A system of distributing video and/or audio information employs digital signal processing to achieve high rates of data compression. The compressed and encoded audio and/or video information is sent over standard telephone, cable or satellite broadcast channels to a receiver specified by a subscriber of the service, preferably in less than real time, for later playback and optional recording on standard audio and/or video tape.

Audio and video transmission and receiving system

A public switched telephone network (PSTN) (10) provides digital video signals from a video information provider (140, 160) to one or more of a plurality of subscriber premises (100, 120). A subscriber uses either a standard telephone instrument (102) over the PSTN or a dedicated control device (130, 132) over an ISDN packet network (38) to order video programming. The request is transmitted to a designated video information provider (140, 160) and digital transmission connectivity is established between the video information provider (140, 160) and the central office serving the subscriber. Connectivity between the central office and subscriber is provided by asymmetrical digital subscriber line interface units over a local loop. The interface units frequency multiplex digital video information with voice information to the subscriber and support transmission of a reverse control channel from the subscriber to the central office for transmission on the ISDN packet data network (38) back to the video information provider (140, 160). The interfaces also allow base band signalling and audio between the central office and the subscriber for conventional telephone instrument connectivity.

Pstn architecture for video-on-demand services.

Two algorithms for block motion estimation that produce performance similar to that of exhaustive search but with computation reduced by a factor of 8 or 16 are presented. The algorithms are based on motion-field and pixel subsampling. A subsampled motion field is first determined by estimating the motion vectors for a fraction of the blocks. The motion vectors for these blocks are determined by using only a fraction of the pixels at any searched location and by alternating the pixel subsampling patterns with the searched locations. The subsampled motion field is then interpolated so that a motion vector is determined for each block of pixels. The algorithms are more robust than previously proposed fast algorithms and both can easily be combined with a hierarchical search strategy. One of the algorithms is fully compatible with MPEG-I. >

New fast algorithms for the estimation of block motion vectors

A public switched telephone network (PSTN) provides digital video signals from a video information provider or digital service bureau to one or more of a plurality of subscriber premises. Administration of orders from subscribers is carried out by a video gateway and file servers at the central office. Asymmetrical digital subscriber line interface (ADSL) units over a local loop carry the necessary signalling between the subscribers and information providers. The interface units frequency multiplex digital video information with voice information to the subscriber and support transmission of a reverse control channel from the subscriber to the central office for transmission back to the information provider. Several enhancements and special features are disclosed.

Video-on-demand services using public switched telephone network

The past decade has witnessed substantial progress towards the application of low-rate speech coders to civilian and military communications as well as computer-related voice applications. Central to this progress has been the development of new speech coders capable of producing high-quality speech at low data rates. Most of these coders incorporate mechanisms to: represent the spectral properties of speech, provide for speech waveform matching, and "optimize" the coder's performance for the human ear. A number of these coders have already been adopted in national and international cellular telephony standards. The objective of this paper is to provide a tutorial overview of speech coding methodologies with emphasis on those algorithms that are part of the recent low-rate standards for cellular communications. Although the emphasis is on the new low-rate coders, we attempt to provide a comprehensive survey by covering some of the traditional methodologies as well. We feel that this approach will not only point out key references but will also provide valuable background to the beginner. The paper starts with a historical perspective and continues with a brief discussion on the speech properties and performance measures. We then proceed with descriptions of waveform coders, sinusoidal transform coders, linear predictive vocoders, and analysis-by-synthesis linear predictive coders. Finally, we present concluding remarks followed by a discussion of opportunities for future research. >

Speech coding: a tutorial review

The Moving Picture Experts Group (MPEG) of ISO is considering a standard for storage applications below 1.5 Mb/s. A video coding algorithm developed at Bellcore that has been proposed as a contribution to the standardization activity of ISO/MPEG is described. The following characteristics of the algorithm are described. The following characteristics of the algorithm are described in detail: frequent periodic reset of the temporal predictor (to provide random access), interpolative coding of every second frame, flexibility of providing adaptive Huffman code tables for digital-storage-media (DSM)-based applications (not feasible for real-time transmissions), 1/2-pel accuracy in motion estimation, and optional quantization matrix. Additional features of reverse playback, fast searches, and random access are described. >

MCPIC: a video coding algorithm for transmission and storage applications

Some digital source coding techniques for speech and video are reviewed. Predictive coding of speech, multipulse and code-excited coders and frequency-domain coders are discussed and compared for the coding of speech signals, and intraframe and still image coding and interframe coding are examined for the coding of image and video signals. The emphasis is on those algorithms that offer high compression while maintaining the perceptual quality of the source signals are discussed. Some algorithms that are general waveform coding algorithms and do not strictly depend on the input source are included. >

Source coding of speech and video signals

A hybrid algorithm has been demonstrated to be an efficient way to encode high-definition television sequences, where each field of the sequence is decomposed into four subbands, with discrete cosine transform applied to the low-frequency subband. The system implementation complexity of such an algorithm and the implementation of low-order filters that perform the subband analysis and synthesis are analyzed. The authors also examine some issues in the connection of the codec to the asynchronous time multiplexing layer for transmission on the CCITT-defined H4 channel. It is shown that a low-cost implementation of the subband/DCT codec for the transmission of distribution quality compressed digital HDTV is feasible at a rate of 135 Mb/s. >

http://ieeexplore.ieee.org/iel2/123/3395/00117333.pdf

C.-T. Chen

Papers

MCPIC: a video coding algorithm for transmission and storage applications

Source coding of speech and video signals

HDTV subband/DCT coding: analysis of system complexity