MPEG Video Streams Research Articles

Design and Implementation of the Stony Brook Video Server

The system architecture of the Stony Brook Video Server (SBVS), which guarantees end-to-end real-time video playback in a client-server setting, is presented. SBVS employs a real-time network access protocol, RETHER, to use existing Ethernet hardware as the underlying communications media. The video server tightly integrates the bandwidth guarantee mechanisms for network transport and disk I/O. SBVS's stream-by-stream disk scheduling scheme optimizes the effective disk bandwidth without incurring significant scheduling overhead. To demonstrate the feasibility of the proposed architecture, we have implemented a prototype called SBVS-1, which can support five concurrent MPEG-1 video streams on an Intel 486DX2/EISA PC. To our knowledge, this system is the first video server that provides an end-to-end performance guarantee from the server's disks to the each user's display over standard Ethernet. This paper describes the implementation details of integrating network and I/O bandwidth guarantee mechanisms, and the performance measurements that drive and/or validate our design decisions. © 1997 by John Wiley & Sons, Ltd.

The effect of multiple time scales and subexponentiality in MPEG video streams on queueing behavior

Guided by the empirical observation that real-time MPEG video streams exhibit both multiple time scale and subexponential characteristics, we construct a video model that captures both of these characteristics and is amenable to queueing analysis. We investigate two fundamental approaches for extracting the model parameters: using sample path and second-order statistics-based methods. The model exhibits the following two canonical queueing behaviors. When strict stability conditions are satisfied, i.e., the conditional mean of each scene is smaller than the capacity of the server, precise modeling of the interscene dynamics (long-term dependency) is not essential for the accurate prediction of small to moderately large queue sizes. In this case, the queue length distribution is determined using quasistationary (perturbation theory) analysis. When weak stability conditions are satisfied, i.e., the conditional mean of at least one scene type is greater than the capacity of the server, the dominant effect for building a large queue size is the subexponential (long-tailed) scene length distribution. In this case, precise modeling of intrascene statistics is of secondary importance for predicting the large queueing behavior. A fluid model, whose arrival process is obtained from the video data by replacing scene statistics with their means, is shown to asymptotically converge to the exact queue distribution. Using the transition scenario of moving from one stability region to the other by a change in the value of the server capacity, we synthesize recent queueing theoretic advances and ad hoc results in video modeling, and unify a broad range of seemingly contradictory experimental observations found in the literature. As a word of caution for the widespread usage of second-order statistics modeling methods, we construct two processes with the same second-order statistics that produce distinctly different queueing behaviors.

IP multicast of real-time MPEG over ATM

IP and ATM multicasting are compared, and current approaches for transporting IP multicast over ATM networks are reviewed. The use of ATM backbones to carry multiple high quality video streams is noted, and the relative strengths and weaknesses of ATM Forum LAN Emulation and recent IETF proposals for carrying IP multicast over ATM are discussed. An experimental application is described for multicasting multiple MPEG-1 video streams between routers connected over a live ATM campus network, and IP multicast services are noted as increasingly important and powerful in the development of the so-called Information Superhighway.

Compressed video processing for cut detection

One of the challenging problems in video databases is the organisation of video information. Segmenting a video into a number of clips and characterising each clip has been suggested as one mechanism for organising video information. This approach requires a suitable method to automatically locate cut points (boundaries between consecutive camera shots in a video). Several existing techniques solve this problem using uncompressed video. Since video is increasingly being captured, moved, and stored in compressed form, there is a need for detecting shot boundaries directly in compressed video. The authors address this issue and show certain feature extraction steps in MPEG compressed video that allow the implementation of most of the existing cut detection methods developed for uncompressed video for MPEG video stream. They also examine the performance of three tests for cut detection by viewing the problem of cut detection as a statistical hypothesis testing problem. As the experimental results indicate, the statistical hypothesis testing approach permits fast and accurate detection of video cuts.

