Showing papers on "Prototype filter published in 1987"
Book•
01 Jan 1987
TL;DR: Introduction to Digital Filters Properties of Finite Impulse-Response Filters Design of Linear-Phase Finite Filters Minimum Phase and Complex Approximation and Comparison of Filtering Alternatives Appendix Index.
Abstract: Introduction to Digital Filters Properties of Finite Impulse-Response Filters Design of Linear-Phase Finite Filters Minimum Phase and Complex Approximation Implementation of Finite Impulse-Response Filters Properties of Infinite Impulse-Response Filters Design of Infinite Impulse-Response Filters Implementation of Infinite-Response Filters Comparison of Filtering Alternatives Appendix Index
908 citations
TL;DR: Analysis and examples indicate that FIR-median hybrid filters preserve details better and are computationally much more efficient than the conventional median and the K-nearest neighbor averaging filters.
Abstract: A new class of median type filters for image processing is proposed. In the filters, linear FIR substructures are used in conjunction with the median operation. The root signals and noise attenuation properties of the FIR-median hybrid filters are analyzed and compared to representative edge preserving filtering operations. The concept of multilevel median operation is introduced to improve the detail preserving property of conventional median and the FIR-median hybrid filters. In the multilevel filters there exists a tradeoff between noise attenuation and detail preservation. The analysis and examples indicate that FIR-median hybrid filters preserve details better and are computationally much more efficient than the conventional median and the K-nearest neighbor averaging filters.
425 citations
TL;DR: In this paper, a new formulation of the detection filter problem is generated by assignment of the closed-loop eigenstructure under certain constraints, and necessary and sufficient conditions for the solution of these algebraic equations are determined which produce a complete theory for detection filters.
Abstract: A new formulation of the detection filter problem is generated by assignment of the closed-loop eigenstructure under certain constraints. Detection filters, which are actually a specific class of observers, fix the output error direction of the system so that it can be associated with a particular failure mode and its known design failure direction. The derivation of detection filters from an eigensystem assignment approach permits a very transparent theory. The detection filter gains and closed-loop eigenvectors are obtained from a set of simultaneous equations. Necessary and sufficient conditions for the solution of these algebraic equations are determined which produce a complete theory for detection filters.
393 citations
TL;DR: This paper introduces a new generalized class of median-type filters which it is shown have essentially the same statistical properties and same type of root signals as the standard median (SM) filters, called FIR-median hybrid (FMH) filters.
Abstract: This paper introduces a new generalized class of median-type filters which we call FIR-median hybrid (FMH) filters. The input signal x(n) is filtered with M linear phase FIR filters, and the output of the FMH filter is the median of the outputs of the FIR filters. The statistical properties of the FMH filters are analyzed for input signals with Gaussian, double exponential, and uniform density functions. It is shown that FMH filters have essentially the same statistical properties and same type of root signals as the standard median (SM) filters. FMH filters preserve corner points and ramps and attenuate impulsive-type noise components effectively. An interesting subclass of the FMH filters requiring only one scaling multiplier, two additions, and three compare/swap operations irrespective of the filter length is introduced. In the examples, speed improvement by a factor of 450 is obtained over an otherwise approximately equivalent SM filter.
284 citations
TL;DR: In this paper, an adaptive median filter is proposed, which allows the simultaneous removal of a combination of signal-dependent and additive random noise in addition to mixed impulse noise in images, processed in a single filtering pass.
Abstract: A novel adaptive median filter is proposed. It allows the simultaneous removal of a combination of signal-dependent and additive random noise in addition to mixed impulse noise in images, processed in a single filtering pass. The adaptation algorithm is based on the local signal-to-noise ratio. An extension of the class of nonlinear mean filters to adaptive filters is considered. The performance of the adaptive median filter is compared to the commonly used median filter and the nonlinear mean filter.
173 citations
TL;DR: This correspondence shows how the design time for equiripple half-band filters can be reduced by a considerable amount and places in evidence new implementation schemes, which simultaneously ensure low passband and stopband sensitivities.
Abstract: Based on a well-known property of FIR half-band filters, this correspondence shows how the design time for equiripple half-band filters can be reduced by a considerable amount. The observation which leads up to this improved procedure also places in evidence new implementation schemes, which simultaneously ensure low passband and stopband sensitivities. Extension of the method to M th-band filter design is also outlined.
