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Showing papers on "Butterworth filter published in 1982"


Book
01 Jan 1982
TL;DR: In this paper, Bilinear Transfer Functions and Frequency Response are discussed in the context of Op-Amp Oscillators, and a ladder design with simulated elements is presented.
Abstract: 1. Introduction 2. Resistor Op-Amp Circuits 3. Bilinear Transfer Functions and Frequency Response 4. Cascade Design with First-Order Circuits 5. The Biquad Circuit 6. Butterworth Low-Pass Filters 7. Butterworth Band-Pass Filters 8. The Chebyshev Response 9. Sensitivity 10. Delay Filters 11. Frequency Transformation 12. Highpass and Band-Elimination Filters 13. Inverse Chebyshev and Cauer Filters 14. Prototype and Frequency-Transformed Ladders 15. Ladder Design with Simulated Elements 16. Leapfrog Simulation of Ladders 17. Switched Capacitor Filters 18. Delay Equalization 19. Op-Amp Oscillators 20. Better Op-Amp Models APPENDIX: SCALING REFERENCES

300 citations


Journal ArticleDOI
TL;DR: In this paper, the basic ϵ-filter and its modifications to a trend-adaptive filter and a two-dimensional filter are described and the effectiveness of the new filter is demonstrated by computer simulation.
Abstract: This paper proposes an ϵ-separating nonlinear digital filter (called an ϵ-filter). This filter is intended for effective filtering of low-amplitude noise superposed on the signal with sharp discontinuities and can be realized by combining a simple nonlinear element with a conventional linear filter. In this paper, the basic ϵ-filter and its modifications to a trend-adaptive filter and a two-dimensional filter are described. The effectiveness of the new filter is demonstrated by computer simulation. Some of its application to EEG analysis, image processing and coding are also presented.

62 citations


Journal ArticleDOI
TL;DR: The modified truncated second-order nonlinear filter was shown to be the correct form of this filter provided a small correction is made in the discrete-time case in this paper.
Abstract: By rederiving the truncated second-order nonlinear filter, it is shown that the original derivations of this filter contain errors, or at least illogical approximations. The so-called modified truncated second-order nonlinear filter is, furthermore, shown to be the correct form of this filter provided a small correction is made in the discrete-time case.

40 citations


Patent
Hansen Jens Dipl Ing1
26 Jun 1982
TL;DR: In this article, a filter and demodulation circuit is proposed which, when used in a radio-frequency receiver, produces an increase in the sensitivity of the receiver by dividing the intermediate frequency into at least two parallel channels at the input.
Abstract: A filter and demodulation circuit is proposed which, when used in a radio-frequency receiver, produces an increase in the sensitivity of the receiver. In the filter and demodulation circuit, the intermediate frequency is divided into at least two parallel channels (11, 12) at the input (10). Each channel contains a series circuit comprising a mixing and oscillator circuit (13, 15; 14, 16), a controllable IF filter (17, 18), a demodulator (19, 20) and a high-pass filter (21) or low-pass filter (22). One transmission channel essentially transmits the modulation frequencies of a first frequency range only and the other transmission channel transmits the modulation frequencies of a second frequency range. The AF voltage at the output of each transmission channel re-adjusts the IF filter of this channel and the oscillator circuit of the mixing and oscillator circuit of the other channel.

