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Showing papers on "Prototype filter published in 1983"


Journal ArticleDOI
TL;DR: In this paper, the authors consider a class of nonlinear filters whose output is given by a linear combination of the order statistics of the input sequence, and choose the coefficients in the linear combination to minimize the output MSE for several noise distributions.
Abstract: We consider a class of nonlinear filters whose output is given by a linear combination of the order statistics of the input sequence. Assuming a constant signal in white noise, the coefficients in the linear combination are chosen to minimize the output MSE for several noise distributions. It is shown that the optimal order statistic filter (OSF) tends toward the median filter as the noise becomes more impulsive. The optimal OSF is applied to an actual noisy image and is shown to perform well, combining properties of both the averaging and median filters. A more general design scheme for applications involving nonconstant signals is also given.

604 citations


Proceedings ArticleDOI
14 Apr 1983
TL;DR: A new method of implementing filter banks for subband coding of speech by combining the quadrature filter characteristic with the polyphase network implementation of filter banks, which requires 35% fewer computations than existing designs.
Abstract: A new method of implementing filter banks for subband coding of speech is presented. First, Esteban's Quadrature Mirror Filter principle is extended to allow the direct synthesis of filter banks with any number of equal size filter bands. Then, by combining the quadrature filter characteristic with the polyphase network implementation of filter banks, a new filter bank structure is obtained which requires 35% fewer computations than existing designs. In this paper the theory of operation of the Polyphase Quadrature Filter is presented and techniques for its efficient implementation are described. Then, examples of filter banks using this approach are shown and compared with other designs, and a simulation of a 16 Kbits/s coder using these filters is presented.

320 citations


Journal ArticleDOI
TL;DR: In this paper, the authors used an 80 MHz differential single-stage CMOS operational amplifier and a fully differential identical-resonator elliptic bandpass ladder filter configuration for communications applications.
Abstract: Bandpass filters for communications applications are realized using an 80 MHz differential single-stage CMOS operational amplifier and a fully differential identical-resonator elliptic bandpass ladder filter configuration. Experimental results are given for a CMOS sixth-order 260 kHz elliptic bandpass filter with a Q-factor of 40, a clock frequency of 4 MHz, and a power dissipation of 70 mW.

250 citations


Journal ArticleDOI
TL;DR: In this article, the remaining unquantized coefficients of a FIR linear phase digital filter when one or more of the filter coefficients takes on discrete values are optimized using the least square response error.
Abstract: An efficient method optimizing (in the least square response error sense) the remaining unquantized coefficients of a FIR linear phase digital filter when one or more of the filter coefficients takes on discrete values is introduced. By incorporating this optimization method into a tree search algorithm and employing a suitable branching policy, an efficient algorithm for the design of high-order discrete coefficient FIR filters is produced. This approach can also be used to design FIR filters on a minimax basis. The minimax criterion is approximated by adjusting the least squares weighting. Results show that the least square criteria is capable of designing filters of order well beyond other approaches by a factor of three for the same computer time. The discrete coefficient spaces discussed include the evenly distributed finite wordlength space as well as the nonuniformly distributed powers-of-two space.

240 citations


Journal ArticleDOI
TL;DR: In this article, specific implementations of the finite impulse response (FIR) block adaptive filter in the frequency domain are presented and some of their important properties are discussed, and the time-domain block adaptive filtering is shown to be equivalent to the frequency-domain adaptive filtering, provided data sectioning is done properly.
Abstract: Specific implementations of the finite impulse response (FIR) block adaptive filter in the frequency domain are presented and some of their important properties are discussed. The time-domain block adaptive filter implemented in the frequency domain is shown to be equivalent to the frequency-domain adaptive filter (derived in the frequency domain), provided data sectioning is done properly. All of the known time- and frequency-domain adaptive filters [1]-[12], [16]-[18] are contained in the set of possible block adaptive filter structures. Thus, the block adaptive filter is generic and its formulation unifies the current theory of time- and frequency-domain FIR adaptive filter structures. A detailed analysis of overlap-save and overlap-add implementations shows that the former is to be preferred for adaptive applications because it requires less computation, a fact that is not true for fixed coefficient filters.

