The authors discuss the problem of mapping algorithms for real-time processing of digital video signals onto a fixed configuration of identical programmable video signal processors. Due to the periodic nature of the algorithms and the small periods that are involved, successive executions of the algorithm have to be interleaved in time. The resulting scheduling problem is mathematically modeled and examined. The authors present a novel solution approach that is based on a divide-and-conquer strategy using phase assignment as the central part. This approach has been implemented and it gives good results for industrially significant video applications. Specifically, the proposed approach has been implemented in only 1300 lines of C and has been applied to a number of problem instances, whose signal flow graphs originate from industrially relevant algorithms, including contour enhancement and progressive scan, noise reduction, 4:3 to 16:9 screen format conversion, and a very elaborate progressive scan algorithm.>

The architecture and programming aspects of a programmable video signal processor are discussed.The processor is an integrated circuit that has a modular architecture with a number of programmable, pipelined processing elements.Networks of these processors can be programmed conveniently with the aid of dedicated programming tools.In this paper the emphasis is on the scheduling of video algorithms and the micro code generation for a network of video signal processors.Due to the periodic nature of the video algorithms and the small periods that are involved, successive executions of the video algorithm have to be interleaved in time.We present a novel solution approach to the scheduling problem using phase assignment as the central part.Results of this approach are presented for industrially significant video applications.

In this paper, we introduce a new approach to design smearing and desmearing filters for data transmission systems to suppress pulse-like disturbances inserted in the communication channel. Our method is based on time domain considerations. It consists of the optimization of two merit factors defined for the impulse response functions of the filters. The first one is a measure of the smearing efficiency, and the second one is a measure of the amount of intersymbol interference caused by the insertion of the two filters in the transmission system. Our method will be applied to smearing and desmearing filters for baseband signals as well as for passband signals. In particular, we show that our method leads to filters having a better smearing efficiency and a simpler implementation than the filters obtained by the classical frequency domain approach.

In this paper a digital transmission unit for the forthcoming integrated services digital network (ISDN) is considered. Basic elements of such a unit are an adaptive echo canceller (EC) and an adaptive decision feedback equalizer (DFE). Each of these filters can be realized by a transversal filter structure or by a lookup table structure. An analysis and comparison of the behavior of these two structures is given, The combination of an EC and a DFE can also be realized by a single lookup table. It is shown that irrespective of the particular realization three different convergence phases can be distinguished, and that the overall convergence time is considerably longer then would be expected from the usual simple theory that ignores error propagation in the DFE. Two variants of a hardware realization of such a digital transmission unit are presented. They provide 144 kbit/s full-duplex transmission on the symmetrical cables of the existing local telephone network. A small transmission bandwidth is obtained by applying NRZ signaling. In the first circuit the EC and DFE are implemented by individual lookup tables, while in the second a combined table is used. Both circuits are to a large extent, implemented digitally. Due to the relatively low sampling frequency and the acceptably low power consumption they are very well suited for large scale integration with present day technology.

This paper describes the application of a commercially available microprocessor (Intel 3000 or Signetics 3000) to a flexible data transmitter and data receiver for high-speed data modems. For the transmitter a quadrature modulation scheme is chosen; the receiver is based on phase-shift compensation techniques and coherent demodulation with an externally derived digital carrier. For the realization with the given microprocessor it has been necessary to adapt the way of executing the various operations (especially the multiplications for the digital filtering) to the available computational capabilities. The resulting microprocessor implementations are also suitable for application in the current medium-speed synchronous data transmission systems.

A digital filter is introduced consisting of a transversal part and a simple recursive network. The coefficients of the transversal part are equal to integer powers of two or zero; thus complicated multipliers are avoided and instead a simple routing circuit is used. Use of several types of the recursive network makes the filter applicable in different frequency ranges. The coefficients of the transversal part can be interpreted as differences of successive values of the impulse response of the filter. It is shown that this difference routing digital filter (DRDF) is especially suited for application in data transmission.

A frequency domain analysis of the digital generation of linearly modulated data waveforms is presented. It is shown that for any rational relation of line carrier frequency and digit frequency, digital data transmitters can be designed economically and adapted to large scale integration (LSI) techniques. Special attention is given to the elimination of digital multipliers throughout the system. The results are illustrated with an example of an all-digital vestigialsideband (VSB) transmitter, with a specially shaped output spectrum which facilitates carrier and clock recovery in the receiver.

CO-modulation is a new type of modulation for the transmission of telegraphic signals. Its primary object is to increase the speed of transmission in a given bandwidth. The essential features of CO-modulation are: 1. The lower frequencies in the information signals are suppressed by means of a high-pass filter before modulation on a carrier. 2. Two channels are provided in the same frequency band by using two carrier-voltages differing90^oin phase. 3. The frequency components which are lacking are regenerated at the receiving end by means of positive feedback, thereby using a feedback network which is adapted to the high-pass filter mentioned in 1. The spectrum in the neighbourhood of the carrier frequency does not contain frequency components arising from the information signals. So a pilot signal of carrier frequency can easily be transmitted for phase-reference, thus allowing coherent detection of the signals transmitted by orthogonal modulation. In the demodulated signals the lower frequencies are missing. It is possible, however, to regenerate the low-frequency components suppressed at the sending end, because the related time-constants are well-known and independent of the transmission path. If these complementary signals are derived at the output and added to the received signals, then the original pulse pattern can be reconstructed. This procedure involves positive feedback which, for telegraphic signals, can easily be made stable. The name CO-modulation stands for complementary orthogonal modulation, thereby indicating the complementary action of suppressing and regenerating frequency components at the sending and receiving end respectively, and the orthogonal modulation by which two channels are supplied in the same frequency band. CO-modulation may be applied to synchronous as well as asynchronous telegraphy. In an experimental system the bandwidth from 700 c/s to 3100 c/s of a normal carrier-telephone channel was used for transmission of 4000 bauds. This corresp0nds to a relative speed of transmission of 1.67 bauds/Hz, which is not far from the theoretical maximum of 2 bauds/Hz as given by the sampling theorem.