SVS COLLEGE OF ENGINEERING

COIMBATORE-642 109
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

Academic Year: 2017-2018

EE6403 DISCRETE TIME SYSTEMS

AND SIGNAL PROCESSING
II EEE – IV SEM

PART A QUESTIONS AND ANSWERS

 UNIT I - INTRODUCTION
1
1. Determine if the system described the equation 𝑦(𝑛) = 𝑥(𝑛) + is causal or non
𝑥(𝑛−1)
causal. [May 2016]
1
𝑦(𝑛) = 𝑥(𝑛) +
𝑥(𝑛 − 1)
𝑛=0
1
𝑦(0) = 𝑥(0) +
𝑥(−1)
𝑛=1
1
𝑦(1) = 𝑥(1) +
𝑥(0)
𝑛=2
1
𝑦(2) = 𝑥(2) +
𝑥(1)
Outputs depend on present and past input therefore the system is causal.

2. What is an anti aliasing filter? [May 2016]

A filter that is used to reject frequency signal before it is sampled to remove the
aliasing of unwanted high frequency signals is called an anti aliasing filter.

3. Given a continuous signal x(t) = 2 cos 300 π t. What is the nyquist rate and fundamental
frequency of the signal. [Dec 2015]
x(t) = 2 cos 300 π t
2 π fm = 300 π
fm=150
Nyquist frequency =2 fm=300Hz

4. Find whether the signal 𝑥(𝑛) = 𝑢(𝑛) is power signal or energy signal? [Dec 2015][May
2013]
Energy
𝑛=∝

𝐸 = ∑ |𝑥(𝑛)|2
𝑛=−∝
𝑛=∝

𝐸 = ∑ |1|2 𝑢(𝑛)
𝑛=−∝
𝑛=∝

𝐸 = ∑|1|2 = 1 + 2 + ⋯ ∝ = ∝
𝑛=0

Power
𝑛=𝑁
1
𝑝 = lim ∑ |𝑥(𝑛)|2
𝑁→∞ 2𝑁 + 1
𝑛=−𝑁
𝑛=𝑁
1
𝑝 = lim ∑ |1|2
𝑁→∞ 2𝑁 + 1
𝑛=0
𝑛=𝑁
1
𝑝 = lim ( ∑ 1𝑛 )
𝑁→∞ 2𝑁 + 1
𝑛=0
1
𝑝 = lim (𝑁 + 1)
𝑁→∞ 2𝑁 + 1
 1
𝑁 (1 + 𝑁)
𝑝 = lim
𝑁→∞ 1
𝑁 (2 + 𝑁)
1
=
2
The energy of the signal is infinite and power is finite. Hence the signal is
power signal.

5. Check if the system described by the difference equation 𝑦(𝑛) = 𝑎 𝑦(𝑛 − 1) + 𝑥(𝑛) with
𝑦(0) = 1 is stable. [May 2015]
Given 𝑦(𝑛) = 𝑎 𝑦(𝑛 − 1) + 𝑥(𝑛)
𝑌(𝑧) − 𝑎 𝑍 −1 𝑌(𝑧) = 𝑋(𝑧)
𝑌(𝑧)(1 − 𝑎 𝑍 −1 ) = 𝑋(𝑧)
𝑌(𝑧) 1
𝐻(𝑧) = =
𝑋(𝑧) 1 − 𝑎 𝑍 −1
ℎ(𝑛) = 𝑎𝑛 𝑢(𝑛)
For stability
∝

∑ |ℎ(𝑛)| < ∞
𝑛=−∝
∞ ∞ ∞
1
∑ |ℎ(𝑛)| = ∑ |𝑎𝑛 | 𝑢(𝑛) = ∑|𝑎𝑛 | = 1 + 𝑎 + 𝑎2 … . . 𝑎∞ = <∞
1−𝑎
𝑛=−∞ 𝑛=−∞ 𝑛=0
System is stable.

6. Differentiate between Energy and power signal. [May 2015]

Energy signal Power signal

Energy of the signal x(n) is Power of the signal x(n) is

𝑛=∝ 𝑛=𝑁
1
𝐸= ∑ |𝑥(𝑛)|2 𝑃 = lim ∑ |𝑥(𝑛)|2
𝑁→∞ 2𝑁 + 1
𝑛=−∝ 𝑛=−𝑁

Signal is called Energy signal if E is finite Signal is called Power signal if E = ∞ and P
and P=0. finite

7. Test whether the system governed by the relation 𝑦(𝑛) = ∑∞

𝑘=−∞ 𝑋(𝑘) is linear time
invariant or not? [Dec 2014]
∞

𝑦(𝑛) = ∑ 𝑥(𝑘)
𝑘=−∞
Response due to delayed input,
∞

𝑦(𝑛, 𝑚) = ∑ 𝑥(𝑘 − 𝑚)
𝑘=−∞
Delayed response,
∞

𝑦(𝑛 − 𝑚) = ∑ 𝑥(𝑘 − 𝑚)
𝑘=−∞
𝑦(𝑛 − 𝑚) = 𝑦(𝑛, 𝑚)
System is time invariant.
8. What is aliasing effect? [Dec 2014] [Dec 2012] [May 2011] [May 2011]
If the signal x(t) sampled at the sampling rate F < 2fm results are in spectral
overlap. This signal cannot be recovered using a low pass filter. This effect is known as
aliasing.

9. Consider the analog signal x(t) = 3 cos 50π t + 10 sin 300π t - cos 100 π t . What is the
Nyquist rate for this signal? [May 2014]
x(t) = 3 cos 50π t + 10 sin 300π t - cos 100 π t
2 π fm1 = 50 π → fm1=25
2 π fm2 = 300 π → fm2=150
2 π fm3 = 100 π →fm3=50
fm =max(fm1, fm2, fm3)
fm=150
Nyquist frequency =2 fm=300Hz

10. State Shannon’s sampling theorem. [May 2014] [May 2011] [Dec 2011][May 2011]
A band limited continuous time signal, with higher frequency fm Hertz, can be
uniquely recovered from its samples provided the sampling rate F ≥ 2fm samples per
second.