197 citations


Journal ArticleDOI
TL;DR: In this paper, a systematic way to design digital filters which allow clear separation of signals with periods of a few days from noise of higher frequency, particularly tidal and inertial, was given.
Abstract: A systematic way is given to design digital filters which allow clear separation of signals with periods of a few days from noise of higher frequency, particularly tidal and inertial. Several examples are given which pass little high-frequency power and none at the principal tidal frequencies. The Lanczos–cosine filter passes too much energy near diurnal frequencies; the Godin filter is better but not optimal. A longer filter is recommended, with flat low-frequency response, a sharp cut-off and very low noise. For current meter records containing inertial motions, it appears desirable to design a filter which specifically suppresses the local inertial frequency.

161 citations


Journal ArticleDOI
TL;DR: In this paper, the admittance inverter parameters of the coupling gaps between resonant sections are deduced from experiment, and bandpass filter design rules are developed, which allows easy filter synthesis from "prototype" low-pass designs.
Abstract: End-coupled resonator bandpass filters built in coplanar waveguide are investigated. The admittance inverter parameters of the coupling gaps between resonant sections are deduced from experiment, and bandpass filter design rules are developed. This allows easy filter synthesis from "prototype" low-pass designs. Measurements of single section resonator quality factors are used to predict filter insertion losses. Several examples of filters realized in coplanar waveguide are presented. Odd-mode coplanar waveguide filter elements that shortcircuit the even coplanar waveguide mode are investigated. Filter tuning, accomplished by adjusting the height of conducting planes above the resonant filter sections, is demonstrated.

156 citations


Journal ArticleDOI
TL;DR: In this article, a two-dimensional (2D) model for the class of causal, recursive, and separable in denominator (CRSD) filters is presented.
Abstract: After introducing a two-dimensional (2-D) model for the class of causal, recursive, and separable in denominator (CRSD) filters, a technique for approximating a given 2-D filter by a CRSD filter is presented. Also, a technique for 2-D CRSD model reduction is considered. Both the stability and minimality properties of the approximate model are explored. Some examples are also given to illustrate the proposed technique.

140 citations


Journal ArticleDOI
TL;DR: A new combined switched-capacitor (SC) frequency-sampling N-path filter is presented, which allows the implementation of very narrow bandpass filters.
Abstract: A new combined switched-capacitor (SC) frequency-sampling N-path filter is presented, which allows the implementation of very narrow bandpass filters. The included frequency-sampling (FS) filter suppresses undesirable passbands of the SC N-path filter. The center frequency f/SUB m/ of the bandpass filter is identical to the circuit clock frequency f/SUB c/. Experimental results are presented for a CMOS SC frequency-sampling four-path filter with second-order filter cells, a center frequency of 1 kHz, and -3-dB passband bandwidth of 11.5 Hz.

85 citations


Journal ArticleDOI
TL;DR: A microcomputer-based filter that removes line induced electrical interference from biopotential signals by learning one period of the noise waveform and subtracting it from the signal is developed.
Abstract: We have developed a microcomputer-based filter that removes line induced electrical interference from biopotential signals by learning one period of the noise waveform and subtracting it from the signal. Since it uses a noise template, the filter can remove noise waveforms containing several harmonics of the 60 Hz line frequency.

63 citations


Journal ArticleDOI
TL;DR: In this paper, the authors present a systematic solution to the problem of designing a stray-insensitive switched-capacitor filter based on the exact simulation of the operation of an LC ladder prototype.
Abstract: This paper presents a systematic solution to the problem of designing a stray-insensitive switched-capacitor filter based on the exact simulation of the operation of an LC ladder prototype. A complete procedure is given for the design of low-pass and bandpass filters with and without transmission zeros.

Proceedings ArticleDOI
01 Jan 1983
TL;DR: Experimental results from a CMOS sixth order 260kHz elliptic bandpass filter with Q of 40, clock frequency of 4MHz and power dissipation of 120mW, will be presented.
Abstract: Bandpass filters for communications applications using an 80MHz differential single-stage CMOS opamp and a fully differential idential-resonator elliptic bandpass ladder filter configuration will be reported. Experimental results from a CMOS sixth order 260kHz elliptic bandpass filter with Q of 40, clock frequency of 4MHz and power dissipation of 120mW, will be presented.

