EP1297627A1 - Interpolationsfilter und verfahren zur digitalen interpolation eines digitalen signals - Google Patents
Interpolationsfilter und verfahren zur digitalen interpolation eines digitalen signalsInfo
- Publication number
- EP1297627A1 EP1297627A1 EP01969330A EP01969330A EP1297627A1 EP 1297627 A1 EP1297627 A1 EP 1297627A1 EP 01969330 A EP01969330 A EP 01969330A EP 01969330 A EP01969330 A EP 01969330A EP 1297627 A1 EP1297627 A1 EP 1297627A1
- Authority
- EP
- European Patent Office
- Prior art keywords
- interpolation filter
- filter
- digital input
- signal
- input signal
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Withdrawn
Links
Classifications
-
- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03H—IMPEDANCE NETWORKS, e.g. RESONANT CIRCUITS; RESONATORS
- H03H17/00—Networks using digital techniques
- H03H17/02—Frequency selective networks
- H03H17/0294—Variable filters; Programmable filters
-
- 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
- H03H17/065—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 integer
- H03H17/0664—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 integer where the output-delivery frequency is lower than the input sampling frequency, i.e. decimation
Definitions
- the invention relates to an interpolation filter and a method for interpolating a digital signal, which can be used in particular for converting the sampling rate.
- Interpolation filters are used as subcircuits in digital circuit systems that require a change in the sampling rate of digital signals. Systems that deal only with simple integer sample rate ratios are not the subject of the invention.
- the associated circuits are referred to as hybrid systems, which consist of a first interpolation filter with a fixed sampling rate ratio and a second interpolation filter.
- the second interpolation filter determines intermediate values that lie at any time between the fixed sample values of the sampling grid after the second interpolation filter and thus permit any sampling rate relationships.
- the first interpolation filter contains a combination of an interpolation device and a digital filter. With the interpolation device, which is also referred to as an oversampling device, "0" values are corresponding to an oversampling factor N between the original ones
- the first interpolation filter is designed such that larger frequency range gaps are formed in the infinitely extending frequency spectrum. It also applies to oversampling that the frequency spectra are reflected at half the original sampling frequency and its multiples.
- a new sampling frequency can be assumed which is in an integer frequency ratio to the original sampling frequency. The digital filter removes the remaining spectral components between the useful signal band and the mirrored frequency band at the new sampling frequency and the associated frequency multiples.
- the digital filter simply functions as a digital low-pass filter, which passes the useful signal frequency range and suppresses the frequency components above it. However, in accordance with the sampling theorem, a mirroring occurs at half the sampling frequency. A digital low pass filter can therefore not suppress multiples of the sampling frequency.
- the spectral signal components at the new sampling frequency and the frequency multiple values must be suppressed in order to implement any sampling rate relationships. If these signal interference components are not suppressed, then signal interference components occur in the useful signal frequency band when generating any sampling rate ratios.
- the first interpolation filter is described in "Proceedings of the IEEE",
- EP-A-0 561 067 describes a method with a hybrid
- a second interpolation filter is implemented as a low-pass filter, which suppresses all signal components whose frequencies are above 1.5 times the value of the original sampling frequency.
- the analog low-pass behavior is achieved with a transversal filter, in which the weighting factors of the stored samples depend on a time difference value.
- Such a low-pass filter not only suppresses the remaining spectral signal components at the frequency multiple values of the new sampling frequency, but also the entire frequency spectral range above a blocking edge. According to a comparable pass / block behavior, such a low-pass filter can be realized only with great effort in comparison with a corresponding comb filter arrangement.
- Theory and VLSI Architectures for Asynchronous Sample Rate Converters describes a method for a sample rate conversion system which, on the one hand, treats the use of simpler sample and hold circuits and, on the other hand, the use of low-pass filters as analog resamplers.
- interference signal frequency ranges the center frequencies of which lie at the frequency multiple values of the new sampling frequency.
- the frequency bandwidth of each signal interference area is equal to twice the frequency bandwidth of the useful signal. If the Nyquist condition for the original digitization is met, the frequency bandwidth of the interference signal range has the maximum value of the original sampling frequency in the limit case.
- the location and bandwidth of all interference areas is in Frequency spectrum defined by the original sampling frequency and the original oversampling factor N.
