DSP Two Marks

III ECE/ DIGITAL SIGNAL PROCESSING/ TWO MARKS QUESTIONS & ANSWERS
PREPARED BY : M .M ANIKANDAN.,M .E.,(Ph.d),ECE DEPT./ AM SEC

1
DFT[ ] x
*
( ) n =

n =0
N 1
x
*
( ) n e
j2
t
kn
N
=

n =0
N 1
x( ) n e
j2
t
kn
N
*
=

n =0
N 1
x( ) n e
j2
t
n ( ) N k
N
*
=X
*
( ) N k
IDFT[ ] X
*
( ) k =
1
N

k =0
N 1
X
*
( ) k e
j2
t
kn
N
=
1
N

k =0
N 1
X( ) k e
j2
t
kn
N
*
=
1
N

k =0
N 1
X( ) k e
j2
t
k ( ) N n
N
*
UNIT I
1. If H(k) is the N-point DFT of a sequence h(n), Prove that H(k) and H(N-K) are complex
conjugates. (Nov2008)
If DFT[x(n)]=X(k)
Then DFT[x*(n)]=X*(N-k)=X*((-k))
N
Proof:
DFT[x*(N-n)] = X*(k)
Proof:
= x*(N-n)
Therefore DFT[x*(N-n)] = X*(k)
2. What are the differences and similarities between DIF and DIT algorithms? (Nov2008)
Sl.No DIT FFT DIF FFT
1 The time domain sequence is
decimated
The DFT x(k) is decimated
2 Input sequence is to be given in bit
reversed order.
The DFT at the output is in bit reversed
order.
3 First calculate 2-point DFTs and
combines them
Decimates the sequence step by step to
2-point sequence and calculate DFT.
4 Suitable for calculating inverse
DFT.
Suitable for calculating DFT.
2
n =0
N 1
x( ) n y
*
( ) n =
1
N

k =0
N 1
X( ) k Y
*
( ) k
n =0
N 1
x( ) n y
*
( ) n =

n =0
N 1
x( ) n
1
N

k =0
N 1
Y( ) k e
j2
t
kn
N
*
=
1
N

n =0
N 1
x( ) n

k =0
N 1
Y
*
( ) k e
j2
t
kn
N
=
1
N

k =0
N 1
Y
*
( ) k

n =0
N 1
x( ) n e
j2
t
kn
N
=
1
N

k =0
N 1
X( ) k Y
*
( ) k
3. Define the properties of convolution. (April 2008, Nov 2005)
Commutative Law x(n)*h(n)= h(n)*x(n)
Associative Law [ x(n)*h1(n)]*h2(n)= x(n)*[h1(n)*h2(n)]
Distributive Law x(n)*[h1(n)]+h2(n)]= x(n)*h1(n)+x(n)*h2(n)
4. Draw the basic butterfly diagram of radix-2 FFT. (April 2005, May 2007 & April 2008)
a A=a+W
n
b

b W
n
B =a W
n
b
-1
5. State and prove parsevals relation for DFT. (Nov 2007)
If DFT[x(n)] = X(k)
and DFT[y(n)] = Y(k)
Then
Proof:
Hence proved
6. What do you mean by the term bit reversal as applied to FFT
In DIT algorithm we can find that for the output sequence to be in a natural order (i.e., X(k) , k=0,1,2,.N-1)
the input sequence has to be stored in a shuffled order. For an 8-point DIT algorithm the input sequence is in the order
x(0), x(4), x(2),x(6),x(1),x(5),x(3) and x(7). We can see that when N is a power of 2 , the input sequence must be
stored in bit-reversal order for the output to be computed in a natural order. For N = 8 the bit-reversal process is
shown in table.
3
Input sample
index
Binary
representation
Bit reversed
binary
Bit reversed sample
index
0 000 000 0
1 001 100 4
2 010 010 2
3 011 110 6
4 100 001 1
5 101 101 5
6 110 011 3
7 111 111 7
7. What are the advantages of FFT algorithm over direct computation of DFT? (May 2007)
- The complex multiplication in the FFT algorithm is reduced by (N/2) log2N times.
- Processing speed is very high compared to the direct computation of DFT.
8. The first five DFT coefficients of a sequence x(n) are x(0) = 20, x(1) = 5+j2, x(2) = 0, x(3)=0.2+j0.4, X(4) = 0.
Determine the remaining DFT coefficients. (May 2007)
By complex conjugate property x(5)=0.2-j0.4,x(6)=0,x(7)=5-j2
9. Define symmetric and Anti symmetric signals. How do you prevent aliasing while sampling a CT signal?
(May 2007)
A real valued signal x(n) is called symmetric if
X (n) = X (-n)
On the other hand, a signal x(n) is called antisymmetric
X (-n) = -X (n)
10. what is the necessary and sufficient condition on the impulse response for stability? (May 2007)
The necessary and sufficient condition for the impulse response is given by
+
|h (n)|<
n=-
11. Define Complex Conjugate of DFT property. (May 2007)
DFT
If x(n)X(k) then
N
X*(n)(X*(-k))
N
= X*(N-K)
12. What is FFT? (Nov 2006)
The fast Fourier transform is an algorithm is used to calculate the DFT. It is based on fundamental principal of
decomposing the computation of DFT of a sequence of the length N in to successively smaller discrete Fourier
Transforms. The FFT algorithm provides speed increase factor when compared with direct computation of the DFT.
4
13. State sampling theorem? (Nov 2006)
Sampling is the process to convert analog time domain continuous signal into discrete time domain
signal. But it is the process of converting only time domain not in amplitude domain.
Nyquist criteria:
We sample the signal based on the following condition i.e., f
s
2f
m
Where f
x
= Sampling frequency
F
m
= maximum signal frequency
If these above conditions are not satisfied we will meet the following demerits after the sampling
process. 1. Guard band , 2. Aliasing Effect
14. What is BIBO Stability? What is necessary and sufficient condition for BIBO stability?
(May 2006 , Nov 2004)
Any system is said to be BIBO stable of and only if every bounded input gives a bounded output.
The BIBO stability depends on the impulse response of the system. The necessary and sufficient condition for
BIBO stability
15. How will you perform linear convolution via circular convolution? (May 2006)
Let the length of x(n) be L, length of h(n) be M. then linear convolution of x(n) and h(n) can be obtained through
following steps.
i. Append x(n) with M-1 zeros. Hence its length will be L+M-1
ii. Append h(n) with L-1 zeros. Hence its length will be L+M-1
iii. Perform circular convolution of above sequences. The result is linear convolution of length
L+M-1
16. How many multiplications and additions are required to compute N-point DFT using radix-2 FFT?
In computing N-point DFT by this method the number of stages of computation will be m-times. The number r
is called the radix of the FFT algorithms.
In radix-2-FFT, the total number of complex additions are reduced to N log
2
N and total number of complex
multiplications are reduced too (N/@)log
2
N.
17. What is decimation-in-time algorithm?
The computation of 8-point DFT using radix -2 DIT FFT, involves three stages of computation. Here N= = 2
3
therefore r=2 and m=3.
The given 8 point sequence is decimated to 2 point sequences. For each 2 point sequence, the 2 point DFT is
computed. From the result of 2-point DFT the 4-point DFT can be computed. From the result of 4-point DFT, the 8-
point DFT can be computed.
Let the given sequence be X(o),X(1),X(2), X(3), X(4), X(5), X(6), X(7) which consists of 8 samples.
18. What is decimation-in-frequency algorithm?
In decimation in frequency algorithm the frequency domain sequence X(k) is decimated. In this method, the
output DFT sequence X(k) is divided into smaller sequence.
5
19. Derive the necessary and sufficient condition for an LTI system to be BIBO stable. (April 2005)
A system is BIBO stable, if for every bounded input, the output is finite. Mathematically if
|x(t)|
<
And
||y(t)|| < then the system is stable.
The necessary and sufficient condition for Continuous time signal is stable if and only if
-
|| h(t) ||
1
= |h(n)| dt <

