Near‐real time reduction of shipboard gravity using Kalman‐filtered GPS measurements

Geophysics ◽  
1991 ◽  
Vol 56 (12) ◽  
pp. 1971-1979 ◽  
Author(s):  
J. F. Genrich ◽  
J.-B. Minster

We have developed a Kalman filter to estimate accurate Eötvös corrections and horizontal ship accelerations from Global Positioning System (GPS) fixes. High‐resolution shipboard gravity measurements are obtained with a newly designed, linear phase, Finite Impulse Response (FIR) low‐pass filter. Both filters are combined to yield accurate, near‐real time, Eötvös‐corrected underway gravity estimates. Error ranges that reflect uncertainty in navigation for these estimates are calculated from autocovariances of Kalman velocity estimates by means of variance propagation expressions for time‐invariant linear digital filters. Estimates of horizontal ship acceleration are combined with a simplified instrument impulse response model in an attempt to remove transient noise from the gravimeter output. We apply the technique to data collected by two shipboard gravimeters, a LaCoste & Romberg Model S Air‐Sea Gravity Meter and a Bell Aerospace BGM-3 Marine Gravity Meter System, operated side‐by‐side on the Scripps R/V Thomas Washington during Leg 1 of the Roundabout expedition. In the absence of significant horizontal accelerations due to course or speed changes, both instruments yield data with good repeatability, characterized by rms differences of less than 1 mGal. Horizontal accelerations generate transient signals that cannot be modeled at present to an accuracy of better than 5 mGal. Difficulties in removing these transients are primarily due to insufficient quantitative knowledge of the response of the instrument, including the gyro‐stabilized platform. This can be determined analytically or empirically.

2018 ◽  
Vol 28 (4) ◽  
pp. 21-27
Author(s):  
I. S. Savinykh ◽  
D. A. Chemasov

Undoubted advantages of finite impulse response filters are their unconditional stability, the absence of limit cycles and the possibility of implementing a filter that does not introduce phase distortion. The disadvantage of such filters is the large cost required to compute the response. This paper considers three-stage interpolated finite impulse response low-pass filters. The maximum values of the interpolation factors are determined. Dependences of the coefficient of computational efficiency and the coefficient of increase in the registers of the three-stage interpolated low-pass filter on the values of the interpolation factors, the widths of the passband and the transition band are obtained. Relations for determining the optimal values of interpolation factors corresponding to the maximal value of computational efficiency coefficient are obtained. In addition, the dependencies of the maximum coefficient of computational efficiency and the optimal coefficient of increase in the registers of the three-stage interpolated low-pass filter on the widths of the passband and the transition band at the optimum values of the interpolation factors are obtained. Considered three-stage interpolated low-pass filters should be used in the case when the required stopband is significantly less than the sampling rate. In this case, three- stage interpolated filters require less computational resources for calculating the response than the two-stage interpolated filters or filter implemented by the transversal structure.


2014 ◽  
Vol 6 ◽  
pp. 129302
Author(s):  
Wenhua Xu ◽  
Hong Bao ◽  
Jianwei Mi ◽  
Guigeng Yang

Due to great flexibility, low damping, and variable structure in the cabin-cable system of five hundred meter Aperture Spherical Radio Telescope (FAST), a real-time digital low-pass filter based on the analysis of frequency is presented in this paper. Firstly, by the Lomb-Scargle theorem, it can obtain the fundamental frequency of cabin-cable system. Then, using the obtained frequency, a digital low-pass filter is designed to filter the measured data. After being filtered, the measured data are used for coarse control. Finally, the results of the experiments on the FAST 5 m model show that calculating the fundamental frequency is accurate and the filter is effective.


2017 ◽  
Vol 55 (9) ◽  
pp. 1579-1588 ◽  
Author(s):  
Ivaylo Christov ◽  
Tatyana Neycheva ◽  
Ramun Schmid ◽  
Todor Stoyanov ◽  
Roger Abächerli

In this work, optimal sparse linear phase Finite impulse response filters are designed using swarm intelligence-based Firefly optimization algorithm. Filters are designed to meet the desired specification with fixed and variable sparsity. The objective function is formulated consisting of three parameters, i. e., maximum passband ripple, maximum stopband ripple and stopband attenuation. The effectiveness of the proposed method is evaluated in two stages. In first stage, the designed filters have been compared with non- sparse in terms of deviation in their specification. The Comparative analysis depicts that the proposed approach of sparse linear phase FIR filter design method performs better than the conventional methods without significantly deviating from the desired specification. The proposed designed filter is then implemented on xilinx ISE14.7(Vertex7) design environment and their performance is compared in terms of time delay, resource utilizaion and frequency of operation. In the second stage, designed sparse FIR filters are compared with earlier state of art sparse FIR filters design techniques.


2021 ◽  
Vol 3 (2) ◽  
pp. 103
Author(s):  
Hendra J. Tarigan

A physical system, Low Pass Filter (LPF) RC Circuit, which serves as an impulse response and a square wave input signal are utilized to derive the continuous time convolution (convolution integrals). How to set up the limits of integration correctly and how the excitation source convolves with the impulse response are explained using a graphical type of solution. This in turn, help minimize the students’ misconceptions about the convolution integral. Further, the effect of varying the circuit elements on the shape of the convolution output plot is presented allowing students to see the connection between a convolution integral and a physical system. PSpice simulation and experiment results are incorporated and are compared with those of the analytical solution associated with the convolution integral.


For digital signal processing, communication systems and VLSI design architectures, an efficient FIR filter is required to eliminate the noise signals. To design an efficient FIR filter, the minimization of two parameters is required such as side lobe attenuation and power. These two parameters can be achieved by designing a FIR filter with the help of Fractional Fourier Transform (FrFT) and Canonical Signed digit (CSD) algorithm. In this work, Finite Impulse Response (FIR) low pass filter is designed by using both FrFT and ordinary Fourier Transform (FT) methods and their frequency responses are compared in terms of side lobe attenuation (SLA). After comparison of both methods, the better results are obtained for an FrFT based design of FIR low pass filter. Apart from this, the FrFT based design of FIR low pass filter is realized in direct form architecture and implemented in VLSI. Further, the Canonical Signed digit (CSD) algorithm is applied for the multiplication process in the architecture implementation to minimize the power consumption. Moreover, frequency response of FIR low pass filter is obtained by using MATLAB software and simulation and synthesis results are obtained by using Xilinx 13.1 ISE.


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