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 ADC DIGITAL DAC Analog

Signal SYSTEM 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 operations like
filtering, multiplication, transformation or amplification etc operations over these digital
signals. The digital output signal from the DSP is given 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. Speech or
voice signal.
Multi-channel signal  signals are generated from multiple sensors or multiple sources
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 finite or
infinite range
Discrete values signal.  signal takes values from a finite set of possible values.

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 period 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 signal is called as
energy signal if its energy if finite. A signal is called power signal if its power is finite.
Ex: Energy of unit sample function is 1.

4. What is maximum range of discrete time frequencies & continuous time

frequencies.

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Discrete time frequencies = -1/2 to 1/2 cycles/sample or -∏ to +∏ rad/samples
Continuous time frequencies = - ∞ to +∞

5. Prove that discrete time signals are periodic only if frequency is rational.
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 even

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?

∞
∑ x (k) h(n – k ) (1)
k= -∞
∞
∑ x (k) δ(n – k ) (2)
k= -∞

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 linear if and
only if T[a1x1(n) + a2x2(n)]=a1T[x1(n)]+a2T[x2(n)] (Superposition Theorem)

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.

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Let X(n) = {2,5,2}. The signal x(n) can also be written as
X(n)= x(-1) δ(n+1) + x(0) δ(n) + x(1) δ(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 signal. Now
the elementary input signals are taken into account and individually given to the system.
Now using linearity property Whatever output response we get for decomposed input
signal, we simply add it & this will provide us total response of the system to any given
input signal.
Linear Convolution states that
y(n) = x(n) * h(n)
∞
y(n) = ∑ x (k) h(n – k )
k= -∞

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.
∞
∑ |h(k)| < ∞
k=-∞

18. How the LSI system is represented by constant coefficient difference

equation. (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 time signal by
taking samples of the continuous time signal at discrete time instants.

20. What is sampling theorem. What is Nyquist rate.

Sampling Theorem states that if the highest frequency in an analog signal is Fmax
and the signal is sampled at the rate fs > 2Fmax then x(t) can be exactly 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

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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 before sampling.

22. What is quantization & coding.

The process of converting a discrete time continuous amplitude signal into 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 integer 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 speech
frequency range is up to 3kHz. But the speech signal also contains some frequency
components more than 3khz. Hence a sampling rate of 6khz will introduce aliasing. Hence
signal should be band limited to avoid aliasing. Thus the signal can be band limited by
passing it through a filter (LPF) which blocks or attenuates 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 quantized
output may differ from the input value. This deviation from the ideal output value is called
the quantization error.

25. What are recursive and non-recursive system

In Recursive systems, the output depends upon past, present, future value 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 Ω and F. while discrete time frequencies
are given as ω and f. Conversion relationships are given as ω = Ω Ts and f=FTs.

27. What is the use of correlation in DSP. How it is related with linear
convolution.
Correlations are nothing but establishing similarity between one set of data and
another. Correlation is closely related to convolution, because the correlation is essentially
convolution of two data sequences in which one of the sequences 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 transmitted 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.

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 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. And 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 domain) 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
∞
X(z) = ∑ x (n) z –n z-Transform.……(1)
n=-∞
Z transform is an infinite power series because summation index varies from -∞ to
∞. But it is useful for values of z for which sum is finite. The values of z for which f (z) is
finite and lie within the region called as “region of convergence (ROC).
ADVANTAGES OF Z TRANSFORM : For calculation of DFT, for analysis and synthesis of
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 duration 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 marginally stable system.
LSI system is causal if and only if the ROC the system function is exterior 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.

