Showing papers on "Butterworth filter published in 2001"

New microstrip "Wiggly-Line" filters with spurious passband suppression

Abstract: In this paper, we present a new parallel-coupled-line microstrip bandpass filter with suppressed spurious passband. Using a continuous perturbation of the width of the coupled lines following a sinusoidal law, the wave impedance is modulated so that the harmonic passband of the filter is rejected while the desired passband response is maintained virtually unaltered. This strip-width perturbation does not require the filter parameters to be recalculated and, this way, the classical design methodology for coupled-line microstrip filters can still be used. At the same time, the fabrication of the resulting filter layout does not involve more difficulties than those for typical coupled-line microstrip filters. To test this novel technique, 3rd-order Butterworth bandpass filters have been designed at 2.5 GHz, with a 10% fractional bandwidth and different values of the perturbation amplitude. It is shown that for a 47.5 % sinusoidal variation of the nominal strip width, a harmonic rejection of more than 40 dB is achieved in measurement while the passband at 2.5 GHz is almost unaltered.

Method of channel frequency allocation for rf and microwave duplexers

Abstract: A method is provided for operating a duplexer including a first tunable bandpass filter, a second tunable bandpass filter and means for coupling the first bandpass filter and the second bandpass filter to an antenna The method comprises the steps of tuning the first tunable bandpass filter to provide a passband corresponding to an assigned transmit frequency, and tuning the second tunable bandpass filter to provide a passband offset from an assigned receive frequency, when the duplexer is operated in a transmit mode When the duplexer is operated in a receive mode, the first tunable bandpass filter is tuned to provide a passband offset from an assigned transmit frequency and the second tunable bandpass filter is tuned to provide a passband corresponding to the assigned receive frequency

The Use of Butterworth Filters for Trend and Cycle Estimation in Economic Time Series

Abstract: Long-term trends and business cycles are usually estimated by applying the Hodrick and Prescott (HP) filter to X-11 seasonally adjusted data. A two-stage procedure is proposed in this article to improve this methodology. The improvement is based on (a) using Butterworth or band-pass filters specifically designed for the problem at hand as an alternative to the HP filter, (b) applying the selected filter to estimated trend cycles instead of to seasonally adjusted series, and (c) using autoregressive integrated moving average models to extend the input series with forecasts and backcasts. It is shown in the article that the HP filter is a Butterworth filter and that, if a model-based method is used for seasonal adjustment, it is possible to give a fully model-based interpretation of the proposed procedure. In this case, one can compute forecasts and mean squared errors of the estimated trends and cycles. The procedure is illustrated with several examples.

Design and Analysis of Analog Filters: A Signal Processing Perspective

Abstract: Preface. 1. Introduction. Part I: Approximation Design and Analysis. 2. Analog Filter Design and Analysis Concepts. 3. Butterworth Filters. 4. Chebyshev Type I Filters. 5. Chebyshev Type II Filters. 6. Elliptic Filters. 7. Bessel Filters. 8. Other Filters. 9. Frequency Transformations. Part II: Implementation and Analysis. 10. Passive Filters. 11. Active Filters. Appendices. References. Index.

Efficient FIR filter for high-speed communication

Abstract: A method for reducing a propagation delay of a digital filter. The digital filter has an input path and an output path and includes a set of delay elements and a number of taps. The taps couples the input path to the output path. Each of the taps includes a coefficient, a multiplier and an adder. Each of the delay elements is disposed between two adjacent taps. The delay elements are placed in both the input path and the output path of the digital filter, such that the digital filter has fewer delay elements in the input path than a direct-form digital filter having the same number of taps in a direct-form structure and has fewer delay elements in the output path than a transposed-form digital filter having the same number of taps in a transposed-form structure, and such that the digital filter has same transfer function as the direct-form digital filter and the transposed-form digital filter.

A 2.7-V CMOS dual-mode baseband filter for PDC and WCDMA

Abstract: A fifth-order analog CMOS RC-opamp baseband filter for a dual-mode cellular phone receiver was designed with maximum component sharing in the two modes, The filter meets the bandwidth specifications of both the PDC and WCDMA standards, which represent the two extremes with respect of the channel bandwidth. The total area of 4.8 mm/sup 2/ was minimized by reducing the filter order from five to three in the PDC mode, Also, the operational amplifiers with adjustable GBW were used to minimize PDC-mode power consumption. The capacitance matrices were made only partially overlapping to reduce the resistance spread, The largest resistors were implemented with T networks and the smallest capacitors with series connections to extend the range of feasible passive component values. The measured integrated input referred noise is 17 /spl mu/V and 47 /spl mu/V in the PDC and WCDMA modes, respectively. The IIP3 is +35 dBV in the WCDMA mode, and the circuit consumes 6.8 mW and 25.4 mW in the PDC and WCDMA modes, respectively. The supply voltage is 2.7 V.