123 citations
TL;DR: In this paper, the authors investigated the properties of a special class of recursive polyphase filters whose subfilters are distinct all-pass filters, and proposed a Remez-type algorithm for designing these filters.
Abstract: We investigate the properties of a special class of recursive polyphase filters whose subfilters are distinct allpass filters. On the basis of the discovered properties, a Remez-type algorithm is constructed for designing these filters. The proposed algorithm is faster and gives higher filter selectivities than other existing methods. Better understanding of the filter properties allows us to select the branch filter orders such that the filter complexity is minimized. In addition to nonlinear-phase filters, the algorithm is applicable to the design of approximately linear-phase filters. Several examples are included illustrating the efficiency of the proposed design scheme. The characteristics of the resulting filters are exposed by means of experimental results, and the implementation of the allpass branch filters is discussed. In a companion paper [33], we use these polyphase filters as basic building blocks in constructing efficient filters for sampling rate alteration. In addition, it is shown how the don't care bands, which cannot be avoided in filters of this type, can be suppressed by using an additional correction filter stage.
113 citations
TL;DR: It is shown that these filters have many oscillatory infinitely long root signals and methods to prevent the possible harmful effect caused by the existence of these roots are suggested.
Abstract: In this correspondence, we analyze the root structures of the standard median (SM) filters, the recursive median (RM) filters, and the FIR median hybrid (FMH) filters. It is shown that these filters have many oscillatory infinitely long root signals. When a section of an oscillatory root is present in a signal, the filter's noise attenuation of the filter is not as good as predicted by statistical measures. We also suggest methods to prevent the possible harmful effect caused by the existence of these roots.
56 citations
06 Apr 1987
TL;DR: The polyphase filter array has been used for efficient implementations of filters with integer sampling rate conversions and the computational complexity is reduced by a factor equal to the sampling rate ratio.
Abstract: The polyphase filter array has been used for efficient implementations of filters with integer sampling rate conversions. [1] The filter in the high sampling rate side is decomposed into its polyphase filters which can be moved to the lower sampling rate side without changing their functions. For FIR filters the computational complexity is reduced by a factor equal to the sampling rate ratio. A rational (L/M) sampling rate conversion system realized with a 1-to-L interpolator followed by an M-to-1 decimator has three sampling rates F, LF and (L/M)F involved. By using the polyphase filter array a filter operating at the sampling rate of LF can be implemented in either the input side or the output side with lower sampling rates. The polyphase filter matrix structure will operate at the sampling rate of F/M, which does not show in the above model and is lower than any one of those three rates. For FIR filters the computational complexity is reduced by a factor of LM compared to the direct realization of the integral filter or by a factor of M (or L) compared to the polyphase filter array realization while the system input-output relation is maintained.
56 citations
TL;DR: In this article, a procedure for designing computationally efficient FIR and IIR fan filters is developed, which are derived from one-dimensional prototypes which are used in changing sampling rates by a factor of two.
Abstract: A procedure for designing computationally efficient FIR and IIR fan filters is developed. The fan filters are derived from one-dimensional prototypes which are used in changing sampling rates by a factor of two. The transfer function B(z) of the prototype is expressed as B(z) = T_1(z^2)+zT_2(z^2) , which leads to a simple expression for the fan filter transfer function. The relation between the passband and stopband deviation requirements of the prototype and the fan filter is derived and employed in the design for meeting two commonly given sets of specifications. A special class of elliptic filters with zeros on the imaginary axis in the z plane is shown to yield computationally efficient IIR fan filters. FIR half-band filters are shown to be suitable one-dimensional prototypes for FIR fan filter design. In this case, symmetry constraints on FIR fan filter impulse response are automatically satisfied.
TL;DR: In this paper, the use of active filters, in conjunction with minimal passive filters, to achieve the required ripple levels with a much smaller overall volume is discussed, and working active filters with gains in excess of 100 at 1 MHz are also presented.
Abstract: As the switching frequency of a dc-converter is raised from 100 kHz to 1 MHz, one would expect the energy storage elements of the circuit to be correspondingly reduced in size. If the circuit must meet input and output ripple requirements of the nature of MIL-STD-461B CEO3, however, the advantage of a higher switching frequency is significantly offset by the higher filter attenuation levels required. The use of active filters, in conjunction with minimal passive filters, to achieve the required ripple levels with a much smaller overall volume is discussed. Working active filters with gains in excess of 100 at 1 MHz are also presented.
