Showing papers on "Butterworth filter published in 1985"

Explicit formulas for lattice wave digital filters

Abstract: Explicit formulas are derived for designing lattice wave digital filters of the most common filter types, for Butterworth, Chebyshev, inverse Chebyshev, and Cauer parameter (elliptic) filter responses. Using these formulas a direct top down design method is obtained and most of the practical design problems can be solved without special knowledge of filter synthesis methods. Since the formulas are simple enough also in the case of elliptic filters, the design process is sufficiently simple to serve as basis in the first part (filter design from specs to algorithm) of silicon compilers or applied to high level programmable digital signal processors.

Low-distortion switched-capacitor filter design techniques

Abstract: The distortion mechanism in switched-capacitor (SC) filters are considered, and closed-form expression relating switched-capacitors filter distortion to circuit parameters are derived. Design techniques for low-distortion applications are discussed and are applied to a sixth-order experimental filter. The filter design uses a fully differential class A/B op amp with a continuous-time common-mode feedback circuit. Distortion measurements show that for 82-dB dynamic range (relative to noise floor) the total harmonic distortion of 0.02% within the whole 4-kHz bandwidth and 0.07% within 20-kHz bandwidth.

An elliptic continuous-time CMOS filter with on-chip automatic tuning

Abstract: A voice-band continuous-time filter is described which was designed based on the technique of fully balanced networks and was fabrication in a 3.5-/spl mu/ CMOS technology. The filter implements a fifth-order elliptic low-pass transfer function with 0.05-dB passband ripple and 3.4 kHz cutoff frequency. A phase-locked loop control system fabricated on the same chip automatically references the frequency response of the filter to an external fixed clock frequency. The cutoff frequency was found to vary by less than 0.1% for an operating temperature range of 0-85/spl deg/C. The absolute value accuracy of the cutoff frequency was 0.5% (standard deviation). With /spl plusmn/5-V power supplies the measured dynamic range of the filter was approximately 100 dB.

Quadrature mirror filter design for an arbitrary number of equal bandwidth channels

Abstract: The two-channel quadrature mirror filter structure of Croisier and Esteban may be extended to an arbitrary number of equal bandwidth channels, given certain restrictions on the bandpass filters. The most serious restriction is that the stopband attenuation of eacli band-pass filter must be high for all frequencies outside twice the nominal 3 dB bandwidth of the filter. This restriction is not really a limiting factor for speech subband waveform coding since high adjacent channel attenuation is a necessity for the confinement of quantization noise. A key feature of our filter structure is that the number of multiplies, adds, and stored coefficients required for implementation is significantly less than those needed for the conventional QMF structure, given the same number of channels. Fortran code for a 16-channel filter structure is listed as an example of efficient implementation.

On prefilters for digital FIR filter design

Abstract: A new family of digital prefilter structures is introduced, based on the Dolph-Chebyshev function These prefilters can be combined with appropriately designed "equalizer" filters based on equiripple methods, leading to efficient FIR digital filter designs Design examples are included, demonstrating the simplicity of the resulting designs, as compared to conventional equiripple designs

Switched bandpass filter

Abstract: A switched bandpass filter includes a plurality of triple tuned bandpass filter sections coupled in parallel between an input and an output terminal and complemented by input and output parallel resonant circuits. Each bandpass filter section includes a switch such as a switching diode by means of which the filter section is either rendered active for passing or inactive for suppressing the frequency band to which it is tuned. Portions of the inactive bandpass filter sections contribute to the resulting frequency response with notches at frequencies outside of the passband of the active filter section. Thus, some components of the inactive bandpass filter sections are reused with the active filter section to gain steep out-of-band rejection slopes. In one embodiment, the switched bandpass filter employs only one switching diode per frequency band and, by reusing two-thirds of the filter components, provides a highly efficient switched bandpass filter arrangement.

The discrete prolate spheroidal filter as a digital signal processing tool

Abstract: The discrete prolate spheroidal (DPS) filter is one of the class of nonrecursive finite impulse response (FIR) filters. The DPS filter, first introduced by Tufts and Francis [1], is superior to other filters in this class in that it has maximum energy concentration in the frequency passband and minimum ringing in the time domain. Slepian [2] gives a complete discussion of DPS function properties. We give a mathematical development of the DPS filter properties, provide information required to easily and accurately construct even many coefficient filters, and compare the properties of this filter with those of the more commonly used filters of the same class. We note that use of the DPS filter allows for particularly useful statements of data time/frequency resolution "cell" values and that overall it forms an especially useful, though little known tool for digital signal processing.

