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This paper develops a tool kit for designing low-pass filters to exhibit the smallest possible phase drop. Based solely on the stopband requirements, it is thus possible to find the best order for a filter to be employed in a feedback loop. That is shown for two much-used filter families, Butterworth and Bessel, in cases where the filter is specified to have a minimum attenuation above a certain frequency. It is argued that the phase drop can be represented by an equivalent filter delay. Design tools are then developed, which do not depend on the precise dynamics of the application process. The tools comprise not only the means for determining the optimal filter order and bandwidth, but also formulae and tables useful for obtaining the resulting filter delay. A simple approximation is subsequently developed, which links the minimum obtainable delay directly to said requirements. The filter order need not be known to apply this expression, and the filter family is represented in it by no more than a single constant. This rule of thumb is finally adapted to the area of anti-aliasing filters and there briefly compared to approximative formulae found in existing literature. [ABSTRACT FROM AUTHOR]

This paper deals with the smart applications of multirate digital signal processing. The two major operations are accomplished in consumer electronics and communication engineering. The process of reducing the sampling frequency of a sampled signal is called decimation. In the usage of decimating filters, only a portion of the out-of-pass band frequencies aliases into the pass band, in systems wherein different parts operate at different sample rates. A filter design, tuned to the aliasing frequencies all of which can otherwise stealth into the pass band, not only provides multiple stop bands but also exhibits computational efficiency and performance superiority over the single stop band design. The proposed method of transmultiplexer using decimation and interpolation filters analysis procedure is not only efficient but also opens up a new vista of being simple and elegant to compute for the desired over and above transmultiplexer. [ABSTRACT FROM AUTHOR]

Mitigation of electromagnetic inference (EMI) is currently a challenge for scien-tists and designers in order to cope with electromagnetic compatibility (EMC) compliance in switching mode power supply (SMPS) and ensure the reliability of the whole system. Standard filtering techniques: passive and active ones present some insufficiency in terms of performance at high frequencies (HF) because analog components would no longer be controllable and this is mainly due to their parasitic elements. So developing EMI digital filters is very interesting, especially with the embedment of a machine control system on a field programmable gate array (FPGA) chip. In this paper, we present a design of an active digital EMI filter (ADF) to be integrated in a drive train system of an electric vehicle (EV). Hardware design as well as FPGA implementation issues have been presented to prove the efficiency of the developed digital filtering structure. [ABSTRACT FROM AUTHOR]

Power line harmonic noise generated by power lines during the seismic data acquisition in land and marine seismic surveys generally appears as a single frequency with 50/60 Hz (or multiples of these frequencies) and contaminates seismic data leading to complicate the identification of fine details in the data. Commonly applied method during seismic data processing to remove the harmonic noise is classical notch filter (or very narrow band-stop filter), however, it also attenuates all recorded data around the notch frequencies and results in a complete loss of available information which corresponds to fine details in the seismic data. In this study, we introduce an application of the algorithm of iterative trimmed and truncated mean filter method (ITTM) to remove the harmonic noise from seismic data, and here, we name the method as local ITTM (LITTM) since we applied it to the seismic data locally in spectral domain. In this method, an optimal value is iteratively searched depending on a threshold value by trimming and truncating process for the spectral amplitude samples within the specified spectral window. Therefore, the LITTM filter converges to the median, but, there is no need to sort the data as in the case of conventional median filters. On the other hand, the LITTM filtering process doesn't require any reference signal or a precise estimate of the fundamental frequency of the harmonic noise, and only approximate frequency band of the noise within the amplitude spectra is considered. The only required parameter of the method is the width of this frequency band in the spectral domain. The LITTM filter is first applied to synthetic data and then we analyze a real marine dataset to compare the quality of the output after removing the power line noise by classical notch, median and proposed LITTM filters. We observe that the power line harmonic noise is completely filtered out by LITTM filter, and unlike the conventional notch filter, without any damage on the available frequencies around the notch frequency band. It also provides a more balanced amplitude spectrum since it does not produce amplitude notches in the spectrum. [ABSTRACT FROM AUTHOR]

Combined digital filters are the basis of fault selection and identification in electric microprocessor protection. The commonly used combined digital filters based on wavelet transform cannot eliminate the interference components of traditional filter, resulting in low reliability of electric microcomputer protection. To this end, a design method of combined digital filter for electric microcomputer protection is proposed based on distributed algorithm. First, the combined digital filter and each implementation module are designed. Then, the principle of distributed algorithm is introduced, and realization of distributed algorithm is analysed, so as to realize the design of combined digital filter. The experimental results show that calculation of the filter designed by this method is simple and effective, and the development of electric microprocessor protection is promoted. [ABSTRACT FROM AUTHOR]

