UNIT I : CLASSIFICATION OF SIGNALS

1. Difference between DSP and ASP.

ASP? Input signal given to the system is analog. Ex R,C,L, OP-AMP etc.
DSP? Input signal given to the system is digital. Ex Digital Computer, Digital
Logic Circuits etc.
a. Compact and light in weight.
b. More accurate i.e less sensitive to environment changes and noise
c. Flexible, programmable and easily up-gradable
d. Easy and lasting storage capacity
e. Less cost.
2. Explain the block diagram of Digital system.
Analog
Analog
Signal
signal
Most of the signals generated are analog in nature. Hence these signals
are converted to digital form by the ADC. The DSP performs signal processing ope
rations like filtering, multiplication, transformation or amplification etc oper
ations over these digital signals. The digital output signal from the DSP is giv
en to the DAC to generate analog signal again.
3. What are single channel - multi-channel signals
Single channel signal ? signal is generated from single sensor or source. Ex. Sp
eech or voice signal.
Multi-channel signal ? signals are generated from multiple sensors or multiple s
ources Ex ECG signals.
Continuous time signals ? defined at any time instance
Discrete time signals ? defined only at sampling instances.
Continuous values signal ? signal amplitude takes on all possible values on a fi
nite or infinite range
Discrete values signal. ? signal takes values from a finite set of possible valu
es.
Analog signals ? Continuous time & continuous amplitude signals
Digital signals ? Discrete time & discrete amplitude signals.
Deterministic signal ? value can be evaluated at any time without certainty.
Random signal ? value can not be evaluated at any instant of time.
Periodic signal ? If x(n+N)= x(n) for all n where N is the fundamental perio
d of the signal. Else non-periodic signals.
Symmetrical(Even) ? if x(n) = x(-n)
Anti-symmetrical(Odd) ? x(-n) = -x(n)
Energy Signal ? Summation of magnitude squared values of x(n). The signa
l is called as energy signal if its energy if finite. A signal is called power s
ignal if its power is finite.
Ex: Energy of unit sample function is 1.
4. What is maximum range of discrete time frequencies & continuous time fre
quencies.
Discrete time frequencies = -1/2 to 1/2 cycles/sample or -? to +? rad/samples
Continuous time frequencies = - 8 to +8
5. Prove that discrete time signals are periodic only if frequency is ratio
nal. What is the condition for periodicity of DT signal.
6. CT periodic signals are converted into DT signals by sampling. DT signal
may not be periodic. Explain this statement with suitable example.
X(t) = sin(10) t ? CT signal (Periodic) Fs=1 t=nTS
X(n)= sin(10) n ? DT signal (Non-periodic).
This is because Discrete frequency is not rational.
7. The highest rate of oscillation is achieved when the discrete frequency
is
?. Explain this statement with suitable example.
8. Prove that any discrete time signal is represented as a combination of e
ven and odd signals with an example.
Even component of signal =[ x(n) + x(-n) ] / 2
Odd component of signal =[ x(n) - x(-n) ] / 2
Example: X(n)={1,2,1} Xe(n)={0.5,1,1,1,0.5} Xo(n)={-0.5,-1,0,1,0.5}
9. Explain the importance of unit sample signal.
Unit sample is given as input to the system H. Output of the system will
be h(n) called as unit impulse response. Once we know the unit impulse response
, we can find out the output of the same system for all type of inputs. (Linear
Convolution).
10. What are different test signals used In DSP.
Unit ramp, unit step and unit sample are three most used test signals in
DSP. Exponential and sine ways can also be used in DSP.
11. Which statement is correct?
8
? x (k) h(n k ) (1)
k= -8
8
? x (k) d(n k ) (2)
k= -8
12. What are Static or dynamic systems.
Static ? Output depends on input sample at same time.
Dynamic ? Output also depends upon past or future samples of input.
TIV ? If its IO characteristic does not change with shift of time.
Linear ? If it satisfies superposition theorem
Let x1(n) and x2(n) are two input sequences, then the system is said to be linea
r if and only if T[a1x1(n) + a2x2(n)]=a1T[x1(n)]+a2T[x2(n)] (Superposition Theor
em)
Causal ? If output of system depends only past and present inputs samples.
Non-causal ? If output of system also depends on future inputs.
Stable ? If every bounded input produces a bounded output.
Unstable ? If any bounded input produces an unbounded output.
13. How the discrete time signal is represented as weighted impulses.
Let X(n) = {2,5,2}. The signal x(n) can also be written as
X(n)= x(-1) d(n+1) + x(0) d(n) + x(1) d(n-1).
14. Explain linear convolution. Will it applicable for Non-linear systems or
 Time variant systems.
In linear convolution we decompose input signal into sum of elementary s
ignal. Now the elementary input signals are taken into account and individually
given to the system. Now using linearity property Whatever output response we ge
t for decomposed input signal, we simply add it & this will provide us total res
ponse of the system to any given input signal.
Linear Convolution states that
y(n) = x(n) * h(n)
8
y(n) = ? x (k) h(n k )
k= -8
15. What are various properties of linear convolution.
Commutative property: x(n) * h(n) = h(n) * x(n)
Associative property: [ x(n) * h1(n) ] * h2(n) = x(n) * [ h1(n) * h2(n
) ]
Distributive property: x(n) * [ h1(n) + h2(n) ] = x(n) * h1(n) + x(n)
* h2(n)
16. Explain when LSI system is causal.
LSI system is causal if and only if h(n) =0 for n<0.
17. Explain when LSI system is stable.
LSI system is stable if its unit sample response is absolutely summable.
8
? |h(k)| < 8
k=-8
18. How the LSI system is represented by constant coefficient difference equ
ation. (Generalized Difference equation)
