Showing papers on "Low-pass filter published in 1995"

Single-phase active power filters for multiple nonlinear loads

Abstract: This paper presents active power filters for single-phase power systems which are comprised of multiple nonlinear loads. The paper provides background on the operation of the filter, the details of the power circuit, the details of the control design, representative waveforms, and spectral performance for a filter which supports a 384 W AC controller and a 900 W uncontrolled bridge rectifier. Experimental data indicate that the active filter typically consumes 3% or less of the average load power, suggesting that a parallel filter is an efficient compensation approach. The spectral performance shows that the active filter brings the system into compliance with IEC-555 for decision frequencies in excess of 30 kHz. A discussion is presented outlining an alternative single-phase active filter which uses two controllable switches and is based on a half-bridge topology. >

A simple method for designing high-quality prototype filters for M-band pseudo QMF banks

Abstract: Discusses a new method for designing the prototype filters necessary to implement M-band pseudo QMF banks. This method does not rely on the traditional nonlinear optimization used in past work but rather optimizes a single parameter on a convex error surface, consistently delivering the best equiripple filter possible while minimizing the overlapped passband distortion. A very simple algorithm for designing lowpass prototype filters suitable for use in pseudo QMF banks is described. To illustrate the applicability of this algorithm, two different filters are designed, both for such applications as wideband audio coding that require high quality reconstructed signals. >

Adaptive postprocessor for block encoded images

Abstract: Blocking effects are a major degradation in block based image coding techniques. In this paper, we propose an adaptive postprocessor for the removal of blocking effects. The adaptation is achieved by changing filter coefficients according to the local characteristics of images and the blocking effects. Here, a space-variant lowpass filter is used to smooth the pixels at block boundaries where the blocking effects are highly visible. In addition, the pixels that are closed to an edge are adaptively regional or directional lowpass filtered if the blocking effects are obvious. As for the pixels at edges, they are left untouched by the postprocessor. Therefore, blocking effects can be removed without blurring the edge sharpness. Performance improvements compared with other techniques are obtained according to simulation results. The proposed postprocessor has 0-1 dB gain (depending on the images) compared with the loop filter of H.261, the linear filter, and the nonlinear space-variant postprocessor. >

An LC filter design method for single-phase PWM inverters

Abstract: In this paper, an analysis and design method of the output LC filter of single-phase PWM inverters is presented. At first, the analytical expressions for the total harmonics of the inductor current and capacitor voltage of the LC filter are derived. As the unique values of the parameters of the LC filter cannot be specified based on the total harmonic of the capacitor voltage alone an additional criterion based on the minimum reactive power of the LC filter is used to specify these parameters. Experimental results are included to verify the derived expressions. >

Voice activity detection driven noise remediator

Abstract: A method and apparatus for improving sound quality in a digital cellular radio system receiver. A voice activity detector uses an energy estimate to detect the presence of speech in a received speech signal in a noise environment. When no speech is present the system attenuates the signal and inserts low pass filtered white noise. In addition, a set of high pass filters are used to filter the signal based upon the background noise level. This high pass filtering is applied to the signal regardless of whether speech is present. Thus, a combination of signal attenuation with insertion of low pass filtered white noise during periods of non-speech, along with high pass filtering of the signal, improves sound quality when decoding speech which has been encoded in a noisy environment.

EMI filter design

Abstract: With today’s electrical and electronics systems requiring increased levels of performance and reliability, the design of robust EMI filters plays a critical role in EMC compliance. Using a mix of practical methods and theoretical analysis, EMI Filter Design, Third Edition presents both a hands-on and academic approach to the design of EMI filters and the selection of components values. The design approaches covered include matrix methods using table data and the use of Fourier analysis, Laplace transforms, and transfer function realization of LC structures. This edition has been fully revised and updated with additional topics and more streamlined content. New to the Third Edition Analysis techniques necessary for passive filter realization Matrix method and transfer function analysis approaches for LC filter structure design A more hands-on look at EMI filters and the overall design process Through this bestselling book’s proven design methodology and practical application of formal techniques, readers learn how to develop simple filter solutions. The authors examine the causes of common- and differential-mode noise and methods of elimination, the source and load impedances for various types of input power interfaces, and the load impedance aspect of EMI filter design. After covering EMI filter structures, topologies, and components, they provide insight into the sizing of components and protection from voltage transients, discuss issues that compromise filter performance, and present a goal for a filter design objective. The text also includes a matrix method for filter design, explains the transfer function method of LC structures and their equivalent polynomials, and gives a circuit design example and analysis techniques. The final chapter presents packaging solutions of EMI filters.