In discrete time system

||h||
1
= |h(n)|
n=-
20. Define DFT pair? (April 2004 & May 2007)
The DTFT pairs are
X(k) = x(n)e
-j2kn/N
x(n) = X(k)e
j2kn/N
21. What is aliasing? (Nov 2003)
If we operate the sampler at f
x
< f
m
, the frequency components of the frequency spectrum will overlap with
each other i.e., the lower frequency of the second frequency component will overlap with higher frequency of the first
frequency component. This overlapping effect is called as Aliasing effect. For avoiding overlapping of high and low
frequency components, we have to use low-pass filter to cut the unwanted high frequency components.
22. Give any two properties of DFT
a)Periodicity x(k+n)=x(k)
b)Linearity DFT{a1x1(n)+a2x2(n)}=a1x1(k)+a2x2(k).
23. Explain Linearity property of DFT
DFT{x (n)}=x (k)&DFT{y (n)}=Y (k)
For any real valued constant a &b .
DFT{a x (n)+b y (n)=a X (k)+a Y (k)
6
1
2
3
1
1
1
2
3
3
1
1
2
2
3
1
1
4
3
2
1
=
4 3 + 6 + 2 +
8 3 + 2 + 3 +
12 6 + 2 + 1 +
4 9 + 4 + 1 +
=
15
16
21
18
24. What are the applications of FFT algorithms? (May/June 2009, April/May 2008)
The applications of FFT algorithm include
1. Linear filtering
2. Correlation
3. Spectrum analysis
25. What are twiddle factors of the DFT? (N/D 2007, M/J 2006)
The complex valued phase factor W
N
is called as twiddle factor,which is an n
th
root of unity as W
N
= e
-j2/N
.
26. How many additions and multiplications are needed to compute N-point FFT? (N/D 2007)
The total number of complex additions = Nlog
2
N
The total number of complex multiplication = N/2 log
2
N.
27. Calculate the number of multiplications in 64 point DFT using FFT? (M/J 2007, 2009)
Number of complex multiplications is given by = N/2 log
2
N
Here N = 64
= 64/2 log
2
64 = 32 log
2
2
6
= (32 x 6)log
2
2 = 192
Therefore Number of multiplications = 192.
28. Find the values of W
N
k
when N=8 and k=2 and also for k=3. (M/J 2007)
We know that W
N
= e
-j2/N
Here N=8, W
8
= e
-j2/8
= e
-j/4
When k=2, W
8
2
= (e
-j/4
)
2
= e
-j/2
= cos(/2) jsin(/2) = -j
When k=3, W
8
3
= (e
-j/4
)
3
= cos(3/4) jsin(3/4) = (-1/2) j(1/2)
29. Determine the circular convolution of the sequence x
1
(n) = {1,2,3,1} and x
2
(n) = {4,3,2,1}. (N/D 2007)
X
3
(n) = {15,16,21,18}
30. Find the linear convolution of {1,0,1} and {2,0,2}. (N/D 2007)
Y(n) = {2,0,4,0,2}
7
31. What is zero padding ? What are its uses?
Let the sequence x(n) has a length L. If we want to find the N-point DFT (N>L) of the sequence x(n), we have to
add (N-L) zeros to the sequence x(n). This is known as zero padding. The uses of padding a sequence with zeros are
(a) We can get better display of the frequency spectrum.
(b) With zero padding, the DFT can be used in linear filtering.
32. Distinguish between linear and circular convolution of two sequences.
Linear convolution Circular convolution
1.If x(n) is a sequence of L number of samples
and h(n) with M number of samples, after
convolution y(n) will contain N= L+M-1
samples.
If x(n) is a sequence of L number of samples and
h(n) with M number of samples, after
convolution y(n) will contain N= Max(L,M)
samples.
2. Linear convolution can be used to find the
response of a linear filter.
Circular convolution cannot be used to find the
3. Zero padding is not necessary to find the
Zero padding is necessary to find the response of
a linear filter.
33. What is meant by sectioned convolution?
If the data sequence x(n) is of long duration, it is very difficult to obtain the output sequence y(n) due to limited
memory of a digital computer. Therefore, the data sequence is divided into smaller sections. These sections are
processed separately one at a time and combined later to get the output.
34. What are the two methods used for the sectioned convolution?
The two methods used for the sectioned convolution are (1) the overlap-add method and (2) overlap-save method.
35. Write briefly about overlap-add method?
In this method the size of the input data block x
i
(n) is L. To each data block we append M-1 zeros and perform N-
point (N = L+ M-1) circular convolution of x
i
(n) with h(n). Since each data block is terminated with M-1 zeros, the
last M-1 points from each output block must be overlapped and added to first M-1 points of the succeeding block.
Hence, this method is called overlap-add method.
36. State the difference between (i) overlap-save method (ii) overlap-add method.
Overlap-save method Overlap-add method
1 In this method the size of the input data block
is N=L+M-1
In this method the size of the input
data block is L.
2 Each data block consists of the last M-1 data
points of the previous data block followed by
L new data points.
Each data block is L points and we
append M-1 zeros to compute N-point
DFT.
3 In each output block M-1 points are corrupted
due to aliasing, as circular convolution is
employed.
In this no corruption due to aliasing, as
linear convolution is performed using
circular convolution.
4 To form the output sequence the first M-1
data points are discarded in each output block
and the remaining data are fitted together.
To form the output sequence, the last
M-1 points from each output block is
added to the first (m-1) points of the
succeeding block.
8
37. What are the steps involved in calculating convolution sum?
The steps involved in calculating sum are
Folding
Shifting
Multiplication
Summation
38. How to obtain the output sequence of linear convolution through circular convolution?
Consider two finite duration sequences x(n) and h(n) of duration L samples and M
samples. The linear convolution of these two sequences produces an output sequence of duration L+M-1
samples, whereas, the circular convolution of x(n) and h(n) give N samples where N=max(L,M).In order to
obtain the number of samples in circular convolution equal to L+M-1, both x(n) and h(n) must be appended
with appropriate number of zero valued samples. In other words by increasing the length of the sequences
x (n) and h(n) to L+M-1 points and then circularly convolving the resulting sequences we obtain the same
result as that of linear convolution.
39. Define circular convolution.
Let x1(n) and x2(n) are finite duration sequences both of length N with DFTs X1(K) and X2(k)
If X3(k)=X1(k)X2(k) then the sequence x3(n) can be obtained by circular convolution defined as
40. Why FFT is needed?
The direct evaluation DFT requires N2 complex multiplications and N2 N complex additions.Thus for
large values of N direct evaluation of the DFT is difficult.By using FFT algorithm the number of complex
computations can be reduced. So we use FFT.
41. Why the computations in FFT algorithm is said to be in place?
Once the butterfly operation is performed on a pair of complex numbers (a,b) to produce (A,B), there
is no need to save the input pair. We can store the result (A,B) in the same locations as (a,b). Since the same
storage locations are used troughout the computation we say that the computations are done in place.
42. 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 for 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.
9
43. What is meant by radix-2 FFT?
The FFT algorithm is most efficient in calculating N point DFT. If the number of output points N can
be expressed as a power of 2 that is N=2M, where M is an integer, then this algorithm is known as radix-2
algorithm.
44. What is overlap-save method?
In this method the data sequence is divided into N point sections xi(n).Each section contains the last
M-1 data points of the previous section followed by L new data points to form a data sequence of length
N=L+M-1.In circular convolution of xi(n) with h(n) the first M-1 points will not agree with the linear
convolution of xi(n) and h(n) because of aliasing, the remaining points will agree with linear convolution.
Hence we discard the first (M-1) points of filtered section xi(n) N h(n). This process is repeated for all
sections and the filtered sections are abutted together.
45. State the properties of DFT.
1) Periodicity
2) Linearity and symmetry
3) Multiplication of two DFTs
4) Circular convolution
5) Time reversal
6) Circular time shift and frequency shift
7) Complex conjugate
8)Circular correlation
46. Define DFT and IDFT (or) What are the analysis and synthesis equations of DFT?
DFT(Analysis Equation)
N-1 nk
IDFT(Synthesis Equation)
47. Define DFT of a discrete time sequence.
The DFT is used to convert a finite discrete time sequence x(n) to an N-point frequency domain sequence
denoted by X(k). The N-point DFT of a finite duration sequence x(n) of length L, where LN is defined as
X(k) =
|
\
|
|
.
x
|
\
|
|
.
ne
j
|
\
|
|
.
2
|
\
|
|
.
t nk
N
N 1
= n 0
for k = 0, 1, 2, ....., (N-1)
10
H( ) e
j
e
=