TL;DR: In this article, the authors proposed two new nonlinear filters for filtering signal-dependent noise, additive noise, and impulsive noise in image processing, based on a generalized homomorphic transformation.
Abstract: In this paper, we propose two new nonlinear filters for filtering signal-dependent noise, additive noise, and impulsive noise in image processing. The first filter proposed is an order statistic filter based on a generalized homomorphic transformation. The second is an adaptive order statistic filter with a variable threshold, which changes according to the noise level. Both of these filters perform well for the different kinds of noise encountered in image processing. They suppress signal-dependent noise, additive noise, and impulsive noise better than median filters, \alpha -trimmed mean filters, general nonlinear mean filters, modified trimmed mean filters, and double-window modified trimmed mean filters. They also preserve the edges of an image better than median filters and are simple to implement.
TL;DR: Complex coefficient digital filters, with applications for processing real sequences, are examined and conventional filter structures, such as parallel, cascade, lattice, and state-space forms, are extended to the complex domain.
Abstract: Complex coefficient digital filters, with applications for processing real sequences, are examined. The method is quite general, and allows any real rational transfer function to be expressed in terms of a complex rational transfer function of reduced order. When implemented in complex hardware form, the reduction of filter order can provide an increase in computational efficiency and speed. Conventional filter structures, such as parallel, cascade, lattice, and state-space forms, are extended to the complex domain. Illustrative examples of complex coefficient filter synthesis are included, along with coefficient sensitivity comparisons between the complex coefficient filters and their real coefficient counterparts.
01 Jan 1987
TL;DR: In this article, a canonical bandpass filter constructed by placing dualmode dielectric resonators coaxially in a cutoff circular waveguide is discussed, and a rigorous analysis of the dual-mode resonator and inter-resonator coupling is performed with the mode matching technique.
Abstract: A canonical bandpass filter constructed by placing dual-mode dielectric resonators coaxially in a cutoff circular waveguide is discussed Rigorous analysis of the dual-mode dielectric resonator and inter-resonator coupling is performed with the mode matching technique Four- and six-pole elliptic bandpass filters are realized at 12 GHz
TL;DR: An iterative technique between the spatial domain and the spectral domain is used to rephase the invariant modes so that a filter composed of a linear combination of modes has the proper overall invariance.
Abstract: A procedure is presented for designing distortion-invariant correlation filters. Optical correlation filters designed using this technique retain full position invariance. The filter design begins by finding the distortion-invariant modes (eigenfunctions) for a particular image. The input image, filter, and correlation response are all spectrally expanded in terms of these orthogonal eigenfunctions. An iterative technique between the spatial domain and the spectral domain is used to rephase the invariant modes so that a filter composed of a linear combination of modes has the proper overall invariance. The iterative technique also controls the information content of the filter by maintaining specified amplitudes for each invariant mode in the filter. Targets are detected by spanning the filter to determine points of constant amplitude.
TL;DR: In this article, two techniques for the design of three-dimensional (3D) filters are introduced, one based on coefficient transformations of two-dimensional circularly symmetric filters and the other based on cascade arrangements of 2-D rotated filters.
Abstract: Two techniques for the design of three-dimensional (3-D) filters are introduced in this paper. The first one is based on coefficient transformations of two-dimensional (2-D) circularly symmetric filters. These filters can be designed by cascading 2-D rotated filters. Stability of the 3-D filters designed is discussed and a stabilization procedure based on cepstrum analysis, is proposed. Stable implementation schemes are introduced. The second technique is based on cascade arrangements of 2-D rotated filters and results in 3-D spherically symmetric filters. Examples of 3-D filters designed on the basis of both techniques introduced are presented. Finally, some comparisons among several 3-D design techniques are given.
Proceedings Article•
01 Sep 1987TL;DR: A bit-parallel third order lowpass filter has been designed and fabricated in a 2 μm CMOS technology and has a bireciprocal transfer function and is fully functional at a sampling frequency of 35 MHz.