Waveguide manifold coupled multiplexer with triple mode filters

Abstract: A multiplexer has a plurality of bandpass filters coupled through E-plane or H-plane T-junctions to a waveguide manifold. Where the multiplexer has four channels and each filter is a six-pole filter, two triple mode cavities make up each filter. Where each filter is a five-pole filter, one triple cavity and one dual mode cavity makes up each filter. Two band edge channel filters are operated to produce an asymmetrical filter function response, thereby causing extra transmission zeros to be created and improving the selectivity of the filter out of the passband. The multiplexer is designed for use in satellite communication systems and can have a reduced volume and weight when compared to previous multiplexers without any sacrifice in electrical performance.

Numerical experiments with a new differentiation filter

Abstract: A dynamic programming filter which provides estimates of the first and second derivative of empirical displacement data is investigated numerically. This filter uses a weighted least squares criteria in estimating the derivatives. The filter equations are presented together with several numerical examples. These examples are taken from references that proposed other techniques.

Nonlinear adaptive filter

Abstract: A nonlinear adaptive filter has an input and an output, and includes an adjustable linear filter and nonlinear filter connected serially between the input and output. The parameters for the adjustable linear filter are derived from signals on the adaptive filter output and from signals on the adaptive filter input. Preferably, the adjustable linear filter is coupled between the adaptive filter input and the nonlinear filter. A parameter computer for adjusting the parameters of the adjustable linear filter includes an input coupled to the output of the nonlinear filter, a processor for processing a signal received from the output of the nonlinear filter, and apparatus for applying the processed output signal to the parameters of the adjustable linear filter. A linear filter for shaping the spectrum of a signal received from the output of the nonlinear filter is preferably coupled between the nonlinear filter output and the parameter computer. In its preferred form, the nonlinear filter portion includes a first processor for generating from an input signal an intermediate signal having a uniform amplitude-density function, and a second processor for generating from the intermediate signal an output signal having a desired amplitude-density function.

Switched capacitor filter utilizing a differential input and output circuit

Abstract: An integrated circuit for filtering signals by having cascaded switched capacitor sampling filters. The circuit includes a transmit section which has an anti-aliasing filter, a core section filter, a highpass filter and an encoder for providing analog-to-digital conversion. Each successive filter is sampled at a lower rate to inhibit anti-aliasing. The circuit also includes a receive section which has a digital-to-analog decoder, an output buffer, a receiver core filter and a power amplifier.

A multi-output second-order digital filter structure for VLSI implementation

Abstract: A multi-output second-order digital filter structure without zero-input and constant-input oscillations is presented. It could be used for the realization of low-pass, high-pass, band-pass, and band-stop Butterworth, Chebyshev, and elliptic digital filters, and all-pass digital filters. The overall structure consists of two delays, three multipliers, and nine adders, which could be adopted for VLSI implementation.

Mixer controlled variable passband finite impulse response filter

Abstract: A mixer controlled, variable passband, finite impulse response filter has a plurality of individual filter function blocks which are cascaded. The filter function of each of the individual filter function blocks is represented by a fixed filter function block having a linear phase response. The junctions between successive filter function blocks form signal taps. The signal from each signal tap is delayed by a compensating delay device so that the signals from each of the plurality of signal taps have the same phase function. Signals from selected signal taps are then combined in a predetermined manner to achieve a desired filter transfer function.

An integrated six-path wave-SC filter

Abstract: Experimental results of a switched capacitor (SC) N-path filter integrated in a CMOS Si-gate technology are reported, with N=6. The circuit is based on the theory of wave-flow networks and uses only fully stray-eliminating (usually referred to as stray-insensitive) SC amplifier and integrator circuits. The two main drawbacks of N-path filters, i.e. unwanted mirror frequencies due to path mismatch and clock feedthrough located in the passbands, as solved by multiplexing large filter parts and by using a third-order high-pass reference filter, respectively. The integrated six-path filter features low power consumption due to the use of dynamic amplifiers.

A single-path frequency-translated switched- capacitor bandpass filter system

Abstract: Conventional switched-capacitor (SC) filter systems have alias and image frequency-translated signal components which are regarded as undesirable and supressed by a low-pass anti-aliasing filter (AAF) and by a low-pass anti-imaging filter (AIF). In this paper, we propose a single-path frequency-translated filter system in which these frequency translations are deliberately utilized in order to realize bandpass responses with very narrow relative bandwidths. The system requires bandpass AAF and AIF which are implemented using SC techniques.

