Search

This paper proposes a new type of hybrid eighth‐mode substrate integrated waveguide (EMSIW) and microstrip line (MSL) bandpass filter (BPF) with mixed electric and magnetic coupling. Firstly, two second‐order EMSIW BPFs with mixed coupling and wide stopband performance are investigated. Then, in order to realise high‐order BPF with decent performance, half‐wavelength (λ/2) open‐circuited MSL resonators are employed in the hybrid BPFs design. In the end, two third‐order BPFs and a fourth‐order BPF are developed by combining the EMSIW and MSL resonators together. Extra transmission zeros can be introduced due to the cross couplings in the trisection and quadruplet topologies. For demonstration, the proposed prototypes are designed, fabricated and measured. Reasonable agreements can be achieved between the measurement and simulation. The proposed third‐order BPFs compete against the state of the art, exhibiting the competitive characteristics of the operating bandwidth, physical footprint and upper‐stopband rejection with less order. The proposed fourth‐order BPF features a compact size of 0.34 × 0.34λ0, a low insertion loss of 1 dB, a broad 3‐dB fractional bandwidth of 22.9% and an excellent lower‐stopband suppression. [ABSTRACT FROM AUTHOR]

SUMMARY Electro-magnetic interface (EMI) noise radiated from the dc cable of photovoltaic power conditioner systems (PCSs) is a serious problem for radio transmission equipment. In order to prevent the interference, the International Special Committee on Radio Interface (CISPR) is discussing to determine a noise regulation value for this issue. To reduce the radiated noise, this paper studies the attenuation characteristics of EMI filters that are connected to both the dc and the ac sides of the PCS. In this paper, it is pointed out that the EMI filters do not show their own attenuation characteristics if those are connected to the PCS. In particular, when EMI filters are connected to both the dc and the ac sides of the PCS, a specific series resonant frequency is observed and deteriorates the attenuation characteristics of both EMI filters. In order to prevent this problem, a design method for EMI filters which can reduce the common-mode EMI noise effectively, is proposed in this paper. Further, the effectiveness of the proposed method is confirmed through experiments. [ABSTRACT FROM AUTHOR]

In this paper, new bridged-T network operating with bandpass (BP) negative group delay (NGD) function is introduced. The circuit design is innovatively based on an inductorless passive RCnetwork. The innovative bridged-T topology is implemented with a R-shunt configuration. The analytical transfer function of voltages and equivalent impedance-matrix are carefully calculated. The BP NGD analysis and design of the bridged-T topology is developed. The BP NGD properties and specifications are analytically investigated. A proof of concept of R-shunt bridged-T prototype with a surface-mounted device printed circuit board is designed, fabricated, and measured to verify the BP NGD behavior from theoretical equations and simulations. As expected, the theoretical, simulation, and measurements results are in very good agreement. From measurement results, the fabricated BP NGD C-shunt bridged-T reaches a delay of about -45 ns and a center frequency of about 0.5 MHz over a -3 dB attenuation. [ABSTRACT FROM AUTHOR]

With the development of photovoltaic power generation technology, a large number of micro-power grid with distributed generation system have appeared in remote areas. It has the longer transmission line between the traditional power grid and the inverter of the distributed generation system. This caused the equivalent network impedance would be increased and would weak the power-grid characteristics. When the photovoltaic inverter is connected to a weaker power grid, the bandwidth of the control system of the inverter will be reduced. Using the traditional PR control to solve the problem is not ideal. This paper proposes fuzzy quasi-PRD control to improve the current quality. The inverter with the LCL filter has the peak of the resonance oscillation, it will affect the stability of the system. The paper adopts the adjustable virtual damping to suppress resonance. At last, the idea is verified by MATLAB/Simulink. The results show that the fuzzy quasi-PRD control method has better control performance, and dynamic tracking characteristics. [ABSTRACT FROM AUTHOR]

The converter is the main component of the converter station and the main noise source of the PLC interference. The accurate modelling of the high-frequency characteristics of the converter is the foundation for the design of the ± 1100 kV UHVDC PLC filter. Based on deep analysis of the high-frequency characteristics of the converter valve, this paper first proposes three high-frequency models of the converter valve. Compared with a real converter valve used in similar HVDC project, model two is chosen to analyse the high-frequency characteristics. Second, the model of Zhundong–Wannan UHVDC transmission project is established in the PSCAD simulation platform to determine the actual high-frequency characteristics of the converter valve. The voltage before the smoothing reactor is calculated, and the high-frequency noise level is obtained by Fourier transform. Considering the practical application of the high-frequency model of the converter valve, this paper finally proposes a formula to express the measured noise level. The noise source expressed by the formula is more practical for research on high-frequency characteristics of the converter station. [ABSTRACT FROM AUTHOR]

