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Showing papers on "Voltage-controlled filter published in 1994"


Journal ArticleDOI
02 Oct 1994
TL;DR: In this article, the authors present an active power filter for three-phase power systems, which is composed of a six-switch threephase inverter, a DC bus capacitor, and an isolation transformer.
Abstract: This paper presents an active power filter for three-phase power systems. The work is motivated by the need for active filtration in a current-source excitation system for a variable-reluctance generator. The active filter is comprised of a six-switch three-phase inverter, a DC bus capacitor, and an isolation transformer. The isolation transformer is required by the application. The leakage inductance associated with each phase of the isolation transformer is used as the series impedance with each phase, by which the inverter is able to actively shape the phase currents in order to compensate for the nonlinearities of all loads within the point of common coupling. The active filter is controlled through two control loops. The inner current regulation loop uses sliding-mode control by virtue of its ease of implementation. The outer voltage loop regulates the average voltage on the DC bus capacitor. The outer voltage loop is responsible for correctly setting the commanded magnitude of the phase currents. This paper presents the analysis, design, and operation of the active filter. Experimental results are provided for the active filter compensating a phase-controlled rectifier which is drawing 10.4 kW. >

161 citations


Journal ArticleDOI
TL;DR: In this paper, a low-power cascadable analogue filter for silicon cochlea implants is described. But it is based on the log-domain filter approach used previously in bipolar technology.
Abstract: A new approach to the design of a low-power cascadable analogue filter for applications such as silicon cochlea implants is described. One section of a possible silicon cochlea based on a second-order continuous-time filter employing MOS technology operating in weak inversion is presented. The idea is based on the recently proposed log-domain filter approach used previously in bipolar technology.

130 citations


Patent
02 Nov 1994
TL;DR: In this article, an analog-to-digital conversion circuit is described which includes a front end sigma-delta modulator circuit, a multi-stage digital decimation filter circuit, and a digital compensation filter circuit.
Abstract: An analog-to-digital conversion circuit is described which includes a front end sigma-delta modulator circuit, a multi-stage digital decimation filter circuit, and a digital compensation filter circuit. An overrange detect circuit is also provided.

110 citations


Patent
07 Mar 1994
TL;DR: In this paper, a 16-bit second-order Sigma-Delta-based converter with a programmable comb filter is presented. But the comb filter uses a 20-bit data path, in order to enable the decimator to provide 16 bits of resolution to the DSP.
Abstract: A "true" 16-bit second order Sigma-Delta based converter that has superior analog components and has a programmable comb filter which is coupled to the digital signal processor. This converter comprises a second order Sigma-Delta modulator and a programmable comb filter. The second order Sigma-Delta modulator dramatically attenuates the baseband quantization noise energy (which in turn increases the resolution of the converter), since its superior amplifiers and comparators enable it to oversample and coarsely quantize the analog input signal at a very high sampling frequency of 12 MHz. The amplifiers are class AB OTAs, which have cross coupled NMOS driven input stages, and cascoded output stages. Also, the common mode voltages are the optimal biasing points, and these voltages are kept constant by a differential input stage, by a PV independent temperature dependent current generator, by optimal device size, and by a common mode feedback circuitry. The programmable comb filter receives the coarsely digitized 1-bit output of the modulator at oversampling frequency F S , and provides a more accurate representation of the input signal to the DSP at slower sampling rate of F S /N. In addition, the comb filter uses a 20-bit data path, in order to enable the decimator (which is formed by the comb filter and by the FIR filter) to provide 16 bits of resolution to the DSP. The output of the programmable comb filter is then supplied to an FIR filter which is realized in the DSP, and this filter removes the remaining out-of-baseband noise.

90 citations


Patent
06 Apr 1994
TL;DR: In this paper, a cross-coupled switched capacitor circuit was used to reduce the size of the capacitors in a biquad switched capacitor filter, which reduced the capacitance by a factor of two.
Abstract: A biquad switched capacitor filter is preferably utilized as the output filter in a sigma delta digital-to-analog converter. The switched capacitor filter uses a cross-coupled switched capacitor circuit which delivers charge to the capacitors on both phases of the clock. As a result, the sizes of the capacitors can be reduced by a factor of two, while delivering the same charge as a single sampling circuit. By using the cross-coupled switching circuit everywhere in the filter, the sensitivity to capacitor mismatches is substantially reduced. The clock phases applied to the stages of the filter are alternated so that there is a one clock cycle delay around each loop containing two filter stages, thereby insuring the stability of the filter.

