This invention claims priority under 35 USC 119 of Australian Provisional Patent Application No. 2002953284 filed Dec. 12, 2002 entitled “Digital Multirate Filtering,” assigned to the assignee of the present invention, and incorporated herein by reference.
The present invention relates broadly to a method for determining filter coefficients for filter stages in a multirate digital filter, and to a multirate digital filter device.
For each of the stages associated with the FIR filter devices 16_1 to 16_n, a rate decimation by factor of 2 is performed prior to filtering by the respective FIR filter devices 16_1, . . . , 16_n. A rate multiplication by a factor of 2 also is performed in each of the stages on the output side, with the outputs being summed over the various stages to generate the summed output 18 of the multirate filter 10. The summers sum signals samples at the same rate.
In multirate filter designs such as the ones shown in
Each sample rate change is achieved, for example, by using a low-pass anti-alias filter.
There is a need in the art for a method of determining filter coefficients for the filter stages of a multirate filter, e.g., a multi-stage filter structure such as shown in
In accordance with a first aspect of the present invention, there is provided a method of determining filter coefficients for filter stages in a multirate digital filter device to achieve a desired filter response. The method includes the steps of:
(a) determining a first plurality of evenly spaced sample points representing the desired response function on a logarithmic time scale, such that the sample points of the first plurality have a geometrically increasing spacing when viewed in a linear time scale, and
(b) determining filter coefficients for each filter stage from an associated group of sample points out of the first plurality of sample points.
In one embodiment, the method includes, prior to step (a), determining a second plurality of sample points representing the desired response function on a logarithmic frequency scale, and deriving the sample points of the first plurality representing the desired response function in the logarithmic time scale from the sample points of the second plurality.
In one embodiment, the step of deriving the sample points of the first plurality from the sample points of the second plurality is further based on a desired phase response of the multirate digital filter device.
The step of deriving the sample points of the first plurality from the sample points of the second plurality may include deconvoluting the desired response function in the logarithmic frequency scale using a set of prototype filter response functions, and deriving the first plurality of sample points representing the desired response function in the logarithmic time scale from a summation of corresponding prototype filter response functions.
In one embodiment, the filter coefficients for each filter stage are determined such that a last tap in one stage is equal to a first tap in the next lower rate filter stage.
Step (b) may include, for each associated group of sample points out of the first plurality of sample points, applying a transform matrix to determine the filter coefficients of the associated filter stage.
Preferably, for at least some of the associated groups of sample points the same transformation matrix is applied to determine the filter coefficients of the respective associated filter stages.
Advantageously, the transformation matrices are based on a substantially inverse filter response characteristic analysis of the individual filter taps of the respective filter stages.
In accordance with a second aspect of the present invention there is provided a multirate digital filter device including
wherein the processor unit is arranged, in use, such that a response function representing the input desired filter response on a logarithmic frequency scale is transformed into a logarithmic time scale, a first plurality of sample points representing the response function in the logarithmic time scale is determined, such that the sample points of the first plurality have an increasing pitch when viewed in a linear time scale, and the filter coefficients for each filter stage are determined from an associated group of sample points out of the first plurality of sample points.
In one embodiment, the processor unit is further arranged, in use, to determine a second plurality of sample points representing the desired response function on a logarithmic frequency scale, and to derive the sample points of the first plurality representing the desired response function in the logarithmic time scale from the sample points of the second plurality.
The processor unit may further be arranged such that, in use, the deriving the sample points of the first plurality from the sample points of the second plurality is further based on a desired phase response of the multirate digital filter device.
In one embodiment, the processor unit is arranged such that, in use, the deriving the sample points of the first plurality from the sample points of the second plurality includes deconvoluting the desired response function in the logarithmic frequency scale using a set of prototype filter response functions, and to derive the first plurality of sample points representing the desired response function in the logarithmic time scale from a summation of corresponding prototype filter response functions.
The device is preferably arranged, such that, in use, filter coefficients for each filter stage are determined such that a last tap in one stage is equal to a first tap in the next lower rate filter stage.
In one embodiment, the processor unit is arranged such that, in use, the determining of the filter coefficients for each filter stage from an associated group of sample points out of the first plurality of sample points includes, for each associated group of sample points out of the first plurality of sample points, applying a transform matrix to determine the filter coefficients of the associated filter stage.
Preferably, the processor unit is arranged such that, in use, for at least some of the associated groups of sample points the same transformation matrix is applied to determine the filter coefficients of the respective associated filter stages.