Summary: A power divider is ideally a lossless reciprocal device performing vector summation of two or more input signals, which is an extremely important circuit component in wireless communication systems or radar systems. In this paper, a general concept of lumped‐element power divider is discussed. Then, a semi‐analytical synthesis technique is presented. General odd‐mode and even‐mode analysis method is proposed for multi‐port symmetrical network and applied in the analysis of N‐way lumped‐element power dividers. The transmission networks and the matching networks can be synthesized step by step, which greatly simplify their analysis. The transmission networks can be synthesized by those well‐known synthesis techniques for two‐port networks, and thus the transmission property of power dividers can be accurately controlled. The matching networks are determined by optimization to ensure certain isolation between the output ports. The design procedure is simple and accurate. Apart from the capability of power division, power dividers are also able to play the role of impedance transform and signal processing. Some examples of power dividers with filtering capability are presented for demonstration. In this paper, a general concept of lumped‐element power divider is discussed. Then, a semi‐analytical synthesis technique is presented. General odd‐mode and even‐mode analysis method is proposed for multi‐port symmetrical network and applied in the analysis of N‐way lumped‐element power dividers. Transmission networks and matching networks can be synthesized step by step, which greatly simplify their analysis. Apart from the capability of power division, power dividers are also able to play the role of impedance transform and signal processing. In this paper, a general concept of lumped‐element power divider is discussed. Then, a semi‐analytical synthesis technique is presented. General odd‐ and even‐mode analysis method is proposed for multi‐port symmetrical network and applied in the analysis of N‐way lumped‐element power dividers. Transmission networks and matching networks can be synthesized step by step, which greatly simplify their analysis. Apart from the capability of power division, power dividers are also able to play the role of impedance transform and signal processing. [ABSTRACT FROM AUTHOR]

This paper represents a new dual‐mode waveguide filter based on a stereometamaterial resonator. A novel resonator is proposed in the paper consisting of two stacked complementary split ring resonators (CSRR) where one of them is twisted in its plane. This resonator provides a transmission zero, which can be easily adjusted in both of the upper and lower stopbands of the filter. By cascading the four of the proposed resonator, an eight order symmetric filter with four transmission zeros are designed. The filter parameters are optimized using the genetic algorithm (GA). This filter is fabricated and the experimental results are in a good agreement with the simulation data. Besides the flexibility in the design, the most important characteristic of the proposed structure is its compactness compared to similar structures. [ABSTRACT FROM AUTHOR]

The reference signals used in a multichannel system are usually strongly crosscorrelated. Preprocessing is normally added to reduce the crosscorrelation between the reference signals because this correlation makes estimating the adaptive filter coefficients difficult. However, this preprocessing is equivalent to warping the reference signals. The transmission of these warped reference signals to an unknown system hinders the essential system operation. On the other hand, it is necessary for uncorrelated components between the reference signals to exist when estimating the adaptive filter coefficients. Preprocessing that increases the ratio of these components is important in improving the convergence characteristics. In this paper, we propose an algorithm that does not transmit preprocessed reference signals to an unknown system, but only uses them to estimate the adaptive filter coefficients, and explain the design conditions needed to implement this algorithm. In other words, with the condition of not increasing the estimation errors, we derive the optimum values of the coefficients that reduce the correlation between the reference signals, which is a feature of this algorithm, and verify its effectiveness in simulations. By employing the proposed algorithm, improvements in the convergence characteristics can be designed without affecting the essential operation of the system. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 88(3): 32–41, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20092 [ABSTRACT FROM AUTHOR]

As a method to obtain the baseband signal from each of several time division multiple access (TDMA) signals that are frequency-multiplexed in the uplink in wireless communication, this paper proposes a method of group demodulation using a multirate filter bank. The proposed method allows demodulation when the TDMA signals are asynchronously received in the uplink. Further, it can be applied to the case in which TDMA signals with different bandwidths are frequency-multiplexed, if the multiple resolution capability of the multirate filter bank is used. Hence, this method is useful in the situation where several different types of signals coexist in such environments as software radio. In the systems discussed in this paper, it is necessary to realize a sharp amplitude response and a linear phase response in the filters of the multirate filter bank with a small computational effort. This is accomplished in this paper by using an infinite impulse response (IIR) filter with approximately linear phase response. © 2001 Scripta Technica, Electron Comm Jpn Pt 1, 85(4): 44–52, 2002 [ABSTRACT FROM AUTHOR]

This paper proposes three balanced multiple‐layer dual‐mode substrate integrated waveguide (SIW) bandpass filters. The first balanced filter has two substrate layers, and two pairs of balanced feeding ports are located on the upper substrate layer. The lower and upper dual‐mode SIW cavities are coupled by two crossed slots in the middle metal layer. The second balanced filter has two substrate layers, and two pairs of balanced feeding ports locate parallelly on the upper and lower substrate layers, respectively. The third balanced filter has three substrate layers, and two pairs of balanced feeding ports locate vertically on the upper and lower substrate layers, respectively. The proposed three balanced filters can produce three, two and one finite‐transmission zeros (FTZs), respectively. Characteristics of the proposed multiple‐layer SIW filters with controllable FTZs' locations are analysed in detail. For the demonstration, some design examples with the centre frequency of 7.5 GHz and bandwidth of 240 MHz are presented, and three design examples are fabricated to verify the performance of the proposed balanced filters. The measured results agree well with simulated ones. [ABSTRACT FROM AUTHOR]

Summary: In this paper, a new structure for a DC‐DC boost converter is proposed. The presented converter provides a high voltage transfer gain with lower duty cycle. Low current and low voltage stress on the switch, enlarged area of operating in continuous conduction mode (CCM), reduced size of the inductors, and the input filter are the main advantages of the proposed converter. The high voltage transfer gain with low number of elements has made it suitable to implement. Hence, using only one switch has made the control of the proposed converter easy. Besides, decreased switching losses and higher efficiency are obtained. The proposed structure is a combination of the Luo converter and a booster unit, which its analysis is studied in three modes, CCM, boundary conduction mode (BCM), and discontinuous conduction mode (DCM). Furthermore, in order to validate the analysis and feasibility of the proposed converter, the experimental results are developed on a low power prototype. In this paper, a new structure for a DC‐DC boost converter is proposed. The presented converter provides a high voltage transfer gain with lower duty cycle. Low current and low voltage stress on the switch, enlarged area of operating in continuous conduction mode, reduced size of the inductors, and the input filter are the main advantages of the proposed converter. [ABSTRACT FROM AUTHOR]

