Showing papers on "Butterworth filter published in 1984"

Multirate filter designs using comb filters

Abstract: Results on multistage multirate digital filter design indicate most of the stages can be designed to control aliasing with only slight regard for the passband which is controlled by a single stage compensator. Because of this, the aliasing controlling stages can be made very simple. This paper considers comb filter structures for decimators and interpolators in multistage structures. Design procedures are developed and examples shown that have a very low multiplication rate, very few filter coefficients, low storage requirements, and a simple structure.

An adaptative, nonlinear edge-preserving filter

Abstract: An adaptative filter whose main feature is to preserve edges and impulses present in the signal is analyzed by the computation of the mean-square error (MSE) of its output sequence. The filter in its more general form is highly nonlinear, resembling the M-type estimators used in robust statistics. A simplified form used here allows the exact computation of the MSE when the filter length is finite. This MSE can be compared to the ones obtained for a median filter and a mean filter. It is shown that for a wide range of the filter and signal parameters such as filter length, edge heights, and impulse width, the performance of the filter proposed in this paper is superior to the other filters mentioned above. An additional advantage of the simplified version of the filter is that in most cases, its computation amounts to a linear adaptative averaging. This contrasts with the amount of calculation required to implement the median filter and any other filter based on the order statistics of the measured samples.

Directional hearing aid

Abstract: A pair of sensitive microphones or transducers are mounted on a user's body, spaced apart by a distance equal to one-half a wavelength of the center frequency of a range of frequencies to be emphasized. By summing the outputs of the two microphones, sound in the broadside or look direction (i.e., the direction the listener faces, the microphones being on a line perpendicular to this direction) are emphasized; sounds in the endfire or side directions are nulled or produce a substantially null response in the region of the center frequency defined by the microphone spacing. A third microphone may be added that is not equally spaced from the microphones on either side, but is spaced to provide half wavelength distances which define maximum and null responses centered at the other points within the frequency range (1-4 KHz) desirable for highly effective hearing. The summed signal from each mircophone pair is bandpass filtered. Three bandpass filters are used. The centers of their pass bands are 1200 Hz, 2250 Hz, and 3600 Hz, respectively. Thus each microphone pair and associated bandpass filter is responsible for providing a directonal receiving capability in its assigned range of frequencies. The frequency ranges are contiguous and overlap slightly. The final output is obtained by summing and amplifying the bandpass filter outputs. A good bandpass filter design is a fourth order Butterworth filter, whose center frequency can be designed to be: ##EQU1##

Embedding Data Into Pictures by Modulo Masking

Abstract: Three systems are proposed for embedding data into industrial quality monochrome analog pictures. The video signal on each scan line is sampled, and a data bit is inserted into a block of three or five pels by modulo masking scrambling the luminance level of only one pel in the block. Prior to transmission, the combined data and video sequence is converted into a continuous signal with a bandwidth that is no greater than that of the original video signal. Using six images each containing 65 536 pels, Systems 1 and 2 embedded an average of 17 430 and 8713 bits per image, while System 3 accommodated data at a constant rate of 21 760 bits/image. The data embedding procedures of Systems 1, 2, and 3 operated with average picture SNR's of 41, 44, and 30 dB, respectively, when the transmission channel was ideal. When the transmission was over a channel composed of a second-order Butterworth filter plus additive noise that yield a channel SNR of 40 dB, no bit errors occurred but System 3 offered the greater safety margin to bit errors than Systems 1 and 2.

Adaptive acoustooptic filter

Abstract: A new adaptive filter utilizing acoustooptic devices in a space integrating architecture is described. Two configurations are presented; one of them, suitable for signal estimation, is shown to approximate the Wiener filter, while the other, suitable for detection, is shown to approximate the matched filter.

Analog RC active filter

Abstract: An RC active filter device which is implemented with integrated circuit technology. Each RC filter element of the device has a distributed series resistance and a distributed shunt capacitance. The series resistance is formed by a strip of resistive material which overlies a resistive plate (or substrate) with an intervening insulating layer to form the shunt capacitance. The filter device can be designed to have a pass-band ripple response which is not affected by variation in nominal absolute resistance and capacitance values resulting from process spreads. These variations only stretch or compress the filter response along the frequency axis. The filter device comprises three RC filter elements and an operational amplifier. The RC filter elements nearest the operational amplifier determine a low pass band, while the input filter element prevents high frequency components outside the low pass band being leaked directly to the filter device output.

