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Showing papers on "Low-pass filter published in 1998"


Patent
27 Feb 1998
TL;DR: In this paper, a transthoracic impedance signal is extracted using a weighted demodulation, and the adaptive filter cutoff frequency is based on the patient's heart rate.
Abstract: A cardiac rhythm management (CRM) device detects transthoracic impedance, extracts ventilation or other information, and adjusts a delivery rate of the CRM therapy accordingly. A four-phase sequence of alternating direction current pulse stimuli is periodically delivered to a patient's thorax. A transthoracic impedance signal is extracted using a weighted demodulation. Signal processing extracts ventilation information and removes cardiac stroke information using an adaptive lowpass filter. The adaptive filter cutoff frequency is based on the patient's heart rate; a higher cutoff frequency is provided for higher heart rates. Peak/valley detection indicates tidal volume, which is integrated to extract minute ventilation (MV). Short and long term averages are formed and compared to establish a MV indicated rate. Rate adjustment ignores MV information when a noise-measurement exceeds a threshold. An interference avoidance circuit delays delivery of the stimuli when telemetry pulses or other interfering signals are detected.

429 citations


Journal ArticleDOI
12 Oct 1998
TL;DR: In this paper, a shunt active filter based on the detection of harmonic voltages at the point of installation is proposed to attenuate harmonic propagation resulting from series/parallel resonance between capacitors for power factor correction and line inductors.
Abstract: This paper focuses on a shunt active filter based on the detection of harmonic voltages at the point of installation. The objective of the active filter is to attenuate harmonic propagation resulting from series/parallel resonance between capacitors for power factor correction and line inductors in a power distribution line. The active filter acts as a low resistor to the external circuit for harmonic frequencies, and it is installed on the end bus of the power distribution line, just like a 50 /spl Omega/ terminator installed on the end terminal of a signal transmission line. Therefore, the function of the active filter is referred to as "harmonic termination" in this paper. Experimental results obtained from a laboratory system rated at 200 V and 20 kW verify that the active filter for the purpose of harmonic termination has the capability of harmonic damping throughout the power distribution line.

268 citations


Patent
09 Jul 1998
TL;DR: In this paper, an optical filter compresses data into and/or derives data from a light signal, and the filter way weight an incident light signal by wavelength over a predetermined wavelength range according to a predetermined function so that the filter performs the dot product of the light signal and the function.
Abstract: In optical filter systems and optical transmission systems, an optical filter compresses data into and/or derives data from a light signal. The filter way weight an incident light signal by wavelength over a predetermined wavelength range according to a predetermined function so that the filter performs the dot product of the light signal and the function.

179 citations


Journal ArticleDOI
TL;DR: In this paper, the group delay of the input reflection coefficients of sequentially tuned resonators has been shown to provide all the information necessary to design and tune filters, and that the group-delay value at the center frequency of the filter can be written quite simply in terms of the low pass prototype values, the LC elements of a bandpass structure, and the coupling coefficients of the inverter coupled filter.
Abstract: The concept of coupling coefficients has been a very useful one in the design of small-to-moderate bandwidth microwave filters. It is shown in this paper that the group delay of the input reflection coefficients of sequentially tuned resonators contains all the information necessary to design and tune filters, and that the group-delay value at the center frequency of the filter can be written quite simply in terms of the low-pass prototype values, the LC elements of a bandpass structure, and the coupling coefficients of the inverter coupled filter. This provides an easy method to measure the key elements of a filter, which is confirmed by results presented in this paper. It is also suggested that since the group delay of the reflection coefficient (i.e., the time taken for energy to get in and out of the coupled resonators) is easily measured, it is a useful conceptual alternative to coupling concepts.

