Client buffering techniques for scalable video broadcasting overbroadband networks with low user delay

In traditional near-VOD (NVOD), the number of streams required is high if the user delay goal is low (say, 2 minutes). We study the use of client buffering to reduce such bandwidth requirement. We first study a scheme based on a streaming approach termed "join-and-stream" (JAS), which broadcasts a movie in a staggered manner and uses short unicast streams to recover the time difference between the broadcast point and the arrival time. We show that such a technique is effective for movies with intermediate arrival rate. We then propose a broadcasting scheme for popular movies termed "stream-bundling." The scheme groups (i.e., "bundles") the server streams into channels of incrementally increasing bandwidth. Such high-speed bundled channels are used to deliver the beginning portion of the videos to the clients, so that the clients can merge with an on-going broadcast stream quickly. By comparing with other previously proposed broadcasting schemes (such as pyramid broadcasting, skyscraper broadcasting and harmonic broadcasting), stream-bundling is shown to achieve similar level of performance with much lower complexity (without many channels to manage and to hop). Using our two schemes, the bandwidth requirement of a system can be reduced significantly (by more than 50% in our examples) as compared with the traditional NVOD, with the cost of only a little client buffering (⩽20% of the movie length)     

Asynchronous and reliable on-demand media broadcast

We discuss wireless broadcasting of multimedia streams within a framework that allows asynchronous media access. Receivers subscribe at any time to the ongoing broadcast session, but are still able to display the media stream from the beginning. A fully scalable broadcasting scheme is presented where the media stream is appropriately segmented, and segments are protected by fountain codes. Erasure-based decoding as well as soft decoding is discussed. Asynchronous data reception and full reliability are achieved at the same time. Depending on its receiving conditions, the receiver adapts its initial playout delay to guarantee high reliability of successful playout.     

Providing VCR Functionality in Multicast Video-on-Demand Systems Using Adaptive Batching

The adaptive batching technique is shown to have better performance than the static scheme in terms of total bandwidth requirement and customer reneging probability in multicast video-on-demand (VoD) systems. In this paper, we first analyze the performance of providing VCR functions in the multicast VoD system using adaptive batching. The result shows that the system may not take any advantage from the adaptive approach if VCR operations are incorporated. In view of this, we then explore the buffer reservation and VCR resume delay to further improve the adaptive system. The trade-off among the buffer size, resume delay, and stream requirement is extensively investigated. It is found that a 30-s resume delay or 20% buffer reservation can significantly reduce the batching stream requirement in the adaptive multicast VoD system.     

Internet Multicasting of IPTV With Essentially-Zero Delay Jitter

A technology for multicasting packetized multimedia streams such as IPTV over the Internet backbone is proposed and explored through extensive simulations. An RSVP or DiffServ algorithm is used to reserve resources (i.e., bandwidth and buffer space) in each packet-switched IP router in an IP multicast tree. Each IP router uses an Input-Queued (IQ) switch architecture with unity speedup. A recently proposed low-jitter scheduling algorithm is used to pre-compute a deterministic transmission schedule for each IP router. The IPTV traffic will be delivered through the multicast tree in a deterministic manner, with bounds on the maximum delay and jitter of each packet (or cell). A playback buffer is used at each destination to filter out residual network jitter and deliver a very low-jitter video stream to each end-user. Detailed simulations of an IPTV distribution network, multicasting 75 high-definition video streams over a fully-saturated IP backbone are presented. The simulations represent the transmission of 129 billion cells of real video data and where performed on a 160-node cluster computing system. In the steady-state, each IP router buffers approx. 2 cells (128 bytes) of video data per multicast output-port. The observed delay jitter is zero when a playback buffer of 15 milliseconds is used. All simulation parameters are presented.      

Efficient Generation of Scalable Transport Stream for High Quality Service in T‐DMB

We introduce an advanced terrestrial digital multimedia broadcasting (AT‐DMB) system that overcomes the limitation of data transmission rates of T‐DMB by doubling it with the same frequency bandwidth. In this letter, we propose an efficient algorithm which generates a scalable transport stream in AT‐DMB by multiplexing certain types of elementary streams encoded using scalable video coding and an MPEG‐surround audio coder for high‐quality multimedia services.     

