DE3047450A1 - Filter for sampling frequency adjustment - has first stage defining blocking region and second stage adjusting transmission region of first stage - Google Patents

Filter for sampling frequency adjustment - has first stage defining blocking region and second stage adjusting transmission region of first stage

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Publication number
DE3047450A1
DE3047450A1 DE19803047450 DE3047450A DE3047450A1 DE 3047450 A1 DE3047450 A1 DE 3047450A1 DE 19803047450 DE19803047450 DE 19803047450 DE 3047450 A DE3047450 A DE 3047450A DE 3047450 A1 DE3047450 A1 DE 3047450A1
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Prior art keywords
filter
stage
sampling frequency
factor
sub
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Granted
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DE19803047450
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German (de)
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DE3047450C2 (en
Inventor
Heinz Dipl.-Ing. 7150 Backnang Göckler
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Robert Bosch GmbH
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Licentia Patent Verwaltungs GmbH
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Priority to DE19803047450 priority Critical patent/DE3047450C2/en
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Publication of DE3047450C2 publication Critical patent/DE3047450C2/en
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    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03H—IMPEDANCE NETWORKS, e.g. RESONANT CIRCUITS; RESONATORS
    • H03H17/00—Networks using digital techniques
    • H03H17/02—Frequency selective networks
    • H03H17/06—Non-recursive filters
    • H03H17/0621—Non-recursive filters with input-sampling frequency and output-delivery frequency which differ, e.g. extrapolation; Anti-aliasing
    • H03H17/0635—Non-recursive filters with input-sampling frequency and output-delivery frequency which differ, e.g. extrapolation; Anti-aliasing characterized by the ratio between the input-sampling and output-delivery frequencies
    • H03H17/0685—Non-recursive filters with input-sampling frequency and output-delivery frequency which differ, e.g. extrapolation; Anti-aliasing characterized by the ratio between the input-sampling and output-delivery frequencies the ratio being rational

Abstract

The filter comprises two filter stages, the first of which is non-recursive and the second of which is recursive or non-recursive with a given phase characteristic. The first filter stage is preceded by a switch (K1) used for increasing the sampling frequency by a factor which is a positive whole number greater than 1, by insertion of a number of zeroes between each two sample values of the input signal (S1). The first stage is followed by a second switch (K2) which reduces the supplied signal by a second factor which comprises a second positive whole number.

Description

Filter zur Änderung der Abtastfreguenz Filter for changing the sampling frequency

Die Erfindung betrifft ein Filter zur Änderung der Abtastfrequenz bei der digitalen oder zeitdiskreten Signalverarbeitung. Solche Filter werden z.B. in der digitalen Tontechnik verwendet.The invention relates to a filter for changing the sampling frequency in digital or time-discrete signal processing. Such filters are e.g. used in digital sound engineering.

Äus der Zeitschrift Proc. IEEE Vol. 61, No. 6, Juni 1973, Seite 692 - 702, ist ein Aufsatz "A Digital Signal Processing Approach to Interpolation von R.W. Schafer und L.R. Rabiner bekannt geworden, in dem im Abschnitt II, D ein Verfahren n zur Veränderung der Abtastfrequenz um den Faktor m beschriem ben wird. Dabei wird ein transversales Filter verwendet, das zur Erfüllung der gestellten Aufgabe einen hohen Filtergrad erfordert, so daß der Aufwand groß wird.From the magazine Proc. IEEE Vol. 61, No. Jun 6, 1973, p. 692 - 702, is an essay "A Digital Signal Processing Approach to Interpolation by R.W. Schafer and L.R. Rabiner became known in which in Section II, D a procedure n for changing the sampling frequency by a factor of m is described. It will a transversal filter is used to accomplish the task at hand requires a high degree of filtering, so that the effort is great.

In der Patentanmeldung P 39 28 705.9 ist ein minimalphasiges Filter zur Anderung der Abtastfrequenz beschrieben, das aus zwei Teilfiltern zusammengesetzt ist. Da sich jedes Teilfilter auf einfache Weise realisieren läßt, ergibt sich insgesamt ein optimales minimalphasiges Filter. Der Nachteil dieses Filters besteht darin, daß die Abtastfrequenz entweder nur um einen ganzzahligen Faktor erhöht oder erniedrigt werden kann.In the patent application P 39 28 705.9 is a minimum phase filter to change the sampling frequency, which is composed of two sub-filters is. Since each sub-filter can be implemented in a simple manner, the overall result is an optimal minimum phase filter. The disadvantage this filter is that the sampling frequency is either only increased by an integer factor can be increased or decreased.

