• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.3
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• pp.506-515
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• 2015

In this paper, we propose a group playback synchronization method using adaptive synchronization delay time by the bit rate of media to synchronize a play position of streaming media between mobile smart devices. This method consists of streaming server-side and client-side synchronization algorithms based on synchronization delay time which includes connection time, control packet transmission time, streaming data buffering time, and synchronization processing time. We implement the Android media player application with synchronization support using the proposed algorithms and present the result of performance evaluation.

• Journal of KIISE
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• v.43 no.7
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• pp.827-833
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• 2016

HTTP Adaptive Streaming (HAS) is a technique that adapts its video quality to network conditions for providing Quality of Experience. In the HAS approach, a video content is encoded at multiple bitrates and the encoded video content is divided into several video segments. A HAS player estimates the network bandwidth and adjusts the video bitrate based on estimated bandwidth. However, the segment scheduler in the conventional HAS player requests video segments periodically without considering TCP. If the waiting duration for the next segment request is quite long, the TCP connection can be initialized and it restarts slow-start. Slow-start causes the reduction in TCP throughput and consequentially leads to low-quality video streaming. In this study, we propose a TCP-aware segment scheduling scheme to improve performance of HAS service. The proposed scheme adjusts request time for the next video request to prevent initialization of TCP connection and also considers the point of scheduling time. The simulation proves that our scheme improves the Quality of Service of the HAS service without buffer underflow issue.

• Journal of KIISE:Information Networking
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• v.37 no.5
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• pp.363-373
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• 2010

The existing QoS mechanisms for video streaming are short of the consideration for various user environments and the characteristic of streaming applying programs. In order to overwhelm this problem, studies on the video streaming protocols exploiting scalable video coding (SVC), which provide spatial, temporal, and qualitative scalability in video coding, are progressing actively. However, these protocols also have the problem to deepen network congestion situation, and to lower fairness between other traffics, as they are not equipped with congestion control mechanisms. SVC based streaming protocols also have the problem to overlook the property of videos encoded in SVC, as the protocols transmit the streaming simply by extracting the bitstream which has the maximum bit rate within available bandwidth of a network. To solve these problems, this study suggests TCP-friendly network adaptive SVC streaming(T-NASS) protocol which considers both network status and SVC bitstream property. T-NASS protocol extracts the optimal SVC bitstream by calculating TCP-friendly transmission rate, and by perceiving the network status on the basis of packet loss rate and explicit congestion notification(ECN). Through the performance estimation using an ns-2 network simulator, this study identified T-NASS protocol extracts the optimal bitstream as it uses TCP-friendly transmission property and perceives the network status, and also identified the video image quality transmitted through T-NASS protocol is improved.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.668-672
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• 2009

In wireless ad-hoc network, knowing the available bandwidth of the time varying channel is imperative for live video streaming applications. This is because the available bandwidth is varying all the time and strictly limited against the large data size of video streaming. Additionally, adapting the encoding rate to the suitable bit-rate for the network, where an overlarge encoding rate induces congestion loss and playback delay, decreases the loss and delay. While some effective rate controlling methods have been proposed and simulated well like VTP (Video Transport Protocol) [1], implementing to cooperate with the encoder and tuning the parameters are still challenging works. In this paper, we show our result of the implementation experiment of VTP based encoding rate controlling method and then introduce some techniques of our parameter tuning for a video streaming application over wireless environment.

• Journal of Broadcast Engineering
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• v.22 no.5
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• pp.600-607
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• 2017

Virtual Reality (VR) has lately gained significant attention primarily driven by the recent market availability of consumer devices, such as mobile phone-based Head Mounted Displays (HMDs). Apart from classic gaming use cases, the delivery of $360^{\circ}$ video is considered as another major application and is expected to be ubiquitous in the near future. However, the delivery and decoding of high-resolution $360^{\circ}$ videos in desirable quality is a challenging task due to network limitations and constraints on available end device decoding and processing. In this paper, we focus on aspects of $360^{\circ}$ video streaming and provide an overview and discussion of possible solutions as well as considerations for future VR video streaming applications. This paper mainly focuses on the status of the standardization activities, Omnidirectional MediA Format (OMAF), to support interoperable $360^{\circ}$ video streaming services. More concretely, MPEG's ongoing work for OMA aims at harmonization of VR video platforms and applications. The paper also discusses the integration in MPEG Dynamic Adaptive Streaming over HTTP (MPEG-DASH), which is considered as $360^{\circ}$ video streaming services with OMAF content. In context of the general OMAF service architecture.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.23 no.7
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• pp.837-844
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• 2019

