Showing papers on "Prototype filter published in 1977"
TL;DR: In this paper, a technique for implementing a finite impulse response (FIR) digital filter in a residue number system (RNS) is presented, and a new hardware implementation of the Chinese Remainder Theorem is proposed for an efficient translation of residue coded outputs into natural numbers.
Abstract: A technique is presented for implementing a finite impulse response (FIR) digital filter in a residue number system (RNS). For many years residue number coding has been recognized as a system which provides a capability for the implementation of high speed multiplication and addition. The advantages of residue coding for the design of high speed FIR filters result from the fact that an FIR requires only the high speed residue operations, i.e., addition and multiplication, while not requiring the slower RNS operations of division or sign detection. A new hardware implementation of the Chinese Remainder Theorem is proposed for an efficient translation of residue coded outputs into natural numbers. A numerical example illustrates the principles of residue encoding, residue arithmetic, and residue decoding for FIR filters. An RNS implementation of a 64th-order dual bandpass filter is compared with several alternative filter structures to illustrate tradeoffs between speed and hardware complexity.
294 citations
TL;DR: A simple, powerful method for suitably combining the results of several passes through the same filter is described in detail, and its computational efficiency is compared to the best possible filter designs meeting the same specifications.
Abstract: When processing data by filters, we often find it necessary to improve the performance of the filter, either by increasing the out-of-band rejection (loss) or by decreasing the error in the passband, or both. A first approach is to process the data by repeated passes through the same filter. Each pass, while increasing the out-of-band loss, also increases the passband error, often to an undesirable level. It also increases the length (order) of the equivalent filter. How can we do a better job of filtering by suitably combining the results of several passes through the same filter? By "better" we mean both less passband error and greater out-of-band, or stopband, loss. This process is called filter sharpening. A simple, powerful method for doing this is described in detail, and its computational efficiency is compared to the best possible filter designs meeting the same specifications. The design method, based on the idea of the amplitude change function, is restricted to symmetric nonrecursive (finite impulse response) filters with piecewise constant pass- and stopbands. Several illustrative examples are given.
242 citations
09 May 1977
TL;DR: In this paper, the amplitude change function is used to improve the performance of symmetric non-recursive (finite impulse response) filters with piecewise constant pass and stopbands.
Abstract: When processing data by filters we often find it necessary to improve the performance of the filter, either by increasing the out-of-band rejection (loss) or by decreasing the error in the passband, or both. A first approach is to process the data by repeated passes through the same filter. Each pass, while increasing the out-of-band loss, also increases the passband error, often to an undesirable level. It also increases the length (order) of the equivalent filter. How can we do a better job of filtering by suitably combining the results of several passes through the same filter? By "better" we mean both less passband error and greater out-of-band, or stopband, loss. This process is called filter sharpening. A simple, powerful method for doing this is described in detail, and its computational efficiency is compared to the best possible filter designs meeting the same specifications. The design method, based on the idea of the amplitude change function, is restricted to symmetric nonrecursive (finite impulse response) filters with piecewise constant pass and stopbands. Several illustrative examples are given.
137 citations
TL;DR: In this paper, it was shown that overflow-stable filters of any order, without any constraints on pole locations within the unit circle, can be realized by parallel-cascade structures of minimum norm systems.
Abstract: In recursive digital filters, the norm of the system matrix is an important design parameter with respect to overflow behavior. Filter realizations that minimize this norm are shown to be free of autonomous overflow limit cycles. Overflow-stable filters of any order, without any constraints on pole locations within the unit circle, can be realized by parallel-cascade structures of minimum norm systems. Minimum norm realizations require the minimum number of delay elements but, in general, more than the minimum number of multiplications and additions.
136 citations
TL;DR: In this paper, the authors established novel stability criteria for multidimensional digital and analog filters with rational transfer functions, which generalize and simplify the stability test for two-dimensional digital filters developed by Huang [4] and significantly simplify the corresponding tests of stability of arbitrary multi-dimensional filters established by Anderson and Jury [6].
