Delivered Video Quality Research Articles

Dynamic Seamless Resource Allocation for Live Video Compression on a Kubernetes Cluster

A solution is proposed on top of Kubernetes to dynamically allocate service resources without service interruption. It is the basis for optimizing a live video compression service. It is demonstrated that dynamic resource allocation can benefit a video compression application, either by reducing the resource consumption, hence costs, or by enhancing delivered video quality. By combining the proposed solution with an elastic encoder and machine learning (ML) for content complexity estimation, a content- and application-aware dynamic resource orchestrator for realtime video compression is designed. Preliminary experimental results using ATEME Titan Live Micro-Services <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">1</sup> encoders demonstrate substantial bitrate reductions on even the most demanding channel.

Tile-Based Edge Caching for 360° Live Video Streaming

<inline-formula xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> <tex-math notation="LaTeX">$360^{o}$ </tex-math></inline-formula> video is becoming an increasingly popular technology on commercial social platforms and vital part of emerging Virtual Reality/Augmented Reality (VR/AR) applications. However, the delivery of <inline-formula xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> <tex-math notation="LaTeX">$360^{o}$ </tex-math></inline-formula> video content in mobile networks is challenging because of its size. The encoding of <inline-formula xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> <tex-math notation="LaTeX">$360^{o}$ </tex-math></inline-formula> video into multiple quality layers and tiles and edge cache-assisted video delivery have been proposed as a remedy to the excess bandwidth requirements of <inline-formula xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> <tex-math notation="LaTeX">$360^{o}$ </tex-math></inline-formula> video delivery systems. Existing works using the above tools have shown promising performance for Video-on-Demand (VoD) <inline-formula xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> <tex-math notation="LaTeX">$360^{o}$ </tex-math></inline-formula> delivery, but they cannot be straightforwardly extended in a live-streaming setup. Motivated by the above, we study edge cache-assisted <inline-formula xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> <tex-math notation="LaTeX">$360^{o}$ </tex-math></inline-formula> live video streaming to increase the overall quality of the delivered <inline-formula xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> <tex-math notation="LaTeX">$360^{o}$ </tex-math></inline-formula> videos to users and reduce the service cost. We employ Long Short-Term Memory (LSTM) networks to forecast the evolution of the content requests and prefetch content to caches. To further enhance the delivered video quality, users located in the overlap of the coverage areas of multiple Small Base Stations (SBSs) are allowed to receive data from any of these SBSs. We evaluate and compare the performance of our algorithm with Least Frequently Used (LFU), Least Recently Used (LRU), and First In First Out (FIFO) algorithms. The results show the superiority of the proposed approach in terms of delivered video quality, cache-hit-ratio and backhaul link usage.

Multicast Optimization for Video Delivery in Multi-RAT Networks

Mobile network operators today need to deliver new types of multimedia content, such as 360° video, to an ever-growing population of users. This creates a pressing need for network mechanisms that can support such demanding services in an economically-efficient fashion. This work focuses on applications that deliver a video file concurrently to multiple users which can utilize different radio technologies, namely cellular and Wi-Fi networks. We propose a mechanism for the orchestration of LTE multicast and Wi-Fi unicast transmissions by optimizing jointly: the design of multicast groups, the spectrum and bandwidth management in both networks, and the video encoding quality for each user (or, group). We consider two key performance criteria: minimizing the utilized LTE spectrum resources, and maximizing the delivered video quality. We formulate the respective optimization problems, prove they are NP-hard, and design an exact algorithm for solving them in near-real time. We employ a wealth of simulation experiments, using 3GPP-compliant parameters, that compare our approach with state-of-the-art benchmarks. Our findings suggest that, in representative scenarios, this multi-RAT orchestration can save up 55% LTE radio resources and increase up to 46% the delivered video quality.

