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Showing papers on "Voltage-controlled filter published in 1996"


Book ChapterDOI
15 Apr 1996
TL;DR: It is shown that most classical techniques used to design finite impulse response (FIR) digital filters can also be used toDesign significantly faster surface smoothing filters and an algorithm to estimate the power spectrum of a signal is described.
Abstract: Smooth surfaces are approximated by polyhedral surfaces for a number of computational purposes. An inherent problem of these approximation algorithms is that the resulting polyhedral surfaces appear faceted. Within a recently introduced signal processing approach to solving this problem [7, 8], surface smoothing corresponds to low-pass filtering. In this paper we look at the filter design problem in more detail. We analyze the stability properties of the low-pass filter described in [7, 8], and show how to minimize its running time. We show that most classical techniques used to design finite impulse response (FIR) digital filters can also be used to design significantly faster surface smoothing filters. Finally, we describe an algorithm to estimate the power spectrum of a signal, and use it to evaluate the performance of the different filter design techniques described in the paper.

239 citations


Proceedings ArticleDOI
03 Mar 1996
TL;DR: In this paper, a synchronous reference frame based controller implements the dynamically varying, negative or positive inductance, by generating active filter inverter voltage commands, which can selectively synthesize multiple active inductances at dominant harmonic frequencies without affecting passive filter impedances at all other frequencies.
Abstract: This paper presents a new control scheme for a parallel hybrid active filter system intended for harmonic compensation of large nonlinear loads upto 20 MVA to meet IEEE 519 recommended harmonic standards. The control scheme is based on the concept of synthesizing a dynamically variable inductance and is used for an active filtering application. A synchronous reference frame based controller implements the dynamically varying, negative or positive inductance, by generating active filter inverter voltage commands. This controller based parallel hybrid active filter system can selectively synthesize multiple active inductances at dominant harmonic frequencies without affecting passive filter impedances at all other frequencies. The controller can be used to provide 'current limiting' function to prevent passive filter overloading under ambient harmonic loads and/or supply voltage distortions. Three implementation variations of parallel hybrid active filter system are presented. This paper also proposes the use of power factor correction capacitors as passive filters for parallel hybrid active filter system, controlled to provide multiple tuned harmonic sinks and to increase cost-effectiveness for high power applications. Simulation results with both PWM and square-wave inverters validate the controller operation for mis-tuned passive filters, single and multiple frequency tuning, to achieve harmonic compensation of a 325 kVA harmonic load under supply voltage harmonics and ambient harmonic loads.

205 citations


Patent
13 Sep 1996
TL;DR: In this article, a method and system for adaptively reducing noise in frames of digitized audio signals that include both speech and background noise is presented, where the filter circuit is adjusted by a filter control circuit adapted for a current frame to exhibit a selected frequency response curve.
Abstract: A method and system are provided for adaptively reducing noise in frames of digitized audio signals that include both speech and background noise. Frames of digitized audio signals are passed through an adjustable, high-pass filter circuit to filter a portion of background noise located in a low frequency range of the digitized signal. The filter circuit is adjusted by a filter control circuit adapted for a current frame to exhibit a selected frequency response curve. The filter control circuit includes a speech detector for detecting the presence or absence of speech in the frames of digitized audio signals. The filter circuit is adjusted when no speech is detected in the current frame. In a first preferred embodiment, the filter control circuit controls the filter circuit by calculating a noise estimate corresponding to the background noise, and adjusting the filter circuit based on the noise estimate. As the noise estimates increase, the filter circuit is adjusted to extract increasing amounts of energy falling in low frequency ranges of speech. In a second preferred embodiment, the filter circuit is adjusted as a function of a noise profile estimate. A noise profile estimate for a current frame is determined as a function of speech detection and is compared to a reference noise profile. Based on this comparison, the filter circuit is adaptively adjusted.