11. What is Nyquist rate for this signal x(t) = 3 cos 600π t + 2cos 1800 π t ? [Dec 2013]
x(t) = 3 cos 600π t + 2cos 1800 π t
2 π fm1 = 600 π → fm1=300
2 π fm2 = 1800 π → fm1=900
fm =max(fm1, fm2,)
fm=900
Nyquist frequency =2 fm=1800Hz

π 30n
12. Determine the fundamental period of the signal 𝑐𝑜𝑠 ( ). [Dec 2013]
105
π 30n
𝑥(𝑛) = 𝑐𝑜𝑠 ( 105 )
30π
𝑤0 = ( )
105
2π 2π
The fundamental period 𝑁 = 𝑤 = 30π . 105=7
0

13. Given a continuous time signal x(t) = 2 cos 500π t What is the Nyquist rate and
fundamental frequency of the signal? [May 2013]
x(t) = 2 cos 500 π t
2 π fm = 500 π
fm=250
Nyquist frequency =2 fm=500Hz

14. What is an LTI system? [Dec 2012]

A linear time invariant (LTI) system follows two principles
 For liner system, the response due to linear combination of inputs is same as
linear combinations of corresponding outputs.
 Time invariance means shift of time origin of input doesnot change the response
of the system.

15. Define Nyquist rate. [May 2012]

𝑤
If the sampling frequency is 𝑤𝑠 , all the frequency above 2𝑠 causing aliasing. This
𝑤𝑠
aliasing can be avoided if the input signal frequencies are below . This frequency is
2
called Nyquist frequency / Nyquist rate.
16. What is a linear time invariant system? [Dec 2011]
If the input output relation of a system does not vary with time, the system is said
to be time invariant or shift invariant.
Example: y(n)= x(n)+x(n-1)

17. What is continuous and discrete time signal?

Continuous time signal:
A signal is said to be a continuous time signal if the amplitude and the time
interval are called continuous time signal. It is denoted by x(t).
Discrete time signal:
A signal is said to be a discrete time signal if the amplitude is continuous but
discrete in time are called discrete time signal. It is denoted by x(n).

18. Teat whether the system y(n)=0.5 x(n)+9 is linear and time invariant system.
Linear
For input x1(n), y1(n)=0.5 x1(n)+9
For input x2(n), y2(n)=0.5 x2(n)+9
a1 y1(n)+ a2 y2(n) = a1[0.5 x1(n)+9]+ a2 [0.5 x2(n)+9]
=0.5[a1 x1(n)+ a2 x2(n)]+9[a1 + a2]
Considering linear combination of inputs
x3(n)=a1 x1(n)+ a2 x2(n)
y3(n)=0.5[a1 x1(n)+ a2 x2(n)]+9
y3(n) ≠ a1 y1(n)+ a2 y2(n) hence system is non linear,

Time invariant
y(n)=0.5 x(n)+9
y(n,k) = y(n-k)
0.5 x(n - k)+9 = 0.5 x(n - k)+9
LHS =RHS
system is time invariant system.

π π
19. Find whether the signal 𝑥(𝑛) = 𝑐𝑜𝑠 (3 𝑛 + 6) is power signal or energy signal? 𝑥(𝑛) =
π π
𝑐𝑜𝑠 (3 𝑛 + 6)
Energy
𝑛=∝

𝐸 = ∑ |𝑥(𝑛)|2
𝑛=−∝
𝑛=∝ 2
π π
𝐸 = ∑ |(𝑐𝑜𝑠 ( 𝑛 + ))|
3 6
𝑛=−∝
𝑛=∝
π π
𝐸 = ∑ (𝑐𝑜𝑠 2 ( 𝑛 + ))
3 6
𝑛=−∝
𝑛=∝ 2π π
1 + 𝑐𝑜𝑠 ( 3 𝑛 + 6)
𝐸= ∑
2
𝑛=−∝
𝑛=∝ 𝑛=∝
1 2π π
𝐸 = ( ∑ 1𝑛 ) + ∑ 𝑐𝑜𝑠 ( 𝑛 + )
2 3 6
𝑛=−∝ 𝑛=−∝
1
E = (∝ −0) =∝
2
Power
 𝑛=𝑁
1
𝑝 = lim ∑ |𝑥(𝑛)|2
𝑁→∞ 2𝑁 + 1
𝑛=−𝑁
𝑛=𝑁 2
1 π π
𝑝 = lim ∑ |(𝑐𝑜𝑠 ( 𝑛 + ))|
𝑁→∞ 2𝑁 + 1 3 6
𝑛=−𝑁
𝑛=𝑁
1 1
𝑝 = lim ( ∑ 1𝑛 ) + 0
𝑁→∞ 2𝑁 + 1 2
𝑛=−𝑁
1 1 1
𝑝 = lim (2𝑁 + 1) = 𝑁
𝑁→∞ 2𝑁 + 1 2 2
The energy of the signal is infinite and power is finite. Hence the signal is
power signal.

20. A discrete time signal x(n)={0,0,1,1,2,0,0...} Sketch the x(n) and x(-n+2) signals.
x(n)

x(-n)

x(-n+2)

21. Determine whether the following signals are periodic, if periodic then compute the
fundamental period.
𝜋62𝑛
a. cos(0.01 𝜋n) b. 𝑠𝑖𝑛 ( 10 )
For periodicity 𝑥(𝑛) = 𝑥(𝑛 + 𝑁)
𝑥(𝑛 + 𝑁) = 𝑐𝑜𝑠 [0.01𝜋 (𝑛 + 𝑁)]
= 𝑐𝑜𝑠 [0.01𝜋 𝑛 + 0.01𝜋𝑁 ]
2𝜋 2𝜋
Fundamental period 𝑇0 = 𝑊 = 0.01𝜋 = 200
For periodicity 𝑥(𝑛) = 𝑥(𝑛 + 𝑁)
𝜋62
𝑥(𝑛 + 𝑁) = 𝑠𝑖𝑛 ( 10 (𝑛 + 𝑁))
𝜋62𝑛 𝜋62𝑁
= 𝑠𝑖𝑛 ( + )
10 10
 Fundamental period
2𝜋 2𝜋 20
𝑇0 = = =
𝑊 62 𝜋 62
10
22. What is Energy and power signal.
Energy of the signal x(n) is
𝑛=∞

𝐸 = ∑ |𝑥(𝑛)|2
𝑛=−∞
Power of the signal x(n) is
𝑛=𝑁
1
𝑃 = lim ∑ |𝑥(𝑛)|2
𝑁→∞ 2𝑁 + 1
𝑛=−𝑁
Signal is called Energy signal if Energy is finite and Power is zero.
Signal is called Power signal if E = infinite and P finite value.