Multimedia Data Flow Traffic Classification Using Intelligent Models Based on Traffic Patterns

Nowadays, there is high interest in modeling the type of multimedia traffic with the purpose of estimating the network resources required to guarantee the quality delivered to the user. In this work we propose a multimedia traffic classification model based on patterns that allows us to differentiate the type of traffic by using video streaming and network characteristics as input parameters. We show that there is low correlation between network parameters and the delivered video quality. Because of this, in addition to network parameters, we also add video streaming parameters in order to improve the efficiency of our system. Finally, it should be noted that, based on the objective video quality received by the user, we have extracted traffic patterns that we use to perfor

A scalable WebRTC-based framework for remote video collaboration applications

Remote video collaboration is common nowadays in conferencing, telehealth and remote teaching applications. To support these low-latency and interactive use cases, Real-Time Communication (RTC) solutions are generally used. WebRTC is an open-source project for real-time browser-based conferencing, developed with a peer-to-peer architecture in mind. In this peer-to-peer architecture, each sending peer needs to encode a separate, independent stream for each receiving peer participating in the remote session, which makes this approach expensive in terms of encoders and not able to scale well for a large number of users. This paper proposes a WebRTC-compliant framework to solve this scalability issue, without impacting the quality delivered to the remote peers. In the proposed framework, each sending peer is only equipped with a limited number of encoders, much smaller than and independent of the number of receiving peers. Consequently, each encoder transmits to a multitude of receivers at the same time, to improve scalability. A centralized node based on the Selective Forwarding Unit (SFU) principle, called conference controller, forwards the best stream to the receiving peers, based on their bandwidth conditions. Moreover, the conference controller dynamically recomputes the encoding bitrates of the sending peers, to maximize the quality delivered to the receiving peers. This approach allows to closely follow the long-term bandwidth variations of the receivers, even with a limited number of encoders at sender-side, and increase the delivered video quality. An integer linear programming formulation for the bitrate recomputation problem is presented, which can be optimally solved when the number of receivers is small. An approximate, scalable method is also proposed using the K-means clustering algorithm. The gains brought by the proposed framework have been confirmed in both simulation and emulation, through a testbed implementation using the Google Chrome browser and the open-source Jitsi-Videobridge software. Particularly, we focus on a remote collaboration scenario where the interaction among the remote participants is dominated by a single peer, as in a remote teaching scenario. When a single sending peer equipped with three encoders transmits to 28 receiving peers, the proposed framework improves the average received video bitrate up to 15%, compared to a static solution where the encoding bitrates do not change over time. Moreover, the dynamic bitrate recomputation is more efficient than a static association in terms of encoders used at sender-side. For the same configuration mentioned above, the same received bitrate is obtained in the static case using four encoders as in the dynamic case using three encoders.

An NFV-Based Video Quality Assessment Method over 5G Small Cell Networks

This article proposes a video quality assessment (VQA) mechanism for next-generation (5G) mobile networks, following the small cell deployment architecture. The proposed method uses the structural similarity (SSIM) index as a reduced reference metric and is suitable for implementation as a virtual network function (VNF) within an IT infrastructure located close to the small cell. It enables the in-service monitoring of the delivered video quality, which is a useful tool for mobile network operators to monitor their customers’ satisfaction. An advantage of the proposed method is that the complex and power-consuming VQA process is performed at the edge of the network, and not at the user equipment itself, thus significantly reducing power consumption. The authors used a small cell experimental testbed to implement and evaluate the performance of the proposed method. The experimental results show that the proposed method can monitor the video quality when the network is degraded.

Optimized Upload Strategies for Live Scalable Video Transmission from Mobile Devices

Sharing live multimedia content is becoming increasingly popular among mobile users. In this article, we study the problem of optimizing video quality in such a scenario using scalable video coding (SVC) and chunked video content. We consider using only standard stateless HTTP servers that do not need to perform additional processing of the video content. Our key contribution is to provide close to optimal algorithms for scheduling video chunk upload for multiple clients having different viewing delays. Given such a set of clients, the problem is to decide which chunks to upload and in which order to upload them so that the quality-delay tradeoff can be optimally balanced. We show by means of simulations that the proposed algorithms can achieve notably better performance than naive solutions in practical cases. Especially the heuristic-based greedy algorithm is a good candidate for deployment on mobile devices because it is not computationally intensive but it still delivers in most cases on-par video quality compared to the more complex local optimization algorithm. We also show that using shorter video segments and being able to predict bandwidth and video chunk properties improve the delivered video quality in certain cases.