This paper presents four frequency selective surfaces (FSSs) to be used in an instantaneous frequency measurement (IFM) systems. The FSS based on simple dipoles have a frequency response corresponding to the characteristics of frequency discriminators. An IFM system with 4‐bits is shown with resolution around 300 MHz. The system contains 16 distinct frequency sub‐bands between 3.21 and 8.75 GHz. CST software simulations and measurements of insertion loss are presented and compared with theoretical results to provide the binary codes of the system. [ABSTRACT FROM AUTHOR]

The work reported in this paper introduces a periodic switching technique applied to continuous-time filters, whose outcome is an equivalent filter with scaled time-constants. The principle behind the method is based on a procedure that extends the integration time by periodically interrupting the normal integration of the filter. The net result is an up scaling of the time constant, inversely proportional to the switching dutycycle. This is particularly suitable for reducing the area occupied by passive devices in integrated circuits, as well as to accurately calibrate the filter dynamics. Previous works have been following this concept in an entirely continuous-time perspective, either focusing on specific circuits or using approximations to provide an extended analysis. This paper includes input/output sampling to derive a closed-form representation for the scaling technique herein coined as 'Filter & Hold' (F&H). A detailed mathematical analysis is described, demonstrating that the F&H concept represents an exact filtering solution. Simulation results and experimental measurements are provided to further validate the theoretical analysis for an F&H vector-filter prototype. [ABSTRACT FROM AUTHOR]

In this paper, a novel and simple dual-polarized frequency-selective rasorber (FSR) which has a wide absorption band including a wide transmission band was presented. Detailed equivalent circuit was derived to accurately predict the frequency responses of the structure. Based on reasonable equivalent analysis of a traditional metal grid structure, a modified structure with improved performance was proposed. The structure consists of a lossy frequency-selective surface (FSS) constructed with metal rings connected by center loaded strips as the top layer, and a broadband transmission FSS at the bottom, while the two layers are separated by air. To validate our proposal, a prototype was fabricated and measured. The measured results are in good agreements with simulations under normal incidence, which indicates a broad absorption bandwidth of 80% that from 3.40 to 8.16 GHz; in the meantime, a wide transmission band with an insertion loss of less than -1.25 dB is ranging from 9.03 to 12.01 GHz. It should be pointed out that, the reflection from the whole structure in the operation band can be kept below -10 dB from 4.15 to 11.87 GHz. [ABSTRACT FROM AUTHOR]

SUMMARY This paper proposes a new control method suitable for active power filters, which can reduce the dc capacitor voltage ripple associated with the third-order harmonic current compensation. The proposed method superimposes a negative-sequence fundamental current on the compensating current to cancel out the active power ripple caused by the third-order harmonic current. As a result, the proposed method has the capability to eliminate the dc capacitor voltage ripple oscillating at double the source frequency. Experimental results obtained by a 10-kW three-phase diode rectifier load verify the validity of the proposed method. The proposed method exhibits a small dc capacitor voltage ripple reduced to 43% of that using the conventional method. [ABSTRACT FROM AUTHOR]

A novel continuously tunable lowpass filter (LPF) with compact size, sharp roll-off and low insertion loss is presented in this paper. The filter employs two varactor diodes, a pair of open-ended coupled lines and a U-shape step impedance line (SIL) with an open-ended stub loaded at the center of the SIL to form a very compact layout. The odd- and even-mode analysis and equivalent circuit model are demonstrated for estimation of the transmission characteristics. Tuning the DC voltage applied on the varactor diodes, the varactor capacitance accordingly changes leading to a varying cutoff frequency fc. The measured results show that the achieved 3-dB fc tuning range is 60.6% (1.15-2.15 GHz). The measured insertion loss (IL) and roll-off rate are 0.2-0.4 dB and 50-73 dB/GHz, respectively. The overall size of the LPF is only 0.005λg2, which shows a competitive advantage comparing with the state-of-the-art work. [ABSTRACT FROM AUTHOR]

To design a highly efficient video codec, comparisons of available techniques front various viewpoints are necessary so that the coding scheme and parameters are chosen appropriately. In recently presented papers and equipment, many systems use motion-compensated interframe prediction and DCT or VQ schemes for the prediction error coding. A loop filter often is adopted to improve coding efficiency. This paper compares DCT and VQ schemes in an interframe coding scheme, in an optimal blocksize for interframe prediction error signal coding and on the Effect of loop filtering from the viewpoints of coding efficiency and reproduced image quality Simulations using actual. image sequences show that the optimal block size is 4 × 4 or 8 × 8, that the loop filter improves significantly the coding efficiency and reproduced image quality, and that DCT and VQ schemes give nearly the same efficiency with an appropriate loop filter. [ABSTRACT FROM AUTHOR]