23. List the properties of discrete time sinusoidal signals.

 A discrete time sinusoidal is periodic only if its frequency is a rational number.
 The highest rate of oscillation in a discrete time sinusoidal is attained when w= 𝜋.

24. What is correlation? What are its types?

The correlation measures the similarity between two signals.
 Cross correlation - similarity between different signals
 Auto correlation - similarity between time shifted version of same signal.

𝜋 𝜋
25. Determine the odd and even components of the signal, 𝑥(𝑛) = 𝑒𝑥𝑝 (𝑗 4 𝑛 + 𝑗 2) where 𝑗 =
√−1.
𝜋 𝜋
𝑥(𝑛) = 𝑒𝑥𝑝 (𝑗 𝑛 + 𝑗 )
4 2
𝜋 𝜋
𝑥(−𝑛) = 𝑒𝑥𝑝 (−𝑗 𝑛 + 𝑗 )
4 2
Odd component
1 1 𝜋 𝜋 𝜋 𝜋
𝑥0 (𝑛) = [𝑥(𝑛) − 𝑥(−𝑛) ] = [𝑒𝑥𝑝 (𝑗 𝑛 − 𝑗 ) − 𝑒𝑥𝑝 (−𝑗 𝑛 − 𝑗 ) ]
2 2 4 2 4 2
1 𝜋 𝜋 𝜋 𝜋 𝜋 𝜋 𝜋 𝜋
= 2 [𝑐𝑜𝑠 (4 𝑛 − 2) + 𝑗 sin (4 𝑛 − 2) −𝑐𝑜𝑠 (4 𝑛 − 2) + 𝑗 sin (4 𝑛 − 2)]
𝜋 𝜋
= 𝑗 sin ( 𝑛 − )
4 2
Even component
1 1 𝜋 𝜋 𝜋 𝜋
𝑥𝑒 (𝑛) = [𝑥(𝑛) − 𝑥(−𝑛) ] = [𝑒𝑥𝑝 (𝑗 𝑛 − 𝑗 ) + 𝑒𝑥𝑝 (−𝑗 𝑛 − 𝑗 ) ]
2 2 4 2 4 2
1 𝜋 𝜋 𝜋 𝜋 𝜋 𝜋 𝜋 𝜋
= 2 [𝑐𝑜𝑠 (4 𝑛 − 2) + 𝑗 sin (4 𝑛 − 2) +𝑐𝑜𝑠 (4 𝑛 − 2) − 𝑗 sin (4 𝑛 − 2)]
𝜋 𝜋
= cos ( 𝑛 − )
4 2
 UNIT II - DISCRETE TIME SYSTEM ANALYSIS
1. Determine the Z-transform and ROC of the following finite duration signals. [May 2016]
i) x(n)={3,2,2,3,5,0,1}
ii) x(n)=δ(n-k)
x(n)={3,2,2,3,5,0,1}
X(Z)=3+2z-1 +2z-2+3z-3+5z-4+z-6
ROC is entire z plane except z=0;
x(n)=δ(n-k)
Z {δ (n-k)} =𝑧 −𝑘 X (z)
ROC is all z except 0 if k >0

2. Compute the convolution of the two sequence x(n)={2,1,0,0,5} and y(n)={2,2,1,1} [May
2016]

2 2 0 0 5
2 4 4 0 0 10
2 4 4 0 0 10
1 2 2 0 0 5
1 2 2 0 0 5
y(n)={4,8,6,4,12,10,5,5}

3. What is ROC of Z transform? State its properties. [Dec 2015] [May 2014][Dec 2012][Dec
2011][May 2011]
The region of convergence (ROC) is defined as the set of all values of z for which
X(z) converges.
 The ROC is ring in the z plane centered at the origin.
 The ROC cannot contain any pole.
 The ROC of a LTI stable system contains the unit circle.
 The ROC must be a connected region.

4. State initial and final value theorem of Z transform. [Dec 2015] [May 2014]
Initial value theorem
If X(z)=Z{x(n)}
𝑥(0) = 𝑙𝑖𝑚 X(z)
𝑍→∞
Final value theorem
If X(z)=Z{x(n)}
𝑥(∞) = 𝑙𝑖𝑚(1 − Z −1 )X(z)
𝑍→1

5. Determine the Z transform of 𝑥(𝑛) = 𝑎𝑛 [May 2015]

Let us take 𝑥(𝑛) = 𝑎𝑛 u(n)
∞

𝑋(𝑧) = ∑ 𝑥(𝑛)𝑧 −𝑛
𝑛=−∞
∞ ∞ ∞
𝑛 −𝑛 𝑛 −𝑛
1 𝑧
𝑋(𝑧) = ∑ 𝑎 u(n)𝑧 = ∑𝑎 𝑧 = ∑(𝑎𝑧 −1 )𝑛 = −1
=
1 − 𝑎𝑧 𝑧−𝑎
𝑛=−∞ 𝑛=0 𝑛=0
6. Determine the Fourier transform of the signal x(t)=sin w0t [May 2015]
∞
𝑋(𝑤) = ∫ 𝑥(𝑡)𝑒 −𝑗𝑤𝑡 𝑑𝑡
−∞
∞
𝑋(𝑤) = ∫ sin w0t 𝑒 −𝑗𝑤𝑡 𝑑𝑡
−∞
∞
𝑒 𝑗𝑤0 𝑡 − 𝑒 −𝑗𝑤0 𝑡
𝑋(𝑤) = ∫ ( ) 𝑒 −𝑗𝑤𝑡 𝑑𝑡
−∞ 2j
∞
𝑒 −𝑗(𝑤−𝑤0 )𝑡 𝑒 −𝑗(𝑤+𝑤0 )𝑡
𝑋(𝑤) = ∫ ( − ) 𝑑𝑡
−∞ 2j 2j
𝑗
= 2 [𝛿(𝑤 − 𝑤0 ) − 𝛿(𝑤 + 𝑤0 )]

7. Find the Z transform and its ROC of the discrete time signals 𝑥(𝑛) = − 𝑎𝑛 𝑢(−𝑛 − 1), a>0.
[Dec 2014]
∞

𝑋(𝑧) = ∑ 𝑥(𝑛)𝑧 −𝑛
𝑛=−∞
−1 ∞ ∞
𝑛 −𝑛 −𝑛 𝑛
1
−1 𝑛
𝑎−1 𝑧
𝑋(𝑧) = ∑ −𝑎 𝑧 = ∑𝑎 𝑧 = ∑(𝑎 𝑧) = −1=
1 − 𝑎−1 𝑧 1 − 𝑎 −1 𝑧
𝑛=−∞ 𝑛=1 𝑛=0