QoS-aware scalable video streaming using data distribution service

Enabling Real-Time video streaming over wireless networks is a challenging task due to the time-varying channel conditions and the limited network resources. The instability of wireless networks leads to problems such as limited and time-varying bandwidth, and unexpected traffic congestion when transmitting a burst of video streams. In Real-Time video streaming, each frame must be delivered and decoded by its playback time. Recently, layer coding (LC) has enabled Real-Time and scalable video streaming to clients of heterogeneous capabilities by dropping upper enhancement layers without the need of re-encoding and with much less bit rate. However, layer coding still facing unfair layer protection problem in which packets from the base or lower layers might be dropped while there is a chance to drop packets from the upper enhancement layers. Losing packets from the base layer can significantly affect the delivered video quality and sometimes lead to an interruption especially in error-prone networks as wireless networks. Architectural solutions at the middleware level introduce higher flexibility, more efficiency in development time and more QoS control. This work investigates the behavior of video streaming over Real-Time publish-subscribe based middleware. We propose and develop an unequal layer protection mechanism for Real-Time video streaming based on the Data Distribution Service (DDS) middleware. A combination of video quality of service (QoS) is proposed to adapt the video transmission to the time-varying network changes. The performance of Real-Time video streaming is measured over IEEE 802.11 g WLAN network. The results show a graceful degradation of video quality while maintaining a robust video streaming free of visible error or interruptions. This is due to the intentionally dropping of some enhancement video stream layers in order to protect lower layers, and therefore maintain a continuous video flow of acceptable quality. Numerical results indicate that the proposed scheme is able to achieve higher throughput and improve the average received video quality especially in a large number of video subscribers.

Identifying QoE optimal adaptation of HTTP adaptive streaming based on subjective studies

HTTP Adaptive Streaming (HAS) technologies, e.g., Apple HLS or MPEG-DASH, automatically adapt the delivered video quality to the available network. This reduces stalling of the video but additionally introduces quality switches, which also influence the user-perceived Quality of Experience (QoE). In this work, we conduct a subjective study to identify the impact of adaptation parameters on QoE. The results indicate that the video quality has to be maximized first, and that the number of quality switches is less important. Based on these results, a method to compute the optimal QoE-optimal adaptation strategy for HAS on a per user basis with mixed-integer linear programming is presented. This QoE-optimal adaptation enables the benchmarking of existing adaptation algorithms for any given network condition. Moreover, the investigated concept is extended to a multi-user IPTV scenario. The question is answered whether video quality, and thereby, the QoE can be shared in a fair manner among the involved users.

비디오 스트리밍 서비스를 위한 MMT 기반 적응적 QoS 연구

This paper discusses QoS enhancement in the Best-effort services of the service plan provided by MPEG Media Transport (MMT) systems for video streaming applications. Among MMT services, i.e. per-flow, per-class, and best-effort services, the server does not provide guaranteed bandwidth for the best-effort service only. Therefore, in the best-effort services, a bandwidth access priority is defined for various services, where the lowest priority is assigned to the low-level video services. To alleviate the issue of bandwidth limitation in the best-effort services, this paper investigates transmission of low-resolution video with low bitrate and up-sampling. Our experimental results prove the superiority of the proposed method in terms of delivered video quality.

In-Network Quality Optimization for Adaptive Video Streaming Services

HTTP adaptive streaming (HAS) services allow the quality of streaming video to be automatically adapted by the client application in face of network and device dynamics. Due to their advantages compared to traditional techniques, HAS-based protocols are widely used for over-the-top (OTT) video streaming. However, they are yet to be adopted in managed environments, such as ISP networks. A major obstacle is the purely client-driven design of current HAS approaches, which leads to excessive quality oscillations, suboptimal behavior, and the inability to enforce management policies. Moreover, the provider has no control over the quality that is provided, which is essential when offering a managed service. This article tackles these challenges and facilitates the adoption of HAS in managed networks. Specifically, several centralized and distributed algorithms and heuristics are proposed that allow nodes inside the network to steer the HAS client's quality selection process. The algorithms are able to enforce management policies by limiting the set of available qualities for specific clients. Additionally, simulation results show that by coordinating the quality selection process across multiple clients, the proposed algorithms significantly reduce quality oscillations by a factor of five and increase the average delivered video quality by at least 14%.

Improved video streaming scheme for internet television with broadband wireless access

Internet television video–streaming schemes do not respond well when access is through a broadband wireless link. A simple but effective negative acknowledgment–based scheme has been designed to meet the needs of Internet television. When used with broadband wireless access, simulations show that the scheme has the better features of both raw UDP transport and the standardised Datagram Congestion Control Protocol but does not have those protocols' disadvantages. The delivered video quality is also found to be superior to multiple–connection, congestion–controlled streaming. This is important for mobile video delivery, as video quality is a key determinant of customer choice.