Difference equation of the generalized LSI system is given as
N M
y(n)=-? ak y(n k)+? bk x(n k)
k=1 k=0
19. What is sampling process. Why it is necessary.
It is the process of converting continuous time signal into a discrete t
ime signal by taking samples of the continuous time signal at discrete time inst
ants.
20. What is sampling theorem. What is Nyquist rate.
Sampling Theorem states that if the highest frequency in an analog signa
l is Fmax and the signal is sampled at the rate fs > 2Fmax then x(t) can be exac
tly recovered from its sample values. This sampling rate is called Nyquist rate
of sampling.
If sampling frequency is less than Nyquist rate, then it is called under
sampling. Under sampling creates aliasing. In aliasing high frequencies appear
as low frequencies.
21. What is aliasing. Explain with example. How to avoid aliasing.
Example:
Case 1: X1(t) = cos 2? (10) t Fs= 40 Hz
i.e t= n/Fs
x1[n]= cos 2?(n/4)= cos (?/2)n
Case 2: X1(t) = cos 2? (50) t Fs= 40 Hz
i.e t= n/Fs
x1[n]= cos 2?(5n/4)=cos 2?(1+ ¼)n
=cos (?/2)n
Thus the frequency 50 Hz, 90 Hz , 130 Hz are alias of the frequency 10 Hz at the
 sampling rate of 40 samples/sec. To avoid aliasing sampling frequency should be
selected as per sampling theorem and pass the signal through pre-alias filter b
efore sampling.
22. What is quantization & coding.
The process of converting a discrete time continuous amplitude signal in
to a digital signal by expressing each sample value as a finite number of digits
is called quantization.
In the encoding operation, the quantization sample value is converted to
the binary equivalent of that quantization level. If 16 quantization levels are
present, 4 bits are required. Thus bits required in the coder is the smallest i
nteger greater than or equal to Log2 L.
23. What is anti-aliasing filter. In which applications it is mostly used.
The sampling rate of 6khz can be used for speech processing because spee
ch frequency range is up to 3kHz. But the speech signal also contains some frequ
ency components more than 3khz. Hence a sampling rate of 6khz will introduce ali
asing. Hence signal should be band limited to avoid aliasing. Thus the signal ca
n be band limited by passing it through a filter (LPF) which blocks or attenuate
s all the frequency components outside the specific bandwidth.
24. Discuss Quantization Noise
After a continuous-time signal has been through the A/D converter, the q
uantized output may differ from the input value. This deviation from the ideal o
utput value is called the quantization error.
25. What are recursive and non-recursive system
In Recursive systems, the output depends upon past, present, future valu
e of inputs as well as past output. In Non-Recursive systems, the output depends
only upon past, present or future values of inputs.
Example y(n)= x(n) + y(n-2) is recursive system and Y(n) = x(n) + x(n-1) is non
recursive system.
26. Explain the frequency relationships between continuous time and discrete
time signals.
Continuous time frequencies are given as O and F. while discrete time fr
equencies are given as ? and f. Conversion relationships are given as ? = O Ts a
nd f=FTs.
27. What is the use of correlation in DSP. How it is related with linear con
volution.
Correlations are nothing but establishing similarity between one set of
data and another. Correlation is closely related to convolution, because the cor
relation is essentially convolution of two data sequences in which one of the se
quences has been reversed.
Applications are in
1) Images processing for (in which different images are compared)
2) In radar and sonar systems for range and position finding in which transmit
ted and reflected waveforms are compared.
3) Correlation is also used in detection and identifying signals in noise.
28. What is the relationship between difference equation and system function
.
System function can be obtained by taking Z transform of the difference
equation.
UNIT II, III ? Z TRANSFORM
1. What is Z transform and ROC. What is the usefulness of ROC. What are the
applications of Z Transform.
For analysis of continuous time LTI system Laplace transform is used. An
d for analysis of discrete time LTI system Z transform is used. Z transform is
mathematical tool used for conversion of time domain into frequency domain (z d
omain) and is a function of the complex valued variable Z. The z transform of a
discrete time signal x(n) denoted by X(z) and given as
8
X(z) = ? x (n) z n z-Transform. (1)
n=-8
Z transform is an infinite power series because summation index varies f
rom -8 to 8. But it is useful for values of z for which sum is finite. The value
s of z for which f (z) is finite and lie within the region called as region of co
nvergence (ROC).
ADVANTAGES OF Z TRANSFORM : For calculation of DFT, for analysis and synthesis o
f digital filter, used for linear filtering, used for finding Linear convolution
, cross-correlation and auto-correlations of sequences.
ADVANTAGES OF ROC: ROC is going to decide whether system is stable or unstable,
the type of sequences causal or anti-causal & decides finite or infinite duratio
n sequences.
2. How poles and zeros & ROC decides the causality and stability of system.
LSI system is stable if and only if the ROC the system function includes
the unit circle. i.e r < 1. Thus Poles inside unit circle gives stable system.
Poles outside unit circle gives unstable system. Poles on unit circle give marg
inally stable system.
LSI system is causal if and only if the ROC the system function is exter
ior to
the circle. i. e |z| > r.
3. Discuss the nature of the signal.
4. Discuss the ROC of the signal & pole-zero plot of the signal.
Consider two cases
case 1: Infinite signal & case 2: Finite signal.

