G.D. Cain
University of Westminster
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Featured researches published by G.D. Cain.
international conference on acoustics, speech, and signal processing | 1994
G.D. Cain; N.P. Murphy; Andrzej Tarczynski
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.<<ETX>>
international conference on acoustics, speech, and signal processing | 1995
G.D. Cain; A. Yardim; P. Henry
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.
IEEE Transactions on Instrumentation and Measurement | 1996
Hanwu Sun; Gregory H. Allen; G.D. Cain
In this paper a new adaptive filter-bank structure suitable for the accurate harmonic measurement in power supply systems is presented. This adaptive measurement technique is based on the resonator-in-a-loop filter-bank structure and includes a new modified Gauss-Newton gradient algorithm. Using this filter-bank structure and a least-squared-error approximation method, simulation results are presented to establish the performance of the technique. The effects of time-varying fundamental frequency are also investigated using this filter-bank structure. Simulation results show that the fundamental frequency can be tracked with very small error. The technique is also successfully applied to a measured voltage signal sampled from the mains to demonstrate performance.
Signal Processing | 1998
Ewa Hermanowicz; Miroslaw Rojewski; G.D. Cain; Andrzej Tarczynski
Abstract A novel family of special discrete-time filters having fractional delay is proposed, where a fractional-sample delay (FSD) filter serves as a versatile building block from which all other special filters of the family are derived. The specific feature of these filters is that the fractional delay is a parameter which can be varied. An illustrative example shows the performance of a specimen filter whose coefficients are obtained from the impulse response of an FIR FSD filter using polynomial approximation. The application of this filter to simultaneous instantaneous frequency estimation and fractional delay of a complex FM signal is also presented.
international conference on acoustics, speech, and signal processing | 1997
Andrzej Tarczynski; Vesa Välimäki; G.D. Cain
Filtering signals sampled on a grid which is nonuniformly distributed in the time domain is not a simple task since the filters 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.
midwest symposium on circuits and systems | 1995
A. Tarczynski; G.D. Cain
The paper presents a new algorithm for designing digital filters approximating a given frequency response. The method combines and modifies known techniques which are utilised in digital filter design such as: Levys linearisation, minimisation of weighted squared error, or Lawsons algorithm. The ultimate goal of the proposed approach is to obtain a design algorithm for filters minimising the Chebyshev norm of the frequency response error. The proposed algorithm is illustrated by sample designs.
midwest symposium on circuits and systems | 1995
Artur Krukowski; G.D. Cain; Izzet Kale
This paper addresses high-order frequency transformation and extends standard Constantinides, Mullis/Franchitti transformations and the exact N-point approach we have previously reported. Using specially designed M/sup th/ order allpass filters, mapping of N features of the prototype filter can be carried out independently (where N<M). The mapping filter is designed taking special care to ensure that there are no excess mapping replicas which may arise if an exact N-point transformation is used. This design viewpoint represents a significant departure from previous approaches to filter transformations.
Sensor Review | 1997
Ivars Bilinskis; G.D. Cain
Addresses the problem of full digital processing of sensor signals at frequencies in the microwave and radio frequency range. Discusses advantages and drawbacks of the emerging digital alias‐free signal processing technology considering it as a new DSP tool prospective for achieving a breakthrough in DSP theory and techniques leading to a stepwise enlarging of the DSP application frequency range.
international conference on acoustics, speech, and signal processing | 1993
G.D. Cain; A. Yardim; Richard C. S. Morling
DSP deserves to be a major focus of professional engineering education, with ready access points provided at virtually all stages of undergraduate, postgraduate, and continuing professional development study. The authors describe one institutions commitment to this concept, where entering undergraduates experience an electronic engineering curriculum designed around a 15% substrate of DSP, with an abundance of supporting project work available in two years of the program. A named DSP masters course is an important bridge to industry and permits simultaneous access to short courses serving both full-time postgraduates and industry-based engineers. A satellite outreach linking a 15-site network of leading European DSP groups via live, interactive tele-seminars is also described.<<ETX>>
IEE Proceedings - Vision, Image, and Signal Processing | 1994
Izzet Kale; J. Gryka; G.D. Cain; B. Beliczynski