In this paper, a blocking-artifact detection method is proposed for motion-compensated frame-rate up-conversion (MC-FRUC) algorithms. In the MC-FRUC process, most of the notable blocking artifacts occur in unreliable motion-compensated regions. To find the unreliable motion-compensated regions, an occlusion detection (OD) method for which the neighboring motion vectors are used is applied. If an occlusion occurs, then the use of median filtering reduces any inappropriate motion-compensated errors; otherwise, the normal motion compensated interpolation (MCI) is applied. In addition, to reduce the occurrence of the blocking artifacts in the MC-FRUC algorithms, a blocking-artifact-reduction method for which the blocking-artifact strength in the frequency domain is applied to the occluded regions is utilized. The experiment results show that the proposed method shows more-favorable peak signal-to-noise ratio (PSNR) values compared with those of the conventional FRUC algorithms regarding full-HD (1920 × 1080) and WQXGA (2560 × 1600) sequences. [ABSTRACT FROM AUTHOR]

Satellite's attitude can be determined by using several different ways. In order to use a recursive filter to determine the attitude especially after crucial intervals for the satellite including detumbling, eclipse, etc., single-frame methods can be used, and singular value decomposition (SVD) method has been selected because of its robustness for this paper. They can be placed for the initial values, or continuous input to the filter in addition to covariance matrix at each step. Also, sensor configurations depending on the position of the Sun are considered and used in the SVD method. Wahba's loss function is minimized with separating the basic matrix including the reference and model vectors into singular values because many researches show the SVD as the most robust algorithm after error analyses between several single-frame methods. Sun sensor, magnetometer, and horizon sensor are the selected common attitude sensors for small satellites in order to compare their attitude determination performances. For this purpose, absolute errors for different sensor configurations in the simulation environment in addition to the RMS errors are presented. The purpose of the paper is to present an attitude determination concept for a nanosatellite. Also, different sensor configurations are considered in the single-frame method with switching logic depending on the vector data. Horizon sensor and sun sensor are the optimum pair for the light side of the Earth on the other hand, during the eclipse; horizon sensor and magnetometer give the most accurate results. Study shows that the proposed algorithm can be performed for initial values to the Kalman filter, and in the transient phase before the filter tuned in practical satellite usage. In the eclipse period, sun-independent sensors are used and changing configurations provide better attitude estimation results to filtering techniques. [ABSTRACT FROM AUTHOR]

This paper proposes an efficient implementation of the generalized labeled multi-Bernoulli (GLMB) filter by combining the prediction and update into a single step. In contrast to an earlier implementation that involves separate truncations in the prediction and update steps, the proposed implementation requires only one truncation procedure for each iteration. Furthermore, we propose an efficient algorithm for truncating the GLMB filtering density based on Gibbs sampling. The resulting implementation has a linear complexity in the number of measurements and quadratic in the number of hypothesized objects. [ABSTRACT FROM PUBLISHER]

This paper describes a novel image filter with superior performance on detail-preserving removal of random-valued impulse noise superimposed on natural gray-scale images. The non-local means filter is in the limelight as a way of Gaussian noise removal with superior performance on detail preservation. By referring the fundamental concept of the non-local means, we had proposed a non-local median filter as a specialized way for random-valued impulse noise removal so far. In the non-local processing, the output of a filter is calculated from pixels in blocks which are similar to the block centered at a pixel of interest. As a result, aggressive noise removal is conducted without destroying the detailed structures in an original image. However, the performance of non-local processing decreases enormously in the case of high noise occurrence probability. A cause of this problem is that the superimposed noise disturbs accurate calculation of the similarity between the blocks. To cope with this problem, we propose an improved non-local median filter which is robust to the high level of corruption by introducing a new similarity measure considering possibility of being the original signal. The effectiveness and validity of the proposed method are verified in a series of experiments using natural gray-scale images. [ABSTRACT FROM AUTHOR]

In this paper, a coarse alignment method based on apparent gravitational motion is proposed. Due to the interference of the complex situations, the true observation vectors, which are calculated by the apparent gravity, are contaminated. The sources of the interference are analyzed in detail, and then a low-pass digital filter is designed in this paper for eliminating the high-frequency noise of the measurement observation vectors. To extract the effective observation vectors from the inertial sensors' outputs, a parameter recognition and vector reconstruction method are designed, where an adaptive Kalman filter is employed to estimate the unknown parameters. Furthermore, a robust filter, which is based on Huber's M-estimation theory, is developed for addressing the outliers of the measurement observation vectors due to the maneuver of the vehicle. A comprehensive experiment, which contains a simulation test and physical test, is designed to verify the performance of the proposed method, and the results show that the proposed method is equivalent to the popular apparent velocity method in swaying mode, but it is superior to the current methods while in moving mode when the strapdown inertial navigation system (SINS) is under entirely self-contained conditions. [ABSTRACT FROM AUTHOR]