5. Discuss the nature of the signal.

6. ROC does not contains poles. Discuss the correctness of this statement.
7. Define pole and zero of the system. What poles and zeros are plotted wit
h respect to unit circle in z plane.
The frequency at which the magnitude of transfer function approaches inf
inity is called pole and the frequency at which magnitude of transfer function b
ecomes zero is called zero. Unit circle is the frequency axis in z plane.
8. What is the use of Unilateral Z transform.
Unilateral Z transform is used to solve the difference equation.
9. Can a pole and zero lie on the same point.
10. What are Dirichlet conditions.
11. Explain JURY'S Stability Algorithm
Jury's stability algorithm says
1. Form the first rows of the table by writing the coefficients of D(z).
 B0 B1 B2 --------- BN
BN BN-1 BN-2 --------- B0
2. Form third and fourth rows of the table by evaluating the determinant CJ
3. This process will continue until you obtain 2N-3 rows with last two havi
ng 3
elements. Y0,Y1,Y2
A digital filter with a system function H(z) is stable, if and only if it passes
the following conditions.
a. D(Z)|Z=1 > 0
b. (-1)N D(Z)|Z=-1 >0
c. |b0|>|bN|, |C0|>|CN-1|
Z Transform Properties
Sr No Property X(n) X(z)
1 Linearity a1 x1(n) + a2 x2(n) a1 X1(z) + a2 X2(z)
2 Time shifting x(n-k) X(z) z k
3 Scaling in z domain an x(n) x(z/a)
4 Time reversal x(-n) x(z-1)
5 Convolution Theorem x1(n) * x2(n) X1(z) X2(z)
Standard Z Transforms
Sr No X(n) Property X(Z) ROC
1 d(n) 1 complete z plane
2 d(n-k) Time shifting z-k except z=0
3 d(n+k) Time shifting zk except z=8
4 u(n) 1/1- z-1 |z| > 1
5 u(-n) Time reversal 1/1- z |z| < 1
6 -u(-n-1) Time reversal z/z- 1 |z| < 1
7 n u(n) Differentiation z-1 / (1- z-1)2 |z| > 1
8 an u(n) Scaling 1/1- (az-1) |z| > |a|
9 -an u(-n-1) 1/1- (az-1) |z| < |a|
10 n an u(n) Differentiation a z-1 / (1- az-1)2 |z| > |a|
11 -n an u(-n-1) Differentiation a z-1 / (1- az-1)2 |z| < |a|
12 cos(?0n) u(n) 1- z-1cos?0
1- 2z-1cos?0+z-2
|z| > 1
13 sin(?0n) u(n) z-1sin?0
1- 2z-1cos?0+z-2
|z| > 1

UNIT IV: FT,DFT AND FFT

1. Why the frequency domain analysis is preferred over time domain analysis
in DSP.
Time domain analysis provides some information like amplitude at sampling instan
t but does not convey frequency content & power, energy spectrum hence frequency
domain analysis is used. Magnitude and phase plot can be obtained from its FT a
nd system characteristic can be described well by using its frequency domain.
2. What is DTFT. Explain the nature of the spectrum of discrete time signal
.
The discrete time Fourier transform of the signal is denoted as X(?). It
is also called as analysis equation. It is given as
8
X(?) = ? x (n) e j?n
 n=-8
Here ? is the frequency of discrete time signal and it takes all possible values
between -? to ?. Hence its Fourier transform is continuous.
Case 1: If x(n) is infinite or finite non-periodic sequence then its spectrum X(
?) is continuous in nature.
Case 2: If x(n) is finite periodic sequence then its spectrum X(?) will be disc
rete.
Inverse DTFT is also called synthesis equation. Here integration is used since X
(?) is the continuous function of ?. Integration limits are -? to ?. And the per
iod of integration is 2?.
3. What is the existence criteria of DTFT. Why it is used.
In the definition of DTFT, there is summation over infinite range of n.
Hence for DTFT to exist, the convergence of this summation is necessary. Hence e
xistence criteria is
8
? |x(n)| < 8
n=-8
IDTFT does not have convergence problem since the integration is over limited r
ange.
4. What are the symmetry properties of FT.