8. Determine the Z transform of for the signal 𝑥(𝑛 ) = 𝛿 (𝑛 − 𝑘) + 𝛿 (𝑛 + 𝑘).[Dec 2013]

Z {δ (n)} =1
Z {δ (n - k)} =𝑧 −𝑘 X (z)
Z {δ (n + k)} =𝑧 𝑘 X (z)
Z{ 𝛿 (𝑛 − 𝑘) + 𝛿 (𝑛 + 𝑘)}=( 𝑧 −𝑘 + 𝑧 𝑘 ) X (z)

9. Prove the convolution property of z – transform. [Dec 2013]

𝑍 {𝑥1 (𝑛)} = X1 (z)
𝑍 {𝑥2 (𝑛)} = X2 (z)
𝑍 {𝑥1 (𝑛) ∗ 𝑥2 (𝑛)} = X1 (z) X2 (z)

10. Given a difference equation 𝑦(𝑛) = 𝑥(𝑛) + 3𝑥(𝑛 − 1) + 2 𝑦(𝑛 − 1). Determine the
system function H (z). [May 2013]
𝑦(𝑛) − 2 𝑦(𝑛 − 1) = 𝑥(𝑛) + 3𝑥(𝑛 − 1)
𝑌(𝑧) − 2 𝑍 −1 𝑌(𝑧) = 𝑋(𝑧) + 3 𝑍 −1 𝑋(𝑧)
𝑌(𝑧)(1 − 2 𝑍 −1 ) = 𝑋(𝑧) (1 + 3 𝑍 −1 )
𝑌(𝑧) 1 + 3 𝑍 −1
𝐻(𝑧) = =
𝑋(𝑧) 1 − 2 𝑍 −1

1 𝑛
11. Find the stability of the system whose impulse response ℎ(𝑛) = (2) 𝑢(𝑛) [May 2013]
For stability
∝

∑ |ℎ(𝑛)| < ∞
𝑛=−∝
1 𝑛
Given ℎ(𝑛) = (2) 𝑢(𝑛)
∞ ∞ ∞
1 𝑛 1 𝑛 1 1 1 1
∑ |ℎ(𝑛)| = ∑ |( ) 𝑢(𝑛) | = ∑ |( ) | = 1 + + … . . ∞ = =2<∞
2 2 2 4 2 1
𝑛=−∞ 𝑛=−∞ 𝑛=0 1−2
 System is stable.

12. Define discrete time Fourier transform pair for a discrete sequence. [Dec 2012]
∞
𝑗𝑤
𝑋(𝑒 ) = ∑ 𝑥(𝑛)𝑒 −𝑗𝑤𝑛
𝑛=−∞

13. Give relation between DTFT and Z transform. [May 2012]

The Z transform of x(n) is given by
∞

𝑋(𝑧) = ∑ 𝑥(𝑛)𝑧 −𝑛
𝑛=−∞
where 𝑧 = 𝑟𝑒 𝑗𝑤
The DTFT of x(n) is given by
∞

𝑋(𝑒 𝑗𝑤 ) = ∑ 𝑥(𝑛)𝑒 −𝑗𝑤𝑛

𝑛=−∞

14. Write the commutative and distributive properties of convolution. [Dec 2011]
Commutative property
𝑥(𝑛 ) ∗ ℎ(𝑛 ) = ℎ(𝑛 ) ∗ 𝑥(𝑛 )
Distributive property
𝑥(𝑛 ) ∗ [ℎ1 (𝑛 ) + ℎ2 (𝑛 )] = 𝑥(𝑛 ) ∗ ℎ1 (𝑛 ) + 𝑥(𝑛 ) ∗ ℎ2 (𝑛 )

15. Write the DTFT for (a) 𝑥(𝑛 ) = 𝑎𝑛 𝑢(𝑛 ), (b) 𝑥(𝑛 ) = 4𝛿 (𝑛 ) − 3𝛿 (𝑛 − 1).

𝑥(𝑛 ) = 𝑎𝑛 𝑢(𝑛 )
∞ ∞ ∞
𝑗𝑤 −𝑗𝑤𝑛 𝑛 −𝑗𝑤𝑛
𝑋(𝑒 ) = ∑ 𝑥(𝑛)𝑒 = ∑ 𝑎 𝑢(𝑛 )𝑒 = ∑ 𝑎𝑛 𝑒 −𝑗𝑤𝑛
𝑛=−∞ 𝑛=−∞ 𝑛=0

∞
1
= ∑(𝑎 𝑒 −𝑗𝑤 )𝑛 =
1 − 𝑎𝑒 −𝑗𝑤
𝑛=0

𝑥(𝑛 ) = 4𝛿 (𝑛 ) − 3𝛿 (𝑛 − 1) .
∞ ∞
𝑗𝑤 −𝑗𝑤𝑛
𝑋(𝑒 ) = ∑ 𝑥(𝑛)𝑒 = ∑(4𝛿 (𝑛 ) − 3𝛿 (𝑛 − 1) ) 𝑒 −𝑗𝑤𝑛
𝑛=−∞ 𝑛=0

∞ ∞

= 4 ∑ 𝛿 (𝑛 )𝑒 −𝑗𝑤𝑛 − 3 ∑ 𝛿 (𝑛 − 1) ) 𝑒 −𝑗𝑤𝑛 = 4 − 3𝑒 −𝑗𝑤

𝑛=0 𝑛=0

16. What is the relation between DFT and Z-Transform?

Let N point DFT of x (n) be X (K) and z transform of x (n) be X (Z). The N point
sequence X(K) can be obtained from X(Z) by evaluating X(Z) at N equally spaced points
around the unit circle.

