Showing papers on "Butterworth filter published in 1983"

Polyphase quadrature filters--A new subband coding technique

Abstract: A new method of implementing filter banks for subband coding of speech is presented. First, Esteban's Quadrature Mirror Filter principle is extended to allow the direct synthesis of filter banks with any number of equal size filter bands. Then, by combining the quadrature filter characteristic with the polyphase network implementation of filter banks, a new filter bank structure is obtained which requires 35% fewer computations than existing designs. In this paper the theory of operation of the Polyphase Quadrature Filter is presented and techniques for its efficient implementation are described. Then, examples of filter banks using this approach are shown and compared with other designs, and a simulation of a 16 Kbits/s coder using these filters is presented.

Segmented transversal filter

Abstract: A segmented digital transversal filter comprising a first transversal filter 4 that eliminates all frequency components from a signal sequence above half the subsampling frequency, a second transversal filter 11 that convolutes the output of the first filter with a decimated and energy compensated lower band of filter coefficients (HI 1 '), a third transversal filter 22 that convolutes the output of the first filter and a decimated and energy-compensated upper band of filter coefficients (HI 2 '), and a fourth transversal filter 32 that convolutes the original signal sequence and the central band of filter coefficients (HI 3 ). Delays 21 and 31 are added so that the outputs of the second, third and fourth filters arrive concurrently at an adder 51, the sum being the output of the invention.

Microwave-frequency response of an optical-fibre delay-line filter

Abstract: The frequency response of a single-mode fibre recirculating delay-line filter has been measured from 0 to 18 GHz using a broadband measurement system that includes an inteferometric waveguide modulator and a high-speed photodetector. An optical-fibre notch filter having a fundamental passband at 740 MHz exhibited 23 overtones that were uniform to within ±1.5 dB over the entire measurement range.

Adjacent-Channel and Quadrature-Channel Interference in Minimum Shift Keying

Abstract: In this paper we present formulas for the computation of error probability in the presence of quadrature-channel or adjacentchannel interference in addition to intersymbol interference in a minimum shift keying system. The filters in the receiver and transmitter are arbitrary but with a finite number of poles. The effect of phase jitter in the main channel, phase and symbol timing misalignment in the interfering channels, and sampling time jitter is taken into account. The probability of error is averaged over the phase and symbol timing misalignment. Numerical results are presented for Butterworth filters in the receiver and transmitter with two, three, and four poles. Curves of error probability as a function of various variables (signal-to-noise ratio, bandwidth of receiver and transmitter filters, number of poles, channel frequency separation, phase jitter, sampling time, and symbol timing and phase misalignment) are presented. The method of this paper can readily be applied to other filters; hence, it can be used in the design and prediction of the performance of digital communication systems.

Design data for digital Butterworth filters in geophysical data processing

Abstract: The analysis of geophysical data by computer often includes post-processing with digital filters. Of particular interest are those digital filters which approximate ideal low pass, high pass, band pass or band reject responses. Design data are presented for digital filters which approximate a Butterworth response. Low pass, high pass, band pass and band reject filters are discussed and it is shown that in general the nth order low pass Butterworth transfer function may be factorised with the denominator consisting of linear and/or quadratic factors. Simple frequency transformations permit the realisation of the associated high pass, band pass and band reject transfer functions for these factors. Application of the bilinear transformation yields the z transforms for these factors which, when used recursively, provide the desired filter response. Fortran computer programs are included and the performance limitations of the digital filters are illustrated by comparison with some ideal Butterworth responses.

High-pass filter of the first order and application thereof to telephony

Abstract: A sampled first order high pass filter having a configuration that eliminates the output offset voltage normally associated with known filters of this general type. It is constructed according to the diagram of the accompanying figure with two operational amplifiers, four capacitors and eight switches actuated at each period in two separate phases a and b. The filter is particularly useful in digital telephone transmission. The filter is placed just up-stream of an analog-digital converter sampled at the same frequency as the filter and the second amplifier serves as a comparator for conversion outside the phases a and b.

