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Showing papers on "Linear phase published in 1990"


Journal ArticleDOI
TL;DR: The dynamic range of phase‐ reconstructed magnetic resonance images is compared to that of magnitude‐reconstructed images, verifying that the noise in the two receiver channels is uncorrelated and an artifact‐free technique is presented for correcting phase spillover, which further extends the phase angle dynamic range.
Abstract: The dynamic range of phase-reconstructed magnetic resonance images is compared to that of magnitude-reconstructed images. From analysis of propagation of errors, the phase angle noise is phase-independent and given in radians by σ (|I|)/|I|, the noise-to-signal ratio of the corresponding magnitude-reconstructed image. As the phase can range from minus π to π the phase angle dynamic range is 2π times that of the signal magnitude. These results agree with experiment, verifying that the noise in the two receiver channels is uncorrelated. An artifact-free technique is presented for correcting phase spillover, which further extends the phase angle dynamic range. The reconstruction-based reference phase is adjusted on a local basis so that the boundary of phase wraparound is reconstructed near the center of the [ − π, π] interval. For a particular flow study, the phase signal-to-noise was extended over twofold by spillover correction, to a value 15 times that of the magnitude signal-to-noise. © 1990 Academic Press, Inc.

202 citations


Journal ArticleDOI
TL;DR: Four methods are presented for optimizing filters in the normalized peak ripple magnitude (NPRM) sense, two of the methods being to the passband gain sectioning technique and several heuristic methods for determining alpha.
Abstract: Four methods are presented for optimizing filters in the normalized peak ripple magnitude (NPRM) sense. Two of the methods being to the passband gain sectioning technique. The other two methods make use of the objective function f= delta - alpha b. Several heuristic methods for determining alpha are also presented. The NPRM is an important performance measure; the absolute peak ripple magnitude and passband gain are less important. In these applications, the passband gain need not be fixed at unity but should be a continuous variable to the optimized. Nevertheless, an upper and a lower bound on the passband gain should be imposed to satisfy overflow and roundoff noise performance requirements. >

166 citations


Journal ArticleDOI
TL;DR: In this article, an autoregressive model is fitted to the signal, and low-pass filtering is performed in the frequency domain by a linear phase FIR filter and differentiation is performed on the high-frequency noise magnification.
Abstract: Smoothing and differentiation of noisy signals are common problems whenever it is difficult or impossible to obtain derivatives by direct measurement. In biomechanics body displacements are frequently assessed and these measurements are affected by noise. To avoid high-frequency noise magnification, data filtering before differentiation is needed. In the approach reported here an autoregressive model is fitted to the signal. This allows the evaluation of the filter bandwidth and the extrapolation of the data. The extrapolation also reduces edge effects. Low-pass filtering is performed in the frequency domain by a linear phase FIR filter and differentiation is performed in the frequency domain. The reported results illustrate the accuracy of the algorithm and its speed (mainly due to the use of the FFT algorithm). Automatic bandwidth selection also guarantees the homogeneity of the results.

166 citations


Journal ArticleDOI
TL;DR: Results are presented on 2-D FIR (two-dimensional finite-impulse-response) filter banks for multirate applications, where conditions for alias-free and perfect signal reconstruction are derived and synthesis structures for paraunitary and nonparaunitary polynomial matrices yield perfect reconstruction filter banks.
Abstract: Results are presented on 2-D FIR (two-dimensional finite-impulse-response) filter banks for multirate applications. The theory is valid for all sampling lattices; conditions for alias-free and perfect signal reconstruction are derived. Synthesis structures for paraunitary and nonparaunitary polynomial matrices are derived, which yield perfect reconstruction filter banks. The degrees of freedom are given for these systems. Linear phase conditions are posed on the polyphase form of filter banks. which is used to derive a design structure for the restricted, but important, case of linear phase filter banks. >