22 citations


Journal ArticleDOI
TL;DR: In this paper, a low-passband insertion loss is achieved by large gap fin-lines and pure metal inserts mounted in the E-plane of rectangular waveguides requiring no supporting dielectrics, which combines the advantages of low-cost etching techniques and the low-loss performance of usual waveguide circuits.
Abstract: Low passband insertion-loss is achieved (1) by large-gap fin-lines, by which the high-Q potential increasing with gap-width is fully utilized, and (2) by pure metal inserts mounted in the E-plane of rectangular waveguides requiring no supporting dielectrics. This design combines the advantages of low-cost etching techniques and the low-loss performance of usual waveguide circuits. The theory described includes both higher-order mode interaction of the discontinuities and the finite thickness of dielectrics, metal fins as well as inserts. An optimizing computer program varies the filter parameters for a given number of resonators until the insertion loss yields a minimum in passband and an optimum in stopband. Data for optimized X-, Ka-, V-, E-, and W-band filters are given. Measurements verify the described theory. Measured minimum pass-band insertion losses are 0.3, 0.7, 1.5 dB for the fin-line filter for midband frequencies of about 12, 34, 75 GHz, and for the metal insert filter 0.1, 0.6, 0.5, and 0.7 dB at 12, 33, 63, and 76 GHz, respectively.

20 citations


Patent
22 Dec 1982
TL;DR: In this article, the antenna duplexer is made compact by a use of a SAW filter, and yet it eliminates the possibility that the SAW filters might be burned, and it avoids additional circuits, e.g., an impedance compensation circuit.
Abstract: An antenna duplexer is made compact by a use of a SAW filter, and yet it eliminates the possibility that the SAW filter might be burned, and it avoids additional circuits, e.g., an impedance compensation circuit. The antenna duplexer comprises a local oscillation filter. A reception filter is coupled to the local oscillation filter. The coupled side is partly constituted by a SAW filter. A transmission filter is coupled to the reception filter and an antenna is coupled between the reception filter and the transmission filter.

17 citations


Journal ArticleDOI
TL;DR: In this paper, the authors show that the unit noise gains of normal realizations of digital filters can be expressed in terms of a set of noise gain parameters that are simply related to the pole locations and pole residues.
Abstract: The unit noise gains for optimal and parallel normal realizations of digital filters can be expressed in terms of a set of noise gain parameters that are simply related to the pole locations and pole residues. These noise gain parameters are shown to be invariant under a class of frequency transformations, and for digital filter transfer functions derived by bilinear transformation of an analog transfer function, are independent of the frequency scaling parameter. As a result, the unit noise gains of normal realizations can be simply related to the performance characteristics of the filter, i.e., to filter order, passband ripple, and stopband gain. These simple relations make it easy for the filter designer to select a structure with acceptable roundbff error. Unit noise gain for normal realizations of Butterworth, Chebyshev, and elliptic filters are plotted for a range of performance characteristics, and compared with optimal state-space structures. These results show that there is no significant difference between the unit noise gains of optimal normal realizations and parallel normal realizations, and that the unit noise gains of optimal state-space structures are significantly lower than the normal forms only for high-order Butterworth filters.

14 citations


Patent
Henry Wurzburg1
13 Aug 1982
TL;DR: In this paper, a switched capacitor all-pass filter with linear magnitude and phase response is presented, and a transfer function which represents an all pass filter and which has a predetermined phase response are provided.
Abstract: A switched capacitor all pass filter which provides an output signal having substantially linear magnitude and phase response is provided. A transfer function which represents an all pass filter and which has a predetermined phase response is provided. A filter/phase equalizer structure having integrating operational amplifiers and various feedback portions which represent the transfer function is also provided.

9 citations


Journal ArticleDOI
TL;DR: The design of a bank of FIR bandpass filters for channel filtering and sample rate alteration in a 24-channel transmultiplexer is considered and it is shown that such a filter bank can be realized by the combination of a weighting network and a discrete cosine transform.
Abstract: The design of a bank of FIR bandpass filters for channel filtering and sample rate alteration in a 24-channel transmultiplexer is considered in this paper. It is shown that such a filter bank can be realized by the combination of a weighting network and a discrete cosine transform. A minimum phase design is suggested for the lowpass prototype filter in order to solve the long absolute delay Problem inherent in linear phase filters. Two approaches to the design of long minimum phase filters are disclosed; one employs frequency sampling design techniques, and the other makes use of the properties of the complex cepstrum of a minimum phase sequence. Filter specifications for the low-pass prototype are discussed and a design example is included. The hardware realization of the filter bank using a multiplier-accumulator as the arithmetic element is also discussed.

