Digital Signal Processing

2Marks

1. What is a continuous and discrete time signal?

Ans:
Continuous time signal: A signal x(t) is said to be continuous
if it is defined for all time t. Continuous time signal arise
naturally when a physical waveform such as acoustics wave or
light wave is converted into a electrical signal. This is effected
by means of transducer.(e.g.) microphone, photocell.
Discrete time signal: A discrete time signal is defined only
at discrete instants of time. The independent variable has
discrete values only, which are uniformly spaced. A discrete
time signal is often derived from the continuous time signal
by sampling it at a uniform rate.

2. Give the classification of signals?

Ans:
Continuous-time and discrete time signals
Even and odd signals
Periodic signals and non-periodic signals
Deterministic signal and Random signal
Energy and Power signal

3. What are the types of systems?

Ans:
Continuous time and
discrete time systems
Linear and Non-linear
systems
Causal and
Non-causal
systems
Static and
Dynamic
systems
Time varying and time in-varying systems
Distributive parameters and
Lumped parameters systems
Stable and Un-stable systems.
4. What are even and odd signals?
Ans:
Even signal: continuous time signal x(t) is said to be even
if it satisfies the condition x(t)=x(-t) for all values of t.
Odd signal: he signal x(t) is said to be odd if it satisfies the
condition x(-t)=-x(t) for all t. In other words even signal is
symmetric about the time origin or the vertical axis, but odd
signals are anti-symmetric about the vertical axis

5. What are deterministic and random signals?

Ans:
Deterministic Signal: deterministic signal is a signal about
which there is no certainty with respect to its value at any
time. Accordingly we find that deterministic signals may be
modeled as completely specified functions of time.
Random signal: random signal is a signal about which there is
uncertainty before its actual occurrence. Such signal may be
viewed as group of signals with each signal in the ensemble
having different wave forms
.(e.g.) The noise developed in a television or radio amplifier
is an example for random signal.

6. What are energy and power signal?

Ans:
Energy signal: signal is referred as an energy signal, if and
only if the total energy of the signal satisfies the condition
0<E<∞. The total energy of the continuous time signal x(t) is
given as
E=limT→∞∫x2 (t)dt, integration limit from –T/2 to +T/2
Power signal: signal is said to be powered signal if it
satisfies the condition 0<P<∞. The average power of a
continuous time signal is given by
P=limT→∞1/T∫x2(t)dt, integration limit is from-T/2 to +T/2

7. What are the operations performed on a signal?

Ans:
Operations performed on dependent variables:
Amplitude scaling: y (t) =cx (t), where c is the scaling factor,
x(t) is the continuous time signal.
Addition: y
(t)=x1(t)
+x2(t)
Multiplicati
on y
(t)=x1(t)x2(
t)
Differentiat
ion: y
(t)=d/dt x(t)
Integration
(t) =∫x(t)dt

Operations performed on independent variables

Time shifting
Amplitude scaling
Time reversal

8. What are elementary signals and name them?

Ans:
The elementary signals serve as a building block for the
construction of more complex signals. They are also
important in their own right, in that they may be used to
model many physical signals that occur in nature.
There are five elementary signals. They are as follows
Unit
step
func
tion
Unit
imp
ulse
func
tion
Ramp function
Exponential function
Sinusoidal function

9. What are the properties of a system?

Ans:
Stability: A system is said to be stable if the input x(t)
satisfies the condition(t) ≤Mx<∞ and the out put satisfies the
condition y(t) ≤My<∞ for all t.
Memory: A system is said to be memory if the output signal
depends on the present and the past inputs.
Invertibility: A system is said to be invertible if the input of
the system con be recovered from the system output.
Time invariance: A system is said to be time invariant if a
time delay or advance of the input signal leads to an
identical time shift in the output signal.
Linearity: A system is said to be linear if it satisfies
the super position principle
i.e.) R(ax1(t)+bx2(t))=ax1(t)
+bx2(t)

10.What is memory system and memory less system?

Ans:
A system is said to be memory system if its output signal at
any time depends on the past values of the input signal.
circuit with inductors capacitors are examples of memory
system..
A system is said to be memory less system if the output at
any time depends on the present values of the input signal.
An electronic circuit with resistors is an example for
memory less system.

11.What is an invertible system?

