Digital time signal filtering method and device for transmission

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3647241, 36472412, 36472418, 348614, 379406, 379410, G06F 1710, H04M 100, H04M 900, H04M 908

Patent

active

057681651

DESCRIPTION:

BRIEF SUMMARY
BACKGROUND OF THE INVENTION

The invention relates to a process for digitally filtering time signals such as those received from a transmission channel or a broadcasting network.
It also relates to a digital filtering device for implementing the process.
The invention is particularly advantageous to realize echo correction at the level of a receiver connected to a transmission system such as a television network.
Generally, digital filtering relates to a digital time signal, that is, one which results from a sampling at the rate of a given sampling period T generally imposed in accordance with requirements that the processing allow for a determined bandwidth, for example. Thus a video signal in conformity with the D2-MAC standard, for example, is sampled for a period of 49.38 nanoseconds.
Digital filtering of such a signal can be performed in the frequency domain, that is, from the discrete Fourier transform of the digital time signal. In this case, the filtering is defined by a transfer function which is also sampled at a given frequency sampling interval df. This transfer function, then is the sampled Fourier transform of the time impulse response of the filter to be produced.
Mathematically, the filtering consists of calculating, at each sampling period T, the convolution product of the time signal and the impulse response of the filter. In the time domain, the result of the convolution produces the filtered signal directly. If the filter is defined by its transfer function in the frequency domain, the convolution is reduced to a simple product (in the complex space) of the discrete Fourier transform of the time signal to be filtered by the transfer function of the filter for each of the frequency values for which the transfer function is defined. The result of this complex product furnishes the discrete Fourier transform of the filtered signal. An inverse discrete Fourier transform makes it possible to return to the time domain.
In order for this filter to be achievable in practice, the number of samplings in the time and frequency domains is obviously limited. This results in conditions and limitations relative to the ranges of definition for time and frequency quantities. Thus, in conformity with Shannon's theorem, the time sampling at the period T imposes a limitation on the bandwidth of the signal which can be filtered. This first condition can be expressed by saying that the sampling frequency 1/T of a real time signal must be more than double its bandwidth DF. The sampling in the frequency domain also involves a second condition: the time signal to be filtered is only taken into account during a limited time interval DT of a duration at most equal to the inverse 1/df of the sampling interval df in the frequency domain. The maximum interval 1/df thus defines a time window which will henceforth be called the "time horizon" of the filter.
The first of the preceding conditions is met by limiting the bandwidth of the time signal by means of filtering and/or by choosing a sampling frequency 1/T which is high enough. The second condition will also be met if the time signal can be represented by means of a limited number of samplings processed by the filter at each sampling period.
If the discrete Fourier transform mentioned above is used, a limited number N of samplings in the time domain is used to calculate the same number N of samplings in the frequency domain. The two preceding conditions can therefore be summed up by the following relations:
A time signal filtering operation by means of a digital filter H is summarized in theory in FIG. 1, which shows the operations performed in the time and frequency domains as well as the correspondences between one domain and the other. The upper part of the figure shows the various time parameters involved in the filtering, while the lower part shows the corresponding parameters (by Fourier transform) in the frequency domain.
Thus, the time signal e(t) to be filtered (assumed to be continuous) is sampled at the period T and converted into a digital quantity by

REFERENCES:
patent: 4701875 (1987-10-01), Konishi et al.
patent: 4797847 (1989-01-01), Duhamel
patent: 4951269 (1990-08-01), Amano et al.
patent: 5276516 (1994-01-01), Bramley
patent: 5473555 (1995-12-01), Potter
Proceedings: ICASSP 87, International Conference on Acoustics, Speech, and Signal Processing Apr. 6, 7, 8, 9, 1987 Registry Hotel, Dallas, Texas vol. 4 of 4.

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