Showing papers on "Butterworth filter published in 1992"

Time-domain filter bank analysis: a new design theory

Abstract: The authors present a new time-domain approach for the analysis and design of a broad class of general analysis/synthesis systems based on M-band filter banks. They derive a set of time-domain conditions for reconstruction which can be used directly in a filter bank design procedure. The general and unrestricted nature of this framework allows for the design of many useful banks. In addition to the complete derivation of the time-domain conditions, they also describe the associated filter bank design procedure and a number of design examples are included. >

High-linearity continuous-time filter in 5-V VLSI CMOS

Abstract: High-linearity self-tuning continuous-time filters, fabricated in a standard 1.6- mu m 5-V CMOS process, are presented. Frequency control is achieved using switchable arrays of highly linear double-polysilicon capacitors in an active RC filter structure, resulting in tunable filters with very low signal distortion. One filter, a Tow-Thomas biquad, exhibits dynamic range and signal linearity of typically 91 dB. Another smaller implementation, a Sallen and Key filter, attains >or=76 dB. Cutoff frequency response is maintained to an accuracy of around +or-5%. >

Lagrange multiplier approaches to the design of two-channel perfect-reconstruction linear-phase FIR filter banks

Abstract: The authors present two approaches to the design of two-channel perfect-reconstruction linear-phase finite-impulse-response (FIR) filter banks. Both approaches analyze and design the impulse responses of the analysis filter bank directly. The synthesis filter bank is then obtained by simply changing the signs of odd-order coefficients in the analysis filter bank. The approach deals with unequal-length filter banks. By designing the lower length filters first, one can take advantage of the fact that the number of variables for designing the higher length filters is more than the number of perfect-reconstruction constraint equations. The second approach generalizes the first, and covers the design for all parts of linear phase perfect reconstruction constraint equations. >

A new variable fractional sample delay filter with nonlinear interpolation

Abstract: A finite impulse response (FIR) filter that can synthesize any fractional sample delay by a nonlinear interpolation technique is presented. Analytically closed-form solutions for the tap weights of such an FIR filter and their frequency responses are also presented. >

Least-mean squares adaptive digital filter havings variable size loop bandwidth

Abstract: An adaptive digital filter uses a weight adjustment unit for adjusting the weights of an adaptive digital filter according to one or more input signals to the digital filter and according to an error signal indicative of the difference between the actual and desired outputs of the digital filter. The weight adjustment unit has a first low-pass filter for low-pass filtering a signal indicative of the product of the error signal and the one or more input signals, a squarer for squaring the output of the first low-pass filter, a second low-pass filter for low-pass filtering the output of the squarer to extract the D.C. component thereof, a third low-pass filter for low-pass filtering a signal indicative of the output of the error signal squared to extract the D.C. component thereof, a dividing unit for dividing the output of the second low-pass filter by the output of the third low-pass filter to provide a loop bandwidth, and a weight calculation unit for providing values for one or more weights of the adaptive digital filter according to the previous values of the weights and the value of the loop bandwidth.

Fast filter bank (FFB)

Abstract: The sliding fast Fourier transform (FFT) filter bank has an exceedingly low complexity of one multiplication per channel per sample. However, its frequency selectivity and passband response are poor. It is shown that the sliding FFT filter bank is in fact a particular member of a new family of fast filter banks (FFBs). In the case of FFT, each cluster of butterflies can in fact be derived from a pair of complementary two-tap (i.e. first-order) prototype FIR filters. The poor selectivity and degraded passband response of the FFT filter bank is a direct consequence of the poor frequency response of the prototype first-order filter. It is shown that by increasing the order of the prototype filters, it is possible to implement a filter bank with arbitrarily good selectivity and flat passband response. The FFB retains the low-complexity feature of the FFT. Because of its very much improved frequency response characteristics, the FFB be suitable for use in many applications where the FFT filter bank is unsuitable. >