Ans:
A system is said to be invertible system if the input of the
system can be recovered from the system output. The set of
operations needed to recover the input as the second system
connected in cascade with the given system such that the
output signal of the second system is equal to the input signal
applied to the system.
H-1{y(t)}=H-1{H{x(t)}}.
12. What are time invariant systems?
Ans:
A system is said to be time invariant system if a time delay or
advance of the input signal leads to an idenditical shift in the
output signal. This implies that a time invariant system
responds idenditically no matter when the input signal is
applied. It also satisfies the condition
R{x(n-k)}=y(n-k).

13. Is a discrete time signal described by the input output

relation y[n]= rnx[n] time invariant.

Ans:
A signal is said to be time invariant if R{x[n-k]}= y[n-k]
R{x[n-k]}=R(x[n]) / x[n]→x[n-k]
=rnx [n-k] ---------------- (1)
y[n-k]=y[n] / n→n-k
=rn-kx [n-k] ------------------- (2)
Equations (1)≠Equation(2)
Hence the signal is time variant.

14. Show that the discrete time system described by the

input-output relationship y[n]
=nx[n] is linear?
Ans:
For a sys to be linear R{a1x1[n]+b1x2[n]}=a1y1[n]
+b1y2[n]
L.H.S:R{ a1x1[n]+b1x2[n] }=R{x[n]} /x[n] → a1x1[n]
+b1x2[n]
= a1 nx1[n]+b1 nx2[n]
------------------- (1)
R.H.S: a1y1[n]+b1y2[n]= a1 nx1[n]+b1 (2
nx2[n]-------------------- )
Equation(1)=Equation(2)
Hence the system is linear

15. What is SISO system and MIMO system?

Ans:
A control system with single input and single output is
referred to as single input single output system. When the
number of plant inputs or the number of plant outputs is
more than one the system is referred to as multiple input
output system. In both the case, the controller may be in the
form of a digital computer or microprocessor in which we
can speak of the digital control systems.
16. What is the output of the system with system function
H1 and H2 when connected in cascade and parallel?
Ans:
When the system with input x(t) is connected in cascade
with the system H1 and H2 the output of the system is
y(t)=H2{H1{x(t)}}
When the system is connected in parallel the output
of the system is given
by y(t)=H1x1(t)
+H2x2(t).

17.What do you mean by periodic and non-periodic

signals?
A signal is
said
to be
perio
dic if
x(n+
N)=x(
n)
Where N is the time period.
A signal is said
to be non-
periodic if
x(n+N)≠x(n) .

18.Determine the convolution sum of two

sequences x(n) = {3, 2, 1, 2} and h(n) = {1,
2, 1, 2}

Ans: y(n) = {3,8,8,12,9,4,4}

19.Find the convolution of the signals
x(n) = 1 n=-2,0,1
= 2 n=-1
elsewher
= 0 e.
Ans: y(n) = {1,1,0,1,-2,0,-1}

20.Detemine the solution of the difference equation

y(n) = 5/6 y(n-1) – 1/6 y(n-2) + x(n) for x(n) = 2n u(n)
Ans: y(n) = -(1/2)n u(n) + 2/3(1/3)n u(n)+8/5 2nu(n)
21.Determine the response y(n), n>=0 of the system
described by the second order difference equation
y(n) – 4y(n-1) + 4y(n-2) = x(n) – x(n-1) when the input
is x(n) = (-1)n u(n) and the initial condition are y(-1) = y(-
2)=1.
Ans:y(n) = (7/9-5/3n)2n u(n) +2/9(-1)n u(n)

22. Differentiate DTFT and DFT

DTFT output is continuous in time where as DFT
output is Discrete in time.

23.Differentiate between DIT and DIF algorithm

DIT – Time is decimated and input is bi reversed
format output in natural order DIF – Frequency is
decimated and input is natural order output is bit
reversed format.

24. How many stages are there for 8 point DFT

8

25 How many multiplication terms are required for

doing DFT by expressional method and FFT method
expression –n2 FFT N /2 log N

26.What are the different types of signal representations?

a. Graphical representation
b. Functional representations
c. Tabular representation
d. Sequence representation

27. Distinguish analog and digital filters

Analog digital
Consists of elements like
Constructed using active or adder,
passive components and it subtractor and delay units and
is it is
described by a difference
described by a differential equation
 equation
Frequency response can be Frequency response can be
changed by changing the changed by changing the filter
components coefficients
It processes and generates Processes and generates digital
analog output output
Output varies due to
external Not influenced by external
conditions conditions

28.Write the expression for order of Butterworth filter?