Abstract: To test the performance of recursive digital filters for high sampling rate applications, such as pulse former filters or filters for intermediate frequencies of TV sets and radios, a bit-parallel third order lowpass filter has been designed and fabricated in a 2 ?m CMOS technology. This filter has a bireciprocal transfer function and is fully functional at a sampling frequency of 35 MHz. In order to obtain such high frequencies with a recursive filter, a carry-save arithmetic has been used rather than a conventional carry-propagate arithmetic. This gives rise to several realization problems which will be described in this contribution. The prototype chip contains about 9000 transistors and occupies an area of 14.7 mm2.
01 Apr 1987
TL;DR: This paper investigates the numerical stability of the well-known Fast Transversal Filters algorithms by introducing redundancy in the algorithm by computing some quantity in two different ways.
Abstract: This paper investigates the numerical stability of the well-known Fast Transversal Filters algorithms. Two different modes of divergence are indicated with an effort to mathematically explain the reason of the appearance of each mode. An original stabilization method is thus introduced. It basically consists in introducing redundancy in the algorithm by computing some quantity in two different ways. The difference of the two values (nonzero because of finite precision) is fed back to the algorithm in order to correct it. The stabilized algorithm requires some more operations but it is always linear in the order of the filter per iteration.
01 Jan 1987
TL;DR: In this paper, Costa and Venetsanopoulos and Goodman modified two-dimensional (2D) rotated digital filters by performing rotation through a somewhat more general transformation, which can lead to an unstable 2-D digital filter when finiteprecision arithmetic is used for the implementation.
Abstract: Two methods for the design of two-dimensional (2-D) rotated digital filters proposed by Costa and Venetsanopoulos and Goodman are modified by performing rotation through a somewhat more general transformation. This modification eliminates nonessential singularities of the second kind, which can lead to an unstable 2-D digital filter when finiteprecision arithmetic is used for the implementation. The paper also describes detailed procedures for the design of 2-D rotated digital filters satisfying prescribed specifications. The procedures can be used for the design of low-pass, high-pass, bandpass, and bandstop filters of the Butterworth, Chebyshev, or elliptic type, and also for the design of wave digital filters. The specifications that can be prescribed are the maximum passband ripple, the minimum stopband attenuation, the passband and stopband edges, and the degree of circularity. Further, zero phase can readily be achieved.
TL;DR: In this paper, the problem of median filtering synthesis is addressed, where the performance of a median filter is achieved through the cascade of several median filters of smaller window size, and a statistical performance criterion is chosen for the system specifications.
Abstract: In this paper, we address the problem of median filtering synthesis, that is, the design of a median filter system realization that will satisfy a given set of specifications. In particular, we address the problem of a cascade median filter realization, where the performance of a median filter is achieved through the cascade of several median filters of smaller window size. Because of the nonlinear nature of these filters, a statistical performance criterion is chosen for the system specifications. In order to evaluate the output of the cascade filter, a method is developed which finds the statistics of the roots of median filters, where roots are signals obtained after several median filter passes. Finally, a VLSI implementation for a cascade median filter system is presented.
Book•
26 Mar 1987
TL;DR: In this article, the authors present a survey of the history of crystal filters and their applications in various applications, including Transformers and Coils, as well as the specification and testing of assembled crystal filters.
Abstract: Quartz-Crystal Resonators. Transformers and Coils. Filter Circuit Theory. Discrete Crystal Filters. Mechanical Design and Manufacture of Discrete Crystal Filters. Monolithic Crystal Filters. Cascaded Monolithic Crystal Filters. Applications of Crystal Filters. Specification and Testing of Assembled Crystal Filters. Using Crystal Filters. Appendix. Index.
TL;DR: A systolic array realization for 2-D digital filters is presented, based on Chan's local state-space model, which has the minimum number of states (minimum delays) and a technique is presented to reduce the number of multipliers to its minimum.
Abstract: A systolic array realization for 2-D digital filters is presented. It is based on Chan's local state-space model. This structure has the minimum number of states (minimum delays). A technique is also presented to reduce the number of multipliers to its minimum. Conditions that the system eigenvalues have to satisfy to ensure low sensitivity with respect to the variations in the multipliers are developed. Several examples are included to illustrate the proposed techniques.
01 Apr 1987
TL;DR: A new technique for the design of FIR fractional-slope phase filters based on Chebyshev approximation is described and analyzed, which results from a formulation of the problem which satisfies the Haar condition, thus allowing the use of the efficient Remez exchange algorithm.