8 citations


Patent
03 Feb 1982
TL;DR: In this paper, the authors considered a band-pass filter with at least one resonant window whose shape determines the width of the pass band of the filter, and applied it to staggered circuit bandpass filters.
Abstract: Filter constituted, in a single cross-section (8) of the guide, by at least one resonant window (10) whose shape determines the width of the pass band of the filter. Application to staggered circuit band-pass filters.

7 citations


Patent
05 Oct 1982
TL;DR: In this paper, a signal processor consisting of a first low pass filter, the signal from which is fed to two first parallel programmable gain amplifiers controlled by a first digital function controller.
Abstract: A signal processor for use in association with a frequency analyzer, can operate in the frequency translation mode or the filter mode. The signal processor comprises a first low pass filter, the signal from which is fed to two first parallel programmable gain amplifiers controlled by a first digital function controller. The signals then pass via two, second low pass filters to two second programmable gain amplifiers controlled by a second digital function controller. The signals from the second amplifiers are then added by an adder and filtered by a third low pass filter. The first controller is controlled by a frequency signal equal to the frequency in the middle of the bandwidth being translated and the first filter is controlled by a frequency which is a multiple of the control frequency to the first controller. In the range translation mode, the second filters and second controller have a control signal equal to the bandwidth frequency and the third filter is controlled by a signal which is a multiple of the control signal to the second filter and the second controller. In the filter mode, the cntrol signals to the first and second controllers are equal to each other and the control signals to the first and third are equal to each other.

Patent
12 Jul 1982
TL;DR: In this article, the phase shifting circuit is formed by elements of a Gaussian type bandpass-ladder filter consisting of at least three sections each representing one pole, and the output is taken from the output of a filter section prior to the last filter section.
Abstract: For demodulating an FM-carrier which is, for example, modulated by a video signal and two sound subcarriers, a Gilbert mulitplier (13) and a phase shifting circuit (12) are used. The phase shifting circuit is formed by elements of a Gaussian type bandpass-ladder filter consisting of at least three sections each representing one pole. The output (10) of the phase shifting circuit is taken from the output of a filter section prior to the last filter section.


Patent
17 Aug 1982
TL;DR: In this paper, a Diplexer for electrical signals comprises a first frequency selective filter (22) coupled between a first input terminal (20A), and a circuit point (24), a second frequency selective filtering (26) coupled with the circuit point and an output terminal, and a third frequency selective filter(28) coupling between a second input node (20B) and the circuit node (24).
Abstract: A diplexer for electrical signals comprises a first frequen­ cy selective filter (22) coupled between a first input terminal (20A), and a circuit point (24), a second frequency selective filter (26) coupled between the circuit point (24) and an output terminal (20C), and third frequency selective filter (28) coupled between a second input terminal (20B) and the circuit point (24). The first filter (22) selects frequencies higher than a first frequency and the second filter (26) selects frequencies lower than a second frequency which is higher than the first frequency, The third filter (28) selects frequen­ cies lower than a third frequency which is lower than the first frequency. In a television receiver, the first and second filters (22,26) serve as a bandpass filter for the UHF band, and the third filter (28) passes the VHF and CATV (cable television) bands.

Journal ArticleDOI
TL;DR: For a digital Butterworth low-pass general- order filter, generalized expressions for the filter coefficients of the transfer function are derived, with representation in direct form as a single high-order filter, rather than as a combination of second-order and first-order subfilters.
Abstract: For a digital Butterworth low-pass general-order filter, generalized expressions for the filter coefficients of the transfer function are derived, with representation in direct form as a single high-order filter, rather than as a combination of second-order and first-order subfilters. By using double precision to overcome the usual problems of numerical accuracy with the direct form, the advantages of simplicity and economy of programming effort are realized.