Mechanical bandpass filter design with lumped element prototypes

Abstract: A method is presented which provides means for the design of mechanical filters by the comparison of whole "pi" sections of the filter with the corresponding sections of a simple electric prototype filter. This replaces the well-known equivalences for each filter component and reduces the accumulation of eventual comparison errors. The results are the computation of broader passbands with more accuracy and more precise narrow-band filter design. This is achieved by first splitting the shunt arms of a mechanical filter ladder in two one-ports, providing a succession of symmetrical "pi" sections, formed by two half-resonators and a coupling element in between. The calculated chain matrices of these sections which are established including more or less mechanical phenomenons of the elements, compatible with the desired approximation and bandwidth. Second, a low-pass filter is synthesized with the imposed filter characteristics, using well-known insertion loss theory. This filter is put into a form which results in a succession of symmetrical "pi" sections as well. The elements of the chain matrices of the mechanical filter sections are compared with those of the transformed electrical prototype; the comparison of the matrix coefficients yield a frequency mapping function, applicable in the same way as in the design of electrical broadband filters. The degree of approximation for each bandwidth is immediately found by comparison of the plotted matrix coefficients. Extensional, torsional, and flexural vibrations are dealt with and some commonly used combinations are calculated with their basic matrices for demonstration purposes.

Universal multiple-output second-order digital filter

Abstract: A universal, second-order, digital-filter structure is proposed, which realises simultaneously a lowpass, a highpass, a bandpass, a notch and allpass transfer function. The structure uses two unit-delays and four multipliers, and is quite attractive for VLSI implementation since a versatile universal chip can be fabricated, which can be used for the implementation of Butterworth, Chebyshev and elliptic digital filters as well as digital equalisers.

Design and Fabrication of SAW Multipole Filters

Abstract: The design and fabrication techniques used to produce SAW 6-pole Butterworth filters at a center frequency of 217 mHz are discussed. The complex and stringent frequency requirements, better than 10 ppm, of coupling 6 individual poles t ogether requires simple yet accurate modeliny t echniques and a reproducible fabrication and trimming process. The approach used in this work is to use the two-pole acoustic section as the basic building block. The response of the two-pole section is then modeled on a microprocessor using a standard AC circuit analysis program. The results from this analysis are used in fabricating and combining the sections. Calculated and measured results are presented.

Efficient FIR filter implementation using microprocessor

Abstract: An efficient approach for direct form FIR filter implementation using microprocessor is presented. The approach uses the positive number and the coefficient partitioning techniques.

Computer generation of a slowly varying pseudorandom process

Abstract: The paper illustrates a computer modelling method for the generation of a highly correlated pseudorandom process with a desired power density spectrum. The samples of the pseudorandom process are generated as the output of a cascade connection of a correlation shaping transversal filter and a reconstruction filter with statistically independent samples fed to the input. Such a generation method needs a much smaller number of mathematical operations compared with the traditional transversal filtering method, and does not require any special or careful filter design for each process parameter, as is necessary in the application of an IIR filter.

A new independently adjustable filter using the operational amplifier pole

Abstract: A new second-order lowpass active filter, based on the single pole roll-off characteristics of the operational amplifier, with only one operational amplitier and a single external capacitor is presented. Independent control of centre frequency, frequency selectivity and passband gain by a single resistor is possible with this design. The filter has low sensitivity to all circuit active and passive parameters and is suitable for low-medium-high Q factor operations over a wide range of frequencies. The filter is suitable for integration.

Passive Arm Filters in I and Q Receivers for MSK-Type Continuous Phase Modulations

Abstract: The use of Butterworth and Chebyshev low-pass filters in suboptimum in-phase ( I ) and quadrature phase ( Q ) receivers is considered for several bandwidth efficient modulations. These modulations are: minimum shift keying (MSK), tamed frequency modulation with rectangular pulse shaping (TFMREC), and MSK with duobinary encoding of the source bits (DMSK). The sensitivity of the modulations with their I and Q receivers to a noisy phase reference and to timing errors is also investigated. It was found that standard, passive low-pass filter receive filters performed quite well when compared to other nonstandard filters. The use of standard filters should thus be considered in any system design. The modulations, with their simple receivers, can be ranked in order of increasing sensitivity to timing errors as follows: MSK, DMSK, TFMREC. This ranking is the reverse to the one obtained when it is done in order of best bandwidth efficiency.

