Article Free Access Share on A real-time scalable software video codec for collaborative applications over packet networks Authors: J. Hartung Multimedia Communications Research Laboratory, Lucent Technologies, Bell Laboratories 600 Mountain Avenue Murray Hill, NJ Multimedia Communications Research Laboratory, Lucent Technologies, Bell Laboratories 600 Mountain Avenue Murray Hill, NJView Profile , A. Jacquin Multimedia Communications Research Laboratory, Lucent Technologies, Bell Laboratories 600 Mountain Avenue Murray Hill, NJ Multimedia Communications Research Laboratory, Lucent Technologies, Bell Laboratories 600 Mountain Avenue Murray Hill, NJView Profile , J. Pawlyk Multimedia Communications Research Laboratory, Lucent Technologies, Bell Laboratories 600 Mountain Avenue Murray Hill, NJ Multimedia Communications Research Laboratory, Lucent Technologies, Bell Laboratories 600 Mountain Avenue Murray Hill, NJView Profile , K. L. Shipley Multimedia Communications Research Laboratory, Lucent Technologies, Bell Laboratories 600 Mountain Avenue Murray Hill, NJ Multimedia Communications Research Laboratory, Lucent Technologies, Bell Laboratories 600 Mountain Avenue Murray Hill, NJView Profile Authors Info & Claims MULTIMEDIA '98: Proceedings of the sixth ACM international conference on MultimediaSeptember 1998 Pages 419–426https://doi.org/10.1145/290747.290814Published:01 September 1998Publication History 8citation352DownloadsMetricsTotal Citations8Total Downloads352Last 12 Months3Last 6 weeks2 Get Citation AlertsNew Citation Alert added!This alert has been successfully added and will be sent to:You will be notified whenever a record that you have chosen has been cited.To manage your alert preferences, click on the button below.Manage my AlertsNew Citation Alert!Please log in to your account Save to BinderSave to BinderCreate a New BinderNameCancelCreateExport CitationPublisher SiteeReaderPDF

This paper describes a prototype video coding platform meant for the conception and testing of multimedia products such as next-generation videophones. The platform is largely based on ITU-T Recommendation H.263, with a number of additional object-oriented quality enhancement features which make it especially well suited for very low bit-rate coding of "head-and-shoulders" video material typical of real-time multimedia applications, video teleconferencing, and video telephony. These features consist of: (1) segmentation into objects of interest, (2) segmentation-based prefiltering, (3) model-assisted rate control, (4) adaptive vector quantization, and finally (5) segmentation-based postfiltering. In the spirit of Recommendation H.263, these enhancements are modular and can be selectively turned on or off, thereby enabling a wide variety of coding modes.

Proposes a codebook adaptation algorithm for very low bit rate, real-time video coding. Although adaptive codebook design has been studied in the past, its implementation at very low coding rates suitable for the MPEG4 standard remains significantly challenging. The coder uses a standard motion compensated predictor with DCT quantization. It is unique in that it uses a hybrid scalar/vector quantizer to code predictor residuals. Bits are dynamically allocated to minimize distortion in the current frame, and scalar quantized blocks are used to adapt the VQ codebook. A codebook adaptation algorithm is described which uses an "equidistortion principle" and a competitive learning algorithm to continuously adapt the codewords. This training algorithm results in an increased use of the more efficient vector quantizer and improved video quality.

A multiprocessor architecture for the real-time implementation of video coding algorithms is described. This MIMD architecture provides an efficient I/O structure for processing video at various resolutions, and has a distributed frame store and I/O structure suited for the implementation of three-dimensional subband coding. A high-quality 384 kb/s coder has been implemented using the video array processor (VAP) as a research platform for developing the coding algorithm.>

The authors describe two DTMF (dual-tone multiple frequency) detection algorithms that are highly efficient in the use of both real time and memory. The first algorithm is based on linear prediction (LP) and can be used to implement up to 32 DTMF detectors on a single 25-MHz NMOS WE DSP32. Using a 50-MHZ WE CMOS DSP32C, up to 45 detectors can be implemented. The second algorithm is based on a slight modification of the Goertzel algorithm and can be used to implement up to 16 DTMF detectors on a single DSP32 and up to 32 detectors on a DSP32C. In each of these implementations no external memory is used. Thus, for DSP32 implementations, the small 40-pin DIP package can be used. While the LP based algorithm is the more efficient of the two, the modified-Goertzel-based detector performs better in the presence of speech.>

Digital signal processors (DSPs) have traditionally been used in real-time applications with very high data throughput. For this reason, system designers have been reluctant to accept the degradation in performance inherent in machine code compiled from high-level languages such as C. The problem is compounded by the fact that DSPs use pipelined architectures to achieve their high data throughput, resulting in hazards and latencies between instructions. Simple compiler implementation cannot take advantage of latent instructions, resulting in a conservative and inefficient executable program. This problem has been addressed in the C compiler package for the AT&T WE DSP32 by the addition of a postoptimizer and an extensive application library. The authors give an overview of the DSP32 family architecture, describe the operation of the basic compiler and optimization strategies, and provide an example of the use of the compiler.>

In 1986 CCITT (International Telegraph and Telephone Consultative Committee) Study Group XVII recommended an international standard for coding wideband speech and music (50-7000 Hz) at 64 Kb/s or less. Some applications envisioned for such a coder are in teleconferencing, commentary channels, and for use in high-grade telephones for ISDN (integrated-services digital network) applications. The authors describe a full-duplex implementation of this algorithm on a single AT&T WE DSP16 digital signal processor (DSP) running at a 55 ns instruction-cycle time. An AT&T T7521 codec is used to digitize the analog signal with 15-bit precision at a 16 kHz sample rate. The codec provides both antialiasing and reconstruction filters on-chip. The codec interfaces directly to the DSP16 serial port, and data are double-buffered in hardware, and transferred into on-chip RAM using the interrupt facility on the DSP device. The parallel port of the device is used for both controlling program execution and interfacing to a 192 kb/s serial bit-stream.>