Predictive coding for FIR filter wordlength reduction

Abstract: A new method of reducing the coefficient wordlength of an FIR filter by utilizing the autocorrelation property of the impulse response is presented. The desired FIR filter is realized as a cascade of a low order all-pole filter modeling the desired FIR filter and another FIR filter representing the modeling error. This technique can also be used simultaneously with other wordlength reduction techniques to enhance the wordlength reduction capability. The resulting filter may be implemented as a parallel combination of a limited number of filters whose coefficient values are - 1, 0 , or + 1 . The coefficient values can also be made equal to powers-of-two.

System for processing audio frequency information for frequency modulation

Abstract: In FM transmitters, particularly land mobile FM transmitters, wherein a speech signal is pre-emphasized between 300 Hz and 3 kHz, whereafter it is filtered to provide a sharp cut-off and wherein the maximum peak frequency deviation is limited in order to maintain channel integrity, the level of undistorted speech can be raised to within 90% of the transmitter peak deviation by directly connecting an amplitude limiter to a filter amplifier which behaves as a level sensitive filter when the speech signal has been limited and prevents ringing and overshoot from occurring. The output from the filter amplifier is coupled to a low pass filter, typically a fourth-order Butterworth filter, via an attenuator which reduces the amplitude of the signal to prevent the onset of further clipping.

Odd order elliptic waveguide cavity filters

Abstract: An odd order bandpass filter has at least one cavity resonating at its resonant frequency in three independent orthogonal modes. The filter has at least one feedback coupling that is made to resonate and change sign at a center frequency. When the filter has two cavities, one being a triple cavity and the other being a dual mode cavity, the filter can be operated to achieve an elliptic function response. Also, the filter of the present invention can achieve a weight and volume reduction when compared to six-pole dual mode filters.

Very low sensitivity FIR filter implementation using 'structural passivity' concept

Abstract: The concept of "structurally bounded" or "structurally passive" FIR filter implementation is introduced, as a means of achieving very low passband sensitivities. The resulting filter structures, called FIRBR structures, can easily be transformed into very low-sensitivity "passive" two-dimensional FIR filter structures. From a layout point of view, the new structures are not any more complicated than the well-known cascade form. The FIRBR structures do not depend, for synthesis, upon continuous-time filter circuits.

A multi-output second-order digital filter without limit cycle oscillations

Abstract: A multi-output second-order digital filter without zero-input, constant-input, and forced overflow oscillations is presented. The filter structure can be used for the simultaneous realization of low-pass, high-pass, band-pass, band-stop, all-pass, and general biquadratic transfer functions using two delays, four multipliers, and eleven adders.

Effect of Narrow-Band Filters on the Error Probability of M-ary FSK with a Limiter-Discriminator-Integrator Detector

Abstract: We derive an equation for the error probability of M -ary frequency shift keying with a limiter-discriminator-integrator detector in the presence of narrow-band filters in the transmitter and receiver. We present numerical results for the case of quaternary symbols, wide-band transmitter filter, and Butterworth filter in the receiver with 2, 3, and 4 poles. We optimize the sampling time and threshold setting for various values of filter bandwidth. We show the sensitivity of the error probability to errors in sampling time and threshold setting.

Coaxial-lead filter, comb-lead filter or interdigital filter having at least four resonators

Abstract: The specified filter circuit consists of at least four resonators which, on the one hand, are coupled directly to one another and in which, on the other hand, design measures are provided by means of which a bridging coupling is achieved. Using such bridging couplings, attenuation poles can be produced in the filter characteristic. It must be possible to set the frequency position of such attenuation poles relatively precisely, in particular also in the case of filters having a relatively large number of resonators. To this end, an additional coupling device, which acts as a concentrated capacitor, is provided in at least one direct coupling.

frequency selective filter circuit

Abstract: A time domain filter having ideal high pass and low pass characteristics is provided by means of a one-shot monostable multivibrator and a flip-flop interconnected by means of an AND gate and an EXCLUSIVE-OR gate wherein the set time for the one-shot is equal to the half-period of the cut-off frequency. The time domain bandpass filter circuit having ideal bandpass filter characteristics comprises a pair of one-shot monostable multivibrators connected in cascade with each other and interconnected with a flip-flop by means of an AND gate. A further application of a modified time domain filter is that of time domain FM demodulator having an optional phase-locked loop.

