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Showing papers on "Voltage-controlled filter published in 1989"


Journal ArticleDOI
TL;DR: An analytical treatment of the two-beam coupling devices is given in a Laplace transform framework in the undepleted pump approximation assuming plane wave inputs to allow a unified treatment ofThe current status of optical novelty filters and related devices is reviewed.
Abstract: A novelty filter detects what is new in a scene and may be likened to a temporal high-pass filter. The current status of optical novelty filters and related devices, based upon four-wave mixing and two-beam coupling in photorefractive media, is reviewed. A detector that shows only what is not new, a monotony filter, may be likened to a temporal low-pass filter. Demonstrations of high- and low-pass and bandpass temporal image filters are then discussed. An analytical treatment of the two-beam coupling devices is given in a Laplace transform framework in the undepleted pump approximation assuming plane wave inputs. This allows a unified treatment of the various filter characteristics. >

138 citations


Journal ArticleDOI
TL;DR: Two simple methods for retrieving a single sinusoid corrupted with noise are proposed, based on the lattice form realization of an adaptive infinite-impulse-response (IIR) notch filter, which have considerable potential in adaptive notch filter applications, especially when the input signal-to-noise ratio is low.
Abstract: Two simple methods for retrieving a single sinusoid corrupted with noise are proposed. They are based on the lattice form realization of an adaptive infinite-impulse-response (IIR) notch filter. The IIR filter is a cascade of second-order all-pole and all-zero filters, and the coefficients of the finite-impulse-response (FIR) section are adapted. The proposed algorithms keep the poles of the filter inside the unit circle. The computer simulation results show that the algorithms have considerable potential in adaptive notch filter applications, especially when the input signal-to-noise ratio is low. >

111 citations


Journal ArticleDOI
TL;DR: A technique for realizing a fully differential filter using an operational amplifier (op amp) without common-mode feedback (CMFB) is presented, which results in a smaller area, a reduced power dissipation, and an improved speed and dynamic range of the filter.
Abstract: A technique for realizing a fully differential filter using an operational amplifier (op amp) without common-mode feedback (CMFB) is presented. This technique results in a smaller area, a reduced power dissipation, and an improved speed and dynamic range of the filter. Experimental results of a biquad switched-capacitor bandpass filter with a center frequency of 3.5 MHz and a Q of 24 are presented. The op amp has an open-loop unity-gain bandwidth of 230 MHz, a phase margin greater than 50 degrees , and a DC gain of 76 dB. The prototype filter occupies only 0.26 mm/sup 2/ and dissipates 6 mW with a 5-V supply. >

60 citations


Journal ArticleDOI
TL;DR: In this article, the dc-side and ac-side active filters were compared with the passive filters, both on the dc side and the ac side of the HVDC terminal, and the results showed that the DC side active filter can filter the three dominant harmonics at the 12th, the 24th and the 36th frequencies.
Abstract: The active filter concepts are compared with the passive filters, both on the dc-side and the ac-side of the HVDC terminal. The Dickinson terminal of the CU HVDC transmission line is used as the basis for this study. Based on this study, the following conclusions are reached for the dc-side and the ac-side active filters: DC-Side Active Filters 1. The active filter is designed for filtering of the three dominant harmonics at the 12th, the 24th and the 36th harmonic frequencies. Computer simulations indicate that these three dominant harmonics can be attenuated to less than 1 % at the transmission line terminal, compared to the harmonic voltages produced by the converter. It should be noted that any other harmonic frequencies which may be troublesome can be included in the injection current to filter these harmonics. Higher order harmonics have small amplitudes; hence, the required total injection current and the total kVAs of the components in the active filter are not changed significantly by adding higher order harmonics. 2. The effective cost of the dc-side active filter is estimated to be 186 k$, which also includes the present worth of the additional 17.3 kW losses in the active filter, as compared to the passive filter. The active filter costs is lower by only 27 k$, as compared to the passivefilter cost of 213 k$. Given the uncertainty in the cost estimates, both filters are comparable in terms of cost.