17. Perform linear convolution for the following sequence x1(n)={1,2,3,4}, x2(n)={1,2,2,1}.

1 2 2 1
1 1 2 2 1
 2 2 4 4 2
3 3 6 6 3
4 4 8 8 4
y(n)={1,4,9,15,16,11,4}

18. Perform circular convolution of x1(n)={1,2,3,4}, x2(n)={1,1,2,2} using matrix method.

1 2 2 1 1 15
1 1 2 2 2 = 17
2 1 1 2 3 15
2 2 1 1 4 13
y(n)={15,17,15,13}

19. List the properties of discrete time sinusoidal signals.

 A discrete time sinusoidal is periodic only if its frequency is a rational number.
 The highest rate of oscillation in a discrete time sinusoidal is attained when w= 𝜋.

20. Distinguish between DFT and DTFT.

DFT DTFT
Obtained by performing sampling
Sampling is performed only in time
operation in both the time and frequency
domain
domains.

Discrete frequency spectrum Continuous function of w.

21. Find the DTFT of 𝑥(𝑛) = −𝑏 𝑛 𝑢(−𝑛 − 1).

∝

𝑋(𝑒 𝑗𝑤 ) = ∑ 𝑥(𝑛). 𝑒 −𝑗𝑤𝑛

𝑛=−∝
∝

𝑋(𝑒 𝑗𝑤 ) = ∑ −𝑏 𝑛 𝑢(−𝑛 − 1). 𝑒 −𝑗𝑤𝑛

𝑛=−∝
1
=
1 − 𝑒 −𝑗𝑤
 UNIT III - DISCRETE FOURIER TRANSFORM

1. Draw the graph of a 4 point DIT FFT butterfly structure for DFT. [May 2016][May
2015][Dec 2013]

𝑊40 =1; 𝑊41 =e-j2π/4 = -j

2. What are the applications of FFT algorithms? [May 2016]

Linear filtering
Correlation analysis
Power spectrum analysis
Frequency analysis

3. Calculate % saving in computing through radix 2 DFT algorithm of DFT coefficients.

Assume N=256. [Dec 2015]
DFT
The number of complex multiplications required using direct computation is
N2=2562=65536
The number of complex addition required using direct computation is
N(N-1)=256(256-1)=65280
FFT
The number of complex multiplication required using FFT is
(N/2) log2 N=(256/2) log2 256=1024
The number of complex addition required in FFT is N log2 N=256 log 2 256=2048
65535
% Saving in Multiplication is 1024 ∗ 100 = 6400%
65280
% Saving in Addition is ∗ 100 = 3187%
2048

4. State the circular frequency shifting properties of DFT [Dec 2015][May 2014]
Let DFT{x (n)} =X (K)
Circular frequency shifting : DFT {x ((n − m))𝑁 } = e−j2πkm/N X(k)

5. Compute the DFT of the sequence x(n)={1, 1, 0, 0}. [May 2015]

DFT of x (n) is given by
𝑁−1
−𝑗2𝜋𝑘𝑛
𝑋(𝑘) = ∑ x(n). 𝑒 𝑁 ; Where k = 0, 1, 2 … . . N − 1
𝑛=0
 𝑊40 =1; 𝑊41 =e-j2π/4 = -j

Input S1 Output
1 1+0(1)=1 1+1(1)=2
0 1-0(1)=1 1+1(-j)=1-j
1 1+0(1)=1 1-1(1)=0
0 1-0(1)=1 1-(-j)=1+j
X(K)={2, 1-j , 0, 1+j}

6. What is zero padding? What are its uses? [Dec 2014]

Let the sequence x(n) has the 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 zero padding’s are
We can get better display of the frequency spectrum.
With zero padding, the DFT can be used in linear filtering.

7. State parsavel’s relation for DFT. [Dec 2014]

Let DFT{x1 (n)} =X1 (K), DFT{x2 (n)} =X 2(K)
𝑁−1 𝑁−1
1
∑ x1 (n) x2∗ (n) = ∑ X1 (k) X2∗ (k)
𝑁
𝑛=0 𝑛=0

8. Compare DIT radix – 2 FFT and DIF radix – 2 FFT [May 2014]
DIT radix – 2 FFT DIF radix – 2 FFT
The time domain sequence is decimated. The frequency domain sequence is
decimated.
When the input is in bit reversed order, the When the input is in bit normal order, the
output will be in normal order and vice output will be in bit reversed order and
versa. vice versa.
In each stage of computations, the phase In each stage of computations, the phase
factors are multiplied before add and factors are multiplied after add and
subtract operations. subtract operations.
The value of N should be expressed such The value of N should be expressed such
that N = 2m and this algorithm consists of m that N = 2m and this algorithm consists of m
stages of computations. stages of computations.
Total number of arithmetic operations is Total number of arithmetic operations is
NlogN complex additions and (N/2) logN NlogN complex additions and (N/2) logN
complex multiplications. complex multiplications.

9. In eight point DIT what is the gain of the signal path that goes from x(7) to X(2)?[Dec
2013]
Gain = 𝑤80 . 𝑤80 (−1). (−𝑗) = 𝑗
10. Find the discrete Fourier transform for x(n)=δ(n) [May 2013]
DFT of x (n) is given by
𝑁−1
−𝑗2𝜋𝑘𝑛
𝑋(𝑘) = ∑ 𝑥(𝑛). 𝑒 𝑁 ; Where K = 0, 1, 2 … . . N − 1
𝑛=0
δ(n)=1 for n=0;
δ(n)=0 for n≠0;
Then, the DFT of the sequence δ (n) is given by,
𝑁−1
−𝑗2𝜋𝑘𝑛 −𝑗2𝜋𝑘.0
𝑋(𝑘) = ∑ δ(n). 𝑒 𝑁 = δ(0). 𝑒 𝑁 =1
𝑛=0

11. Draw the basic butterfly flow graph for the computation in the DIF FFT algorithm. [May
2013]

𝑊40 =1; 𝑊41 =e-j2π/4 = -j

12. Define discrete Fourier transform pair for a discrete sequence. [Dec 2012]
𝑁−1
−𝑗2𝜋𝑘𝑛
𝑋(𝑘) = ∑ x(n). 𝑒 𝑁 ; Where k = 0, 1, 2 … . . N − 1
𝑛=0

𝑁−1
1 𝑗2𝜋𝑘𝑛
𝑥(𝑛) = ∑ X(k). 𝑒 𝑁 ; Where N = 0, 1, 2 … . . N − 1
𝑁
𝑛=0

13. Find the 4 point DFT of the sequence x(n)={1, 1}. [Dec 2012]
After zero padding x(n)={1, 1,0,0}.
DFT of x (n) is given by
𝑁−1
−𝑗2𝜋𝑘𝑛
𝑋(𝑘) = ∑ x(n). 𝑒 𝑁 ; Where k = 0, 1, 2 … . . N − 1
𝑛=0