The expression is N=log (λ /€) 1/2/log (1/k) ½

29.Write the expression for the order of chebyshev filter?

N=cosh-1(λ /e)/cosh-1(1/k)

30.Write the various frequency transformations in analog

domain?
LPF to LPF:s=s/ c
LPF to HPF:s= c/s
LPF to BPF:s=s2xlxu/s(xu-xl)
LPF to BSF:s=s(xu-xl)?s2=xlxu. X=

31.Write the steps in designing chebyshev filter?

1. Find the order of the filter.
2. Find the value of major and minor axis. λ
3. Calculate the poles.
4. Find the denominator function using the above poles.
5. The numerator polynomial value depends on the value of
n.
If n is odd: put s=0 in the denominator polynomial.
If n is even put s=0 and divide it by (1+e2)1/2
32.Write down the steps for designing a Butterworth
filter?

1. From the given specifications find the order of

the filter
2. find the transfer function from the value of N
 3. Find c
4 find the transfer function ha(s) for the above value of c by su
s by that value.

33.State the equation for finding the poles in

chebyshev filter sk=acos¢k+jbsin
¢k,where ¢k=∏/2+(2k-1)/2n)∏

34.State the steps to design digital IIR filter using bilinear method
Substitute s by 2/T (z-1/z+1), where T=2/Ώ (tan (w/2) in h(s) to
get h (z)

35. What is warping effect?

For smaller values of w there exist linear relationship between

w and .but for larger values of w the relationship is nonlinear.
This introduces distortion in the frequency axis. This effect
compresses the magnitude and phase response. This effect is
called warping effect

36. Write a note on pre warping.

The effect of the non linear compression at high frequencies can

be compensated. When the desired magnitude response is
piecewise constant over frequency, this compression can be
compensated by introducing a suitable rescaling or prewar ping
the critical frequencies.

37.Give the bilinear transform equation between s

plane and z plane s=2/T (z-1/z+1)

38.Why impulse invariant method is not preferred in the design

of IIR filters other than low pass filter?
In this method the mapping from s plane to z plane is many to
one. Thus there ire an infinite number of poles that map to the
same location in the z plane, producing an aliasing effect. It is
inappropriate in designing high pass filters. Therefore this
method is not much preferred.

39.By impulse invariant method obtain the digital filter

transfer function and the differential equation of the analog
filter h(s) =1/s+1
H (z) =1/1-e-Tz-1

Y/x(s) =1/s+1
Cross multiplying and taking inverse lap lace we get,
D/dt(y(t)+y(t)=x(t)

40. What is meant by impulse invariant method?

In this method of digitizing an analog filter, the impulse
response of the resulting digital filter is a sampled version of the
impulse response of the analog filter. For e.g. if the transfer
function is of the form, 1/s-p, then
H (z) =1/1-e-pTz-1
41.What do you understand by backward difference?
One of the simplest methods of converting analog to
digital filter is to approximate the differential equation by
an equivalent difference equation. d/dt(y(t)/t=nT=(y(nT)-
y(nT-T))/T

42.What are the properties of chebyshev filter?

1. The magnitude response of the chebyshev filter exhibits ripple
either in the stop band or the pass band.
2. The poles of this filter lies on the ellipse

43.Give the Butterworth filter transfer function and its

magnitude characteristics for different orders of filter.
The transfer function of the Butterworth
filter is given by H (jΏ) =1/1+j (Ώ/Ώc) N

44.Give the magnitude function of Butterworth filter.

The magnitude function of
Butterworth filter is |h(jΏ)=1/[1+
(Ώ/Ώc)2N]1/2 ,N=1,2,3,4,….

45. Give the equation for the order N, major, minor axis of
an ellipse in case of chebyshev filter?
The
1
order is given by N=cosh-1(((10.1αp)-1/10.1αs-1)1/2))/cosh-
Ώs/Ώp
A= (µ1/N-µ-1/N)/2Ωp
B=Ωp (µ1/N+ µ-1/N)/2
46. Give the expression for poles and zeroes of a chebyshev type 2
filters
The zeroes of chebyshev type 2 filter SK=jΏs/sinkФk, k=1….N
The poles of this
filter xk+jyk xk=
Ώsσk/ Ώs2+σk2
yk=ΏsΏk/ Ώs2+σk2 σk=acosФk
47. How can you design a digital filter from analog filter?
Digital filter can de designed from analog filter using the
following methods
1. Approximation of derivatives
2. Impulse invariant method
3. Bilinear transformation

48. write down bilinear

transformation.
s=2/T (z-1/z+1)

49.List the Butterworth polynomial for various orders.