Parasitic-capacitance-insensitive current-mode filters using operational transresistance amplifiers

Abstract: A parasitic-capacitance-insensitive MOSFET-C integrator and differentiator using operational transresistance amplifiers are proposed and experimental results demonstrated. The bandwidths of these circuits are also independent of their gains. The required capacitances are smaller than those in previous work. Two new configurations of universal current-mode biquad filters based on these circuits are presented. A second-order bandpass filter constructed using universal biquad filters and another third-order Chebyshev lowpass filter with 0.1 dB passband ripple are breadboarded and the measured results presented.

Device and method for providing high speed data transfer through a drop line of a power line carrier communication system

Abstract: A method and device for increasing the bandwidth of a drop line in a power line carrier communication system for facilitating high speed transmission of communication data. The drop line is divided into two sections, each containing both high frequency communication data components and low frequency power components. A high pass filter is provided for passing the high frequency components to a receiver device and a low pass filter is provided for passing the low frequency power components to an electrical system in a residence. The low pass filter is also used for blocking, from the drop line, high frequency noise generated by electrical appliances operating from the electrical system.

Design of efficient M-band coders with linear-phase and perfect-reconstruction properties

Abstract: This paper presents a new design technique for obtaining M-band orthogonal coders where M=2/sup i/. The structures obtained using the proposed technique have the perfect reconstruction property. Furthermore, all filters that constitute the subband coder are linear-phase FIR-type filters. In contrast with conventional design techniques that attempt to find a unitary alias-component matrix in the frequency domain, we carry out the design in the time domain, based on time-domain orthonormality constraints that the filters must satisfy. The M-band design problem is reduced to the problem of finding a suitable lowpass filter h/sub 0/(n). Once a suitable lowpass filter is found, the remaining (M-1) filters of the coder are obtained through the use of shuffling operators on the lowpass filter. This approach leads to a set of filters that use the same numerical coefficient values in different shift positions, allowing very efficient numerical implementation of the subband coder. In addition, by imposing further constraints on the lowpass branch impulse response h/sub 0/(n), we are able to construct continuous bases of M-channel wavelets with good regularity properties. Design examples are presented for four-, eight-, and 16-band coders, along with examples of continuous wavelet bases that they generate. >

Slow asymptotic convergence of LMS acoustic echo cancelers

Abstract: In most acoustic echo canceler (AEC) applications, an adaptive finite impulse response (FIR) filter is employed with coefficients that are computed using the LMS algorithm. The paper establishes a theoretical basis for the slow asymptotic convergence that is often noted in practice for such applications. The analytical approach expresses the mean-square error trajectory in terms of eigenmodes and then applies the asymptotic theory of Toeplitz matrices to obtain a solution that is based on a general characterization of the actual room impulse response. The method leads to good approximations even for a moderate number of taps (N>16) and applies to both full-band and subband designs. Explicit mathematical expressions of the mean-square error convergence are derived for bandlimited white noise, a first-order Markov process, and, more generally, pth-order rational spectra and a direct power-law model, which relates to lowpass FIR filters. These expressions show that the asymptotic convergence is generally slow, being at best of order 1/t for bandlimited white noise. It is argued that input filter design cannot do much to improve slow convergence. However, the theory suggests postfiltering as a remedy that would be useful for the full-band LMS AEC and may also be applicable to subband designs. >

A 3 V 12-55 MHz BiCMOS pseudo-differential continuous-time filter

Abstract: The reduction of the supply voltage forces one to develop system and circuit solutions able to achieve the same performance previously obtained with higher supply voltage. In this paper, a second-order low-pass continuous-time filter operating at a 3 V power supply is presented. The prototype filter is implemented using a highly linear pseudo-differential transconductor. The input common-mode signal is canceled at the transconductor level using a feed-forward path. The output common mode voltage is controlled at the filter level using lossy integrators. A prototype cell has been realized in 1.2 /spl mu/m BiCMOS technology. The pole frequency can be tuned in the range 12-55 MHz. A THD of -40 dB is achieved for signals up to 1 V/sub pp/ at 10 MHz. The dynamic range is approximately 60 dB.

