• Journal of KIISE:Computer Systems and Theory
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• v.33 no.1_2
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• pp.137-146
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• 2006

Traditional live video streaming systems support the limited field of view (FOV) of image to the remote users. A server system based on the pan/tilt camera provides a user with wide view by changing the view direction of the camera mechanically. But, when many clients try to access to the server, this system can not offer their own view to every user simultaneously, and moreover it has the delay by camera motion. In order to offer wide views to several users, we propose new streaming system using the panorama image that has wide view. Our system is a kind of implementation of software pan/tilt camera. The server acquires panorama video and sends a part of the video to clients. Then, each client can control their own view. We need the effective way to reduce the average transmission data size and server burden to the compression because generally the full size of panorama video is too big to be served by the real-time streaming. To solve this problem, we propose an strip-based video compression and adaptive transmission of the compressed multiple strip videos. Experimental results show that our system can be adapted quickly to the change of view and the number of clients. Furthermore, proposed method effectively reduce the transmission data.

• 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.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.4 no.5
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• pp.691-708
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• 2010

An important issue of supporting multi-users with diverse quality-of-service (QoS) requirements over wireless networks is how to optimize the systematic scheduling by intelligently utilizing the available network resource while, at the same time, to meet each communication service QoS requirement. In this work, we study the problem of a variety of communication services over multi-media heterogeneous cognitive OFDM system. We first divide the communication services into two parts. Multimedia applications such as broadband voice transmission and real-time video streaming are very delay-sensitive (DS) and need guaranteed throughput. On the other side, services like file transmission and email service are relatively delay tolerant (DT) so varying-rate transmission is acceptable. Then, we formulate the scheduling as a convex optimization problem, and propose low complexity distributed solutions by jointly considering channel assignment, bit allocation, and power allocation. Unlike prior works that do not care computational complexity. Furthermore, we propose the FAASA (Fairness Assured Adaptive Sub-carrier Allocation) algorithm for both DS and DT users, which is a dynamic sub-carrier allocation algorithm in order to maximize throughput while taking into account fairness. We provide extensive simulation results which demonstrate the effectiveness of our proposed schemes.

• Journal of Advanced Navigation Technology
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• v.13 no.4
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• pp.523-531
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• 2009

In this study, we propose a joint source/channel rate control algorithm for video encoder targeting packet erasure channel. Based on the buffer constraints of video communication systems, encoding rate constraint is presented. After defining source distortion models for coded and skipped video frames and a channel distortion model for packet errors and their propagation, an average distortion model of received video is proposed for a given encoding window. Finally, we define an optimization problem to minimize the average distortion for given channel rates and packet loss rates by controlling spatio-temporal parameters of source video and FEC block sizes. Then, we propose a window-based algorithm to solve the problem in real-time.

• Journal of Korea Multimedia Society
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• v.15 no.12
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• pp.1464-1474
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• 2012

In this paper, we propose a QoS (Quality of Service)-aware and cooperative resource reservation scheme using cross-layer link adaptation for wireless high definition video transmission through UWB (Ultra Wide Band) network with D-MAC (Distributed Medium Access Control). A wireless high definition video transmission system usually requires stable high throughput even without line-of-sight, e.g., a destination device in another room separated by a wall. Since the WiMedia D-MAC supporting DRP (Distributed Reservation Protocol) scheme causes lots of DRP resource reservation conflicts due to failure of beacon detection in wireless channel environment, overall performances of the WiMedia D-MAC can be deteriorated. And the current WiMedia MAC standard has not considered QoS provisioning even though QoS parameters such as a range of service rates are provided to each traffic stream. Therefore, we propose Relay DRP protocol with QoS-based relay node selection criterion, which makes a relay path to avoid DRP resource reservation conflicts and guarantee QoS more stably through cross-layer link adaptation of cooperative relay transmission scheme and is compliant with the current WiMedia D-MAC protocol. Simulation results demonstrate performance improvements of the proposed method for throughput and QoS provisioning.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.1B
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• pp.26-38
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• 2006