53 citations


Journal ArticleDOI
TL;DR: An analytical method is developed for estimating the probability density function (pdf) of this envelope for different kinds of filter responses and for realistic combinations of phase noise severity and filter bandwidth.
Abstract: Consideration is given to the case of an optical pulse containing phase noise which is passed through either an optical filter or (following heterodyne lightwave detection) an electrical filter Because of the phase noise, the envelope of the filter output at any instant is a random variable An analytical method is developed for estimating the probability density function (pdf) of this envelope for different kinds of filter responses and for realistic combinations of phase noise severity and filter bandwidth Obvious applications are to detection analyses of coherent lightwave systems, wherein finite laser linewidths constitute an important source of impairment For each of the several types of filters considered, the envelope PDF can be accurately fitted by an exponential function approximation, where the decay constant is related in a simple way to known system parameters >

51 citations


Patent
Andre Tore Mikael1
07 Apr 1989
TL;DR: An adaptive digital filter including a non-recursive part and a recursive part, which can be updated in a simple and reliable manner, is presented in this article, where a linear combination is formed with adaptive weighting factors (W0-W3) from the output signals of the recursive filters.
Abstract: An adaptive digital filter including a non-recursive part and a recursive part, and which can be updated in a simple and reliable manner. The recursive part of the filter has a plurality of separate, permanently set recursive filters (13-16) with different impulse responses, and a linear combination is formed with adaptive weighting factors (W0-W3) from the output signals of the recursive filters (13-16). The filter is updated by a single (e(n)) being utilized for updating the non-recursive part (11) of the filter and the adaptive weighting factors (W0-W3) in the recursive part of the filter.

50 citations


Journal ArticleDOI
TL;DR: In this article, an active filter that removes harmonics from the AC line by the injection of pulse-width-modulated (PWM) controlled currents is described, and the performance characteristics of the proposed method are investigated by analysis and simulation.
Abstract: An active filter that removes harmonics from the AC line by the injection of pulse-width-modulated (PWM) controlled currents is described. These compensating currents are optimized for the removal of all harmonic components up to a specified order using a filter that requires no supply other than the line to which it is connected. The required bidirectional power flow in the filter is achieved by establishing an internal inductor current within upper and lower limits by a purely fundamental current flow into the filter from the line. The performance characteristics of the proposed method are investigated by analysis and simulation. Experimental results from a microcomputer-controlled model establish the validity of the approach. >

40 citations


Patent
29 Nov 1989
TL;DR: In this paper, a continuously adaptive phase-locked loop synthesizer is described, in which error correction pulses from a phase detector are separated into narrow pulse width and wide pulse width pulses, which are coupled to the loop filter to provide a VCO control voltage on a control line connected to the output of the filter.
Abstract: A continuously adaptive phase locked loop synthesizer is disclosed in which error correction pulses from a phase detector are separated into narrow pulse width and wide pulse width pulses. The wide pulse width pulses are coupled to the loop filter to enable a rapid charge of the loop filter to provide a VCO control voltage on a control line connected to the output of the loop filter. The narrow pulse width pulses are filtered by a narrow bandwidth filter before being applied to the loop filter thus enabling a slow charge of the loop filter. The narrow bandwidth filter is decoupled from the control line but referenced to the control line voltage.

30 citations


Patent
Fernando Gen-Kuong1
30 Oct 1989
TL;DR: In this article, a general biquadratic filter circuit is described which uses operational transconductance amplifiers and a bias current is used to control the transconductances of the amplifiers which in turn control the corner frequency.
Abstract: A general biquadratic filter circuit is described which uses operational transconductance amplifiers. A bias current controls the transconductance (gm) of the amplifiers which in turn control the corner frequency (Wo) of the circuit. The circuit is configured with three potential sources which can be preselected to either be at a ground level or at a desired voltage level. By preselecting which of the potentials are grounded and which are at a desired voltage level, the circuit can become either a lowpass, highpass, bandpass or notch filter.