154 citations


Journal ArticleDOI
TL;DR: In this paper, a second-order active bandpass filter using integrated inductors was implemented in Si bipolar technology, which uses special techniques to make the quality factor and the center frequency tunable.
Abstract: A second-order active bandpass filter using integrated inductors was implemented in Si bipolar technology. The filter uses special techniques to make the quality factor and the center frequency tunable. For a nominal center frequency of 1.8 GHz and a quality factor of 35, the filter has 1 dB compression dynamic range of 40 dB, and draws 8.7 mA from a 2.8 V supply.

145 citations


Journal ArticleDOI
TL;DR: In this paper, the authors present design techniques and performance bounds for implementing Q-enhanced, LC bandpass filters in silicon IC technologies, which offer significant advantages over switched capacitor and Gm-C based designs, including higher frequency of operation and lower power consumption.
Abstract: This paper presents design techniques and performance bounds for implementing Q-enhanced, LC bandpass filters in silicon IC technologies. These filters offer significant advantages over switched capacitor and Gm-C based designs, including higher frequency of operation and lower power consumption for a given dynamic range. A prototype 200 MHz, fourth-order filter implemented in a 2 /spl mu/m n-well CMOS process is described, and measured performance is compared with theoretical predictions. The prototype filter operates at a selectivity Q of 100 and draws less than 8 mA when operating from 3 to 5 V supplies, making it potentially suitable for use as a first IF filter in modern cellular and PCS receivers.

123 citations


Journal ArticleDOI
TL;DR: In this paper, a new universal voltage-mode second-order filter with three inputs and one output employing two current conveyors, two capacitors and three resistors is presented, with the same number of passive elements, the new filter employs one less current conveyor than the latest published paper.
Abstract: A new universal voltage-mode second-order filter with three inputs and one output employing two current conveyors, two capacitors and three resistors is presented. With the same number of passive elements, the new filter employs one less current conveyor than the latest published paper. Also, the new filter does not need a voltage follower. The new circuit still offers the following advantageous features of the published filter of this category: realization of all-pass, notch, high-pass, bandpass and low-pass signals from the same configuration, no requirements for component-matching conditions, orthogonal control of ω0 and Q, low active and passive sensitivities.

80 citations


Journal ArticleDOI
TL;DR: In this paper, a simple nonlinear (quadratic) filter is shown to demodulate bandpass sampled AM signals efficiently, based upon a discrete version of the recently introduced Teager-Kaiser energy operator.
Abstract: A simple nonlinear (quadratic) filter is shown to demodulate bandpass sampled AM signals efficiently. The filter is based upon a discrete version of the recently introduced Teager-Kaiser energy operator, but also closely resembles a complex digital sampling demodulator. Such a filter can also be implemented in analogue circuitry.

72 citations


Patent
25 Oct 1996
TL;DR: In this article, an active filter (12) is proposed to cancel ripple cancellation in a DC-DC converter with an output filter inductor (L) and an adaptive tuning scheme compensates for inductance variation and drift.
Abstract: An active filter (12) provides ripple cancellation in a DC-DC converter (52). The feedforward filter (12) applies to any converter with an output filter inductor (L). The filter (12) is inherently stable, performs in both continuous and discontinuous conduction modes, and applies to resonant converters. A suitable linear amplifier (60), combined with a current transformer (CT), results in a low loss implementation. An adaptive tuning scheme compensates for inductance variation and drift. The filter (12) is usable with buck, push-pull, and boost converter topologies. The result is output ripple below 10 mVRMS. The filter (12) is effective on converters with outputs as low as 2 V and currents beyond 30 A.

71 citations


Patent
01 Mar 1996
TL;DR: In this paper, a parallel hybrid active filter system for harmonic compensation of large nonlinear loads is provided, which includes a passive filter connected in series with an inverter that is controlled to produce a dynamically variable inductance at selected harmonic frequencies.
Abstract: A parallel hybrid active filter system for harmonic compensation of large non-linear loads is provided. The hybrid filter includes a passive filter connected in series with an inverter that is controlled to produce a dynamically variable inductance at selected harmonic frequencies. The passive filter may include passive capacitive and inductive elements, or may include a power factor correction capacitor alone, with all the inductance for the hybrid filter provided by the active filter inverter. The active filter inverter is controlled to provide the dynamically variable inductance by a synchronous reference frame (SRF) based controller that generates active filter inverter voltage commands that are fed to a PWM or square wave modulated voltage source inverter (VSI). The SRF controller includes an inductance command generator that generates the inductance value necessary to provide harmonic compensation from measured three phase load and filter currents transformed into a two phase synchronous rotating reference frame. A single inverter may be controlled to implement variable inductances for compensation of multiple harmonic frequencies by superposition of active filter inverter voltage commands from multiple SRF controllers. The active filter inverter is also preferably controlled by a SRF based DC bus controller to provide for maintenance of a DC bus voltage providing power to the inverter without affecting the harmonic compensation of the hybrid filter.

