Showing papers on "Low-pass filter published in 2003"

An explicit filtering method for large eddy simulation of compressible flows

Abstract: A method for large eddy simulation (LES) is presented in which the sub-grid-scale modeling is achieved by filtering procedures alone The procedure derives from a deconvolution model, and provides a mathematically consistent approximation of unresolved terms arising from any type of nonlinearity The formal steps of primary filtering to obtain LES equations, approximate deconvolution to construct the subgrid model term and regularization are combined into an equivalent filter This filter should be an almost perfect low pass filter below a cut-off wavenumber and then fall off smoothly The procedure has been applied to a pressure-velocity-entropy formulation of the Navier–Stokes equations for compressible flow to perform LES of two fully developed, turbulent, supersonic channel flows and has been assessed by comparison against direct numerical simulation (DNS) data Mach numbers are 15 and 30, and Reynolds numbers are 3000 and 6000, respectively Effects of filter cut-off location, choice of differentiation scheme (a fifth-order compact upwind formula and a symmetric sixth-order compact formula were used), and grid refinement are examined The effects are consistent with, and are readily understood by reference to, filtering characteristics of the differentiation and the LES filter All simulations demonstrate a uniform convergence towards their respective DNS solutions

Compact elliptic-function low-pass filters using microstrip stepped-impedance hairpin resonators

Abstract: A compact elliptic-function low-pass filter using microstrip stepped-impendance hairpin resonators and their equivalent-circuit models are developed. The prototype filters are synthesized from the equivalent-circuit model using available element-value tables. To optimize the performance of the filters, electromagnetic simulation is used to tune the dimensions of the prototype filters. The filter using multiple cascaded hairpin resonators provides a very sharp cutoff frequency response with low insertion loss. Furthermore, to increase the rejection-band bandwidth, additional attenuation poles are added in the filter. The filters are evaluated by experiment and simulation with good agreement. This simple equivalent-circuit model provides a useful method to design and understand this type of filters and other relative circuits.

A compact second-order LTCC bandpass filter with two finite transmission zeros

Abstract: A novel implementation and associated design formula for a compact low-temperature cofired ceramics (LTCC) lumped-element second-order bandpass filter are proposed in this paper. The filter schematic that provides two finite transmission zeros is well known. It is shown in the paper that the filter schematic is built on a pair of conventional inductive coupled resonator tanks with a feedback capacitor between input and output. While revealing its working mechanism both graphically and mathematically, a simple design procedure for such a compact filter is also given. The proposed filter has been implemented in a six-layer ceramic substrate using LTCC technology, showing promising application potentials in miniaturized mobile terminals and Bluetooth RF front-ends. The measured results agree very well with the full-wave electromagnetic designed responses.

A robust open-source algorithm to detect onset and duration of QRS complexes

Abstract: This paper presents a novel algorithm to detect onset and duration of QRS complexes. After low-pass filtering, the ECG signal is converted to a curve length signal by a transform in which a nonlinear scaling factor is introduced to enhance the QRS complex and to suppress unwanted noise. Adaptive thresholds are applied to the length signal to determine the onset and duration of the QRS complex. The algorithm was evaluated with the complete set of single channel ECGs (signal O) from the MIT-BlH Arrhythmia Database, and achieved a gross QRS sensitivity of 99.65% and a gross QRS positive predictive accuracy of 99.77%. The QRS onset determination is very stable and is insensitive to QRS morphology change. The noise tolerance of the algorithm was evaluated using the MIT-BIH Noise Stress Test Database. The C source code for the single-channel algorithm has been contributed to PhysioToolkit and is freely available from PhysioNet (www.physionet.org).

Improving passive filter compensation performance with active techniques

Abstract: This paper presents the performance analysis of a hybrid filter composed of passive and active filters connected in series. The analysis is done by evaluating the influence of passive filter parameters variations and the effects that different active power filter's gain have in the compensation performance of the hybrid scheme. The compensation performance is quantified by evaluating the attenuation factor in a power distribution system energizing high-power nonlinear loads compensated with passive filters and then improved with the connection of a series active power filter. Finally, compensation characteristics of the hybrid topology are tested on a 10-kVA experimental setup.