Sr No Sequence DTFT
1 X*(n) X*(- ?)
2 X*(-n) X*(?)
3 XR(n) Xe(?)=1/2 [ X(?) + X*(-?)]
4 jXI(n) Xo(?)=1/2 [ X(?) - X*(-?)]
5 Xe(n) XR(?)
6 Xo(n) jXI(?)
DTFT Properties:
Sr No Property Time domain Sequence Frequency Domain Sequence
1
Periodicity
x(n)
X(?+2?k)= X(?)
2
Linearity
a1x1(n)+a2x2(n)
a1X1(?)+a2X2(?)
3
Time Shifting
x(n-k)
e-j?k X(?)
4
Time Reversal
x(-n)
X(-?)
5
Convolution
x1(n) * x2(n)
X1(?)+ X2(?)
6
Frequency Shifting
e-j?on x(n)
X(?- ?0)
7
Scaling
x(pn)
X(?/p)
8
Differentiation
-j n x(n)
d/d? [X(?)]
9
Parseval's Theorem Energy of the signal is given by
E= 1/2? ? |X(?)|2 d?

DFT Properties:
Sr No Property Time domain Sequence Frequency Domain Sequence
1
Periodicity
x(n)
X(k+N)= X(k)
2
Linearity
a1x1(n)+a2x2(n)
a1X1(k)+a2X2(k)
3
Circular Time Shift
X((n-k))N
e-j2?kl/N X(k)
4
Time Reversal
X((-n))N
X((-k))N
5
Circular Convolution
x1(n) N x2(n) N-1
? x1(n) x2((m-n))N
n=0
6
Circular frequency Shifting
ej2?kl/N X(n)
X((k-l))N
7
Parseval's Theorem Energy of the signal is given by
N-1
E= 1/N ? |X(k)|2
K=0

5. Why DFT's are used in frequency domain analysis in place of DTFT.

FT is the continuous function of x(n) and the range of ? is from - ? to
? or 0 to 2?. while DFT is calculated only at discrete values of ?. Thus DFT is
discrete in nature which is sampling version of FT and thus mostly used in analy
sis of discrete signals.
For Discrete time signals x(n) , Fourier Transform is denoted as x(?) & given by
8
X(?) = ? x (n) e j?n
n=-8
DFT is denoted by x(k) and given by (?= 2 ? k/N)
N-1
X(k) = ? x (n) e j2 ? kn / N
n=0
6. Circular convolution and Linear convolution are same or different.
Multiplication of two sequences in time domain is called as Linear convo
lution while Multiplication of two sequences in frequency domain is called as ci
rcular convolution. They are one and same but they differ in total number of sam
ples in it.
7. What are overlap save and add method. Why these methods are used.
When the input data sequence is long then it requires large time to get
the output sequence. Hence other techniques are used to filter long data sequenc
es. Instead of finding the output of complete input sequence, it is broken into
small length sequences. The output due to these small length sequences are compu
ted fast. The outputs due to these small length sequences are fitted one after a
nother to get the final output response.
8. What is FFT. In which applications it is preferred over DFT.
Large number of the applications such as filtering, correlation analysis
, spectrum analysis require calculation of DFT. But direct computation of DFT re
quire large number of computations and hence processor remain busy. Hence specia
l algorithms are developed to compute DFT quickly called as Fast Fourier algorit
hms (FFT).
The radix-2 FFT algorithms are based on divide and conquer approach. In
this method, the N-point DFT is successively decomposed into smaller DFT s. Becaus
e of this decomposition, the number of computations are reduced.
9. If input signal x(n) contains 4 samples. How many samples will be presen
t in its DFT. What will happen if it contains less than 4 samples.
10. What is the difference between DITFFT and DIFFFT.
Sr No DIT FFT DIF FFT
1 DITFFT algorithms are based upon decomposition of the input sequence int
o smaller and smaller sub sequences. DIFFFT algorithms are based upon decompo
sition of the output sequence into smaller and smaller sub sequences.
2 In this input sequence x(n) is splitted into even and odd numbered sampl
es In this output sequence X(k) is considered to be splitted into even and
odd numbered samples
3 Splitting operation is done on time domain sequence. Splitting operat
ion is done on frequency domain sequence.
4 In DIT FFT input sequence is in bit reversed order while the output sequ
ence is in natural order. In DIFFFT, input sequence is in natural order. A
nd DFT should be read in bit reversed order.
11. What is use of Goertzel Algorithm.
If DFT is to be calculated at selected points only then, Goertzel algori
thms are used. Goertzel algorithms are used to calculated DFT as linear filterin
g operations and required less number of calculations.
12. State the relationship between ZT and FT.
There is a close relationship between Z transform and Fourier transform.
If we replace the complex variable z by e j?, then Z transform is reduced to Four
ier transform.
13. What mathematical tools are used to convert the signals from time domain
to frequency domain.
14. What are Dirichlet conditions.
15. Expansion in time domain is equivalent to compression in frequency
domain. Discuss this statement with an example.
16. If two sequences are multiplied in time domain what will be effect on th
eir DFT's.
17. Circular Convolution can be obtained from linear convolution but vice-ve
rsa is not possible. Discuss this statement with an example.
18. State any two applications of A) Linear convolution B)Circular Convoluti
on C) DFT D) FFT
19. What is use of bit reversal technique. Where it is used.
Decimal Memory Address x(n) in binary (Natural Order) Memory Address in bit re
versed order New Address in decimal
0 0 0 0 0 0 0 0
1 0 0 1 1 0 0 4
2 0 1 0 0 1 0 2
3 0 1 1 1 1 0 6
4 1 0 0 0 0 1 1
5 1 0 1 1 0 1 5
6 1 1 0 0 1 1 3
7 1 1 1 1 1 1 7
Table shows first column of memory address in decimal and second column as binar
y. Third column indicates bit reverse values. As FFT is to be implemented on dig
ital computer simple integer division by 2 method is used for implementing bit r
eversal algorithms.
20. Explain In Place computation and Memory requirement concept.
a
A= a + WNr b