New solution for the ideal filter approximation problem

Abstract: A new solution is proposed for the ideal filter approximation problem consisting of a cascade connection of a non-minimum-phase filter and a phase-correcting network which are synthesised simultaneously. As illustrated by numerical examples, the new solution is more economical than the conventional Cauer filter and phase corrector combination when comparison is made on the basis of the total number of elements used in practical realisation. It is also shown that filters with a monotonic passband response can be used to advantage over the filters with weighted Chebyshev approximation of the passband magnitude response.

Commutating filter passing only the fundamental frequency and odd harmonics thereof

Abstract: A commutating filter circuit producing only the fundamental frequency and odd harmonics of an input signal comprises first and second commutating filters each connected to receive the input signal and coupled to pass an output signal to the non-inverting and inverting terminals, respectively, of a differential amplifier. The first filter comprises N stages, each having a time constant of RC and a filter sampling frequency of f 1 . The second filter comprises N/2 stages, each having a time constant of 2RC and a filter sampling frequency of 2f 1 . The resultant signal from the differential amplifier contains only the fundamental frequencies and odd harmonics of the input signal.

Wideband Agile Transversal Filter

Abstract: The development of a Wideband Agile Transversal Filter is made possible by the use of dual gate GaAs FET transistor arrays. The transversal filter design has a bandwidth of greater than 2.0 GHz and the filter transfer function can be changed in less than 10 nanoseconds. The design has been configured in a four section block that can be cascaded to obtain multiple delay functions and thus very complex filter transfer functions.

Transmission filter with biquadratic filter stages.

Abstract: and of a second flter comprising a leapfrog arrangement of biquadratic filter stages the first and last ones of which are dissipative for reactive pseudo-power and each have a transfer function proportional to (Z -/+ 1) , the other biquadratic filter stages being non-dissipative for reactive pseudo-power. Various embodiments of the biquadratic filter stages are disclosed.

Perceptual identification and discrimination of phase distortions

Abstract: Experimental results on the audibility of phase distortion produced by minimum-phase 4kHz and 15 kHz anti-alias filters are reported. Numerous impulse response pairs with identical spectral magnitudes but linear-phase and minimum-phase responses were compared by five test subjects who listened dichotically on earphones. Group-delay distortion was doubled, progressively, until at least 67% mean correct discrimination was attained. At 4kHz, the phase distortion introduced by a cascade of two eighth-order Butterworth filter pairs was audible as was that from only a single pair of seventh-order elliptic filters. At 15kHz, the cascade of up to 4 pairs of seventh-order elliptic filters introduced no perceptible effects. Experimental details and several representative filter frequency and transient responses are included.

Inductive electrical signal filter with reduced impedance

Abstract: A translating circuit couples an electrical signal filter to a signal path conveying signals to be filtered. The translating circuit exhibits unity current gain and a voltage gain other than unity between the signal path and the filter, and applies signals from the signal path to the filter. By setting the scale factor (e.g., attenuation) of the translating circuit, the impedance of the filter can be set to any desired level without affecting the operating parameters of the signal path. In the case of an inductive filter, the translating circuit permits the filter to be designed with a lower impedance, and a smaller value inductor less susceptible to picking up interference signals likely to adversely affect signals in the signal path, without requiring a corresponding adjustment in the operating parameters of the signal path.

The noise gain of a digital butterworth ladder filter

Abstract: We show that any digital Butterworth ladder filter has a noise gain of N/2, where N is the filter order. The result is obtained by linking the noise gain to the average time delay of an analog Butterworth filter, which we show to be N/4E.