Abstract: A new technique for the design of FIR fractional-slope phase filters based on Chebyshev approximation is described and analyzed. The technique results from a formulation of the problem which satisfies the Haar condition, thus allowing the use of the efficient Remez exchange algorithm. The new design is implemented with a modification of the McClellan-Parks-Rabiner FIR filter design program. The resulting fractional-slope phase filters are shown to have a complex error function that is essentially equiripple in magnitude. The new technique may be used for designing parallel elements of a multirate filter, such as the polyphase interpolation filter, or for stand-alone filters used as fractional-slope phase shifters. The advantages of the new technique are the simplicity and numerical stability of the design program and the lack of restrictions on phase slope specification.
TL;DR: The measurements show that by careful optimization of the gain bandwidth, slew rate, and gain of the amplifiers, high-performance low-power SC filters can be constructed.
Abstract: A micropower fourth-order elliptical switched-capacitor (SC) low-pass filter for biomedical applications has been designed and measured. The charge transfer error of an SC integrator using a transconductance amplifier is discussed. Also first-order noise and PSRR calculations are performed and compared with the results of simulations and measurements. The measurements show that by careful optimization of the gain bandwidth, slew rate, and gain of the amplifiers, high-performance low-power SC filters can be constructed. The cutoff frequency of the filter is 5 kHz, the ripple in the passband is 0.27 dB, and stopband rejection is 49 dB. The power consumption of the filter is 190 /spl mu/W with /spl plusmn/2.5-V power supplies. The dynamic range of the filter is 75 dB, and the total harmonic distortion over the whole passband range is below 0.25% for a 2-V/SUB pp/ input signal. The PSRR of the filter is above 40 dB at frequencies below 3 kHz.
TL;DR: In this article, the adaptive median hybrid (AMH) filters were proposed to estimate the current signal value from future signal values and from past input or output signal values, and the output of the overall filter is the median of the adaptive filter outputs and the input signal value.
Abstract: In this paper, adaptive filter structures suitable for filtering signals with rapidly varying characteristics are presented. In the proposed Adaptive Median Hybrid (AMH) filters, adaptive filter substructures are used to estimate the current signal value from future signal values and from past input or output signal values. The output of the overall filter is the median of the adaptive filter outputs and the input signal value. Using computer simulations, the convergence properties of the filters in stationary and nonstationary signals have been analyzed. Due to the median operation, the AMH filters adapt and preserve abrupt changes in signal statistics substantially better than conventional adaptive filters.
01 Apr 1987
TL;DR: A new class of FIR Median Hybrid (FMH) filters which contain linear FIR substructures to estimate the current signal value using forward and backward prediction to obtain a filter which attenuates noise on constant and ramp signals.
Abstract: In this paper, we introduce a new class of FIR Median Hybrid (FMH) filters which contain linear FIR substructures to estimate the current signal value using forward and backward prediction The output of the overall filter is the median of the predicted values and the actual signal value in the middle of the filter window Predictors maximizing the signal to noise ratio on signal sections described by an 1th order polynomial are derived The ramp enhancement filters are shown to attenuate the noise on a ramp signal better than the Standard Median (SM) filters The new predictive FMH filters are shown to have root signals which do not exist for the SM filters, eg triangular waves By combining the level and the ramp enhancement FMH filters, we obtain a filter which attenuates noise on constant and ramp signals
06 Apr 1987
TL;DR: A new type of nonlinear filters, the Adaptive Median Hybrid (AMH) filters, for the suppression and detection of short duration interferences and two types of AMH filters are introduced, the AMH filter with separate adaptive substructures (SAMH) and the AMh filter with coupled substructure (CAMH), which have different convergence properties and implementation.
Abstract: In this paper, we introduce a new type of nonlinear filters, the Adaptive Median Hybrid (AMH) filters, for the suppression and detection of short duration interferences. In the AMH filters, adaptive filter substructures are used to estimate the current signal value from the future and past signal values. The output of the overall filter is the median of the adaptive filter outputs and the current signal value. This kind of nonlinear filter structure is shown to adapt and preserve rapid changes in signal characteristics well. However, it filters out short duration interferences. By examining the difference between the original and filtered data, interferences can be detected. We introduce two types of AMH filters, the AMH filter with separate adaptive substructures (SAMH) and the AMH filter with coupled substructures (CAMH), which have different convergence properties and implementation. We use both synthetic and real data (speech and electroencephalogram (EEG)) to show the applicability of the proposed filters.