Abstract: Novel stability criteria are established, for multidimensional digital and analog filters with rational transfer functions. The criteria generalize and simplify the stability test for two-dimensional digital filters developed by Huang [4], and significantly simplify the corresponding tests of stability of arbitrary multidimensional filters established by Anderson and Jury [6].
133 citations
TL;DR: A simple resampling technique is presented that extends the range of designs to conversions between any two rates and can vary slightly as in a practical situation where the input signal and output signal are under the control of autonomous clocks.
Abstract: Filtering is necessary in decimation (decreasing the sampling rate of) or interpolation (increasing the sampling rate of a digital signal. If the rate change is substantial, the process is more efficient when the decimation or interpolation occurs in stages rather than in one step. Half-band filters are particularly efficient for effecting octave changes in sampling rate and nine digital filters are presented, eight of them half-band filters, to be used as components of multistage interpolators and decimators. Also presented is a procedure for combining the filters to produce multistage designs that meet a very wide range of accuracy requirements (stopband attenuation to 77 dB, passband ripple as low as 0.00014). The nine filters admit changes between sampling rates above 4W, where W is the nominal bandwidth of the signal. Established design techniques may be used to obtain efficient filters for conversion between 4W Hz sampling and 2W Hz, the "baseband sampling rate." With these multistage filters, the possible interpolation and decimation ratios are all integer multiples of powers of two. To overcome this restriction we present a simple resampling technique that extends the range of designs to conversions between any two rates. The interpolation or decimation ratio need not be an integer or even rational. In fact, it can vary slightly as in a practical situation where the input signal and output signal are under the control of autonomous clocks. We demonstrate the approach by means of several design examples and compare its results with those obtained from the optimization scheme of Crochiere and Rabiner.
121 citations
TL;DR: In this paper, a new structure is proposed for the implementation of 2-D FIR digital filters designed by transformations of 1-D filters, which requires a minimum number of multiplications and uses directly as coefficients the impulse response samples of the original filter.
Abstract: A new structure is proposed for the implementation of 2-D FIR digital filters designed by transformations of 1-D filters. This structure requires a minimum number of multiplications and uses directly as coefficients the impulse response samples of the 1-D prototype filter. The roundoff noise of the new structure is analyzed in detail and is shown to be superior to that of previous implementations for all examples studied.
107 citations
TL;DR: The possibilities for extending the class of lowpass recursive digital filters to include high pass, bandpass, and bandstop filters are described, and experience with a PDP 11 computer has shown that these filters may be programmed simply using machine code, and that online operation at sampling rates up to about 8 kHz is possible.
Abstract: After reviewing the design of a class of lowpass recursive digital filters having integer multiplier and linear phase characteristics, the possibilities for extending the class to include high pass, bandpass, and bandstop (‘notch’) filters are described. Experience with a PDP 11 computer has shown that these filters may be programmed simply using machine code, and that online operation at sampling rates up to about 8 kHz is possible. The practical application of such filters is illustrated by using a notch desgin to remove mains-frequency interference from an e.c.g. waveform.
104 citations
TL;DR: This filter is a very compact structure, it radiates power significantly less than conventional shunt-stub and coupled-line filters and is also less susceptible to the influence of other components and lines in its vicinity.
Abstract: The paper describes the design of a new class of microstrip bandstop filter. This filter is a very compact structure, it radiates power significantly less than conventional shunt-stub and coupled-line filters and is also less susceptible to the influence of other components and lines in its vicinity.
101 citations
TL;DR: In this paper, a bandpass filter and a discriminator for r.f. signals have been constructed by using fiber-optic delay lines, and their frequency-response characteristics have been investigated.
Abstract: A bandpass filter and a discriminator for r.f. signals have been constructed by using fibre-optic delay lines, and their frequency-response characteristics have been investigated. Both devices had a fundamental frequency of 193 MHz, and the filter had a 3 dB passband of 10 MHz.