26 citations


PatentDOI
Michael A. Deaett1
TL;DR: In this article, a time-varying recursive filter is proposed, where the multipliers following successive delay elements of the filter have sets of normalized covariance matrix coefficients which are stored and which have been obtained from the normalized autocorrelation coefficients for each of a plurality of time frames of the sampled transient signal which is to be synthesized from the stored coefficients.
Abstract: This invention utilizes a time-varying recursive filter where the multipliers following successive delay elements of the filter have sets of normalized covariance matrix coefficients which are stored and which have been obtained from the normalized autocorrelation coefficients for each of a plurality of time frames of the sampled transient signal which is to be synthesized from the stored coefficients. The normalization constant during a frame is also applied as a scale factor to the recursive filter. The input of the filter is a pseudo-random noise generator signal applied to the recursive filter at the sample rate. A plurality of successive time frames of operation of the recursive filter and with a set of coefficients for each time frame provides the entire synthesized transient signal. An analog and digital implementation of the synthesizer are described.

16 citations


Patent
Michael Edwin Barnard1
29 Sep 1989
TL;DR: A zero-IF radio receiver circuit comprises an input filter (3), quadrature mixers (9,10), d.c.-blocking capacitors (24,26), and a demodulator (22).
Abstract: A zero-IF radio receiver circuit comprises an input filter (3), quadrature mixers (9,10), d.c.-blocking capacitors (24,26) and a demodulator (22). The circuit is partly integrated on a semiconductor chip and an inductive component of the input filter comprises one or more of the chip bond-wires. In order to compensate for the inevitable variation of the inductance of these bond-wires from circuit to circuit part of the signal from the local oscillator (12) is added to the input signal before its application to the input filter and the d.c. component of the resulting output from one mixer (10), which component is representative of the phase shift of the local oscillator signal produced by the input filter and hence of any tuning error of this filter, is applied to a tuning control input (30) of the filter to reduce the error. Alternatively the tuning control signal may be adjusted to maximize the sum of the squares of the d.c. components of the signals in the two IF channels and hence minimize the attenuation produced by the input filter.

Patent
09 Jun 1989
TL;DR: In this paper, a high-pass filter is used for the recognition of a plasma which is supplied with energy by means of an ac current of preset frequency, which is detected by a suitable sensor and feeds to a highpass filter.
Abstract: The invention relates to a circuit configuration for the recognition of a plasma which is supplied with energy by means of an ac current of preset frequency. This ac current is detected by way of a suitable sensor and feeds to a highpass filter. This highpass filter permits those frequencies to pass through which lie above the preset frequency. The output signal of the highpass filter is rectified through a rectifier, freed of its residual ripple through a lowpass filter, and supplied to an evaluation device.

Proceedings ArticleDOI
08 May 1989
TL;DR: In this article, a transistor-only bandpass/lowpass filter based on g/sub m/C integrators that is capable of realizing critical frequencies in excess in 10 MHz is described.
Abstract: A new continuous-time transistor-only bandpass/lowpass filter based on g/sub m//C integrators that is capable of realizing critical frequencies in excess in 10 MHz is described. The center frequency, Q, and voltage gain are all voltage-tunable over a 10:1 range. The integrators use a novel, fully balanced, source-degenerated transconductor which maintains less than 1% nonlinearity for differential inputs up to 2 V. A second-order state variable filter with a nominal f/sub 0/=3.2 MHz and Q=5 consumes only 20 mW from a single-ended 5-V supply. >

Proceedings ArticleDOI
W.F. McGee1
08 May 1989
TL;DR: A technique is presented for designing a bank of filters with the interpolating property: given N arbitrary frequencies (f/sub m/), the mth filter has unity response at the frequency f/ sub m/ and zero response at any of the other frequencies.
Abstract: A technique is presented for designing a bank of filters with the interpolating property: given N arbitrary frequencies (f/sub m/), the mth filter has unity response at the frequency f/sub m/ and zero response at the other frequencies. The filter bank realization may be viewed either as a statistic digital filter or as an adaptive filter wherein N oscillators are summed with time-varying complex weights; the modulated weights represent the filter bank outputs. These filter banks are useful as front-end processors for telephone tone receivers, acoustic signal processing, and general spectrum analysis. >