148 citations


Journal ArticleDOI
TL;DR: The authors develop structures for FIR (finite impulse response) perfect-reconstruction QMF (quadrature mirror filter) banks, which cover a subclass of systems that yield linear-phase analysis filters for arbitrary M.
Abstract: The authors develop structures for FIR (finite impulse response) perfect-reconstruction QMF (quadrature mirror filter) banks, which cover a subclass of systems that yield linear-phase analysis filters for arbitrary M. The parameters of these structures can be optimized to design analysis filters with minimum stopband energy which at the same time have linear phase and satisfy the perfect-reconstruction property. If there are M subbands, then depending on whether the coefficients h/sub k/(n) of each analysis filter are symmetric or antisymmetric, several combinations of filter banks are possible. Some of these permit perfect reconstruction and some do not. For a given M, the authors develop a formula for the number of combinations for a subclass of linear-phase perfect-reconstruction structures. As an example, they elaborate on a perfect-reconstruction linear-phase lattice structure for three channels. The lattice structure is such that, regardless of the parameter values, the QMF bank enjoys the perfect-reconstruction property while at the same time the analysis filters have linear phase. A design example is presented, along with tables of impulse response coefficients. >

98 citations



Patent
29 May 1990
TL;DR: In this paper, a second phase locking loop is added to a conventional phase lock loop to form a phase tracking system which provides a precisely constant phase shift over a full range of frequencies of the signal captured by the conventional phase locked loop.
Abstract: The output signal of a phase tracking system maintains phase coherence with the dominant input signal while tracking the frequency of that same dominant input signal without additional phase shifting circuitry Circuitry having a second phase locked loop is added to a conventional phase locked loop to form a phase tracking system which provides a precisely constant phase shift over a full range of frequencies of the signal captured by the conventional phase locked loop

84 citations


Patent
05 Feb 1990
TL;DR: In this paper, the phase angle of the magnetic flux is detected and the phase current is computed on the basis of phase current, and a q-axis-current control phase and a reference potential phase are determined based on the result of pulse-width-modulation computation and phase angle.
Abstract: A current component, in which a phase angle is different by a predetermined reference angle from a magnetic flux occurring in an inductive load, is brought to a q-axis current. The q-axis current is computed on the basis of phase current. A phase angle of the magnetic flux is detected. A q-axis-current control phase and a reference potential phase are determined on the basis of the result of pulse-width-modulation computation and the phase angle. The q-axis-current control phase is controlled in a pulse-width-modulation manner. A potential of the remaining reference potential phase if fixed to a reference potential. Thus, interference among the phases is prevented to reduce current ripple.

71 citations


Patent
30 Jul 1990
TL;DR: In this paper, a method and apparatus for determining the phase and amplitude accuracy of continuous-phase-modulated signals is described, where a modulated RF signal generated by a transmitter is down converted to a relatively low intermediate frequency which is filtered and sampled by a high sampling rate analog-to-digital converter.
Abstract: A method and apparatus for determining the phase and amplitude accuracy of continuous-phase-modulated signals is described. A modulated RF signal generated by a transmitter is down converted to a relatively low intermediate frequency which is filtered and sampled by a high sampling rate analog-to-digital converter. A digital signal processor processes the digital signals to produce a measured phase function corresponding to the modulated RF signal. From the measured amplitude and phase functions, an estimate of the ideal reference phase function corresponding to the modulated RF signal is calculated and synthesized. The reference phase function is compared to the measured phase function to determine the phase function from which the modulated RF signal phase error and frequency error are computed.

55 citations


Patent
09 Aug 1990
TL;DR: In this article, a linear phase progression in the correlation process is used to measure the frequency offset of a receiver employing complex correlation, which is indicative of carrier frequency offset equalization.
Abstract: In a receiver employing complex correlation, there is provided a mechanism for carrier frequency offset equalization. There exists a linear phase progression in the correlation process that is indicative of frequency offset. One method involves curve fitting the phase progression in order to warp the sync word against which subsequent correlation is performed rather than controlling the reference frequency of the receiver. The other successively approximates the phase progression by pre- and re-warping the synchronization reference word within the correlation process itself rather than controlling the reference frequency of the receiver.

51 citations


Journal ArticleDOI
TL;DR: In this article, a new class of composite filters called the linear phase coefficient composite filters (LPCCFs) is introduced, which considers the training set selection and the filter design simultaneously.
Abstract: A new class of composite filters called the linear phase coefficient composite filters (LPCCFs) is introduced. Unlike previous approaches to composite filter design, this method considers the training set selection and the filter design simultaneously. The LPCCF is obtained by summing the training images weighted by complex weights of unit magnitude and linear phase. This paper also presents a way of combining the outputs of a bank of LPCCFs. Computer simulations are also included to quantify the performance of this approach.