Constant Z bandswitched input filter

Abstract: A constant impedance (Z) bandswitched input filter for use in a wideband signal receiver includes a low pass filter, a high pass filter and a bandpass filter for reducing intermodulation distortion as well as for providing a flat input impedance characteristic. The bandpass filter resonates at the crossover frequency between the low and high pass filters for minimizing insertion loss while maintaining a constant termination impedance across the entire input signal bandwidth to reduce the input standing wave ratio (VSWR). Switching means are provided for coupling the high and low pass filters as well as the bandpass filter in circuit depending upon the frequency band tuned to. The constant Z bandswitched input filter is particularly adapted for use in a wide band cable television (CATV) system for substantially reducing intermodulation distortion and signal reflections arising from impedance mismatches, but has application in any cable operated multichannel communications system.

Dielectric Resonators Band Pass Filter with High Attenuation Rate.

Abstract: A bandpass filter with high attenuation cut off has been realized by cascading a bandpass filter containing dielectric resonators and a two dielectric resonators bandstop filter. The filter response has a higher cut off attenuation rate [26 db/10 MHz) compared to the original bandpass filter (14 db/10 MHz). The attenuation characteristics and return loss response show that the bandpass and the center frequency remain inchanged at 40 MHz and 4.015GHz respectively. The insertion losses are low (1db).

Evanescent mode triple ridge lowpass harmonic filter

Abstract: A waveguide lowpass filter has successive groups of three separate ridges spaced longitudinally in said filter. A center ridge of each filter has a larger cross-sectional area than two side ridges which are the same size. All ridges are parallel to one another in a longitudinal direction. The groups are associated with shunt capacitances and the spaces between adjacent groups are associated with series inductances in an evanescent mode. In operation, the filter is capable of supporting a TE10 mode in the passband and three modes, TE10, TE20 and TE30 in the stopband. The filter has a relatively high power handling capability as compared to previous evanescent mode lowpass harmonic filters.

Direct synthesis of filter transfer functions

Abstract: In this paper we propose a new procedure for the direct synthesis of complex transfer functions for a new class of selective filter. Conditional mean square approximation with Chebyshev weights, with three Lagrangian multipliers (for transfer-function values at cutoff frequency, at the origin (ω = 0) and for the phase at the cutoff frequency) gives a new class of filter derived in explicit form. By the choice of the transfer-function phase we can control the shape of amplitude characteristic in the passband or the filter selectivity.

Switchable bandpass filter

Abstract: A bandpass filter suitable for use in modem circuitry and other applications which require the filter's center frequency to be switched is disclosed. The filter circuit has the characteristic that although the center frequency is changed the overall gain of the filter remains constant. This characteristic results from using a plurality of filter sections which each have low individual gain and providing compensating gain at the input of the filter. In addition, compensating gain elements are used in each filter section so that when the center frequency of the filter is changed, the gain of each filter stage remains constant.

Self-Tuning Filter for Instrumentation

Abstract: A microprocessor-controlled filter that tunes automatically to the fundamental of the input signal is described. Hardware and software implementation are shown, and the error in maximum amplitude selection is discussed. Filter performance using a switched capacitor tunable circuit is documented.

Integrated microwave filter and method of constructing same

Abstract: Cascaded dual-mode resonance cavities are separated by plates having each a cruciform iris therein, the arms of which have lengths determined by a direct and precise procedure starting from a conventional Butterworth prototype filter. The transfer function parameters are altered to change the filter response and the group delay performance is determined and compared to an ideal flat response for producing a penalty function signal from the difference in order to modify the parameters again in such direction that the penalty function signal is likely to be reduced. The process is repeated until the penalty function signal is minimum, whereby a compact structure is realized which provides an attenuation response having sharp cut-off slopes at the edges of the signal bandwidth together with a close to flat in-band group delay response.