162 citations


Journal ArticleDOI
TL;DR: A novel multifunction optical filter with a Michelson-Gires-Tournois interferometer (MGTI) for future smart wavelength-division-multiplexed network system applications with interesting features, including that theoretical visibility is always unity regardless of the mirror reflectance value.
Abstract: We propose using a novel multifunction optical filter with a Michelson–Gires–Tournois interferometer (MGTI) for future smart wavelength-division-multiplexed network system applications. The MGTI filter is a typical Michelson interferometer in which one of its reflecting mirrors is replaced with a Gires–Tournois resonator. One unique feature of this device is that it can function as a channel-passing (CP), a channel-dropping (CD), or a wide-bandpass (BP) filter, depending on the interferometer arm-length difference. Other interesting features are that (1) the linewidths of both the CP and the CD filter are twice as narrow as that of a typical Fabry–Perot filter with similar parameters, (2) theoretical visibility is always unity regardless of the mirror reflectance value, and (3) the BP filter has an excellent boxlike response function. Numerical results showing these characteristics are presented.

146 citations


Proceedings ArticleDOI
17 May 1998
TL;DR: In this article, the authors proposed a hybrid active filter for the damping of harmonic resonance in industrial power systems, which consists of a small-rated active filter and a 5th-tuned passive filter.
Abstract: This paper proposes a hybrid active filter for the damping of harmonic resonance in industrial power systems. The hybrid filter consists of a small-rated active filter and a 5th-tuned passive filter. The active filter is characterized by detecting the 5th-harmonic current flowing into the passive filter. It is controlled in such a way as to behave as a negative or positive resistor by adjusting a feedback gain from a negative to positive value, and vice versa. The negative resistor presented by the active filter cancels a positive resistor inherent in the passive filter, so that the hybrid filter acts as an ideal passive filter with infinite quality factor. This significantly improves damping the harmonic resonance, compared with the passive filter used alone. Moreover, the active filter acts as a positive resistor to prevent an excessive harmonic current from flowing into the passive filter. Experimental results obtained from a 20-kW laboratory model verify the viability and effectiveness of the hybrid active filter proposed in this paper.

141 citations


Journal ArticleDOI
TL;DR: In this paper, a computationally efficient numerical filter is presented which suppresses these defects by taking advantage of the particular appearance of ring artifacts in the Fourier transforms of the recorded sinograms.
Abstract: Defective and insufficient calibrated detector elements introduce ring or half-circle artifacts in microtomographic image reconstructions. A computationally efficient numerical filter is presented which suppresses these defects by taking advantage of the particular appearance of ring artifacts in the Fourier transforms of the recorded sinograms. The performance of the filter is demonstrated on experimental data taken with high-energy synchrotron radiation in phase-contrast (outline) mode.

134 citations


Patent
07 Nov 1998
TL;DR: In this paper, a cascade of two filters (114, 118) along with a short bulk delay (110) is used to model the feedback path of a hearing aid, and the second filter does not use a separate probe signal.
Abstract: Feedback cancellation apparatus uses a cascade of two filters (114, 118) along with a short bulk delay (110). The first filter (114) is adapted when the hearing aid is turned on in the ear. This filter adapts quickly using a white noise probe signal (216), and then the filter coefficients are frozen. The first filter models parts of the hearing-aid feedback path that are essentially constant over the course of the day. The second filter (118) adapts while the hearing aid is in use and does not use a separate probe signal. This filter provides a rapid correction to the feedback path model when the hearing aid goes unstable, and more slowly tracks perturbations in the feedback path that occur in daily use. The delay (110) shifts the filter response to make the most effective use of the limited number of filter coefficients.

131 citations


Journal ArticleDOI
TL;DR: In this paper, a direct cascading of a wide band combine filter to a TE01 mode dielectric resonator (DR) filter is proposed to suppress the spurious response of the DR cavity filter.
Abstract: This paper presents the state of the art of high-Q TE01 mode DR cavity filters for PCS wireless base station applications. In order to have TE01 mode filter to be competitive with other high-Q cavity technologies, employment of nonadjacent coupling to implement advanced filter features and easy filter machining and integration are essential. The quadruplet and trisections are regarded as basic blocks to implement symmetric and asymmetric transmission zeros in filter stop band. The relative alignment of the magnetic mode field across the coupled adjacent cavities is analyzed to identify the sign of nonadjacent coupling. A direct cascading of a wide band combine filter to a TE01 mode dielectric resonator (DR) filter is proposed to suppress the spurious response of the DR cavity filter. This approach simplifies the integration between the DR filter and the spurious suppression device and has been proved to be very cost effective. Experimental eight- and six-pole quasi-elliptic function filters show the typical performances. To take advantage of the special property of magnetic mode field alignment across the adjacent cavities, a five-pole canonical asymmetric filter with three transmission zeros in low side is implemented. We believe this filter is a new design for high-Q cavity filter, while a three-pole elliptic function filter is new for DR filter technology.