Der Erfindung liegt daher die Aufgabe zugrunde, ein Filter n zur Änderung der Abtastfrequenz um den Faktor m anzugeben, wobei n und m natürliche Zahlen sind und n * 1 und m g 1 sein soll und das mit geringerem Aufwand als bisher möglich realisiert werden kann.The invention is therefore based on the object of providing a filter n for change the sampling frequency by the factor m, where n and m are natural numbers and n * 1 and m g 1 should be and that with less effort than previously possible can be realized.

Die Aufgabe wird gelöst wie im Anspruch 1 beschrieben, die Unteransprüche geben vorteilhafte Ausgestaltungen an.The object is achieved as described in claim 1, the subclaims indicate advantageous configurations.

Im folgenden sei die Erfindung anhand eines Ausführungsbeispiels mit Hilfe der Figuren näher erläutert.In the following the invention is based on an exemplary embodiment Explained in more detail using the figures.

Figur 1 zeigt eine erfindungsgemäße Anordnung. Vor dem ersten Teilfilter wird mit Hilfe des symbolisch dargestellten Umschalters K1 die Abtastfrequenz fA1 durch Hinzufügen von n-1 Nullwerten zwischen je zwei Abtastwerten um den Faktor n erhöht. Das erste Teilfilter hat die Wirkungsfunktion H5 und realisiert damit den Sperrbereich der Gesamtwirkungsfunktion.Figure 1 shows an arrangement according to the invention. Before the first partial filter the sampling frequency fA1 is set with the aid of the symbolically represented switch K1 by adding n-1 zero values between every two samples by the factor n increased. The first sub-filter has the action function H5 and thus realizes it the blocking range of the overall action function.

Wenn man anschließend nur jeden m-ten Abtastwert weiterverarbeitet, dies ist symbolisch durch den Schalter K2 dargestellt, dann wird die Abtastfrequenz um den Faktor m erniedrigt. Insgesamt ergibt sich dadurch eine Veränderung der n n Abtastfrequenz um den Faktor m' so daß fA2 = m . fA1 wird.If one then only further processes every mth sample, this is symbolically represented by the switch K2, then the sampling frequency decreased by the factor m. Overall, this results in a change in the n n sampling frequency by the factor m 'so that fA2 = m. fA1 becomes.

Das zweite Teilfilter hat die Wirkungsfunktion Hp und entzerrt damit den Durchlaßbereich des ersten Teilfilters Hs.The second sub-filter has the action function Hp and thus equalizes the pass band of the first sub-filter Hs.

Figur 2a zeigt die spektrale Darstellung der Wirkungsfunktion H5 und das Spektrum S1 der Signale s1(kT) bzw. s1(kT/n), wobei der Faktor n = 5 ist.Figure 2a shows the spectral representation of the action function H5 and the spectrum S1 of the signals s1 (kT) or s1 (kT / n), where the factor n = 5.

Figur 2b zeigt das verbleibende Spektrum S2 des Signals s2(kT/n) nach der Filterung durch das erste Teilfilter. Figur 2c zeigt die spektrale Darstellung der Wirkungsfunktion Hp und das Spektrum S3 des Signals s3(kTn) nach dem Schalter K2, wobei der Faktor m = 4 ist. Die Figur 2d zeigt das Spekm trum S4 des Signals s4(kTW) nach der Filterung des zweiten Teilfilters. Die Figur 2e zeigt das Produkt der Wirkungsfunktionen Hp Für die Realisierung des ersten Teilfilters wird vorteilhaft eine nicht rekursive Filterstruktur verwendet, wie es in dem Aufsatz Proc. IEEE Vol. 61, No. 6, Juni 1973, Seite 692 - 702 beschrieben ist. Dagegen können für das zweite Teilfilter sowohl rekursive als auch nicht rekursive Filterstrukturen zum Einsatz kommen, wie es in den Unteransprüchen ausgeführt ist.FIG. 2b shows the remaining spectrum S2 of the signal s2 (kT / n) after filtering through the first sub-filter. Figure 2c shows the spectral representation the action function Hp and the spectrum S3 of the signal s3 (kTn) after the switch K2, where the factor m = 4. FIG. 2d shows the spectrum S4 of the signal s4 (kTW) after filtering the second sub-filter. Figure 2e shows the product of the action functions Hp It is advantageous to implement the first sub-filter uses a non-recursive filter structure as described in the article Proc. IEEE Vol. 61, No. 6, June 1973, pages 692-702. On the other hand, for that second sub-filter for both recursive and non-recursive filter structures Use come as it is set out in the subclaims.