Passband filter having an asymmetrical filter response

Abstract: A filter, such as a transmit filter of a duplexer, includes an array of acoustic resonators that cooperate to establish an asymmetrically shaped filter response over a target frequency passband. The acoustic resonators are preferably film bulk acoustic resonators (FBARs). The filter response defines an insertion loss profile in which a minimum insertion loss within the target passband is located at or near a first end of the frequency passband, while the maximum insertion loss is located at or near the opposite end of the frequency passband. In the transmit filter embodiment, the minimum insertion loss is at or near the high frequency end of the filter response, which is tailored by selectively locating poles and zeros of the array of FBARs.

Further developments in an ac-based capacitance tomography system

Abstract: An ac-based electrical capacitance tomography (ECT) system has been developed at UMIST, with sinusoidal excitation and phase-sensitive demodulation (PSD). In this article the overall system design is briefly described and then discussed in detail are the key design features, needed to achieve a high signal-to-noise ratio and a high data acquisition rate, including transient process analysis of the first stage, design of an analog Butterworth low-pass filter and optimization of the data acquisition process. The experiments show that the ac-based system has signal variance 19 times better than the previous design, the charge/discharge ECT system, and that a data acquisition rate of 140 frames per second is achieved for a 12-electrode system.

The design of microstrip six-pole quasi-elliptic filter with linear phase response using extracted-pole technique

Abstract: The development of microstrip filters has been in great demand due to the rapid growth of wireless communication systems in this decade. Quasi-elliptic response filters are very popular in communication systems because of their high selectivity, which is introduced by a pair of transmission zeros. A number of ways of implementing the quasi-elliptic response filter on microstrip have been studied over the last two decades, i.e., the cascaded quadruplet filter, canonical filter, and extracted-pole filter. However, there is very little information in the literature giving the design details for microstrip extracted-pole filters. In this paper, design equations of the extracted-pole filter for microstrip are reviewed. A new class of microstrip filter is also presented here. This class of filter will have a quasi-elliptic function response and at the same time linear phase in the passband. The linear phase of the filter is introduced by an in-phase cross coupling, while the transmission zero is realized using an extracted-pole technique. Experimental results, together with a theoretical comparison between the group delay of this design, and the conventional quasi-elliptic six-pole filter are also presented.

Efficient tunable IIR and allpass filter structures

Abstract: A new recursive filter structure is proposed which can be controlled on-line using a single parameter. The structure can be used for interpolation in timing synchronisation of digital communications receivers. The technique is illustrated with an example of the implementation of a tunable fractional delay allpass filter using the Thiran design technique.

A 0 dB-IL, 2140/spl plusmn/30 MHz bandpass filter utilizing Q-enhanced spiral inductors in standard CMOS

Abstract: A 3/sup rd/-order Chebyshev bandpass filter, that employs on-chip passive elements with Q-enhancement technique, achieves an insertion loss of 0 dB and a passband of 60 MHz around a center frequency of 2140 MHz. Fabricated in a 0.25-/spl mu/m CMOS, the filter operates with 2.5-V supply and 7 mA. The filter occupies an area of 1.3 mm /spl times/2.7 mm.

Input filter design for sliding mode controlled matrix converters

Abstract: This paper presents the input filter design for sliding mode controlled matrix converters. A global sliding mode control approach is considered for the converter with input filter to guarantee near unity input power factor operation and voltage or output current control. The sliding surfaces are directly obtained from the global system equations (matrix converter and input filter) written in the phase canonical form. The association of these sliding surfaces with the state-space vectors technique allows the choice of the most appropriate switching strategy. The proposed new input filter design considers the maximum allowed displacement factor introduced by the filter, as well as the ripple present at the capacitor voltages. Simulation results are obtained and discussed.