- the N-fold oversampling of the original digital sampling sequence has the effect that the relative frequency bandwidth of the interference signal ranges in the frequency spectrum is reduced by a factor of 1 / N in relation to the new sampling frequency. This facilitates the separation of the useful signal frequency band from the respective interference signal frequency range, since the transition range between the pass and the blocking frequency range for the second interpolation filter is increased. This reduces the circuitry required for the second interpolation filter. However, this is paid for by a higher circuit complexity for the smoothing filter in the first interpolation filter.
- EP 0 696 848 AI therefore proposed a method for digital interpolation of signals which leads to a very high signal-to-noise ratio with, at the same time, low circuit complexity for the filter system, which consists of a first and second interpolation filter.
- weighting factors or filter coefficients are multiplied by delayed input values of a digital signal having a first clock frequency, the delay being dependent on a time difference value which is determined by the time of interpolation and by the time grid of the first clock signal ,
- the filter coefficients of the interpolation filter are determined by the impulse response h (t) in the time domain.
- the associated transfer function H (F) has a signal damping behavior in the frequency domain, which essentially relates to the signal interference ranges lying at the frequency multiples of the first clock frequency is limited. Each of these signal interference areas in the frequency spectrum are assigned at least two adjacent zeros. If there are double-order zeros, at least one of the interference areas and the associated periodic interference areas is assigned at least one further zero of the transfer function H (F).
- the amplitude response of the interpolation filter described in EP 0 696 841 AI is comb-shaped and, because of the narrowband interference signal frequency ranges, has only a very narrowband useful signal frequency range.
- the invention provides an interpolation filter for filtering a digital input signal, the amplitude response of which has a low-pass attenuation curve in the useful signal frequency range of the digital input signal.
- the interpolation filter according to the invention offers the advantage that broadband digital input signals can also be processed.
- interpolation filter according to the invention can also be used for analog / digital converters with the highest sampling frequencies, since in practice Applications the entire circuit is calculated on only one to four times the useful signal bandwidth.
- the low sampling frequencies or the long clock periods T of the digital signal processing offer the advantage that the components of the interpolation filter, for example demultiplexers, operate at low frequencies and are therefore particularly simple to implement in terms of circuitry.
- the interpolation filter is followed by a high-pass filter to compensate for the low-pass amplitude response of the interpolation filter.
- the group delay of the interpolation filter advantageously runs essentially constant.
- the digital input signal which is filtered by the interpolation filter according to the invention, is preferably an equidistant digital signal with a predetermined clock period T in .
- the group delay of the interpolation filter according to the invention is preferably adjustable within the clock period T ⁇ n of the digital input signal.
- the ratio of the clock periods of the digital input signal T in and the digital output signal T aUs filtered by the interpolation filter is preferably adjustable.
- the interpolation filter and the downstream high-pass filter together have a sinc filter characteristic.
- a further interpolation filter for narrowing the useful signal frequency range is preferably connected upstream of the interpolation filter.
- the upstream interpolation filter is preferably a polyphase filter.
- the interpolation filter consists of a filter coefficient generator for generating filter coefficients as a function of a base function, a multiplier for multiplying the digital input signal with the generated filter coefficients, and an accumulator for accumulating the by multiplication weighted digital input signal.
- the basic function is preferably stored in a memory device of the interpolation filter.
- the interpolation filter according to the invention has a basic function generating device for generating the basic function as a function of basic functions.
- a memory device is preferably provided for storing the basic functions.
- this has a controllable switching device which can be switched as a digital output signal for reading out the weighted digital input signal.
- the accumulator consists of an adder and a register, the output of which is fed back to an input of the adder.
- the invention also provides a method for digital interpolation of a digital input signal with the features specified in claim 16.
- the invention provides a method for digitally interpolating a digital input signal with the following steps, namely
- the filter coefficients of the interpolation filter are preferably determined as a function of a basic function.
- This basic function is preferably stored beforehand in a memory.
- the basic function is generated according to a further embodiment of the method according to the invention from predetermined basic functions.
- a first basic function is preferably a time-limited potentiated sine function.
- the second basic function is preferably a sample hold function of the first order.
- a multiplicity of filter coefficient sets of the interpolation filter are generated as a function of the basic function, each of which has an essentially identical amplitude response but different group delays in the useful signal frequency range, with the filter coefficient set subsequently being used to determine the filter coefficients of the Interpolation filter is selected, the group delay ⁇ corresponds to the set group delay.