Radon discussed the reconstruction problem of a two-variable function from its line integrals and the algebraic representation of this problem. He also discussed the reconstruction from curved-line integrals curved-line integrals and extended the discussion to the reconstruction problem of a multivariable function from its hyperplane integrals. For the first two of these problems, reconstruction algorithms have been developed. They have been applied to X-ray CT (Computed Tomography). The third problem has been studied in connection with ultrasonic CT. On the other hand, very little engineering development has been reported on the fourth problem. However, in a practical problem, it is desired to find a three-dimensional structure based on its projections measured as plane integrals. Also, in statistical analysis, a joint probability density function of multiple variables is often desired based on its marginal distributions. Moreover, the reconstruction problem in multidimensional spaces is of mathematical interest. In this paper, a foundation to this problem and theoretical formulas for a filtered back-projection method for reconstruction are given in a simple form based on a discussion similar to that in CT. For numerical calculation filtering is usually implemented by convolution in CT. In this paper, extensions of the kernel functions for convolution to the multidimensional problem are also presented. [ABSTRACT FROM AUTHOR]

This paper proposes a method of realizing an adaptive normalized ARMA lattice filter. This filter was proposed in a previous paper and can be realized with an arbitrary ARM A order and with all the lattice coefficients less than one. Furthermore, these coefficients are equal, so the structure is simplified. The adaptive realization method proposed in this paper retains these features. In addition, we perform spectral estimation of a time-varying signal with frequency weighting and model identification of an unknown input system. In our paper we describe an experiment with a real speech signal to prove that our algorithm can be used for frequency-weighted spectral estimation. The use of real data to perform spectral estimation has not been attempted in other references. [ABSTRACT FROM AUTHOR]

This paper presents a method of designing hybrid analog/asymmetrical square-root (SR) FIR filters. In addition to the conventional frequency domain constraints, the proposed method considers time-domain constraints as well, including the inter-symbol interference (ISI) and the opening of the eye pattern at the receiver output. This paper also reviews a systematic way to find the discrete-time equivalence of analog parts in a band-limited digital communication system. Thus, a phase equalizer can be easily realized to compensate for the nonlinear phase responses of the analog components. With the hybrid analog/SR FIR filter co-design, examples show that using the proposed method can result in a more robust ISI performance in the presence of the receiver clock jitter. [ABSTRACT FROM AUTHOR]

Cross‐coupled microwave filter with flexible topology is an essential component in a modern communication system. Among the various filter networks, the use of an extended box section can effectively reduce the difficulty of filter design, but the synthesis approach for this popular structure is not well developed. This paper presents the synthesis method based on the folded coupling matrix to derive the extended box sections. At the same time, an eighth‐degree filter with extended sixth‐degree box sections is synthesized to demonstrate the correctness of the approach. Finally, the new approach's feature to derive advanced circuits such as peculiar quintets is discussed, indicating that it has certain universality. [ABSTRACT FROM AUTHOR]

This paper describes the improvement of the skirt characteristics of a bandpass filter (BPF) by using mixed tapped resonators. In this paper, we try to examine the transmission characteristics of the BPFs with a function to control the number of attenuation poles based on the use of a λ/2 resonator and a λ/4 resonator as the basis of the distributed constant line resonators. As a result, we have confirmed that realization of the attenuation poles has a greater effect than choice of the number of resonators or choice and location control of the numbers of realized poles. The miniaturization of the presented BPF is also accomplished as well as location of the attenuation poles mentioned above. It has been confirmed that our proposal enables the control of the number of attenuation poles of the BPF. © 2005 Wiley Periodicals, Inc. Electr Eng Jpn, 151(4): 36–46, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/eej.20074 [ABSTRACT FROM AUTHOR]

Highly tunable diodes and multidiode biasing configurations can be employed to enhance the tuning range of frequency-selective surfaces (FSS). The paper considers a variety of such diodes and configurations, demonstrating experimentally a wideband tuning that can be in excess of 500 MHz. However, the presence of varactor biasing electrodes leads to the distortion of the FSS response. The paper describes how the problem can be eliminated through the novel use of surface-mount resistors. The experimental results are obtained in a waveguide setup, within the 1.6–2.7-GHz range. © 2005 Wiley Periodicals, Inc. Microwave Opt Technol Lett 44: 412–416, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/mop.20651 [ABSTRACT FROM AUTHOR]

This paper studies an adaptive filter with a restricted linear sum of the moments of tap coefficients and investigates its applications. In general, when a sampled value filter or a digital filter represented by a surface acoustic filter is realized, the maximum absolute values or the variances of the tap coefficients are restricted. In this paper, restrictions on the maximum values or the variances are imposed by means of a restricted linear sum of several moments around an arbitrary positive number, facilitating the realization of the filter. The number of the moments and the weights are determined in such a way that the frequency characteristics of the filter after restriction satisfy the given specifications. The adaptive system consists of an ideal filter without restriction on the tap coefficients and an adaptive filter placed in parallel. The adaptive filter consists of a transversal filter. Its adaptive algorithm is a gradient algorithm with a small amount of processing, with a restricted linear sum of the moments of the tap coefficients. Since the proof of uniqueness of the converged value of the adaptive algorithm and the upper limit of the loop gain in the algorithm are formulated, the filter designer can easily obtain filter parameters with restricted tap coefficients. Finally, verification of the discussed results is rigorously carried out by computer simulations. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 88(3): 11–20, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20126 [ABSTRACT FROM AUTHOR]

