss protection (ULP) method using a new forward error correction (FEC) mechanism for robust scalable video streaming over wireless networks. For an efficient FEC assignment considering video quality, we first introduce a simple and efficient performance metric, the layer-based recovery rate (LRR), for quantifying the unequal error propagation effects of the temporal and quality layers on the basis of packet losses. LRR is based on the unequal importance in both the temporal and the quality layers of a hierarchical scalable video coding structure. Then, the proposed ULP-LRR method assigns an appropriate number of FEC packets on the basis of the LRR to protect the video layers against packet lossy network environments. Compared with conventional ULP algorithms, the proposed ULP-LRR algorithm demonstrates a higher performance for various error-prone wireless channel statuses.

• The KIPS Transactions:PartC
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• v.10C no.4
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• pp.479-484
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• 2003

In the current Internet, it is difficult for an Internet Phone to guarantee the QoS due to variable network conditions such as packet loss rate, delay and bandwidth. In addition, the QoS of an Internet Video Phone is more hard to guarantee because of video data. In this paper, we investigate application-level QoS control schemes that can adapt to variable network conditions, and describe an error control scheme and a congestion control scheme. Based on these QoS control schemes, we have designed and implemented an Internet Video Phone System that supports adaptive audio and video delivery. Through experiments, we found that the Internet Video Phone can reduce the packet loss rate considerably as well as adjust the transmission rate considering other TCP flows.

• Journal of Korea Multimedia Society
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• v.17 no.9
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• pp.1084-1091
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• 2014

When demand for transmitting multimedia data increases, network congestion is more likely to occur and users will suffer high loss rate as well as high delay. In order to enhance quality-of-service (QoS) of video multicasting, we need to raise transmission reliability and reduce end-to-end delay. This paper proposes a routing mechanism for a OpenFlow-based small-scale network in order to multicast video reliably with low delay. In our method, multipath routing will be applied to Multiple Description (MD) Coded video to exploit its multi-description property. Through performance evaluation, our method shows improvement on loss rate, delay and video distortion.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.6
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• pp.1147-1165
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• 2011

We present a simple and cost effective rate control scheme for streaming video over a wireless channel by using the information of mobile devices' buffer level. To prevent buffer fullness and emptiness at receivers, the server should be able to adjust sending rate according to receivers' buffer status. We propose methods to adjust sending rate based on the buffer level and discrete derivative of the buffer occupancy. To be compatible with existing network protocols, we provide methods to adjust sending rate by changing the inter-packet delay (IPD) at the server side. At every round-trip time, adjustments of sending rate are made in order to achieve responsiveness to sudden changes of buffer availabilities. A series of simulations and the prototype system showed that the proposed methods did not cause buffer overflows and it can maintain smoother rate control and react to bandwidth changes promptly.