Journal ArticleDOI
TL;DR: A general scheme for changing sampling rates is developed, and a method of designing recursive (IIR) filters for use in this general scheme is presented, and the use of recursive filters with approximately linear phase led to a performance better than that using nonrecursive filters.
Abstract: A general scheme for changing sampling rates is developed, and a method of designing recursive (IIR) filters for use in this general scheme is presented. If phase response is ignored, then this scheme provides a considerable savings in computations over nonrecursive (FIR) schemes. The use of recursive filters with approximately linear phase also led to a performance better than that using nonrecursive filters. The quantization effects in the new scheme are minimal.

Journal ArticleDOI
TL;DR: In this paper, a technique for designing recursive and non-recursive two-dimensional digital filters by the application of a complex transformation to a one-dimensional low-pass filter is presented.
Abstract: This paper develops a technique for designing recursive and nonrecursive two-dimensional digital filters by the application of a complex transformation to a one-dimensional low-pass filter. A set of transformed filters is presented. The appropriate combination of these filters produces zero-phase fan filters. The advantage of such an approach is that the resulting fan filters are inherently stable and no optimization is needed. Several examples are presented for both finite impulse response (FIR) and infinite impulse response (IIR) fan filters and quadrant fan filters. The paper also includes implementation techniques using the properties of finite area arrays which reduce both computational and storage requirements compared with conventional implementations.

Journal ArticleDOI
TL;DR: A new initialization scheme for 3-band filters which improves convergence in the design of bandstop filters and an improved search algorithm which can be used to locate the maxima of the error function is described.
Abstract: Two techniques are described which can be used in the design of weighted-Chebyshev nonrecursive digital filters. The first technique is a new initialization scheme for 3-band filters which improves convergence in the design of bandstop filters. The second technique is an improved search algorithm which can be used to locate the maxima of the error function. It incorporates cubic interpolation with a selective step-by-step search and it replaces the exhaustive step-by-step search proposed by Rabiner, McClellan, and Parks in the past as well as the selective step-by-step search proposed by Antoniou recently. The two techniques as well as an improved technique for the rejection of superfluous maxima of the error function, proposed recently by Antoniou, are used to construct a filter-design package which can be used for the design of equiripple nonrecursive filters. When applied for the design of a large number of low-pass, high-pass, bandpass, and bandstop filters, the new design method leads to an average saving in the amount of computation of about 87 percent relative to the amount required by the Rabiner, McClellan, and Parks method. It is demonstrated that this improvement is achieved without degrading the robustness of the Remez exchange algorithm to a significant extent.

Journal ArticleDOI
TL;DR: An algorithm is presented for designing finite impulse response (FIR) recursive digital filters that require few multiplies to produce good frequency response, and uses dynamic programming techniques to find the best least-squares piecewise-exponential approximation to a desired impulse response of length P.
Abstract: An algorithm is presented for designing finite impulse response (FIR) recursive digital filters that require few multiplies to produce good frequency response, The process of reducing the number of multiplies needed to implement a digital filter is called thinning. This thinning algorithm uses dynamic programming techniques to find the best least-squares piecewise-exponential approximation to a desired impulse response of length P. Because these filters are implemented recursively, the number of arithmetic operations is independent of the model filter length P and is dependent only on the number of pieces or segments S used in the approximation ( S \ll P ). Examples of thinned narrow-band, broad-band, lowpass, and bandpass filters are given. Several of these thinned filters require fewer than one-third the number of multiplications required for the corresponding model filter, while still retaining desirable frequency response characteristics. The effects of coefficient quantization and finite precision arithmetic are also discussed.

Journal ArticleDOI
TL;DR: Results from this device show that the method of realizing such a filter is viable for many applications and maybe suitable to replace more conventional types of microwave filter realized using TEM-mode resonators.
Abstract: A design method for narrow-band suspended substrate strip-line filters having a true bandpass structure is presented. A generalized Chebyshev low-pass prototype is used, resulting in a convenient form of realization in suspended substrate stripline. A prototype device, designed with the aid of a computer program, is given as an example. Results from this device show that the method of realizing such a filter is viable for many applications and maybe suitable to replace more conventional types of microwave filter realized using TEM-mode resonators.

Journal ArticleDOI
TL;DR: A modification of the trasformation method used for the design of two-dimensional (2D) circularly symmetric finite impulse response (FIR) digital filters from one-dimensional filters is described in this article.
Abstract: A modification of the trasformation method used for the design of two-dimensional (2-D) circularly symmetric finite impulse response (FIR) digital filters from one-dimensional (1-D) filters is described. This modification entails the embedding of variable parameters in the different transformations applied to the different factors of the 1-D reference function. This new method results in the design of 2-D FIR filters whose frequency response characteristics meet the cutoff boundary specifications more closely than the transformations without the modification. This method is quite useful for the design of 2-D FIR filters with multiple cutoff boundaries such as bandpass filters.