n =

h( ) n e
jn
e
= 1 e
j2
e
+
= e
j
e
[ ] e
j
e
e
j
e
+ = e
j
e
( ) 2jsine = 2je
j
e
sine
UNIT II
1. Show that the filter with h (n) = [-1, 0, 1] is a linear phase filter. (Nov 2008, May 2007)

From the above equation we can find () = - which is the proportional to . Hence the filter h(n) is a linear
Phase filter.
2. What are the merits and demerits of FIR filters? (Nov 2005 & April 2008)
FIR filters that have ideal linear phase characteristics can be easily designed.
FIR filters realized non-recursively are always stable.
Errors arising from quantization of signals and finite word length effects are usually less critical for FIR filter
designs as these realization do not have feedback FIR filters are implemented through FFT algorithms, which greatly
reduced its processing time.
3. In the design of FIR digital filters, how is Kaiser window different from other windows? (Nov 2007)
It provides flexibility for the designer to select the side lobe level and N. 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.
4. State the condition for a digital filter to be causal and stable. (May 2007)
The response of the causal system to an input does not depend on future values of that input, but depends only on
the present and/or past values of the input. A filter is said to be stable, bounded-input bounded output stable, if every
bounded input produces a bounded output. A bounded signal has amplitude that remains finite.
5. What is the condition satisfied by linear phase FIR filter? (Nov/Dec 2003 & May 2007)
Linear phase is of the form () = k
Here k is constant. Thus phase shift is linearly proportional to frequency. For linear phase, the impulse response
should satisfy following condition.
h (n) = h (M-1-n)
6. Give any two properties of Butterworth filter and chebyshev filter. (Nov/Dec 2006, May/June 2006, Apr 2005 &
Nov 2004)
a. The magnitude response of the Butterworth filter decreases monotonically as the frequency increases ()
from 0 to .
b. The magnitude response of the Butterworth filter closely approximates the ideal response as the order N
increases.
c. The poles on the Butterworth filter lies on the circle.
d. The magnitude response of the chebyshev type-I filter exhibits ripple in the pass band.
e. The poles of the Chebyshev type-I filter lies on an ellipse.
7. What are the desirable and undesirable features of FIR Filters? (May2006)
The width of the main lobe should be small and it should contain as much of total energy as possible.
The side lobes should decrease in energy rapidly as w tends to
11
8. Define Hanning and Blackman window functions. (May 2006)
The window function of a causal hanning window is given by
W
Hann
(n) = 0.5 0.5cos2n/ (M-1), 0nM-1
0, Otherwise
The window function of non-causal Hanning window I s expressed by
W
Hann
(n) = 0.5 + 0.5cos2n/ (M-1), 0|n|(M-1)/2
0, Otherwise
The width of the main lobe is approximately 8/M and the peak of the first side lobe is at -32dB.
The window function of a causal Blackman window is expressed by
W
B
(n) = 0.42 0.5 cos2n/ (M-1) +0.08 cos4n/(M-1), 0nM-1
= 0, otherwise
The window function of a non causal Blackman window is expressed by
W
B
(n) = 0.42 + 0.5 cos2n/ (M-1) +0.08 cos4n/(M-1), 0|n|(M-1)/2
= 0, otherwise
The width of the main lobe is approximately 12/M and the peak of the first side lobe is at -58dB.
9. Write the magnitude function of Butterworth filter. What is the effect of varying order of N on magnitude and
phase response? (Nov 2005)
|H(j)|
2
= 1 / [ 1 + (/
C
)
2N
] where N= 1,2,3,.
10. Mention the necessary and sufficient condition for linear phase characteristics in FIR filter. (Nov 2005)
The necessary and sufficient conditions is that the phase function should be linear function w, which in turn
requires constant phase delay (or) constant phase and group delay i.e., Q(w) w
Q(w) = - w -w
11. What is linear phase? What is the condition to be satisfied by the impulse response in order to have a linear
phase? (Apr 2005 & Nov 2003)
For a filter to have linear phase the phase response (w) w is the angular frequency.
The linear phase filter does not alter the shape of the signal. The necessary and sufficient condition for a filter to
have linear phase.
h(n) = h(N-1-n); 0 n N-1
12. List the characteristics of FIR filters designed using window functions. (Nov 2004)
the Fourier transform of the window function W(e
jw
) should have a small width of main lobe
containing as much of the total energy as possible
the fourier transform of the window function W(e
jw
) should have side lobes that decrease in
energy rapidly as w to . Some of the most frequently used window functions are described in
the following sections.
12
13. Give the Kaiser Window function. (Apr 2004)
The Kaiser Window function is given by
W
K
(n) = I
0
() / I
0
() , for |n| (M-1)/2
Where is an independent variable determined by Kaiser.
= [ 1 (2n/M-1)
2
]
14. What are the different types of filters based on impulse response?
Based on impulse response the filters are of two types
1. IIR filter
2. FIR filter
The IIR filters are of recursive type, whereby the present output sample depends on the present input,
past input samples and output samples. The FIR filters are of non recursive type, whereby the present
output sample depends on the present input sample and previous input samples.
15. What are the different types of filters based on frequency response?
Based on frequency response the filters can be classified as
1. Lowpass filter
2. Highpass filter
3. Bandpass filter
4. Bandreject filter
16. What are the advantages and disadvantages of FIR filters?
Advantages:
1. FIR filters have exact linear phase.
2. FIR filters are always stable.
3. FIR filters can be realized in both recursive and non recursive structure.
4. Filters with any arbitrary magnitude response can be tackled using FIR sequence.
Disadvantages:
1. For the same filter specifications the order of FIR filter design can be as high as 5 to 10 times that
in an IIR design.
2. Large storage requirement is requirement
3. Powerful computational facilities required for the implementation.
17. What are the design techniques of designing FIR filters?
There are three well known methods for designing FIR filters with linear phase .They are
(1.)Window method
(2.)Frequency sampling method
(3.)Optimal or minimax design.
18. What is Gibbs phenomenon?
One possible way of finding an FIR filter that approximates H(ejw) would be to truncate the infinite
Fourier series at n=(N-1/2).Direct truncation of the series will lead to fixed percentage overshoots and
undershoots before and after an approximated discontinuity in the frequency response.
13
19. What are the desirable characteristics of the window function?
20. List the steps involved in the design of FIR filters using windows.
21. What are the advantages of Kaiser Window?
It provides flexibility for the designer to select the side lobe level and N
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
22. 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 are identified as DFT coefficients. The
filter coefficients are then determined as the IDFT of this set of samples.
23. 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
24. Draw the direct form realization of FIR system.
25. When cascade form realization is preferred in FIR filters?
The cascade form realization is preferred when complex zeros with absolute magnitude is less than one.
26. State the equations used to convert the lattice filter coefficients to direct form FIR Filter
coefficient.
27. Draw the direct form realization of a linear Phase FIR system for N even.
15
28. Draw the direct form realization of a linear Phase FIR system for N odd
29. Draw the M stage lattice filter.
30. State the equations used to convert the FIR filter coefficients to the lattice filter Coefficient.
31. What are the properties of FIR filter?
1. FIR filter is always stable.
2. A realizable filter can always be obtained.
3. FIR filter has a linear phase response.
32. What do you understand by linear phase response?
For a linear phase filter () , the linear filter does not alter the shape of the original signal. If the phase
response of the filter is nonlinear the output signal may be distorted one. In many cases a linear phase characteristic is
required throughout the passband of the filter to preserve the shape of a given signal within the passband. An IIR filter
16
cannot produce a linear phase. The FIR filter can give linear phase, when the impulse response of the filter is
symmetric about its mid-point.
33. What are the disadvantages of Fourier series method?
In designing FIR filter using Fourier series method the infinite duration impulse response is truncated at
n = (N-1/2). Direct truncation of the series will lead to fixed percentage overshoots and undershoots before and after
an approximated discontinuity in the frequency response.
34. What is the need for employing window technic for FIR filter design? OR What is window and why it is
necessary?
One possible way of finding an FIR filter that approximates H(e
j
) would be to truncate the infinite Fourier series at
n = (N-1/2). Abrupt truncation of the series will lead to oscillations in the passband and stopband. These oscillations
can be reduced through the use of less abrupt truncation of the Fourier series. This can be achieved by multiplying the
infinite impulse response by a finite weighing sequence w(h), called a window.
35. Define Rectangular and Hamming window functions.
36. Compare FIR and IIR filters. (May 2007)
Sl.No IIR FIR
1 H(n) is infinite duration H(n) is finite duration
2 Poles as well as zeros are present. Sometimes all