101 citations


Journal ArticleDOI
TL;DR: A class of novel millimetric uniplanar series resonators are presented, which can be used in monolithic and hybrid uni Planar microwave integrated circuits (MIC's) and are able to demonstrate low radiation and compactness characteristics, which are attractive for passive and active monolithicand hybrid integrated circuits.
Abstract: A class of novel millimetric uniplanar series resonators are presented, which can be used in monolithic and hybrid uniplanar microwave integrated circuits (MIC's). The proposed structures are able to demonstrate low radiation and compactness characteristics, which are attractive for passive and active monolithic and hybrid integrated circuits. A principle of achieving these high-quality circuits is described and also confirmed by experimental and theoretical results, which are in good agreement up to 50 GHz. To illustrate the features of the proposed series resonators and demonstrate their effectiveness, two classes of miniature coplanar waveguide (CPW) filters (namely, low-pass and bandpass) are designed using these resonators. The developed low-pass filter has some important advantages such as low insertion loss in passband, very wide stopband, high cutoff rates, small size, low number of elements, and an effective control of spurious signals. On the other hand, the newly developed bandpass filter provides an alternative, yet compact, structure to classical filters. Obviously, many other classes of filters or passive components can also be designed.

93 citations


Journal ArticleDOI
TL;DR: The development of a filter bank structure which combines the flexibility of the short-time Fourier transform (STFT) with the implementation efficiency of the polyphase filter bank decomposition, meeting these requirements and leading to a hardware-efficient implementation, is presented.
Abstract: An approach is presented to realizing a digital channelized receiver for signal intercept applications that provides a hardware efficient implementation of a uniform filter bank in which the number of filters K is greater than the decimation factor M. The proposed architecture allows simple channel arbitration logic to be used and provides reliable instantaneous frequency measurements, even in adjacent channel crossover regions. In the proposed implementation of the filter bank, K is related to M by K=FM where F is an integer. It is shown that the optimum selection of F allows the instantaneous frequency measurement to be made in the channel crossover region and the arbitration function to be based solely on the instantaneous frequency measurement. The development of a filter bank structure which combines the flexibility of the short-time Fourier transform (STFT) with the implementation efficiency of the polyphase filter bank decomposition, meeting these requirements and leading to a hardware-efficient implementation, is presented.

Journal ArticleDOI
TL;DR: It is shown that the reduced-rank output signal computed via truncated (Q)SVD is identical to that from an array of parallelly connected analysis-synthesis finite impulse response (FIR) filter pairs.
Abstract: We show that the reduced-rank output signal computed via truncated (Q)SVD is identical to that from an array of parallelly connected analysis-synthesis finite impulse response (FIR) filter pairs. The filter coefficients are determined by the (Q)SVD, and the filters provide an explicit description of the reduced-rank noise reduction algorithm in the frequency domain.

Journal ArticleDOI
TL;DR: In this article, a new comb filter design method using fractional sample delay is presented, where the specification of the comb filter is transformed into that of fractional delay filter design.
Abstract: In this paper, a new comb filter design method using fractional sample delay is presented. First, the specification of the comb filter design is transformed into that of fractional delay filter design. Then, conventional finite impulse response (FIR) and allpass filter design techniques are directly applied to design fractional delay filter with transformed specification. Next, we develop a constrained fractional delay filter design approach to improve the performance of the direct design method. Finally, several design examples and an experiment of power line interference removal in an electrocardiogram (ECG) signal is demonstrated to illustrate the effectiveness of this new design approach.