Land cover classification using very high spatial resolution (VHSR) imaging plays a very important role in remote sensing applications. However, image noise usually reduces the classification accuracy of VHSR images. Image spatial filters have been recently adopted to improve VHSR image land cover classification. In this study, a new object-based image filter using topology and feature constraints is proposed, where an object is considered as a central object and has irregular shapes and various numbers of neighbors depending on the nature of the surroundings. First, multi-scale segmentation is used to generate a homogeneous image object and extract the corresponding vectors. Then, topology and feature constraints are proposed to select the adjacent objects, which present similar materials to the central object. Third, the feature of the central object is smoothed by the average of the selected objects' feature. This proposed approach is validated on three VHSR images, ranging from a fixed-wing aerial image to UAV images. The performance of the proposed approach is compared to a standard object-based approach (OO), object correlative index (OCI) spatial feature based method, a recursive filter (RF), and a rolling guided filter (RGF), and has shown a 6%-18% improvement in overall accuracy. [ABSTRACT FROM AUTHOR]

In this paper, a modified GrabCut algorithm is proposed using a clustering technique to reduce image noise. GrabCut is an image segmentation method based on GraphCut starting with a user-specified bounding box around the object to be segmented. In the modified version, the original image is filtered using the median filter to reduce noise and then the quantized image using K-means algorithm is used for the normal GrabCut method for object segmentation. This new process showed that it improved the object segmentation performance a lot and the extract segmentation result compared to the standard method. [ABSTRACT FROM AUTHOR]

This paper presents a new method for designing digital recursive integrators and differentiators by using nonlinear stochastic global optimization based on particle swarm optimization (PSO). A modified PSO is used to optimize the unknown coefficients of second-order and third-order recursive integrators in order to obtain a better magnitude response closer to the ideal integrator. The coefficients of the state-of-the-art available filters in the literature are chosen as initial points in the PSO process. By choosing a good starting point, convergence to an optimal solution is greatly facilitated. A dynamic modification of the fitness function used in the PSO process leads to the design of a set of digital integrators each optimized for a specific frequency range. Then, second-order and third-order recursive digital differentiators are designed by inverting and stabilizing the transfer functions of the designed recursive integrators. The obtained stabilized differentiators are further optimized using PSO to further improve their performance. The magnitude responses of the designed filters outperform the existing integrators and differentiators. Copyright © 2015 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

SUMMARY In this paper, we propose an L1-approximation for the design of FIR digital filters with complex coefficients. The L1-approximation has the advantage of having a flatter passband and a small overshoot around the discontinuity point as compared with the L2-approximation and [ABSTRACT FROM AUTHOR]

Recommender systems (RSs) use information filtering to recommend information of interest (to a user). Similarly, personalisation can be adopted for recommendations in e-market. We propose a new and innovative system called as interest-based recommender system (IBRS) for personalisation of recommendations. The IBRS is an agent-based RS that takes into account user's preferences. It can transform a standard product (or service) into a dedicated solution for an individual. The system discovers interesting product alternatives based on user's underlying mental attitudes (likes and dislikes) during the repair process using argumentation. The proposed method combines a hybrid RS approach with automated argumentation-based reasoning between agents. The system improves results by improving the recommendation repair activity. We consider a book recommendation application, for experiment to carry out the system's (objective and subjective) evaluation using standard metrics. The experiments confirm that the proposed IBRS improves user's acceptance of the product as compared with a traditional hybrid method and an argumentation-enabled state-of-the-art recommendation method. The system has been found to be more effective than its traditional counterpart when dealing with the new user problems. [ABSTRACT FROM AUTHOR]

The paper presents a concept of an innovative interaction structure for a digital electric vehicle (EV) dashboard. The structure connects interactions between vehicle, driver and traffic infrastructure, in order to help users driving in a conscious way, informing them about their performances and providing tools able to modify driving behaviour. Through the Systemic Design approach, it is possible to move from a quantitative configuration (set on consumption) to a new one set on resource optimization. The achievement is a new layout for the information visualization system designed for an electric vehicle able to communicate to the driver the environmental impact of its drive style. [ABSTRACT FROM AUTHOR]

We analyze a two-stage echo cancellation technique for on-channel repeaters (OCRs) based on the reconstruction of the coupling channel impulse response via digital filtering. The estimation of the coupling channel between the transmitting and the receiving antennas is achieved through the injection in the repeated signal of locally generated low-power pseudo-noise sequences or trains of short pulses. The analytical model developed in this paper permits the investigation of some important performance figures, such as the mean rejection ratio, the echo suppression and the echo suppression at nominal position. In addition, a new analytical framework dealing with canceller stability issues, arising due to the non perfect echo cancellation, is derived. Specifically, an upper bound on the probability of instability of the canceller is provided. As shown in the numerical results, this framework permits the identification of the main OCR design guidelines. Finally, we will validate our analytical models by showing their accordance with the experimental measurements realized with the OCR prototypes we developed. [ABSTRACT FROM AUTHOR]