Journal ArticleDOI
TL;DR: In this paper, the symmetric form filter has been described and its sensitivity characteristics, which shows that the sensitivity of center frequency with respect to the gain-bandwidth product (GB) is lower than that of the companion form 1 filter realisation.
Abstract: A new filter realisation known as the symmetric form filter has already becn described and its noise/linearity performance analysed. A study is made of its sensitivity characteristics, which shows that the sensitivity of centre frequency, ?p, with respect to the gain-bandwidth product (GB) is lower than that of the companion form 1 filter realisation.

Journal ArticleDOI
TL;DR: In this paper, a new SAW resonator filter on LiTaO3 (X-112°Y) substrate has been developed, which is a transducer coupling 3-poles filter with Butterworth response.
Abstract: A new SAW resonator filter on LiTaO3 (X-112°Y) substrate has been developed. The filter is a transducer coupling 3-poles filter with Butterworth response. It is constructed monolithically and requires no external elements. 3.5 dB minimum insertion loss, 5 10 kHz 3 dB-bandwidth, and 46 dB image signal rejection level are obtained at 280 MHz. This new filter, designed for a front-end filter, enables single super-heterodyne radio paging receivers to be offered.

Journal ArticleDOI

01 Jan 1982
TL;DR: The intention of the thesis is to present an overview of the con cepts of filter design along with two significant developments: a prehensive filter design computer program and the theoretical develop ment of an Nth order elliptic filter design procedure.
Abstract: AN ACTIVE FILTER DESIGN PROGRAM (theory and application) Author: Louis R. Gabello Advisor: Dr. Edward R. Salem This thesis deals with the design of filters in the frequency do main. The intention of the thesis is to present an overview of the con cepts of filter design along with two significant developments: a com prehensive filter design computer program and the theoretical develop ment of an Nth order elliptic filter design procedure. The overview is presented in a fashion which accents the filter design process. The topics discussed include defining the attenuation requirements, normalization, determining the poles and zeros, denormalization and implementation. For each of these topics the text ad dresses the fundamental filter types (low pass through band stop) . Within the topic of determining the poles and zeros, three classical approximations are discussed: the Butterworth, Chebyshev and elliptic function. The overview is concluded by illustrating selected methods of implementing the basic filter types using infinite gain multiple feed back (IGMF) active filters. The second major portion of the thesis discusses the structure, use and results of a computer program called FILTER. The program is very extensive and encompasses all the design processes developed within the thesis. The user of the program experiences an interactive session where the design of the filter is guided from parameter entries through the response evaluation and finally the determination of component

Journal ArticleDOI
TL;DR: In this article, a new method for realizing a first-order difference equation without operational amplifiers (op-amps) is proposed, which is applied to a low-pass filter using leap frog configuration.
Abstract: A new method for realizing a first-order difference equation without operational amplifiers (op-amps) is proposed. This circuit is applied to a low-pass filter using leap frog configuration. In order to remove the op-amp from the circuit, unique methods are introduced at the adding and subtracting portions in the circuit, As a result, using the first-order recursive circuit of which dynamic range per one-order is only 20 dB, the fourth-order low-pass filter with dynamic range of 40 dB is realized, extending from 100 Hz to 100 kHz of sampling frequency.

Journal ArticleDOI
TL;DR: In this paper, the feasibility of realising a high-order LC filter with a small set of different capacitor values, without sacrificing the frequency response specifications, is indicated, and this idea could be conveniently adopted in other filter structures also, for example the FDNR transformed filter realisations.
Abstract: The feasibility of realising a high-order LC filter with a small set of different capacitor values, without sacrificing the frequency response specifications, is indicated. This idea could be conveniently adopted in other filter structures also—for example the FDNR transformed filter realisations.