Journal ArticleDOI
TL;DR: A multistage adaptive filtering system which generates the current reference delaylessly and accurately is introduced, making it possible to extract the sinusoidal active current from the distorted waveform without harmful phase shift, even when the frequency and amplitude alter simultaneously.
Abstract: Active power filters are used to eliminate AC harmonic currents by injecting equal but opposite compensating currents. Successful control of active filters requires, among other things, an accurate current reference. In this paper, we introduce a multistage adaptive filtering system which generates the current reference delaylessly and accurately. Our filter structure combines a low-pass prefilter and an adaptive predictive filter, making it possible to extract the sinusoidal active current from the distorted waveform without harmful phase shift, even when the frequency and amplitude alter simultaneously. Although active filters are typically used to compensate for the supply harmonics, where the fundamental frequency remains almost constant, we will show that our filter structure can also be applied in applications where the frequency alters rapidly.

Patent
05 Jun 1998
TL;DR: In this article, a filter network having the capability of establishing multiple, tunable notch frequencies is described. But the authors focus on the downlink of the network, where a notch filter path is established for each notch frequency and includes a bandpass filter and inverter.
Abstract: A filter network having the capability of establishing multiple, tunable notch frequencies A notch filter path is established for each notch frequency and includes a bandpass filter and inverter An input RF signal covering a wide frequency range is applied to all the notch filter paths Each notch filter path produces an output spectrum that is equal in magnitude and 180° out of phase with respect to an undesired frequency spectrum A combiner circuit combines the outputs of each notch filter path in parallel with the RF input signal to produce an RF output signal with all desired spectra unchanged and all undesired spectra attenuated

Journal ArticleDOI
TL;DR: This work proposes a method of designing an arbitrary high-order low-pass filter (Q filter) by utilizing the concept of the phase lead compensator, and shows that this method yields more robust and improved results than the conventional load torque observer.
Abstract: The average speed detection method involves a measurement delay, which can cause a serious instability problem to the unknown load torque observer. The instability can be cured by inserting an artificial delay into the torque-filtering path of the observer. Also, by utilizing the concept of the phase lead compensator, we propose a method of designing an arbitrary high-order low-pass filter (Q filter). Through the results of simulation and experiments, we show that our proposed method yields more robust and improved results than the conventional load torque observer.

Journal ArticleDOI
TL;DR: Under quite general assumptions the problem can be formulated as a linear program, and solved with well-known efficient optimization techniques, and the performance of the optimal signals, measured by their combined bandwidth and noise immunity, is analyzed.
Abstract: We study the design of optimal signals for bandwidth-efficient linear coded modulation. Previous results show that for linear channels with intersymbol interference (ISI), reduced-search decoding algorithms have near-maximum-likelihood error performance, but with much smaller complexity than the Viterbi decoder. Consequently, the controlled ISI introduced by a lowpass filter can be practically used for bandwidth reduction. Such spectrum shaping filters comprise an explicit coded modulation, for which we seek the optimal design. We simultaneously constrain the bandwidth and maximize the minimum Euclidean distance between signals. We show that under quite general assumptions the problem can be formulated as a linear program, and solved with well-known efficient optimization techniques. Numerical results are presented, and the performance of the optimal signals, measured by their combined bandwidth and noise immunity, is analyzed. The new codes are comparable to set-partition (TCM) trellis codes. Tests of an M-algorithm decoder confirm this and show that the performance occurs at small detection complexity.

Proceedings ArticleDOI
04 May 1998
TL;DR: The main objectives are to find out which constraints in the filter topologies, if any, must be observed along the evolutionary process and to study the problem of convergence to parsimonious circuits.
Abstract: We present in this work the application of a set of different evolutionary methodologies in the problem of electronic filter design. The main objectives are to find out which constraints in the filter topologies, if any, must be observed along the evolutionary process and to study the problem of convergence to parsimonious circuits. The new area of evolutionary electronics is introduced, an evolutionary methodology based on variable length representation is presented and the results on the evolution of low-pass and band-pass filters are described.