This paper focuses on the design problem of a recursive networked strong tracking filter (NSTF) for a class of nonlinear networked systems with parameter perturbations and unknown inputs. The sensors for the networked system are allowed to be spatially distributed in a large geographical area, and signals are transmitted via a shared communication channel with limited capacity. For this kind of system structure, the measurements from different sensors may experience probabilistic data loss with different probabilities. A series of Bernoulli sequences is employed to describe the multiple packet dropout rates. Parameter perturbations and unknown inputs in the system are considered in the filter design process. A recursive networked extended Kalman filter is first derived in the least mean square sense by taking the packet dropout phenomenon into account. Then, a fading factor is introduced in the filter structure in order to cope with the parameter perturbations and unknown system inputs, and a recursive NSTF is derived by developing the so-called networked orthogonal principle. It is shown that the proposed NSTF is capable of providing satisfactory estimation results even in the presence of system parameter perturbations and/or unknown system inputs. A simulation study is carried out on a practical Internet-based three-tank system, and the estimation results show the effectiveness and applicability of the proposed filtering techniques. [ABSTRACT FROM PUBLISHER]

The most commonly used intensity measure in ground-motion prediction equations is the pseudo-absolute response spectral acceleration (PSA), for response periods from 0.01 to 10 s (or frequencies from 0.1 to 100 Hz). PSAs are often derived from recorded ground motions, and these motions are usually filtered to remove high and low frequencies before the PSAs are computed. In this article we are only concerned with the removal of high frequencies. In modern digital recordings, this filtering corresponds at least to an anti-aliasing filter applied before conversion to digital values. Additional high-cut filtering is sometimes applied both to digital and to analog records to reduce high-frequency noise. Potential errors on the short-period (high-frequency) response spectral values are expected if the true ground motion has significant energy at frequencies above that of the anti-aliasing filter. This is especially important for areas where the instrumental sample rate and the associated anti-aliasing filter corner frequency (above which significant energy in the time series is removed) are low relative to the frequencies contained in the true ground motions. A ground-motion simulation study was conducted to investigate these effects and to develop guidance for defining the usable bandwidth for high-frequency PSA. The primary conclusion is that if the ratio of the maximum Fourier acceleration spectrum (FAS) to the FAS at a frequency $$f_{saa} $$ corresponding to the start of the anti-aliasing filter is more than about 10, then PSA for frequencies above $$f_{saa} $$ should be little affected by the recording process, because the ground-motion frequencies that control the response spectra will be less than $$f_{saa} $$ . A second topic of this article concerns the resampling of the digital acceleration time series to a higher sample rate often used in the computation of short-period PSA. We confirm previous findings that sinc-function interpolation is preferred to the standard practice of using linear time interpolation for the resampling. [ABSTRACT FROM AUTHOR]

This paper has developed two techniques to reduce the parasitic parameters and improve the high-frequency performance of the planar integrated electromagnetic interference (EMI) filter. First, each developed technique has been analyzed, respectively, then the performance of an EMI filter with both the developed techniques has been researched. The research contents are as follows. 1) Analyzing the improvement of the equivalent parallel capacitance (EPC) for a common mode (CM) EMI filter by two CM windings overlapped and interleaved layout. The EPC of the integrated EMI filter is about 14% of the reference EMI filter. 2) Analyzing the improvement of the equivalent series inductance (ESL) in the parallel branch for a CM EMI filter by each CM winding coupled with two ground winding layers. The resonance frequency brought by ESL is shifted to a higher frequency. At 30 MHz, the insertion loss is increased by 45 dB compared with the reference. In the mean time, the structure is equivalent to a two-stage L-type low-pass filter, which can largely increase the high-frequency attenuation compared with the one-stage L-type low-pass filter for the reference one. (3) Researching the performance of an EMI filter with both the proposed techniques. The developed EMI filter can not only decrease the EPC, but also reduce the negative effect of ESL. The developed EMI filter improves the high-frequency performance dramatically compared with the EMI filter with only one of the developed techniques. In the mean time, the differential mode (DM) capacitor is formed by the two overlapped and interleaved windings, and does not need an extra lumped DM capacitor. Therefore, CM inductors, CM capacitors, and DM capacitors can be integrated in the overlapped and interleaved L–C winding structure, so miniaturized EMI filter can be realized. The experiment results indicate effectiveness of the developed integrated EMI filters, and the cost and complexity is the same as the reference EMI filter. [ABSTRACT FROM PUBLISHER]