In this paper, a microstrip filter has been designed, fabricated, and measured for application in ultra-wideband antennas. By using the combination of a band-stop filter and a low-pass filter, the band-selective characteristics can be demonstrated. In order to enhance the suppression of the out-of-band signals, a defected ground structure (DGS) with stopband performance is used. The passband responses of the filter, which range below 2.8 GHz and from 5.0 to 6.0 GHz, cover most commercial wireless communication systems. The LC-lumped equivalent circuit of the filter has been extracted and simulated by using transmission-line analysis. © 2004 Wiley Periodicals, Inc. Microwave Opt Technol Lett 41: 248–251, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/mop.20107 [ABSTRACT FROM AUTHOR]

In this paper, a signal extension method is proposed for realization of nonextension convolution in a two-channel orthogonal FIR filter bank. Methods of performing nonextension convolution include the symmetric extension method for a linear phase filter bank and the periodic extension method applicable to an arbitrary filter bank. However, the periodic extension method is not appropriate for image compression because it becomes discontinuous at the signal edge. In this paper, for application to an orthogonal filter bank, the input signal of the decomposed filter bank is extended in such a way that the input signal in the synthesized filter bank has symmetry similar to that in the symmetric extension method. In this way, nonextension convolution is realized. In this case, it is shown that there are degrees of freedom in the extension signals determined by the input signal and the coefficients of the decomposed filter bank. The extended signal is calculated by singularity decomposition in such a way that the extended signal becomes smooth. Finally, the proposed method is applied to the coding of static images and the effectiveness of the method is demonstrated. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(3): 41–48, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10125 [ABSTRACT FROM AUTHOR]

An optimum filter is designed for separating quasi-periodic signals (those with amplitude and period changing randomly but a rough periodicity kept) in a noisy environment. In the first half of the paper, a Wiener filter is derived with steadiness assumed in the variations of the amplitude and fundamental frequency. It is shown that the frequency characteristics are (1) comb characteristics of a constant BW (bandwidth) type if only the amplitude variations are taken into consideration, (2) comb characteristics of a constant Q type if only the variations of the fundamental frequency are taken into account, and (3) comb characteristics transitional from a constant BW type to a constant Q type as the harmonic changes from a lower order to a higher order when both are taken into account. In the latter half of the paper, a Kalman filter analysis is carried out for the case in which the statistical nature of the amplitude/fundamental frequency variations is time-varying. The random nature of the complex phase term is treated as state-dependent noise. By describing the signal model in terms of the Ito probability differential equation, the Kalman filter equation for the use of this model as a state space model is solved. In particular, it is shown by approximate solution that the center frequency and occupancy bandwidth of each narrow-band filter are determined to follow the variations of the amplitude and fundamental frequency. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 86(9): 43–53, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10067 [ABSTRACT FROM AUTHOR]

In the problem of restoration of images degraded by impulsive noise, a method has been proposed to eliminate impulsive noise by using the noise position image formed with noise detectors. However, such a method estimates the signal values of the pixels to be processed by using only the nondegraded pixels within the filter window. Hence, if the majority of the window is occupied by impulsive noise, the signal information needed for estimation becomes insufficient and the estimation accuracy is significantly reduced. In addition, in the previous method, average value filters and median filters are used for estimation of the signal value of the processing points. No processing has been included that takes account of the directionality (shape) of image signals such as edge components and detail components. In the present paper, a new noise elimination filter is proposed that attempts to improve restoration accuracy by addressing these two problems. In the proposed filter, first the degraded pixels within the filter window are precompensated for the problem of insufficient original signal pixels in the filter window, and then the signal values at the processing points are estimated by the filter once sufficient signal information is obtained. Preservation of the directionality of the image signals is improved by extending the directional difference filter. Its effectiveness is verified by real image processing results. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 86(5): 65–75, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.1149 [ABSTRACT FROM AUTHOR]

This paper describes a systematic study of compensated bridges in coplanar technology. If bridges are necessary for filtering the parasitic coupled slot-line mode excited by coplanar discontinuities, they modify the electrical characteristics associated with the fundamental even mode. Based on a low-pass filter approach, a very simple technique is used to compensate for this parasitic effect. Here, emphasis is placed on the flexibility of this compensation technique, which allows us to reduce the technological constraints. Several solutions based on this approach are presented in the case of transmission lines, discontinuities such as tee junctions, and classical distributed bandpass filters. This study is valid for air bridges in classical coplanar technology, as well as for integrated dielectric bridges in multilayer coplanar technology. The concepts developed in this paper are illustrated theoretically and experimentally on multilayer devices over the 1–50 GHz frequency band. © 2002 Wiley Periodicals, Inc. Microwave Opt Technol Lett 32: 402–408, 2002. [ABSTRACT FROM AUTHOR]

Electric filters have a relevant importance in electronic systems because they are present in almost any electronic system. For example, communication systems, as many other electric systems, make intensive use of filtering to separate unwanted noise from the desired signal. Unfortunately, filter design is an intensive computational task requiring a significant amount of numerical calculations to obtain either the parameters of a filter transfer function or the element values for a filter circuit realization. This paper describes a software package whose purpose is to provide a tool to be used as a teaching aid in analog and digital filter design courses. The feature of this package is that it uses MATLAB [2] for the numerical computations. The main advantage of the filter design software package described in this paper is that makes uses of one of the MATLAB toolboxes, the signals toolbox (which is used for analog and digital filter design), but used with an interface that makes possible even to the novice user to readily design filters, either analog or digital, without any previous knowledge of MATLAB or the signals toolbox. © 2002 Wiley Periodicals, Inc. Comput Appl Eng Educ 9: 259–264, 2001; Published online in Wiley InterScience (www.interscience.wiley.com.); DOI 10.1002/cae.cae.10004 [ABSTRACT FROM AUTHOR]