Sliding-window transform with integrated windowing

Abstract: A system for a sliding-window transform with integrated windowing is described. The system provides a Direct Fourier Transform kernel with an integrated windowing filter having a desired number of stages. In one embodiment, the windowing filter is a lowpass filter. In one embodiment, the lowpass filter has a rectangular filter transfer characteristic. The DFT includes a complex multiplier. A first portion of the windowing filter is provided before the complex multiplier and can be implemented using real arithmetic. A second portion of the windowing filter is provided after the complex multiplier and is implemented using complex arithmetic. In one embodiment, the filter weights of the second portion of the windowing filter are unity and thus no multiplier is needed for the filter weights in the second portion of the windowing filter.

Filter-and-sum beamformer with adjustable filter characteristics

Abstract: We introduce a polynomial filter structure for filter-and-sum beamforming applied to a microphone array application. The structure is a multidimensional extension of the well known Farrow structure, which has mainly been used for fractional delay filtering and interpolation of I-D signals. The proposed method enables an easy, smooth and efficient control of the beamforming filter characteristics by adjusting only a single control variable, e.g., for dynamic beam steering. The optimization method for polynomial beamforming filter design is presented and illustrated with simulations of beamforming filter characteristics. The design example is given for a linear array of four omni-directional microphones and a polynomial FIR filter with 20-tap delay lines.

Novel analytical model for design and implementation of three-phase active power filter controller

Abstract: The paper presents a simple analytical model for the design and implementation of a three-phase active power filter controller. Voltage decouplers and pole-zero cancellation are used in current regulators to simplify the current control plant to a first-order delay type. This simplification is made by considering the delay times caused by the lowpass filter of reference current calculation circuits, line inductors of an active power filter and the feedback circuit of a DC-link voltage. From the derived analytical model, the cutoff frequency of the lowpass filter and controller parameters can be appropriately determined to increase the harmonic current compensating capability of an active power filter and accelerate the dynamic response of the DC-link voltage. Analytical and experimental results indicate that the proposed active power filter can largely improve the total harmonic distortion of current and correct the power factor to unity with balanced and unbalanced loads.

Rapid design of SAW oscillator electronics for sensor applications

Abstract: This paper presents electronic techniques and a general methodology for the rapid design of two-port surface acoustic wave oscillators. By using a methodical approach, the stability problems and much of the time-consuming experimentation common to radio frequency (RF) oscillator design was avoided. Several oscillators in the range of 200 MHz to 1 GHz were designed using variations of the same basic circuit. Circuit designs included both two-port resonator and delay line SAW devices. The electronics were small in size ( 1.75 cm ×1.75 cm ) and inexpensive ( The first step of this systematic method was to identify a readily available RFIC amplifier that met the specifications of the oscillator. The specifications included amplifier gain, bandwidth and maximum input power. Choice of the proper amplifier allowed the same circuit to be used for several SAW oscillators spanning a large frequency range. Next a passive LC filter was designed to limit the open loop gain to a small frequency region around the SAW device’s fundamental frequency. This filter eliminates the problems of mode hopping and oscillation at harmonics. The S-parameters of the oscillator were then measured in an open-loop configuration to determine the phase shift requisite for the closed loop oscillation condition of 0° phase shift. To achieve stable oscillation, a passive LC phase-shifting filter was designed. Using the properties of Butterworth filters and a simple computer program, filters of exact phase shift were designed. The last step of design was the use of microstrip layout techniques to reduce wave reflections and susceptibility to electromagnetic interference.

Improving time-domain measurements with a network analyzer using a robust rational interpolation technique

Abstract: A method to efficiently and accurately compute a time-domain waveform from a network-analyzer frequency-domain measurement is presented in this paper The method is based on a robust interpolation technique to construct a pole-residue representation of the response of the device-under-test First, the rational function is expressed in terms of Chebyshev polynomials, instead of the usual power series, to accurately determine the poles of the network over a wide frequency range The properties of a passive system are then utilized to efficiently calculate the residues The resulting pole-residue model is analytically transformed to obtain the time-domain response in any time window, beyond the limitations of the discrete Fourier transform (DFT) technique Unlike the DFT technique, the method does not require a large number of equally spaced harmonically related frequency points The parametric model can also be used to economically store large measurement data The proposed procedure is computationally inexpensive and less sensitive to numerical instability To illustrate the validity of the method, examples of frequency- and time-domain measurements of a Beatty structure and simulation data of a low-pass Butterworth filter are given

An efficient approach for designing nearly perfect-reconstruction cosine-modulated and modified DFT filter banks