An Integrated 18th Order VIS-SC Pseudo-N-Path Filter

Abstract: Experimental results of an 18th order integrated bandpass filter in VIS-SC technique will be presented. This filter has been designed for use as "channel filter" e.g. in a PCM/FDM transmultiplexer.

Design of recursive digital filters for the selection or rejection of a particular velocity in seismic signals

Abstract: In this paper a new type of “velocity-selecting/rejecting filter” which passes or stops a particular event in a seismic signal is proposed. The velocity-selecting filter is based on a time-space band-pass filter with sharp passband for a particular direction, and similarly, the velocity-rejecting filter is based on a time-space band-stop filter. A technique for designing such filters, in terms of an infinite-impulse-response (IIR) filter, is presented, in which a rotated version of separable filter is used. Finally, numerical examples are included to illustrate the design theory.

Electrical filter comprising resonant circuits.

Abstract: A filter, such as a bandpass filter for use at intermediate frequencies in a radio transmission system, includes at least one filter stage comprising a common-emitter transistor (1) having resonant circuits (4, 5) in both its collector and emitter circuits. The ratio of the impedances of the resonant circuits (4, 5) determines the frequency response of the filter. The resonant circuits (4, 5) are isolated from one another by the transistor (1) so that they can be individually tuned. The presence of the transistor (1) also enables the input return loss of the filter to be determined independently of the tuning of the resonant circuits (4, 5). Individual filter stages can be cascaded.

FIR low-pass filter design using parametric filter technique

Abstract: This paper describes a new design method for finite impulse response (FIR) filters. FIR low-pass filters which have quasi-equiripple characteristics in the stopband and arbitrary ones in the passband are designed. Applying the parametric filter design technique, the stopband zeros are given in the closed form. Hence, only passband zeros are iteratively approximated to shape passband characteristics. This method can be applied for both the minimum and the linear phase cases.

ARMA digital lattice filter based on a linear prediction theory

Abstract: This paper proposes some elementary sections of an ARMA lattice filter. an ARMA lattice filter with an arbitrary AR order and an arbitrary MA order is presented. A conventional ARMA lattice filter has already been constructed by a method used to develop an AR lattice filter to a 2-dimensional filter. Therefore the conventional filter is not suitable for a filter realization when an AR order and an MA order differ from each other, and for the design of a minimum realized ARMA model. the most important characteristic of lattice filters is an orthogonality between every prediction error calculated at each elementary section. Employing this characteristic, we employ the AR lattice filter in a speech analysis, synthesis, an equalizer, and so on. It is shown that the proposed ARMA lattice filter satisfies an orthogonality. Furthermore, the ARMA lattice inverse filter which is an analyzer for an observed waveform and the ARMA lattice filter which is a synthesizer for the waveform are presented.

Analogue wave filter device

Abstract: An RC active filter device which is implemeted with integrated circuit technology. Each RC filter element of the device has a distributed series resistance and a distributed shunt capacitance. The series resistance is formed by a strip of resistive material which overlies a resistive plate (or substrate) with an intervening insulating layer to form the shunt capacitance. The filter device can be designed to have a pass-band ripple response which is not affected by variation in nominal absolute resistance and capacitance values resulting from process spreads. These variations only stretch or compress the filter response along the frequency axis. The filter device shown in Figure 2 comprises three RC filter elements 1, 2 and 11, and an operational amplifier 3. The RC filter elements 1 and 2 determine a low pass band. The filter element 11 prevents high frequency components outside the low pass band being leaked directly to the filter device output.

High-frequency GaAs transversal filter

Abstract: A transversal filter integrated circuit, fabricated using 1 μm GaAs MESFET technology, is described. The filter uses 280 MESFETs, has a clocked delay line and on-chip tap weighting to give three different filter responses. Operation has been demonstrated up to a clock frequency of 400 MHz, which is believed to be the fastest reported operation of a clocked transversal filter.

Group Carrier Supply Ceramic Filter

Abstract: A group carrier supply ceramic filter for use in an FDM carrier system was examined. For the filter, a ladder filter configuration composed of nine wire-mounted ceramic disk resonators and eight multilayer ceramic capacitors was adopted. Because of refinement of the filter design, only rough adjustment was fully sufficient for the frequency adjustment on the resonator. The filter exhibited analogous characteristics designed according to image parameter theory; both its temperature and aging characteristics were satisfactory. The stopband attenuation was more than 100 dB, due to suppression of spurious vibration. The filter size was reduced to 1/2 that of the usual quartz crystal filter.