pole filters are also designed.
These are all zero filters.
3 These filters use feedback from output. They are
recursive filters.
These filters do not use feedback. They are
nonrecursive.
4 Nonlinear phase response. Linear phase is
obtained if H(z) = Z
-1
H(Z
-1
)
Linear phase response for h(n) = h(m-1-n)
5 These filters are to be designed for stability These are inherently stable filters
6 Number of multiplication requirement is less. More
7 More complexity of implementation Less complexity of implementation
17
8 Less memory is required More memory is required
9 Design procedure is complication Less complicated
10 Design methods:
1. Bilinear Transform
2. Impulse invariance.
Design methods:
1. Windowing
2. Frequency sampling
11 Can be used where sharp cutoff characteristics
with minimum order are required
Used where linear phase characteristic is
essential.
UNIT III
1. What is prewarping? (Nov 2003,2008)
When bilinear transformation is applied, the discrete time frequency is related continuous time frequency as,
= 2tan
-1
T/2
This equation shows that frequency relationship is highly nonlinear. It is also called frequency warping. This effect can
be nullified by applying prewarping. The specifications of equivalent analog filter are obtained by following relationship,
= 2/T tan /2
This is called prewarping relationship.
2. What is the relation betweeen analog and digital frequency in impulse invariant transformation?(April 2008)
T=
3. State the condition for a digital filter to be causal and stable. (May 2007)
The response of the causal system to an input does not depend on future values of that input, but depends only on the
present and/or past values of the input.
A filter is said to be stable, bounded-input bounded output stable, if every bounded input produces a bounded output.
A bounded signal has amplitude that remains finite.
4. Find the digital transfer function H (z) by using impulse invariant method for the analog transfer function H(s) =
1/(S+2). Assume T=0.5sec.
H(s) = 1/(s+2)
The system function of the digital filter is obtained by
H (z) = 1/ (1-e
-2T
z
-1
)
Since T=o.5 sec
H (z) = 1/ (1-.067Z
-1
)
5. Mention any two procedures for digitizing the transfer function of an analog filter. (Nov 2006)
1. Impulse Invariant Technique
2. Bilinear Transform Technique
18
6. what are the parameters that can be obtained from the chebyshev filter specification? (Nov 2006 May 2007)
(or)
Give the equation for the order N, major, minor and axis of an ellipse in case of chebyshev filter. (Nov 2005)
N cosh
-1
(/) / cosh
-1
(
S
/
P
)
Where = (10
0.1s
1)
= (10
0.1p
1)
7. What are the advantages and disadvantages of bilinear transformation? (May 2006)
Advantages:
The bilinear transformation provides one-to-one mapping.
Stable continuous systems can be mapped into realizable, stable digital systems.
There is no aliasing.
Disadvantage:
The mapping is highly non-linear producing frequency, compression at high frequencies.
Neither the impulse response nor the phase response of the analog filter is preserved in a digital filter
obtained by bilinear transformation.
8. What is impulse invariant mapping? What is its limitation? (Apr/May 2005)
The philosophy of this technique is to transform an analog prototype filter into an IIR discrete time filter whose
impulse response [h(n)] is a sampled version of the analog filters impulse response, multiplied by T.
This procedure involves choosing the response of the digital filter as an equi-spaced sampled version of the analog
filter.
9. What is frequency warping? (Nov2004 & May 2007)
The bilinear transform is a method of compressing the infinite, straight analog frequency axis to a finite one long
enough to wrap around the unit circle only once. This is also sometimes called frequency warping. This introduces a
distortion in the frequency. This is undone by pre-warping the critical frequencies of the analog filter (cutoff frequency,
center frequency) such that when the analog filter is transformed into the digital filter, the designed digital filter will meet
the desired specifications.
10. What are the limitations of impulse invariant mapping technique? (Apr2004)
The impulse invariance technique is appropriate only for band limited filter like low pass filter. Impulse invariance
design for high pass or band stop continuous-time filters, require additional band limiting to avoid severe aliasing
distortion, if impulse designed is used. Thus this method is not preferred in the design of IIR filters other than low-pass
filters.
11. Give the transform relation for converting low pass to band pass in digital domain. (Apr 2004)
Low pass with cut off frequency
C
to band pass with lower cut-off frequency
1
and higher cut-off
frequency
2
:
S -------------
C
( s
2
+ 1 2) / s (
2
-
1
)
19
The system function of the high pass filter is then
H(s) = H
p
{
C
( s
2
+ 1 2) / s (
2
-
1
)}
12. Give the bilinear transformation. (Nov2003)
The bilinear transformation method overcomes the effect of aliasing that is caused due to the analog frequency
response containing components at or beyond the nyquist frequency. The bilinear transform is a method of compressing
the infinite, straight analog frequency axis to a finite one long enough to wrap around the unit circle only once.
S = (2/T) (Z-1) (Z+1)
13. State the structure of IIR filter?
IIR filters are of recursive type whereby the present o/p sample depends on present i/p, past i/p
samples and o/p samples. The design of IIR filter is realizable and stable. The impulse response h(n) for
a realizable filter is h(n)=0 for n 0
14. State the advantage of direct form II structure over direct form I structure.
In direct form II structure, the number of memory locations required is less than that of direct form I
structure.
15. How one can design digital filters from analog filters?
Map the desired digital filter specifications into those for an equivalent analog filter.
Derive the analog transfer function for the analog prototype.
Transform the transfer function of the analog prototype into an equivalent digital filter transfer
function.
16. What do you understand by backward difference?
One of the simplest method for converting an analog filter into a digital filter is to approximate the
differential equation by an equivalent difference equation.
d/dt y(t)=y(nT)-y(nT-T)/T
The above equation is called backward difference equation.
17. What is the mapping procedure between S-plane & Z-plane in the method of mapping
differentials? What are its characteristics?
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 Zplane.
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.
18. What is meant by impulse invariant method of designing IIR filter?
In this method of digitizing an analog filter, the impulse response of resulting digital filter is a sampled
version of the impulse response of the analog filter. The transfer function of analog filter in partial
fraction form.
20
19. Give the bilinear transform equation between S-plane & Z-plane.
S=2/T(1-Z
-1
/1+Z
-1
)
20. What are the properties of bilinear transformation?
The mapping for the bilinear transformation is a one-to-one mapping that is for every point Z, there is
exactly one corresponding point S, and vice-versa.
The j-axis maps on to the unit circle |z|=1,the left half of the s-plane maps to the interior of the unit
circle |z|=1 and the half of the s-plane maps on to the exterior of the unit circle |z|=1.
21. Define an IIR filter.
The filter designed by considering all the infinite samples of impulse response are called IIR filters.
The impulse response is obtained by taking inverse fourier transform of ideal frequency response.
22. Distinguish between IIR and FIR filters.
The filter design starts from ideal frequency response. By taking inverse fourier transform of ideal
frequency response,the desired impulse response is obtained, which consists of infinite number of samples.
The digital filters designed by selecting only N samples of the impulse response are called FIR filters.
The digital filters designed by selecting all the infinite samples of impulse response are called IIR filters.
23. What are the requirements for an analog filter to be stable and causal?
(i) The analog filter transfer function H
a
(s) should be a rational function os s and the coefficients of
s should be real.
(ii) The poles should lie on the left half of s-plane.
(iii) The number of zeros should be less than or equal to number of poles.
24. Write a brief note on the design of IIR filter. (OR) How a digital IIR filter is designed?
For designing a digital IIR filter, first an equivalent analog filter is designed using any one of the
approximation technique and given specifications. The result of the analog filter design will be an analog
filter transfer function H
a
(s). The analog filter transfer function is transformed to digital filter transfer
function H(z) using either Bilinear or Impulse invariant transformation.
25. Mention the important features of IIR filters.
(i) The physically realizable IIR filters does not have linear phase.
(ii) The IIR filter specifications includes the desired characteristics for the magnitude response only.
26. What are the advantages and disadvantages of digital filters?
Advantages of digital filters:
(i) High thermal stability due to absence of resistors, inductors and capacitors.
(ii) The performance characteristics like accuracy, dynamic range, stability and tolerance can be
enhanced by increasing the length of the registers.
(iii) The digital filters are programmable.
(iv) Multiplexing and adaptive filtering are possible.