Active power line conditioner with fundamental negative sequence compensation

Abstract: An improved active power line conditioner is disclosed. A series inverter is controlled by a series filter controller which performs synchronous transformations on a load current to generate a series filter feedforward signal corresponding to the fundamental components of the load current. The series filter controller also generates a series filter reference signal corresponding to a negative sequence fundamental output voltage. The series filter feedforward signal and the series filter reference signal are combined to form a series filter compensation signal. The series filter compensation signal is applied to the series inverter to generate sinusoidal input currents, with negative sequence fundamental output voltage compensation, for a non-linear load. A parallel inverter is controlled by a parallel filter controller which performs synchronous transformations to generate a parallel filter feedforward signal corresponding to the harmonic ripple components of the load current. The parallel filter controller also generates a parallel filter reference signal corresponding to a negative sequence fundamental source current. The parallel filter feedforward signal and the parallel filter reference signal are combined to form a parallel filter compensation signal. The parallel filter compensation signal is applied to the parallel inverter to generate sinusoidal voltages, with source current negative sequence fundamental compensation, for the non-linear load.

Adjustable filter for differential microphones

Abstract: A method and apparatus for providing a differential microphone with a desired frequency response are disclosed. The desired frequency response is provided by operation of a filter, having an adjustable frequency response, coupled to the microphone. The frequency response of the filter is set by operation of a controller, also coupled to the microphone, based on signals received from the microphone. The desired frequency response may be determined based upon the distance between the microphone and a source of sound, and may comprise both a relative frequency response and absolute output level. The frequency response of the filter may comprise the substantial inverse of the frequency response of the microphone to provide a flat response. Furthermore, the filter may comprise a Butterworth filter.

Quadrature mirror filter banks and method

Abstract: M-channel pseudo-quadrature-mirror-filter (QMF) banks using a prototype filter having a linear-phase spectral-factor of a 2Mth band filter. The overall transfer function of the analysis filter/synthesis filter system is a delay, and the aliasing cancellation has all the significant aliasing terms canceled. Consequently, the aliasing level at the output of the pseudo-QMF banks is comparable to the stopband attenuation of the prototype filter, with the error at the output of the analysis filter/synthesis filter system approximately equal to the aliasing error at the level of the stopband attenuation. The pseudo-QMF banks have the stopband attenuation of the analysis filters and thus synthesis filters of -100 dB. The resulting reconstruction error is also on the order of -100 dB. Optimization of the pseudo-QMF banks by a quadratic-constrained least-squares formulation converges very fast as both a cost function and constraints are quadratic functions with respect to unknown parameters, providing a much higher stopband attenuation compared to previous filter banks.

Combination surge and diplex filter for CATV distribution systems

Abstract: A combination surge and diplex filter is provided for a CATV distribution amplifier. In a first preferred implementation, an integrated surge and high pass filter section is provided when the CATV amplifier is used only in the forward direction. Provision is made for a plug-in low pass filter section which is matched to the high pass filter section. The plug-in low pass section is used to form a diplex filter if the reverse direction capability of the CATV system is required from the distribution amplifier. In a second preferred implementation, the integrated surge and high pass filter section is segmented into a surge filter segment which is designed as a divisible part of the high pass filter section. A plug-in module is then provided with a high pass filter segment having those components of the high pass filter section which were not necessary for surge protection and a low pass filter section matched to the high pass filter section. The plug-in module is utilized to form a diplex filter, if the reverse direction capability of the CATV system is required from the distribution amplifier.

