Control system using harmonic filters

Miscellaneous active electrical nonlinear devices – circuits – and – Specific signal discriminating without subsequent control – By frequency

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327 42, 327 46, 327 48, 327555, 327557, H04B 110

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054690870

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BRIEF SUMMARY
INTRODUCTION

The invention relates to a harmonic filter which is a signal processing means for obtaining the complex amplitude of a single harmonic component from a signal which contains one or more harmonic components. The filter can be used in active or adaptive control systems for attenuating disturbances.
The idea of separating a reference signal into separate, fixed frequency bands and filtering each band independently is well known and was first used in attempts to control transformer noise. Examples of this are described in U.S. Pat. No. 2,776,020 to W. B. Conover et al and in the article to K. Kido and S. Onoda, "Automatic Control of Acoustic Noise Emitted from Power Transformer by Synthesizing Directivity". Science Reports of the Research Institutes, Tohoku University (RITU), Japan. Series B: Technology. Part 1: Reports of the Institute of Electrical Communication (RIEC), Vol 23, 97-110.
The general idea has been generalized for use with broadband signals and digital systems as noted in U.S. Pat. No. 4,423,289 to M. A. Swinbanks and by I. D. McNicol in "Adaptive Cancellation of Sound in Ducts", M. Eng. Sci. Thesis, Dept. Electrical and Electronic Engineering, University of Adelaide, Australia, (1985).
These systems are all feedforward systems, since the controller output is obtained by filtering a reference signal. These systems use narrow band filters to obtain the real, time varying signal at fixed frequencies. Other approaches, also described in McNicol, use Fourier transform techniques to obtain the complex components of the reference signal and the residual signal at fixed frequencies.
An extension to this approach, which allows for varying frequencies in the disturbance, is to use a timing signal from the source of the disturbance to trigger a sampling device. This results in an exact number of samples in each cycle of the noise, and the harmonic components can then be obtained by a Discrete or Fast Fourier Transform as described in U.S. Pat. No. 4,490,841 to G. B. B. Chaplin et al. With the approach described both input and output processes are synchronized with the timing signal. In the special case of fixed frequencies this approach is equivalent to a feedforward system.
The approaches differ in the way the controller output is obtained and adjusted. In one approach the output is generated by filtering reference signals. The amplitude and phase of each signal is adjusted in the time domain by a variable filter as in Swinbanks, while in the other approach the controller output is updated in the frequency domain using the Discrete Fourier Transform of the residual signal as in Chaplin for varying frequencies, and for fixed frequencies in "Adaptive Filtering in the Frequency Domain" by Dentino et al, IEEE Proceedings, Vol 69, No. 12, pages 474-75 (1978).
The first approach can be implemented digitally by using a frequency sampling filter followed by a two-coefficient FIR filter or by using a frequency sampling filter followed by a Hilbert transformer and two single coefficient filters.
These techniques are described in the aforementioned Dentino reference and in "Adaptive Frequency Sampling Filters" by R. R. Bitmead and B. Anderson, IEEE Transactions, ASSP, Vol. 29, No. 3, pages 684-93 (1981).
For application to active control the standard update methods must be modified to account for the response of the physical system. This can be done by filtering the reference signal through a model of the physical system. An example of this is the "filtered-x LMS" algorithm described in Adaptive Signal Processing by B. Widrow and S. D. Stearns, Prentice Hall (1985). Similarly, this approach has been used for periodic noise, see "A Multichannel Adaptive Algorithm for the Active Control of Start-Up Transients", by S. J. Elliot and I. M. Stothers, Proceedings of Euromech 213, Marseilles (1986). Nelson and Elliot generate reference signals for each harmonic and then filter these signals through two coefficient filters. These parallel filters, one for each harmonic, are then adapted using the filtered-x LM

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patent: 4328591 (1982-05-01), Baghdady
patent: 4489280 (1984-12-01), Bennet, Jr. et al.
patent: 4513249 (1985-04-01), Baghdady

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