64 citations


Patent
07 Nov 1996
TL;DR: In this paper, a low pass filter with a notch at 60 Hz and a bandpass filter which amplifies signals in a frequency range from 10-40 Hz has been presented for low-cost heart rate monitor.
Abstract: A convenient low-cost heart rate monitor. In one embodiment, a digital filter structure includes a low pass filter having a notch at 60 Hz and a bandpass filter which amplifies signals in a frequency range from 10-40 Hz and has a notch at 60 Hz. This digital filter has a recursive structure and uses integer coefficients to simplify and speed up the calculations. A four bit microcontroller may implement the digital filter. The output of the digital filter is subject to enhancement signal processing to emphasize QRS complexes indicative of human heartbeats.

59 citations


Journal ArticleDOI
TL;DR: Two new current-mode universal filters are presented that use unity gain current and voltage followers and can simultaneously realise lowpass, highpass and bandpass responses without any changes in the circuit topology.
Abstract: Two new current-mode universal filters are presented. The proposed filters use unity gain current and voltage followers. The first filter has three inputs and one output and can realise lowpass, highpass and bandpass responses without any changes in the circuit topology. Realisation of notch and allpass responses can be easily achieved without adding any additional active elements. The second filter has three inputs and one output and can simultaneously realise lowpass, highpass and bandpass responses without any changes in the circuit topology. Realisation of notch and allpass responses can be easily achieved without adding any additional active elements. The proposed circuits enjoy low active and passive sensitivities.

Journal ArticleDOI
TL;DR: In this paper, a universal voltage-mode second-order filter circuit is presented, which has three inputs and one low-impedance output and can realize all the standard filter functions; lowpass, highpass, bandpass, notch and allpass, without changing the passive elements.
Abstract: A new universal voltage-mode second-order filter circuit is presented. The circuit has three inputs and one-low-impedance output and can realize all the standard filter functions; lowpass, highpass, bandpass, notch and allpass, without changing the passive elements. The proposed circuit uses only five passive components and enjoys independent control of the natural frequency and the bandwidth, and orthogonal control of the natural frequency and the quality factor as well as low active and passive sensitivities.

Patent
31 May 1996
TL;DR: In this article, a switching device for changing the cut-off frequency of the low-pass filter is provided for the high-frequency input circuit, which is controlled to be on or off in response to the receiving frequency, thereby changing the passband of the high frequency input circuit.
Abstract: A double super-heterodyne receiver has a high-frequency input circuit that includes a high-frequency amplifying device, a low-pass filter and a high-pass filter used for obtaining bandpass characteristics for the receiving band. A switching device for changing the cut-off frequency of, at least, the low-pass filter is further provided for the high-frequency input circuit. The switching device is controlled to be on or off in response to the receiving frequency, thereby changing the passband of the high-frequency input circuit. Image disturbance characteristics can thus be improved.

Patent
Masanori Ueda1, Osamu Ikata1, Hideki Ohmori1, Yoshiro Fujiwara1, Kazushi Hashimoto1 
20 Aug 1996
TL;DR: In this article, a filter device includes at least two filter elements provided in a package, each of the filter elements passing only signals within a predetermined frequency band, the predetermined frequency bands having center frequencies which are distinct from each other.
Abstract: A filter device includes at least two filter elements provided in a package, each of the filter elements passing only signals within a predetermined frequency band, the predetermined frequency bands of the filter elements having center frequencies which are distinct from each other. An input terminal is connected to and shared by respective inputs of the filter elements. An output terminal is connected to and shared by respective outputs of the filter elements.