A quadratic-constrained least-squares approach to the design of digital filter banks

Abstract: The author formulates the filter bank design problem as a quadratic-constrained least-square minimization problem. The solution of the minimization problem converges very fast since the cost function and the constraints are quadratic functions with respect to the unknown parameters. The cosine-modulated and the two-channel linear-phase filter banks are designed using the proposed formulation. Compared to other design methods, the proposed technique yields PR (perfect reconstruction) filter banks with much higher stopband attenuation. The filter designed using the new approximation could be used as an initialization filter in a conventional PR filter bank design. >

System and method of producing adaptive FIR digital filter with non-linear frequency resolution

Abstract: A filter and method of adapting a digital filter provides a single fixed finite impulse response (FIR) filter adaptively from measured data, in a manner whereby the filter's frequency and time resolution can be controlled. The resulting filter exhibits properties which allow it to be efficiently implemented in various multi-rate configurations. Specifically, the system and method produce an FIR filter with high resolution at low frequencies by having a large number of coefficients, but reduces resolution at higher frequencies by allowing only a fraction of the coefficients to adapt to the high frequency part of the signal. This is accomplished by using a multi-rate, segmented adaptation procedure, such that resolution and bandwidth are controlled independently at each state. If desired, the resulting filter can be made to approximate constant Q resolution. In addition, by adapting only a short part of the filter at a time, misadjustment is minimized.

Multi-passband dielectric filter construction having filter portions with dissimilarly-sized resonators

Abstract: A filter duplexer, such as a filter duplexer for a radio transceiver, of minimum dimensions is disclosed. A first filter portion of the duplexer filter includes resonators of a first geometric configuration, and a second filter circuit portion of the duplexer filter comprises resonators of a second geometric configuration. The geometric configuration of the two filter circuit portions are dissimilar such that relative characteristic admittances of the resonators of the respective filter circuit portions are dissimilar. Because the resonators of the two filter circuit portions are of dissimilar electrical characteristics, a desired frequency response of the duplexer filter may be obtained with similar resonator loading capacitances.

Intermediate frequency filter having a ceramic filter

Abstract: An intermediate frequency filter for an intermediate frequency used in the receiver section of a digital radio communications apparatus is composed of an amplitude-flat ceramic filter incorporating two ceramic resonators, a buffer means and a single tuning circuit having the center frequency thereof within the bandwidth of the amplitude-flat ceramic filter, all being connected in series with each other. The center frequency of the single tuning circuit and Q are adjusted to approximate the amplitude characteristic of the intermediate frequency filter to that of TBT filter (0.4≦parameter m≦1.0).

Error probability of digital modulation in satellite mobile, land mobile, and Gaussian channels with narrow-band receiver filter

Abstract: It is shown how to derive formulas for the error probability for M-ary differential phase shift keying with differential phase detection (DPD) and M-ary frequency shift keying with DPD, limiter-discriminator detection and limiter-discriminator-integrator detection in the satellite mobile channel (SMC) with narrowband receiver filter if such formulas are available for the Gaussian channel. The modification of the formulas involves only a redefinition of the noise power and autocorrelation function. Since the SMC contains as special cases the land mobile (Rayleigh) channel and the Gaussian channel, the derived formulas are valid for these channels as well. In fact the formula for the land mobile channel is in many cases reduced to a closed form, which does not contain an integral. The author computes the error probability for the four systems, and compares their performance assuming a third-order butterworth filter and M=2,4,8 symbols. >

Digital filter for filtering and decimating delta sigma modulator output signals

Abstract: An oversampled interpolative delta sigma analog-to-digital converter including a delta sigma modulator is provided with a cascade of bit-slice elements at the output of the modulator to form a filter/decimator. Each bit-slice element includes a filter circuit that filters the bit-rate signal in accordance with an arbitrary filter impulse response input signal to provide the converter with a filter characteristic that can be controllably varied without modifying the filter hardware. In each bit-slice element, an adder circuit and a delay circuit decimate the bit-rate signal produced by the delta sigma modulator to provide a digital output signal at a clock cycle rate equal to the Nyquist rate. The filter/decimator also provides encoding of the delta sigma output in 2's complement format.

IIR filter design via optimal Hankel-norm approximation

Abstract: A new technique for digital filter design is presented. Based on the singular value decomposition of the Hankel matrix, balanced realisation and all-pass functions, an IIR filter is obtained via an optimal Hankel–norm approximation. The error between the optimal filter with order r and the desired filter is found to be equal to the (r + l)th singular value of the Hankel matrices. The designed low-pass filter and the differentiator are given to illustrate the proposed design algorithm.