TL;DR: In this article, the average phase of certain parts in the frequency plane was studied for zero-order non-coherent optical filtering by subtraction of images recorded with two different filters.
TL;DR: A recently introduced new stability test for single-dimensional digital filters is extended to apply to two-dimensional filters, and the procedure for the implementation of this test is discussed.
Abstract: A recently introduced new stability test for single-dimensional digital filters is extended to apply to two-dimensional filters, and the procedure for the implementation of this test is discussed. Nontrivial examples for both discrete and continuous two-dimensional filters are used to illustrate the procedure.
TL;DR: In this article, a generalization of Ramachandran and Lakshminarayanan's |ω|-filter has been proposed for 3D reconstruction of a density function, based on a direct convolution algorithm.
Abstract: The 3-D reconstruction of a density function is based on a direct convolution algorithm developed first by Ramachandran and Lakshiminarayanan. Their method adopts a particular choice of weighting function or filter which is called here an |ω|-filter. In some cases this choice of filter had an undesirable oscillatory response. To remedy this problem Shepp and Logan found a weighting function which produced a better reconstruction of a head section. The filter functions of Ramachandran and Lakshminarayanan and Shepp and Logan are only two of many choices for an |ω|-filter. Shepp and Logan's filter was the best for the early tomographic machines. Their filter function provided both a damped response to the cut-off frequency and a low sensitivity to noise. For the new tomographic machines, however, it is desirable to find filters that are not sensitive to counting noise, sample size and sample spacing as the previous filters. Here a study and generalization is made of the previous |ω|-filters. It extends the important filters of Ramachandran and Lakshiminarayanan, and Shepp and Logan to a class of generalized |ω|-filters. A generalized |ω|-filter can be chosen to have both good accuracy and a flexibility to cope with noise. A detailed comparison is made among the different possible filter shapes with respect to their responses to simulated data and noise. Finally in this paper it is demonstrated that a substantial reduction in the x-ray exposure time can be accomplished by choosing the appropriate generalized |ω|-filter.
01 Jan 1977
TL;DR: The design of a fully-integrated NMOS analog sampled-data recursive bandpass filter is described and experimental results for the critical elements and for partially integrated feasibility models in both metal-gate and silicon-gate NMOS technologies are discussed.
Abstract: MONOLITHIC INTEGRATION of precision audio frequency filters has been a difficult objective to reach. Active RC filters using numerous off-chip precision RC elements have been in use for years’. Transversal filters using CCD or BBD approaches occupy very large silicon areas when fully integrated with all peripheral circuits2. Fully integrated digital filters require precision linear A/D coders and also require a large silicon area . Analog sampleddata recursive filters have been described previously, but these employed many off-chip precision components4’5. This paper will describe the design of a fully-integrated NMOS analog sampled-data recursive bandpass filter, and discuss experimental results for the critical elements and for partially integrated feasibility models in both metal-gate and silicon-gate NMOS technologies.
TL;DR: In this paper, a technique for rotating the frequency responses of separable filters is developed, where transfer functions having rational powers of z are introduced and realized by input/output signal array interpolations.
Abstract: A technique for rotating the frequency responses of separable filters is developed. In this technique transfer functions having rational powers of z are introduced and realized by input/output signal array interpolations. Several applications of this technique to designing two-dimensional recursive filters are presented. Two- and multidimensional manipulations are performed by a series of one-dimensional manipulations.
TL;DR: Companion work on the design of envelope-constrained filters is extended and shown to provide an easily implementable adaptive filter with a structure quite similar to that of other adaptive filters based on least-squares techniques.
Abstract: Companion work on the design of envelope-constrained filters is extended and shown to provide an easily implementable adaptive filter with a structure quite similar to that of other adaptive filters based on least-squares techniques. Behavior of the new filter in noise is examined, and a variety of other extensions are discussed. An application to TV channel equalization is explored in some detail.
