In this experiment a digital FIR filter is designed using Frequency sampling method.In FSM we subsitute w=2(pi)k/n.5 parameters namely, Attenuation in both pass and stop bands in dB, pass band and stop band frequencies in Hertz and Sampling frequency in Hertz are taken from user. Once we obtain the desired frequency response, Hd[w], we can directly convert them into the frequency domain using the conventional Discrete Fourier Transform algorithm, H[k]. The final output sequence, h[n] can be obtained by performing the Inverse Discrete Fourier Transform algorithm. The final output sequence, h[n] is always symmetric about the point of symmetry i.e. N/2 . Discontinuity is observed in phase plot between lobes and also when the spectrum goes out of the range that is from -pi to +pi.
Simplest method of filter design.
ReplyDeleteYes,it is better than windowing method
DeleteMany values of DFT ij filter design are zero which makes computation simpler.
DeleteHowever, time aliasing of the output signal, if undersampled, can be a major drawback.
ReplyDeleteBut with the correct choice of sampling frequency we can easily design a good filter
ReplyDeleteProper choice of sampling frequency is an important factor in designing filter
DeleteSpectrum goes out of range for -pi to pi
ReplyDelete