To construct seamless collaborative environments, what all participants intent should be delivered, and visual elements such gesture, facial expression, and ambiance should be shared with all participants. In this paper, we propose high-quality video service to support DV(digital video) and HDV(high-definition DV) based on Access Grid(AG) which is a prevalent collaborative system. The proposed service is designed for employing versatile media tools and codecs with SDP(session description protocol) and SAP(session announcement protocol). We also design network-adaptive video transmission module to mitigate the impact of network fluctuation. This periodically monitors multicast performance and controls frame rate on sender side considering network condition. The experimental results over the test bed show that proposed service enhances quality of AG video service and provides seamless high-quality video transport by mitigating the impact of network fluctuation.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.4
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• pp.105-112
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• 2011

We propose an adaptive load balancing scheme to transport real-time video streams efficiently in this paper. The playback buffer level of a video requesting client is high or low temporarily in mobile environment. This scheme attempts to allocate more network bandwidth to serve a video request with the lower buffer level preferentially. In this scheme, the amount of network bandwidth is dynamically allocated to the requesting clients according to their playback buffer levels in a distributed mobile system. In order to improve the quality of service and real-time performance of individual video playback, the proposed load balancing scheme tries to maximize the number of frames that are transported successfully to the client prior to their playback times. Fair services can also be provided to all the concurrent clients by making their playback situation more adaptive. The performance of this load balancing scheme is compared with that of other static load balancing scheme through extensive simulation experiments, resulting in the higher ratio of frames transmitted successfully within given deadlines.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.8
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• pp.4092-4104
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• 2017

In this paper, an efficient 4K and 8K UHD(Ultra High Definition) transmission scheme is proposed on the convergence networks with broadcasting and LTE(Long Term Evolution) by using CoMP(Coordinated Multi-Point). A video data is compressed and divided into BL(Base Layer), E(Enhanced layer)1, E2 and E3 by scalable HEVC(High Efficiency Video Coding). The divided layers can be combined by the scalable HEVC such as mobile HD, full HD, 4K and 8K UHD(Ultra High Definition). The divided layers are transmitted through the convergence networks with DVB-T2(Digital Video Broadcasting-$2^{nd}$ Generation Terrestrial) broadcasting system and LTE CoMP. This scheme transmits mobile HD and full HD layers through DVB-T2 broadcasting system by using M-PLP(Multiple-physical Layer Pipes), and adaptively transmits 4K or 8K UHD layer through LTE CoMP with MMT(MPEG Media Transport) server. An adaptive transmitting and receiving scheme in the LTE CoMP system provides 4K or 8K UHD layer to a user according to the user status. The proposed scheme is verified by showing the system-level simulation results which is better BER(bit-error-rate) performance than the conventional scheme. The results show that the proposed scheme provides the stable video contents to the user especially at the cell edge.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.7B
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• pp.583-591
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• 2008

This paper proposes a resource allocation algorithm for minimizing the overall distortion of multiple users in orthogonal frequency division multiplex (OFDM). The proposed algorithm exploits the diversity of multiuser and the rate-distortion function using packet distortion model in a system with limited resources. We first induce a rate-distortion function considering error concealment and error propagation properties of H.264 video structures. Then we perform adaptive resource allocation utilizing multiuser diversity for minimizing the overall video quality degradation. We also consider the proportional rate which is pre-determined for each user. Simulation results show that compared to the previous time division multiple access method and the resource allocation method maximizing data rate, the proposed rate allocation algorithm substantially improves the received video quality.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.2A
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• pp.148-158
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• 2008

This paper proposes an efficient signal to noise ratio (SNR) estimation algorithm and its hardware implementation for adaptive transmission system using M-ary modulation scheme. In this paper, we present the implementation results of the proposed algorithm for the second generation digital video broadcasting via satellite (DVB-S2) system, and the proposed algorithm can be tailored to the other communication systems using adaptive transmissions. We built a look-up table (LUT) using the theoretical background of the received signal distribution, and by using this LUT we need just two comparators and a counter for the hardware implementation. For this reason, the hardware of the proposed scheme produces accurate estimation results even with extremely low complexity. The simulation results investigated in this paper reveal that the proposed method can produce estimation results within the specified SNR range in the DVB-S2 system, and it requires a few hundreds of samples for average estimation error of about 1 dB.