Digital Filter Design

An approximate method for rapid multiplication or divi- sion with relatively simple digital circuitry is described. The algo- rithm consists of computing approximate binary logarithms, adding or subtracting the logarithms, and computing the approximate anti- logarithm of the resultant. Using a criteria of minimum mean square error, coefficients for the approximations are developed. An error analysis is given for three cases in which the algorithm is useful. Finally, applications to digital filtering computations are considered which illustrate that log-antilog multiplication is not simpler than an array multiplier for computing single products, but is useful for parallel digital filter banks and multiplicative digital filters.

In this search, an important methodology has been presented for communicated information rectification utilizing advanced channel windowing approach. The modern data communication technologies are ensured with numerous challenges because of their unpredictability and arrangement. Various digital transmission topologies in 4G can't fulfill the requirements in future arrangements, therefore, alternative multicarrier modulation (MCM) becoming the nominated approaches among all other data transmission techniques. Wherein prototype filter configuration is a fundamental system based on which the synthesis and analysis filters are derived. This paper presents a complete review on the ongoing advances of finite impulse response (FIR) filter plan procedures in MCM based correspondence frameworks. Initially, the essential issues are tried, taking into consideration the presentation of available data signal applicants and the FIR filter design concept. At that point the techniques for FIR filter configuration are summed up in subtleties and are center around the accompanying three group’s recurrence testing strategies, windowing based strategies and advancement-based techniques. At last, the exhibitions of different FIR structure strategies are assessed and measured by power spectral density (PSD) and bit error rate (BER), and variable MCM plots as well as their potential prototype filters are examined.

The resolution and the linearity of time measurements made with the CMS electromagnetic calorimeter are studied with samples of data from test beam electrons, cosmic rays, and beam-produced muons. The resulting time resolution measured by lead tungstate crystals is better than 100 ps for energy deposits larger than 10 GeV. Crystal-to-crystal synchronization with a precision of 500 ps is performed using muons produced with the first LHC beams in 2008.

We report the discovery of a new phenomenon in observational astronomy: high-frequency oscillations (HFO) in the brightness of stars and planets. They were detected during an international campaign of synchronized high-temporal-resolution observations of variable stars. The observations were carried out at two observatories on four telescopes using high-speed photometers. The detected HFOs are non-stationary modulated waves with a discrete spectrum ranging from fractions to tens of Hz and, probably, significantly higher. Their amplitudes range from thousandths to hundredths of a magnitude. It was noted that variations in stellar brightness are accompanied by shifts in the positions of stars (SIM) in the focal plane of telescopes. The amplitudes of SIM motion can be estimated at between a few hundredths and a few tenths of an arc second, and their speed at up to a few hundredths of an arc second per second. The SIM light curves show the elliptical orbits of the reference star. A promising hypothesis is that SIM phenomena are caused by gravitational waves. It follows that the angle of incidence of a light beam fluctuates at the frequency of the gravitational waves and with an amplitude proportional to the amplitude of the gravitational waves. Observations of HFO and SIM phenomena confirm the existence of background gravitational waves.

An accurate phasor estimation is an important task for a correct operation of control and protection devices. The presence of aperiodic components in electrical signals is a major problem for the correct performance of digital filters. In addition, their stability is heavily conditioned by the distortion resulting from the noise inherent to signal processing. Consequently, a complete study of the behaviour of any filtering algorithm (in the presence of noise and aperiodic components) must be carried out before its implementation. Usually, numerical performance indices are used to measure the accuracy of a proposed method. In order to ease the analysis and comparison of different digital filters, a new set of graphic tools is presented in this paper. To illustrate the application of these graphic tools, a case study is presented with three different filtering algorithms developed for protection relaying.

The amplitude of the signal collected from the PbWO(4) crystals of the CMS electromagnetic calorimeter is reconstructed by a digital filtering technique. The amplitude reconstruction has been studied with test beam data recorded from a fully equipped barrel supermodule. Issues specific to data taken in the test beam are investigated, and the implementation of the method for CMS data taking is discussed

ccelerometers have been used for field measurements of wind-induced response of buildings. However, wind-induced response consists of a static component, i.e. a mean value, and a dynamic fluctuating component. The static component is difficult to measure by accelerometers. An RTK-GPS (Leica MC1000) has a nominal accuracy of +/-1 cm +1 ppm for horizontal displacements and +/-2cm +2ppm for vertical displacements with a sampling rate of 10 Hz. This study aims to demonstrate the feasibility of RTK-GPS for wind-induced response measurements and its efficiency in measuring the displacement of a full-scale tower. As the first experiment, the accuracy of Real-Time Kinematic-Global Positioning System (RTK-GPS) in measuring sinusoidal displacements was examined, using an electronic exciter. When the vibration frequency was lower than 2Hz and the vibration amplitude was larger than 2 cm, RTK-GPS results seemed to closely follow the actual displacement. The efficiency of RTK-GPS was then demons...

Until April 2007 the MAGIC telescope used a 300 MSamples/s FADC system to sample the shaped PMT signals produced by the captured Cherenkov photons of air showers. Different algorithms to reconstruct the signal from the read-out samples (extractors) have been implemented and are described and compared. Criteria based on the obtained charge and time resolution/bias are defined and used to judge the different extractors, by applying them to calibration, cosmic and pedestal signals. The achievable charge and time resolution have been derived as functions of the incident number of photo-electrons.

The amplitude of the signal collected from the PbWO 4 crystals of the CMS electromagnetic calorimeter is reconstructed by a digital filtering technique. The amplitude reconstruction has been studied with test beam data recorded from a fully equipped barrel supermodule. Issues specific to data taken in the test beam are investigated, and the implementation of the method for CMS data taking is discussed.