Journal ArticleDOI
TL;DR: The wavelet adaptive filter for the removal of baseline wandering in ECG signals is described and shows a lower ST-segment distortion than the standard filter and the adaptive filter.
Abstract: A wavelet adaptive filter (WAF) for the removal of baseline wandering in ECG signals is described. The WAF consists of two parts. The first part is a wavelet transform that decomposes the ECG signal into seven frequency bands using Vaidyanathan-Hoang wavelets. The second part is an adaptive filter that uses the signal of the seventh lowest-frequency band among the wavelet transformed signals as primary input and a constant as reference input. To evaluate the performance of the WAF, two baseline wandering elimination filters are used, a commercial standard filter with a cutoff frequency of 0.5 Hz and a general adaptive filter. The MIT/BIH database and the European ST-T database are used for the evaluation. The WAF performs better in the average power of eliminated noise than the standard filter and adaptive filter. Furthermore, it shows a lower ST-segment distortion than the standard filter and the adaptive filter.

Patent
20 May 1998
TL;DR: In this article, an adaptive line noise detection and cancellation system has been proposed to identify, signal and remove contamination from an ECG signal wherein the ECG signals are conditioned to remove various portions of the signal prior to processing in various noise detectors while minimizing the signal conditioning effect of the filters on the signal.
Abstract: An adaptive line noise detection and cancellation system having a baseline wander filter, high and low pass filters, an adaptive line noise canceler and various noise detectors is provided to identify, signal and remove contamination from an ECG signal wherein the ECG signal is conditioned to remove various portions of the ECG signal prior to processing in various noise detectors while minimizing the signal conditioning effect of the filters on the ECG signal and while further providing the operator with the ability to manually or automatically activate the filters and to indicate the status of the filters on a printout or display.

Journal ArticleDOI
TL;DR: In this article, a new class of nonsymmetric maximally flat low-pass finite impulse response (FIR) filters is described, which achieves a smaller delay than symmetric filters while maintaining relatively constant group delay around /spl omega/=0.
Abstract: This paper describes a new class of nonsymmetric maximally flat low-pass finite impulse response (FIR) filters. By subjecting the magnitude and group delay responses (individually) to differing numbers of flatness constraints, the new filters are obtained. It is found that these filters achieve a smaller delay than symmetric filters while maintaining relatively constant group delay around /spl omega/=0, with no degradation of the frequency response magnitude. The design of these filters is initially investigated using Grubner bases. An analytic design technique, applicable to a subset of the forgoing filters, is provided that does not depend on Grubner basis computations.

Patent
Lane Allen Smith1
10 Feb 1998
TL;DR: In this article, an integrated circuit, e.g., an AC '97 conforming audio codec, includes a digital filter and gain module including multiple channels of gain control and multiple channels for digital filtering, which can be added or added by increasing the clock speed without changing the digital filter design.
Abstract: An integrated circuit, e.g. an AC '97 conforming audio codec, includes a digital filter and gain module including multiple channels of gain control and multiple channels of digital filtering. A gain control module includes an overflow check of data samples requiring differing lengths of clamping. Each channel of the digital filter includes a finite impulse response (FIR) filter, and an infinite impulse response (IIR) filter. The digital filtering is implemented largely in hardware independent of the number of channels required and/or independent of the required order of the filtering. Thus, filter channels can be added or additional filtering implemented merely by increasing the clock speed without changing the digital filter design. The FIR filter is capable of being reset each frame to prevent a DC buildup at internal nodes. The IIR filter performs a plurality of 2nd order biquadratic equations in an overall average of as few as four clock cycles per 2nd order biquad. A RAM is used to store the state variables for the 2nd order biquadratic equations. The state variable RAM is reset by controlling the clear input of latches at an input and/or the output of the state variable RAM, and the state variable RAM is addressed by a delta counter which is independent of the particular number of filter channels or filter orders implemented. Test patterns may be inserted between functional modules of an integrated circuit such as the disclosed audio codec by appropriate control of the preset and clear inputs to output latches of the functional modules.