This paper describes pipeline realization of a gradient-type adaptive filter on the algorithmic level. By pipeline realization, the longest path (critical path) of the filter can be reduced so that the throughput characteristic can be improved. In addition, the reduction of the critical path relaxes the restriction on the operators to be used and the power consumption can be decreased. Evaluation criteria of the pipeline realization are throughput, convergence, latency, and computational complexity. Using the delayed LMS adaptive filter as one of the gradient-type adaptive filters, it is shown that pipeline realization is possible that not only has a high throughput but also convergence and latency identical to those in the LMS adaptive filter taking into account the hardware complexity. © 2001 Scripta Technica, Electron Comm Jpn Pt 3, 84(5): 33–42, 2001 [ABSTRACT FROM AUTHOR]

This paper describes cavity structure which can be used to implement triple-mode filters in a rectangular waveguide with asymmetric transmission zeros. The structure described is very simple to manufacture, and is therefore particularly suitable for diplexer applications in the Ka-band and above. As an application example, a three-pole filter is demonstrated, thereby fully validating the proposed triple-mode configuration. © 2001 John Wiley & Sons, Inc. Microwave Opt Technol Lett 28: 228–231, 2001. [ABSTRACT FROM AUTHOR]

Aiming at the elimination of Gaussian noise and impulsive noise, the authors have proposed the median filter and mean hybrid (MMH) filter, which combines the mean filter, which is useful in eliminating Gaussian noise, with the median filter, which is useful in eliminating impulsive noise. This paper proposes the learning MMH (LMMH) filter with the learning function, where the mean filter and the median filter in the MMH filter are replaced by the FIR filter and the order statistics (OS) filter, respectively, which are more general filters. The LMMH filter is optimized by learning, and the information concerning the position and the order in the training signal are well reflected on the LMMH filter, which improves the accuracy of the signal restoration, compared to the MMH filter. This paper further proposes a method by which the learning signal to be used in the learning of the LMMH filter can be constructed based only on the signal to be processed, which is degraded by the noise. By applying both the LMMH filter and the method of constructing the learning signal, a precise method for learning of the LMMH filter is provided, even in actual situations where the original signal for the signal to be processed is unknown. It is shown through an application example that the LMMH filter obtained by the learning exhibits a stable and better performance than the MMH filter. © 1999 Scripta Technica, Electron Comm Jpn Pt 3, 83(3): 95–107, 2000 [ABSTRACT FROM AUTHOR]

This paper proposes a new time‐resolved spectral analysis method based on a modification to the linear predictive coding (LPC) method for enhancing the identification of the dominant frequencies of a signal. The method described here is based on a z‐plane analysis of the LPC poles. These poles are used to produce a series of reduced order filter transfer functions which can accurately identify and estimate the frequency of the dominant spectral features. The standard LPC method has been shown to suffer from a sensitivity to noise and its performance is dependent on the filter order. The proposed method can accurately identify the dominant frequency components over a range of filter orders and is shown to be robust in the presence of noise. Compared with traditional time‐resolved methods, it is a parameterised method where the identification of the dominant frequency changes can be directly obtained in the form of frequency measurements. In a series of 10,000 Monte Carlo experiments on single component and multiple component signals, this LPC pole processing method outperforms the standard LPC method by accurately identifying the dominant frequency components in the signals. [ABSTRACT FROM AUTHOR]

In this paper, a new class of estimators for permanent magnet synchronous motors is proposed. Using a novel representation of the permanent magnet synchronous motor dynamics and some suitable filtering, we obtain new solutions to two important problems: (a) estimation of the stator resistance and inductance and (b) flux estimation with known electrical parameters. Neither one of the schemes requires knowledge of the mechanical parameters nor of the magnetic flux constant. Conditions for global convergence are given in terms of persistency of excitation requirements that, for the flux observation problem, is imposed only to the rotor speed. Both stability proofs require the availability of rotor speed. However, for the flux observer, extensive simulations using the speed estimate obtained from the estimator (instead of the actual rotor speed) illustrate the robustness of the proposed scheme vis-à-vis this assumption. Simulations also show that the new flux observer outperforms other existing designs. Copyright ©2015 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

In this paper, a class of cross-coupled differential filters is presented. The filters implemented are second order, using two transistors cross-coupled surrounded by passive RLC components. The input of each filter is differential and so is its output. Biasing of each filter is straightforward, and the stability of the filters is examined. Several of the filters were built experimentally using discrete components and simulated using SPICE in a 0.18 µm Taiwan Semiconductor Manufacturing Company (TSMC) complementary metal-oxide semiconductor (CMOS) process, confirming the validity of the filters. Copyright © 2015 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