Patent
30 Jan 1996
TL;DR: In this paper, a double-domain, analog transversal equalizer comprising a plurality of serially connected, active, analog filter sections with associated tap locations, each filter section having a programmable frequency response and adaptively controlled one-bit delay.
Abstract: Apparatus for performing time and frequency-domain filtering in a sampled communication channel for the transmission of binary information. A received signal is provided to a double-domain, analog transversal equalizer comprising a plurality of serially connected, active, analog filter sections with associated tap locations, each filter section having a programmable frequency response and adaptively controlled one-bit delay. The filter sections provide sequential, cumulative frequency domain filtering as the received signal propagates through the equalizer. The transfer function of each filter section has a linear phase polynomial denominator and a real (at s=jω) polynomial numerator, facilitating independent control of the delay and frequency response of each filter section. A plurality of analog multipliers multiply signals present at the tap locations by analog tap weight signals to generate a plurality of product signals, and an analog summer adds the product signals to generate an output filtered signal before sampling occurs. An LMS error-based adaptive tap weight control circuit iteratively generates optimal analog tap weight signals, a delay control circuit provides the necessary timing signals at the bit-rate to control the time delay of each analog filter section and a location of zeros circuit controls the location of zeros in the transfer function, and hence the frequency response, of each filter section. The double-domain transversal equalizer provides aperiodic and controlled frequency response at low frequencies and does not require a noise suppressing prefilter.

Patent
22 Nov 1996
TL;DR: In this article, a radio frequency filter has a band-pass type frequency response which is controllable in a manner such that the frequency response may be moved between the transmission frequency (TX') and reception frequency (RX') of associated radio equipment.
Abstract: A radio frequency filter has a band-pass type frequency response which is controllable in a manner such that the frequency response may be moved between the transmission frequency (TX') and reception frequency (RX') of associated radio equipment. Thus the radio frequency filter may be used both as a transmission filter and as receiving filter, provided that the transmission and reception take place at different times. The radio frequency filter includes a change-over switch, which connects the radio frequency filter to a transmitter when the pass band of the radio frequency filter is in the transmission frequency, and which connects to the receiver when the pass band of the radio frequency filter is in the reception frequency.

Patent
06 Aug 1996
TL;DR: In this article, an integrated harmonic response suppression filter directly in or on the dielectric ceramic monolithic block is proposed, which can result in a substantial savings in space, cost and part count in an electronic telecommunications device.
Abstract: A ceramic filter (100) with integrated harmonic response suppression has a ceramic monolithic block filter having a predetermined passband defined by tuned resonators located between an input and an output (116); and at least one of a harmonic trap filter, a lowpass filter and a lowpass microstrip filter, each having an inductive and a capacitive component. This is achievable with a design which incorporates an integrated harmonic response suppression filter directly in or on the dielectric ceramic monolithic block. This can result in a substantial savings in space, cost, and part count in an electronic telecommunications device.


Proceedings ArticleDOI
12 May 1996
TL;DR: An adaptive analog notch filter is proposed for applications requiring high frequency adaptive signal processing and is capable of rejecting a sinusoid and with a second notch filter section attached, is able to reject a pair of sinusoids.
Abstract: An adaptive analog notch filter is proposed for applications requiring high frequency adaptive signal processing. Based on a log filter circuit topology, this filter design suggests new circuitry for the implementation of analog adaptive filters. Simulation results demonstrate that the filter is capable of rejecting a sinusoid, and with a second notch filter section attached, is capable of rejecting a pair of sinusoids. The circuit design and the adaptation method are described.

Patent
30 Jul 1996
TL;DR: In this article, a matched filter and a sliding correlator are used in parallel, and the first acquisition and holding operation is executed by the matched filter, a correlating operation is performed by a sliding correlation, and a voltage supply to the matching filter is stopped.
Abstract: A filter circuit largely reducing electric power consumption compared with a conventional one, as well as realizing the initial acquisition in high enough speed. In a filter circuit according to the present invention, a matched filter and a sliding correlator are used in parallel; the first acquisition and holding is executed by a matched filter, a correlating operation is executed by a sliding correlator and a voltage supply to the matched filter is stopped.