b WNr
B= a - WNr b
From values a and b new values A and B are computed. Once A and B are co
mputed, there is no need to store a and b. Thus same memory locations can be use
d to store A and B where a and b were stored hence called as In place computatio
n. The advantage of in place computation is that it reduces memory requirement.
Thus for computation of one butterfly, four memory locations are required for st
oring two complex numbers A and B.
21. Can FFT Algorithms are applicable for the values of N which are not powe
r of 2. Example N=12.
Yes, In such cases sequence is padded with sufficient number of zeros su
ch that the value of N becomes the power of 2. Alternately (Another method)
If N=12, It can be divided into 3 sequence of 4 samples each. These sequences wi
ll be as follows
First Sequence: x(0), x(3), x(6), x(9)
Second sequence: x(1), x(4), x(7), x(10)
Third sequence: x(2), x(5), x(8), x(11)
Now 4 point DFT's are calculated and then proceed further.

UNIT V ? DIGITAL FILTER

1. What is Sinc function.
Sinc pulse represents impulse response of ideal LPF while impuls
e train represents ideal sampling function.
2. What is inversibility property.
If the system is invertible then HH-1=1. This means if the two systems a
re cascaded, output is same as input. Thus the condition for system to be inver
tible in terms of impulse response is h(n)*h-1(n) = d(n).
3. Difference between analog and digital filter.
Analog filters are used for filtering analog signals while digital filte
rs are used for digital signals. Analog filters are designed with various compon
ents like resistor, inductor and capacitor and digital Filters are designed with
digital hardware like FF, counters shift registers, ALU and software s like C or
assembly language.
Digital filters are more accurate, less sensitive to environmental chan
ges, most flexible, programmable and stable.
4. What are ideal filter characteristic.
1. Ideal filters have a constant gain (usually taken as unity gain)
passband characteristic and zero gain in their stop band.
2. Ideal filters have a linear phase characteristic within their pa
ssband.
3. Ideal filters also have constant magnitude characteristic.
5. What are notch and Comb filters. What are its applications.
A notch filter is a filter that contains one or more deep notches or ide
ally perfect nulls in its frequency response characteristic. Notch filters are u
seful in many applications where specific frequency components must be eliminate
d. Example Instrumentation and recording systems required that the power-line fr
equency 60Hz and its harmonics be eliminated.
comb filters are similar to notch filters in which the nulls occur perio
dically across the frequency band similar with periodically spaced teeth.
Frequency response characteristic of notch filter |H(?)| is as shown
?o ?1 ?
6. What are digital resonators. In which applications they are used.
A digital resonator is a special two pole bandpass filter with a pair of
complex conjugate poles located near the unit circle. The name resonator refers
to the fact that the filter has a larger magnitude response in the vicinity of
the pole locations. Digital resonators are useful in many applications, includin
g simple bandpass filtering and speech generations.
7. What is difference between FIR and IIR filter
FIR system has finite duration unit sample response. i.e h(n)=0 for n<0
and n = M
IIR system has infinite duration unit sample response. i. e h(n) = 0 for n<0
FIR systems are non recursive. Thus output of FIR filter depends upon present an
d past inputs while IIR systems are recursive. Thus output of IIR filter depends
upon present and past inputs as well as past outputs.
FIR filters are most stable, requires limited memory. In IIR filters sta
bility can not be guaranteed and requires infinite memory.
8. In which applications FIR filters are designed.
FIR filters can have an exactly linear phase response so that no phase d
istortion is introduced in the signal by the filter. Hence FIR filters are gener
ally used if no phase distortion is desired. Example: Data Transmission over a l
ong distance and speech processing FIR filters are used.
9. In which applications IIR filters are designed.
IIR filters are generally used if sharp cutoff and high throughput is re
quired. Also Analogue filters can be easily and readily transformed into equival
ent IIR digital filter.
10. How the stable filters can be designed.
All poles should be placed inside the unit circle on order for the filte
r to be stable. However zeros can be placed anywhere in the z plane.
1. FIR filters are all zero filters hence they are always stable.
2. IIR filters are stable only when all poles of the filter are inside unit
circle.
11. Difference between impulse invariance and BZT method.
Impulse invariance: In this method IIR filters are designed having a unit sampl
e response h(n) that is sampled version of the impulse response of the analog fi
lter. Hence small value of T is selected to minimize the effect of aliasing. Fre
quency relationship is linear and all poles are mapped But the main disadvantage
of this method is that it does not correspond to simple algebraic mapping of S
plane to the Z plane. Thus the mapping from analog frequency to digital fre
quency is many to one.
Bilinear transformation Method: The bilinear transformation is a conformal mappi
ng that transforms the j O axis into the unit circle in the z plane only once, t
hus avoiding aliasing of frequency components. But Frequency relationship is non
-linear. Frequency warping or frequency compression is due to non-linearity.
Impulse invariance method is generally used for designing low frequencies filter
like LPF. while for designing of LPF, HPF and almost all types of Band pass and
band stop filters BZT method is used.
12. Plot Mapping between analog and digital filter frequencies in BZT method
.