Journal ArticleDOI
TL;DR: In this article, a 4-MHz, fifth-order elliptic low-pass Gm-C filter is described whose characteristics are tuned by an on-chip automatic tuning circuit.
Abstract: A 4-MHz, fifth-order elliptic low-pass Gm-C filter is described whose characteristics are tuned by an on-chip automatic tuning circuit. The tuning circuit uses only one integrator as the master of tuning instead of problematic voltage controlled oscillator (VCO) and voltage controlled filter (VCF). MOS transistors in linear operation region perform the voltage-to-current conversion in an operational transconductance amplifier, and thereby we achieved /spl plusmn/1.5 V operation. A prototype filter was implemented in a 0.8-/spl mu/m double-poly, double-metal CMOS process. The filter exhibits the dynamic range of 57.6 dB and dissipates 10 mW with /spl plusmn/1.5-V supply. The stopband attenuation is better than 45.0 dB and the passband ripple is smaller than 1.0 dB.

Patent
23 Feb 1998
TL;DR: In this paper, a digital filter and gain module is implemented in hardware independent of the number of channels required and/or the required order of the filtering. But the digital filter channels can be added or additional filtering implemented merely by increasing the clock speed without changing the digital filtering design.
Abstract: An integrated circuit, e.g. an Audio Codec (AC) '97 conforming audio codec, includes a digital filter and gain module including multiple channels of gain control and multiple channels of digital filtering. A gain control module includes an overflow check of data samples requiring differing lengths of clamping. Each channel of the digital filter includes a finite impulse response (FIR) filter, and an infinite impulse response (IIR) filter. The digital filtering is implemented largely in hardware independent of the number of channels required and/or independent of the required order of the filtering. Thus, filter channels can be added or additional filtering implemented merely by increasing the clock speed without changing the digital filter design. The FIR filter is capable of being reset each frame to prevent a direct current (DC) buildup at internal nodes. The IIR filter performs a plurality of 2 nd order biquadratic equations in an overall average of as few as four clock cycles per 2 nd order biquad. A random access memory (RAM) is used to store the state variables for the 2 nd order biquadratic equations. The state variable RAM is reset by controlling the clear input of latches at an input and/or the output of the state variable RAM, and the state variable RAM is addressed by a delta counter which is independent of the particular number of filter channels or filter orders implemented. Test patterns may be inserted between functional blocks of an integrated circuit such as the disclosed audio codec by appropriate control of the preset and clear inputs to output latches of the functional blocks.

Patent
Joel Page1, Edwin de Angel1, Wai Laing Lee1, Lei Wang1, Hong Helena Zheng1, Chung-Kai Chow1 
16 Sep 1998
TL;DR: In this paper, a phase shifter is implemented using a polyphase filter and the amount of delay imparted by the phase shifters is determined by a particular set of coefficients selected from a plurality of such coefficients.
Abstract: A phase shifter is implemented using a polyphase filter. The filter is preferably a linear phase Finite Impulse Response (FIR) filter. The amount of delay imparted by the phase shifter is determined by a particular set of coefficients selected from a plurality of such coefficients. Storage requirements are reduced by taking advantage of symmetries in the coefficients for the filters. Memory requirements are further reduced by partitioning the polyphase filter into two polyphase filters and using one to set a rough delay amount and the other to set a fine delay amount between rough delay amount settings. The particular amount of delay may be set by an external synchronization signal.

Patent
12 Jan 1998
TL;DR: In this article, an optical disc reproduction method and a reproduction apparatus capable of accurately extracting a BCA code without being affected by mirror modulation of an RF signal is presented. But this method is limited to the case of binary codes.
Abstract: The present invention provides an optical disc reproduction method and an optical disc reproduction apparatus capable of accurately extracting a BCA code without being affected by mirror modulation of an RF signal. An RF signal is peak-held via a buffer consisting of an operational amplifier according to time constants of a capacitor and a resistor. This RF signals subjected to low pass filter processing by the capacitor and the resistor prior to a predetermined binarization, thus enabling extraction of a BCA signal form the aforementioned RF signal.