The objective of this article is to present an accurate model for behavioral modeling and digital predistortion of power amplifiers (PAs) using physical knowledge. Starting with the physically meaningful block model, we present a novel approach to simplify the block model and propose a simplified physical knowledge-based (SPK) model. The SPK model's performance was experimentally assessed by two types of PAs (a LDMOS Doherty PA and a GaN Doherty PA) and two signals (a single carrier 16QAM signal and a 2-carrier WCDMA signal). All experimental results prove the superiority of the SPK model. Compared with the 1st-dynamic deviation reduction (DDR) approach and the 2nd-simplified DDR approach, the SPK approach achieves average ACPR improvements of 4.4 dB and 2.5 dB, respectively. © 2012 Wiley Periodicals, Inc. Int J RF and Microwave CAE 23: 682-689, 2013. [ABSTRACT FROM AUTHOR]

When the three-dimensional (3-D) video system includes a multiview video generation technique using depth data to provide more realistic 3-D viewing experiences, accurate depth map acquisition is an important task. In order to generate the precise depth map in real time, we can build a camera fusion system with multiple color cameras and one time-of-flight (TOF) camera; however, this method is associated with depth errors, such as depth flickering, empty holes in the warped depth map, and mixed pixels around object boundaries. In this paper, we propose three different methods for depth error reduction to minimize such depth errors. In order to reduce depth flickering in the temporal domain, we propose a temporal enhancement method using a modified joint bilateral filtering at the TOF camera side. Then, we fill the empty holes in the warped depth map by selecting a virtual depth and applying a weighted depth filtering method. After hole filling, we remove mixed pixels and replace them with new depth values using an adaptive joint multilateral filter. Experimental results show that the proposed method reduces depth errors significantly in near real time. [ABSTRACT FROM AUTHOR]

Understanding of the runoff generation processes is important in understanding the magnitude and dynamics of groundwater discharge. However, these processes continue to be difficult to quantify and conceptualize. In this study, two digital filter based separation modules, the Recursive filtering method (RDF) and a generalization of the recursive digital filter (GRDF) were 1991-2002 in the Hableh Roud River at the stream gauge of Bonkuh, Semnan province. A technique for assessing the recession constants of the sub flows based on calibration the average value of the inverse of the value of the inverse of the slope of the linear path in the recession periods of an Ln (q)-time graph is presented. The result show that, the GRDF method gave higher (Baseflow index) BFI values than the RDF method with less variability and the mean baseflow calculated on an annual basis, ranged from 3.27 m³/s to 4.04 m³/s over period of study by RDF and GRDF, respectively. Since the true values of the baseflow index are unknown, it cannot be said, which one of the methods gives the best estimates. [ABSTRACT FROM AUTHOR]

Multipath is one of the prominent error sources in Global Positioning System (GPS), which adversely affects the accuracy of the user's position. This paper describes the estimation of frequency components of multipath error signal using spectral analysis and its effective mitigation using time-varying digital filters. Four types of 2nd order filters, namely, Butterworth, Type I and II Chebyshev and Elliptic filters, are examined for mitigation of multipath and their performance is compared. It is observed that by applying digital filters of different cut-off frequencies over the spectrum of the multipath, one can significantly reduce the multipath errors. The designed filters are applied to GPS data acquired from a typical urban environment site and found that Butterworth filter reduced the error most effectively. To extract the magnitude of error contribution in low and medium multipath frequency ranges, its mitigation is carried out with Butterworth filter. The performance of the filter is validated by the spectral analysis for multipath signals before and after filtering process. However, this spectral analysis followed by digital filtering is different from the approaches reported elsewhere in the open literature. This new approach is very useful in identifying least-affecting multipath sites for GPS applications. [ABSTRACT FROM AUTHOR]

A kind of scale invariant features transformation (SIFT for short) operators on DWT domain are proposed for watermarking algorithm. Firstly, the low frequency of the image is obtained by DWT. And then the SIFT transformation is used to calculate the key feature points for the low frequency sub-image. Based on the chosen space's key points with moderate scale, a circular area as watermark embedding area is constructed. According to the research and final results, the novel digital watermark algorithm is proposed benefiting from the characteristics of SIFT's key points and local time-frequency of DWT. The algorithm not only has good robustness to resist on such operations as compression, shearing, noise addition, median filtering and scaling, but also has good inhibition to possible watermark fake verification. [ABSTRACT FROM AUTHOR]

Iterative image reconstruction can dramatically improve the image quality in X-ray computed tomography (CT), but the computation involves iterative steps of 3D forward- and back-projection, which impedes routine clinical use. To accelerate forward-projection, we analyze the CT geometry to identify the intrinsic parallelism and data access sequence for a highly parallel hardware architecture. To improve the efficiency of this architecture, we propose a water-filling buffer to remove pipeline stalls, and an out-of-order sectored processing to reduce the off-chip memory access by up to three orders of magnitude. We make a floating-point to fixed-point conversion based on numerical simulations and demonstrate comparable image quality at a much lower implementation cost. As a proof of concept, a 5-stage fully pipelined, 55-way parallel separable-footprint forward-projector is prototyped on a Xilinx Virtex-5 FPGA for a throughput of 925.8 million voxel projections/s at 200 MHz clock frequency, 4.6 times higher than an optimized 16-threaded program running on an 8-core 2.8-GHz CPU. A similar architecture can be applied to back-projection for a complete iterative image reconstruction system. The proposed algorithm and architecture can also be applied to hardware platforms such as graphics processing unit and digital signal processor to achieve significant accelerations. [ABSTRACT FROM PUBLISHER]