Journal ArticleDOI
TL;DR: Phase detection of Young's fringes is applied to a highly precise retardation measurement using a simple common-path polarizing interferometer with a birefringent wedge and a polarizer.
Abstract: Phase detection of Young’s fringes is applied to a highly precise retardation measurement. A simple common-path polarizing interferometer is used with a birefringent wedge and a polarizer. The birefringent wedge introduces a spatially linear phase difference between orthogonally polarized light and Young’s fringes are formed on an image sensor. The phase difference between the orthogonally polarized components of light is proportional to the phase of Young’s fringes. Thus, the retardation is equal to the Young’s fringes’ phase change before and after insertion of the retarder into the common-path interferometer. The phase of Young’s fringes is calculated from the Fourier cosine and sine integrals of the fringe profile. The experimental results for wave plates, a Soleil-Babinet compensator, and a Pockels cell are presented with error estimates. The accuracy of the retardation measurement is experimentally estimated to be greater than λ/2100.

Book ChapterDOI
23 Apr 1990
TL;DR: The method of phase differences is shown to be equivalent to a Newton-Raphson root finding iteration through the resolutions of band-pass filtering, and ideas are extended into 2-D where it is possible to extract both vertical and horizontal disparities.
Abstract: We apply the notion that phase differences can be used to interpret disparity between a pair of stereoscopic images Indeed, phase relationships can also be used to obtain probabilistic measures both edges and corners, as well as the directional instantaneous frequency of an image field The method of phase differences is shown to be equivalent to a Newton-Raphson root finding iteration through the resolutions of band-pass filtering The method does, however, suffer from stability problems, and in particular stationary phase and aliasing The stability problems associated with this technique are implicitly derived from the mechanism used to interpet disparity, which in general requires an assumption of linear phase and the local instantaneous frequency We present two techniques Firstly, we use the centre frequency of the applied band-pass filter to interpret disparity This interpretation, however, suffers heavily from phase error and requires considerable damping prior to convergence Secondly, we use the derivative of phase to obtain the instantaneous frequency from an image, which is then used to improve the disparity estimate These ideas are extended into 2-D where it is possible to extract both vertical and horizontal disparities

Journal ArticleDOI
TL;DR: The relevance of the local phase information for texture discrimination and image segmentation is studied and experimental results seem to confirm the importance of theLocal phase information.

PatentDOI
TL;DR: In this article, the average phase difference between consecutive ones of phase vectors obtained at an interval T equal to the period of transmitted wave is detected, a plurality of phase difference values are added and averaged to obtain an average phase contrast value, and a Doppler frequency is calculated from the average difference value to obtain the average speed of the object, in order to remove an error caused by aliasing of an angle in the course of the arithmetic operation for obtaining the average contrast value.
Abstract: A flow meter for detecting the average speed of an object in such a manner that an ultrasonic pulsed continuous wave is repeatedly transmitted toward the object to obtain a phase vector from the reception signal of the wave reflected from the object, the phase difference between consecutive ones of phase vectors obtained at an interval T equal to the period of transmitted wave is detected, a plurality of phase difference values are added and averaged to obtain an average phase difference value, and a Doppler frequency is calculated from the average phase difference value to obtain the average speed of the object, in which in order to remove an error caused by aliasing of an angle in the course of the arithmetic operation for obtaining the average phase difference value, a plurality of presumed average phase difference values having the error are calculated from the average phase difference value, the difference between the change of a phase caused by adding the phase difference values successively and the change of phase based upon each of the presumed average phase difference values is estimated to select the most appropriate one of the presumed average phase difference values, and the selected average phase difference value is used as a corrected average phase difference value.