TL;DR: In this paper, it was shown that the power fed back into the filter input can, under adverse conditions, nearly equal the power leaving the output, which can be a cause for parasitic oscillations in case of digital filters.
Abstract: If a filter is used in multiplex telephone equipment, it actually operates in a loop due to the presence of two-wire/four-wire terminating equipment. Due to this, the power fed back into the filter input can, under adverse conditions, nearly equal the power leaving the output. This can be a cause for parasitic oscillations in case of digital filters. If properly designed, however, wave digital filters and nonrecursive digital filters remain stable.
TL;DR: This paper considers the problem of optimizing spatial frequency domain filters for detecting a class of edges in images and shows that the optimal filter represents the Laplacian operator in image space followed by a low pass filter with a cutoff frequency.
Abstract: Edge detection and enhancement are required in a number of important image processing applications. In this paper we consider the problem of optimizing spatial frequency domain filters for detecting a class of edges in images. The filter is optimum in that it produces maximum energy in the vicinity of the location of the edge for a given spatial resolution I and the bandwidth Ω. We show that the filter transfer function can be specified in terms of the prolate spheroidal wavefunctions for a given space–bandwidth product IΩ. Further we show that for values of IΩ less than 2, the optimal filter represents the Laplacian operator in image space followed by a low pass filter with a cutoff frequency Ω.
TL;DR: In this paper, two multipliers are proposed which realize a completely general fractional multiply and are suitable for digital-filtering applications, but they require a fixed table look-up read-only memory.
Abstract: A recently proposed residue-number-arithmetic digital filter offers major cost and speed advantages over binary-arithmetic digital filters, but suffers one major drawback. The filter coefficients must be constant, since the lack of a fast method of multiplication by a fraction in residue arithmetic requires the coefficients to be realised by a fixed table look-up read-only memory. Two multipliers are proposed which realise a completely general fractional multiply and are suitable for digital-filtering applications.
TL;DR: In this paper, the generalized immittance converter (GIC) when embedded in an arbitrary network is derived based on minimizing the dependence of the GIC transfer function on the op amps' characteristics.
Abstract: Design equations are derived for the generalized immittance converter (GIC) when embedded in an arbitrary network. The derivation is based on minimizing the dependence of the GIC transfer function on the op amps' characteristics. Equations for the optimum design of the GIC when simulating inductance and frequency-dependent negative-resistance (FDNR) elements follow as special cases. The optimized GIC's are applied in the design of bandpass filters using Gorski-Popiel's ladder embedding technique. The design procedure is presented in detail and its application is illustrated through the design of a telephone-channel bandpass filter of the twelfth order.
01 Dec 1977
TL;DR: In this paper, a minimum-phase CCD low-pass transversal filter is compared to a linear-phase filter with the same magnitude characteristics, and it is shown that the minimum phase design can offer up to an order of magnitude improvement in group delay, and is also less sensitive to transfer inefficiency and tap weight error than the linear phase design.
Abstract: The calculated and measured response of a minimum-phase CCD low-pass transversal filter is compared to a linear-phase design with the same magnitude characteristics. It is shown that the minimum-phase design can offer up to an order of magnitude improvement in group delay, and is also less sensitive to transfer inefficiency and tap-weight error than the linear-phase design.
TL;DR: In this article, a filter tuning method based on the match of measured and computed input impedances for a short-circuited filter is described, and two singly terminated filters, an 8-pole Chebyshev filter and a 6-pole pseudoelliptic function filter, are tuned using this method.
Abstract: This paper describes a filter tuning method based upon the match of measured and computed input impedances for a short-circuited filter. Two singly terminated filters, an 8-pole Chebyshev filter, and a 6-pole pseudoelliptic function filter tuned by using this method have demonstrated excellent performance.
TL;DR: In this article, the authors proposed a covariance-invariant response matching technique for digital filter synthesis, based on the frequency response of these designs, which is superior to the methods of impulseinvariance and bilinear-z as a response matching design technique.