Until April 2007 the MAGIC telescope used a 300 MSamples/s FADC system to sample the shaped PMT signals produced by the captured Cherenkov photons of air showers. Different algorithms to reconstruct the signal from the read-out samples (extractors) have been implemented and are described and compared. Criteria based on the obtained charge and time resolution/bias are defined and used to judge the different extractors, by applying them to calibration, cosmic and pedestal signals. The achievable charge and time resolution have been derived as functions of the incident number of photo-electrons.

In this paper we present a general approach for the design
of LeGall’s 5/3 linear phase perfect reconstruction filterbank. Using our generalized approach, we can design
the 5/3 filters with the coefficients having different finite
wordlengths (precision) - thus this can be seen as a generalization of the LeGall 5/3 filters. We construct design examples with the filters having different wordlengths, and
also analyze the frequency characteristics of the designed
filters.

In this work we investigate the effects of pansharpening on multitemporal multispectral (MS) and panchromatic (PAN) images for applications of change detection (CD). The rationale of the analysis consists in understanding in what conditions and to which extent the merging process can improve the results of a standard unsupervised method of change detection. In order to properly analyze and study this problem, different multiresolution fusion algorithms are tested to compare the CD results obtained from original MS and PAN data and from spatially enhanced MS data. Several experiments are carried out by applying some fusion techniques to coregistered Quickbird images acquired at different times. From these experiments some interesting conclusions are derived on the effectiveness and the appropriateness of the different investigated multiresolution fusion techniques with respect to the change detection application.

It is a well-known result of estimation theory that biased estimators can outperform unbiased ones in terms of expected quadratic error. In steady-state, many adaptive filtering algorithms offer an unbiased estimation of both the reference signal and the unknown true parameter vector. In this correspondence, we propose a simple yet effective scheme for adaptively biasing the weights of adaptive filters using an output multiplicative factor. We give theoretical results that show that the proposed configuration is able to provide a convenient bias vs variance tradeoff, leading to reductions in the filter mean-square error, especially in situations with a low signal-to-noise ratio (SNR). After reinterpreting the biased estimator as the combination of the original filter and a filter with constant output equal to 0, we propose practical schemes to adaptively adjust the multiplicative factor. Experiments are carried out for the normalized leastmean-squares (NLMS) adaptive filter, improving its mean-square performance in stationary situations and during the convergence phase.

ISAAC is a highly capable, highly reusable, modular, and integrated FPGA-based common instrument control and computing platform for a wide range of instrument needs as defined in the Earth Science National Research Council (NRC) Decadal Survey Report. This paper presents its motivation, technical approach, and the infrastructure elements. It also describes the first prototype, ISAAC I, and its application in the design of SMAP L-band radar digital filter.

An approach for algorithm specifications and development is described for SMAP's radar onboard processor with multi-stage demodulation and decimation bandpass digital filter. Point target simulation is used to verify and validate the filter design with the usual radar performance parameters. Preliminary FPGA implementation is also discussed.

J-DSP + is a java-based object-oriented programming environment that was developed at Arizona State University for use in the undergraduate DSP class . In this paper, we describe innovative software extensions on J-DSP to accommodate on-line laboratories for speech processing, image processing, and communications systems. Significant modifications in the object-oriented GUI of J-DSP that enable simulation of feedback systems are also presented. The speech processing functions enable on-line simulations of speech coding algorithms and include PCM and ADPCM quantization as well as more elaborate algorithms such as the LPC and the CELP. Image processing functionalities include development of 2-D signal processing capabilities including 2-D-FFT, 2-D-filter design, and 2-D graphics and picture processing. Communications functionality covers several aspects of analog and digital modulation and demodulation. On-line laboratory exercises have been developed in the aforementioned areas and posted on a web site (). This site also includes on-line evaluation forms for the exercises. Statistical and qualitative evaluations that assess the learning experiences of the students that use J-DSP are presented.

Linear filters are often employed in most signal processing applications. As a matter of fact, they are well understood within a uniform theory of discrete linear systems. However, many physical systems exhibit some nonlinear behaviour, and in certain situations linear filters perform poorly. One case is the problem of acoustic echo cancellation, where the digital filter employed has to identify as close as possible the acoustic echo path that is found to be highly nonlinear. In this situation a better system identification can be achieved by a nonlinear filter. The problem is to find a nonlinear filter structure able to realize a good approximation of the echo path without any significant increase of the computational load. Conventional Volterra filters are well suited for modelling that system but they generally need too many computational resources for a real time implementation. In this paper we consider some low complexity nonlinear filters in order to find out a filter structure able to achieve performances close to those of the Volterra filter, but with a reduced increase of the computational load in comparison to the linear filters commonly employed in commercial acoustic echo cancellers.

Linear filters are often employed in most signal processing applications. As a matter of fact, they are well understood within a uniform theory of discrete linear systems. However, many physical systems exhibit some nonlinear behaviour, and in certain situations linear filters perform poorly. One case is the problem of acoustic echo cancellation, where the digital filter employed has to identify as close as possible the acoustic echo path that is found to be highly nonlinear. In this situation a better system identification can be achieved by a nonlinear filter. The problem is to find a nonlinear filter structure able to realize a good approximation of the echo path without any significant increase of the computational load. Conventional Volterra filters are well suited for modelling that system but they generally need too many computational resources for a real time implementation. In this paper we consider some low complexity nonlinear filters in order to find out a filter structure able to achieve performances close to those of the Volterra filter, but with a reduced increase of the computational load in comparison to the linear filters commonly employed in commercial acoustic echo cancellers.