- FIG. 1 shows a typical circuit arrangement which contains the interpolation filter according to the invention
- 3b shows the group delay curve of an interpolation filter according to the invention
- 4a shows the amplitude response of a first exemplary interpolation filter according to the invention
- FIG. 4b shows the associated group delay curve of the interpolation filter according to the invention with the amplitude response according to FIG. 4a;
- 5a shows the amplitude response of a further interpolation filter according to the invention
- 5b shows the group delay curve of the interpolation filter with the amplitude response shown in FIG. 5a;
- FIG. 6 shows an example of a basic function which is used to determine the filter coefficients of the interpolation filter according to the invention
- FIG. 7 shows the course of the group delay of a preferred embodiment of the interpolation filter according to the invention with the basic function shown in FIG. 6 in comparison to the curve of the group delay of an interpolation filter according to the prior art.
- FIG. 1 shows a typical circuit arrangement in which the interpolation filter according to the invention is used to filter a digital input signal.
- the interpolation filter 5 has setting lines 6, 7 for setting the target group delay ⁇ and the decimation factor K.
- the interpolation filter 5 filters the digital input signal present on the line 4 and outputs a filtered one digital output signal via a signal line 8 to a downstream high-pass filter 9.
- the high-pass filter 9 filters the filtered output signal of the interpolation filter 5 according to the invention, which is present on the line 8, and emits a corresponding filtered output signal via a line 10.
- the signal at the interpolation filter 5 digital input signal has a clock frequency f ⁇ n, which corresponds to the sampling frequency f abt branch of the analog / digital converter. 2
- the filtered digital output signal applied to the signal output line 8 has an output clock frequency f out .
- the decimation factor K which can be set via the setting line 7, indicates the ratio between the input frequency f n of the digital input signal and the output frequency f from the filtered digital output signal.
- the interpolation filter 5 has an amplitude response with a low-pass-shaped attenuation curve in the useful signal frequency range of the digital input signal present on line 4. Due to the low-pass attenuation curve of the interpolation filter, signal distortions of the digitized output signal of the interpolation filter 5 occur.
- the downstream high-pass filter 9 is used to eliminate these distortions by compensating the low-pass amplitude response of the interpolation filter 5 by an amplitude response that is complementary to it.
- FIG. 2 shows a preferred embodiment of the interpolation filter 5 according to the invention shown in FIG. 1.
- the interpolation filter 5 has a signal input 11 for receiving a digital input signal.
- the digital signal input 11 of the interpolation filter 5 is connected to a multiplication device 13 via a line 12.
- the multiplication device 13 multiplies the digital input signal present on the line 12 by filter coefficients or weighting factors which are present on a line 14 of the interpolation filter 5.
- the filter coefficients of the interpolation filter 5 are generated in a filter coefficient generator 15 of the interpolation filter 5.
- the filter coefficient generating device 15 is connected via internal setting lines 16, 17 to setting connections 18, 19 of the interpolation filter 5.
- the desired decimation factor K can be set via the setting connection 18 of the interpolation filter 5.
- the desired group delay ⁇ of the interpolation filter 5 can be set at the setting connection 19.
- the filter coefficient generator 15 generates the filter coefficients depending on a basic function.
- the base function in the embodiment shown in FIG. 2 is stored in a memory device 20 and is read out via an internal line 21 by the filter coefficient generator 15.
- the basic function is not stored in advance, but is generated by a basic function generation device as a function of basic functions.
- the basic functions are preferably stored in a memory device.
- the multiplication-weighted digital input signal passes from the multiplication device 13 via an internal line 22 to an accumulator 23 for accumulation of the weighted digital input signal.
- the accumulator 23 contains an adder 24 which is connected on the output side to a register 26 via a line 25.
- the output line 27 of the register 26 is connected to a via a line 28 1 1 CN ⁇ P 1 1 P 1 1 J 1 1 1 c ⁇ m 1 oo P rö G ⁇ GPG CO CO N 1 £ rö
- the group delay time ⁇ of the interpolation filter 5 in the useful signal frequency band .DELTA.f nut for the digital input signal is substantially constant and does not expire until the higher frequency in the frequency ranges auseinan-.
- the group delays caused by the different filter coefficient sets, which are generated based on the basic function by the filter coefficient generator 15, are different.