Recursive radar clutter filter

Abstract: A recursive radar filter arranged to provide filtering of ground clutter signals from radar echoes in the form of short batches of pulses. The filter includes a feed-forward section with delay, adder, and fixed amplifier elements. The filter also includes a feedback section which includes similar elements. Two time-varying amplifiers are included in the filter which change gain as the pulses are applied to the input of the filter. In a three delay version of the filter, the gain of one of the time-varying amplifiers is such that the feed-forward section of the filter does not pass any pulses until the fourth pulse has been applied to the filter. After the third pulse, the gain of this time-varying amplifier increases on each successive input pulse. In another embodiment, the two time-varying amplifiers are eliminated by making all of the fixed gain amplifiers time-varying in relationship with the interpulse period of the pulse input signal. Both arrangements allow recursive filtering without the need for extra pulses or pulse tapering to control transients in the filter.

A generalized Chebyshev channelizing filter

Abstract: In this paper some issues concerning the design and implementation of a satellite transponder's channelizing filter of degree 6 are discussed. In particular, the construction of the transmission coefficient, the synthesis procedure and the realization are discussed. the amplitude and the group delay responses of the filter are optimized simultaneously to meet prescribed requirements. the prototype filter has been synthesized as a cross-coupled array. Explicit formulae for the element values in terms of the real and the imaginary parts of the poles of the transmission coefficient are derived. Single mode direct coupled cavity realization with positive and negative couplings has been developed. an experimental filter has been constructed, the measured responses are very close to the computed ones.

Octave multiplex filter

Abstract: PURPOSE: To designate the characteristic at a logarithmically uniform interval at a frequency region by constituting each filter stage with two digital pair filters of low pass and band pass characteristic and changing a clock frequency unit delay time of a delay element constituting a digital filter of each stage so as to use the total filter multiplexedly. CONSTITUTION: A unit delay time is selected as 1/f 0 .2 -4 sec to decide the coefficient of an FIR filter, a clock frequency is selected as f 0 Hz to an input signal having an actual sampling interval 1/f 0 sec, and the delay time per stage of the delay element S 4 -1 is selected as nT=1/f 0 .2 -2 sec while keeping the coefficients the same. Similarly, the digital filter is selected similarly in other stage. In order to cancel the pass band of the odd number order, a signal is inputted to a digital filter F 4 having a low pass characteristic whose delay time per delay element stage is 1/f 0 .2 -3 sec. In a figure (b) of the output of the filter F 4 , the output is added to an output of a digital filter S 3 by an adder circuit D 4 and the result is as shown in figure (c). The processing above is repeated similarly and an output of an object characteristic is obtained from the final digital filter F 0 . COPYRIGHT: (C)1987,JPO&Japio

The Kalman-Bucy Filter

Abstract: Equation (3.1 or 3) mathematically describes the dynamic model in the Kalman-Bucy filter, whose output is a stochastic vector process to be estimated on the basis of noise-contaminated observations. To complete the Kalman-Bucy model for the estimation problem, we now define the observation process.

Performance measures in the selection of reconstruction filters for SPECT imaging

Abstract: Filter selection in SPECT image reconstruction poses an implicit tradeoff between image smoothness, image contrast and noise texture. It is known that the Ramachandran or ramp filter provides the greatest contrast (and resolution) in a reconstructed image at the expense of poor noise handling. In order to improve detection of either hot or cold lesions, practical experience dictates that some image smoothing must be provided. Thus, the question of the ''optimal'' filter selection has been the subject of this investigation. Using an extension of the approach as originally developed by Beck and Metz, the authors derived a figure-of-merit (FOM) that quantifies the performance of the reconstruction algorithm being tested. The FOM can be separated into two components: smoothness factor (SF) which is a measure of noise reduction and contrast factor (CF) which is a measure of the spatial resolution of the filter being tested. Each component is measured separately. The optimum filter should maintain the contrast of the Ramachandran filter (CF = 0.313) but improve the noise handling (SF = 8.86). A variety of commercially available filters were tested: rectangular, Hann, and Butterworth. Also examined were the order of the filter and the interpolation method (nearest-neighbor, linear, and Fourier methodsmore » with padding of zeros). The authors found that high order Butterworth filters performed best in maintaining contrast but reducing noise. Using a cutoff frequency of 0.200 (Nyquist = 0.500) and an order of 30, the filter provided a CF = 0.313 and a SF = 30.31. Linear interpolation was marginally better.« less