88 citations


Patent
14 Jun 1994
TL;DR: In this article, an active filter controller uses synchronous transformations to identify selected harmonic reference components corresponding to individual harmonics of the three-phase load currents and then injects a permissible percentage of harmonics into the supply voltage to reduce the voltage distortion at the passive filter.
Abstract: A power line conditioner includes an active filter coupled, in series, and a passive filter coupled, in parallel, to a three-phase power distribution network The three-phase power distribution network includes a voltage source that induces three-phase input currents at a first end of the three-phase power distribution network A load, circulating three-phase load currents, is positioned at a second end of the three-phase power distribution network The active filter controller of the invention uses synchronous transformations to identify selected harmonic reference components corresponding to individual harmonics of the three-phase load currents The selected harmonic reference components are multiplied by a predetermined factor corresponding to a permissible percentage of the individual harmonics that may be injected into the supply voltage This results in active filter reference signal components that are applied to the active filter In response to the active filter reference signal components, the active filter injects a permissible percentage of harmonics into the supply voltage so as to reduce the voltage distortion at the passive filter By reducing voltage distortion, the controlled injection of harmonics into the supply allows for a simplified passive filter design Consequently, the passive filter may be implemented as a power factor correction capacitor

78 citations


Journal ArticleDOI
TL;DR: In this paper, a universal voltage-mode second-order filter with three inputs and one output employing three current conveyors, one voltage-follower, two capacitors and three resistors is presented.
Abstract: A universal voltage-mode second-order filter with three inputs and one output employing three current conveyors, one voltage-follower, two capacitors and three resistors is presented. The proposed circuit offers the following advantageous features: realisation of allpass, notch, highpass, bandpass and lowpass signals from the same configuration, no requirements for component-matching conditions, orthogonal control of omega /sub o/ and Q, low active and passive sensitivities and cascadability. >

77 citations


Patent
15 Nov 1994
TL;DR: In this paper, an active filter controller uses synchronous transformations on the input currents to identify a negative sequence fundamental signal and a positive-sequence fundamental signal, while filtering all harmonic components within the input current.
Abstract: The apparatus includes an active filter, coupled in series, and a passive filter, coupled in parallel, to a power distribution network. The power distribution network includes a voltage source that induces input currents at a first end of the power distribution network. Nonlinear loads and other conditions on the power distribution network cause unbalanced power signals. The active filter controller of the invention uses synchronous transformations on the input currents to identify a negative sequence fundamental signal and a positive sequence fundamental signal, while filtering all harmonic components within the input currents. The negative sequence fundamental signal and the positive sequence fundamental signal are combined to form an active filter reference signal which is applied to the active filter. In response to the active filter reference signal, the active filter operates as a current controlled harmonic voltage source, carrying only the fundamental current, while only injecting harmonic voltages. Consequently, the active filter is operated as a harmonic isolator between the supply and load.

60 citations


Journal ArticleDOI
TL;DR: An active power filter for compensating voltage sags that occur on a weak AC power system is described, and a design procedure based on IEEE/ANSI voltage withstand tolerance (IEEE standard 446-1987) is proposed.
Abstract: An active power filter for compensating voltage sags that occur on a weak AC power system is described. The proposed active power filter is especially suitable in situations where sensitive data processing and other critical loads are to be operated on a weak AC system. The proposed filter is fast acting and simple in design. A design procedure based on IEEE/ANSI voltage withstand tolerance (IEEE standard 446-1987) is proposed. Laboratory tests on a prototype filter show fast response and linear correction characteristics. >

