We present a single-chip, MPEG-2 main profile at main level, audio and video encoder and decoder. It combines a RISC core, a 24-bit DSP, video and audio interface units, and several dedicated processing units. A programmable video interface unit supports multiple modes of pre- and post-processing and on-screen display (OSD). The codec has been implemented using a standard-cell library in 0.18 /spl mu/m CMOS technology.

In this paper we present a single chip MPEG-2, MP@ML, video encoder/decoder. It combines a RISC core and several dedicated processing units organized as a process pipeline. It includes a generic 16-bit host interface and direct interfaces to video modulators and demodulators. It has been implemented using a standard-cell library, in a four-layer metal, 0.35 /spl mu/m CMOS technology. Its small size (less than 3 million transistors) and low implementation cost make it suitable for a wide range of consumer applications.

Most of the current video coding standards that use motion compensation for interframe redundancy reduction provide a 1-dimensional VLC (variable length coding) scheme to separately encode the horizontal and vertical components of the differential motion vectors (DMVs). It can be derived, however, from information theory that coding the DMV components separately is less efficient than coding them jointly as a single entity. In this report, we propose a two-dimensional VLC scheme to code the differential motion vectors for videophone applications. Experimental results based on five commonly used video test sequences show that a 24% average saving of motion vector bits, as compared to the 1-D scheme in H.263, has been achieved without sacrificing picture quality at all.

MPEG1 video coding standard is an international standard primarily aimed at storage applications, and has been adopted in 1992. As a second work item by the ISO MPEG committee, MPEG2 standard is being considered for broader applications. In this paper, we investigate the advantages of the more complicated MPEG2 standard over the MPEG1. An overview on the technology of the standards and their major differences is given. Simulation results are presented to compare the performance of the two standards in terms of quality, quantization, bit count, and buffer content. It is shown that the MPEG2 yields better performance especially for highly interlaced video sequence.

The Motion Picture Experts Group (MPEG) video-coding algorithm is regarded as a promising coding algorithm for coding full-motion video. However, since MPEG was originally designed for storage applications, some problems must be solved before the algorithm can be applied to interactive services. Due to the use of periodic intraframe coding and bidirectional interframe prediction, the end-to end delay of the MPEG algorithm is much larger than that of the H.261 algorithm. In packet video transmission, the large peak in bit rate caused by periodic intraframe coding may lower performance of statistical multiplexing. In this paper, real-time video transmission using the Hybrid Extended MPEG (Bellcore's proposal to ISO/MPEG) is considered. First, the end-to-end delay of the Hybrid Extended MPEG algorithm is analyzed. Then several schemes to reduce the delay are considered and compared with regular coding schemes in terms of image quality, end-to-end delay and performance of statistical multiplexing. Error resilience of the presented schemes is also tested by simulations assuming cell loss. It is shown that the presented schemes improved the end-to-end delay and performance of statistical multiplexing significantly.>

Video coding is a key to successful visual communications. An interframe video coding algorithm using hybrid motion-compensated prediction and interpolation is considered for coding studio quality video at a bit rate of over 5 Mb/s. Interframe coding without a buffer control strategy usually results in variable bit rates. Although packet networks may be capable of handling variable bit rates, in some applications, a constant bit rate is more desirable either for a simpler network configuration or for channels with fixed bandwidth. A self-governing rate buffer control strategy that can automatically steer the coder to a pseudoconstant bit rate is considered. This self-governing rate buffer control strategy employs more progressive quantization parameters, and constrains quantizer adjustments so that a smoother quality transition can be attained. Simulation results illustrate the performance of the pseudoconstant bit rate coder with this buffer control strategy.

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.>

An adaptive transform coding algorithm using a quadtree-based variable blocksize DCT (discrete cosine transform) is introduced to achieve a better tradeoff between bit rate and image quality. The choice of appropriate blocksize is determined by a mean-based decision rule that can discriminate various image contents for better visual quality. Some simulation results are given. It is found that the same or better image quality can be obtained with lower average bit rate.>

Reconstruction from phase information via convex projections is considered for deblurring smeared images when the blurring function does not introduce any phase. The effects of initialization and constraints in the algorithm are investigated. It is shown by examples that the number of quantization levels when the smeared image is digitized determines the quality of the restored image.

A solution to the regularized signal restoration problem is formulated in the context of projections onto convex sets. An extended Wiener set is defined based on the Wiener solution in the frequency domain. A constrained Wiener solution is determined as a point in the intersection of the extended Wiener set and other deterministic constraint sets. The relationship between the proposed technique and the constrained Miller-Tikhonov regularization developed earlier is established.

Robust matched filtering is considered for multiple-input systems, with problems formulated in both frequency and time domains being given specific treatment. Robust solutions are found for bounded classes containing well-defined but not exactly specified input characteristics. Several properties for some special cases of interest are examined, and an example is given to illustrate the usefulness of the results. Finally, the results are applied to a narrowband spatial array system, for which the robust scheme has an interesting and useful characteristic.

Robust Wiener filtering has previously been considered for the single-input (scalar) case where there is no channel distortion and where the signal to be estimated is the source signal itself. Here, these results are extended to the multiple-input (vector) case where linear channel distortion is allowed and the signal to be estimated is a linear-filtered version of the source signal. The results are obtained from those for the single-input ease by modifying the constraints on signal and noise characteristics. Such a modification is motivated by examining the expression of the mean-squared error for the optimum filter.