N Denominator polynomial
1 S+1
2 S2+.707s+1
3 (s+1)(s2+s+1)
4 (s2+.7653s+1)(s2+1.84s+1)
5 (s+1)(s2+.6183s+1)(s2+1.618s+1)
6 (s2+1.93s+1)(s2+.707s+1)(s2+.5s+1)
7 (s+1)(s2+1.809s+1)(s2+1.24s+1)(s2+.48s+1)

50. Differentiate Butterworth and Chebyshev filter.

Butterworth dampimg factor 1.44 chebyshev 1.06

Butterworth flat responsedamped response.

51. What is filter?

Filter is a frequency selective device ,which amplify
particular range of frequencies and attenuate
particular range of frequencies.

52.What are the types of digital filter according to their impulse

 response?
IIR(Infinite impulse
response )filter
FIR(Finite Impulse
Response)filter.

53.How phase distortion and delay distortion are introduced?

The phase distortion is introduced when the phase
characteristics of a filter is nonlinear with in the desired
frequency band.
The delay distortion is introduced when the delay is not
constant with in the desired frequency band.

54. what is mean by FIR filter?

The filter designed by selecting finite number of samples
of impulse response (h(n) obtained from inverse fourier
transform of desired frequency response H(w)) are called
FIR filters

55.Write the steps involved in FIR filter design

Choose the desired frequency
response Hd(w) Take the inverse
fourier transform and obtain Hd(n)
Convert the infinite duration
sequence Hd(n) to h(n) Take Z
transform of h(n) to get H(Z)

56.What are advantages of FIR filter?

Linear phase FIR filter can be easily designed .Efficient
realization of FIR filter exists as both recursive and non-
recursive structures. FIR filter realized non-recursively stable.
The round off noise can be made small in non recursive
realization of FIR filter.

57. What are the disadvantages of FIR FILTER

The duration of impulse response should be large to realize
sharp cutoff filters. The non integral delay can lead to
problems in some signal processing
applications.

58.What is the necessary and sufficient condition for the linear

phase characteristic of a FIR filter?
The phase function should be a linear function of w, which
 inturn requires constant group delay and phase delay.

59.List the well known design technique for linear phase FIR filter
design?
Fourier series method and window method
Frequency sampling method.
Optimal filter design method.

60. Define IIR filter?

The filter designed by considering all the infinite samples of
impulse response are called IIR filter.

61. For what kind of application , the antisymmetrical impulse

response can be used?
The ant symmetrical impulse response can be used to design
Hilbert transforms and differentiators.

62. For what kind of application , the symmetrical impulse

response can be used?
The impulse response ,which is symmetric having odd number
of samples can be used to design all types of filters ,i.e ,
lowpass,highpass,bandpass and band reject. The symmetric
impulse response having even number of samples can be used to
design lowpass and bandpass filter.

63.What is the reason that FIR filter is always stable?

FIR filter is always stable because all its poles are at the origin.

64.What condition on the FIR sequence h(n) are to be imposed n

order that this filter can be called a liner phase filter?
The conditions are
(i) Symmetric condition h(n)=h(N-1-n)
(ii) Antisymmetric condition h(n)=-h(N-1-
n)

65.Under what conditions a finite duration sequence h(n) will

yield constant group delay in its frequency response
characteristics and not the phase delay?
If the impulse response is anti symmetrical ,satisfying the
condition
 H(n)=-h(N-1-n)
The frequency response of FIR filter will have constant
group delay and not the phase delay .

66. State the condition for a digital filter to be causal and stable?
A digital filter is causal if its impluse response h(n)=0 for n<0.
A digital filter is stable if its impulse response is absolutely
summable ,i.e,
∞
∑
h(n)
<∞
n=-
∞

67.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.

68.When cascade from realization is preferred in FIR filters?

The cascade from realization is preferred when complex
zeros with absolute magnitude less than one.