- the filter coefficient generator 15 compares the group delays ⁇ with that via the setting line 17 set target group delay ⁇ so ⁇ and selects the filter coefficient set whose group delay within the useful signal frequency range ⁇ f use corresponds to the set target group delay.
- the filter coefficient set is selected in which the deviation between the group delay ⁇ constant in the useful signal frequency range and the target group delay ⁇ S oi ⁇ is minimal.
- FIGS. 5a, 5b show a further example of an interpolation filter 5 according to the invention, the useful signal frequency range of which is approximately 0.24 fi n . It can be seen from FIGS. 5a, 5b that the attenuation curve inside and outside the useful signal frequency range is low-pass.
- FIGS. 4a, 4b shows the course of the basic function BF (x) used for the interpolation filter shown in FIGS. 4a, 4b.
- the interpolation filter 5 can be followed by a high-pass filter 9 in order to compensate for distortions which arise due to the low-pass-shaped attenuation curve of the amplitude response of the interpolation filter 5.
- the series connection of the interpolation filter 5 with the high-pass filter 9 preferably has a sinc filter characteristic.
- the interpolation filter 5 can be preceded by another interpolation filter of conventional type for narrowing the useful signal frequency range.
- This upstream interpolation filter can be a polyphase filter.
- the filter coefficients of the adjustable interpolation filter 5 are determined in such a way that the amplitude response has a low-pass attenuation curve in the useful signal frequency range ⁇ f util of the digital input signal.
- the filter coefficients of the interpolation filter 5 become dependent determined by a basic function BF.
- This basic function BF is either stored beforehand in an internal memory 20 of the interpolation filter 5 or generated by a basic function generating device on the basis of predetermined basic functions GF.
- Two fundamental basic functions are preferably used, the first basic function being a time-limited potentiated sine function with the following equation:
- the second fundamental basic function is a first order hold function with the following equation:
- h 2 (t) ⁇ (t) - ⁇ (tn) (4) where ⁇ (tn) is the unit jump at time n.
- the basic functions BF can either consist of the basic functions GF according to equations (3), (4) themselves or can be generated by linking operations of the basic functions in the basic function generating device.
- the link operations include the following operations:
- the basic function can be stored as a sampled impulse response in a memory device 20, for example a ROM memory, of the interpolation filter 5.
- a memory device 20 for example a ROM memory
- Generation device 15 read out. It is also possible to approximate the impulse response of the basic function BF as a whole or in sections using polynomials.
- the basic functions BF can also be generated on the basis of the basic functions GF by multiple operative linking.
- the interpolation filter according to the invention meets various requirements.
- the difference in the amplitude responses of the individual polyphases is minimized with a given circuit complexity.
- the group delays ⁇ of the individual polyphases run essentially constant within a clock period T in of the digital input signal.
- Each individual polyphase has an amplitude difference of at least 2 dB.
- the interpolation filter according to the invention has a low-pass characteristic.
- Hybrid systems can also be constructed with the interpolation filter according to the invention.
- the interpolation filter is divided into two poly phases, with two architectures being available for implementation.
- the first architecture the even filter coefficients are multiplied by one polyphase and the odd filter coefficients by the other polyphase.
- a low pass signal is generated by adding the two poly phases. This signal is then folded using the sampled time-continuous filter.
- a high-pass signal is also generated by subtracting one polyphase from the other. Thereupon every second sample value is inverted in the continuous-time filter before a signal convolution is carried out. Finally, the folded low-pass and high-pass signals are added together.
- FIG. 7 shows the group delay curve of an interpolation filter 5 according to the invention in comparison to the group delay curve of a conventional interpolation filter according to the prior art, which has a sinc filter characteristic.