Journal ArticleDOI
TL;DR: In this article, a universal current-mode filter with single input and arbitrary number of outputs is presented, which uses second-generation current-conveyors, grounded resistors and grounded capacitors.

Patent
Achim Degenhardt1
23 Feb 1996
TL;DR: In this paper, an adaptive balance filter is proposed, which includes a transmission path, a reception path, and an adaptive filter having a signal output, a coefficient output for outputting filter coefficients, an error signal input, and a signal input coupled to the transmission path.
Abstract: An adaptive balance filter includes a transmission path, a reception path, and an adaptive filter having a signal output, a coefficient output for outputting filter coefficients, an error signal input, and a signal input coupled to the transmission path. There is a main filter which has a signal output, a signal input coupled to the transmission path, and a coefficient input terminal. There is also a first subtractor which has one input coupled to the reception path, another input coupled to the signal output of the main filter, and an output forming a further course of the reception path. A second subtractor has one input coupled to the reception path, another input coupled to the signal output of the adaptive filter, and an output coupled to the error signal input of the adaptive filter. A transfer device is connected between the adaptive filter and the main filter and has a control input for loading the filter coefficients of the adaptive filter into the main filter upon command of a corresponding copy signal. A transfer control device having first, second and third inputs is provided. The transfer control device ascertains echo attenuations of the adaptive filter and the main filter from the first, second and third inputs, comparing two echo attenuations with one another, and sending a corresponding copy signal to the transfer device in the event that the echo attenuation of the adaptive filter is higher than the echo attenuation of the main filter.

Journal ArticleDOI
TL;DR: The two-parallelogram filter banks are constructed and design, which is the class of 2-D filter banks in which the supports of the analysis and synthesis filters consist of two parallelograms, and the cosine-modulated versions of a prototype that has a parallelogram support are derived.
Abstract: It is well known that the analysis and synthesis filters of orthonormal DFT filter banks can not have good frequency selectivity The reason for this is that each of the analysis and synthesis filters have only one passband Such frequency stacking (or configuration) in general does not allow alias cancellation when the individual filters have good stopband attenuation A frequency stacking of this nature is called nonpermissible and should be avoided if good filters are desired In a usual M-channel filter bank with real-coefficient filters, the analysis and synthesis filters have two passbands It can be shown that the configuration is permissible in this case Many designs proposed in the past demonstrate that filter banks with such configurations can have perfect reconstruction and be good filters at the same time We develop the two-parallelogram filter banks, which is the class of 2-D filter banks in which the supports of the analysis and synthesis filters consist of two parallelograms The two-parallelogram filter banks are analyzed from a pictorial viewpoint by exploiting the concept of permissibility Based on this analysis, we construct and design a special type of two-parallelogram filter banks, namely, cosine-modulated filter banks (CMFB) In two-parallelogram CMFB, the analysis and synthesis filters are cosine-modulated versions of a prototype that has a parallelogram support Necessary and sufficient conditions for perfect reconstruction of two-parallelogram CMFB are derived

Book
01 Jan 1996
TL;DR: Introduction to filters and filter design software analogue filter approximation functions analogue lowpass, highpass, bandpass and bandstop filters analogue frequency response calculation and display analogue filter implementation using active filters.
Abstract: Introduction to filters and filter design software analogue filter approximation functions analogue lowpass, highpass, bandpass and bandstop filters analogue frequency response calculation and display analogue filter implementation using active filters introduction to discrete-time systems infinite impulse response (IIR) digital filter design finite impulse response (FIR) digital filter design digital filter implementation using C. Appendices: technical references code organization.

Journal ArticleDOI
TL;DR: In this paper, an integrated-optic variable bandwidth and tunable center frequency filter using a transversal-form programmable optical filter with 32 taps was presented. But the authors did not report the realisation of an integrated optical variable bandwidth.
Abstract: The authors report the realisation of an integrated-optic variable bandwidth and tunable centre frequency filter using a transversal-form programmable optical filter with 32 taps. The bandwidth is varied from 44 to 84 GHz with a fixed centre frequency. Also, the centre frequency is tuned over 105 GHz with a fixed bandwidth.