69. What are the disadvantage 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 .
70.What are Gibbs oscillations?

One possible way of finding an FIR filter that approximates

H(ejω)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.

71. What are the desirable characteristics of the windows?

The desirable charaterstics of the window are
 1.The central lobe of the frequency response of the
window should contain most of the energy and should
be narrow.
2.The highest side lobe level of the frequency
response should be small. 3.The sides lobes of the
frequency response should decrease in energy
rapidly as ω tends to π .

For example, the binary number 01.1100 has the value 1.75 in decimal.
72. Calculate DFT of the sequence x(n)={1,1,2,2}
N-1
x(k)= x(n)e-j2ðnk/N K=0,1,2,3,…N-1
n=0
3
x(k)= x(n)e-j2ðnk/N K=0,1,2,3
n=0
N=4
= x(0)+x(1)e-jkð/2+x(2)e-jkð+x(3)e-j3kð/2
= 1+ e-jkð/2-2e-jkð-2e-j3kð/2 K=0,1,2,3
73. List any four properties of DFT
a. Periodicity
b. Linearity
c. Time reversal
d. Circular time shift

74.Why FFT is needed?

FFT is needed to compute DFT with reduced number of
calculations. DFT is required for spectrum analysis and filtering
operations on the
signals using digital computers.
75. Calculate the number of multiplications needed in the
calculation of DFT
and FFT with 64 point sequence.
The number of complex multiplications required using direct
computation
is N2 = 642 = 4096.
The number of complex multiplications required using FFT is
N log2 N = 64 log264 = 192
76. What is the main advantage of FFT?
FFT reduces the computation time required to compute discrete fourier
transform.
77. Calculate the number of multiplications needed in the
calculation of DFT using FFT with 32 point sequence.
The number of complex multiplications required using FFT is
N log2N = 32 log232 = 80
78. What is FFT?
FFT is a method for computing the DFT with reduced number of
calculations using symmetry and periodicity properties of twiddle
factor WkN .The computational efficiency is achieved by decomposing
of an N-point DFT into successively smaller DFTs to increase the
speed of computation.
79. How many multiplications and additions are required to
compute N-point DFT using radix-2 FFT?
N log2N multiplications and N log2N additions

80.What is the objective of spectrum estimation?

The main objective of spectrum estimation is the determination
of the power spectral density of a random process. The estimated PSD
provides information about the structure of the random process which
can be used for modeling, prediction or filtering of the deserved
process.

81.List out the addressing modes supported by C5X processors?

1. Direct addressing
2. Indirect addressing
3. Immediate addressing
4. Dedicated-register addressing
5. Memory-mapped register addressing
6. Circular addressing

82.what is meant by block floating point representation? What are

its advantages?
In block point arithmetic the set of signals to be handled is divided
into blocks. Each block have the same value for the exponent. The
arithmetic operations with in the block uses fixed point arithmetic &
only one exponent per block is stored thus saving memory. This
representation of numbers is more suitable in certain FFT flow graph
& in digital audio applications.
83.what are the advantages of floating point arithmetic?
1. Large dynamic range
2. Over flow in floating point representation is unlike.

84.what are the three-quantization errors to finite word length

registers in digital filters? 1. Input quantization error 2. Coefficient
quantization error 3. Product quantization
error

85.How the multiplication & addition are carried out in floating

point arithmetic?
In floating point arithmetic, multiplication are carried out as follows,
Let f1 = M1*2c1 and f2 = M2*2c2. Then f3 = f1*f2 = (M1*M2)
2(c1+c2)
That is, mantissa is multiplied using fixed-point arithmetic and
the exponents are added.
The sum of two floating-point number is carried out by shifting the
bits of the mantissa of the smaller number to the right until the
exponents of the two numbers are equal and then adding the
mantissas.

86.What do you understand by input quantization error?

In digital signal processing, the continuous time input signals are
converted into digital using a b-bit ACD. The representation of
continuous signal amplitude by a fixed digit produce an error, which is
known as input quantization error.

87.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
88.what is the relationship between truncation error e and the bits
b for representing a decimal into binary?
For a 2's complement representation, the error due to truncation for
both positive and negative values of x is 0>=xt-x>-2-b
Where b is the number of bits and xt is the truncated value of x.
The equation holds good for both sign magnitude, 1's complement
if x>0
If x<0, then for sign magnitude and for 1's complement the
truncation error satisfies.