On the design and characterization of femtoampere current-mode circuits

Abstract: In this paper, we show and validate a reliable circuit design technique based on source voltage shifting for current-mode signal processing down to femtoamperes. The technique involves specific-current extractors and logarithmic current splitters for obtaining on-chip subpicoampere currents. It also uses a special on-chip sawtooth oscillator to monitor and measure currents down to a few femtoamperes. This way, subpicoampere currents are characterized without driving them off chip and requiring expensive instrumentation with complicated low leakage setups. A special current mirror is also introduced for reliably replicating such low currents. As an example, a simple log-domain first-order low-pass filter is implemented that uses a 100-fF capacitor and a 3.5-fA bias current to achieve a cutoff frequency of 0.5 Hz. A technique for characterizing noise at these currents is also described and verified. Finally, transistor mismatch measurements are provided and discussed. Experimental measurements are shown throughout the paper, obtained from prototypes fabricated in the AMS 0.35-/spl mu/m three-metal two-poly standard CMOS process.

A channelized digital ultrawideband receiver

Abstract: A channelized digital ultrawideband (UWB) receiver that efficiently samples the UWB signal at a fraction of the chip frequency is proposed. The received signal is channelized in the frequency domain by employing a bank of mixers and low-pass filters. After sampling at a much reduced frequency, digital synthesis filters optimally estimate the transmitted signals. The signal-to-noise ratio (SNR) of the proposed receiver has been solved and compared against an ideal conventional receiver, which is defined as a receiver that samples at the signal Nyquist rate. When finite resolution analog-to-digital converters (ADC) are employed in the presence of a large narrowband interferer, the proposed receiver significantly outperforms the ideal conventional receiver. For example, the SNR of the proposed receiver is as much as 20 dB higher than the ideal conventional receiver when a 4-bit ADC is used in the presence of a 50 dB (relative to the noise floor) brickwall narrowband interferer with a bandwidth of 15% of the chip frequency.

Overlapped complex-modulated transmultiplexer filters with simplified design and superior stopbands

Abstract: A simple method for the design of the finite-impulse-response prototype filter for maximally decimated overlapped complex-modulated transmultiplexers with near perfect reconstruction property is presented. The procedure is unified for all values of overlap factor and leads to a prototype filter with excellent frequency selectivity and fast sidelobe fall-off rate. The high stopband attenuation and fast sidelobe fall-off rate, which is justified analytically, make the proposed filters suitable candidates for high-speed data communication applications employing multicarrier modulation.

An adaptive filter for synchronous extraction of harmonics and distortions

Abstract: This paper provides an adaptive filter for synchronous detection and extraction of harmonics. The filter can be used as integral part of the control system of a power electronic apparatus (e.g., STATCOM, APF, and UPFC) to generate the desired control signals. Stability and convergence analyses of the adaptive filter are presented based on the dynamical systems theory. Performance of the filter is verified as a means for reference signal generation in a shunt active power filter.

All-polymer RC filter circuits fabricated with inkjet printing technology

Abstract: All-polymer RC filter circuits by all-inkjet printing technology are fabricated for the first time. Conductive polymers such as polyaniline and poly(3, 4-ethylenedioxythiophene) have been used as the electrode material of the capacitor as well as the resistor material. The fabrication process and the characteristics of the printed capacitor and RC filter have been demonstrated. Simulation of the printed RC circuit has been demonstrated and compared with the experimental measurement results. A detail discussion has been given about the all-polymer capacitor and RC filter.

Comparisons in circuit configuration and filtering performance between hybrid and pure shunt active filters

Abstract: This paper deals with a hybrid shunt active filter for harmonic compensation of a three-phase diode rectifier in a 480-V adjustable-speed drive system. The hybrid filter is formed by a three-phase single-tuned current LC filter for the 7-th harmonic frequency and a small-rated three-phase voltage-source PWM inverter. Computer simulation is carried out to compare the hybrid filter with a pure shunt active filter in terms of circuit configuration and filtering performance. The simulation results indicate effectiveness and viability of the hybrid filter. In addition, theoretical analysis confirms the validity of the simulation results.

Method and apparatus for channel equalization

Abstract: A method and apparatus is disclosed to overcome the effects of intersymbol interference during data transmission. Overcoming the effects of intersymbol interference makes possible higher data transmission rates for a given error rate. In one embodiment a receiver-transmitter pair is configured with a precode filter at the transmit side and a feed forward filter and a feedback filter at the receive side. Filter coefficients are calculated to reduce the undesirable effects of the channel, such as intersymbol interference. In one embodiment a training process occurs with the feedforward filter and a feedback filter, such that the first N coefficients of the feedback filter are set to zero. Thereafter, the coefficients of the feedforward filter are subject to spectral factorization and separated into minimum phase roots and maximum phase roots. The minimum phase roots comprise the precode filter coefficients and the maximum phase roots are established as feedforward filter coefficients.