52 citations


Patent
21 Oct 1994
TL;DR: In this paper, a narrow band gain enhancing filter for direct access storage devices is presented, where the initial states of the filter are stored as phase, frequency and amplitude of a sine function.
Abstract: A method and apparatus, in a direct access storage device including a head positioned for interaction with a data storage medium mounted on a rotating spindle, an actuator for positioning the head, and a servo control loop for positioning the actuator. The improvement comprises: a narrow band gain enhancing filter for connection in the servo control loop. The filter has a response frequency related to the rotational frequency of the spindle. The filter has programmable initial states. Also provided is a switch for switching in the narrow band filter as the head approaches a target position on the data storage medium. Initial states of the filter are determined and supplied to program the filter. At least one additional filter having a peak response frequency at a harmonic of the frequency of rotation of the spindle, may be provided. Initial states of the additional filters are determined and supplied to program the additional filter. The filter may include successive delay circuits for inputs to the filter, which circuits provide delayed outputs; multipliers for multiplying selected delayed outputs by a constant; and a summer for summing the delay outputs after the delayed outputs are multiplied by the multipliers. The initial states may be stored as phase, frequency and amplitude of a sine function.

51 citations


PatentDOI
TL;DR: In this paper, an arrangement for converting an electric signal into an acoustic or a mechanic signal comprising a transducer (11), a linear or nonlinear filter (1) with controllable parameters, a sensor (12), a controller (24), a reference filter (20), and a summer (17).
Abstract: An arrangement is provided for converting an electric signal into an acoustic or a mechanic signal comprising a transducer (11), a linear or nonlinear filter (1) with controllable parameters, a sensor (12), a controller (24), a reference filter (20) and a summer (17). The filter (1) is connected to the electric input of the transducer and is adaptively adjusted to compensate for the linear and/or nonlinear distortions of the transducer and to realize a desired overall transfer characteristic. The filter has for every controllable filter parameter an additional output (7) supplying a gradient signal to the controller and a control input (10). The summer (17) provides an error signal derived from the sensor output and reference filter output. The controller contains a circuit (53) for filtering the gradient signal and/or a circuit (25) for filtering the error signal, a multiplier (51) and an integrator (57) for producing a control signal to update every filter parameter. This arrangement omits off-line pre-training and adapts on-line for changing transducer characteristics caused by temperature, ageing and so on.

Patent
08 Aug 1994
TL;DR: In this paper, a high-frequency integrated continuous-time filter with built-in test mode is presented, which provides the ability to easily track the cutoff frequency of the filter without the additional power and area requirements and noise sources present in prior art master/slave tuning schemes.
Abstract: A high-frequency integrated continuous-time filter with built-in test mode. The present invention provides the ability to easily track the cutoff frequency of the filter without the additional power and area requirements and noise sources present in prior art master/slave tuning schemes. Furthermore, the filter being tested is the actual filter that is used to process signals, unlike the prior art where a similar but separate filter or oscillator is used to tune the bias values for both circuits. Better tuning accuracy is thus obtained in the present invention. The circuit is designed to oscillate in test mode at the cutoff frequency of the filter. Oscillation is achieved by moving the poles of the filter from the left half-plane either onto the imaginary axis or into the right half-plane. The filter frequency accuracy is established by trimming the frequency of the oscillation in test mode during wafer probe or by adjusting the circuit biasing to tune the cutoff frequency in test mode during power-up or between reads in a memory system. The oscillation is disabled during normal operation of the filter.

Patent
01 Feb 1994
TL;DR: In this article, a speaker amplification system incorporates an adaptable notch filter that can dynamically adapt to the frequency of feedback oscillations caused by acoustic feedback frequency in order to remove the feedback oscillation before they are amplified above an audible level.
Abstract: A speaker amplification system incorporates an adaptable notch filter that can dynamically adapt to the frequency of feedback oscillations caused by acoustic feedback frequency in order to remove the feedback oscillation before they are amplified above an audible level. The adaptable notch filter is implemented as an adaptable finite impulse response filter by means of a digital signal processor. The finite impulse response filter also has an adaptable convergence factor which minimizes noise introduced by the filter. A special limiting routine is used to limit the values of the finite impulse response filter coefficients to prevent the filter from "blowing up" due to large input signal disturbances. The speaker amplification system also includes circuitry which provides multiple bands of equalization, with each band being capable of amplification. The mullet-band equalizer function allows hearing impaired people to adjust the frequency response of the telephones handset speaker, to provide better intelligibility, due to a large increase in volume over normal telephone handset receivers.