A high-dynamic-range integrated continuous-time bandpass filter

Abstract: An eighth-order Butterworth bandpass filter, operating at 100 kHz with a quality factor of 14.3, is presented. The filter features an optimized dynamic range, a large tuning range, and a small occupied chip area of 0.25 mm/sup 2/ owing to very simple circuitry. Measurements show a very accurate realization of the desired transfer function, a high dynamic range of 62 dB, and a tuning range from 50 to 200 kHz. It is shown how the dynamic range can be improved to a theoretical maximum if circuit simplicity is sacrificed. >

A method for exceptionally low noise single channel recordings.

Abstract: We present a method whereby, with integrating electronics, quartz patch electrodes and a novel use of silicone oil, background noise levels as low as .083 pA RMS in a 5 kHz bandwidth (4-pole Butterworth filter) have been achieved in single channel patch clamp recordings. These approaches result in much higher signal to noise ratios for single channel recording than have previously been reported and should allow many investigators to significantly reduce noise at a constant bandwidth or to increase their recording bandwidths by several kHz.

Device and method for adjusting a bandpass filter, especially a combiner filter

Abstract: The invention relates to a frequency adjustment method for an RF bandpass filter, especially a combiner filter, wherein the center frequency of the passband of the RF bandpass filter (22) is adjusted in response to the RF power reflected from the input of the bandpass filter or passed through the bandpass filter. In the method a sample signal (Pr) proportional to the RF power reflected from the input of the bandpass filter or passed through the bandpass filter is mixed (29) with a signal of substantially the transmitting frequency, the mixing result is lowpass-filtered (24) and the bandpass filter (22) is adjusted in response to the lowpass-filtered mixing result (24a).

Data filter tuning for constant density recording applications

Abstract: An arrangement for tuning a data filter in a mass storage system. The filter cutoff frequency (low pass) may be set as a percentage above or below the incoming data rate to achieve the desired "eye opening" in read data. The frequency of a reference oscillator in a phase-locked-loop (PLL) and the cutoff frequency of the filter is controlled by a common control signal from the PLL. The PLL, locked to a scaled multiple of the data rate, determines the cutoff frequency of the filter.

The design of arbitrary-length cosine-modulated filter banks and wavelets, satisfying perfect reconstruction

Abstract: FIR filter banks which satisfy the perfect-reconstruction (PR) property can be obtained by cosine modulation of a linear-phase prototype filter of length N=2mM, where M is the number of channels. A PR cosine-modulated filter bank is presented for which the length of the prototype filter is arbitrary. Additional regularity conditions are imposed on the filter bank to obtain the cosine-modulated orthonormal bases of compactly supported wavelets. Design examples are given. >

Design of multiplierless FIR filters by multiple use of the same filter

Abstract: A novel approach to the design of multiplierless filters, based on the ACF (amplitude change function) [1], is discussed. The prototype filter chosen is a CCOS (the cascade of the cosine functions) which requires no multipliers and only some adders. The required filter specifications are met by multiple use of the same CCOS filter. Effects due to coefficient quantisation do not arise when using the new approach. No multipliers are required to implement this filter.

The effect of I,Q mismatch errors on adaptive cancellation

Abstract: The effects of in-phase (I) and quadrature-phase (Q) amplitude errors and low-pass-filter (LPF) errors on adaptive cancellers are investigated. I,Q errors occur because of errors in the synthesis process of the mixers and LPFs designed to be identical for each input channel. These I,Q errors among the channels result in cancellation degradation. Tapped delay line transversal filters have been proposed as a way to compensate for these errors and thus improve cancellation performance. However, it is shown that if there is any LPF mismatch, then transversal filtering has a small effect on improving canceler performance. The use of individual I,Q adaptive transversal filter weighting is suggested as a means of completely eliminating the phase amplitude errors, and making the canceler performance responsive to transversal filter compensation. >