89.what is meant rounding? Discuss its effect on all types of

number representation?
Rounding a number to b bits is accomplished by choosing the
rounded result as the b bit number closest to the original number
unrounded.
For fixed point arithmetic, the error made by rounding a number to b bits satisfy the
inequality
-2-b 2-b
-----<=xt-x<=
--------
2 2

for all three types of number systems, i.e., 2's complement, 1's
complement & sign magnitude.

For floating point number the error made by rounding a number to

b bits satisfy the inequality
-2-b<=E<=2-b where E=xt-x
--------
x
90.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.
91.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. Some times the pole locations will be changed in
such a way that the system may drive into instability.

92.which realization is less sensitive to the process of quantization?

Cascade form.

93.what is meant by quantization step size?

Let us assume a sinusoidal signal varying between +1 and -1
having a dynamic range 2. If the ADC used to convert the sinusoidal
signal employs b+1 bits including sign bit, the number of levels
available for quantizing x(n) is 2b+1. Thus the interval between
successive levels

q= 2 =2-b
--------
2b+1
Where q is known as quantization step size.

94.How would you relate the steady-state noise power due to

quantization and the b bits representing the binary sequence?
Steady state noise power
Where b is the number of bits excluding sign bit.

95.what is overflow oscillation?

The addition of two fixed-point arithmetic numbers cause over
flow 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
over flow oscillations.

96.what are the methods used to prevent overflow?

There are two methods used to
prevent overflow 1. Saturation
arithmetic 2. Scaling

97.what are the two kinds of limit cycle behavior in DSP?

1.zero input limit cycle
 oscillations
2.Overflow limit cycle
oscillations

98.What is meant by "dead band" of the filter

The limit cycle occur as a result of quantization effect in
multiplication. The amplitudes of the output during a limit cycle are
confined to a range of values called the dead band of the filter.

99.Explain briefly the need for scaling in the digital filter

implementation.
To prevent overflow, the signal level at certain points in the
digital filter must be scaled so that no overflow occurs in the
adder.

100.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

101.Define a system?
A system is a physical device or algorithm that performs an operation
on the signal

102.What is digital signal processing?

The dsp refers processing of signal by digital system.