Robust Video Communication over an Urban VANET

Video communication within a Vehicular Ad Hoc Network (VANET) has the potential to be of considerable benefit in an urban emergency, as it allows emergency vehicles approaching the scene to better understand the nature of the emergency. However, the lack of centralized routing and network resource management within a VANET is an impediment to video streaming. To overcome these problems the paper pioneers source-coding techniques for VANET video streaming. The paper firstly investigates two practical multiple-path schemes, Video Redundancy Coding (VRC) and the H.264/AVC codec's redundant frames. The VRC scheme is reinforced by gradual decoder refresh to improve the delivered video quality. Evaluation shows that multiple-path 'redundant frames' achieves acceptable video quality at some destinations, whereas VRC is insufficient. The paper also demonstrates a third source coding scheme, single-path streaming with Flexible Macroblock Ordering, which is also capable of delivery of reasonable quality video. Therefore, video communication between vehicles is indeed shown to be feasible in an urban emergency if the suitable source coding techniques are selected.

Interval Type-2 Fuzzy Logic Congestion Control for Video Streaming Across IP Networks

Intelligent congestion control is vital for encoded video streaming of a clip or film, as network traffic volatility and the associated uncertainties require constant adjustment of the bit rate. Existing solutions, including the standard transmission control protocol (TCP) friendly rate control equation-based congestion controller, are prone to fluctuations in their sending rate and may respond only when packet loss has already occurred. This is a major problem, because both fluctuations and packet loss affect the end-user's perception of the delivered video. A type-1 (T1) fuzzy logic congestion controller (FLC) can operate at video display rates and can reduce packet loss and rate fluctuations, despite uncertainties in measurements of delay arising from congestion and network traffic volatility. However, a T1 FLC employing precise T1 fuzzy sets cannot fully cope with the uncertainties associated with such dynamic network environments. A type-2 FLC using type-2 fuzzy sets can handle such uncertainties to produce improved performance. This paper proposes an interval type-2 FLC that achieves a superior delivered video quality compared with existing traditional controllers and a T1 FLC. To show the response in different network scenarios, tests demonstrate the response both in the presence of typical Internet cross-traffic as well as when other video streams occupy a bottleneck on an All-Internet protocol (IP) network. As All-IP networks are intended for multimedia traffic, it is important to develop a form of congestion control that can transfer to them from the mixed traffic environment of the Internet. It was found that the proposed type-2 FLC, although it is specifically designed for Internet conditions, can also successfully react to the network conditions of an All-IP network. When the control inputs were subject to noise, the type-2 FLC resulted in an order of magnitude performance improvement in comparison with the T1 FLC. The type-2 FLC also showed reduced packet loss when compared with the other controllers, again resulting in superior delivered video quality. When judged by established criteria, such as TCP-friendliness and delayed feedback, fuzzy logic congestion control offers a flexible solution to network bottlenecks. These findings offer the type-2 FLC as a way forward for congestion control of video streaming across packet-switched IP networks.

Emergency Video Multi-Path Transfer over Ad Hoc Wireless Networks

Ad Hoc networks utilize multi-hop radio relay without the need for a base station, supporting mobility and allowing them to be quickly deployed in an emergency. Real-time video communication across an ad hoc network allows helpers to better understand the nature of the problem within a disaster area but the lack of centralized routing and network resource management is challenging, particularly if the wireless nodes have limited processing power. Multi-path solutions have been proposed for video transfer. This paper investigates two practical schemes, Video Redundancy Coding and a proposal of the paper, H.264 codec redundant frames. The paper reports that redundant frames when used in combination with multipath do result in as much as 10 dB improvement in delivered video quality, making the difference between acceptable and unacceptable visual communication in a disaster scenario.

Fuzzy logic congestion control for broadband wireless IPTV

It is demonstrated that interval type-2 fuzzy logic control (IT2 FLC) is more robust than traditional fuzzy logic congestion control of video streaming. An IT2 FLC is compared to the well-known TFRC and TEAR congestion controllers for internet multimedia streaming. On an all-IP network with broadband wireless access, delivered video quality is improved with the IT2 FLC by about 1 dB for each client, once the offered traffic (up to 50 video streams) exceeds the capacity available to video over the wireless link.