Provision of continuous VCR functions in interactive broadcast VoD systems

Video broadcasting is one of the feasible solutions to implement a large-scale video-on-demand (VoD) system. Nevertheless, it is still an open issue for the provision of continuous VCR functions in a delay insensitive broadcast VoD system. In this paper, we propose to jointly optimize an active buffer management scheme with contingency channels to support the VCR functions in an efficient protocol called partitioned broadcasting. We develop a greedy channel management scheme by exploiting the property of the broadcasting protocol such that the system bandwidth capacity can be fully utilized. Incorporating the channel management scheme with the partitioned video broadcast, the VoD system not only provides delay insensitive services but also handles all the interactive requests. Extensive simulation results demonstrate that the partitioned broadcasting system outperforms the traditional system based on the staggered broadcasting protocols. It is found that 20 broadcasting channels and 10 contingency channels are sufficient to support on average 720 customers for a single video with less that one second start-up delay and all types of VCR functions.     

基于多模式匹配的网络视频流识别与分类算法

快速发现网络中的视频流是进行网络视频监督及管理的前提与基础。本文通过分析网络视频流数据包的特征,提出了一种基于多模式匹配思想的网络视频流快速发现与分类算法,该算法利用不同视频流的特征建立匹配机,只需对网络数据包进行一次不完全扫描,就可以判断出数据包中是否含有视频流及类型。实验结果表明,与普通的协议解析方法相比,在满足准确性的前提下,所提算法具有更好的时间性能。     

Multimedia Stream Binding for a Pan-European Services Platform

The ability to carry stream information over a range of network types in a managed way will become an essential requirement for telecommunications network operators as future services evolve to include the transmission of audio/video and bulk data streams. This paper details the architecture of a working pan-European demonstrator offering multimedia services through the integration of stream control, as specified in the Object Management Group's (OMG's) control and management of audio/video streams, with a TINA-compliant service management environment. This CORBA-based demonstrator, known as the EURESCOM services platform (ESP), has been developed by six European telecommunications companies within EURESCOM project P715. Various modelling concepts, as defined by TINA, ODP and OMG, have been used and verified in a first prototype implementation, with connectivity provided by commercially available multimedia technology such as H.320/323 products. This EURESCOM services platform prototype is one of the first demonstrators world-wide to implement OMG's control and management of audio/video streams specification.     

Delivering scalable video with QoS to the home

User satisfaction is a key factor in the success of novel multimedia services. Yet, to enable service providers and network operators to control and maximize the quality (QoS, QoE) of delivered video streams, quite some challenges remain. In this paper, we particularly focus on three of them. First of all, objectively measuring video quality requires appropriate quality metrics and methods of assessing them in a real-time fashion. Secondly, the recent Scalable Video Coding (SVC) format opens opportunities for adapting video to the available (network) resources, yet the appropriate configuration of video encoding as well as real-time streaming adaptation are largely unaddressed research areas. Thirdly, while bandwidth reservation mechanisms in access/core networks do exist, service providers lack a means for guaranteeing QoS in the increasingly complex home networks (which they are not in full control of). In this paper we offer a broad view on these interrelated issues, by presenting the developments originating in a Flemish research project (including proof-of-concept demonstrations). From a developmental perspective, we propose an architecture combining a real-time video quality monitoring platform, on-the-fly adaptation (optimizing the video quality) and QoS reservation in a heterogeneous home network based on UPnP QoS?v3. From a research perspective, we propose a new subjective test procedure that revealed user preference for temporal scalability over quality scalability. In addition, an extensive study on optimizing HD SVC encoding in IPTV scenarios with fluctuating bandwidth showed that under certain bandwidth constraints (prohibiting sufficient fidelity) spatial scalability is a better option than quality scalability.     

基于嵌入式操作系统的码流检测仪的设计

主要介绍了嵌入式实时操作系统及在数字电视广播码流监测方面的开发应用。以研究开发用于监控数字电视网络信号的码流监测仪为例，阐述了怎样开发具有网络功能的嵌入式设备。     

The Split and Merge protocol for interactive video-on-demand

A true video-on-demand (VoD) system lets users view any video program, at any time, and perform any VCR-like user interactions. To reduce the per-user video delivery cost, multiple users may be batched and share the same video stream. Existing sharing schemes do not allow true VoD. A new protocol, called Split and Merge (SAM), does allow true VoD. SAM also provides an innovative way to merge these individuals back into the batching streams when they resume normal play mode     