103. What are the steps involved in digital signal processing?

a. Converting the analog signal to digital signal ,which is performed by
A/D converter
b. Processing the digital signal by digital systems.
c. Converting the digital output signal from the digital system to analog
signal by D/A converter.
104. What are the advantages of DSP?
a. The programme can be modified easily for better Performance.
b. Better accuracy can be achieved by using adaptive algorithm.
c. The digital signal can be easily stored and transported.
d. Digital systems are cheaper than analog equal lent.
105. Give some applications of DSP?
a. Speech processing
b. Communications
c. Biomedical
106. Write the difference equation governing the Nth order LTI
system.
NM
Y(n)=å ak y(n-k) +å bk x(n-k)
k=1 k=0
a. N is the order of the system
b. ak & bk are constant coefficients
c. y(n)&x(n) are output and input to the system
107. List the various methods of classifying discrete time systems?
a. Static and dynamic systems.
b. Time invariant and time variant
c. Linear and nonlinear
d. Causal and no causal
e. Stable and unstable
f. FIR and IIR systems
g. Recursive and non recursive systems
108. What are static and dynamic systems? Give examples?
A discrete time system is called static(memory less)if it’s output at any
instant n dependent on the input sample at the same time (but does not
depend on past or future samples).If the response depends on past or
future samples, then the system is called dynamic system.
Eg.y(n)=ax(n) static system
Y(n)=ax(n)+bx(n-1)
109. Define time invariant system?
A system is said to be time invariant if it’s input output characteristics
does not change with time. Let H be a system and H{X(n)}=Y(n).now
if H{X(nk)}=Y(n-k) then the system H is called time invariant.
110. What is linear and nonlinear systems?
If a system satisfies superposition and homogeneity principles then the
system is called linear otherwise it is called nonlinear If H is a
system,X1(n) and X2(n) are inputs a and b are constants then
H{aX1(n)+BX2(n)}=aH{X1(n)}+bH{X2(n)} then His linear.
111. What is a causal system give an example?
A system is said to be causal, if the output of the system at any time n
depends on present and past inputs ,but does not depend on future
inputs.
E.g.(n)=x(n)+x(n-1)
112. Define a stable system?
Any relaxed system is said to be bounded input bounded output stable
if and only if every bounded input yields a bounded output.
μ
å h(n)< μ where h(n)is impulse response of the system
n=-μ
113. What is LTI system?
A linear time invariant system is defined that a system obeys both
linearity and time invariant properties.
If a system satisfies superposition and homogeneity principles then the
system is called linear
A system is said to be time invariant if it’s input output characteristics
does not change with time.
114. What are FIR and IIR systems?
FIR (finite impulse response):this type of system has an impulse
response
which is zero outside the finite time interval eg. h(n)=0 for n<0 and
n>N
IIR (Infinite Impulse Response):An IIR system exhibits an impulse
response
of infinite duration.
115. State sampling theorem.
A band limited continuous time signal ,with higher frequency fc Hz can
be uniquely recovered from it’s samples provided that the sampling rate
F>2fc samples per second.
116. Show whether the system is linear?
Y(n)=n x(n)
H{aX1(n)+BX2(n)}=a H{X1(n)}+b H{X2(n)} then H is linear.
a H{X1(n)}+b H{X2(n)}=anx1(n)+bnx2(n) ------------(1)
H{aX1(n)+BX2(n)}= anx1(n)+bnx2(n) --------------(2)
(1)=(2) So the system is linear.
117. Show whether the system is linear?
Y(n)=nx2(n)
Since x2(n) term is present in the system which implies non linearity in
to the system. Therefore the system is nonlinear.
118. Determine if the following system is time invariant or time
variant?
Y(n)=x(n)+x(n-1)
If the input is delayed by k units in time we have y(n,k)=H{x(n-
k)}=x(nk)+
x(n-k-1)
If we delay the output by k units then y(n-k)= x(n-k)+x(n-k-1) So the
system is time invariant.
119. Determine if the system described by the following equation is
causal or not?
Y(n)=x(n2)
For n = -1
Y(-1)=x(1)
For n = 2 Y(2) = x(4)
Therefore the output of the system depends on future input and hence
the
system is non causal.
120. Define unit sample response of a system and what is it’s
significance?
The response of a system denoted as h(n),obtained from a discrete time
system when the input signal is a unit sample sequence is known as unit
sample response.
121. Define z transform?
The Z transform of a discrete time signal x(n) is defined as
X(z) = x(n)z-n
n= -
where z is a complex variable. In polar form z=re-jw
122. What is meant by ROC?
The region of convergence (ROC) is defined as the set of all values of z
for which x(z) converges.
123. Explain about the roc of causal and anti-causal infinite
sequences?
For causal system the roc is exterior to the circle of radius r.
For anti causal system it is interior to the circle of radius r.
124. Explain about the roc of causal and anti causal finite
sequences
For causal system the roc is entire z plane except z=0.
For anti causal system it is entire z plane except z= .
125. What are the properties of roc?
a. The roc is a ring or disk in the z plane centered at the origin.
b. The roc cannot contain any pole.
c. The roc must be a connected region
d. The roc of an LTI stable system contains the unit circle.
126. Explain the linearity property of the z transform
If z{x1(n)}=x1(z) and z{x2(n)}=x2(z) then z{ax1(n)+bx2(n)}=ax1(z)
+bx2(z)
a&b are constants.
127. State the time shifting property of the z transform
If z{x(n)}=x(z) then z{x(n-k)}=z-kx(z)
128. State the scaling property of the z transform
If z{x(n)}=x(z) then z{anx(n)}=x(a-1z)
129. State the time reversal property of the z transform
If z{x(n)}=x(z) then z{x(-n)}=x(z-1)
130. Explain convolution property of the z transform
If z{x(n)}=x(z) & z{h(n)}=h(z) then z {x(n)*h(n)}=x(z)h(z)
131. Explain the multiplication property of z transform
If z{x(n)}=x(z) & z{h(n)}=h(z) then
z {x(n) h(n)}= 1/2ðj c x(ã)h(z/ã) ã-1d ã
132. State final value theorem of z tramsform
If x(n) is causal z{x(n)}=x(z), where the roc of x(z) includes, but
it is not necessary to confined to - z ->1 and (z-1)x(z)has no pole on
or outside the unit circle then
x( ) = lt (z-1)
x(z) z› 1
133. Define system function?
The ratio between z transform of out put signal y(z) to z
transform of input signal x(z) is called system function of the
particular system
Y(z)
H(z)= ---------
X(Z)
134. What are the conditions of stability of a
 causal system ? All the poles of the system
are with in the unit circle.
The sum of impulse response for all values of n is bounded

h(n) <
n=-
135. Determine z transform and roc of the signal {1,2,3,4}

-
X(z) = x(n)z-n
n =-
3
= x(n)z -n =x(0)z-0+x(1)z-1+x(2)z-2+x(3)z-3
n=0
= 1z-0+2z-1+3z-2+4z-3
roc is entire z plane except z = 0
136. Determine z transform and roc of the signal {1,2,3,4}