Patent
22 Aug 1990
TL;DR: In this paper, a phase equalization system for a digital-to-analog converter (DAC) includes a digital portion (10) having an interpolation section (14) for receiving a digital input and increasing the sampling frequency thereof for input to a delta-sigma modulator (16).
Abstract: A phase equalization system for a digital-to-analog converter (DAC) includes a digital portion (10) having an interpolation section (14) for receiving a digital input and increasing the sampling frequency thereof for input to a delta-sigma modulator (16). A summing junction (24) is disposed between the interpolation circuit (14) and the delta-sigma modudlator (16) to allow an offset voltage to be summed therewith. This provides for D.C. offset, this offset being controlled by a calibration control (40). The output of the digital section (10) is input in an analog section (12), which has a one-bit DAC 21) that is input to an analog filter (22) for converting and filtering the one-bit digital stream output by the delta-sigma modulator (16). The interpolation circuit (14) includes a three stage interpolation filter comprising a first stage (50), a second stage (52) and a third stage (54). The second stage (52) is comprised of a finite impulse response filter (FIR) that has a nonlinear phase response. The nonlinear phase response of the interpolation filter (52) compensates for the phase deviation of the analog filter (22) from a linear phase response. Therefore, the composite phase provided by the combination of the phase equalization in the digital section (10) and the phase nonlinearity in the analog section (12) will result in a linear overall phase relationship for the DAC.

Patent
Shousei Yoshida1
13 Sep 1990
TL;DR: In this article, the authors proposed a carrier recovery system combining a narrow band carrier phase estimator and a wide band CPA, and a digital phase demodulator using this system.
Abstract: A carrier recovery system combining a narrow band carrier phase estimator and a wide band carrier phase estimator, and a digital phase demodulator using this carrier recovery system. The narrow band carrier phase estimator extracts a carrier phase having a low pulse slip rate from received signal. This carrier phase, however, cannot track fast phase fluctuation due to fading. On the other hand, the wide band carrier phase estimator tracks fast phase fluctuation due to fading, and extracts a carrier phase from the received signal. From this carrier phase is subtracted the other carrier phase obtained by the narrow band carrier phase estimator, and the balance, after its fluctuating range is limited, is added to the carrier phase obtained by the narrow band carrier phase estimator, thereby making it possible to obtain a carrier phase which can track fast phase fluctuations due to fading without sacrificing the phase slip rate. By using the carrier phase obtained in this manner, there can be structured a digital phase demodulator suitable for mobile communication systems based on coherent detection. If carrier component is extracted by reversely modulating the received signal with reference signal obtained by subjecting to decision the signal demodulated by this digital phase demodulator, a carrier phase is recovered by filtering the extracted carrier component in a wide band, and the received signal is synchronously detected with this carrier phase, the error rate can be further reduced.

Journal ArticleDOI
TL;DR: In this paper, the authors investigated some approaches for designing one-dimensional linear phase finite-duration impulse-responses (FIR) notch filters, which are based on the modification of several established design techniques of linear phase FIR band-selective filters.
Abstract: This paper investigates some approaches for designing one-dimensional linear phase finite-duration impulse-responses (FIR) notch filters, which are based on the modification of several established design techniques of linear phase FIR band-selective filters Based on extensive design examples and theoretical analysis, formulae have been developed for estimating the length of a linear phase FIR notch filter meeting the given specifications In addition, the design of two-dimensional linear phase FIR notch filters is briefly considered Illustrative examples are included

Journal ArticleDOI
TL;DR: In this paper, a solution of the problem of electromagnetic scattering of obliquely incident plane waves by homogeneous, nonlinear anisotropic cylindrical structures is obtained, where the medium of the scatterer is characterized by Volterra-type integrals for the electric and magnetic flux density vectors D and B, respectively.
Abstract: The solution of the problem of electromagnetic scattering of obliquely incident plane waves by homogeneous, nonlinear anisotropic cylindrical structures is obtained. The medium of the scatterer is characterized by Volterra-type integrals for the electric and magnetic flux density vectors D and B, respectively. The nonlinear problem is solved using the perturbation method. The effects of nonlinearities on the field properties both inside and outside the scatterer, together with the effect on the radar cross section, are investigated for the fundamental frequency components. To demonstrate the validity of the approach, the results obtained by the perturbation method are compared with those obtained using the plane wave representation method of D. Censor (1983), where the iteration method is used to solve the resulting dispersion equation. The results are in very good agreement in both amplitude and phase of the fields for the case of very weak nonlinearity. When the relative magnitude of the nonlinear component of the permittivity is increased, the iteration method shows a faster divergence of the phase from the linear phase. >