21
Disadvantages of digital filters:
(i) The bandwidth of the discrete signal is limited by the sampling frequency.
(ii) The performance of the digital filter depends on the hardware used to implement the filter.
27. What is the main objective of impulse invariant transformation?
The objective of this method is to develop an IIR filter transfer function whose impulse response is the
sampled version of the impulse response of the analog filter. Therefore the frequency response characteristic
of the analog filter is preserved.
28. What is the importance of poles in filter design?
The stability of a filter is related to the location of the poles. For a stable analog filter the poles should lie
on the left half of s-plane. For a stable digital filter the poles should lie inside the unit circle in the z-plane.
29. What is aliasing?
The phenomena of high frequency sinusoidal components acquiring the identity of low frequency
sinusoidal components after sampling is called aliasing.i.e. aliasing is higher frequencies impersonating
lower frequencies. The aliasing problem will arise if the sampling rate does not satisfy the Nyquist sampling
criteria.

30. What is aliasing problem in impulse invariant method of designing digital filters?Why it is absent
in bilinear transformation?
In impulse invariant mapping, the analog frequencies in the interval(2k-1)/T(2k+1)/T (where k is
an integer) maps into corresponding values of digital frequencies in the interval -. Hence the mapping
of to is many-to-one.
This will result in high frequency components acquiring the identity of the low frequency components if
the analog filter is not bandlimited. This effect is called aliasing. The aliasing can be avoided in bandlimited
filters by choosing very small values of sampling time(or very high sampling frequency).
The bilinear mapping is one-to-one mapping and so there is no effect of aliasing.
31. What is butterworth approximation?
In butterworth approximation, the error function is selected such that the magnitude is maximally flat in
the origin(i.e.,at =0) and monotonically decreasing with increasing .
32. Compare the impulse invariant and bilinear transformations.