X(z) = x(n)z-n
n=-

0
X(z)= x(n)z -n = x(-3)z3+x(-2)z2+x(-1)z1+x(0)
n=-3
= 4+3z1+2z2+1z3
ROC is entire z plane except z=

137. Determine z transform and roc of the signal {1,2,3,4}

X(z)= x(n) z-n

n=-
2
X(z)= x(n)z -n = x(-1)z1 + x(0)z0 + x(1)z-1 +
x(2)z -2 n=-1
= 1z1+2+3z-1+4z-2
ROC is entire z plane except z= , 0
138. Find the z transform and roc of anu(n)
X(z)
= x(n)z-n
n=-
roc- z - >
n -n -1
X(z) = a z =1/(1-az ) a.
n=0
139. Find the z transform and roc of -anu(-n-1)
X(z)
= x(n)z-n
n=-
-1
X(z)=
- a n z-n
n= -
-
=-(a = 1/(1-az-1) roc- z - <
1 n
z) a.
n=1

140. The z-transform of a sequence x(n) is x(z),what is the z

transform of nx(n)
If z{x(n)}=x(z) then z{nx(n)}=-zd(x(z))/dz

141. Find the z-transform of (a) A digital impulse (b) A digital

step. (a)Since x(n) is zero except for n = 0, where x(n) is 1,
we find x(z) = 1.
(b) Since x(n) is zero except for n 0, where x(n) is 1, we find

1
x(z) = Z-n =
1 – z-
1
n=0
142. What is the relationship between z-transform and DTFT?
The z-transform of x(n) is given by
x(z) x(n) Z-n ; where z = rej
ω
= ……………….. (1)
n=-
Substituting z in x(z) we get,
 x(z)
= x(n) r-ne-jωn ………………. (2)
n=-
The Fourier transform of x(n) is given by
x(ej ω ) = x(n) e-jωn ………………..(3)
n=-
Equation (2) and (3) are identical, when r
= 1.
In the z-plane this corresponds to the locus of points on the unit
circle - z - = 1 .
Hence X(ej ω ) is equalj to H(z) evaluated along the unit circle, or
X(e ) = x(z)-For X(e ω ) to exist, the ROC of x(z) must include
jω

the unit circle.

143. What are the different methods of evaluating inverse
z-transform? It can be evaluated using several
methods.
Long division method
Partial fraction expansion method
Residue method
Convolution method
144.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 x(k).The N point DFT of
a finite sequence x(n) of length L,(L<N) is defined as

N-1
x(k)= x(n)e-j2πnk/N K=0,1,2,3,…N-1

145. What are the different methods of evaluating inverse z-

transform?
It can be evaluated using several methods.
i. Long division method
ii. Partial fraction expansion method
iii. Residue method
iv. Convolution method

146. 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 x(k).The N point DFT of a finite
sequence x(n) of length L,(L<N) is defined as N-1
x(k)= x(n)e-j2ðnk/N K=0,1,2,3,…N-1
n=0
147. Define IDTFT
The IDTFT of the sequence of length N is defined as
N-1 X(n)=(1/N ) x(k)ej2ðnk/N n=0,1,2,3,…N-1
k=0
148. What is the drawback in DTFT?
The drawback in discrete time fourier transform is that it is continuous
function of w and cannot be processed by digital systems.

149.State periodicity property with respect to DFT.

If x(k) is N-point DFT of a finite duration sequence
x(n), then x(n+N) = x(n) for all n.
x(k+N) = x(k) for all k.
150.State periodicity property with respect to DFT.
If x1(k) and x2(k) are N-point DFTs of finite duration sequences x1(n)
and x2(n), then DFT [a x1(n) + b x2(n)] = a x1(k) + b x2(k), a, b are
constants
zj =
eω