Abstract: Efficient two-step algorithms are described for optimizing the stopband response of the prototype filter for cosine-modulated and modified DFT filter banks either in the minimax or in the least-mean-square sense subject to the maximum allowable aliasing and amplitude errors. The first step involves finding a good start-up solution using a simple technique. This solution is improved in the second step by using nonlinear optimization. Several examples are included illustrating the flexibility of the proposed approach for making compromises between the required filter lengths and the aliasing and amplitude errors. These examples show that by allowing very small amplitude and aliasing errors, the stopband performance of the resulting filter bank is significantly improved compared to the corresponding perfect-reconstruction filter bank. Alternatively, the filter orders and, consequently, the overall delay can be significantly reduced to achieve practically the same performance.

Radio receiver having a dynamic bandwidth filter and method therefor

Abstract: A radio receiver (100) has an IF (intermediate frequency) filter (200) for dynamically adjusting its intermediate frequency. The filter (200) includes a filter bank (301), power estimator circuits (308, 310, 312), and weighting circuits (314, 316, 318). The filter bank (301) generates sub-bands, each sub-band having a predetermined frequency range. The power estimators (308, 310, 312) provide an estimated power in each sub-band. A filter control (320) uses the power estimates to determine a percentage of each sub-band signal that is permitted to be coupled a summation circuit (319). The summation circuit (319) sums the weighted sub-band signals to provide a filtered output signal to a demodulator (212).

Constant impedance filter

Abstract: A constant impedance filter maintains a constant input impedance for frequencies that are both inside the filter passband and outside the filter passband The constant input impedance appears as a pure resistance The constant impedance filter includes a plurality of filter poles that are connected in series Each of the filter poles include an inductor, a capacitor, and a resistor The value of the inductor, the capacitor, and the resistor are selected to provide a constant input impedance over frequency for each pole of the filter, which produces a constant input impedance for the entire filter over frequency The constant impedance filter can be implemented as a low pass filter, a high pass filter, or a bandpass filter Furthermore, the constant impedance filter can be implemented in a single-ended configuration or a differential configuration

Programmable surface acoustic wave (SAW) filter

Abstract: A novel programmable SAW filter with switchable multi-element interdigital transducers (IDTs) controlled by a microprocessor or a computer is provided that realizes the tunability of both center frequency and bandwidth of the SAW filter. The filter possesses the feature of the programmability of both center frequency and 3 dB bandwidth. As an example design, the center frequency of the SAW filter ranges from 126.8 MHz to 199.1 MHz while the 3 dB bandwidth ranges from 18.8 MHz to 58.9 MHz. The multi-input configuration increases the programmability of the device and improves insertion loss. A matching network for the programmable SAW filter further improves insertion loss level and stopband attenuation. A resistance weighting method has been applied to improve in band ripple with the passband ripple being reduced from 6.44 dB to 1.37 dB after resistance weighting. The prototype of programmable SAW filter simplifies the device structure and fabrication process by eliminating the tap weighting and summing circuits, resulting in a smaller device and lower costs. Moreover, frequency response shaping is realized without apodization.

Antenna duplexer with divided and grounded transmission line

Abstract: An antenna duplexer has a first filter and a second filter coupled to respective signal terminals and to a common node. The common node is coupled to an antenna terminal. The first filter has a higher-frequency passband than the second filter. A transmission-line circuit, inserted between the first filter and the common node, increases the impedance of the first filter in its lower stopband. The transmission-line circuit includes an internal node, a first transmission line coupling the internal node to the common node, a second transmission line coupling the internal node to the first filter, and a grounding circuit such as a capacitor coupling the internal node to ground. The grounding circuit further enhances the lower stopband impedance of the first filter.

Design of linear combination of weighted medians

Abstract: This paper introduces a novel nonlinear filtering structure: the linear combination of weighted medians (LCWM). The proposed filtering scheme is modeled on the structure and design procedure of the linear-phase FIR highpass (HP) filter in that the linear-phase FIR HP filter can be obtained by changing the sign of the filter coefficients of the FIR lowpass (LP) filter in the odd positions. The HP filter can be represented as the difference between two LP subfilters that have all positive coefficients. This representation of the FIR HP filter is analogous to the difference of estimates (DoE) such as the difference of medians (DoM). The DoM is essentially a nonlinear HP filter that is commonly used in edge detection. Based on this observation, we introduce a class of LCWM filters whose output is given by a linear combination of weighted medians of the input sequence. We propose a method of designing the 1-D and 2-D LCWM filters satisfying required frequency specifications. The proposed method adopts a transformation from the FIR filter to the LCWM filter. We show that the proposed LCWM filter can offer various frequency filtering characteristics including "LP," "bandpass (BP)," and "HP" responses.