Sigma-delta modulator with spectrally shaped feedback

Abstract: This paper explores the use of finite-impulse response (FIR) filters in the feedback path of a low-pass sigma-delta modulator in order to combat some nonideal effects encountered in an analog implementation. In this approach, the filter corresponding to the first integrator is a lowpass filter which smoothes out the feedback waveform by attenuating the high-frequency quantization noise. This lowpass filtering decreases the power consumption of a switched-capacitor implementation and alternatively reduces the sensitivity to clock jitter in a continuous-time structure. A design methodology ensuring the stability of the system is presented. Theoretical analysis and simulations show that the FIR filters allow a continuous-time single-bit modulator to achieve the jitter performance of a comparable multibit modulator.

Digital filter adaptively learning filter coefficient

Abstract: A linear filter feeds back an error signal generated using the output of an adaptive filter to a learning circuit. The learning circuit learns the coefficients of the adaptive filter using the error signal. The coefficients of the linear filter are determined depending on the generation method of a target signal for error minimization. In a signal interpolation type timing recovery digital filter, the coefficients of an inverse interpolation filter are determined in such a way as to minimize an error of a signal after interpolation.

Unintended filtering in a typical photodiode detection system for optical tweezers

Abstract: We characterize the frequency-dependent response of a photo detection system based on a Si-PIN photodiode and a laser with wavelength 1064 nm, a system commonly used with optical tweezers We chopped the laser beam with chopper frequencies from 200 Hz to 14 kHz, and found an exponentially delayed response of the detection system with a characteristic delay time of ∼20 μs The physical mechanism causing this time delay is silicon’s transparency to 1064 nm light: Photons are absorbed and create charge carriers not only in the diode’s depletion layer, where they are detected within nano-seconds, but predominantly in the n-layer, where they remain undetected till transported out by thermal diffusion The diode’s response is dominated by this delay which can be characterized as a first-order low-pass filter with a 3dB-frequency of 8–9 kHz, depending on laser intensity Measurements exploiting frequencies near or above this 3dB-frequency must be corrected for this unintended filter effect We describe how to do

On the eigenfilter design method and its applications: a tutorial

Abstract: The eigenfilter method for digital filter design involves the computation of filter coefficients as the eigenvector of an appropriate Hermitian matrix. Because of its low complexity as compared to other methods as well as its ability to incorporate various time and frequency-domain constraints easily, the eigenfilter method has been found to be very useful. In this paper, we present a review of the eigenfilter design method for a wide variety of filters, including linear-phase finite impulse response (FIR) filters, nonlinear-phase FIR filters, all-pass infinite impulse response (IIR) filters, arbitrary response IIR filters, and multidimensional filters. Also, we focus on applications of the eigenfilter method in multistage filter design, spectral/spacial beamforming, and in the design of channel-shortening equalizers for communications applications.

A reduced-size dual-mode bandpass filter with capacitively loaded open-loop arms

Abstract: A novel type of dual-mode microstrip bandpass filter using degenerate modes of a dual-mode microstrip square loop resonator with capacitively loaded open-loop arms is proposed. Such a dual-mode bandpass filter with a 0.75% bandwidth at the center frequency of 1.603 GHz is designed and fabricated to demonstrate the design of reduced-size microstrip filters. It is shown that the proposed filter has a size reduction of about 59% at the same center frequency, as compared with the dual-mode bandpass filters such as microstrip patch, cross-slotted patch, square loop and ring resonator filter.

Convergence analysis of a deficient-length LMS filter and optimal-length sequence to model exponential decay impulse response

Abstract: This letter analyzes the mean-square convergence of a deficient-length least mean-square adaptive filter whose length is less than that of the unknown system and proves that filter length and the envelope of impulse response are the important factors in convergence rate control For the impulse response with an exponential decay envelope, which covers a large set of physical systems, eg, acoustic echo path, an optimal filter length sequence is figured out to achieve the fastest convergence The simulations of an exact exponential decay envelope and of a real-life echo path in a car environment are performed via computer, and the results validate our analysis well

Adaptive filtering using multiplicative general parameters for zero-crossing detection

Abstract: An adaptive, robust, and computationally efficient disturbance reduction method for line-frequency zero-crossing detectors is proposed. Our adaptive system consists of a fixed finite-impulse response filter block and two multiplicative general parameters. The predictive filter is able to handle wide variations in the line frequency without harmfully adapting to the harmonic components.