Landscapes
- Physics & Mathematics (AREA)
- Engineering & Computer Science (AREA)
- Computer Hardware Design (AREA)
- Mathematical Physics (AREA)
- Complex Calculations (AREA)
- Analogue/Digital Conversion (AREA)
- Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)
Abstract
Description
Claims
Applications Claiming Priority (3)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
DE10032520A DE10032520A1 (de) | 2000-07-05 | 2000-07-05 | Interpolationsfilter und Verfahren zur digitalen Interpolation eines digitalen Signals |
DE10032520 | 2000-07-05 | ||
PCT/EP2001/007543 WO2002003550A1 (de) | 2000-07-05 | 2001-07-02 | Interpolationsfilter und verfahren zur digitalen interpolation eines digitalen signals |
Publications (1)
Publication Number | Publication Date |
---|---|
EP1297627A1 true EP1297627A1 (de) | 2003-04-02 |
Family
ID=7647772
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
EP01969330A Withdrawn EP1297627A1 (de) | 2000-07-05 | 2001-07-02 | Interpolationsfilter und verfahren zur digitalen interpolation eines digitalen signals |
Country Status (4)
Country | Link |
---|---|
US (1) | US7225213B2 (de) |
EP (1) | EP1297627A1 (de) |
DE (1) | DE10032520A1 (de) |
WO (1) | WO2002003550A1 (de) |
Families Citing this family (7)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US7627021B2 (en) * | 2003-01-30 | 2009-12-01 | The Mitre Corporation | Interference canceling CDMA mobile station receiver |
US7606309B2 (en) * | 2004-09-30 | 2009-10-20 | Intel Corporation | Motion estimation for video processing using 2-D (spatial) convolution |
DE102006045794A1 (de) * | 2006-09-26 | 2008-03-27 | Micronas Gmbh | Vorrichtung und Verfahren zum polyphasigen Resampling |
CN101915931B (zh) * | 2010-07-09 | 2012-07-04 | 中国人民解放军国防科学技术大学 | 高精度延迟滤波器的多级插值设计方法 |
US9002917B2 (en) * | 2010-07-30 | 2015-04-07 | National Instruments Corporation | Generating filter coefficients for a multi-channel notch rejection filter |
US9424696B2 (en) | 2012-10-04 | 2016-08-23 | Zonar Systems, Inc. | Virtual trainer for in vehicle driver coaching and to collect metrics to improve driver performance |
CN108226636B (zh) * | 2016-12-15 | 2021-06-11 | 欧姆龙株式会社 | 自动滤波方法和装置 |
Family Cites Families (9)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
ATE14358T1 (de) | 1980-11-26 | 1985-08-15 | Studer Willi Ag | Verfahren und schaltungsanordnung zur umsetzung der abtastfrequenz einer abtastfolge unter umgehung der konversion in ein kontinuierliches signal. |
GB9205614D0 (en) * | 1992-03-14 | 1992-04-29 | Innovision Ltd | Sample rate converter suitable for converting between digital video formats |
US5475628A (en) * | 1992-09-30 | 1995-12-12 | Analog Devices, Inc. | Asynchronous digital sample rate converter |
US5717617A (en) * | 1993-04-16 | 1998-02-10 | Harris Corporation | Rate change filter and method |
US5548540A (en) * | 1994-06-24 | 1996-08-20 | General Electric Company | Decimation filter having a selectable decimation ratio |
DE59409276D1 (de) * | 1994-08-08 | 2000-05-11 | Micronas Intermetall Gmbh | Verfahren zur digitalen Interpolation von Signalen |
KR100299139B1 (ko) * | 1997-12-31 | 2001-11-14 | 윤종용 | 데시메이션여파기장치및방법 |
US6487573B1 (en) * | 1999-03-26 | 2002-11-26 | Texas Instruments Incorporated | Multi-rate digital filter for audio sample-rate conversion |
US6772181B1 (en) * | 1999-10-29 | 2004-08-03 | Pentomics, Inc. | Apparatus and method for trigonometric interpolation |
-
2000
- 2000-07-05 DE DE10032520A patent/DE10032520A1/de not_active Withdrawn
-
2001
- 2001-07-02 EP EP01969330A patent/EP1297627A1/de not_active Withdrawn
- 2001-07-02 WO PCT/EP2001/007543 patent/WO2002003550A1/de active Application Filing
- 2001-07-02 US US10/070,203 patent/US7225213B2/en not_active Expired - Fee Related
Non-Patent Citations (1)
Title |
---|
See references of WO0203550A1 * |
Also Published As
Publication number | Publication date |
---|---|
US7225213B2 (en) | 2007-05-29 |
DE10032520A1 (de) | 2002-01-24 |