Surface acoustic wave filter with a passband formed by a longitudinally coupled filter and a resonator inductance

Abstract: A surface acoustic wave filter includes a longitudinally-coupled resonator-type surface acoustic wave filter having at least two interdigital transducers disposed on a piezoelectric substrate along the propagation direction of a surface acoustic wave, and at least one surface acoustic wave resonator connected between an input terminal and/or an output terminal and the longitudinally-coupled resonator-type surface acoustic wave filter. In this surface acoustic wave filter, a pass band is formed by utilizing at least one of the resonant modes of the longitudinally-coupled resonator-type surface acoustic wave filter and the inductance of the surface acoustic wave resonator.

Composite lc filter circuit and composite lc filter component

Abstract: PROBLEM TO BE SOLVED: To provide a composite LC filter with a small insertion loss where the effect of a terminating state of input/output terminals on a pass band is small and to provide a composite LC filter component. SOLUTION: A band pass filter F11 is connected between a common input output terminal 12 and an input output terminal 13 and a band pass filter F12 is connected between the common input output terminal 12 and an input output terminal 14 in the composite LC filter circuit 11. The band pass filter F11 comprises a series resonance circuit including an inductor L1 and a capacitor C1 and has a pass band f1 whose center frequency is the series resonance frequency. The band pass filter F12 comprises a series resonance circuit including an inductor L2 and a capacitor C2 and has a pass band f2 (>f1) whose center frequency is the series resonance frequency.

A new characteristics-variable two-input/output filter in D-module-designs, realizations, and equivalence

Abstract: This paper proposes a new n-th order two-input/output filter in a module for two-phase signal processing. The filter can make a distinction between positive and negative frequencies of two-phase signals and can allow one to process the signals based on frequency-polarity. The filter can also change dynamically its filtering characteristics by simply injecting a shift-signal to the filter. Filtering effects equivalent to ones by parallel arranged filters with vector rotators can be obtained, but in a much simpler manner. A total of unified analyses of attractive general properties of the proposed filter is also given for easy designs and realization. The effectiveness of the analyses and usefulness of the filter are examined and confirmed through experiments. The proposed two-input/output filter in module has potential usefulness to a variety of polyphase signal filtering applications including sensorless vector controls of three-phase AC motors that require to process positive or/and negative sequences based on frequency and polarity.

Frequency-tunable notch filter

Abstract: A new type of frequency-tunable active notch filter achieves frequency selectivity through interaction among input derived signal components that are passed through parallel signal channels in a forward-only direction. The notch filter differs from earlier channelized notch filters by using multiple, instead of just one, bandpass channels that maintain required forward signal flow in the main, passband-determining signal path without signal distortion at passband frequencies. The new approach has been experimentally verified with a hybrid-integrated three-channel filter whose 40-dB-deep band-reject notch can be continuously tuned, with the help of voltage-controlled variable-capacitance elements, from 9.5 to 10.5 GHz. A single-pole bandpass filter tunes in frequency with the help of only one variable capacitance element, yet still maintains constant passband width across the tuning span. One feature of the bandpass filter is the achievement of constant notch bandwidth across the entire frequency-tuning span of the notch filter.

IIR filters based on frequency-response masking approach

Abstract: In this paper, IIR digital filter design based on the frequency-response masking approach is presented. The overall filter structure is based on the periodic model filter, complementary periodic model filter, and masking filters. The periodic model filter is obtained by replacing each delay in the prototype low-pass IIR filter by M delays. Masking filters are designed to eliminate unwanted bands from the periodic model filter or complementary periodic model filter. The transition bandwidth of the overall filter is M times smaller to that of the prototype filter. This technique can be used to design sharp low-pass, high-pass, band-pass, and band-stop filters. IIR model filters are implemented as low-sensitivity wave lattice digital filters (WDF-s) consisting of two all-pass branches in parallel. In all-pass sections, each delay is replaced with M delays to obtain periodic model filter. This way maximal sample frequency of the filter is M times increased. We show that the transfer function of the model filter is an elliptic minimal Q factors (EMQF) filter suitable for the efficient WDF implementation.