Multipath routing for video delivery over bandwidth-limited networks

The delivery of quality video service often requires high bandwidth with low delay or cost in network transmission. Current routing protocols such as those used in the Internet are mainly based on the single-path approach (e.g., the shortest-path routing). This approach cannot meet the end-to-end bandwidth requirement when the video is streamed over bandwidth-limited networks. In order to overcome this limitation, we propose multipath routing, where the video takes multiple paths to reach its destination(s), thereby increasing the aggregate throughput. We consider both unicast (point-to-point) and multicast scenarios. For unicast, we present an efficient multipath heuristic (of complexity O(|V|/sup 3/)), which achieves high bandwidth with low delay. Given a set of path lengths, we then present and prove a simple data scheduling algorithm as implemented at the server, which achieves the theoretical minimum end-to-end delay. For a network with unit-capacity links, the algorithm, when combined with disjoint-path routing, offers an exact and efficient solution to meet a bandwidth requirement with minimum delay. For multicast, we study the construction of multiple trees for layered video to satisfy the user bandwidth requirements. We propose two efficient heuristics on how such trees can be constructed so as to minimize the cost of their aggregation subject to a delay constraint.     

Effective multi-program broadcasting of prerecorded video using VBR MPEG-2 coding

The tradeoff between picture quality and bandwidth usage is a prominent issue in the world of broadcasting. Since broadcasters are able to transmit multiple streams simultaneously in a channel, they face the challenge of guaranteeing the contracted picture quality required by each of the transmitted video streams while maximizing the number of video streams carried in each channel. We have developed an easy to implement MPEG-2 based multi-program video coding system suitable for digital TV broadcast, video on demand, and high definition TV over broadcast satellite networks with limited bandwidth. Compared to present broadcast systems and for the same level of contracted picture quality, our system greatly increases the number of video streams transmitted in each channel. As a result, either a large number of transponders can be freed to carry real-time broadcasting or the level of picture quality can be significantly increased. By switching from tape storage to video server technology, the need for numerous playback (VTR) systems at the headend is eliminated. In addition, the most of the complete MPEG-2 encoders are replaced by much less complex MPEG-2 transcoders. All this means a much more cost-effective solution for broadcast stations.     

Four paradigms for indexing video conferences

Meetings in which participants are linked by video, audio, and shared computer applications produce several parallel information streams. We created a meeting indexer, Jabber, that uses content-based indexing of the audio stream to access these parallel streams. It performs speech recognition on the audio stream, then groups the recognized words into semantically linked trees. The user interface is designed to display information with minimal distraction during meetings     

Interactive Broadcasting System for VBR Encoded Videos

Video broadcasting has been proved to be an efficient technique to increase the scalability of a video-on-demand (VoD) system. In this paper, we address the problems in providing interactive functions for VBR encoded videos in a broadcast VoD system. A traffic smoothing scheme is proposed to support the VCR functions in delivering VBR videos over CBR channels by the staggered broadcasting protocol. By introducing a small buffering delay, the customers are able to join back to the broadcasting groups after the interactive functions. A system model is then developed to determine the optimal parameters such that the system can meet the delay requirement as well as provide the expected quality of service to the customers. The results show that the proposed system framework is very efficient in terms of bandwidth requirement and buttering delay to provide interactive VoD services.     

An integrated voice and data multiple-access scheme for a land-mobile satellite system

A Land-Mobile Satellite System (LMSS) is a satellite-based communications network which provides voice and data communications to mobile users in a vast geographical area. By placing a "relay tower" at a height of 22300 mi, an LMSS can provide ubiquitous radio communication to vehicles roaming in remote or thinly populated area. LMSS is capable of supporting a variety of services, such as two-way alphanumeric service, paging service, full-duplex voice service, and half-duplex dispatch service. A Network Management Center (NMC) will handle the channel requests, channel assignments, and in general the network control functions. A pool of channels is managed at the NMC to be shared by all mobile users. An integrated demand-assigned multiple-access protocol has been developed for the experimental LMSS. The pool of channels is divided into reservation channels and information channels. The information channels can be assigned by the NMC to be either voice channels or data channels. Each mobile user must send a request through one of the reservation channels to the NMC via the ALOHA random-access scheme. Once the request is received and processed, the NMC will examine the current traffic condition and assign an information channel to the user. NMC will periodically update the partitions between the reservation channels, voice channels, and data channels to optimize system performance. Data channel requests are queued at the NMC while voice channel requests are blocked calls cleared. Various operational scenarios have been investigated. Tradeoffs between the data and voice users for a given delay requirement and a given voice call blocking probability have been studied. In addition, performance impacts of such technological advancements as satellite on-board switching and variable bandwidth assignment are discussed.     