Advantageously, the processor unit is arranged, in use, to base the transformation matrices on a substantially inverse filter response characteristic analysis of the individual filter taps of the respective filter stages.
In accordance with a third aspect of the present invention, there is provided a data storage medium having stored thereon computer readable data for instructing a computer to execute a method as defined in the first aspect.
Preferred embodiments of the present invention will now be described, by way of example only, with reference to the accompanying drawings.
The starting point in
The logarithmic frequency scale samples, e.g. 22 are initially “transformed” into logarithmic time scale samples, e.g. 34 of representing the (impulse) response function 30 along a logarithmic time scale 32, as illustrated in
In
It has been recognized by the applicant that the geometrically increasing spacing on the linear time scale of the samples, e.g. 34, make this sampling well suited for determining filter coefficients of a multirate digital filter device having a multi-stage architecture. As illustrated in
In the following, a conversion process will be described by way of an example, to convert a time-domain filter response with equally spaced time samples on a logarithmic time scale, e.g., a response such as that shown in
In one embodiment, to calculate the filter coefficients for the various filter stages 60_0 to 60_n, transformation matrices 62_0 to 62_n are applied to respective groups 64_0 to 64_n, each of associated consecutive sample points out of the sample points T0 to TN. Note as shown in
In one embodiment, the matrices 62_0 to 62_n are based on a pseudo inverse filter response characteristic analysis of the individual filter taps of the respective filter stages 60_0 to 60_n, which can be derived for a given multirate digital filter configuration.
It has been found by the applicant that in a typical multirate digital filter configuration such as the one illustrated in
As illustrated in
Turning to
In the following, details of a conversion process from the filter response on a logarithmic frequency scale (compare
Turning now to
As illustrated in
A=shift(A′, N−3)*W0
B=shift(B′, N−3)*W1
Ci=shift(C′, N−2−i)*Wi(i=1 . . . N−2)
D=D′*WN
A summation of the fitted prototype response functions yields a fit to the overall desired response function of the multirate digital filter device. For each prototype frequency-domain curve (A′, B′, C′, D′), there is a corresponding time domain impulse response. It will be appreciated by one skilled in the art that a downward shift along the logarithmic frequency scale of each prototype frequency response (as used to produce A, B, and each Ci), corresponds to an equivalent upward shift on the algorithmic time scale. Based on these shifted time-domain prototype filters, the plurality of equally spaced sampling points, e.g. 70 in the logarithmic time scale 72 can be derived as illustrated in
We note that it will be appreciated by the person skilled in the art that for the desired filter response function, W0 to WN, shown in
It has been found by the applicant that the real and imaginary parts of the frequency response, W0 to WN, correspond to even and odd time-domain impulse response functions respectively. Furthermore, if the frequency response, W0 to WN, is causal, then this implies that only the real part of the response needs to be computed. For example, if the real and imaginary parts of W0 to WN are converted into their corresponding impulse responses:
hr=Convert((W0), (W1), . . . (WN))
hi=Convert(ℑ(W0), ℑ(W1), . . . ℑ(WN))
then we know that hr will be an even function, and hi will be an odd function. Therefore:
hr(i)=hr(−i)
hi(i)=−hi(−i)
but, because we have chosen our frequency response to ensure that the impulse response will be causal, we therefore know that:
h(i)=hr(i)+hi(i)=0 {i<0}
and this implies that:
hr(i)=−hi(i) {i<0)
and the odd and even properties of hr and hi therefore mean that:
hr(i)=hi(i) {i>0)
this means that we can compute h(i) from hr(i) by the following method:
It has therefore been found by the applicant that, if the frequency response, W0 to WN, is generated or chosen to ensure that the resulting impulse response is causal, then the imaginary parts of the sequence, W0 to WN, can be ignored, and the real part can be used to compute the impulse response by the equation given above.
It will be appreciated by the person skilled in the art that numerous modifications and/or variations may be made to the present invention as shown in the specific embodiments without departing from the spirit or scope of the invention as broadly described. The present embodiments are, therefore, to be considered in all respects to be illustrative and not restrictive.
In the claims that follow and in the description of the invention, except where the context requires otherwise due to express language or necessary implication the word “comprising” is used in the sense of “including,” i.e., the features specified may be associated with further features in various embodiments of the invention.
Number | Name | Date | Kind |
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5272656 | Genereux | Dec 1993 | A |
5732002 | Lee et al. | Mar 1998 | A |
6680972 | Liljeryd et al. | Jan 2004 | B1 |
Number | Date | Country | |
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20040122879 A1 | Jun 2004 | US |