Vibration/noise active control system for vehicles

Abstract: A vibration/noise active control system for automotive vehicles which generates a canceling signal for canceling vibrations and noises generated within the compartment of the vehicle, based on a reference signal related to the vibrations and noises by means of an adaptive filter. The canceling signal is converted into a canceling sound. The adaptive filter has a filter coefficient thereof changed based on the cancellation error between the canceling sound and the vibrations and noises within the compartment, detected by an error sensor, and the reference signal. A memory stores a plurality of filter coefficient values of the adaptive filter corresponding to a plurality of predetermined traveling conditions of the vehicle. A discriminator reads out a filter coefficient value of the adaptive filter corresponding to the detected predetermined traveling condition of the vehicle, supplies the read-out filter coefficient value to the adaptive filter to generate the canceling signal, and stores a filter coefficient value of the adaptive filter which has been changed based on the generated canceling signal.

Slope transforms: theory and application to nonlinear signal processing

Abstract: Fourier transforms are among the most useful linear signal transformations for quantifying the frequency content of signals and for analyzing their processing by linear time-invariant systems. Some nonlinear signal transforms are developed that can provide information about the slope content of signals and are useful analytic tools for large classes of nonlinear systems. Many of their theoretical properties are examined, showing a striking conceptual resemblance to Fourier transforms and their application to linear systems. These novel transforms, called slope transforms, are originally derived from the eigenvalues of morphological dilation and erosion systems, where the corresponding eigenfunctions are lines /spl alpha/t+b parameterized by their slope /spl alpha/. They obey a nonlinear superposition principle of the supremum- or infimum-of-sums type. Applied to the impulse response of dilation or erosion systems, the slope transforms provide a slope response function for these systems, which allows their analysis and design in a transform domain, the slope domain. Applied to arbitrary signals, the slope transforms provide information about upper or lower tangents to the signal's graph at varying slopes. The upper or lower envelopes of the signal can be obtained from the inverse transforms. Overall, the slope transforms provide a new transform domain for signals and morphological systems where time lines become slope impulses, time cones become slope bandpass filters, and time dilation/erosion transform into addition of slope transforms. Their application to the design of slope-selective filters is also presented. >

A multidimensional nonlinear edge-preserving filter for magnetic resonance image restoration

Abstract: The paper presents a multidimensional nonlinear edge-preserving filter for restoration and enhancement of magnetic resonance images (MRI). The filter uses both interframe (parametric or temporal) and intraframe (spatial) information to filter the additive noise from an MRI scene sequence. It combines the approximate maximum likelihood (equivalently, least squares) estimate of the interframe pixels, using MRI signal models, with a trimmed spatial smoothing algorithm, using a Euclidean distance discriminator to preserve partial volume and edge information. (Partial volume information is generated from voxels containing a mixture of different tissues.) Since the filter's structure is parallel, its implementation on a parallel processing computer is straightforward. Details of the filter implementation for a sequence of four multiple spin-echo images is explained, and the effects of filter parameters (neighborhood size and threshold value) on the computation time and performance of the filter is discussed. The filter is applied to MRI simulation and brain studies, serving as a preprocessing procedure for the eigenimage filter. (The eigenimage filter generates a composite image in which a feature of interest is segmented from the surrounding interfering features.) It outperforms conventional pre and post-processing filters, including spatial smoothing, low-pass filtering with a Gaussian kernel, median filtering, and combined vector median with average filtering. >

The design of nonuniform modulated filterbanks

Abstract: We present an approach to nonuniform filterbank design based on modulated filters and the principle of adjacent channel aliasing cancellation. The approach is a generalization of pseudo QMF designs to nonuniform channel arrangements and the essential idea is to form nonuniform filterbanks from uniform sections joined by "transition" filters. All channel filters are formed by modulating lowpass prototypes. To meet the aliasing cancellation conditions, the transition filters are modulated from complex lowpass prototypes, although the resulting channel filters are still real. The approach provides a computationally simple method for designing filterbanks with large numbers of channels because of the fact that channel filters are modulated from a few lowpass prototypes and because the lowpass prototype designs can be scaled. That is, designs for narrow channel spacing can be produced from designs for wider channel spacing by interpolating the lowpass prototype impulse response. The development is limited to the design of nonuniform filterbanks with integer decimation factors, although we demonstrate that the technique can be used to produce excellent rational decimation factor designs by combining channel outputs using an inverse polyphase transform.