Journal ArticleDOI
TL;DR: It turns out that the classical elliptic filters are seldom the best representatives of the filter class discussed in the paper, so simple explanations of some properties of the filters with denominator order lower than numerator order are given.
Abstract: This paper presents an efficient algorithm for the design of low-pass recursive digital filters with Chebyshev passband and stopband, all zeros on the unit circle, and different order numerator and denominator. The procedure takes advantage of the possibility of generating analytically magnitude squared functions with Chebyshev passband and adjustable zeros or Chebyshev stopband and adjustable poles. The resulting algorithm requires only one approximation interval making it more efficient than other existing design procedures. The number of multiplications per sample required in realizing the resulting filters is discussed and the optimal denominator and numerator orders are considered in narrow-band, wide-band, and intermediate applications. It turns out that the classical elliptic filters are seldom the best representatives of the filter class discussed in the paper. Simple explanations of some properties of the filters with denominator order lower than numerator order are given, such as the existence of an extra ripple in the passband and the minimum attainable passband ripple.

Journal ArticleDOI
TL;DR: These methods are used to develop digital versions of Butterworth and Chebychev filters and the bilinear transformation is used to derive the z-transforms of the filters from their s-plane continuous time descriptions.

Patent
10 Aug 1983
TL;DR: In this article, a non-invasive clinical instrument provides for in vivo detection of a restriction in an artery using Switched-Capacitor bandpass filters, which are used to remove noise and amplify a complex electrical signal.
Abstract: A non-invasive clinical instrument provides for in vivo detection of a restriction in an artery. A microphone is applied to the body over the artery and generates a complex electrical signal which is processed to remove noise and amplified. A digital bandpass filter circuit includes a plurality of individual bandpass filters, each of which covers 1/3 of an octave and covers adjacent octaves to cover the complete spectrum between about 2 to 2,000 hertz. Switched-capacitor bandpass filters are used. Each three bandpass filters separated by one-third of an octave are connected such that setting the center bandpass filter automatically resets the two adjacent bandpass filters to cover a single octave in steps of one-third. The sensed signal which is an AC analog signal is transmitted by an analog low pass filter to block undesirably higher frequency components from the digital filters. Digital low-pass filters connect the analog low-pass filter to the bandpass filters. A display unit includes a multiplexor for sequentially transmitting the several bandpass signals to the horizontal input in proper timed relation. Each bandpass filter includes an AC (RMS) to DC (average) converter for establishing an appropriate DC output signal, which is passed through a sample and hold circuit to freeze the display.

Journal ArticleDOI
TL;DR: A universal second-order switched-capacitor filter section has been fabricated on an NMOS chip that can perform all five basic filter types as well as a sine wave oscillator without external components, while requiring only an external clock.
Abstract: A universal second-order switched-capacitor filter section has been fabricated on an NMOS chip. The device can perform all five basic filter types as well as a sine wave oscillator without external components, while requiring only an external clock. The filter type is determined by selecting one or more of three input pins. The filter response is determined by ten external programming pins which may be either digitally controlled or hard wired.

Proceedings ArticleDOI
Sharad Singhal1, B. Atal
01 Apr 1983
TL;DR: The possibility that multi-pulse excitation can approximate the all-pole filter excitation sufficiently closely and obtain the optimum filter parameters for this excitation is examined.
Abstract: Present LPC analysis procedures assume that the input to the all-pole filter is white; the filter parameters are obtained by minimizing the mean-squared error between the filter output samples and their values obtained by linear prediction on the basis of past output samples. It is well known that these procedures often do not yield accurate filter parameters for periodic (or quasi-periodic) signals such as voiced speech. To compensate for the periodic nature of speech, an estimate of the excitation of the all-pole filter has to be made. Multi-pulse LPC obtains the best excitation for a specified bit rate by minimizing a weighted mean-squared criterion representing subjectively important differences between original and synthetic speech signals. In this paper we examine the possibility that multi-pulse excitation can approximate the all-pole filter excitation sufficiently closely and obtain the optimum filter parameters for this excitation.