Patent
14 Mar 1989
TL;DR: In this article, a self-adaptive predictive encoder for speech encoding, typically in a 16 kHz wide band, has been proposed, consisting of a comparator receiving, on an additive input, successive samples of a signal to be encoded and, on a subtractive input, an estimated signal delivered by a prediction loop which receives the error signal from the output of the comparator.
Abstract: A differential encoder for speech encoding, typically in a 16 kHz wide band, has a self adaptive predictive filter, comprising a comparator receiving, on an additive input, successive samples of a signal to be encoded and, on a subtractive input, an estimated signal delivered by a prediction loop which receives the error signal from the output of the comparator, the prediction loop comprising a self-adaptive predictive filter, a gain control circuit and a multiplier which receives the outputs of the predictive filter and of the gain control circuit and whose output is connected to the subtractive input The predictive filter comprises a plurality of parallel channels assigned to different mutually adjacent spectral bands, each channel comprising an input pass-band filter, a quadratic detector fed by the pass-band filter, and a peak detector whose input is connected to the quadratic detector and whose output is applied to a multiplier which also receives the output signal from the respective pass-band filter, the outputs of all multipliers driving a common summer A decoder for recovering the speech has a symmetric construction

Journal ArticleDOI
26 Jun 1989
TL;DR: In this paper, an analog-amplitude discrete-time recursive filter is proposed for frequency compensation of power processors, which copes with converters that chop up to 0.5 MHz and seems to be better than time invariant compensation networks and DSP-based filters.
Abstract: An analog-amplitude discrete-time recursive filter is proposed for the frequency compensation of power processors. It has been applied to a converter of the type described by S. Cuk (1986), and it was allowed to reach an almost ideal profile of G/sub loop/. The filter copes with converters that chop up to 0.5 MHz and seems to be better than time-invariant compensation networks and DSP-based filters. Exact small-signal analysis of the switching cell and of the compensation filter is carried out, and experimental results of gain and phase frequency dependencies are reported. >

Patent
23 Oct 1989
TL;DR: In this paper, a tunable bandpass filter for radio frequency energy with a phase-locked loop for tracking an input signal and to control the filter to keep the center frequency of the passband coincident with the frequency of input signal is shown using a Yttrium Iron Garnet filter as a frequency determining element and as a passive dispersive reference element for a frequency discriminator.
Abstract: A tunable bandpass filter for radio frequency energy with a phase-locked loop for tracking an input signal and to control the filter to keep the center frequency of the passband coincident with the frequency of the input signal is shown Using a Yttrium Iron Garnet (YIG) filter as a frequency determining element and as a passive dispersive reference element for a frequency discriminator, the bandpass filter uses the output signal of the discriminator to form a fine tuning signal to control the center frequency of the passband of the YIG filter

Journal ArticleDOI
C.L. Perry1
TL;DR: A high performance integrated bipolar design, based on a simulation of a passive LC ladder filter is described, with results.
Abstract: The design and some results obtained from a low-frequency low-pass transconductor-capacitor filter integrated in a standard bipolar process are discussed. A comparison is made between the results of the transconductor-capacitor filter design approach and of a gyrator filter design approach. The comparison indicates the advantages of the transconductor-capacitor design over a gyrator filter design. >

Patent
Michael Edwin Barnard1
06 Oct 1989
TL;DR: A zero-IF radio receiver circuit comprises an input filter (3), quadrature mixers (9,10), d.c.-blocking capacitors (24,26), and a demodulator (22).
Abstract: A zero-IF radio receiver circuit comprises an input filter (3), quadrature mixers (9,10), d.c.-blocking capacitors (24,26) and a demodulator (22). The circuit is partly integrated on a semiconductor chip and an inductive component of the input filter comprises one or more of the chip bond-wires. In order to compensate for the inevitable variation of the inductance of these bond-wires from circuit to circuit part of the signal from the local oscillator (12) is added to the input signal before its application to the input filter and the d.c. component of the resulting output from one mixer (10), which component is representative of the phase shift of the local oscillator signal produced by the input filter and hence of any tuning error of this filter, is applied to a tuning control input (30) of the filter to reduce the error. Alternatively the tuning control signal may be adjusted to maximise the sum of the squares of the d.c. components of the signals in the two IF channels and hence minimise the attenuation produced by the input filter.