𝑊40 =1; 𝑊41 =e-j2π/4 = -j

Input S1 Output
 1 1+0(1)=1 1+1(1)=2
0 1-0(1)=1 1+1(-j)=1-j
1 1+0(1)=1 1-1(1)=0
0 1-0(1)=1 1-(-j)=1+j
X(K)={2, 1-j , 0, 1+j}

14. The first five DFT value for N=8 is as follows X(k)={28, -4+j9.656, -4+4j, -4+j1.656, -4, ...}
compute rest of three DFT values
X(n)=X*(N-n)
X(5)=X*(8-5)=X*(3)= -4 - j1.656
X(6)=X*(8-6)=X*(2)= -4 - 4j
X(7)=X*(8-7)=X*(1)= -4 - j9.656

15. Compute 4 point IDFT for X(k)={2, 3+j, -4, 3-j)

𝑊40 =1; 𝑊41 =e-j2π/4 = -j

Input S1 Output
2 2+(-4)=-2 (-2)+( 6)=4
-4 2-(-4)=6 6 + (-2j)(-j)=4
3-j 3-j +(3+j)=6 (-2)-( 6)=-8
3+j 3-j -(3+j)=-2j 6 - (-2j)(-j)=8
x(n)={1,1,-2,2}

16. What is meant by radix-2 FFT?

The FFT algorithm is most efficient algorithm for 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 FFT algorithm.

17. In direct computation of N-point DFT of a sequence, how many multiplications and
additions are required? (Or) How is FFT faster? How many multiplications and additions
are required to compute N point DFT using radix-2 FFT? [Dec 2009, May 2010]
FFT is faster because it requires less number of complex multiplications and
Complex additions compared to direct computation of DFT.

Operation FFT DFT

𝑁
Complex multiplications 𝑙𝑜𝑔2 𝑁 𝑁2
2

Complex additions 𝑁 𝑙𝑜𝑔2 𝑁 N ( N - 1)

18. List any two properties of DFT.

Let DFT{x (n)} =X (K), DFT{x1 (n)} =X1 (K), DFT{x2 (n)} =X 2(K)
 Periodicity: X (K+N) =X (K) for all K.
 Linearity: DFT[a1 x1 (n)+a2 x2(n)]=a1 X1 (K)+a2 X2 (K)
  DFT of time reversed sequence: DFT[ x(N-n)]=X(N-K)
 Circular convolution :DFT[x1(n)*x2(n)]=X1(K) X2(K)

19. Compute the IDFT of Y(k)={1, 0, 1, 0)

𝑊40 =1; 𝑊41 =e-j2π/4 = -j

Input S1 Output
1 1+1=2 (2)+( 0)=2
1 1-1=0 0 + (0)(-j)=0
0 0+0=0 (2)-( 0)=2
0 0-0=0 0 - (0)(-j)=0
x(n)={0.5, 0, 0.5, 0}

20. State the circular time shifting and circular frequency shifting properties of DFT
Circular time reversal : DFT {x ((−n))𝑁 } = X((−k))𝑁 =X (N-k)
Circular frequency shifting : DFT {x ((n − m))𝑁 } = e−j2πkm/N X(k)

21. Distinguish between linear convolution and Circular Convolution.

S. No Linear Convolution Circular Convolution
1. If x(n) is a sequence of L number of If x(n) is a sequence of L number of
samples and h(n) with m number of samples and h(n) with m number of
samples, after convolution y(n) will samples, after convolution y(n) will
contain N = L + M – 1 samples. contain N = Max(L,M) samples
2. Linear convolution can be used to find Circular convolution can be used to
the response of a linear filter. find the response of a linear filter
3. Zero padding is not necessary to find Zero padding is necessary to find the
the response of a linear filter. response of a linear filter.

22. What is bit reversal?

When the binary representation of one number is the mirror image of the binary
representation of the other, then both the numbers are said to be in bit reversal order.
For example, in a three-bit system, binary equivalent of one and four are bit-reversed
values of each other, since the three-bit binary representation of one, 001, is the mirror
image of the three-bit binary representation of four, 100.
 UNIT IV DESIGN OF DIGITAL FILTERS
1. Obtain the cascade realization for the system function. [May 2016]

2. Mention the advantages of FIR filter over IIR filter. [May 2016]
1. FIR filter have exact linear phase.
2. FIR filters are always stable.
3. FIR filter can be realized in both recursive and recursive structures.
4. Excellent design methods are available for various kinds of FIR filter.
5. FIR filter are free of limit cycle oscillation, when implemented on a finite word
length digital system.

3. Define prewarping effect? Why it is employed? [Dec 2015][May 2014][Dec 2012][May

2012]
In bilinear transformation, the relation between analog and digital frequencies is
nonlinear. This non-linear relationship introduces distortion in frequency axis, when the
‘s’ plane is mapped into ‘z’ plane using bilinear transformation. This effect is known as
frequency warping. The pre-warping is performed as follows:
2 ω
Ω = T tan ( 2 ),
In above equation, Ω and ω are analog and digital frequencies respectively. T is
nothing but a sampling rate. Pre-warping is necessary to eliminate the effect of warping
on amplitude response.

4. The impulse response of analog filter is given in Figure. Let h(n)=h(nT), where T=1.
Determine the system function. [Dec 2015][Dec 2013]

h(n) = {0,1,2,3,4,5,4,3,2,1,0}
10

𝐻(𝑧) = ∑ ℎ(𝑛)𝑧 −𝑛
𝑛=0
= 𝑧 −1+2𝑧 −2 + 3𝑧 −3 + 4𝑧 −4 + 5𝑧 −5 + 4𝑧 −6 + 3𝑧 −7 +2𝑧 −8 + 1𝑧 −9
5. Comment on pass band and stop band characteristics of butter worth filter. [May 2015]
In butter worth filter the transfer function is monotonic in both pass band and
stop band.
2 2
6. Realize the following causal linear phase FIR filter function 𝐻(𝑧) = 3 + 𝑧 −1 + 3 𝑧 −2 [May
2015]
2 2
𝐻(𝑧) = 3 + 𝑧 −1 + 3 𝑧 −2
2
𝐻(𝑧) = (1 + 𝑧 −2 ) + 𝑧 −1
3
7. What are the properties of Chebyshev filter. [Dee 2014][May 2013]
 The magnitude response is equi-ripple in the pass band and monotonic in the stop
band or vive-versa.
 The Chebyshev type – I filters are all pole designs.
 All poles lie on the ellipse.