The Coriolis mass flow meter, for its features of directly measuring mass flow and high precision, is widely utilized in the flow measuring fields like batch filling and commercial trade. However, the unsatisfactory dynamic performance is one major constraint to Coriolis mass flow meter from being even more widely used in batch filling industry. To solve the problem, an experimental system which generates refined flow step stimulus signals for Coriolis mass flow meter was designed and based on which experimental and theoretical study on dynamic performances of typical Coriolis mass flow meter was carried out. On top of this, a time-domain recursive digital filter was designed to provide dynamic compensation and the experiments showed this filter can reduce the step response time of Coriolis mass flow meter by about 50%. [ABSTRACT FROM AUTHOR]

The sideways heat equation concerns an inverse problem where an unknown and inaccessible surface temperature is estimated using sensor data gathered from an interior point of a slab. The article presents how the ill-posed problem is solved using a finite-difference technique, where the space variable is continuous and the time variable is discrete. The inverse procedure is based on a recursive deconvolution algorithm, derived from the direct solution. Regularization is completed with digital filters. The test runs give promising results. However, due to the time derivative approximation and disturbances, abrupt changes in the estimated signal cannot be perfectly reproduced. [ABSTRACT FROM AUTHOR]

This paper presents a novel method to design the input filter for a direct ac/ac matrix converter using genetic algorithms (GA) optimization. The input filter for a matrix converter is a very important and critical part of the conversion structure and careful design is necessary to ensure high input power quality, compactness, and stability. The GA will optimize structure and parameters of the input filter as a function of different factors such as energy storage, weight, and volume. The effectiveness of this design method is demonstrated through a wide range of simulations using Saber and experimental results on a laboratory prototype. The same methodology could also be adapted and applied to any converter configuration such as, for example, traditional voltage source converters. [ABSTRACT FROM PUBLISHER]

In this paper, we improve iterative greedy search algorithms in which atoms are selected serially over iterations, i.e., one-by-one over iterations. For serial atom selection, we devise two new schemes to select an atom from a set of potential atoms in each iteration. The two new schemes lead to two new algorithms. For both the algorithms, in each iteration, the set of potential atoms is found using a standard matched filter. In case of the first scheme, we propose an orthogonal projection strategy that selects an atom from the set of potential atoms. Then, for the second scheme, we propose a look-ahead strategy such that the selection of an atom in the current iteration has an effect on the future iterations. The use of look-ahead strategy requires a higher computational resource. To achieve a tradeoff between performance and complexity, we use the two new schemes in cascade and develop a third new algorithm. Through experimental evaluations, we compare the proposed algorithms with existing greedy search and convex relaxation algorithms. [ABSTRACT FROM PUBLISHER]

Many signal processing problems are tackled by filtering the signal for subsequent feature classification or regression. Both steps are critical and need to be designed carefully to deal with the particular statistical characteristics of both signal and noise. Optimal design of the filter and the classifier are typically aborded in a separated way, thus leading to suboptimal classification schemes. This paper proposes an efficient methodology to learn an optimal signal filter and a support vector machine (SVM) classifier jointly. In particular, we derive algorithms to solve the optimization problem, prove its theoretical convergence, and discuss different filter regularizers for automated scaling and selection of the feature channels. The latter gives rise to different formulations with the appealing properties of sparseness and noise-robustness. We illustrate the performance of the method in several problems. First, linear and nonlinear toy classification examples, under the presence of both Gaussian and convolutional noise, show the robustness of the proposed methods. The approach is then evaluated on two challenging real life datasets: BCI time series classification and multispectral image segmentation. In all the examples, large margin filtering shows competitive classification performances while offering the advantage of interpretability of the filtered channels retrieved. [ABSTRACT FROM PUBLISHER]

In this article, the quasi-TEM approach is successfully used to analyze coupled-microstrip lines of finite length, which are usually used for the coupling between the source/load and resonators in microstrip filters. First, an orthogonal expansion method is used to extract the per-length inductance and capacitance matrices. The authors then obtain the impedance matrix or admittance matrix, which relates the voltages and currents at both ends of coupled-microstrip lines. To verify our theory, an example is presented, i.e., a Y-shaped dual-mode bandpass microstrip filter, for which, the equivalent circuit model based on the quasi-TEM analysis of coupled-microstrip lines is accurately set up. The approach in this article is applicable to planar microstrip lines with arbitrary number of conductors, under the quasi-TEM assumption. © 2011 Wiley Periodicals, Inc. Int J RF and Microwave CAE, 2012. [ABSTRACT FROM AUTHOR]