The paper presents performance analysis of least-mean-square algorithm based adaptive filter embedded with constant false alarm rate (CFAR) detector for the purpose of better detection of target under non-homogeneous clutter environment in radar application. The objective of this paper is to develop a method by redesigning the radar detector in such a way to emphasize the target response and de-emphasize the clutter response. The hardware implementation using pipeline technique for the adaptive filter reveals its capability to support high sampling frequency, which is an ardent necessity for high performance radar. The moderate area-delay-product and low power consumption have made it suitable for hardware realization for such application. The extensive MATLAB simulation of proposed design shows remarkable improvement of detection performance in terms of signal-to-noise ratio of 17 dB considering probability of detection at 0.8 over the generic cell averaging CFAR (CA-CFAR). Copyright © 2015 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

Conventional synthesis techniques for general Chebyshev lowpass, highpass, and bandstop filters are usually based on general Chebyshev filtering function, from which lowpass prototype is set up. By applying the lowpass to lowpass, highpass, or bandstop frequency transformation on realization networks of lowpass prototypes, final realization networks of general Chebyshev lowpass, highpass, and bandstop filters are obtained. In this paper, direct synthesis techniques for general Chebyshev lowpass, highpass, and bandstop filters are discussed. Transmission zeros can be placed wherever they are desired to control the performance of the filters. Unlike conventional synthesis techniques, they derive filtering polynomials constituting network parameters directly in the lowpass, highpass, or bandstop domain, which might find applications in analogue and digital filter design. Several examples are presented for demonstration. Copyright © 2015 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

The transmission-type periodic branching filter presently is used as the multiplexer/demultiplexer for communication satellites. This type of filter can be applied to the submillimeter band multibeam communication satellite, in an attempt to realize large-capacity communication for the efficient use of frequencies in the future. It can also be applied to the branching filter in the optical communication, aiming similarly at the large-capacity communication, if the filter is constructed using the dielectric transmission line. To realize such an application, it is desired to improve the performance of the conventional periodic branching filter to realize a wider bandwidth and sharper roll-off of the amplitude transmission response. The conventional periodic branching filter is composed of two 90-deg hybrids, a quarter-wavelength transmission line and a resonator. This type of filter has a narrow usable relative bandwidth of approximately 6 percent, with a frequency response of third-order elliptic function type. This paper proposes a new transmission-type periodic branching filter with two resonators. It is composed of two hybrids and two resonators of different kinds. A bandwidth of approximately twice the width of the conventional filter end an approximately three-times sharper roll-off of the amplitude transmission response, with the frequency response of fifth-order elliptic function type, are obtained. This paper discusses the theoretical characteristics, design method and the result of experiment for this periodic branching filter. As a result of experimental construction, it was verified that the wide-band, low-loss property and the sharp roll-off of amplitude characteristic can be realized. [ABSTRACT FROM AUTHOR]

This paper proposes a simple synthesis method of impedance matching and transforming filter. This method is based on the use of a multistage ladder network simulating a non-uniform transmission line. When the filter characteristic functions defined in this paper satisfy the distortionless conditions of the voltage and current waveforms the artificial transmission line mentioned above almost operates as an ideal lowpass filter. As an example the characteristics of 1 - 2ω transformable type lowpass filter are evaluated and the obtained results almost satisfy the conditions for the ideal filter. [ABSTRACT FROM AUTHOR]

This paper describes the properties of the second-order digital bandpass filter functions which are obtained from the second-order analog bandpass functions by the bilinear z-transformation. Taking note of the magnitude and the argument of the poles of the bilinear-transformed transfer function, we can express the center frequency, the bandwidth and the variable range of the center frequency of the bandpass filter as simple functions of these parameters. In particular, when the poles are complex conjugate, it can be shown that the bandwidth does not change with argument of the poles if the magnitude of the poles is constant. It seems that in the past the quality factor Q of digital filters has not been investigated fully. In this paper we introduce two definitions of Q from the time and frequency domains, and show that they are numerically equivalent if the ratio they are numerically equivalent if the ratio of the product of the center frequency and sampling rate to Q is less than a certain value. Finally, we discuss the relation between the pole argument and the filter dynamic range when the second-order digital bandpass function is realized in direct form. [ABSTRACT FROM AUTHOR]

This paper presents a synthesis method for the LC simulation-type active RC filter. In the proposed method, the second-order active RC filter, which has been used as the unit section in the traditional cascade structure. The mode equation of the original LC filter is set up and the simulated using the multi-input circuit for each node voltage, which is a modification of the second-order active RC circuit. The desired circuit is then obtained. As the original second-order active RC circuit is summarized in a table as the design formula. The low element-sensitivity property of the resistor-terminated LC filter is retained in the proposed circuit. In particular, the element sensitivity is low when the relative errors among R- and C-elements are small as in the thin-film hybrid circuit. Finally, a design example and experimental result are shown for the fifth-order elliptic filter, verifying that the circuit operates in accordance with the theory. [ABSTRACT FROM AUTHOR]

In the field of system modeling, circuit synthesis and filter design, it is very important to approximate a given characteristic function by a rational function of an appropriate order. Padé approximation has been known for a long time and is one of the rational function approximation methods used extensively because of its simplicity. This paper proposes to investigate N-point Padé approximation, from the system-theoretic viewpoint. The N-point Padé approximation is a rational function interpolation including the derivative of arbitrary order and, compared to regular Padé approximation and rational Hermite interpolation. In this paper we discuss the following subjects on rational function approximation of given characteristic function: its explicit form, minimal order realization problem, Nuttall's compact form, its relation to continued fractions, and the numerical efficient iterative algorithm. These are more general discussions, even including regular Padé approximation. Also, we give numerical results of the approximation of a transfer function for continuous and discrete systems as an example of application of N-point Padé approximation. [ABSTRACT FROM AUTHOR]