Programmable two-part matched filter for spread spectrum

Abstract: A spread-spectrum-matched-filter apparatus including a code generator, a programmable-matched filter, a frame-matched filter, and a controller. The code generator generates replicas of a pilot-chip-sequence signal and a data-chip-sequence signal, which are used to set the programmable-impulse response of the programmable-matched filter. The programmable-impulse response is alternately changed between that of the pilot-chip-sequence signal and the data-chip-sequence signal so that the programmable-matched filter alternately detects the pilot-spread-spectrum channel and the data-spread-spectrum channel embedded in a received spread-spectrum signal. The frame-matched filter detects a despread-pilot-bit-sequence signal and generates a peak-pilot-correlation signal which can be used for timing of the controller as to when to trigger changing the programmable-impulse response of the programmable-matched filter.

Filtering of harmonic currents and damping of resonances in power systems with a hybrid-active filter

Abstract: A novel hybrid-active filter topology to minimize electric utility current harmonics at high power levels is presented in this paper. The proposed topology combines both passive and active filters to obtain the lowest power converter VA rating as compared to the power converter rating in the active filter and the series-active hybrid filter configurations. This is demonstrated with experimental results from a laboratory model. Transient and steady state operation of the hybrid-active filter is also presented. The control of the proposed filter is extended to provide damping of resonances due to neighboring loads. Simulation results showing the effect of active damping are presented. >

Directional ear device with adaptive bandwidth and gain control

Abstract: An ear device protects a user from damaging sound levels while permitting the user to hear and understand conversation in a noisy environment. The device includes an enclosure system for at least partially isolating the user's ear drums from ambient sounds, at least on directional microphone, an adaptive band pass filter and a speaker. The adaptive band pass filter further includes adaptive high pass and adaptive low pass filters which adaptively and independently control the range of the pass band depending upon the characteristics of the noise in the user's environment. A speaker transmits the processed signal to the user. The adaptive filter also includes a detector connected to receive the processed signal which controls the gain of the system to prevent overcompression of the audio signal. The adaptive filter further includes an adaptive compression circuit utilizing multiple time constants to control the response time of the adaptive filter under various conditions.

Direct conversion receiver

Abstract: A direct conversion receiver for a radio system has an antenna and blocking filter connected to an amplifier. The input signal is split and mixed with an in-phase signal and a quadrature phase signal generated by an oscillator in a mixer circuit. An output from each mixer circuit is applied to a low pass filter and to an input of a limiting circuit. The output from each low pass filter is applied to respective circuits, a first of which is arranged to sum the in-phase and quadrature phase signals, and the second of which is arranged to subtract the in-phase and quadrature phase signals to generate a respective output signal having an axis intermediate the in-phase and quadrature phase signal axes. These signals, together with the in-phase and quadrature phase signals are passed through a limiting circuit respectively, to a decoder circuit for recovering the data. The output of the limiting circuits represent signals quantized to eight possible phase states separated by 45°. In an alternative embodiment a ratiometric combiner may be used instead of the summation circuits, particularly for π/4-DQPSK modulation, where the phase excursion per symbol is ±45° or ±135° depending upon the bit pair combination. In the latter embodiment a minimum of eight axes are generated giving sixteen sectors and 22.5° phase resolution.

Plasma treatment apparatus and method

Abstract: A plasma treatment apparatus comprising a chamber earthed, a vacuum pump for exhausting the chamber, a suscepter on which a wafer is mounted, a shower electrode arranged in the chamber, opposing to the suscepter, a unit for supplying plasma generating gas to the wafer on the suscepter through the shower electrode, a first radio frequency power source for adding radio frequency voltage, which has a first frequency f 1 , to both of the suscepter and the shower electrode, a second radio frequency power source for adding radio frequency voltage, which has a second frequency f 2 higher than the first frequency f 1 , at least to one of the suscepter and the shower electrode, a transformer whose primary side is connected to the first radio frequency power source and whose secondary side to first and second electrodes, and a low pass filter arranged in a circuit on the secondary side of the transformer, and serving to allow radio frequency voltage, which has the first frequency f 1 , to pass through it but to cut off radio frequency voltage, which has the second frequency f 2 , while plasma is being generated.