PERFORMANCE ANALYSIS OF HETEROGENEOUS TRAFFIC ON AN INTEGRATED NETWORK LINK WITH FINITE WAITING ROOM AND ANTICIPATED-RELEASE RESERVATION POLICY

We consider a system where the superposition of two heterogeneous Poisson traffic streams is offered to an integrated network link in synchronous transfer mode, where one stream follows the blocked-and-cleared mode (‘loss’ mode) and the other can wait (finitely) if bandwidth is not available for connection at the time of arrival (‘hold’ mode). We assume that each stream has different bandwidth requirements per call. A reservation scheme, called anticipated-release policy, is introduced where an arrival is accepted into a waiting room only if the amount of time this customer is expected to wait is within acceptable limits. For such a loss/hold system, we provide analytical performance models for exponential service time distributions for both streams as well as for the non-exponential service time distribution case for the traffic stream in ‘hold’ mode. We also present a method on how to model the waiting time distribution of the traffic stream with ‘hold’ mode. From numerical studies, we observe that blocking can be reduced considerably for both services just by introducing a small waiting room for one traffic class compared to ‘loss’ mode for both traffic classes. Furthermore, this holds true for the case when a maximum tolerable time limit is imposed on the waiting. Finally, our results indicate that this loss/hold scenario with limited waiting room appears to be virtually insensitive to the service time distribution of the ‘hold’ mode traffic.     

MPEG-2数字视频广播传输流中节目参考时钟的处理

MPEG 2标准系统层中定义的传输流已经在事实上成为数字电视领域中系统层传输的普遍标准。数字视频广播系统中传输流的处理由复用器完成。由于节目参考时钟是编解码器中共同系统时钟的标签 ,精确度要求非常严格 ,因此复用器研制的一个难点就是节目参考时钟的处理。文中利用复用器的传输流输出端对节目参考时钟进行精确插入 ,充分利用了数字信号处理器和 9位先入先出缓存的特性 ,精确而简易地满足了数字视频广播系统的时序要求。     

FEC scheme for a TDM-OFDM based satellite radio broadcasting system

There are two licenses, 12.5 MHz each, in the S-band for digital satellite-to-vehicle radio broadcasting in the United States. The potential advantages of such a system is that a motorist can enjoy commercial-free music, digital-quality sound and seamless coast-to-coast coverage. One proposal for the broadcast system is to have two satellites covering the continental USA at any given time. There will also be terrestrial repeaters in cities where the receivers on the vehicles cannot see the two satellites. The channels of these systems are affected by Rician, Rayleigh and flat fading caused by shadowing. This paper proposes a forward error correction (FEC) scheme that is not only robust against fading but also enables a low-delay tuning channel so that minimum tuning delay will occur when a user is switching and selecting programs. The scheme uses multiple source coded bit streams employing different interleaver depths. Large interleavers are used to ensure good decoded signal quality while small interleavers are used to minimize tuning delay. The proposed scheme also ensures that a program will not be interrupted by momentary shadowing frequently experienced by a motorist when, for example, a vehicle goes under a highway overpass. The impact of interleaver design on the real-time end-to-end delay and fading due to shadowing is analyzed. Finally, the channel code performance in Rician and Rayleigh fading channels are also presented     

Characterization of slice-based H.264/AVC encoded video traffic using token buckets

An explicit slice-based mode type selection scheme for use in H.264/AVC has recently been developed, which reduces the burstiness effect of standard frame-based H.264/AVC by breaking up the Group of Picture structure. In this paper, slice-based encoded video streams are characterized using the token bucket traffic model and compared to standard frame-based encoded streams. Both lossless, loss bounded and delay bounded token bucket models are investigated and the high quantiles are found for the amount of loss. Loss above the amount given by the high quantiles will happen only with a very small probability. It is shown that the reduced burstiness for the slice-based video encoding leads to lower token bucket parameters compared to frame-based video encoding for a stream without scene changes, and a larger reduction in the token bucket parameters compared to the frame-based video encoding when a small amount of delay or loss is allowed for the stream with scene changes. Next, reshuffling of the frames of the video streams with scene changes is employed to better understand the effects of long-range dependence on the token bucket parameters. Only small effects are found from reshuffling the scenes, but reshuffling of the frames inside the scenes leads to lower token bucket parameters. Finally, an approach to estimate the parameters for the token bucket model using simple characteristics of the slice-based stream is developed.