A new framework for MPEG video delivery over heterogeneous networks

The failure of legacy packet markers, such as the Single Rate Three Color Marker (SRTCM) and the Two Rate Three Color Marker (TRTCM), to protect important data from being dropped, renders a poor delivery of video quality over Differentiated Services (DiffServ) networks due to the hierarchical structure of MPEG video. Therefore, we propose a content-aware packet marking scheme, which we call Two Markers System (TMS), to improve the delivery quality of MPEG video streams. TMS places marker modules at both the video source and the edge of a DiffServ network. The source marking modules act at the application-level to set an importance indicator in the IP header of the MPEG frame packet by evaluating the contribution of the packet to the expected picture quality perceived by an end user. Meanwhile, at the edge of the DiffServ network, the Enhanced Token Bucket Three Color Marker (ETBTCM) is proposed to perform packet marking at the network-level in accordance with the pre-assigned importance of individual video frame packets. When the traffic flow is measured as out-of-profile, the less important packets will be marked with higher probability than the more important ones. Therefore, in the event of network congestion, less important frame packets can be dropped before more important ones. Consequently, a better delivered video quality can be provided. Further, due to the convenience offered by wireless networks, more users choose mobile devices to connect to the Internet. Therefore, a Video Differentiator is suggested, placed at the entry to the IEEE 802.11e wireless network, to provide traffic differentiation and better service for video traffic. We use authentic MPEG4 video traffic traces to evaluate this framework. Results show that better video quality can be delivered over our proposed new framework.

Unequal Protection of Video Streaming through Adaptive Modulation with a Trizone Buffer over Bluetooth Enhanced Data Rate

Bluetooth enhanced data rate wireless channel can support higher-quality video streams compared to previous versions of Bluetooth. Packet loss when transmitting compressed data has an effect on the delivered video quality that endures over multiple frames. To reduce the impact of radio frequency noise and interference, this paper proposes adaptive modulation based on content type at the video frame level and content importance at the macroblock level. Because the bit rate of protected data is reduced, the paper proposes buffer management to reduce the risk of buffer overflow. A trizone buffer is introduced, with a varying unequal protection policy in each zone. Application of this policy together with adaptive modulation results in up to 4 dB improvement in objective video quality compared to fixed rate scheme for an additive white Gaussian noise channel and around 10 dB for a Gilbert-Elliott channel. The paper also reports a consistent improvement in video quality over a scheme that adapts to channel conditions by varying the data rate without accounting for the video frame packet type or buffer congestion.

Buffer size resilience for Bluetooth video streams from adaptive ARQ with active discard

ARQ in various forms is an important defence against RF error bursts but needs to be adaptive when delay sensitive encoded video is transmitted. It is demonstrated that adaptive ARQ itself is sensitive to mobile device buffer size and should be combined with an active discard policy based on picture type. For some Bluetooth send buffer sizes, this leads to up to 10 dB improvement in delivered video quality over adaptive ARQ alone.

Cross-Layer Design for Video Transmission over Wireless Rician Slow-Fading Channels Using an Adaptive Multiresolution Modulation and Coding Scheme

We describe a multilayered video transport scheme for wireless channels capable of adapting to channel conditions in order to maximize end-to-end quality of service (QoS). This scheme combines a scalable H.263+ video source coder with unequal error protection (UEP) across layers. The UEP is achieved by employing different channel codes together with a multiresolution modulation approach to transport the different priority layers. Adaptivity to channel conditions is provided through a joint source-channel coding (JSCC) approach which attempts to jointly optimize the source and channel coding rates together with the modulation parameters to obtain the maximum achievable end-to-end QoS for the prevailing channel conditions. In this work, we model the wireless links as slow-fading Rician channel where the channel conditions can be described in terms of the channel signal-to-noise ratio (SNR) and the ratio of specular-to-diffuse energy . The multiresolution modulation/coding scheme consists of binary rate-compatible punctured convolutional (RCPC) codes used together with nonuniform phase-shift keyed (PSK) signaling constellations. Results indicate that this adaptive JSCC scheme employing scalable video encoding together with a multiresolution modulation/coding approach leads to significant improvements in delivered video quality for specified channel conditions. In particular, the approach results in considerably improved graceful degradation properties for decreasing channel SNR.