LC simulation by active RC circuits generally requires an equal or a higher number of active elements than the order of the filter. This is one of the main difficulties that delays their practical applications as compared to cascade realizations. This paper proposes a method to simulate impedance-scaled LCR filters using an equal or a lower number of voltage followers than the number of nodes in the original LCR filter. The influence of the imperfections of the voltage followers is also investigated. As simple examples a 4th-order bandpass filter and a 4th-order lowpass filter are realized. Measurements and sensitivity analysis are performed on the 4th-order bandpass filter and the results demonstrate the high performances of the method as compared to the other methods. [ABSTRACT FROM AUTHOR]

Most of the literatures dealing with the design of 2-D digital filters, have considered design problems for the restricted structures. This is because it is difficult to test the stability of general 2-D digital filter structures. Thus, globally optimal 2-D digital filters are difficult to design. This paper presents a necessary and sufficient condition for the asymptotic stability of 2-D spatial systems represented by the local state-space model. Also, the relationship between model parameters and stability is shown. The optimal design of 2-D digital filters generally is formulated on the basis of the stability condition. The optimal design is reduced further into an unconstrained nonlinear minimization problem. The design method for 2-D spatial domain specifications is proposed as a special case of the formulation. Numerical examples are given to illustrate the effectiveness of the method. Finally, the new optimization algorithm for the design is considered. [ABSTRACT FROM AUTHOR]

In the acoustic echo canceller, the speaker input signal is sued as a reference signal to update adaptive filter coefficients, and is sometimes composed of only the far-end background noise. This paper proposes a configuration in which the estimation error is kept constant also for a voiceless noise period, while continuously updating the coefficients. The summational normalization LMS is used as the adaptive algorithm, where the products of the residual echo and the transmitted signal (i.e., the numerators of the coefficient update resulting from the normalized least mean square (NLMS) algorithm), as well as the norm of the transmitted signal (i.e., the denominator of the forementioned), are accumulated, respectively, for a certain number, and the ratio of the two sums is used as the coefficient update. The summation number is controlled by the power of the transmitted signal. When, for example, the configuration is such that the coefficients are updated only when the sum of the norms reaches a certain value (i.e., the threshold), the echo cancellation is maintained at the desired value, independently of the power of the transmitted signal. The cancellation is enhanced in the period where the power of the transmitted signal is increased. In addition to the control method, which can calculate the power of the environmental noise needed in determining the threshold, this paper analyzes the structure of the filter. [ABSTRACT FROM AUTHOR]

This paper discusses a method for configuration of a single operational amplifier filter circuit using one operational amplifier (Op Amp) and LCR elements. The proposed single Op Amp filter circuit has a constant resistance input impedance and a zero output impedance. It allows realization of any stable transfer function. In this paper, a method for realization of an allpass function is investigated in particular and its design method is presented. Actually, a computer simulation and an experiment with a test circuit made of discrete components are carried out for a circuit to realize a fifth-order allpass function. The circuit performance is compared with that of a cascaded scheme of Deliyannis circuits. The element sensitivity in regard to the LCR elements, the frequency characteristics of the Op Amp and the sensitivity of the inductor L with respect to Q are derived. It is shown that the proposed allpass circuit has low element sensitivity and the number of necessary reactance elements is identical to the order of the transfer function. [ABSTRACT FROM AUTHOR]

Harmonic interference problems generated by bulk semiconductor power converters become more serious in power systems with their wide use in industrial applications. Shunt passive filters consisting of LC tune filters or shunt active filters using PWM inverters have been used to suppress harmonic currents in distribution systems. To install the passive or active filters, it is necessary to find exactly a dominant harmonic source in distribution feeders. A harmonic power on each phase and order at a receiving terminal has been investigated by using a digital power meter including an FFT analyzer. However, the conventional detection method has a problem in that it is difficult to discuss exactly harmonic power flow because the detected harmonic power is much smaller compared with a fundamental active power at the receiving terminal. In this paper, the harmonic power in three phases, which can be detected on real time, is defined by applying the pq theory. Next, it is verified by a digital simulation that the proposed method can discuss harmonic flows more easily than the conventional method. As a result, it is more simple and precise to find the harmonic source. Moreover, measurement errors of the harmonic power are revealed quantitatively, and the validity of the proposed method is demonstrated by digital simulation. [ABSTRACT FROM AUTHOR]

This paper proses a new PWM pattern which reduces the switching frequency of GTO elements by 30 percent, reduce the harmonic content and is better suited for current control with variable modulation factor as compared with the conventional patterns. In the proposed method, the frequency spectrum of switching function is weighed inversely proportional to the harmonic order utilizing the low pass L-C filter inserted to the power source. The performance function thus derived is used to define the evaluation function which represents the sum of squares of harmonic amplitudes. The optimal PWM pattern is determined so as to minimize the evaluation function using Park vectors. This paper also proposes an optimal filter whose resonance frequency is determined so as to reduce the distortion factor of line current to less than a specified limit, which is set at 5 percent in this paper. Constants L and C are determined so as to minimize the filter capacity. Results of experiments using microcomputer are presented to show that the proposed method yields a line current with very small distortion factor. [ABSTRACT FROM AUTHOR]