Proceedings ArticleDOI
01 May 1990
TL;DR: In this paper, a recursive structure for implementing long linear-phase finite impulse response (FIR) filters using a very small number of multipliers is introduced, which uses the principle of switching and resetting between two identical copies of the same IIR filter.
Abstract: Recursive structures are introduced for implementing long linear-phase finite impulse response (FIR) filters using a very small number of multipliers. The implementation uses the principle of switching and resetting between two identical copies of the same infinite impulse response (IIR) filter. The impulse response of the resulting filters is a truncated and shifted version of the response of a filter G(z)G(z/sup -1/) where G(z) is a stable IIR stable and G(z/sup -1/) is the corresponding unstable IIR filter. The filters are implemented as a parallel connection of several branches, each branch generating a truncated response corresponding to a complex conjugate pole pair and its mirror-image pair. The truncation is performed using a feedforward term which provides pole-zero cancellations. To stabilize the pole-zero cancellations, and to avoid the quantization error from growing too much, the branch filters are implemented by applying the principle of switching and resetting. It is shown, by means of an example, that by using this approach a nearly optimum FIR filter of order larger than 500 using just 17 adjustable parameters can be designed. >

Patent
Melvin G. Montgomery1
26 Dec 1990
TL;DR: In this paper, a phase shift keyed telemetry system is described, which is particularly adapted to demodulation of phase shift-keyed signals where the medium of transmission is non-ideal, such that transitional cycles are present at each change of phase of the transmitted data.
Abstract: A phase shift keyed telemetry system is disclosed, which is particularly adapted to demodulation of phase shift keyed signals where the medium of transmission is non-ideal, such that transitional cycles are present at each change of phase of the transmitted data. Demodulation is accomplished by comparison of the input signal to a reference signal generated from the input signal itself. The reference signal is generated by frequency dividing the input signal, preferably after full wave rectifying, by a multiple which is larger than the number of transitional cycles expected; the frequency divided signal is then multiplied in frequency to substantially match the frequency of the input signal. The use of the divider and multiplier removes the effects of transitional cycles of varying phase in generating the reference signal. A phase locked loop in the reference signal generating circuit also reduces the effects of frequency and phase shifting during the transitional cycles, as well as providing tolerance for missing cycles due to system noise.

Journal ArticleDOI
TL;DR: In this paper, an alternative method based on oversampling is proposed to correct the time-domain data to obtain a flat baseline in the resulting spectra, which can be used for protein 3D structure determination by NMR spectroscopy.

Patent
07 Feb 1990
TL;DR: In this paper, an optical notching filter is described, in which an optical spatial filter is formed by a spatial light modulator which is written into by light, and a write light source can be independent of the read light source, such that any suitable write or read light sources can be utilized therein, either with single pass or recursive optical arrangements.
Abstract: An optical notching filter is disclosed in which an optical spatial filter therein is formed by a spatial light modulator which is written into by light. The optical notching filter system of the present invention provides a linear phase and amplitude response with applications for high speed adaptive filtering to filter out noise and signals of no interest. In one particular disclosed embodiment, the write with light optical notching filter system of the present invention is designed to adaptively excise interference from Radio Frequency (RF) signals, such as are utilized in communications or radar systems. The technical approach of the present invention provides a more compact system design than similar electronically addressed optical notching filtering systems, and also significantly reduces the complexity and attendant power requirements of the system. Different embodiments of a write with light optical notching filter system can operate with either nonochromatic or polychromatic light sources. Moreover, the write light source can be independent of the read light source, such that any suitable write or read light source can be utilized therein, either with single pass or recursive optical arrangements.

Patent
09 Apr 1990
TL;DR: In this article, a crystal oscillator circuit for use with overtone crystals restricts its resonant frequency to either fundamental or overtone frequencies, by adjusting a current source which changes time delay, without using inductors or other reactive components.
Abstract: A crystal oscillator circuit for use with overtone crystals restricts its resonant frequency to either fundamental or overtone frequencies. An inverting amplifier, which provides 180 degrees of phase shift by inverting an input signal, includes components which effectively provide another 180 degree phase shift at a particular frequency by adding a time delay. The 180 degree phase shift by a signal inverting amplifier, with the included 180 degree phase shift provided by the time delay comprises an oscillator by the addition of a crystal with a resonant frequency equal to the particular frequency determined by the time delay. Suppression of unwanted oscillation frequencies is accomplished by adjusting a current source, which changes time delay, without using inductors or other reactive components.