DPSK signaling over hard limiting channel in the presence of intersymbol interference

Abstract: This paper considers the performance evaluation of binary differentially-phase-shift-keyed signals which have been transmitted through a hard limiting channel in the presence of intersymbol interference and additive Gaussian noise. An expression for the probability of error is derived which is in the form of an infinite series. In the analysis, the effects of Gaussian noise in the uplink and downlink are taken into consideration. The intersymbol interference is assumed to be present in the uplink only. In order to illustrate the applicability of the method, numerical results are computed for a channel which employs a Butterworth filter.

An adaptive digital filter

Abstract: An adaptive digital filter which is used, for instance, for a prediction filter in an ADPCM modulator and an ADPCM demodulator, for providing stable operation and minimum phase shift response has been found. The filter (Fig. 4, Fig. 5) has three parallel branches (A, B, C), two (A, B) of them have a plurality of series connected non-recursive filter elements (A1-An, B1-Bn) each of which has degree not larger than three relating to an operator (Z―1), and third branch (C) is merely a conductive line. Tap coefficients (c1, c2, ...., d1, d2, ....) of non-recursive elements change according to solutions of the transfer function of the filter. A numerator and/or a denominator of the transfer function of the present filter is a Chebychev polynominal relating to an operator (Z―1), and has a zero point and/or a pole. The solutions (wi, vi) of a numerator and a denominator locate alternately on a unit circle (Fig. 6) on a Z-1 plane. When solutions of the transfer function do not locate alternately on said unit circle, tap coefficients (c1, c2, ...., d1, d2, ....) are not updated, since non-alternate solutions do not assure stable operation and/or minimum phase shift response of the filter.

Design and Analysis of Linear Predictor-Corrector Digital Filters

Abstract: In application, it is believed that the third order linear predictor-corrector digital filter, or commonly known as the α-β-, tracking filter. can be derived from the Kaiman filter. In this paper, we characterize the values of α,β, and , so that theα-β-, tracking filter is indeed a limiting Kaiman filter, and derive the input to observation noise ratios in terms of the stochastic parameters as functions of α,β and , to allow the user to design the α-β-, filter to obtain near-optimal performancez-transform is used to uncouple the filter and to study stability.

ACSL (Advanced Continuous Simulation Language) Simulation of a Third Order Butterworth Filter Using Implementations Derived From Both Z and S Domain Representations.

Abstract: : This report describes the implementation of a Third Order Butterworth filter in the ACSL Observations and comparisons are made with inputs and outputs Both plots of the S and Z transfer functions were made with an ACSL program The program and Bode plots are both described in this report (Author)

A bit serial LDI recursive digital filter

Abstract: The practical implementation of a bit serial lossless discrete integrator (LDI)[1] recursive ladder filter suitable for implementation as a single integrated circuit is described. The low coefficient sensitivity and simplicity of the LDI signal flowgraph make the filter structure suitable for implementing high quality digital filters. A bit serial integrated circuit filter has been designed and simulated using the silicon compiler system FIRST[2,3]. The LDI filter coefficients which yield a Chebyscheff transfer function characteristic are found using an exact synthesis method[4]. The finite precision time domain response of the filter is simulated using the FIRST simulator and the magnitude response is verified by calculating the fourier transform of the filter unit sample response. The LDI filter implementation makes use of an alternate clocking scheme which simplifies the signal flow graph. This is a form of multiplexing which is easily implemented using bit serial arithmetic.

Digital linear-phase low-pass filter

Abstract: The low-pass filter indicated contains delay elements, multipliers and summers. The delay line contains two delay components, balanced with respect to the centre tap, with the delay time T and, in the more distant area, delay components with the delay time 2T. The low-pass filter comprises the cascade connection of a transverse filter and an attenuation-free resonator. The digital multipliers can be designed in a manner known per se by means of shift circuits. The special circuit provides a digital broadband low-pass filter in which the pass band and stop band are equally broad. It can be used in particular in the transmission of television signals.