Patent
15 Jun 1989
TL;DR: In this article, an output of an outline extracting circuit is transmitted through a band limiting filter for suppressing bands other than an objective frequency band to be enhanced and, thereafter, it is supplied to a noise slicing circuit for eliminating micro amplitude noises.
Abstract: According to the present invention, in an image enhancer, an output of an outline extracting circuit is transmitted through a band limiting filter for suppressing bands other than an objective frequency band to be enhanced and, thereafter, it is supplied to a noise slicing circuit for eliminating micro amplitude noises. The noises near the cut-off frequency of the filter can be more effectively suppressed and the deterioration of the waveform characteristic of the outline enhancement signal due to the filter can be also improved.

Journal ArticleDOI
TL;DR: A low-complexity multiplierless design of a half-band recursive digital filter is presented which requires only nine adders and five delays and could be implemented on a single VLSI chip to obtain a high-performance half- band filter.
Abstract: A low-complexity multiplierless design of a half-band recursive digital filter is presented. The filter structure is realized as a bireciprocal lattice wave digital filter. The filter coefficients are represented in a canonic signed-digit code with only two nonzero digits, and thus each filter coefficient can be implemented with only a single adder or subtracter. A fifth-order elliptic filter section is presented which requires only nine adders and five delays. Thus, a cascade of three or four of these fifth-order building blocks could be implemented on a single VLSI chip to obtain a high-performance half-band filter. >

Patent
01 Sep 1989
TL;DR: In this article, a time-variant filter bank is proposed to reduce the number of coefficients to be stored by combining the complex multiplier with the adjacent filter, resulting in a lower computing effort.
Abstract: Filter bank for frequency-division multiplexing or frequency- division demultiplexing of L-channel signals with transversal filters at reduced or increased sampling rate, the L-filters being implemented by means of a cascade and a complex multiplier being inserted between the individual filter cascades, characterised in that the multiplier is combined with the filter of the preceding and following cascade stage. Such filter banks are mainly used in transmultiplexers for conversion from FDM into TDM and conversely, mainly in satellite engineering. It is the aim of the invention to reduce the required computing power. By combining the complex multiplier with the adjacent filter, a time-variant filter is produced but the filters of the other cascade stages are time-invariant as before and contain real coefficients. Seen overall, a lower computing effort is required, the number of coefficients to be stored increasing only insignificantly (Figure 2a and 2b).

Proceedings ArticleDOI
23 May 1989
TL;DR: An examination is made of the coherent signal-subspace method for wideband direction-finding and a direct approach to filter design that does not involve explicit computation of the filter response is suggested.
Abstract: An examination is made of the coherent signal-subspace method for wideband direction-finding. The preprocessor consists of a multichannel finite-impulse-response filter and is suited for real-time applications. In order to accomplish faster filter design and a simple filter structure with few interconnections, band transformations are proposed. A direct approach to filter design that does not involve explicit computation of the filter response is suggested. >

Proceedings ArticleDOI
01 Jan 1989
TL;DR: An efficient procedure for the design of interpolated FIR (IFIR) filters with linear phase that uses the uniform B-spline function a6 an interpolator and solves the optimal Chehyshev approx.
Abstract: This paper presents an efficient procedure for the design of interpolated FIR (IFIR) filters with linear phase. The algorithm uses the uniform B-spline function a6 an interpolator and solves the optimal Chehyshev approx. imation problem on the appropriate subinterval. The technique can he used for the design of general lowpass, highpass and bandpass filters. Although the number ofmuLtipLicstions of the IFIR filter is dependent on the bandwidth and the center frequency of the desired filter, our program provider the mininun number of multiplications achievable and nearly always provides a substantial reduction when compared to Parks-MeClellan designs.

