國立虎尾科技大學 |

Application Layer System Design for Real-Time Video Communications.

紀錄類型:	書目-語言資料,手稿 : Monograph/item
正題名/作者:	Application Layer System Design for Real-Time Video Communications./
作者:	Kurdoglu, Eymen.
面頁冊數:	1 online resource (119 pages)
附註:	Source: Dissertation Abstracts International, Volume: 78-10(E), Section: B.
標題:	Computer engineering. -
電子資源:	click for full text (PQDT)
ISBN:	9781369851649

Application Layer System Design for Real-Time Video Communications.
Kurdoglu, Eymen.

Application Layer System Design for Real-Time Video Communications. - 1 online resource (119 pages)

Source: Dissertation Abstracts International, Volume: 78-10(E), Section: B.

Thesis (Ph.D.)--Polytechnic Institute of New York University, 2017.

Includes bibliographical references

Given that the world is becoming ever more connected each day, real-time video communications is an avenue of technology with untapped potential that can bring the human beings ever closer. The main objective of this thesis is to improve the state-of-the-art in the field of real-time video communications with respect to the end-users' quality of experience, through application layer system design, which jointly considers the video and the networking components.

Electronic reproduction.
Ann Arbor, Mich. :
ProQuest,
2018

Mode of access: World Wide Web

ISBN: 9781369851649Subjects--Topical Terms:

569006
Computer engineering.
Index Terms--Genre/Form:

554714
Electronic books.

Application Layer System Design for Real-Time Video Communications.
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500 $a Source: Dissertation Abstracts International, Volume: 78-10(E), Section: B.
500 $a Advisers: Yao Wang; Yong Liu.
502 $a Thesis (Ph.D.)--Polytechnic Institute of New York University, 2017.
504 $a Includes bibliographical references
520 $a Given that the world is becoming ever more connected each day, real-time video communications is an avenue of technology with untapped potential that can bring the human beings ever closer. The main objective of this thesis is to improve the state-of-the-art in the field of real-time video communications with respect to the end-users' quality of experience, through application layer system design, which jointly considers the video and the networking components.
520 $a We first focus on designing peer-to-peer multi-party video conferencing systems, where the users with different uplink-downlink capacities send their videos using multicast trees. One way to deal with user bandwidth heterogeneity is employing layered video coding, generating multiple layers with different rates, whereas an alternative is partitioning the receivers of each source and disseminating a different non-layered video version within each group. Here, we aim to maximize the average received video quality for both systems under uplink-downlink capacity constraints, while constraining the number of hops the packets traverse to two. We first show any multicast tree is equivalent to a collection of 1-hop and 2-hop trees, under user uplink-downlink capacity constraints. This reveals that the packet overlay hop count can be limited to two without sacrificing the achievable rate performance. Assuming a fine granularity scalable stream that can be truncated at any rate, we propose an algorithm that solves for the number of video layers, layer rates and distribution trees for the layered system. For the partitioned simulcast system, we develop an algorithm to determine the receiver partitions along with the video rate and the distribution trees for each group. Through numerical comparison, we show that the partitioned simulcast system achieves the same average receiving quality as the ideal layered system without any coding overhead, and better quality than the layered system with a coding overhead of only 10% for the 4-user systems simulated. The two systems perform similarly for the 6-user case if the layered coding overhead is 10%, and partitioned simulcast achieves higher received quality when the layered coding overhead is beyond 10%.
520 $a Next, we study interactive video calls between two users, where at least one of the users is connected over a cellular network. It is known that cellular links present highly-varying network bandwidth and packet delays. If the sending rate of the video call exceeds the available bandwidth, the video frames may be excessively delayed, destroying the interactivity of the video call. Here, we present Rebera, a cross-layer design of proactive congestion control, video encoding and rate adaptation, to maximize the video transmission rate while keeping the one-way frame delays sufficiently low. Rebera actively measures the available bandwidth in real time by employing the video frames as packet trains. Using an online linear adaptive filter, Rebera makes a history-based prediction of the future capacity, and determines a bit budget for the video rate adaptation. Rebera uses the hierarchical-P video encoding structure to provide error resilience and to ease rate adaptation, while maintaining low encoding complexity and delay. Furthermore, Rebera decides in real time whether to send or discard an encoded frame, according to the budget, thereby preventing self-congestion and minimizing the packet delays. Our experiments with real cellular link traces show that Rebera can, on average, deliver higher bandwidth utilization and shorter packet delays than Apple's FaceTime.
520 $a Finally, we consider video calls affected by bursty packet losses, where forward error correction (FEC) is applied on a per-frame basis due to tight delay constraints. In this scenario, both the encoding (eFR) and the decoded (dFR) frame rates are crucial; a high eFR at low bitrates leads to larger quantization stepsizes (QS), smaller frames, hence suboptimal FEC, while a low eFR at high bitrates diminishes the perceptual quality. Coincidently, damaged frames and others predicted from them are typically discarded at receiver, reducing dFR. To mitigate frame losses, hierarchical-P coding (hPP) can be used, but at the cost of lower coding efficiency than IPP..I coding (IPP), which itself is prone to abrupt freezing in case of loss. Here, we study the received video call quality maximization for both hPP and IPP by jointly optimizing eFR, QS and the FEC redundancy rates, under the sending bitrate constraint. Employing Q-STAR, a perceptual quality model that depends on QS and average dFR, along with R-STAR, a bitrate model that depends on eFR and QS, we cast the problem as a combinatorial optimization problem, and employ exhaustive search and hill climbing methods to solve explicitly for the eFR and the video bitrate. We also use a greedy FEC packet distribution algorithm to determine the FEC redundancy rate for each frame. We then show that, for iid losses, (i) the FEC bitrate ratio is an affine function of the packet loss rate, (ii) the bitrate range where low eFR is preferred gets wider for higher packet loss rates, (iii) layers are protected more evenly at higher bitrates, and (iv) IPP, while achieving higher Q-STAR scores, is prone to abrupt freezing that is not considered by the Q-STAR model. For bursty losses, we show that (i) layer redundancies are much higher, rising with the mean burst length and reaching up to 80%, (ii) hPP achieves higher Q-STAR scores than IPP in case of longer bursts, and (iii) the mean and the variance of decoded frame distances are significantly smaller with hPP.
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538 $a Mode of access: World Wide Web
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710 2 $a Polytechnic Institute of New York University. $b Electrical and Computer Engineering. $3 1182851
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