Electrical computers: arithmetic processing and calculating – Electrical digital calculating computer – Particular function performed
Reexamination Certificate
2001-01-02
2002-09-03
Malzahn, David H. (Department: 2124)
Electrical computers: arithmetic processing and calculating
Electrical digital calculating computer
Particular function performed
Reexamination Certificate
active
06446103
ABSTRACT:
BACKGROUND OF THE INVENTION
Field of the Invention
The present invention relates to a nonrecursive digital filter and a method for calculating the coefficients of the filter.
Numerous products contain filters (e.g. tone controls of a television set or car radio, frequency splitters of loud-speaker systems) whose characteristic must be shiftable in the frequency direction.
If these filters are realized by digital signal processing, then recursive filters are usually used despite a number of disadvantages. This filter type is characterized in that it has only few parameters (coefficients) which have to be varied in order to scale the frequency response of the filter (e.g. in order to vary the cut off frequency) without having to vary the sampling frequency of the filter.
“Zeitdiskrete Signalverarbeitung” [Discrete time signal processing] by Oppenheim and Schafer, Oldenburg Verlag, Munich, 1992, page 152 et seq. refers to the possibility of shifting the frequency response of a digital filter by varying the sampling frequency. However, this approach generally requires anti-aliassing filters or decimation filters with a variable cut-off frequency, i.e. the problem is merely shifted.
In this context, European published patent application EP 0 401 562 discloses a nonrecursive digital filter having a filter calculation unit which calculates an output signal from an input signal. The filter calculation unit is connected to a coefficient calculation unit, which supplies the coefficients for the filter calculation unit, and the coefficient calculation unit is connected to a coefficient memory, an operator interface, and a control unit.
Furthermore, the paper by Sangil Park, “REAL TIME PITCH (FREQUENCY) SHIFTING TECHNIQUES” in Proceedings of the International Conference on Consumer Electronics (ICCE), New York, USA, IEEE, Vol. Conf. 101 Jun. 1991, pages 296-97 disclosed a method for shifting the frequency/pitch of a signal in real time, in which, in the method, the signal is sampled at a second sampling rate, whose ratio with respect to the original sampling rate is chosen to correspond to the ratio of original pitch (frequency) to desired pitch (frequency), and in which the output samples are once more output at the original sampling rate.
SUMMARY OF THE INVENTION
The object of the present invention is to provide a nonrecursive digital filter which overcomes the above-noted deficiencies and disadvantages of the prior art devices and methods of this general kind, and the frequency response of which can be shifted in a simple manner at a constant sampling frequency. To that end, it is also an object of the present invention to provide the required calculation method for the coefficients of this filter.
With the above and other objects in view there is provided, in accordance with the invention, a method of calculating coefficients of a nonrecursive digital filter for shifting a frequency response of the filter from one characteristic frequency fc
1
to another characteristic frequency fc
2
The method comprises the following method steps: determining (i.e., prescribing or calculating) coefficients h1 (n) of a prototype of the filter for a given sampling frequency fa
1
;
calculating a continuous time impulse response h(t) associated with the prototype;
sampling the impulse response h(t) at a sampling frequency
fa2
=
fa1
*
fc1
fc2
,
and thereby producing new coefficients h2 (n); and
operating the filter with the new coefficient h2 (n) at fa
1
.
With the above and other objects in view there is also provided, in accordance with the invention, a further method of calculating the coefficients of a nonrecursive digital filter for the purpose of shifting the frequency response of this filter from one characteristic frequency fc
1
to another characteristic frequency fc
2
The novel method has the following method steps:
determining coefficients h1 (n) of a prototype of the filter for a given sampling frequency fa
1
;
calculating new coefficients h2 (n) from the coefficients h1 (n) by interpolating the values of a continuous time impulse response h(t) at an interval 1/fa
2
, where
fa2
=
fa1
fc1
fc2
;
and
operating the filter with the new coefficient h2 (n) at fa
1
.
In accordance with an added feature of the invention, polynomials are used for the interpolation. Preferably, only polynomials of order 0 to 3 are used.
In accordance with an additional feature of the invention, the continuous time filter is sampled starting from a temporal midpoint.
In accordance with another feature of the invention, the filter prototype is defined with N coefficients h1 (n), where n=0,1, . . . , N−1, and the interpolation is performed at the points
t
=
N
-
1
2
*
fa1
+
k
fa2
where
k=0;+/−1;+/−2, . . .
k being incremented or decremented until all the coefficients h1 (n) used for the respective interpolation have the value 0.
In accordance with a further feature of the invention, only the coefficients where k=0; +1; +2 . . . are calculated since the coefficients with negative k correspond to the corresponding coefficients with positive k.
In accordance with again an added feature of the invention, the new coefficients h2 (n) are multiplied by the samples of a sine or cosine function having a frequency fo.
It is known from systems theory (Fourier transformation) that shifting of the frequency response of a filter and opposite shifting of the impulse response of the filter correspond to one another. The coefficients of a digital nonrecursive filter (prototype of the frequency characteristic) represent samples of such an impulse response. In principle, the continuous time profile of the impulse response can then be calculated from the samples, with a sampling frequency fa
1
being prescribed. Coefficients of a nonrecursive digital filter are again obtained by sampling the continuous time signal at another sampling frequency fa
2
. If this filter is then operated at the sampling frequency fa
1
, the desired shifted frequency characteristic is obtained. If fa
2
is chosen to be greater than fa
1
, then the frequency characteristic of the filter is compressed, and for fa
2
less than fa
1
the characteristic is expanded. In this case, the actual problem consists in the interpolation of the new coefficients from the coefficients of the prototype. This interpolation corresponds to the combination of reconstruction of the continuous time impulse response and sampling thereof at fa
2
. The realization details are explained in detail below:
Let H(f) be the Fourier transform of the impulse response h(t) of a linear system, in short: h(t)
H(f). The following holds true in that case: h(a*t) 1/a*H(f/a), i.e. if the impulse response h(t) is expanded (a<1), then the transform H(f) is compressed, and vice versa. In addition, the transform H(f) is “amplified” by the factor 1/a.
In principle, this property is also preserved after the sampling of the impulse response, as long as the sampling theorem is not violated. If the theorem is violated, care must be taken to ensure that the desired frequency characteristic H(f/a) is not varied impermissibly.
Assume that a digital filter is operated at the sampling frequency fa
1
and then has the characteristic frequency (e.g. 3 dB—cut off frequency of a low pass filter) fc
1
. If the same filter is then operated at the sampling frequency fa
2
, the new characteristic frequency fc
1
=fc
1
*(fa
2
/fa
1
) is produced.
The calculation of the coefficients of a new nonrecursive filter from the coefficients of a prescribed nonrecursive filter (called prototype hereinafter) with the aim of shifting the characteristic frequency from fc
1
to fc
2
is done as follows:
Step 1: Calculate coefficients h1 (n) of the prototype for the sampling frequency fa
1
.
Step 2: Calculate the associated continuous time impulse response h(t)
Step 3: Sampling of the impulse response h(t) at the sampling frequency fa
2
=(fc
1
/fc
2
)*fa
1
.
Step 4: operation of the new filter with the coefficients h2 (n) at fa
1
.
In order to be able
Greenberg Laurence A.
Infineon - Technologies AG
Malzahn David H.
Mayback Gregory L.
Stemer Werner H.
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