Digital filter sharpening aims to improve filtering by cascading structures employing multiple use of an original prototype's unchanged coefficient set. Starting with a visitor-designed FIR prototype, we demonstrate the operation of three different sharpening methods, comparing their various advantages and shortcomings. Two tools are employed: a Simulink model, facilitating visual examination of virtual instrument displays, and a MATLAB package that provides a compact equivalent filter model of the sharpened resultant.

Filtering signals sampled on a grid which is nonuniformly distributed in the time domain is not a simple task since the filter's coefficients have to be time varying. They must be updated at each sampling instant. The filtering becomes even more complicated when it has to be optimal (or at least suboptimal) in the sense of a certain design criterion. In this paper we present an effective algorithm for FIR filtering aiming at minimisation of the energy of the filtering error signal. The approach provides a solution which resembles the weighted least squares design method for FIR filters of uniformly sampled signals.

Windowing is an attractive avenue for creating asymmetric FIR impulse response sequences for use as fractional-delay elements. However, traditional (symmetric) windows are not usable, leading to a need for offsetting and purpose-optimization of the windows' defining parameters. It is shown that as few as three terms of such a modified raised cosine window deliver enormous improvements over simple sinc fractional-sample delays.

Non-optimal FIR filters used for fractional-sample delay, despite their wideband nature, are shown to benefit significantly from application of windowing. Here simple raised-cosine windows prove to be very effective, particularly if they are cast as asymmetric modifications of their conventional forms. The offset von Hann window is surprisingly potent when the number of coefficients is large and window offset coincides with the fractional delay required of the overall filter.

Four main FIR filter types are examined for applicability to the task of fractional-sample delay filtering. Since a balance is sought between the small magnitude of the complex approximation error and rapid re-design for employment in variable-delay situations, techniques with closed-form design formulae have been favoured. Size of fractional delay and oddness or evenness of the filter exhibit simple, but fundamentally important interplay, and an irreducible error at the Nyquist frequency always prevails. Guidelines for choosing between these closed-form candidate filters are suggested. The popular sinc delayer is shown to be a particularly poor performer.>

In this paper we demonstrate application of the fractional sample delayor (FSD) to constructing sampling rate converters for incommensurate sampling rates. We use Lagrange polynomials for interpolating the values of the signal between sampling instants. It is shown that to deploy this technique effectively one has to use polynomials of high degree. Otherwise the interpolation error for high frequencies (above half Nyquist) is very big. Increasing the polynomial's degree leads inevitably to numerical problems. The interpolation process may be improved when additional samples of the signal are generated before interpolation. It is shown that FSD can be used for achieving this task. The final version of the sampling rate converter comprises two FSDs and a very simple four-point interpolator based on a Lagrange polynomial.>

A MININET "station" acts as a user concentrator into the MININET word-switched network where each word (of up to 16 bits) entering a network "port" is mapped into a single packet for immediate transmission through the network. The stations maintain highly transparent "virtual connections" between user devices. Among the important requirements for MININET are that, under saturation conditions, the communication resources of the network be shared fairly between the user devices and that the network possesses very short end-to-end propagation delays. The design of a station meeting these requirements, capable of processing over one million packets per second, is described. Its architecture is based on special-purpose processors which are functionally distributed. The biggest design challenge was to minimize the processing delay in the node while avoiding potential blockages and remaining fair. The final solution has been to use a master arbiter, which looks at the readiness of the ports and channels in conjunction with connection, routing, and flow control information. The station management entities reside in a general-purpose microcomputer. These monitor and modify the operation of the high-speed processors, establish virtual connections, and support the routing algorithm.

In this paper we review the acoustical and statistical principles underlying the operation of a two-input signal processing speech dereverberation technique (J.Allen et al., 1977) and describe modifications to this technique based upon experimental and analytical studies of this process. These experimental modifications include selectable short-time binaural coherence estimation techniques, frequency domain smoothing of coherence measures on a critical-band basis, and suppression rules based on thresholded coherence estimates. The effect of these modifications and choices of processing parameters will be discussed in relation to changes in spectrograms and the quality and intelligibility of processed signals.

This paper describes a control protocol for high-speed transfers between transceivers in a data communication system. One particularly suitable application of current topical interest would be to control lines linking nodes in a local area packet-switching system. The protocol applies for fixed-length data packets which must ultimately be received at their destination in sequence. With these specifications, a set of rules which is easy to implement in practice is established. The result is a packet interlock method of numbering packet transmissions and jumping back to retransmit a series of packets under error conditions. Problems associated with multiple errors, dummy packets, initial synchronization and resynchronization are investigated and found to be readily overcome by the proposed protocol.