Dynamic region of interest transcoding for multipoint video conferencing

This paper presents a region of interest transcoding scheme for multipoint video conferencing to enhance the visual quality. In a multipoint videoconference, usually there are only one or two active conferees at one time which are the regions of interest to the other conferees involved. We propose a Dynamic Sub-Window Skipping (DSWS) scheme to firstly identify the active participants from the multiple incoming encoded video streams by calculating the motion activity of each sub-window, and secondly reduce the frame-rates of the motion inactive participants by skipping these less-important subwindows. The bits saved by the skipping operation are reallocated to the active sub-windows to enhance the regions of interest. We also propose a low-complexity scheme to compose and trace the unavailable motion vectors with a good accuracy in the dropped inactive sub-windows after performing the DSWS. Simulation results show that the proposed methods not only significantly improve the visual quality on the active subwindows without introducing serious visual quality degradation in the inactive ones, but also reduce the computational complexity and avoid whole-frame skipping. Moreover, the proposed algorithm is fully compatible with the H.263 video coding standard. 1

Low-complexity and high-quality frame-skipping transcoder for continuous presence multipoint video conferencing

2003-2004 > Academic research: refereed > Publication in refereed journalVersion of RecordPublishe

dOTM: a mechanism for distributing centralized multi-party video conferencing in the cloud

One of the key factors for a given application to take advantage of cloud computing is the ability to scale in an efficient, fast and reliable way. In centralized multi-party video conferencing, dynamically scaling a running conversation is a complex problem. In this paper we propose a methodology to divide the Multipoint Control Unit (the video conferencing server) into more simple units, broadcasters. Each broadcaster receives the media from a participant, processes it and forwards it to the rest. These broadcasters can be distributed among a group of CPUs. By using this methodology, video conferencing systems can scale in a more granular way, improving the deployment

Random Linear Network Coding for 5G Mobile Video Delivery

An exponential increase in mobile video delivery will continue with the demand for higher resolution, multi-view and large-scale multicast video services. Novel fifth generation (5G) 3GPP New Radio (NR) standard will bring a number of new opportunities for optimizing video delivery across both 5G core and radio access networks. One of the promising approaches for video quality adaptation, throughput enhancement and erasure protection is the use of packet-level random linear network coding (RLNC). In this review paper, we discuss the integration of RLNC into the 5G NR standard, building upon the ideas and opportunities identified in 4G LTE. We explicitly identify and discuss in detail novel 5G NR features that provide support for RLNC-based video delivery in 5G, thus pointing out to the promising avenues for future research.Comment: Invited paper for Special Issue "Network and Rateless Coding for Video Streaming" - MDPI Informatio

DCT-based video frame-skipping transcoder

2002-2003 > Academic research: refereed > Refereed conference paperVersion of RecordPublishe

SatCom Today in Canada: Significant Research: Broadband Satellite Communications List of CITR related Publications (1998-2003)

Journal Papers Conference Papers Contributions to Standards Canadian Space Agency Recent Publication

Video quality estimation of {DCCP} streaming over wireless networks

International audienceThis paper describes a streaming architecture simulation model above Network Simulator 2 (NS2) which allows to define specific transport properties. Multimedia contents are specific because they are time-dependent and they can undergo small deterioration if necessary. We simulate such a congestion control that has the ability to decrease the multimedia quality in case of network congestion in order to decrease packet losses and packet delivery delays. We integrate this video congestion control inside DCCP (Datagram Congestion Control Protocol) and TFRC (TCP Friendly Rate Control). The transcoding of the multimedia contents is realized thanks to the NetMoVie simulation model which is an RTP mixer. We compare the adaptive transport solution to the classic transport solution without any adaptive mechanism. The Peak Signal-to-Noise Ratio (PSNR) of the received multimedia contents is measured and compared for better visualization

Bit Rate Control for Real-time Multipoint Video Conferencing

With the rapid development of video compression and network technology, real-time video communications has become a popular part of our daily life. Rate control is needed to satisfy the expectation of high quality and to make it possible to transmit over limited bandwidth. The objective of this thesis is to design a rate control scheme for a real-time Transcoding-Compositing Multipoint Video Conferencing System, which operates exclusively in the DCT domain. In this Transcoding-Compositing system, the mode of the composited frame should firstly be decided before encoding the composited image. A mode decision method relying on Karhunen-Loeve scene change detection is proposed. A new linear source Rate-Distortion model is developed in the - domain ( is the percentage of zero), based on which rate control scheme is designed. The designed rate control scheme is parted into three levels: Frame Level, Sub-frame Level, and Macroblock Level. Frame Level rate control decides the bit budget for each frame based on the buffer fullness. Sub-frame Level rate control optimizes the distribution of the bit budget among the decimated sub-images. Based on the linear source model, Macroblock Level rate control carries out an adaptive procedure to precisely control the number of encoding bits for each sub-image