Proceedings ArticleDOI
05 May 1996
TL;DR: In this article, a 14th order CMOS transconductance-C (Gm-C) bandpass filter for video applications is described, which achieves 75 dB dynamic range over 700 kHz noise bandwidth.
Abstract: A 14th order CMOS transconductance-C (Gm-C) bandpass filter for video applications is described. By using highly linear Gm-C integrators, the filter achieves 75 dB dynamic range over 700 kHz noise bandwidth. The measured IM3 @ 600 kHz is -61 dB for a 4 Vpp input signal. On-chip automatic frequency tuning provides more than 300% center frequency range of the filter with 1% frequency accuracy. The 0.7 /spl mu/m CMOS filter measures 4.8 mm/sup 2/ and consumes 70 mW from a single 5 V power supply.

Patent
29 Oct 1996
TL;DR: In this paper, a method and apparatus for equalizing a communications signal transmitted through a transmission medium includes an integrated continuous-time filter and circuitry for developing a feedback signal to compensate for distortion in the signal caused by the transmission medium.
Abstract: A method and apparatus for equalizing a communications signal transmitted through a transmission medium includes an integrated continuous-time filter and circuitry for developing a feedback signal to compensate for distortion in the signal caused by the transmission medium. The control signal adjusts the transfer characteristics of the integrated continuous-time filter thereby compensating for loss and distortion of the signal caused by the transmission medium and further tunes the integrated continuous-time filter thereby compensating for semiconductor process variations in the integrated continuous-time filter.

Patent
18 Nov 1996
TL;DR: In this paper, a method and apparatus for calibrating an analog equalizer in a sampled amplitude read channel is disclosed wherein the filter's frequency response is measured and calibrated directly, by injecting a known periodic signal into the analog filter and measuring a spectrum value at a predetermined frequency.
Abstract: A method and apparatus for calibrating an analog equalizer in a sampled amplitude read channel is disclosed wherein the filter's frequency response is measured and calibrated directly. This is accomplished by injecting a known periodic signal into the analog filter and measuring a spectrum value at a predetermined frequency. The filter parameters are adjusted accordingly until the spectrum reaches a predetermined target value. In the preferred embodiment, the analog filter comprises at least one second order low pass filter (referred to as a biquad filter), and the filter's spectrum is adjusted relative to the well known parameters f o and Q. Specifically, the parameters f o and Q are optimized relative to a power measurement at predetermined harmonics of the input signal. In this manner, the present invention enables auto-calibration of the analog equalizer without reading any data from the disc. Furthermore, the calibration process can be executed during the storage system's normal operation without significantly degrading its overall performance.

Patent
15 May 1996
TL;DR: In this article, a differential input/differential output OTA is proposed for voltage controlled filter applications that employ two resistively degenerated matched pairs of bipolar transistors; one pair provides differential input, the other provides differential output and feedback to the first pair, thus canceling AC distortions.
Abstract: A differential input/differential output OTA includes two resistively degenerated matched pairs of bipolar transistors; one pair provides differential input, the other provides differential output and feedback to the first pair, thus canceling AC distortions. In voltage controlled filter applications that employ the new OTA, control feedthrough is significantly reduced when compared to similar low pass filters which employ conventional OTAs.

Patent
04 Oct 1996
TL;DR: In this article, a filter co-processor within a DSP takes advantage of the orthogonal nature of modulated signals during the equalization process, leading to increased processing when compared to the prior art.
Abstract: A filter co-processor (103, 109 and 109 fig.) within a Digital Signal Processor (DSP) takes advantage of the orthogonal nature of modulated signals during the equalization process. Since, after reception, only certain real/imaginary values of the received signals (112) are useful for demodulation, the filter co-processor only processes those values to estimate the transmitted signal. By processing only those values for demodulation, the filter co-processor is able to process information in a given amount of time, leading to increased processing when compared to the prior art.