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Filters Applications i. Separation of signals that have been combined Eg. Separating mothers heartbeat from baby's heart beat in EKG ii. Restoration of signals that have been distorted To restore music recorded with poor quality equipment Analog or Digital filters can be used Digital filters can achieve far superior results Can change parameters easily to accommodate specification changes at almost no cost Faster roll-offs ( high order filters are possible) But Digital filters can not be used for extremely high frequency applications due to limitations in sampling frequencies and processing times. Types of frequency selective Filters Low-pass, High-pass, Band-pass, and Band-reject filters (All-pass filters are also available )

Conversion from LPF to other types All can be implemented using Low-pass filters i. High-pass from Low-pass filters 1) Spectral inversion Change the sign of all the samples of the filter kernel [i.e. the impulse response h(n)] and add one to the sample at the center of symmetry [i.e., new kernel = (n)-h(n)]

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For this to work The original filter kernel must have left-right symmetry (linear phase) impulse must be added at the center of symmetry 2) Spectral reversal Change the sign of every other sample. (equivalent to multiplying by sinusoid of f=0.5 ==> modulation ==> shift in frequency domain by 0.5)

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iii. Band-reject filters Adding kernels that do not have overlapping pass bands

By combining filters in cascade and in parallel, various filters can be implemented Eg. Spectral inversion of band-pass filter to implement a band-reject filter Ideal filters are not realizable E.g. Ideal LPF H(For - h(n) = sin(nc)/nFor all n Non-causal h(n) 0 for n<0

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h(n) 0 for n<0 Unstable |h(n)| not less than Practical Digital Filter Design Considerations: Considerations: a. Causality Ideal filters are not causal and therefore not physically realizable Restrictions imposed due to causality i. The frequency response H() cannot be zero, except at a finite set of points in frequency This is a result of Paley-Wiener theorem, which we are not going to discuss (Please refer Chapter 8 of Digital Signal Processing text book by Proakis & Manolakis) ii. The magnitude |H()| cannot be constant at any finite range of frequencies and the transition from pass band to stop band cannot be infinitely sharp. This is a result of the Gibbs phenomenon which we are not going to discuss. (See the text book for more information)

iii. The real and imaginary parts of H() are interdependent and are related by the discrete Hilbert transform. As a result, the magnitude |H()| and phase () of H() cannot be chosen arbitrarily b. Linear Phase Filters that has linear phase with frequency (constant group delay) No phase distortion

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Where B() is real, and are constant Note: depending on whether B is positive or negative, pi radians should also be added to the angle. This may lead to a piece-wise linear phase since the range of angles is limited to + and - pi radians. These filters satisfy the following symmetry or asymmetry condition

i.e., if z1 is a root, 1/z1 is also a root For real h(n) , z*1 and 1/ z*1 are also roots (complex conjugate pair)

c. FIR vs IIR implementation If linear phase is a requirement choose a FIR filter Choose a FIR filter if linear phase is a requirement. Either FIR or IIR can be used if linear phase is not a requirement. Finite Impulse Response (FIR) and Infinite Impulse Response (IIR) filters FIR EE325-385 Page 5

Use recursion Weighted addition of input and output = output Faster Steeper cutoffs can be obtained with much lower order filters than FIR filters. IIR filter has lower side lobes in stop-band than FIR filter with same number of parameters (filter order) Number of parameters determine the resources Memory and computational complexity Higher the order - higher the number of parameters Only some FIR digital filters have linear phase and IIR digital filters (or any type of analog filters) do not have linear phase. d. Specifications of a practical frequency selective filters Using amplitude |H()|

FIR IIR

Use convolution Weighted addition of input = output Better performance, but slower

Specify i) Maximum tolerable passband ripple 1 (or band edge value 1/(1+2) if H2 is used) ii) Maximum tolerable stopband ripple 2 iii) Passband edge frequency p iv) Stopband edge frequency s Passband ripple in dB = 20 log 10(p) Eg. s= 0.001 ==> -60 dB

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Eg. s= 0.001 ==> -60 dB Stop-band ripple in dB = 20 log 10(s) Eg. p = 0.707 (i.e. 1/sqrt(2)) ==> -3 dB Transition width = s - p Frequency selective filters We only look at LTI filters specified by difference equation

Specify filter characteristics in frequency domain Determine coefficients of FIR or IIR filter that closely approximates the frequency response specifications Basic digital filter design involves approximating the ideal frequency response with a system that has the frequency response H() as above, and selecting the coefficients {ak} and {bk}

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