? 2 tan -1 (OT/2)

OT

13. What is frequency warping. Why it is used in filter design.

In BZT Frequency relationship is non-linear. Frequency warping or freque
ncy compression is due to non-linearity. Frequency warping means amplitude respo
nse of digital filter is expanded at the lower frequencies and compressed at
the higher frequencies in comparison of the analog filter. But the main disadva
ntage of frequency warping is that it does change the shape of the desired filte
r frequency response.
14. What are different approximation. how it is useful in filter design.
No Practical filters can provide the ideal characteristic. Hence approxi
mation of the ideal characteristic are used. Such approximations are standard an
d used for filter design. Such three approximations are regularly used. Butterw
orth Filter Approximation, Chebyshev Filter Approximation and Elliptic Filter Ap
proximation
Butterworth filters are defined by the property that the magnitude respo
nse is maximally flat in the passband.
15. State the mapping between Z Plane and S plane in Impulse Invariance meth
od or Bilinear Transformation method.
1) Left side of s-plane is mapped inside the unit circle.
2) Right side of s-plane is mapped outside the unit circle.
3) jO axis is in s-plane is mapped on the unit circle.
Im[z]
jO

Re(z)
s
 Z-Plane S-Plane
16. What is all pass filter. What are its applications.
An all pass filter is defined as a system that has a constant magnitude
response for all frequencies.
|H(?)| = 1 for 0 = ? < ?
The simplest example of an all pass filter is a pure delay system with system fu
nction
H(z) = Z-k. This is a low pass filter that has a linear phase characteristic.
All Pass filters find application as phase equalizers. When placed in ca
scade with a system that has an undesired phase response, a phase equalizers is
designed to compensate for the poor phase characteristic of the system and there
fore to produce an overall linear phase response.
17. FIR filter are always stable. Explain.
In FIR Impulse response of the system is given as
H(n) = bn for 0 = n = M-1
= 0 otherwise.
i.e Y(n) = b0 x(n) + b1 x(n-1) + .. + bM-1 x(n-M+1)
Thus y(n) is bounded if input x(n) is bounded. This means FIR system produces bo
unded output for every bounded input. Hence FIR systems are always stable.
18. What are the various method used for FIR & IIR filter design
The various methods used for IIR Filer design are as follows
1. Approximation of derivatives
2. Impulse Invariance
3. Bilinear Transformation
The various method used for FIR Filer design are as follows
1. Windowing Method
2. DFT method
3. Frequency sampling Method. (IFT Method)
19. What are Gibbs phenomenon
Impulse response of an ideal LPF is as shown in Fig.