Impulse invariant transformation Bilinear transformation
(i) It is many-to-one mapping. (i) It is one-to-one mapping.
(ii) The relation between analog and
digital frequency is linear.
(ii) The relation between analog and digital
frequency is nonlinear.
(iii) To prevent the problem of aliasing
the analog filters should be bandlimited.
(iii) There is no problem of aliasing and so the
analog filter need not be bandlimited.
(iv) The magnitude and phase response
of analog filter can be preserved by
choosing low sampling time or high
sampling frequency.
(iv) Due to the effect of warping, the phase
response of analog filter cannot be preserved.
But the magnitude response can be preserved
by prewarping.
22
33. Write the properties of butterworth filter.
(i) The butterworth filters are all pole designs.
(ii) At the cutoff frequency
c
, the magnitude of normalized butterworth filter is 1/2.
(iii) The filter order N, completely specifies the filter and as the value of N increases the magnitude
response approaches the ideal response.
(iv) The magnitude is maximally flat at the origin and monotonocally decreasing with increasing .
34. What is Chebyshev approximation?
In chebyshev approximation, the approximation function is selected such that the error is minimized
over a prescribed band of frequencies.
35. What is type-I chebyshev approximation?
In type-I chebyshev approximation, the error function is selected such that, the magnitude response is
equiripple in the passband and monotonic in the stopband.
36. What is type-II chebyshev approximation?
In type-I chebyshev approximation, the error function is selected such that, the magnitude response is
monotonic in the passband and equiripple in the stopband. The type-II magnitude response is called inverse
Chebyshev response.
37. How the order of the filter affects the frequency response of chebyshev filter.
From the magnitude response of type-I chebyshev filter it can be observed that the magnitude response
approaches the ideal response as the order of the filter is increased.
38. How the poles of chebyshev transfer function are located in s-plane?
The poles of the chebyshev transfer function symmetrically lies on an ellipse in s-plane.
39. Write the properties of chebyshev type-I filters.
(i) The magnitude response is equiripple in the passband and monotonic in the stopband.
(ii) The chebyshev type-I filters are all pole designs.
(iii) The normalized magnitude function has a value of 1/(1+
2
) at the cutoff frequency
c
.
(iv) The magnitude response approaches the ideal response as the value of N increases.
40.Compare the Butterworth and chebyshev Type-I filters.
Butterworth Chebyshev Type-I
(i) All pole design. (i) All pole design.
(ii) The poles lie on a circle in s-plane. (ii) The poles lie on an ellipse in s-plane.
(iii) The magnitude response is maximally
flat at the origin and monotonically
decreasing function of .
(iii) The magnitude response is equiripple in
passband and monotonically decreasing in
the stopband.
(iv) The normalized magnitude response
has a value of 1/2 at the cutoff frequency
c
.
(iv) The normalized magnitude response has a
value of 1/(1+
2
) at the cutoff frequency
c
.
23
(v) Only few parameters has to be
calculated to determine the transfer
function.
(v) A large number of parameter has to be
calculated to determine the transfer
function.
UNIT - IV
1. Express the fraction (-9/32) in sign magnitude, 2s complement notations using 6 bits. (Nov 2008)
Sign magnitude : 1.01001
2s complement : 1.10111
2. What are the various factors which degrade the performance of digital filter implementation when
finite word length is used? (Nov 2008)
3. What are the three types of quantization error occurred in digital systems? ( Nov 2006 & Apr 2008)
- Input quantization error
- coefficient quantization error
- product quantization error
4. What is meant by limit cycle oscillations? ((May 2006,Apr 2005 May 2007, Nov 2007 & Apr 2008)
In fixed point addition, overflow occurs due to excess of results bit, which are stored at the registers. Due
to this overflow, oscillation will occur in the system. Thus oscillation is called as an overflow limit cycle
oscillation.
5. Express the fraction(-7/32) in signed magnitude and twos complement notations using 6 bits. (Nov
2007)
Sign magnitude : 1.00111 2s complement : 1.11001
6. Express the fraction 7/8 and -7/8 in sign magnitude, 2s complement and 1s complement. (Nov 2006)
7/8 -7/8
Sign magnitude : 0.111 1.111
1s complement : 0.000 1.000
2s complement : 0.001 1.001
7. Define Sampling rate conversion. (May 2007)
Sampling rate conversion is the process of converting a signal from one sampling rate to another,
while changing the information carried by the signal as little as possible.
Sample rate conversion needed because different systems use different sampling rates.
8. Convert the number 0.21 into equivalent 6-bit fixed point number. (May 2007)
0.001101
24
9. Why rounding is preferred to truncation in realizing digital filter?(May2007)
Error introduced due to rounding operation is less compared to truncation. Similarly quantization
error due to rounding is independent of arithmetic operation. And mean of rounding error is zero. Hence
rounding is preferred over truncation in realizing digital filter.
10. What are the different quantization methods? (Nov 2006)
- amplitude quantization
- vector quantization
- scalar quantization
11. What is zero padding? Does zero padding improve the frequency resolution in the spectral estimate?
(Nov 2006)
The process of lengthening a sequence by adding zerovalued samples is called appending with
zeros or zero padding.
12. List the advantages of floating point arithmetic. (Nov 2006)
- Large dynamic range
- Occurrence of overflow is very rare
- Higher accuracy
13. Give the expression for signal to quantization noise ratio and calculate the improvement with an
increase of 2 bits to the existing bit.(Nov2006,Nov2005)
SNR
A / D
= 16.81+6.02b-20log
10
(R
FS
/
x
) dB.
With b= 2 bits increase, the signal to noise ratio will increase by 6.02 X 2 = 12dB.
14. Draw the probability density function for rounding. (Nov 2005)
Shows the probability density function of error in rounding operation.
15. Compare fixed point and floating point representations. (May/Jun 2006)
Fixed Point Arithmetic Floating Point Arithmetic
- It covers only the
dynamic range.
- Compared to FPA,
accuracy is poor
- Compared to FPA it is
low cost and easy to
design
- It is preferred for real
time operation system
- Errors occurs only for
multiplication
- Processing speed is
high
- It covers a large range of
numbers
- It attains its higher accuracy
- Hardware implementation is
costlier and difficult to design
- It is not preferred for real
time operations.
- Truncation and rounding
errors occur both for
multiplication and addition
- Processing speed is low
25
16. What is dead band? (Nov 2004)
In a limit cycle the amplitude of the output are confined to a range of value, which is called dead band.
17. How can overflow limit cycles be eliminated? (Nov 2004)
- Saturation Arithmetic
- Scaling
18. What is zero input limit cycle oscillation? (Apr 2004)
Zero Input Limit Cycles:
Zero input limit cycles are usually of lower amplitude in comparison with overflow limit cycles. If the
system enters to the limit cycles oscillations, it will continue even after input attains zero range.
19. What is steady state noise power at the output of an LTI system due to the quantization at the input to
L bits? (Nov 2003 &Apr 2004)
The steady state noise power is basically the variance of output noise.