Patent
28 Feb 1994
TL;DR: In this article, the SAW filter and the amplifier are inductively or capacitively coupled to the resonators depending upon the type of filter that is required, and if three or more resonators are present in the first filter then an amplifier can also be integrated in the filter.
Abstract: A filter, particularly suitable for radio frequency applications comprises a first filter which, in turn, comprises at least a pair of intercoupled resonators (RES1, RES2), and a SAW filter (SAW) which is coupled between the two intercoupled resonators so that a signal input to the filter at its input (IN) is coupled through the SAW filter to provide an output signal at its output (OUT). This provides a filter with all the advantages of a SAW filter, but with the ability to withstand the high power requirements of radio frequency applications. If three or more resonators are present in the first filter, then an amplifier can also be integrated in the filter. The SAW filter (and the amplifier if present) are inductively or capacitively coupled to the resonators depending upon the type of filter that is required.

Patent
09 Jun 1994
TL;DR: In this paper, the sampling frequency of the plurality of digital filter functions is lower than the master clock of the signal processing circuit and operating a plurality of filters on a time-sharing basis.
Abstract: An optical disk player provided with a digital servo circuit uses a time-sharing multifunctional digital filter, in which a plurality of filter functions are realized with a single filter. By lowering the sampling frequency of the plurality of digital filter functions than the master clock of the signal processing circuit and operating the plurality of digital filters on a time-sharing basis, the number of steps of the signal processing command can be decreased and the circuit scale can be reduced. By constructing a High Pass Tracking Zero Cross HPF and an anti-shock BPF of a digital filter using the same step, reduction in the circuit scale and increase in the sampling rate can be attained.

Patent
14 Oct 1994
TL;DR: Amplitude Shift Keying (ASK) modulation system and method with an ASK demodulator that is implemented with an analog-emulating digital bandpass filter is described in this paper.
Abstract: Amplitude Shift Keying (ASK) modulation system and method with an ASK demodulator that is implemented with an analog-emulating digital bandpass filter. The bandpass filter also generates a carrier detect signal when it detects a carrier frequency that passes the filter pass band.

Patent
15 Dec 1994
TL;DR: A variable Q reflection mode filter has a three-port circulator, one port of which is terminated by a one-port filter as mentioned in this paper, which includes a ladder network of resonators, successive of the resonators having progressively reducing Q values.
Abstract: A variable Q reflection mode filter has a three-port circulator, one port of which is terminated by a one-port filter. The filter includes a ladder network of resonators, successive of the resonators having progressively reducing Q values. The filter increases the overall 3 dB resonance point of the filter to reduce the unloaded Q factors required for particular applications. The filter may be arranged to provide a maximally flat response, an equiripple response, or a quasi-ripple response. Transmission zeros may also be incorporated to further enhance the effective unloaded Q.

Patent
02 Nov 1994
TL;DR: In this paper, a multi-stage digital decimation filter was proposed for a sigma-delta modulator, which includes a sinc6 filter, a half-band linear phase FIR filter and a symmetric, linear phase, FIR filter.
Abstract: This invention is for a multi-stage digital decimation filter, which includes a sinc6 filter, a half-band linear phase FIR filter and a symmetric, linear phase, FIR filter and a digital compensation filter. The multi-stage decimation filter down-samples the output of a sigma-delta modulator by a factor of about 64. The compensation filter provides compensation for rolloff introduced by the multi-stage decimation filter.

Journal ArticleDOI
TL;DR: In this paper, a new multifunction filter using minimum components is presented, which possesses single input and four outputs can simultaneously generate second-order high-pass, bandpass, and low-pass filtering functions at individual output terminals.
Abstract: A new multifunction filter using minimum components is presented. This filter which possesses single input and four outputs can simultaneously generate second-order highpass, bandpass, and lowpass filtering functions at individual output terminals. The filter has the following merits; it uses only two grounded capacitors which makes it suitable for integrated circuit implementation, no matching condition is required, offering multifunction outputs and has low passive and active sensitivities. Experimental results agree very well with the theoretical result.