WO2002003550A1 (de) | 2002-01-10 |
US20020184278A1 (en) | 2002-12-05 |
Similar Documents
Publication | Publication Date | Title |
---|---|---|
EP0889588B1 (de) | Filterkombination zur Abtastratenumsetzung | |
DE60207841T2 (de) | Alias-reduktion unter verwendung komplex-exponentiell modulierter filterbänke | |
DE69831271T2 (de) | Vorrichtung und verfahren der dezimierungsfilterung | |
DE3044208C2 (de) | Interpolator zur Erhöhung der Wortgeschwindigkeit eines digitalen Signals | |
EP0696848B1 (de) | Verfahren zur digitalen Interpolation von Signalen | |
DE60219064T2 (de) | Aufwartsabtaster-, abtastrateumsetzungs- und abtastfrequenzerniedrigungsstufen enthaltendes zeitdiskretes filter | |
DE10317698B4 (de) | Verfahren zum Entwerfen von Polynomen zur Steuerung des Veränderns von anpassungsfähigen Digitalfiltern | |
EP1297627A1 (de) | Interpolationsfilter und verfahren zur digitalen interpolation eines digitalen signals | |
WO2000035096A2 (de) | Analog-digital-umsetzer | |
DE19521610B4 (de) | Dezimationsfilter unter Verwendung einer Nullfüllschaltung zur Lieferung eines wählbaren Dezimationsverhältnisses sowie Verfahren zur Dezimationsfilterung | |
DE60034964T2 (de) | Programmierbarer convolver | |
EP0234452B1 (de) | Digitale Schaltungsanordung zur Abtastratenänderung und Signalfilterung und Verfahren zu ihrem Entwurf | |
DE102011116217A1 (de) | Verwendung eines multilevel-pulsweitenmodulierten Signals zur Realzeit-Rauschauslöschung | |
DE19510655B4 (de) | Schaltungsanordnung zum Filtern eines Stroms quantisierter elektrischer Signale und Verfahren zum Filtern eines Stoms quantisierter elektrischer Signale | |
EP1092269A2 (de) | Verfahren zur digitalen taktrückgewinnung und selektiven filterung | |
DE3044582A1 (de) | Digitaler verstaerker, insbesondere zur verwendung in einer digitalen fernsprech-teilnehmerschaltung | |
DE69921327T2 (de) | Digital/analog-wandler | |
DE102005018858B4 (de) | Digitales Filter und Verfahren zur Bestimmung seiner Koeffizienten | |
DE10317701B4 (de) | Verfahren und Digitalsignalverarbeitungseinheit zur Erzeugung von Filterkoeffizienten für Digitalfilter mit veränderlicher Bandbreite | |
DE4337134A1 (de) | Verfahren zur Aufbereitung eines digitalen Frequenzmultiplexsignals | |
DE19510656A1 (de) | Dezimierungs-Schaltungsanordnung und Verfahren zum Filtern quantisierter Signale | |
DE102006054776A1 (de) | Vorrichtung und Verfahren für die Sigma-Delta-Modulation | |
EP0402519B1 (de) | Verfahren und Anordnung zur Verbesserung des Dynamikbereichs eines adaptiven rekursiven Netzwerks zur Verarbeitung zeitdiskreter Signale | |
DE60028739T2 (de) | Digital-/analog-wandler | |
EP0731566A2 (de) | Schaltungsanordnung zur Umsetzung eines 1-Bit-Digital-signals in ein Analogsignal |
Legal Events
Date | Code | Title | Description |
---|---|---|---|
PUAI | Public reference made under article 153(3) epc to a published international application that has entered the european phase |
Free format text: ORIGINAL CODE: 0009012 |
|
17P | Request for examination filed |
Effective date: 20020308 |
|
AK | Designated contracting states |
Kind code of ref document: A1 Designated state(s): AT BE CH CY DE DK ES FI FR GB GR IE IT LI LU MC NL PT SE TR Designated state(s): AT BE CH CY DE DK ES FI FR GB GR IE IT LI LU MC NL PT SE TR |
|
AX | Request for extension of the european patent |
Extension state: AL LT LV MK RO SI |
|
RBV | Designated contracting states (corrected) |
Designated state(s): AT BE DE GB |
|
17Q | First examination report despatched |
Effective date: 20060726 |
|
GRAP | Despatch of communication of intention to grant a patent |
Free format text: ORIGINAL CODE: EPIDOSNIGR1 |
|
RBV | Designated contracting states (corrected) |
Designated state(s): DE GB |
|
STAA | Information on the status of an ep patent application or granted ep patent |
Free format text: STATUS: THE APPLICATION IS DEEMED TO BE WITHDRAWN |
|
18D | Application deemed to be withdrawn |
Effective date: 20090213 |