P =
e
2
.1/2 |H()|
2
dw
-
Here
e
2
is the variance of input error signal.
e
2
= 2
-2L
R
FS
2
/48

v
2
= 2
-2L
R
FS
2
/48 X |H ()|
2
dw
-
This equation gives steady state noise power due to quantization.
20. What is meant by finite word length effects in digital filters? (Nov 2003)
The digital implementation of the filter has finite accuracy. When numbers are represented in digital
form, errors are introduced due to their finite accuracy. These errors generate finite precision effects or finite
word length effects.
When multiplication or addition is performed in digital filter, the result is to be represented by finite
word length (bits). Therefore the result is quantized so that it can be represented by finite word register. This
quantization error can create noise or oscillations in the output. These effects are called finite word length
effects.
21. What is round-off noise error?
Rounding operation is performed only on magnitude of the number. Hence round-off noise error is
independent of type of fixed point representation. If the number is represented by b
u
bits before quantization
and b bits after quantization, then maximum round-off error will be (2
_b
-2
-bu
)/2. It is symmetric about zero.
- Overflow is rare
phenomenon
- Overflow is a range
phenomenon
26
22. What is meant by fixed point arithmetic? Give example?
In the fixed point arithmetic, the digits to the left of the decimal point represent the integer part of the
number and digits to the right of the decimal point represent fractional part of the number. For example,
(1458.568)
10
(1101.101)
2
are the fixed point numbers note that base of the number system is also written outside
the bracket.
23. what are the different types of arithmetic in digital systems?
There are three types of arithmetic used in digital systems. They are fixed point arithmetic, floating
point ,block floating point arithmetic.
24. What are the different types of fixed point arithmetic?
Depending on the negative numbers are represented there are three forms of fixed point arithmetic.
They are sign magnitude,1s complement,2s complement
25. What is meant by sign magnitude representation?
For sign magnitude representation the leading binary digit is used to represent the sign. If it is equal
to 1 the number is negative, otherwise it is positive.
26. What is meant by 1s complement form?
In 1,s complement form the positive number is represented as in the sign magnitude form. To obtain
the negative of the positive number ,complement all the bits of the positive number.
27. What is meant by 2s complement form?
In 2s complement form the positive number is represented as in the sign magnitude form. To obtain
the negative of the positive number ,complement all the bits of the positive number and add 1 to the LSB.
28. What is meant by floating pint representation?
In floating point form the positive number is represented as F =2CM,where is mantissa, is a fraction
such that1/2<M<1and C the exponent can be either positive or negative.
29. What are the advantages of floating pint representation?
1.Large dynamic range 2.overflow is unlikely.
30. What is input quantization error?
The filter coefficients are computed to infinite precision in theory. But in digital computation the filter
coefficients are represented in binary and are stored in registers. If a b bit register is used the filter coefficients
must be rounded or truncated to b bits ,which produces an error.
31.What is product quantization error?
The product quantization errors arise at the out put of the multiplier. Multiplication of a b bit data with a b bit
coefficient results a product having 2b bits. Since a b bit register is used the multiplier output will be rounded or
truncated to b bits which produces the error.
27
32.What are the different quantization methods?
Truncation and Rounding
33. What is overflow oscillations?
The addition of two fixed point arithmetic numbers cause overflow when the sum exceeds the word size available to
store the sum. This overflow caused by adder make the filter output to oscillate between maximum amplitude limits. Such
limit cycles have been referred to as overflow oscillations.
34. What are the two kinds of limit cycle behavior in DSP?
(i) Zero limit cycle oscillations
(ii) Overflow limit cycle oscillations
35. What is meant by quantization step size?
Let us assume a sinusoidal signal varying between +1 and -1 having a dynamic range 2. If ADC used to convert the
sinusoidal signal employs b+1 bits including sign bit, the number levels available for quantizing x(n) is 2
b+1
. Thus the
interval between successive levels. q= 2/(2
b+1
) = 2
-b
. where q is known as quantization step size.
36.Explain briefly the need for scaling in the digital filter implementation.
To prevent overflow, the signal level at certain points in the digital filters must be scaled so that no overflow occurs in
the adder.
37. Why the limit cycle problem does not exist when FIR digital filter is realized in direct form or cascade form?
In the case of FIR filters, there are no limit cycle oscillations, if the filter is realized in direct form or cascade form
since these structures have no feedback.
38. Why rounding is preferred to truncation in realizing digital filter?
(i) The quantization error due to rounding is independent of the type arithmetic.
(ii) The mean of rounding error is zero.
(iii) The variance of the rounding error signal is low.
39. What is truncation?
Truncation is a process of discarding all bits less significant than LSB that is retained
40. What is Rounding?
Rounding a number to b bits is accomplished by choosing a rounded result as the b bit number closest number
being unrounded.
41. State some applications of DSP?
Speech processing ,Image processing, Radar signal processing.
42. what is meant by A/D conversion noise?
A DSP contains a device, A/D converter that operates on the analog input x(t) to produce xq(t) which
is binary sequence of 0s and 1s. At first the signal x(t) is sampled at regular intervals to produce a
sequence x(n) is of infinite precision. Each sample x(n) is expressed in terms of a finite number of bits
given the sequence xq(n). The difference signal e(n)=xq(n)-x(n) is called A/D conversion noise.
28
43.what is the effect of quantization on pole location?
Quantization of coefficients in digital filters lead to slight changes in their value. This change in
value of filter coefficients modify the pole-zero locations. Sometimes the pole locations will be changed
in such a way that the system may drive into instability.
44.which realization is less sensitive to the process of quantization?
Cascade form.
UNIT V
1. What are the factors that may be considered when selecting a DSP processor for an application?
(Nov 2008)
Architectural features, Execution speed, Type of arithmetic and Word length.
2. State the merit and demerit of multiported memories?(May 2007,Nov 2008)
3. What is meant by pipelining? (Apr2008 & Nov 2007)
A pipeline is the continuous and somewhat overlapped movement of instruction to the processor or in the
arithmetic steps taken by the processor to perform an instruction. With pipelining, the computer architecture
allows the next instructions to be fetched while the processor is performing arithmetic operations, holding
them in a buffer close to the processor until each instruction operation can be performed. The staging of
instruction fetching is continuous. The result is an increase in the number of instructions that can be
performed during a given time period.
4. What is the principal features of the harvard Architecture?(Apr 2008)
The Harvard architecture has two separate memories for their instructions and data. It is capable of
simultaneous reading an instruction code and reading or writing a memory or peripheral.
5. Differentiate between von Neumann and Harvard architecture? (May 2007)
Sl.No Harvard Architecture Von-Neumann Architecture
1 Separate memories for
program and data.
It shares same memory for program
and data.
2 The speed of execution in
Harvard architecture is high
The speed of execution is increased
by pipelining
3 In this architecture having a
common interval address and
data bus.
It is having a separate interval address
and data bus.
4 It is not suitable for DSP
processors.
It is normally used for Harvard
architecture.
29
6. Give the digital signal processing application with the TMS 320 family. (Nov 2006)
- DSP processors should have circular buffers to support circular shift operations.
- The DSP processor should be able to perform multiply and accumulate operations very fast.
- DSP processors should have multiple pointers to support multiple operands jumps and shifts.
7. What is the advantage of Harvard architecture of TMS 320 series? (Nov 2006)
- It shares same memory for program and data
- The speed of execution is increased by pipelining
- It is having a separate interval address and data bus.
- It is normally used for Harvard architecture
8. What are the desirable features of DSP Processors? (Nov 2006)
o DSP processors should have multiple registers so that data exchange from register to register is fast.
o DSP operations require multiple operands simultaneously. Hence DSP processor should have multiple
operand fetch capacity.
o DSP processors should have circular buffers to support circular shift operations.
o The DSP processor should be able to perform multiply and accumulate operations very fast.
o DSP processors should have multiple pointers to support multiple operands jumps and shifts.
o Multi processing ability.
9. What are the different types of DSP Architecture?
Von-Neumann Architecture
Harvard Architecture
Modified Harvard Architecture
VLIW Architecture
10. Define MAC unit?
The dedicated hardware unit is called MAC. It is called multiplier-accumulator. It is one of the
computational unit in processor. The complete MAC operation is executed in one clock cycle.
The DSP processors have a special instruction called MACD. This means multiply accumulate with
data shift.
11. Mention the Addressing modes in DSP processors.
- Short immediate addressing
- Short Direct Addressing
- Memory-mapped Addressing
- Indirect Addressing
- 6.5 bit reversed addressing mode
- Circular addressing
12. State the features f TMS3205C5x series of DSP processors.
Powerful 16 bit CPU
TDM port
16X16 bit multiplies / Add operations can be performed in single cycle.
30
224KX16 bit maximum addressable external memory space.
Full duplex synchronous serial port for coder / decoder interface.
On-chip scan based emulation logic.
Boundary scan
Low power dissipation
IEEE standard text access ports
13. Define Parallel logic unit?
It executes logic operations on the data without affecting the contents of ACC. PLU provides bit
manipulation which can be used to set, clear, test or toggle bits in data memory control or status registers.
14. Define scaling shifter?
The scaling shifter has a 16 bit input connected to the data bus and 32 bit output connected to the
ALU. The scaling shifter produces a left shift of 0 to 16 bits on the input data. The other shifters perform
numerical scaling, bit extraction, extended precision arithmetic and overflow prevention.
15. Define ARAU in TMS320C5X processor?
ARAU meant Auxiliary register and auxiliary register arithmetic unit. These register are used for
temporary data storage. The auxiliary register file is connected to the auxiliary register arithmetic unit. The
contents of the auxiliary register can be ARAU helps to speed up the operations of CALU.
16. What are the Interrupts available in TMS320C5X processors?
It has four general purpose interrupts.
- INT4
- INT1
- RS (Reset)
- NMI (Non Maskable interrupt)
-
17. What are the addressing modes available in TMS320C5X processors?
- Direct
- Indirect
- Immediate
- Register
- Memory mapped
- Circular Addressing
18. Write the syntax of assembly language syntax.
- The source statement can contain following four ordered fields. i.e.,
[Label][:] mnemonic [operand list] [; comment]
- The source statement follows following guidelines
- All the statements must begin with a label, a blank, an asterisk or a semicolon.
- Labels may be placed before the instruction mnemonic on the same line or on the proceeding line
in the first column.
- Each field must be separated with blanks.
- If comment begins in column 1 it must have semicolon or asterisk at its beginning. In other
columns, comments can begin with semicolon.
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19. What are the different buses of TMS320C5X and their functions?
The C5X architecture has four buses and their functions are as follows:
Program bus (PB):
It carries the instruction code and immediate operands from program memory space to the CPU.
Program address bus (PAB):
It provides addresses to program memory space for both reads and writes.