High input impedance voltage-mode universal biquadratic filter using two OTAs and one CCII

Abstract: A three-input and single-output voltage-mode universal biquadratic filter with high input impedance, using only two operational transconductance amplifiers (OTAs), one plus-type second-generation current conveyor (CCII) and two capacitors, is presented. The proposed circuit can realize all the standard filter responses, that is, highpass, bandpass, lowpass, notch and allpass filters, from the same configuration. The proposed circuit has no requirements for component matching conditions.

A novel open-loop control method for a current-source active power filter

Abstract: This paper presents a new control system for a current-source active power filter. The harmonic current compensation is realized using only a feedforward control of the load currents. The LC filter resonance of the converter is damped in an open-loop manner using the dynamic equations of the supply filter. The benefits of the proposed control system are that it is simple and straightforward, the number of measurements and sensors can be minimized, and also that the LC filter size can be optimized according to proper harmonic distortion level of the supply currents without care about the stability issues of the closed-loop system. This usually leads to reduction of the filter size. Also, the changes in fundamental current components of the active filter can be effectively realized when the active power filter can also be used as a fast reactive power compensator. The control system also includes the calculation delay compensation of the digital control system. The control system is realized using a single-chip Motorola MPC555 microcontroller. The tests with the prototype show effective current harmonic compensating performance of the nonlinear loads.

Low-pass filter for a pll, phase-locked loop and semiconductor integrated circuit

Abstract: The invention provides a low-pass filter suitably used as a loop filter for a PLL or a DLL that has a filtering characteristic equivalent to that of a conventional one and can be realized in a smaller circuit area. The low-pass filter includes first filtering means (31) for accepting, as an input, an input signal to the low-pass filter and outputting a first voltage; a circuit element (311) included in the first filtering means (31) for allowing a first current to flow in accordance with the first voltage; current generating means (32) for generating a second current at a given rate to the first current; second filtering means (33) for accepting, as an input, the second current and outputting a second voltage; and adding means (34) for adding the first voltage and the second voltage and outputting an output signal of the low-pass filter, in which the second current is set to be smaller than the first current.

A 1.5-V 75-dB dynamic range third-order G/sub m/-C filter integrated in a 0.18-/spl mu/m standard digital CMOS process

Abstract: A design methodology of a CMOS linear transconductor for low-voltage and low-power filters is proposed in this paper. It is applied to the analog baseband filter used in a transceiver designed for wireless sensor networks. The transconductor linearization scheme is based on regulating the drain voltage of triode-biased input transistors through an active-cascode loop. A third-order Butterworth low-pass filter implemented with this transconductor is integrated in a 0.18-/spl mu/m standard digital CMOS process. The filter can operate down to 1.2-V supply voltage with a cutoff frequency ranging from 15 to 85 kHz. The 1% total harmonic distortion dynamic range measured at 1.5 V for 20-kHz input signal and 50-kHz cutoff frequency is 75 dB, while dissipating 240 /spl mu/W.

System and method for monitoring and indicating a condition of a filter element in a fluid delivery system

Abstract: A filter condition sensing circuit includes current and voltage sensors and a data processor. The current and voltage sensors produce signals indicating magnitudes of electrical current and voltage, respectively, provided to an electric motor used to move a fluid (a gas or liquid) through a filter element. During a first time period, the data processor uses the signals to determine an acceptable range of electrical power required by the motor. During a second time period, the data processor uses the signals to determine a magnitude of electrical power required by the motor. When the magnitude of electrical power is within the acceptable range of electrical power, the data processor generates a signal indicating a normal condition of the filter element. A described filter condition indicating system includes the circuit. A method is for determining a condition of a filter element.

Cable detector with decimating filter and filtering method

Abstract: An electromagnetic cable detection or location device is disclosed comprising a detection stage (600) to produce a detection signal, a decimating filter (610) having an input and an output, the input being coupled to the receiving stage to filter the detection signal and output a down-sampled filtered signal with a decimation radio of N, and an output stage (620), coupled to the output of the decimating filter, to output an indication signal based upon the down-sampled filtered signal. The decimating filter used is a Finite Impulse Response (FIR) filter having a frequency response corresponding to that of a cascade of m SINC filters, wherein m is an integer greater than 1. A method of decimating a signal in a device is also disclosed.