Journal ArticleDOI
TL;DR: In this article, a novel phase correction method is presented, which starts from phase data that may be corrupted with (measurement) noise, due to an appropriate choice of the model, the phase intercept distortion is avoided.
Abstract: A novel phase correction method is presented, which starts from phase data that may be corrupted with (measurement) noise. Due to an appropriate choice of the model, the phase intercept distortion is avoided. Some practical examples and experimental results are given. >

Proceedings ArticleDOI
03 Apr 1990
TL;DR: A technique for designing IIR (infinite impulse response) filters with a complex-valued frequency response is proposed, whose real and imaginary parts represent the log magnitude and phase errors, respectively, is minimized using an iterative weighted-least-squares technique in the frequency domain.
Abstract: A technique for designing IIR (infinite impulse response) filters with a complex-valued frequency response is proposed The complex logarithmic error, whose real and imaginary parts represent the log magnitude and phase errors, respectively, is minimized in the Chebyshev sense using an iterative weighted-least-squares (WLS) technique in the frequency domain Both the log magnitude and phase errors of the resulting filter become nearly equiripple Filter coefficients at each iteration are efficiently computed using a fast recursive algorithm for a set of linear equations which is derived from the WLS problem >

Patent
Jose I. Suarez1
29 Nov 1990
TL;DR: In this article, a phase shifter is coupled with a voltage controlled oscillator to the first integrator and provides the phase shifted signal that represents a function of the phase of the signal from the VCO.
Abstract: A fast locking phase locked loop includes a first integrator that provides a signal representing a function of the mathematical or ideal integral of the phase difference between an input signal and a feedback signal. A voltage controlled oscillator is coupled to the first integrator and provides a signal to a phase shifter that provides the phase shifted signal that represents a function of the phase of the signal from the VCO, and a function of the integral of the phase difference between the integrated signals.

Journal ArticleDOI
TL;DR: In this paper, the solitary-wave solutions of the four-wave equations are studied and their relevance to four wave mixing in finite media is discussed, and the authors show that by controlling the linear phase mismatch judiciously, it is often possible to effect a complete transfer of action from the pump waves to the probe and signal waves.
Abstract: The solitary-wave solutions of the four-wave equations are studied, and their relevance to four-wave mixing in finite media is discussed. In general, the transfer of action from the pump waves to the probe and signal waves is limited by nonlinear phase shifts that detune the interaction. However, by controlling the linear phase mismatch judiciously, it is often possible to effect a complete transfer of action from the pump waves to the probe and signal waves.

Patent
15 May 1990
TL;DR: In this paper, a closed-loop configuration of an adder, a filter, and a delay circuit is presented, where the output of the adder is fed back to the filter via the all-pass filter and delay circuit as the feedback signal.
Abstract: A musical tone synthesizing apparatus has a closed-loop configuration including an adder, a filter and a delay circuit. The adder adds its feedback signal to a signal to be synthesized which is applied from an external device. The filter constructed as the all-pass filter has a frequency characteristic by which a phase delay between its input and output signals is varied in response to a frequency variation of its input signal. Such all-pass filter includes a delay element having a delay time which is set longer than a predetermined unit delay time corresponding to a sampling period to be employed. Herein, the output of adder is fed back to the adder via the all-pass filter and delay circuit as the feedback signal. Thus, a signal circulating the closed-loop is picked up as a synthesized musical tone signal.

Journal ArticleDOI
TL;DR: In this paper, a design technique and an associated algorithm for selective linear-phase low-pass and high-pass sampled data filters are presented for realization as leap-frog lossless discrete integrator (LDI) switched-capacitor structures and also in wave digital form.
Abstract: A design technique, and the associated algorithm, are presented for selective linear-phase low-pass and high-pass sampled data filters. The resulting filter transfer functions are particularly suitable for realization as leap-frog lossless discrete integrator (LDI) switched-capacitor structures and also in wave digital form. Due to the specific form of the transfer function, the filters generally have no lumped-element counterparts. The technique leads to an optimum compromise between amplitude selectively and passband phase linearity under the restriction of a prescribed form of the transfer function, and allows the use of finite j omega -axis zeros of transmission for achieving high amplitude selectivity. The transfer functions presented are also capable of realization as distributed microwave filters, but these require the use of coupled-wire networks in the form of distributed Brune sections, if j omega -axis zeros of transmission are introduced. >