Data read bus (DB):
It interconnects various elements of the CPU to data memory space.
Data read address bus (DAB):
It provides the address to access the data memory space.
20. List the on-chip peripherals in 5X.
The C5X DSP on-chip peripherals available are as follows:
1. Clock Generator
2. Hardware Timer
3. Software-Programmable Wait-State Generators
4. Parallel I/O Ports
5. Host Port Interface (HPI)
6. Serial Port
7. Buffered Serial Port (BSP)
8. Time-Division Multiplexed (TDM) Serial Port
9. User-Maskable Interrupts
21. What are the applications of PDSPs?
Digital cell phones, automated inspection, voicemail, motor control, video conferencing, Noise
cancellation, Medical imaging, speech synthesis, satellite communication etc.
22. What are the different stages in pipeling?
(i)The fetch phase, (ii) The decode phase, (iii) Memory read phase, (iv) The execute phase.
23. Why do we need DSP processors?
Use a DSP processor when the following are required:
Cost saving
Smaller size
Low power consumption
Processing of many high frequency signals in real-time
24. What are the basic instruction features of DSP?
Ar i t hmet i c oper at i ons such as add, subt r act and mul t i pl y
Logic operations such as AND,OR,XOR and NOT
Multiply and accumulate(MAC) operation
Signal scaling operations for scaling the signal before and/or after digital signal processing
It is important that dedicated high-speed hardware be provided to carry out these operations.
32
25. What are the basic hardware features of DSP?
On-chip r egi st er s f or st or age of i nt er medi at e r esul t s
On-chip memories for signal samples (RAM)
On-chip program memory for programs and fixed data such as filter coefficients (ROM)
26. What are the advantages and disadvantages of VLIW architecture?
Advant ages of VLIW ar chi t ect ur es
Increased performance
Better compiler targets
Potentially scalable
Disadvantages of VLIW architectures
o Increased memory use
o High power consumption
o Misleading MIPS ratings
27. What are the three features of ALU?
Thr ee f eat ur es
(i) Status Flags sign , zero, carry, and overflow
(ii) Overflow Management limit the accumulator contents
(iii) Register File improves the efficiency
28. Write the operations of address generation unit.
1. Get t i ng a new val ue f r om an i mmedi at e oper and, a r egi st er , or a memor y l ocat i on
2. Incrementing or decrementing the current address
3. Adding or subtracting an offset to the current address
4. Adding or subtracting an offset to the current address, comparing the new address with the limits defined for a
circular addressing mode, and generating a new address as per the circular addressing mode algorithm
5. Generating a new address from the current address by applying the bit-reversed addressing mode algorithm
29. What are the seven types of indirect addressing?
Auto increment
Auto decrement
Post indexing by adding the contents of AR0
Post indexing by subtracting the contents of AR0
Single indirect addressing with no increment
Single indirect addressing with no decrement
Bit reversed addressing
30. List out the two types of immediate addressing.
1. Short immediate addressing
2. Long immediate addressing
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31. List out the five memory mapped registers.
CBSR1 Circular Buffer 1 Start Register
CBSR2 Circular Buffer 2 Start Register
CBER1 Circular Buffer 1 End Register
CBER2 Circular Buffer 2 End Register
CBCR Circular Buffer Control Register
32. Mention the two types of long immediate addressing mode.
- One-operand instructions
- Two-operand instructions

DSP Two Marks

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III ECE/ DIGITAL SIGNAL PROCESSING/ TWO MARKS QUESTIONS & ANSWERS

PREPARED BY : M .M ANIKANDAN.,M .E.,(Ph.d),ECE DEPT./ AM SEC

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