The paper proposes the use of a DSP family ADMC330 several ways to control an active filter. It is made available as filter simulation in Matlab-Simulink environment, and the results confirm the effectiveness of the filter. This paper presents four schemes using both types of construction, with and without feedback loop, for three-phase voltage sources but balanced nonlinear load (no neutral wire). [ABSTRACT FROM AUTHOR]

Abstract: In this paper, we propose a new discrete-time model which relates uniform samples and K-th order nonuniform (KONU) samples. Typically, such situations may occur in practice where the sampling rates of analog-to-digital converters (ADCs) in a cluster of sensors may be different. This paper also outlines a digital signal processing approach to reconstructing uniform samples from the nonuniform samples produced by time interleaving analog-to-digital converters (TI-ADCs) with different sampling rates and time-mismatches. [Copyright &y& Elsevier]

This paper presents a three-pole tunable combline bandpass filter with center frequency, bandwidth, and zero control. The filter is designed on a Duroid substrate with \varepsilonr=10.2 and h=25\ \mil. A frequency range of 1.5–2.2 GHz with a 1-dB bandwidth tuning from 50 to 170 MHz (2.2%–11.2% fractional bandwidth) is achieved. The transmission zero can also be controlled, and a zero location of 1.37–1.64 GHz is demonstrated at center frequency fo of 2.05 GHz. The measured third-order intermodulation intercept point and 1-dB power compression point at midband (1.85 GHz) and a bandwidth of 110 MHz are >15 and 8 dBm, respectively. To our knowledge, this is the first three-pole combline tunable bandpass filter with center frequency, bandwidth, and transmission zero control. [ABSTRACT FROM AUTHOR]

In this paper, we propose and develop a partially shielded tunable filter structure using a varactor-loaded split-ring resonator. The novel physical topology of the filter is suitable for vertical stacking of filters and enables a compact multilayer tunable filter for applications requiring a very wide tuning range. The capacitance range and ratio needed to achieve the designated tuning range are presented, and analytical solutions are given for the second and third higher order modes for synthesis in the spurious-free filter operation range. Two prototypes are developed. A partially shielded tunable filter using varactor-loaded split-ring resonators is developed to verify the design method and to demonstrate the wide tuning range of a tunable filter. To further extend the tuning range, a filter bank with three stacked tunable filters at different frequency bands are designed and developed. With slight overlap between neighboring bands, a continuous tuning range of more than 6:1 is achieved. [ABSTRACT FROM AUTHOR]

Abstract: Real-time digital pulse shaping techniques allow synthesis of pulse shapes that have been difficult to realize using the traditional analog methods. Using real-time digital shapers, triangular/trapezoidal filters can be synthesized in real time. These filters exhibit digital control on the rise time, fall time, and flat-top of the trapezoidal shape. Thus, the trapezoidal shape can be adjusted for optimum performance at different distributions of the series and parallel noise. The trapezoidal weighting function (WF) represents the optimum time-limited pulse shape when only parallel and series noises are present in the detector system. In the presence of 1/F noise, the optimum WF changes depending on the 1/F noise contribution. In this paper, we report on the results of the evaluation of new filter types for processing signals from CANBERRA high purity germanium (HPGe) and passivated, implanted, planar silicon (PIPS) detectors. The objective of the evaluation is to determine improvements in performance over the current trapezoidal (digital) filter. The evaluation is performed using a customized CANBERRA digital signal processing unit that is fitted with new FPGA designs and any required firmware modifications to support operation of the new filters. The evaluated filters include the Cusp, one-over-F (1/F), and pseudo-Gaussian filters. The results are compared with the CANBERRA trapezoidal shaper. [Copyright &y& Elsevier]

In this work, new trace configurations using stepped impedance elements and open-circuited stubs to reduce far-end crosstalk in printed circuit board (PCB) interconnects are introduced. The goal of this study is to reduce crosstalk without using additional PCB components in the design. Specifically, we use stepped impedance elements and open-circuited stubs of uniform length and width on the victim traces to suppress high-frequency electromagnetic interference and to equalize propagation delays or capacitive and inductive couplings between PCB traces. The overall design is very similar to the usual low-pass filter configurations, which are difficult to implement in the prototype testing. The proposed approach shows remarkably better results when compared with conventional intervening guard trace schemes. © 2011 Wiley Periodicals, Inc. Int J RF and Microwave CAE, 2011. [ABSTRACT FROM AUTHOR]

