DSP Important Tips

Unit: 1 Realization of Digital Filter

1) For direct form I realization total no of storage element required : M+N (where M is
no of poles and N is no of zeroes)
2) For direct form II realization total no of storage element required: Max (M, N) (where
M is no of poles and N is no of zeroes) i.e. if M>N then the total no of storage required
is M.
3) For direct form II realization, you have to show the structure with combined
common delays (refer Salivahanan page no: 459 fig 9.7)
4) Two pair Ladder configuration for IIR filter:

If the system function is described by the following form:

H (z)

2
3

z 4 z z 1 2

(i)

We have to find P(z) and Q(z) ,

P(z)= even polynomial
Q(z)= odd polynomial

From Eqn no (i),

P(z)= 4 z 2 2

(ii)

Q(z)= z 3 z 1

(iii)

Now P(z)/Q(z) =

We get,

P(z)
Q(z)

z 3 z 1
, Now dividing and factoring P(z)/Q(z) ratio
4 z 2 2

= 0 z 1

1
1 z 1/ 2 z 1
1

Routh Array Calculation is same like the ladder structure

Routh Array:
Z-3
Z-2
Z-1
1

a3 = 1
a2 = 4
b1= 1/2
c1= 1/4

a1=1
a0 = 2
0
0

Now we have calculated, b1= (a1 a2 a3 a0)/ a2

a3 1
a
b 1
;1 2 8; 2 1
a2 4
b1
c1 4

These is the structure of two pair ladder structure. Now replace the value of 0 , 1 , 2 to
the above structure
We get,

5) For continued fraction ladder structure you have to show the total continued
function as well as the ladder structure.
6) If canonical form is stated then solve the problem with direct form II

Unit: 2 Design of IIR Digital Filters

1) For Butterworth and Chebyshev filter use the formula in decibel or convert it to
normal value. When converting to normal value by the formula = 20log A, if the
attenuation term is stated then take the negative sign of the dB value.
2) The order of the filter is always taken as the next integer calculated, i.e. if N =5.01
calculate it will be taken as N=6.
3) For stable filter R.H.S. poles should be neglected i.e. pole with positive real value
should be neglected.

Unit: 3 Finite Impulse Response Filter Design:

1) If the window length is not given but the desired frequency response is given of a
linear phase filter

Hd () = {

3 ,
0,

||

Then compare the frequency response with

Then

(1)
2

(1)

= 3, so we can calculate M= 7

This M will be the filter length.

2) If the range of the frequency response is given as

|| then we have to

consider as negative and positive. In this case the limit of the will be -to

to and to .
4
4
3) You have to remember all the window function in causal form (It is given in the
salivahanan)

Unit: 4 Discrete Fourier transform:

1) If two sequence is given as x1=(1,2,3,4) and x2=(1,1,2,2) , arrow is indicating the zero
Position.
Here the two sequence is started at -2 position, so the convoluted sequence will start from
-2+-2=-4 position
Like y= (convolution of x1 & x2) = (1, 3, 7, 13, 14, 14, 8) and the arrow is indicating the
zero position.
2) For overlap add and overlap save method the total length of the sequence will be
equal to the length of the linear convolution between them.
3) If the convolution is stated to be calculated using time domain method then you have
to use the time domain formula.

Unit: 5 FFT:
1) Comparison of computational complexity between DFT and FFT are
DFT
Complex Addition: N(N-1)
Complex Multiplication: N2

FFT
Nlog2N
N/2log2N

2) For N=6 and N=12 values you have to use composite radix technique (refer in
salivahanan)
3) Study radix 3 FFT technique of N=9.
4) For DIT FFT the input is time domain and in bit reversal form and output is in arranged
form where as in DIF FFT technique input is in arranged form and the output is in bit
reversal form.