Journal ArticleDOI
TL;DR: In this paper, the synthesis, analysis, and experimental measurements of a metal grid angular filter are described, and the behavior of this angular filter, both in frequency and angle, is analyzed.
Abstract: The synthesis, analysis, and experimental measurements of a metal grid angular filter are described. Angular filters are devices that can be added to existing antennas in order to reduce the radiation over a specified angular region while allowing radiation over other angular sectors to pass relatively unaffected. The behavior of this angular filter, both in frequency and angle, is analyzed in this paper, and these results are compared with measurements.

Journal ArticleDOI
TL;DR: In this article, a design principle is proposed to reduce filtering requirements at each stage and move the filter operations to low sampling-rate stages and thus result in a lower arithmetic rate.
Abstract: A theorem is introduced which is useful in deriving equivalent multirate filter structures. Frequency responses of multistage multirate filters are derived and defined by deriving their equivalent one-stage filters. A design principle is proposed to reduce filtering requirements at each stage and move the filter operations to low-sampling-rate stages and thus result in a lower arithmetic rate. Optimum FIR and IIR multistage multirate filter designs are developed based on this principle. The new design has a one-point passband specification for each decimator and/or interpolator stage resulting in a wider transition region and lower filter order. Examples are given to explain the design procedure, and comparisons are made to show the superiority of the new filters.

Proceedings ArticleDOI
V. Jain1, R. Crochiere
01 Apr 1983
TL;DR: A novel technique for the design of analysis/synthesis filter banks, such as quadrature-mirror filter (QMF) banks, is presented, based upon a sectional approximation of the filter bank wherein the synthesis section is taken to be known from the previous iteration, and the analysis section is optimized according to the performance criterion.
Abstract: A novel technique for the design of analysis/synthesis filter banks, such as quadrature-mirror filter (QMF) banks, is presented. The technique is iterative, but each iteration step is mathematically simple. It is based upon a sectional approximation of the filter bank wherein the synthesis section is taken to be known from the previous iteration, and the analysis section is optimized according to the performance criterion. The strategy is then reversed, and the synthesis section optimized. At each iteration, each of these two steps results in an eigenvalue-eigenvector problem of a matrix with dimensionality equal to the number of taps. The technique is exemplified in detail for the QMF bank design problem.

Proceedings ArticleDOI
01 Jan 1983
TL;DR: In this paper, a method for designing the geometry of SAW bandpass filters with arbitrary passband and group delay response is presented, which allows the design of filters with smooth amplitude and phase response in the passband.
Abstract: A method for designing the geometry of SAW bandpass filters with arbitrary passband and group delay response is presented. The 6-function model describes the relation between filter geometry and\ frequency response. (No second order effects are included in the design procedure). The design problem is defined so that the method of linear programming can be used. The geometry is determined by optimizing the stop band rejection under the restriction of a fixed filter length. This method allows the design of filters with smooth amplitude and phase response in the passband.

Proceedings ArticleDOI
14 Apr 1983
TL;DR: The problem of designing lowpass FIR digital filters that are very flat at zero frequency, smooth in the passband, and minimax in the stopband is treated.
Abstract: We treat the problem of designing lowpass FIR digital filters that are very flat at zero frequency, smooth in the passband, and minimax in the stopband. The cases of lowpass and lowpass-differentiator are studied in particular. An effective design algorithm is described based on linear programming with a single equality constraint on the first derivative at the origin plus a concavity constraint in the passband. An empirical design equation for estimating model order in the odd-length lowpass-differentiator case is given.

Proceedings ArticleDOI
01 Apr 1983
TL;DR: An attempt is made in the present paper to derive simplified expressions for coefficient and internal data wordlengths as a function of specified parameters.
Abstract: The field of applications of adaptive digital filters is growing and, as for fixed coefficient filters, system designers and project engineers need simple means for computational complexity assessment. An attempt is made in the present paper to derive simplified expressions for coefficient and internal data wordlengths as a function of specified parameters. Only the gradient technique in direct and cascade forms is considered.

Journal ArticleDOI
TL;DR: In this article, a complementary relation between a dual-mode bandpass and bandstop waveguide filter is found, and two trial samples of bandstop filters are constructed to demonstrate the principle.
Abstract: A complementary relation between a dual-mode bandpass and bandstop waveguide filter is found. Then a new idea for constructing a bandstop filter is developed. Two trial samples of bandstop filters are constructed to demonstrate the principle.