As the demand for video streaming has been rapidly increasing recently, video streaming schemes for increasing the efficiency of radio resource has attracted a lot of attention. In this paper, we propose an adaptive video streaming scheme to enhance the efficiency of video streaming by using receivers' caching capability. The proposed streaming scheme can transmit video data on a broadcast basis even when two clients request different video data, only if specific conditions satisfied, while existing schemes can only transmit video data on a broadcast basis only when two clients request the same video data. In this paper, we mathematically derive the average transmission time of the proposed scheme and the approximation of the average transmission time. The accuracy of the mathematical analysis is verified by simulations. Mathematical analysis and simulation results show that the proposed scheme can significantly reduce the average transmission time, compared to the existing scheme.

• Journal of Korea Multimedia Society
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• v.21 no.8
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• pp.942-951
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• 2018

In these days, 360 degree videos, which is provided via VR, have high resolution and quality of 8K. These kinds of videos inevitably require streaming technology which guarantees QoS. Therefore, we suggest MPEG DASH HTTP protocol which stably provides streaming services of high quality videos in 8K, which are 360 degree tile-encoded, on low-spec devices such as OTT and IPTV Settop and Tile Segment Cache management structure for servers operating streaming services.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.1
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• pp.414-435
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• 2017

As network adaptive streaming technology becomes increasingly common, transport protocol also becomes important in guaranteeing the quality of multimedia streaming. At the same time, because of the appearance of high-quality video such as Ultra High Definition (UHD), preventing buffering as well as preserving high quality while deploying a streaming service becomes important. The Internet Engineering Task Force recently published Multipath TCP (MPTCP). MPTCP improves the maximum transmission rate by simultaneously transmitting data over different paths with multiple TCP subflows. However, MPTCP cannot preserve high quality, because the MPTCP subflows slowly increase the transmission rate, and upon detecting a packet loss, drastically halve the transmission rate. In this paper, we propose a new multipath congestion control scheme for high-quality multimedia streaming. The proposed scheme preserves high quality of video by adaptively adjusting the increasing parameter of subflows according to the network status. The proposed scheme also increases network efficiency by providing load balancing and stability, and by supporting fairness with single-flow congestion control schemes.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.8
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• pp.3856-3872
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• 2018

HTTP-based adaptive streaming (HAS) has recently been widely deployed on the Internet. In the HAS system, a video content is encoded at multiple bitrates and the encoded video content is segmented into small parts of fixed durations. The HAS client requests a video segment and stores it in the playout buffer. The rate adaptation algorithm employed in HAS clients dynamically determines the video bitrate depending on the time-varying bandwidth. Many studies have shown that an efficient rate adaptation algorithm is critical to ensuring quality-of-experience in HAS systems. However, existing algorithms have problems estimating the network bandwidth because bandwidth estimation is performed on the client-side application stack. Without the help of transport layer protocols, it is difficult to achieve accurate bandwidth estimation due to the inherent segment-based transmission of the HAS. In this paper, we propose an alternative approach that utilizes the playout buffer occupancy rather than using bandwidth estimates obtained from the application layer. We start with a queueing analysis of the playout buffer. Then, we present a buffer-aware rate adaptation algorithm that is solely based on the mean buffer occupancy. Our simulation results show that compared to conventional algorithms, the proposed algorithm achieves very smooth video quality while delivering a similar average video bitrate.

• IEMEK Journal of Embedded Systems and Applications
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• v.17 no.5
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• pp.297-307
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• 2022

The simple procedural segment selection algorithm commonly used in Dynamic Adaptive Streaming over HTTP (DASH) reveals severe weakness to provide high-quality streaming services in the integrated mobile networks of various wired and wireless links. A major issue could be how to properly cope with dynamically changing underlying network conditions. The key to meet it should be to make the segment selection algorithm much more adaptive to fluctuation of network traffics. This paper presents a system architecture that replaces the existing procedural segment selection algorithm with a deep reinforcement learning algorithm based on the Asynchronous Advantage Actor-Critic (A3C). The distributed A3C-based deep learning server is designed and implemented to allow multiple clients in different network conditions to stream videos simultaneously, collect learning data quickly, and learn asynchronously, resulting in greatly improved learning speed as the number of video clients increases. The performance analysis shows that the proposed algorithm outperforms both the conventional DASH algorithm and the Deep Q-Network algorithm in terms of the user's quality of experience and the speed of deep learning.