Journal ArticleDOI
TL;DR: In this paper, an integrate-and-dump filter operating at 1 Gbit/s has been demonstrated for NRZ signal pulses, and noise filtering with the I&D circuit yielded the same error rate performance as an 800 MHz lowpass filter.
Abstract: An integrate-and-dump filter operating at 1 Gbit/s has been demonstrated. For NRZ signal pulses, noise filtering with the I&D circuit yielded the same error rate performance as an 800 MHz lowpass filter. Unlike the lowpass filter, the I&D filter can reduce the degradation caused by certain kinds of timing jitter, and introduces no intersymbol interference.

Proceedings ArticleDOI
08 May 1989
TL;DR: An analytical formula for the filter design is developed, which allows one to design the required low-pass FIR prototype filter with much less computational complexity and to obtain better filter bank performance.
Abstract: Consideration is given to the filter design problem for a recently proposed polyphase filter bank with an arbitrary number of subband channels. On the basis of the perfect reconstruction condition for the 1-D case, an analytical formula for the filter design is developed. Compared with direct numerical design algorithms, this analytical formula allows one to design the required low-pass FIR prototype filter with much less computational complexity and to obtain better filter bank performance. A design example is given for illustration. >

Journal ArticleDOI
TL;DR: The authors present an efficient computer-aided-design procedure that systematically utilizes a lookup table containing scattering parameters from variously dimensioned E-plane fins to achieve an optimal combination of filter parameters by proper selection of the E-planes from the table and appropriate determination of the other filter elements.
Abstract: The authors present an efficient computer-aided-design procedure that systematically utilizes a lookup table containing scattering parameters from variously dimensioned E-plane fins. The main idea is to achieve an optimal combination of filter parameters by proper selection of the E-plane fins from the table and appropriate determination of the other filter elements in order to satisfy the given filter specifications. The technique of selecting the proper fin from the table is explained. The relationship between the desired center frequency of the filter and the approximate resonant frequency of the single fin in the table is shown. The relationship between each design parameter and the filter characteristic is presented. The algorithm is applied to the design of a bandpass filter with two E-plane fins operating in the Ka-band. The algorithm is verified by comparing the characteristics of the designed filter with the experimental results. >

01 Jan 1989
TL;DR: A new structure for the VLSI implementation of bit/serial adaptive IIR filter is presented, built at a bit level consisting of only gated full adders and the coefficients of the filter can be updated in real time for the time invariant and adaptive filtering.
Abstract: A new structure for the VLSI implementation of bit/serial adaptive IIR filter is presented. The system is built at a bit level consisting of only gated full adders. This approach allows recursive operation of IIR filter to be implemented with minimal delay time and chip area. The coefficients of the filter can be updated in real time for the time invariant and adaptive filtering. The fourth order filter is implemented on a

Proceedings ArticleDOI
27 Nov 1989
TL;DR: It is shown that, when sinusoidal interference is present at the carrier frequency, a system employing a linear prediction filter can significantly outperform a systems employing a rectangular notch filter.
Abstract: The authors analyze the performance of a direct sequence spread spectrum acquisition system employing a linear prediction filter to suppress a tone interferer located at the carrier frequency. They present a bound on the probability of error in the search mode and expressions for detection and false-alarm probabilities in the lock mode. They also give numerical evaluations of these expressions and compare them to analogous results on the the performance of an acquisition system employing a rectangular notch filter. It is shown that, when sinusoidal interference is present at the carrier frequency, a system employing a linear prediction filter can significantly outperform a system employing a rectangular notch filter. >

Patent
Arthur James Edwards1
03 Apr 1989
TL;DR: In this article, a switchable filter has characteristics (e.g., time constant) that are changed by a low offset, bipolar switch that is turned off and on to periodically connect/disconnect an additional circuit element to the filter.
Abstract: A switchable filter has characteristics (e.g., time constant) that are changed by a low offset, bipolar switch that is turned off and on to periodically connect/disconnect an additional circuit element to the filter. The preferred embodiment is specially designed for use in an automotive voltage regulator.