Abstract: When discretizing continuous-time filters, one is often interested in preserving a property termed covariance-invariance. Techniques are outlined for synthesizing discrete-time filters which are covariance-invariant with corresponding continuous-time filters. The synthesis techniques involve straightforward matrix decompositions or polynomial root-finding algorithms that can easily be programmed on a digital computer. Applications of the technique to digital filter synthesis are outlined, with example designs presented for covariance-invariant Butterworth and Chebyshev digital filters. Based on the frequency response of these designs it is argued that the method of covariance-invariance is superior to the methods of impulse-invariance and bilinear-z as a response matching design technique for the synthesis of digital filters. This superiority is especially apparent at sampling rates that are marginal with respect to filter critical frequencies. Moreover, the covariance-invariant designs are stably invertible solutions to a so-called spectral factorization problem. This property may be important in inverse filtering applications.
TL;DR: It is shown that the design of 2-variable filter functions using this approach reduces to the problem of identifying a suitable 2- variable reactance function g(s_1,s_2) and the realization of a stable single-variable transfer function T(s) .
Abstract: This paper describes a technique for approximating 2-variable filter specifications in the continuous or analog domain. It is shown that the design of 2-variable filter functions using this approach reduces to the problem of identifying a suitable 2-variable reactance function g(s_1,s_2) and the realization of a stable single-variable transfer function T(s) . Then T(g(s_1,s_2)) is the desired 2-variable stable transfer function which is guaranteed to have at least one realization whenever T(s) and g(s_1,s_2) are realizable. Applications of the theory developed in this paper are presented in the design of lumped-distributed filters and 2-dimensional digital filters.
TL;DR: Low-loss surface-acoustic-wave (SAW) filters with loss as low as -0.6 dB at 35 MHz, -2.3 dB at 320 MHz, and -50 dB sidelobes are commonly constructed as discussed by the authors.
Abstract: For several years surface-acoustic-wave (SAW) devices have looked promising as future components for receiver front-end and intermediate frequency filters. However, high insertion loss and a time domain triple transit signal have limited the utilization of these devices. The low loss SAW filter overcomes both previous problems and is a milestone in electronic component development. Filters with loss as low as -0.6 dB at 35 MHz, -2.3 dB at 320 MHz, and -50 dB sidelobes are routinely constructed. This paper discusses surface-wave filters in general, and then specifically the design of low loss SAW filters.
TL;DR: In this article, a closed-form solution to the problem of synthesizing recursive filters with specified gain and phase characteristics is presented, which is shown to be numerically well conditioned, and to result in BIBO stable realizations.
Abstract: A closed-form solution to the problem of synthesizing recursive filters with specified gain and phase characteristics is presented. The method is shown to be numerically well conditioned, and to result in BIBO stable realizations. By means of a free design parameter, the filter response can be modified according to given criteria in a particular application. Examples indicate that the method yields, with a standard setting of the free parameter, designs very close to optimal in terms of frequency-domain performance.
TL;DR: The operational amplifier rolloff characteristics and a single capacitor are used for obtaining an inverting bandpass function and the resulting filters have an extended frequency range of operation and a reduced number of external capacitors.
Abstract: The operational amplifier rolloff characteristics and a single capacitor are used for obtaining an inverting bandpass function. The filter performance depends on the gain-bandwidth product of the operational amplifier. Experimental results are included. The amplifier rolloff characteristics can be utilised in deriving transfer functions. The resulting filters have an extended frequency range of operation and a reduced number of external capacitors.
01 Jan 1977
TL;DR: In this paper, it was shown that the reactance transformation for an analog filter corresponds to an all-pass function transformation in the case of a digital filter, and this result could be used to obtain in a simple way the spectral transformations for digital filters derived by Constantinides.
Abstract: It is shown that the reactance transformation for an analog filter corresponds to an all-pass function transformation in the case of a digital filter. It is further shown how this result could be used to obtain in a simple way the spectral transformations for digital filters derived by Constantinides.