In Fourier series method, limits of summation index is -8 to 8. But fil

ter must have finite terms. Hence limit of summation index change to -Q to Q whe
re Q is some finite integer. But this type of truncation may result in poor conv
ergence of the series. Abrupt truncation of infinite series is equivalent to mul
tiplying infinite series with rectangular sequence. i.e at the point of disconti
nuity some oscillation may be observed in resultant series.
Consider the example of LPF having desired frequency response Hd (?) as
shown in figure. The oscillations or ringing takes place near band-edge of the f
ilter. This oscillation or ringing is generated because of side lobes in the fre
quency response W(?) of the window function. This oscillatory behavior is called
"Gibbs Phenomenon".
20. Ideal filter are not physically realizable. Why.
LSI system is causal if its unit sample response satisfies following condition.
h(n) = 0 for n<0
In above figure h(n) extends -8 to 8. Hence h(n) ?0 for n<0. This means causalit
y condition is not satisfied by the ideal low pass filter. Hence ideal filters a
re anti-causal and thus are not physically realizable.
21. FIR Filters always provides linear phase response. Explain.
The phase or angle of H(?) is given as
-? M-1 for |H (?)| > 0
 2
Angle H(?) =
-? M-1 for |H (?)| < 0
2
In above equations M is constant. Hence Phase of H(?) is linear function of ?. T
hat is phase is linearly proportional to frequency. When |H(?)} changes sign, ph
ase changes by ?. Thus FIR filters are linear phase filters. This is important
feature of FIR Filters.
22. For Speech processing or data transmission which type of filter are pref
erred.
FIR filter always provides linear phase response. This specifies that th
e signals in the pass band will suffer no dispersion Hence when the user wants
no phase distortion, then FIR filters are preferable over IIR. Phase distortion
always degrade the system performance. In various applications like speech proce
ssing, data transmission over long distance FIR filters are more preferable due
to this characteristic. Another reason is that quantization noise can be
made negligible in FIR filters.
23. How FIR filters can be classified.
FIR filters can be classified into two types. Symmetric and Anti-symmetric FIR f
ilters
1 Unit sample response of FIR filters is symmetric if it satisfies followi
ng condition
h(n)= h(M-1-n) n=0,1,2 .M-1
2. Unit sample response of FIR filters is Anti-symmetric if it satisfies fo
llowing condition
h(n)= -h(M-1-n) n=0,1,2 .M-1
24. Why FIR needs higher orders for similar magnitude response compared to I
IR filters.
Impulse response of ideal low pass filter is as shown in fig. In order t
o have finite terms we will multiply this infinite series with rectangular windo
w which will generate desired frequency response. But some oscillation or ringin
g effect will be observed at the point of truncation. This effect is known as Gi
bbs Phenomenon.
As M increases this side lobes becomes narrow and oscillatory behavior d
ecreases. As an example, the impulse response for a LPF is truncated with M=9,25
and an infinite number of samples is as shown.
25. What are Windows techniques? How they are selected.
Impulse response of ideal filter is infinite but in FIR filter, h(n) is
finite. Hence in order to truncate infinite impulse response to finite range we
will multiply it to window and thus practically implemented. There are various t
ypes of windows like rectangular, triangular, hamming, Hanning window etc.
The windows are selected depending upon the transition width of main lobe and am
plitudes of sidelobes. The windows are selected such that Gibb's phenomenon is r
educed.
The particular window is selected depending upon minimum stop band atten
uation.
26. What are different window functions used for design of FIR filters.
Different types of windows functions are available which reduce ringing
effect. These are Triangular window, Blackman, Hamming window, Hanning Window a
nd Kaiser window.
a. FIR filters designed using hamming window has reduced sidelobes compared
to rectangular window.
b. Blackman window has very small sidelobes but increased width of main lob
e. In Kaiser window has reduced side lobes and transition band is narrow and hen
ce mostly used.

27. What are the constraints to be imposed while designing filters from it p
ole zero plot.
Filters can be designed from its pole zero plot. Following two constraints shoul
d be imposed while designing the filters.
1. All poles should be placed inside the unit circle on order for the filte
r to be stable. However zeros can be placed anywhere in the z plane. FIR filters
are all zero filters hence they are always stable. IIR filters are stable only
when all poles of the filter are inside unit circle.
2. All complex poles and zeros occur in complex conjugate pairs in order fo
r the filter coefficients to be real.
In the design of low pass filters, the poles should be placed near the unit circ
le at points corresponding to low frequencies ( near ?=0)and zeros should be pla
ced near or on unit circle at points corresponding to high frequencies (near ?=?
). The opposite is true for high pass filters.
28. Which window is better. Short duration window or long duration window.
Long Duration window. Because the length of window must be infinite in i
deal case.
29. What are frequency transformation techniques. Why they are used.
Frequency transformation techniques are used to generate High pass filter, Bandp
ass and bandstop filter from the lowpass filter system function.
Sr No Type of transformation Transformation ( Replace s by)
1 High Pass ?hp
s
?hp = Password edge frequency of HPF
2 Band Pass (s2 + ?l ?h )
s (?h - ?l )
?h - higher band edge frequency
?l - Lower band edge frequency
3 Band Stop s (?h - ?l)
s2+ ?h ?l
?h - higher band edge frequency
?l - Lower band edge frequency