Abstract: Digital cameras use a single electronic sensor overlaid with a color filter array (CFA) to capture images for reducing the production cost. Only one primary color is sampled in each pixel such that the missing color primaries must be reconstructed by interpolation. In this paper, a hybrid demosaicking scheme for CFA interpolation is proposed to increase the visual quality of the restored image by artifact suppression in the edge region. A preliminary interpolation is first implemented by bilinear interpolation. The color difference model is then used to update green channel. The red and blue channels are successively updated using the proposed hue model that can reduce the distortion around edges. Finally, an iteratively adaptive algorithm using an edge-sensing interpolation mechanism and a luminance-color difference model is exploited to improve visual quality of the restored image by artifact suppression in the edge region. Simulation results demonstrate that the performance of the proposed algorithm is better than that of the existing approaches in terms of peak signal-to-noise ratio (PSNR) and visual quality. [Copyright &y& Elsevier]

Abstract: A novel impulsive noise elimination filter combining a double noise detector with an adaptive neural-fuzzy inference system (ANFIS) is proposed. The proposed filter has a two-phase scheme for removing impulse noise: detecting noisy pixels and removing noise. In the detecting noisy pixels phase, noise candidates identified with the noise detection algorithm of the adaptive median filter are judged again by local fuzzy membership function to improve the accurate rate of noise detection. In the removing noise phase, the corrupted pixels are restored using adaptive neural-fuzzy inference system based on their neighbor uncorrupted pixels. Experimental results show that the proposed filter yields a better performance than the recent filters in terms of details preservation and noise suppression, and our scheme can remove impulsive noise with noise level as high as 90%. [Copyright &y& Elsevier]

A new dynamic harmonic estimator is presented as an extension of the fast Fourier transform (FFT), which assumes a fluctuating complex envelope at each harmonic. This estimator is able to estimate harmonics that are time varying inside the observation window. The extension receives the name “Taylor–Fourier transform (TFT)” since it is based on the McLaurin series expansion of each complex envelope. Better estimates of the dynamic harmonics are obtained due to the fact that the Fourier subspace is contained in the subspace generated by the Taylor–Fourier basis. The coefficients of the TFT have a physical meaning: they represent instantaneous samples of the first derivatives of the complex envelope, with all of them calculated at once through a linear transform. The Taylor–Fourier estimator can be seen as a bank of maximally flat finite-impulse-response filters, with the frequency response of ideal differentiators about each harmonic frequency. In addition to cleaner harmonic phasor estimates under dynamic conditions, among the new estimates are the instantaneous frequency and first derivatives of each harmonic. Two examples are presented to evaluate the performance of the proposed estimator. [ABSTRACT FROM AUTHOR]

Abstract: This paper seeks to elucidate the design and implementation of an instrumentation amplifier, filters, LabVIEW-based spike detection, and automatic spike counting to detect pleasure sensation in the rat using invasive BCI. This method involved sites related to pleasure, and after acquiring signals from the ventral pallidum, facial motor cortex, and orbitofrontal cortex, these signals were analyzed to assess the pleasure sensation in the rat. The results illustrated a decreased spike rate in the motor and orbitofrontal cortices and an increased spike rate in the ventral pallidum during pleasure. The pleasure detection experiment was conducted four times to obtain the mean values of spike rates. The motor cortex had 9spikes/s, the orbitofrontal cortex had 18spikes/s and the ventral pallidum had 34spikes/s. The correlation coefficient is above 78%, effectiveness of the experiment. [ABSTRACT FROM AUTHOR]

Abstract: In this paper, a hybrid filter topology is proposed to reduce torque pulsation, switching voltage harmonics and EMI noises in PMSM with direct torque hysteresis controllers. The filter topology consists of IGBT-based active filter (AF), LC filter in the main inverter output and trap filter in the motor side. The coupling of the AF-compensated voltage and the motor main voltage is achieved through series transformer. The AF is characterized by detecting the harmonics in the motor phase voltages and uses hysteresis voltage control method to provide almost sinusoidal voltage to the motor windings. The active filter uses hysteresis voltage controller while the motor main circuit uses hysteresis direct torque control. The simulation results of this combined control structure show considerable torque ripple or pulsation reduction in steady state range and adequate dynamic torque performance as well as considerable harmonics and EMI noise reduction. [ABSTRACT FROM AUTHOR]

The development of multiple communication standards and services has created the need for a flexible and efficient computational platform for baseband signal processing. Using a set of heterogeneous reconfigurable execution units (RCEUS) and a homogeneous control mechanism, the proposed reconfigurable architecture achieves a large computational capability while still providing a high degree of flexibility. Software tools and a library of commonly used algorithms are also proposed in this paper to provide a convenient framework for hardware generation and algorithm mapping. In this way, the architecture can be specified in a high-level language and it also provides increased hardware resource usage. Finally, we evaluate the system's performance on representative algorithms, specifically a 32-tap finite impulse response (FIR) filter and a 256-point fast Fourier transform (FFT), and compare them with commercial digital signal processor (DSP) chips as well as with other reconfigurable and multi-core architectures. [ABSTRACT FROM AUTHOR]