Journal ArticleDOI
01 Aug 1998
TL;DR: A simple but effective operator for the reduction of blocking artifacts is presented, together with a simple hardware implementation, based on the rational filter approach: the operator is expressed as a ratio between a linear and a polynomial function of the input data.
Abstract: A simple but effective operator for the reduction of blocking artifacts is presented, together with a simple hardware implementation. The method is based on the rational filter approach: the operator is expressed as a ratio between a linear and a polynomial function of the input data. Such filters have been proved to outperform other conventional methods in other applications, such as noise smoothing, thanks to their capability of adapting gradually to the local image characteristics. The filter is capable of biasing its behaviour in order to achieve good performance both in uniform areas, where linear smoothing is needed, and in textured zones, where nonlinear and directional filtering is required. An activity detector is embedded in the expression of the operator itself so that the biasing of the behaviour of the filter is smooth and not based on fixed thresholds. A solution for the hardware implementation of the scheme is presented in detail. Despite the simplifications imposed by the hardware design, the filter retains the same efficiency as the original algorithm.

Proceedings ArticleDOI
14 Dec 1998
TL;DR: In this article, a frequency-to-voltage converter (FVC) for integrated CMOS mixed-signal (analog/digital) applications is presented, which can operate theoretically up to 100 MHz and its operating frequency range can easily be changed.
Abstract: In this paper we present a new high performance frequency-to-voltage converter (FVC) dedicated for integrated CMOS mixed-signal (analog/digital) applications. The circuit is very fast and requires a very small silicon area for integration. The output voltage generated by this FVC does not present any AC ripples and is proportional to the period of a square wave form input signal. The time response of this FVC is very small and is approximately equal to eight cycles of the input signal within a precision of 0.4%. The circuit can operate theoretically up to 100 MHz and its operating frequency range can easily be changed.

Journal ArticleDOI
TL;DR: In this correspondence, the Zetterberg-Zhang algorithm is modified and the mean square error (MSE) measure for transients is defined and the optimal transient suppressor to cancel the transients down to a desired level at the minimum complexity of implementation.
Abstract: A new method for suppressing transients in recursive infinite impulse response (IIR) digital filters is proposed. The technique is based on modifying the state (delay) variables of the filter when coefficients are changed so that the filter enters a new state smoothly without transient attacks, as originally proposed by Zetterberg and Zhang (1988). In this correspondence, we modify the Zetterberg-Zhang algorithm to render it feasible for efficient implementation. We define a mean square error (MSE) measure for transients and determine the optimal transient suppressor to cancel the transients down to a desired level at the minimum complexity of implementation. The application of the method to all-pole and direct-form II (DF II) IIR filter sections is studied in detail. Time-varying recursive filtering with transient elimination is illustrated for tunable fractional delay filters and variable-bandwidth lowpass filters.

Journal ArticleDOI
TL;DR: Shifting the expansion frequency from the real frequency axis to the lower half of the complex frequency plane reduces the pollution due to dominant resonant modes and results in a much wider convergence range for the moment-matching AWE technique.
Abstract: This paper describes an efficient algorithm to evaluate the spectral response of passive microwave devices. The method is based on the combination of the tangential-vector finite-element method (TVFEM) for modeling three-dimensional (3-D) microwave passive components and the asymptotic waveform evaluation (AWE) technique for efficiently computing the spectral responses. Unlike previous AWE approaches, which use direct matrix factorization to solve for the moments, we employ a preconditioned conjugate gradient (PCG) method. It is observed that the iterative PCG solver converges much faster by solving only the additional components of the higher moments outside the span of previous moments. Moreover, this paper discusses the effect of shifting the expansion frequency from the real frequency axis to the lower half of the complex frequency plane. Through several numerical examples, a waveguide with an obstacle inside, mitered 90/spl deg/ E- and H-plane waveguide bend, microstrip low-pass filter, and microstrip patch antenna, we show that shifting reduces the pollution due to dominant resonant modes and, consequently, results in a much wider convergence range for the moment-matching AWE technique.