Proceedings ArticleDOI
13 Feb 1994
TL;DR: In this article, an active power filter for single-phase systems which are comprised of multiple nonlinear loads is presented, where the inverter switches are controlled to shape the current through the filter inductor such that the line current is in phase with and of the same shape as, the input voltage.
Abstract: This paper presents an active power filter for single-phase systems which are comprised of multiple nonlinear loads. The active filter is based on a standard H-bridge inverter. The AC side of the inverter is connected in parallel with the other nonlinear loads through a filter inductance; the DC side of the inverter is connected to a filter capacitor. The inverter switches are controlled to shape the current through the filter inductor such that the line current is in phase with, and of the same shape as, the input voltage. Sliding-mode control is used to perform the active wave-shaping of the current. The paper provides the details of the power circuit design, the details of the control design, representative waveforms, and spectral performance for a filter which supports a 364 W AC controller and a 884 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 of the active filter easily meets the constraints of IEC555. >

Journal ArticleDOI
TL;DR: In this paper, a simple algebraic method for 2D IIR digital notch filter design was proposed, which not only has closed-form transfer function but also satisfies the bounded-input/bounded-output (BIBO) stability condition.
Abstract: In this paper, the two dimensional (2D) IIR digital notch filter design problem is investigated. First, we develop a simple algebraic method for 2D IIR notch filter design. This approach not only has closed-form transfer function but also satisfies the bounded-input/bounded-output (BIBO) stability condition. Next, we apply the 2D notch filter to eliminate sinusoidal/narrowband interferences superimposed on an image. Simulation results are presented to demonstrate the effectiveness of this approach. >

Patent
Hiroshi Kimura1, Ryutaro Horita1, Hase Kenichi1, Kunio Watanabe1, Takashi Nara1 
23 Dec 1994
TL;DR: An active filter control apparatus for controlling or tuning an active filter having a variable cut-off frequency is described in this article, where a control circuit is used to control or tune the cutoff frequency of the active filter and a characteristic correction generator for generating a correction signal to correct a group delay.
Abstract: An active filter control apparatus for controlling or tuning an active filter having a variable cut-off frequency. The active filter control apparatus includes a control circuit for controlling or tuning the cut-off frequency of the active filter and a characteristic correction generator for generating a correction signal to correct a group delay characteristic of the active filter in accordance with a set cut-off frequency. The characteristic correction includes a correction signal generator for generating the correction signal in accordance with a set correction amount. The cut-off frequency controller controls tunes the characteristic of the active filter in accordance with the correction signal. Preferably, the apparatus is formed of a one-chip LSI integrated on one chip. The control apparatus can be utilized to control the speed in a recording/reproducing apparatus such as a optical disk drive or a magnetic tape drive apparatus.

Patent
Paul Anthony Moore1
13 Dec 1994
TL;DR: In this article, a radio receiver has a frequency down conversion stage (28) including a first passive filter (50, 51, 52) tuned to a lower frequency, and an amplifier (56, 58) functioning as a voltage to current converter (54), which can be shunted by a second passive filter in response to actuation of symmetrically arranged switching devices.
Abstract: A radio receiver has a frequency down conversion stage (28) including a first passive filter (50, 51, 52) tuned to a lower frequency. An amplifier (56, 58) functioning as a voltage to current converter (54) comprises a resistor (60) which can be shunted by a second passive filter (70, 71, 74) in response to actuation of symmetrically arranged switching devices (72, 73). When the second passive filter shunts the resistor the overall pass band is raised to a higher frequency. Additionally the amplifier has a lower noise figure when the second passive filter shunts the transconductor.

Journal ArticleDOI
TL;DR: Approximate filters based on a phase-only filter for reliable recognition of objects and good light efficiency and discrimination capability close to that of the optimal filter can be obtained.
Abstract: Approximate filters based on a phase-only filter for reliable recognition of objects are proposed. Good light efficiency and discrimination capability close to that of the optimal filter can be obtained. Computer simulation results are presented and discussed.

Proceedings ArticleDOI
05 Dec 1994
TL;DR: In this article, the authors introduce a new under-decimated system, where the number of channels is less than the decimation ratio in the subbands, and the overall response is equivalent to a tunable multilevel filter.
Abstract: In this paper, we introduce a new under-decimated system. A filter bank is said to be under-decimated if the number of channels is more than the decimation ratio in the subbands. Two types of low-complexity filter banks can be used for the new system, the DFT filter bank and cosine modulated filter bank. The setup of the under-decimated system has 2M channels but is decimated only by M. In both the DFT filter bank case and cosine modulated filter bank case, the system is approximately alias free and the overall response is equivalent to a tunable multilevel filter. The properties of DFT filter banks and cosine modulated filter banks can be exploited to simultaneously achieve parallelism, computational saving and lower working rate. Furthermore, in both filter banks the implementation cost of the analysis bank is comparable to that of one prototype filter plus some low complexity matrices. The individual analysis and synthesis filters have complex coefficients in the DFT filter bank but have real coefficients in the cosine modulated filter bank.