A performance analysis of an object-based I/O architecture in a video server environment

In this paper, we present a performance analysis of how effective video server applications can be supported on personal computers (PCs) connected through a local area network (LAN). We considered both the standard 16-Mbit/s token ring and a 100-Mbit/s token ring, which follows closely the specifications for the Fiber Distributed Data Interface (FDDI). We examined three I/O architectures for a PC-based video server: an interrupt-driven I/O architecture, a peer-to-peer I/O architecture, and a concurrent, object-based I/O architecture that we proposed. The video server must support multiple MPEG-1 video streams at the same time to multiple clients on the LAN. We found that the network protocol layers require a lot of processing power, and that an implementation of our proposed I/O architecture, which takes advantage of the available power of the host processor to off-load I/O adapters, can deliver much better performance, and is more cost-effective, than other I/O architectures in a video server environment.

Performance analysis of a peer-to-peer I/O architecture in video server environments

In the personal computing (PC) and workstation environments, more and more I/O adapters are becoming complete functional subsystems that are intelligent enough to handle I/O operations on their own without much intervention from the host processor. The IBM Subsystem Control Block (SCB) architecture has been defined to enhance the potential of these intelligent adapters by defining services and conventions that deliver command information and data to and from the adapters. The SCB architecture has two operating modes. The Locate Mode represents the conventional, interrupt-driven I/O protocol used in many current personal computers. The Move Mode embodies an advanced, peer-to-peer I/O protocol proposed for the next generation of personal computers. In this paper, we would like to discuss and present a performance analysis of the SCB architecture in typical video server environments. In particular, we study a video server capable of providing support for simultaneous MPEG-1 video streams to multiple clients on a 16-Mbps token-ring network. We also consider the performance impact of a hypothetical 100-Mbps token-ring technology on the video server performance.

Algorithmic and architectural enhancements for real-time MPEG-1 decoding on a general purpose RISC workstation

Traditional video decoders require use of specially designed video decompression processors. We present novel algorithmic and architectural enhancements that allowed for the first time the real-time decompression of MPEG-1 video and audio streams on a low-end, general purpose RISC processor. For video decompression, efficient algorithmic implementations were derived by examining the Huffman decoder, the inverse quantizer and the inverse DCT as a single system. For audio decompression, a new DCT based implementation of the subband filtering operation yields 30% speed improvement in the audio decoding process and 17% speed improvement in overall audio and video decoding. Besides algorithmic enhancements, a new set of multimedia instructions and minor changes in the design of a traditional RISC ALU allowed increased parallelism of pixel-based operations with minimal design and control overhead. Experimental results show that with the synergistic combination of algorithmic and architectural enhancements a multimedia-enhanced RISC processor can achieve higher decoding rates than generic RISC and CISC processors, even when these processors operate at higher clock rates and have larger instruction and data caches. >

Evaluation of UMTS RLC Parameters for MPEG-4 Video Streaming

Universal Mobile Telecommunication Systems (UMTS) known as Third Generation (3G) mobile phone systems using Wideband Code Division Multiple Access (WCDMA) standard aim to provide a high bit rate of services to enable a high quality of multimedia communications. The 3G specification entity, 3GPP, has defined a reliable link layer protocol, Radio Link Control (RLC), for hiding transmission errors from upper layers. Due to the complexity of the protocol and the a number of parameter configurations available, there are many ways that can degrade a system performance, such as incorrect parameters configuration, buffer missmanagement, protocol stalling, etc. MPEG-4 is widely used for video stream services due to its good video quality at low bit rate. In this paper, we have studied on the impacts of RLC parameters, especially RLC timer and polling mechanism, for MPEG-4 video stream transmission by using a computer simulation. Our simulation results provide some suggestions value to optimize such parameters, e.g. RLC time and polling.

Machine reads typewritten pages

We propose a novel object-oriented design approach for developing a platform-neutral IP/TV presentation system for playing live TV-like MPEG video streams on the Internet multicast by RTP-compliant servers. Unlike proprietary viewers, such as Cisco's IP/TV viewers, which run only on Windows platforms, our system runs on any Java Virtual Machine environments. We demonstrate the effectiveness of our design approach via actual experimentations. We also propose a Java-based relay agent architecture for relaying RTP streams across multicast-enabled routers. We show that our system is particularly suitable to run as relay agents in this architecture to tackle the cross-platform issue and to accommodate various audio or video media sources efficiently. By adhering to the Internet and multimedia standards, our system is generically applicable to various media streaming systems.