8. What are the advantages of FIR filter? [Dee 2014]

1. FIR filter have exact linear phase.
2. FIR filters are always stable.
3. FIR filter can be realized in both recursive and recursive structures.
4. Excellent design methods are available for various kinds of FIR filter.
5. FIR filter are free of limit cycle oscillation, when implemented on a finite word
length digital system.

9. Give hamming window function. [May 2014]

The weighing function for the hamming window is given by
2𝜋 n 𝑁−1 𝑁−1
WH (n) = {0.54+0.46cos (N−1) -( 2 )≤𝑛 ≤( 2 )
0 otherwise

1+0.8𝑧 −1
10. Is the given transfer function 𝐻(𝑧) = 1−0.9𝑧 −1 represents low pass filter or high pass
filter? [ Dec 2013]
1 + 0.8𝑧 −1
𝐻(𝑧) =
1 − 0.9𝑧 −1
Pole =-0.9, zero=0.8
Zero close to point at (1,0) and the pole is close to point at (-1,0). Therefore the
given transfer function belongs to high pass filter.

11. Name the two methods for digitizing the transfer function of an analog filter. [May 2013]
1. Impulse invariant method
2. Bilinear transformation

12. What is the need for employing window for designing FIR filter?[Dec 2012]
The windows are finite duration sequence used to modify the impulse response of the
FIR filters in order to reduce the ripples in the pass band and stop band and also to
achieve the desired transition from pass band to stop band.

13. What is Gibbs phenomenon?[May 2012]

One possible way of finding an FIR filter that approximates H d(e j)would be to
truncate the infinite Fourier series at n=  (N-1/2).Abrupt truncation of the series will
 lead to oscillation both in pass band and is stop band .This phenomenon is known as
Gibbs phenomenon.

14. Distinguish between FIR and IIR filters. [May 2012]

S. No FIR filter IIR filter
These filters can be easily designed to These filters do not have linear
1
have perfectly linear phase phase.
FIR filters can be realized recursively IIR filters can be realized
2
and non-recursively recursively
Greater flexibility to control the shape Less flexibility, usually limited to
3
of their magnitude response kind of filters
Errors due to round-off noise are less
The round-off noise in IIR filters
4 severe in FIR filters, mainly because
are more
feedback is not used

15. Define condition for stability. [May 2012]

The left half plane of S plane should map into the inside of the unit circle in the z plane.
Thus a stable analog filter will be converted to a stable digital filter.

16. Compare the impulse invariant and bilinear transformations. [Dec 2011]
S.
Impulse Invariant Transformation Bilinear transformation
No
1 It is many-to-one mapping It is one-to-one mapping
The relation between analog and The relation between analog and digital
2
digital frequency is linear frequency is non-linear
To prevent the problem of aliasing
No problem of aliasing and so the
3 the analog filters should be band
analog filters need not be band limited
limited
The magnitude and phase response Due to the effect of warping, the phase
of analog filter can be preserved by response of analog filters cannot be
4
choosing low sampling time or high preserved. But the magnitude response
sampling frequency can be preserved by pre-warping

17. What is linear phase characteristic of an FIR filter? [Dec 2011]

The linear phase characteristic of an FIR filter is that the phase function should be a
linear function of w, which in turn requires constant phase and group delay.

18. “IIR filter does not have linear phase” – Justify [Dec 2015]
A physically realizable and stable IIR filter cannot have linear phase. A linear phase filter
must have a transfer function that satisfies the condition.
H(z)=±z-N H(z-1)
where z-N represents a delay. But above equation tells us that for every pole inside the
unit circle there is a pole outside the unit circle. Hence the filter would be unstable.
Therefore, a causal and stable IIR filter cannot have linear phase.

19. What is meant by bilinear transformation method of designing IIR filter? [May 2016]
[May 2015] [Dec 2013]
Bilinear transformation is a one to one mapping from the s-domain to the z-domain. That
is, the bilinear transformation is a conformal mapping that transforms the j Ω axis into
the unit circle in the z plane only once, thus avoiding the aliasing of frequency
components. Also the transformation of a stable analog filter result in a stable digital
filter as all the poles in the left half of the s plane are mapped inside the unit circle of the
z plane. The bilinear mapping is a one to one mapping and it is accomplished when
 2 1 − 𝑍 −1
𝑆= ( )
𝑇 1 + 𝑍 −1
20. Compare analog and digital filters. [Dec 2014]
Analog filter Digital filter
Constructed using active or passive Consists of elements like adder,
components and it is described by a Multiplier and delay units and it is
differential equation described by a difference equation
Frequency response can be changed by Frequency response can be changed by
changing the components changing the filter coefficients
It processes and generates analog output Processes and generates digital output
Output varies due to external conditions Not influenced by external conditions

21. Sketch the mapping of s-plane and z-plane in approximation of derivatives. [Dec 2014]
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 ½ centered at Z= ½ in the Z-
plane.
 The right half of S-plane maps into the region outside the circle of radius ½ in the
Z-plane.
 The j Ω-axis maps onto the perimeter of the circle of radius ½ in the Z-plane.

22. Draw the direct form realization of IIR system. [May 2016] [May 2015] [May 2014]
Direct form- I

Direct form- II

23. Mention the properties of Butterworth filter.

 The Butterworth filters are all pole designs.
 At the cut-off frequency Ωc the magnitude of normalized Butterworth filter is
1/√2.
 The filter order ‘n’ completely specifies the filter and as the value of N increases
the magnitude response approaches the ideal response.
24. What is meant by aliasing? [Dec 2012]
When the sampling frequency is less than twice of the highest frequency content of the
signal, then the aliasing is frequency domain takes place. In aliasing, the high frequencies
of the signal mix with lower frequencies and create distortion in frequency spectrum.

25. What are the limitations of impulse invariant method of designing IIR filter? [May 2012]
In this method the mapping from s plane to z plane is many to one. i.e. ,all the poles in
the s plane between the intervals (2k-1)π/T to (2k+1) π /T .Thus there are an infinite
number of poles that map to the same location in the z plane, producing an aliasing
effect. Due to spectrum aliasing the impulse invariant method is inappropriate in
designing high pass filters. That is why the impulse method is not much preferred in the
design of IIR filters other than low pass filter.

