FIR Filter Design Analysis for Power Line Interference in ECG Signals

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IJIRST –International Journal for Innovative Research in Science & Technology| Volume 1 | Issue 6 | November 2014 ISSN (online): 2349-6010

FIR Filter Design Analysis For Power Line Interference In ECG Signals Pankaj Srivastava ME Student ECE National Institute of Technical Teachers Training and Research, Chandigarh, India

Rajesh Mehra Associate Professor ECE National Institute of Technical Teachers Training and Research, Chandigarh, India

Abstract Pick up of hum from power line is a very common phenomenon in the measurement of ECG signals. It is of prime need to reduce the variations coming due to power line so that one can analyse the most of the critical points in the measured signal. Power Line Interference (PLI) may seriously degrade the ECG signal, rendering the ECG analysis inaccurate. In this paper power line interference in the electrocardiogram (ECG) measurement is reduced with the help of FIR filters. FIR filter is designed & simulated to reduce the power line interference in ECG measurement. Developed Filter has been designed and analysed using two Window techniques namely Blackman and Kaiser techniques. Both Window based Filters are designed and simulated using MATLAB. It can be observed that from the simulated result that Blackman based design is better in terms of reduced noise performance, as compared to Kaiser Window based design. Keywords: ECG, FIR, Kaiser, Blackman and PLI. _______________________________________________________________________________________________________

I. INTRODUCTION Digital Signal processing is concerned with the digital representation of the signal and the use of the digital processor to analyze, modify or extract information from signal. To remove noise and interference from the signal and to obtain the spectrum are the reason for processing a digital signal. Flexibility, no drift in performance with temperature, superior performance is the some advantages of digital signal processing. The major DSP operations are Convolution, Correlation, filtering, transformation and modulation. Digital filters are very important part of DSP. Filters have two uses: signal separation and signal restoration. Signal separation is needed when a signal has been contaminated with interference, noise, or other signals. Signal restoration is used when a signal has been distorted in some way. The two major type of digital filter are FIR and IIR filter. A Finite Impulse Response (FIR) digital filter is one whose impulse response is of finite duration. The general difference equation for a FIR digital filter is ( ) ∑ ( ) (1) where y(n) is the filter output at discrete time instance n, b k is the k-th feed forward tap, or filter coefficient, and x(n-k) is the filter input delayed by k samples. The Σ denotes summation from k = 0 to k = M -1 where M is the number of feed forward taps in the FIR filter. FIR filters are simple to design and they are guaranteed to be bounded input-bounded output (BIBO) stable.FIR filters can have an exactly linear phase response. The impulse response of an IIR filter is of infinite duration. The general difference equation for an IIR digital filter is ( ) ∑ ( ) ∑ ( ) (2) Where ak and bk are the coefficient of the filter, i.e the current output sample, y(n), is a function of past outputs as well as present and past input sample, means IIR is feedback type system. IIR filters are useful for high-speed designs because they typically require a lower number of multiplies compared to FIR filters. IIR filters can also be designed to have a frequency response that is a discrete version of the frequency response of an analog filter. Another important feature of DSP is to extract power estimation of the signal. Its major use is in the astronomy and biomedical areas. Power density spectrum estimation methods can be divided into two main groups—nonparametric and parametric—where the Periodogram and window methods belong to the former. Windows may be fixed, e.g., the Hanning window; others have parameters for adjusting the side-lobe attenuation, e.g., the Kaiser window. The window methods decrease the variance of the spectrum estimate by smoothing.

II. THE WINDOW TECHNIQUE When the desired frequency response Hd(ejω) of the system has abrupt transitions (as in the case of an ideal low pass filter), then the impulse response hd(n) has infinite duration. The most obvious way to approximate such a filter (system) is to truncate its impulse response to, say, M +1 sample. The impulse response of the new filter (assuming hd(n) is casual) is thus: The last equation can also be rewritten as

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