Book
01 Dec 1994
TL;DR: Electronic Filter Analysis and Synthesis helps you save time and effort in writing CAD and analysis programs for electronic filters, and provides explicit details on how to synthesize lowpass, bandpass,Bandstop, and highpass realizations for passive, active, digital and switched capacitors.
Abstract: Electronic Filter Analysis and Synthesis helps you save time and effort in writing CAD and analysis programs for electronic filters, and provides explicit details on how to synthesize lowpass, bandpass, bandstop, and highpass realizations for passive, active, digital and switched capacitors.

Patent
19 Jul 1994
TL;DR: In this article, a tunable filter (25) is described for use in loop control circuits, which has a time constant which is determined by a resistor (40) and a capacitor.
Abstract: A tunable filter (25) is described for use in loop control circuits. The filter (25) has a time constant which is determined by a resistor (40) and a capacitor. The capacitor is simulated by the series combination of an impedance converter (41) and a variable resistor, such as a field effect transistor (42). The resistance of the transistor (42) is determined by a control signal present on a control line (32) and the impedance converter (41) converts the resistance into an equivalent capacitive reactance. Therefore, the control signal on the control line (32) effectively controls the capacitance present at a node (43) and, in conjunction with the resistor (40), determines the time constant, and therefore the response time and bandwidth, of the filter (25). Multiple pole, lowpass, bandpass, highpass, and bandstop filters can be constructed. The impedance converter (41) uses a very small capacitor to simulate a large capacitance value at the node (43). This small capacitance value also allows the filter (25) to be fabricated as part of an integrated circuit.

PatentDOI
TL;DR: In this article, a weighting filter is applied to the difference signal and the weighted difference is used to calculate an energy measure, which is then used to control the codebook search.
Abstract: Analysis by synthesis calculates a difference by subtracting (130) synthesized speech from input speech. The synthesized speech is formed by exciting long and short term filters (124,126) with excitation vectors from a codebook store (114) which is searched by codebook generation (120). A weighting filter (132) is applied to the difference signal and the weighted difference is used to calculate an energy measure (134) which is used to control the codebook search (140). The weighting filter is an Rth-order filter controlled with calculated coefficients. The method for calculating coefficients models the frequency response of L Pth-order filters by a single Rth-order filter, where the order R ~ LxP. This method increases the control of a speech coder filter without an increase in hardware complexity.

Journal ArticleDOI
01 Jun 1994
TL;DR: In this article, a novel filter design approach to digital I/Q demodulation is proposed, and two possible realisations are presented using this approach: the first one is based on the high-pass filter method which is suitable for B≤f0 and while the other realisation is a low-pass filtering method suitable for b≤ f0, where B and f0 are the IF signal bandwidth and the IF frequency, respectively.
Abstract: A novel filter design approach to digital I/Q demodulation is proposed. Two possible realisations are presented using this approach. The first one is based on the highpass filter method which is suitable for B≤f0 and while the other realisation is based on the lowpass filter method suitable for B≤f0, where B and f0 are the IF signal bandwidth and the IF frequency, respectively. Both new realisations maintain the advantages of an earlier lowpass approach such as zero DC offset, matched channel frequency responses, and good performance over a wide bandwidth. At the same time, the new highpass filter realisation method possesses higher computational efficiency than other wideband approaches reported in the literature.

Patent
07 Oct 1994
TL;DR: In this article, the authors proposed a dual-band filter with a bandpass characteristic over a first predetermined range of frequencies and a notch, or band-reject characteristic over the second predetermined range.
Abstract: A dual-band filter providing a bandpass characteristic over a first predetermined range of frequencies, and a notch, or band-reject characteristic over a second predetermined range of frequencies. In one embodiment, the notch characteristic occurs within the bandpass characteristic, yielding a filter frequency response with two asymmetric passbands separated by a reject band. This is particularly suited to cellular Band A receive filter applications, but the design is easily adapted to other uses. Appropriate tuning can restore symmetry to the passbands.