Journal ArticleDOI
TL;DR: In this article, a hardware implementation of a fourth and an eighth order filter, realized with one time-shared second-order section is described, and a software package is developed to simulate the filter performance, and validate the experimentally measured response.
Abstract: In vocoder systems speech is spectrally analysed with filter banks containing 16 to 19 narrowband filters. Digital techniques can be used to realize these filter banks in an economical manner by using single time multiplexed second order filters. A hardware implementation of a fourth and an eighth order filter, realized with one time-shared second order section is described. A software package is developed to simulate the filter performance, and validate the experimentally measured response. The CMOS design of a programmable filter is presented. A second order section is time multiplexed to realize up to 32 poles of filtering. The configuration is mask programmable and the filter coefficients are programmable with a UV-PROM. The bank of contiguous bandpass filters for vocoder use can be implemented with three multi-function filter chips.

Patent
22 Apr 1982
TL;DR: In this article, a test pulse compression filter is used with test pulse sequences carried out in order to operate a single sideband or double sideband data transmission process with suppressed carrier, which is used in media with linear distortion and frequency shift as a result of the Doppler effect.
Abstract: The pulse compression filter is used with test pulse sequences carried out in order to operate a single sideband or double sideband data transmission process with suppressed carrier. It is used in media with linear distortion and frequency shift as a result of the Doppler effect. The data are transmitted in the form of data blocks preceded by the test pulse sequence. The test pulse sequence is used to effect correction to the filter parameters and to continuously compensate the frequency in two orthogonal channels. The filter coefficients of the compression filter are equal to zero. They are derived from the bit values of the time inverse sequence associated with the pseudo noise sequence. A positive value of the time invese sequence produces a finite filter coefficient and a negative one produces a zero filter coefficient.

Journal ArticleDOI
TL;DR: In this paper, methods to design IIR digital filters with good attenuation and group delay characteristics are described, and applications of the methods to some common filter design problems are presented.

Proceedings ArticleDOI
T. T. N. Bucher1
01 Oct 1982
TL;DR: In this article, it was shown that if the angle modulated signal is random, but stationary in the wide sense, the power or energy spectrum is the convolution of the corresponding spectra of the angle-modulated wave and of the amplitude pulse.
Abstract: For signals used as Time Division Multiple Access and Slow Frequency Hopping A-J waveforms, it is desirable to be able to predict the spectrum occupancy of pulses containing a number of angle modulated symbols. It is shown that if the angle modulated signal is random, but stationary in the wide sense, the power or energy spectrum is the convolution of the corresponding spectra of the angle modulated wave and of the amplitude pulse. Spectrum occupancy can be obtained by integrating the resulting spectrum. Using a 9845S desk computer, spectrum occupancy has been estimated for FSK waves having peak to peak deviation ratios of 0.5 (MSK) and 0.7 (approximately optimum for limiter-discriminator). The modulating pulse may be a square wave of duration equal to from one to sixteen FSK symbols, or such a pulse passed through a Butterworth filter (with 2, 4 or 6 poles) or an ideal Gaussian filter. The filtered pulses have a one symbol rise time. It is concluded that square wave modulation should be avoided unless the duration is substantially longer than 16 bits. The simple 2-pole Butterworth filter confines the spectrum about as well as the ideal Gaussian, and much better than the square pulse. Pulsed FSK signals should be provided with a rise (and fall) time of the order of one modulation period. The amplitude variation should be continuous from zero to maximum, and probably the first derivative of the variation should also be continuous.



Journal ArticleDOI
TL;DR: The synthetic method of electrical passive networks is successfully applied to the development of an acoustic antialiasing filter for an electret condenser microphone that can be realized with only one acoustic resistance that makes the adjustment at the production easy.
Abstract: The synthetic method of electrical passive networks is successfully applied to the development of an acoustic antialiasing filter for an electret condenser microphone. The Butterworth type lowpass filter of order nine is realized as a tandem connection of an acoustic filter of order five and an electronic filter of order four which is imbeded in the microphone amplifier. The merit of this synthetic design is that the filter can be so realized with only one acoustic resistance that makes the adjustment at the production easy.