UNIT VI: DSP PROCESSOR

1. What are the requirements of DSP processor. How It differs from general
Processor.
The most fundamental mathematical operation in DSP is sum of products also calle
d as dot of products.
Y(n)= h(0)*x(n) + h(1)*x(n-1) + + h(N-1)*x(n-N)
This operation is mostly used in digital filter designing, DFT, FFT and many oth
er DSP applications. A DSP is optimized to perform repetitive mathematical opera
tions such as the dot product. There are four basic requirements of DSP processo
r to optimize the performance They are
1) Fast arithmetic
2) Fast Execution - Dual operand fetch
3) Fast data exchange
4) Circular buffering
Sr No Requirements Features of DSP processor
1 Fast Arithmetic Faster MACs means higher bandwidth.
Able to support general purpose math functions, should have ALU and a programmab
le shifter function for bit manipulation.
Powerful interrupt structure and timers
2 Fast Execution Parallel Execution is required in place of sequential.
Instructions are executed in single cycle of clock called as True instruction c
ycle as oppose to multiple clock cycle.
Multiple operands are fetched simultaneously. Multi-processing Ability and queue
, pipelining facility
Address generation by DAG's and program sequencer.
3 Fast data Exchange Multiple registers, Separate program and data me
mory and Multiple operands fetch capacity
4 Circular shift operations Circular Buffers
2. What are different microprocessor architectures. Which is mostly used in DSP
processor.
There are mainly three types of microprocessor architectures present.
1. Von-Neumann architecture
2. Harvard architecture
3. Analog devices Modified Harvard architecture.
Harvard Architecture is common to many DSP processors. The processor can simulta
neously access two memory blanks using two independent sets of buses allowing op
erands to be loaded while fetching instructions.
Von-Neumann memory architecture is common among microcontrollers Since there is
only one data bus, operands can not be loaded while instructions are fetched.
3. Explain core architecture of ADSP-21xx processor.
ADSP-21xx family DSP's are used in high speed numeric processing applications.
ADSP-21xx architecture consists of
Five Internal Buses
Program Memory Address(PMA)
Data memory address (DMA)
Program memory data(PMD)
Data memory data (DMD)
Result (R)
Three Computational Units
Arithmetic logic unit (ALU)
Multiply-accumulate (MAC)
Shifter
Two Data Address generators (DAG)
Program sequencer
On chip peripheral Options
RAM or ROM
Data Memory RAM
Serial Port
Timer
Host Interface Port
DMA Port

FEATURES OF ADSP-21xx PROCESSOR

1. 16 bit fixed DSP microprocessor
2. Enhanced Harvard architecture for three bus performance.
3. Separate on chip buses for program and data memory.
4. 25 MIPS, 40 ns maximum instruction set 25Mhz frequency.
5. Single cycle instruction execution i.e True instruction cycle.
4. What are ADSP-21xx Development tools
Various development tools such as assembler , linker, debugger, and simulator ar
e available for ADSP-21xx family.
The system builder is the software development tool for describing the c
onfiguration of the target system's memory and I/O. The ranges for program mem
ory(PM) and data memory(DM) are described.
The assembler translated source code into object code modules. The sourc
e code is written in assembly language file (.DSP) Assembler reads .DSP file an
d generates four
output filed with the same root name. Object file(.OBJ), Code File(.CDE), Initia
lization File (.INT), List File(.LST) etc.
The linker is a program used to join together object files into one larg
e object file. The linker produces a link file which contains the binary codes f
or all the combined modules.
A debugger is a program which allows user to load object code program in
to system memory, execute the program and debug it.
Difference between DSP and General Purpose Processor
Sr Parameter General Purpose Processor DSP Processor
1 Instruction Cycle Multiple clock cycles required for execution of
one instruction Single cycle of the clock is needed.
2 Instruction execution Sequential Execution Parallel execution - Pip
elining involved
3 Operand fetch from the memory Sequential Multiple operand fetch c
apability
4 Memories No separate memory Separate program and data memory
5 Instruction set Mostly Contains Data movement instructions Contains
complex addition, multiplication & shifting instructions
6 Address generation PC is used DAG and Program Sequencer
7 On chip address and data buses Single pair of buses PMA,DMA, PMD and
DMD
8 Computational Units ALU ALU, MAC and Shifter
5. What are the different functions used in MATLAB related with DSP.
Sr No Function Application
1 conv(hn,xn) Linear Convolution of two sequences.
2 xcorr(x1n,x2n) Cross Correlation of two sequences.
3 xcorr(x) Auto correlation of sequence
3 fft(xn) DFT of x(n) using FFT algorithm
4 ifft(xn) IDFT of x(n) using FFT algorithm
5 Overlpsav (x,h,N) Implement Overlap save method to perform block c
onvolution.
5 zplane (b,a) Plot Pole zero plot.
6 freqz (b,a) Plot Magnitude phase plot.
7 freqs (b,a) Compute the frequency response of an analog filter.
8 bilinear(z,p,k,fs) Bilinear transformation
9 boxvar (M) Rectangular Window
10 hanning (M) Hanning Window
11 hamming (M) Hamming Window
12 kaiser(M) Kaiser Window