LeerseiteBlank page

Claims (5)

Patentansprüche 9 Filter zur änderung der Abtastfrequenz bei der digitalen oder zeitdiskreten Signalverarbeitung, wobei das Filter aus zwei Teilfiltern zusammengesetzt ist, wobei alle Nullstellen der Wirkungsfunktion des ersten Teilfilters auf dem Einheitskreis der z-Ebene liegen und damit die Sperrbereiche realisierbar sind und wobei mit Hilfe des zweiten Teilfilters die Durchlaßbereiche des ersten Teilfilters entzerrbar sind, dadurch gekennzeichnet, daß zur Veränden rung der Abtastfrequenz um den Faktor m wobei n, m positive ganze Zahlen und n W 1 und m ; 1 ist, vor dem ersten Teilfilter die Abtastfrequenz durch Hinzufügen von n-1 Nullen zwischen je zwei Abtastwerten des Eingangssignals um den Faktor n erhöht wird, daß im ersten Teilfilter die um den Faktor n erhöhte Abtastfrequenz verarbeitbar ist, daß im ersten Teilfilter die um den Faktor n erhöhte Abtastfrequenz um den Faktor m erniedrigbar ist und daß im nachgeschalteten zweiten Teilfilter das Signal mit der um den n Faktor m veränderten Abtastfrequenz verarbeitbar ist. Claims 9 filter for changing the sampling frequency at the digital or time-discrete signal processing, whereby the filter consists of two sub-filters is composed, with all zeros of the action function of the first sub-filter lie on the unit circle of the z-plane and thus the restricted areas can be implemented are and with the aid of the second sub-filter the passbands of the first Partial filters can be equalized, characterized in that for changing the sampling frequency by the factor m where n, m are positive integers and n W 1 and m; 1 is before first sub-filter the sampling frequency by adding n-1 zeros between each two samples of the input signal is increased by the factor n that in the first Partial filter, the sampling frequency increased by the factor n can be processed that in the first Partial filter, the sampling frequency increased by the factor n can be decreased by the factor m is and that in the downstream second sub-filter the signal with the by the n factor m changed sampling frequency can be processed. 2. Filter nach Anspruch 1, dadurch gekennzeichnet, daß das erste Teil filter als nicht rekursives Filter ausgebildet ist.2. Filter according to claim 1, characterized in that the first part filter is designed as a non-recursive filter. 3. Filter nach Anspruch 1, dadurch gekennzeichnet, daß das zweite Teilfilter als rekursives Filter ausgebildet ist.3. Filter according to claim 1, characterized in that the second Partial filter is designed as a recursive filter. 4. Filter nach Anspruch 1, dadurch gekennzeichnet, daß das zweite Teilfilter als rein rekursives Filter ausgebildet ist.4. Filter according to claim 1, characterized in that the second Partial filter is designed as a purely recursive filter. 5. Filter nach Anspruch 1, dadurch gekennzeichnet, daß das zweite Teilfilter als nicht rekursives Filter mit beliebigem Phsenverlauf ausgebildet ist.5. Filter according to claim 1, characterized in that the second Partial filter is designed as a non-recursive filter with any phase curve.
DE19803047450 1980-12-17 1980-12-17 Filter for changing the sampling frequency Expired DE3047450C2 (en)

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Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
DE3418011A1 (en) * 1983-08-11 1985-02-28 LGZ Landis & Gyr Zug AG, Zug Bandpass filter for receiving a sound signal transmitted via an electrical power supply network
DE3605927A1 (en) * 1986-02-25 1987-08-27 Standard Elektrik Lorenz Ag Digital interpolator
US5433208A (en) * 1991-03-01 1995-07-18 Siemens Aktiengesellschaft Device for filtering out baseline fluctuations from physiological measurement signals

Citations (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
DE3028705A1 (en) * 1980-07-29 1982-02-11 Licentia Patent-Verwaltungs-Gmbh, 6000 Frankfurt Minimal phase digital filter - adjusts sampling frequency for signal processing system using two series stages to multiply sampling rate

Patent Citations (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
DE3028705A1 (en) * 1980-07-29 1982-02-11 Licentia Patent-Verwaltungs-Gmbh, 6000 Frankfurt Minimal phase digital filter - adjusts sampling frequency for signal processing system using two series stages to multiply sampling rate

Non-Patent Citations (2)

* Cited by examiner, † Cited by third party
Title
DE-Z.: Frequenz 28 (1974) H.3, S.82-85 *
US-Z.: Proceedings of the IEEE Vol.61, No.6, Juni 1973, S.692-702 *

Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
DE3418011A1 (en) * 1983-08-11 1985-02-28 LGZ Landis & Gyr Zug AG, Zug Bandpass filter for receiving a sound signal transmitted via an electrical power supply network
DE3605927A1 (en) * 1986-02-25 1987-08-27 Standard Elektrik Lorenz Ag Digital interpolator
US5433208A (en) * 1991-03-01 1995-07-18 Siemens Aktiengesellschaft Device for filtering out baseline fluctuations from physiological measurement signals

Also Published As

Publication number Publication date
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