Digital filter circuit for use in oversampling type analog/digital converter

Abstract: A digital filter circuit has two decimation filters, a first and a second, for performing two decimations with respect to a digital data of a predetermined sampling rate. The first decimation filter has a function of producing processing signals for a filter coefficient of the second decimation filter and providing such operational signal to the second decimation filter. Since the output of the first decimation filter is not a coded numeral value but is an output in the form of a processing signal for the filter coefficient of the second decimation filter, the first decimation filter can take the form of a decoder circuit and not an operational circuit and the second decimation filter can be simpler than that having a multiplier circuit of a conventional circuit. The scale of the overall circuit can be reduced and this contributes in the enhancement of high integration of large scale integrated circuits (LSIs), in the scaling down of chip areas and in reducing the manufacturing cost.

Acceleration effects in crystal filters-a tutorial

Abstract: The quasi-static method for the analysis of vibration-induced modulation in crystal filters is briefly reviewed, and a dynamic method, in which the filter is treated as a linear network with time-varying elements, is introduced. The dynamic method, which allows determination of both amplitude and phase modulation due to vibration, is illustrated by examples. It is then applied to the analysis of the spectrum clean-up case, consisting of a frequency source with an output filter, both of which are undergoing the same acceleration. >

Tunable wideband active filter

Abstract: A wideband tunable (programmable) filter includes first-order and second-order filter circuits employing bipolar transistors. The bandwidth of the filter circuits are set by the product of the dynamic emitter resistance of the transistors and directly coupled on-chip capacitances. The filter bandwidth is inversely proportional to the time constant. To adjust the resonant frequency of the filter networks, emitter currents are varied, which in turn controls the emitter resistances. As the resonant frequency of the filter network is varied, the gain and normalized frequency response of the various output nodes remain constant. In the second-order configuration, the damping ratio of the filter is set using "positive" feedback, for a Q of greater than one-half, with a loop gain of less than one. Temperature compensation is accomplished by use of a temperature "dependent" voltage network. The output of this network provides the sum of the continuous temperature compensating voltage component and the externally controlled tuning voltage for the resonant frequency. This voltage is converted to currents that control the bandwidth of the filter so as to minimize dependence of the resonant frequency upon temperature.

Highpass filters for detecting late potentials

Abstract: A nonrecursive monotonous binomial highpass filter is suggested for the analysis of the signal averaged electrocardiograms (ECGs). Theoretical background and details of the appropriate design of these filters are given. The application of these digital filters to signal averaged ECGs provides an undistorted detection of late potentials. Time and phase behavior is improved with respect to the recursive Butterworth filter of fourth order, which has been used frequently and successfully to identify late potentials. >

Generalized feedforward filters with complex poles

Abstract: The authors propose an extension to an existing structure, the gamma filter, replacing the real pole on the tap-to-tap transfer function with a pair of complex conjugate poles and a zero. The new structure is, like the gamma filter, an IIR filter with restricted feedback whose stability is trivial to check. While the gamma filter decouples the memory depth from the filter order for low-pass signals, the proposed structure decouples the memory depth and the central frequency from the filter order for band-pass signals. The learning equations of the model parameters are presented and shown to introduce an additive O(p) complexity to the backpropagation algorithm, where p is the filter order. The error surface for a linear filter is investigated in a system identification context, and the presence of local minima is confirmed. The performance of the proposed model was found to be better than that of the time-delay neural net in a nonlinear system identification context. >

RC filter for low and very low frequency applications

Abstract: An RC filter for low or very low frequency applications, comprising a resistor between the filter input and output, and an amplifier connected after the resistor and having an output fed back to the amplifier input through a capacitor. This simple design allows the known Miller Effect to be utilized to produce a filter having a high time constant while employing small-size components which occupy little space in integrated circuits.