Control method for a combined active filter system employing a current source converter and a high pass filter

Abstract: This paper describes a control method with a combined filter system which senses load current, source current, and line voltage to create reference signals for an active filter. The transfer function of the active filter is identified and is used for the control system design. It is shown that the source current feedback is most effective in suppressing the harmonic-enlarging effects due to parallel resonance and the harmonic current generated by source harmonic voltages. A small setup controlled by a DSP was built, and the validity of the proposed method was demonstrated by experimental results. >

Micro-receiver for receiving a high frequency frequency-modulated or phase-modulated signal

Abstract: A micro-receiver for receiving a high frequency frequency modulated or phase modulated signal, including a single integrated circuit in BiCMOS technology and on which is integrated a high frequency amplifier, an oscillator, a modulator, an intermediate frequency filter, an intermediate frequency amplifier, a demodulator, a low pass filter and a low frequency amplifier. The integrated circuit includes in addition a muting or squelch circuit, a voltage multiplier as well as a standby circuit. The circuits working in a lower frequency range are implemented in CMOS technology, and operated in a very energy-saving way with a single-cell battery with a low battery voltage of 1.3 V. The voltage multiplier provides a higher operating voltage for the circuits working in a higher frequency range. Because of the high degree of integration made possible by implementing the intermediate frequency and low frequency circuits in CMOS technology, it is possible to dispose the receiver, including a battery and earphone, in a housing which can be inserted into the external auditory canal of a person.

Design of optimal digital filter using a parallel genetic algorithm

Abstract: This paper describes the use of a parallel genetic algorithm to design a direct form of a finite word length (FWL), finite impulse response (FIR) low pass digital filter. The results of the proposed design technique are compared to two other proposed methods, an integer programming technique and roundoff of the optimal floating point set of coefficients. Using the Chebyshev metric as the comparison criterion, design of the low pass filter by the parallel genetic algorithm technique proved to be superior to the other methodologies. >

Digital filter with decimated frequency response

Abstract: A method for changing the frequency of a low-pass Finite Impulse Response (FIR) filter with a fixed frequency clock utilizes a decimation-by-coefficient technique. The decimation-by-coefficient method utilizes a single set of coefficients that are stored in a coefficient Read Only Memory (ROM) (64). Data is input to an elastic buffer (60) with multiplications performed by a multiplication circuit (62). To realize a low frequency filter, all coefficients are utilized in the multiplication operations with sequential multiplies. These are accumulated in register (70), this providing a high precision filter. To increase frequency by a factor of two--to decimate the coefficients by a factor of two, it is only necessary to utilize every other coefficient, such that only a single fixed clock (78) is required.

A transfer function approach to harmonic filter design

Abstract: This paper details a new transfer function approach in passive harmonic filter design for industrial and commercial power system applications. Filter placement along with six common filter configurations are presented. Harmonic impedance, voltage division and current division transfer functions are derived and used in a practical filter design procedure which incorporates IEEE-519 distortion limits directly into the design and component specification process. A simple four step filter design procedure is outlined and used in a variable speed motor drive pumping plant application.

Speech enhancement based on temporal processing

Abstract: Finite impulse response (FIR) Wiener-like filters are applied to time trajectories of the cubic-root compressed short-term power spectrum of noisy speech recorded over cellular telephone communications. Informal listenings indicate that the technique brings a noticeable improvement to the quality of processed noisy speech while not causing any significant degradation to clean speech. Alternative filter structures are being investigated as well as other potential applications in cellular channel compensation and narrowband to wideband speech mapping.

Synthesis of fiber-optic delay line filters

Abstract: This paper deals with the synthesis of fiber-optic delay line filters and is organized as follows. We show how filters with negative coefficients can be implemented using positive structures and differential detection at the optoelectronic conversion. This possibility implies that not only lowpass, but bandpass and highpass filters with arbitrary transfer functions may be implemented. Synthesis methods for positive filters based on modified versions of well-known time and frequency domain techniques are presented which take into account the restriction on the positive nature of the filter coefficients. We make use of these results to describe the synthesis procedure for arbitrary filters using fiber-optic structures. Some designs of lowpass, bandpass and highpass filters are presented as examples, and a possible implementation of these structures with a view of future optoelectronic integration is presented. >

Universal filter using unity-gain cells

Abstract: The authors propose the use of both current and voltage followers for implementation of a universal filter. The aim is to show how, by using only unity-gain cells as active components, a universal filter can be obtained with improved performance over classical filters.