26. What are the properties of FIR filter? [May 2016] [May 2015][Dec 2014]
 FIR filters are particularly useful for applications where exact linear phase response is
required.
 The FIR filter is generally implemented in a non-recursive way which guarantees a
stable filter.
 The output depends on the present input and previous inputs only. It does not depend
on previous output.

27. What are the desirable characteristics of the windows? [May 2016]
 The central lobe of the frequency response of the window should contain most of the
energy and should be narrow.
 The highest side lobe level of the frequency response should be small.
 The side lobes of the frequency response should decrease in energy rapidly as w
tends to π

28. What is the necessary and sufficient condition for linear phase characteristics in FIR
filter? (Or) What is linear phase characteristic of an FIR filter? [Dec 2015]
The linear phase characteristic of an FIR filter is that the phase function should be a
linear function of w, which in turn requires constant phase and group delay.
Impulse response, h (n) = ±h (N-1-n)

29. State Gibb’s phenomenon [May 2016] [Dec 2013]

In Fourier series method of FIR filter design, the infinite duration impulse response is
truncated to finite duration impulse response. This abrupt truncation of impulse
response introduces oscillations in the pass band and stop band. This effect is known as
Gibb’s phenomenon (or Gibb’s Oscillation).

30. What are the characteristics of FIR filters designed using windows.
 The width of the transition band depends on the type of window.
 The width of the transition band can be made narrow by increasing the value of N
where N is the length of the window sequence.
 The attenuation in the stop band is fixed for a given window, except in case of Kaiser
window where it is variable.
 UNIT V - DIGITAL SIGNAL PROCESSORS
1. What are the advantages of Harvard architecture in a DSP processor? [Dec 2015]
Harvard architecture is capable of simultaneously reading and instruction code and
reading or writing a memory or peripheral as port of the execution of the previous
instruction.

2. How is a DSP processor applicable for motor control applications? [Dec 2015]
The motor control applications are controlled by the DSP processor by connection the
appliance through relay or opto-couplers.

3. How do digital signal processors differ from other processor? [May 2015]
The digital signal processors are microprocessor specially designed for efficient
implementation of digital signal processing system.

4. State any two applications of DSP. [May 2015]

Telecommunication – echo cancellation in telephone networks.
Instrumentation and control – Spectrum analysis, Digital filters etc.,
Speech processing – speech analysis methods used in automatic speech recognition.

5. What are the different stages in pipelining? [Dec 2014]

 Fetch
 Decode
 Read
 Execute

6. List the various registers used with ARAU of DSP processor? [Dec 2014][May 2014]
The ARAU contains eight 16 bit auxiliary registers AR0- AR7, A 3 bit Auxiliary
register Pointer (ARP), a 16 bit index register and a 16 bit auxiliary register compare
register(ARCR).

7. What are the different buses of TMS 320C54x processor and list their functions? [May
2014]
PB : Program bus and PAM: Program address bus
Program memory bus to read opcode and immediate operant.
CB : C bus and CAB : C address bus
DB : D bus and DAB : D address bus
Two independent data memory buses to read two data simultaneously from
memory.
EB : E bus and EAB : E address bus
Data memory buses to write data in data memory.

8. Mention one important feature of Harvard architecture. [May 2013]

The Harvard architecture has two memory blocks to store code and data separately and
the two memory blocks are connected to CPU by separate buses for simultaneous access
of code and data.

9. What is the advantage of pipelining? [May 2013]

Number of instructions can be executed in parallel, hence speed is high.

10. What is pipelining? What are the different stages in pipelining? [Dec 2012][Dec 2011]
Pipelining is a process by breaking down its instructions into a series of discrete pipeline
stages which can be completed in sequence by specialized hardware.
Different stages in pipelining are
Fetch, Decode, Read and Execute.
11. What is the function of parallel logic unit in DSP processor? [Dec 2012]
The parallel logic unit is an additional logic unit that permits logic operations without
affecting accumulator or product register. It performs Boolean operation or bit
manipulations. It can set, clear or toggle bits in the status register, control register and in
any data memory location.

12. Give the special features of DSP processors. [Dec 2011]

Harvard architecture
VLIW Architecture
Multiplier accumulate unit
Pipelining

13. Write short notes on general purpose DSP processors

General-purpose digital signal processors are basically high speed microprocessors with
hard ware architecture and instruction set optimized for DSP operations. These
processors make extensive use of parallelism, Harvard architecture, pipelining and
dedicated hardware whenever possible to perform time consuming operations

14. Write notes on special purpose DSP processors.

There are two types of special; purpose hardware.
(i) Hardware designed for efficient execution of specific DSP algorithms such as digital
filter, FFT.
(ii) Hardware designed for specific applications, for example telecommunication, digital
audio.
15. What about of Harvard architecture?
The principal feature of Harvard architecture is that the program and the data memories
lie in t o separate spaces, permitting full overlap of instruction fetch and execution.

16. What are the types of MAC is available?

There are two types MAC’S available
 Dedicated & integrated
 Separate multiplier and integrated unit

17. What is meant by pipeline technique?

The pipeline technique is used to allow overall instruction executions to overlap. That is
where all four phases operate in parallel. By adapting this technique, execution speed is
increased.

18. What are four phases available in pipeline technique?

The four phases are
(i) Fetch
(ii) Decode
(iii)Read
(iv) Execution

19. Write down the name of the addressing modes.

 Direct addressing.
 Indirect addressing.
 Bit-reversed addressing.
 Immediate addressing.
 Short immediate addressing.
 Long immediate addressing.
 Circular addressing
20. What are the instructions used for block transfer in C5X Processors?
The BLDD, BLDP and BLPD instructions use the BMAR to point at the source or
destination space of a block move. The MADD and MADS also use the BMAR to address
an operand in program memory for a multiply accumulator operation

21. What is meant by auxiliary register file?

The auxiliary register file contains eight memory-mapped auxiliary registers (AR0-AR7),
which can be used for indirect addressing of the data memory or for temporary data
storage.

22. Write the name of various part of C5X hardware.

 Central arithmetic logic unit (CALU)
 Parallel logic unit (PLU)
 Auxiliary register arithmetic unit (ARAU)
 Memory-mapped registers.
 Program controller.