Demodulated chrominance signal filter with peaked response

Abstract: A filter network suitable for processing a signal such as the demodulated "I" chrominance component of a composite color television signal is disclosed The filter network comprises plural cascaded mutually interactive resonant LC sections for providing wideband filtering, amplitude peaking over a prescribed range of signal frequencies, and signal delay Mismatched resistors terminate the filter network and constitute the primary DC impedance of the filter network, and the LC sections exhibit selected impedances at frequencies within the peaking frequency range

Recursive ϵ‐nonlinear digital filter

Abstract: A recursive ϵ-nonlinear filter that removes random noise from signals with sharp discontinuities is proposed. It consists of a linear recursive low-pass filter with the addition of simple nonlinear fuctions. the proposed filter with a smaller number of taps is as effective as the nonrecursive ϵ-filter repored previously. Methodologies, realizations, and stability problems of the proposed filter are discussed. It is shown that the filter must be realized in cascade form for stable operation. Realizations of the multistage recursive ϵ-filter should be used from consideration of stability.

Optimization Of Target Detection With Analog Focal Plane Filters

Abstract: A mathematical model for the two -pole and four -pole analog focal plane filters is formu-lated which gives the filter pole locations in terms of the circuit parameters. The fre=cuency response and impulse response of the filters are computed, and the impulse responseis used to calculate the filter output for targets of various dwell times, assuming aGaussian model for the optical point spread function. The effects of varying the individualcomponent values in the four -pole filter upon pole locations are also considered.IntroductionThe analog focal -plane processor provides background suppression through temporal filter-ing without the need for digital data storage. This approach utilizes a passive RC filterconnected to each detector in the focal plane array. One hardware implementation of thisapproach uses what is known as "z- technology ". In order to evaluate the potential of thisapproach and to model its performance, a detailed analysis of typical RC filter designs isrequired. Such an analysis should address the following issues:Response to various targets,Response to noise,Clutter leakage, andEffects imposed by the circuit upon signal processing.This paper describes the target response analysis for two filter designs. The first ofthese is a two -pole configuration while the second filter has four poles.Analysis of each filter design begins with a derivation of the transfer function in termsof the R and C values. This transfer function is used to locate the filter poles, and thislatter information can be used to determine the filter frequency response and impulse re-

On qualifying windows for FIR filter design

Abstract: Although several measures of window performance have been defined, none of these directly characterizes the window's application to FIR filter design. But as the number of filter coefficients tends to infinity the peak ripple in the filter response tends to a value that is dependent only on the window. This value, the "Gibbs number," closely approximates the peak ripple for practical filters.

A new method for the design of 2-dimensional stable recursive digital filters satisfying prescribed magnitude and group delay response

Abstract: This paper presents a technique for the design of 2-Dimensional (2-D) stable recursive digital filters satisfying prescribed magnitude and constant group delay response. This technique uses the properties of positive definite matrices and their application in generating 2-variable Very Strictly Hurwitz Polynomials (VSHP) which will be assigned to the denominator of a 2-D analog reference filter. Bi-linear transformations is then applied to the transfer function of the 2-D analog reference filter to obtain the discrete version of the 2-D filter. Parameters of the discrete 2-D filter can be used as the variables of optimization to minimize the least mean square error of the desired and designed magnitude and group delay response of the filter. This technique is illustrated by examples.

An LC simulation active RC low‐pass notch filter by minimum number of capacitors

Abstract: The most important measure in judging the advantages or disadvantages of a synthesis method for an active RC filter is the element sensitivity. One of the powerful methods for minimizing element sensitivity is the well-known leapfrog-type synthesis which simulates a doubly terminated LC filter. In this paper, a method for the synthesis of an LC simulation active RC filter is proposed which can realize a low-pass notch filter with a minimum number of capacitors. First, how the signs are taken for the node voltages and inductor currents so as to minimize not only the number of capacitors but also the number of operational amplifiers is described; voltage and current relations are derived; and a circuit for realizing these relations and a method for synthesizing an odd-order filter are shown. Next, the effects of the variation of RC elements and the GB product of the operational amplifiers are described. Finally, a fifthorder low-pass notch filter is designed and an experiment is carried out. Furthermore, sensitivity calculation is performed using a computer and it is shown that the element sensitivity of this circuit is about 80% that of a circuit realized by conventional leapfrog synthesis.