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EC2302 - DIGITAL SIGNAL PROCESSING
UNIT I
DISCRETE FOURIER TRANSFORM
1. Define DFT and IDFT (or) what the analysis and synthesis equations are of
DFT?
DFT(Analysis Equation)
N-1
nk
X(k)= x(n) WN ,
n=0
WN = e-j2/N
IDFT(Synthesis Equation)
N-1
- nk
WN = e-j2/N
k=0
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2
3
Linear convolution
Circular convolution
Zero padding is not necessary to find the Zero padding is necessary to find the response
response of a linear filter.
of a filter.
19. What are diffe rences between overlap-save and overlap-add methods.
No
Overlap-save method
Overlap-add method
In this method the size of the input data In this method the size of the input data
block is N=L+M-1
block is L
20. What are the differences and similarities between DIF and DIT algorithms?
Differences:
1) The input is bit reversed while the output is in natural order fo r DIT, whereas for
DIF the output is bit reversed while the input is in natural order.
2) The DIF butterfly is slightly different from the DIT butterfly, the difference being
that the complex multiplication takes place after the add-subtract operation in DIF.
Similarities:
Both algorithms require same number of operations to compute the DFT.Both
algorithms can be done in place and both need to perform bit reversal at some place
during the computation.
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EC2302 - DIGITAL SIGNAL PROCESSING
UNIT II
INFINITE IMPULSE RESPONSE DIGITAL FILTERS
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The mapping procedure between S-plane & Z-plane in the method of mapping
of differentials is given by
H(Z) =H(S)|S=(1-Z-1 )/T
The above mapping has the following characteristics
The left half of S-plane maps inside a circle of radius centered at Z= in
the Z-plane.
The right half of S-plane maps into the region outside the circle of radius in
the Z-plane.
The j -axis maps onto the perimeter of the circle of radius in the Z-plane.
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EC2302 - DIGITAL SIGNAL PROCESSING
UNIT III
FINITE IMPULSE RESPONSE DIGITAL FILTERS
The
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hd (n)=1/2 Hd(w)ejwn dw
-
2. Multiply the infinite impulse response with a chosen window sequence w(n) of
length N to obtain filter coefficients h(n),i.e.,
h(n)= hd(n)w(n) for |n|(N-1)/2
= 0
otherwise
3. Find the transfer function of the realizable filter
(N-1)/2
H(z)=z-(N-1)/2 [h(0)+h(n)(zn +z-n )]
n=0
8. What are the desirable characteristics of the window function?
The desirable characteristics of the window are
1. The central lobe of the frequency response of the window should
contain most
of the energy and should be narrow.
2. The highest side lobe level of the frequency response should be small.
3 .The side lobes of the frequency response should decrease in energy
rapidly as tends to .
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a. Rectangular window
b. Hamming window
c. Hanning window
d. Bartlett window
e. Kaiser window
a. Rectangular window:
The equation for Rectangular window is given by
W(n)= 1
0 n M-1
0
otherwise
b. Hamming window:
The equation for Hamming window is given by
WH(n)= 0.54-0.46 cos 2n/M-1
0 n M-1
0
otherwise
c. Hanning window:
The equation for Hanning window is given by
WHn (n)= 0.5[1- cos 2n/M-1 ]
0 n M-1
0
otherwise
d. Bartlett window:
The equation for Bartlett window is given by
WT (n)= 1-2|n-(M-1)/2|
0 n M-1
M-1
0
otherwise
e. Kaiser window:
The equation for Kaiser window is given by
Wk (n)= Io [1-( 2n/N-1)2 ]
for |n| N-1
Io ()
2
0
otherwise
where is an independent parameter.
10. What is the necessary and sufficient condition for linear phase characte ristic
in FIR filte r?
The necessary and sufficient condition for linear phase characteristic in
FIR filter is, the impulse response h(n) of the system should have the
symmetry property i.e.,
H(n) = h(N-1-n)
where N is the duration of the sequence.
11. What are the advantages of Kaiser window?
o It provides flexibility for the designer to select the side lobe level and N
o It has the attractive property that the side lobe level can be varied continuously
from the low value in the Blackman window to the high value in the
rectangular window
12. What is the principle of designing FIR filter using frequency sampling
method?
In frequency sampling method the desired magnitude response is sampled
and a linear phase response is specified .The samples of desired frequency response
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EC2302 - DIGITAL SIGNAL PROCESSING
are identified as DFT coefficients. The filter coefficients are then determined as the
IDFT of this set of samples.
13. For what type of filters frequency sampling method is suitable?
Frequency sampling method is attractive for narrow band frequency selective
filters where only a few of the samples of the frequency response are non zero.
14. Draw the direct form realization of FIR system.
1. z
1
1
1
1
1
1
15. Draw the direct form realization of a linear Phase FIR system for N even.
1
16.Draw the direct form realization of a linear Phase FIR system for N odd
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19. State the equations used to convert the lattice filter coefficients to direct
form FIR
Filter coefficient.
m(0) = 1
m(m) = km
m(k) = m-1 (k) + m(m) m-1 (m-k)
20. State the equations used to convert the FIR filter coefficients to the lattice
filter
Coefficient.
For an M_stage filter , m-1 (0) =1 and km = m (m)
m-1 (k) = m(k) - m(m) m(m-k) , 1km-1
1-m2 (m)
UNIT IV
FINITE WORD LENGTH EFFECTS
1. What is transposition theore m & transposed structure?
The transpose of a structure is defined by the following operations.
Reverse the directions of all branches in the signal flow graph
Interchange the input and outputs.
Reverse the roles of all nodes in the flow graph.
Summing points become branching points.
Branching points become summing points.
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filters?
17. What are the two types of limit cycle behavior of DSP?.
1. Zero limit cycle behavior
2.Over flow limit cycle behavior
18. What are the methods to prevent overflow?
1. Saturation arithmetic
2.Scaling
19. State some applications of DSP?
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Speech processing
Image processing
Radar signal processing.
UNIT V
MULTIRATE SIGNAL PROCESSING
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Y(exp(jw))= 0.5[X(exp(jw/2))+X(exp(j(w/2-pi)))
5. What is the need for anti aliasing filter prior to down sampling?
The spectra obtained after down sampling a signal by a factor M is the sum of
all the uniformly shifted and stretched version of original spectrum scaled by a factor
1/M, then down sampling will cause aliasing. In order to avoid aliasing the signal x(n)
is to be band limited to plus or minus pi/M . This can be done by filtering the signal
x(n) with a low pass filter with a cutoff frequency of pi/M. This filter is known as
antialiasing filter.
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