The objective of this
experiment was to design a digital filter for the given input specifications
using frequency sampling method. In this method the desired frequency response
Hd(w) is sampled at w=2*pi*k/N ( where N = length of the signal) The frequency
samples are thus obtained are taken as DFT coefficients H[k]. FIR filter
response is found by taken IDFT of H[k]. Magnitude and phase spectrum were
plotted for this. Pass band and stop band attenuation values were verified.
In frequency sampling, when the desired frequency response is undersampled, the resulting impulse response will be time aliased
ReplyDeleteFSM method has less chance of aliasing error than wondow method
ReplyDeleteFSM is computationally efficient hence it's good to use
ReplyDeleteDiscontinuity is observed in phase plot between lobes and also when the phase spectrum goes out of the range of -pi to pi.
ReplyDeleteZero crossing was observed in phase plot wherever magnitude was zero in the magnitude response.
ReplyDeleteThis method is useful for the design of non-prototype filters where the desired magnitude response can take any irregular shape
ReplyDeletePhase spectrum is linear within the positive lobes of magnitude spectrum.
ReplyDeleteIf compared to other filter methods FSM is easier to implement but not the optimum digital filter design method.
ReplyDeleteIt gives time aliased output of under sampled
ReplyDeleteFrequency sampling realization is computationally more efficient than direct form realization
ReplyDelete