The idea of moving picture expert group (MPEG) bitstream scaling relates to altering or scaling the amount of data in a previously compressed MPEG bitstream. The new scaled bitstream conforms to constraints that are not known nor considered when the original preceded bitstream was constructed. Numerous applications for video transmission and storage are being developed based on the MPEG video coding standard. Applications such as video on demand, trick-play track on digital video tape recorders (VTR's) and extended-play recording on VTR's motivate the idea of bitstream scaling. In this paper, we present several bitstream scaling methods for the purpose of reducing the rate of constant bit rate (CBR) encoded bitstreams. The different methods have varying hardware implementation complexity and associated trade-offs in resulting image quality. Simulation results on MPEG test sequences demonstrate the typical performance trade-offs of the methods.

In this paper, we discuss the coding strategy when a two-layer digital HDTV service is considered. We propose to use-three different strategies in performing compression to yield the best quality based on the respective requirements on bandwidth, picture quality and efficiency. These three modes are as follows. Strategy A: if the bandwidth for the enhancement layer is not sufficient to achieve high quality, the base layer encoder should perform the compression at a reduced spatial resolution using the combined bandwidth. The top layer video signal will be obtained from upsampling the base layer. Strategy B: if the bandwidth is sufficient to support quality requirements at both layers, the typical two-layer spatial scalable coding scheme is used. Strategy C: if the quality requirement at the top layer is critical, a single layer compression at a full spatial resolution is used. To receive the low resolution signal, certain bit stream scaling can be used to get a usable quality signal. A theoretical qualitative analysis gives an insight of our experimental results. Such a strategy is useful in designing future HDTV services for television sets with different capabilities, resolutions and complexities. The results are also applicable to the problem of HDTV/standard TV compatibility.

The Moving Picture Expert Group (MPEG) video coding standard has been proposed for a variety of applications for video transmission and storage. Several applications such as video on demand, trick-play track on digital VTRs and extended-play recording VTRs motivate the idea of bitstream scaling which intends to reduce the compressed bitstream size according to the outgoing channel capacity. We propose several methods for implementation of bitstream scaling. The different methods have varying hardware implementation complexity, with each having its own degree of tradeoff between required hardware and resulting image quality.

The US HDTV process has fostered substantial research and development activity over the last several years. The Advisory Committee on Advanced Television Service (ACATS) was formed to advise the FCC on the technology and systems suitable for delivery of high definition service over terrestrial broadcast channels. Four digital HDTV systems where tested at the Advanced Television Testing Center. All the systems gave excellent performance, but the results were inconclusive and a plan for a second round of tests was prepared. As each of the four systems where being readied for retest. The proponents of the four individual digital HDTV proposals worked together to define a single HDTV system which incorporated the best technology from the individual systems. The consortium of companies, called the Grand Alliance (GA), announced a combined system and submitted it to ACATS for consideration. After ACATS certification, the GA began construction of a prototype system to submit for laboratory testing at the end of 1994. This paper describes the video compression subsystem and the hardware prototype. The preprocessing, motion estimation, quantization, and rate control subsystems are described. The system uses bidirectional motion compensation, discrete cosine transform, quantization and Huffman coding. The resulting bitstream is input into a transport system which uses fixed length packets. The multiplex transport stream is input into the 8-VSB transmission system. Finally, the specifics of the hardware implementation are described and some simulation results are presented.>

The letter considers modeling methodologies of variable bit-rate (VBR) video sources for performance evaluation of integrated networks. The authors consider an example in which compressed H.261 standard VBR video is transmitted over a local area network carrying both video and data. A group-of-block (GOB) level source model has been constructed using TES (transform-expand-sample)-a new general methodology for generating stationary autocorrelated time series, which can simultaneously capture a prescribed marginal distribution and autocorrelation function. The TES model is used to drive CSMA/CD network simulations for a performance evaluation study. Numerical comparisons with a conventional autoregressive (AR) modeling approach are given.>

The packet-oriented transport approach used in the advanced digital television (ADTV) system for terrestrial HDTV broadcast is described. ADTV achieves robust HDTV delivery on terrestrial simulcast channels via MPEG video compression, prioritization of MPEG data, and 'cell-relay' type packet transport in conjunction with a two-tier physical transmission scheme. General design issues relevant to the development of the proposed transport protocol are discussed. ADTV's prioritization algorithm for partitioning MPEG-encoded video into high-priority (HP) and standard-priority (SP) bit streams is outlined. The data transport format supporting these prioritized compressed video bit streams is described. The three principal sublayers of the ADTV transport protocol are discussed in terms of specific functions, impact of system performance, and hardware implementation factors. A proof-of-concept simulation model that incorporates transport encoding and decoding functionality is outlined, and performance evaluation results are given for illustrative transmission scenarios.>

The authors consider modeling methodologies of variable-bit-rate video sources for performance evaluation of integrated networks. They consider an example in which compressed video is transmitted over a local area network carrying both video and data. A new modeling methodology called transform-expand-sample (TES) which is a method for generating autocorrelated variates, is used. TES is a nonparametric method which can accurately capture the histogram and approximate autocorrelation function of any data set. The TES model was used to drive a network simulation for performance measures.>

An investigation of statistically based approaches to the design of buffer control algorithms for interfacing a compressed digital video source to a constant rate channel is presented. As a first step to a quantitative methodology for the design and evaluation of adaptive buffer control algorithms, a detailed statistical characterization of the various encoding modes of an example broadcast quality intra/interframe differential-pulse-code-modulation (DPCM) algorithm is obtained from extensive simulation. The statistics presented include simple measures such as the overall encoded rate distributions for each encoding mode and more detailed intermode and intramode statistics. Two specific adaptive mode control algorithms are proposed. Using simulation over a large sea of representative images, the performance of the proposed statistically based algorithms is compared to that of a conventional buffer-level-based control heuristic which does not require source characterization.>

The problem of designing shared access packet-transport-based transmission systems for compressed video signals is studied. The feasibility of using conventional link-level and transport-level protocol services to transmit compressed video is examined by focusing on two practically important scenarios for compressed video transmission: (1) multipoint-to-multipoint video transmission using a 200 Mb/s implicit token passing (ITP) fiber-optic local area network (LAN); and (2) point-to-multipoint broadcast video distribution using a 90 Mb/s packet time-division multiplexing (packet-TDM) direct-broadcast satellite channel. To evaluate the performance of such shared-access broadband packet video systems accurate simulation models were developed that were driven by realistic 'broadcast quality' compressed video sources for the ITP-LAN and packet-TDM systems. The models were used to determine design tradeoffs between channel throughput, video quality (measured by clipping probability), and the transport-level and media-access-level protocol features and parameters implemented in the packet video network interface unit.>