Pulse or digital communications – Equalizers – Automatic
Patent
1994-12-19
1996-07-23
Chin, Stephen
Pulse or digital communications
Equalizers
Automatic
375229, 375350, 370 321, 36472419, H03H 730
Patent
active
055397735
DESCRIPTION:
BRIEF SUMMARY
The present invention relates to a method and to an apparatus for ghost cancelling and/or equalizing.
BACKGROUND
In "A tutorial on ghost cancelling", W. Ciciora et al, IEEE Transactions on Consumer Electronics, Vol. 25, pages 9-44, 25. February 1979, the principle of ghost cancelling is described. A received test signal is compared with an ideal version of this test signal. The coefficients of a correction filter applied to the received test signal are adjusted in such a way that the filter output signal matches as much as possible to the ideal test signal. But the filter length is limited and in a lot of cases no sufficient ghost cancellation is possible.
In "Adaptive Noise Cancelling: Principles and Applications", B. Widrow et al, Proceedings of the IEEE, pages 1692-1716, Vol. 63, No. 12, December 1975, an LMS (least-mean-square) adaptive filter for noise cancelling is introduced, wherein a new filter coefficient w.sub.i(j+1) is calculated according to the formula adaptation constant, .epsilon..sub.j is the difference between the respective filter output y.sub.j and the corresponding value of the ideal reference signal and x.sub.ij is the respective value from the filter input signal.
INVENTION
It is one object of the invention to disclose a method for ghost cancelling and/or equalizing, where also more complex distortions and ghost signals, respectively, can be handled and where the speed and stability of adaptation is improved. This object is reached by the inventive method disclosed in claim 1.
A first long LMS adaptive filter which may have more than 1000 taps minimizes the error .epsilon. between the filtered output signal y of the distorted input signal x (a transmitted reference signal) and an ideal reference signal. Normally this ideal reference signal is stored within the ghost canceller/equalizer and corresponds to the reference signal which is transmitted e.g. in a specific TV line. A positioner takes coefficients from the first LMS filter and determines the most useful subset of (say 5) small filters (say 16 taps each) separated by variable delays. The position information calculated in the positioner comprises the lengths of these delays.
The subset of filters forms together with the variable delays a second LMS adaptive filter with a tap number corresponding to the first LMS filter tap number. The coefficients corresponding to the delay `taps` can be regarded as set to zero and the locations of the delay parts of the second LMS filter remain fixed. The coefficients of the filter parts (corresponding to the small filters) are modified adaptively until the best match between the filtered received reference signal and the ideal reference signal is achieved. This process will recommence if the ghost or the received channel changes.
The first and second LMS filter need not work in real time. After establishing of the filter coefficient modification of the subsets these coefficients are copied to a real time main filter working on the whole input signal (e.g. a TV signal) with exception of said reference signal.
The main filter comprises a respective subset of adjustable small filters and a set of adjustable delays. In the following a modified equation how the error is defined.
In principle the inventive method consists in ghost cancelling and/or equalizing, wherein the coefficients of a digital filter are in principle continuously adapted to minimize the error between the filter output signal of a distorted reference signal and an ideal reference signal which is stored or which can be decoded correctly after reception, and wherein said digital filter is a first LMS filter or filter function with a high tap number and the coefficients of this filter or filter function are used by comparing stepwise values derived from such coefficients of said first LMS filter which represent the tap positions of a small filter with a reduced number of coefficients to detect depending on the level of said values the position of one or more variable delays which connect in series two or more of said small fi
REFERENCES:
patent: 5045945 (1991-09-01), Herman et al.
patent: 5111481 (1992-05-01), Chen et al.
patent: 5161017 (1992-11-01), Sato
1988 IEEE Int. Symp. on CAS; 7-9 Jun. 1988 ESPOO FI; pp. 1931-1934; S. Poole: "A CMOS Subscriber Line Audio Processing Circuit Including Adaptive Balance".
IEEE Acoustics, Speech and Signal Processing Magazine, vol. 34, No. 2, Apr. 1986, New York US, pp. 309-316; R. W. Harris "A Variable Step (VS) Adaptive Filter Algorithm".
Bruneau Nelly
Hackett Andrew
Knee Michael
Zipp Andreas
Chin Stephen
Ghebretinsae T.
Herrmann Eric P.
Thomson Consumer Electronics S.A.
Tripoli Joseph S.
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