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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 with
respect to unit circle in z plane.
The frequency at which the magnitude of transfer function approaches infinity is
called pole and the frequency at which magnitude of transfer function becomes 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

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 B0 BN-J
BN BJ
3. This process will continue until you obtain 2N-3 rows with last two having 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 δ(n) 1 complete z plane
2 δ(n-k) Time shifting z-k except z=0
3 δ(n+k) Time shifting zk except z=∞
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(ω0n) u(n) 1- z-1cosω0 |z| > 1
1- 2z-1cosω0+z-2
13 sin(ω0n) u(n) z-1sinω0 |z| > 1
1- 2z-1cosω0+z-2

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 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 instant 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 and 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(ω). It is also called
as analysis equation. It is given as
∞
X(ω) = ∑ x (n) e –jωn
n=-∞
Here ω 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(ω) is
continuous in nature.
Case 2: If x(n) is finite periodic sequence then its spectrum X(ω) will be discrete.
Inverse DTFT is also called synthesis equation. Here integration is used since X(ω) is the
continuous function of ω. Integration limits are -∏ to ∏. And the period 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 existence criteria is
∞
∑ |x(n)| < ∞
n=-∞
IDTFT does not have convergence problem since the integration is over limited range.

4. What are the symmetry properties of FT.

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 Sr No Sequence DTFT
1 X*(n) X*(- ω)
2 X*(-n) X*(ω)
3 XR(n) Xe(ω)=1/2 [ X(ω) + X*(-ω)]
4 jXI(n) Xo(ω)=1/2 [ X(ω) - X*(-ω)]
5 Xe(n) XR(ω)
6 Xo(n) jXI(ω)
DTFT Properties:

Sr No Property Time domain Sequence Frequency Domain Sequence

1 Periodicity x(n) X(ω+2∏k)= X(ω)

2 Linearity a1x1(n)+a2x2(n) a1X1(ω)+a2X2(ω)

3 Time Shifting x(n-k) e-jωk X(ω)

4 Time Reversal x(-n) X(-ω)

5 Convolution x1(n) * x2(n) X1(ω)+ X2(ω)

6 Frequency Shifting e-jωon x(n) X(ω- ω0)

7 Scaling x(pn) X(ω/p)

8 Differentiation -j n x(n) d/dω [X(ω)]

Energy of the signal is given by
9 Parseval's Theorem E= 1/2∏ ∫ |X(ω)|2 dω

DFT Properties:

Sr Property Time domain Frequency Domain

No Sequence 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

N-1
5 Circular Convolution x1(n) N x2(n) ∑ x1(n) x2((m-n))N
n=0

6 Circular frequency Shifting ej2∏kl/N X(n) X((k-l))N

Energy of the signal is given by
7 Parseval's Theorem N-1
E= 1/N ∑ |X(k)|2
K=0

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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 ω is from - ∏ to ∏ or 0 to 2∏.
while DFT is calculated only at discrete values of ω. Thus DFT is discrete in nature which is
sampling version of FT and thus mostly used in analysis of discrete signals.
For Discrete time signals x(n) , Fourier Transform is denoted as x(ω) & given by
∞
X(ω) = ∑ x (n) e –jωn
n=-∞
DFT is denoted by x(k) and given by (ω= 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 convolution while
Multiplication of two sequences in frequency domain is called as circular convolution. They
are one and same but they differ in total number of samples 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 sequences. 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 computed fast. The outputs due to these
small length sequences are fitted one after another 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 require large number of
computations and hence processor remain busy. Hence special algorithms are developed to
compute DFT quickly called as Fast Fourier algorithms (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. Because of this
decomposition, the number of computations are reduced.

9. If input signal x(n) contains 4 samples. How many samples will be present
in its DFT. What will happen if it contains less than 4 samples.

10. What is the difference between DITFFT and DIFFFT.

Sr DIT FFT DIF FFT
No
1 DITFFT algorithms are based upon
decomposition of the input sequence into
smaller and smaller sub sequences.
2 In this input sequence x(n) is splitted
into even and odd numbered samples
3 Splitting operation is done on time
domain sequence.
4 In DIT FFT input sequence is in bit
reversed order while the output sequence
is in natural order.
DIFFFT algorithms are based upon
decomposition of the output sequence
into smaller and smaller sub sequences.
In this output sequence X(k) is
considered to be splitted into even and
odd numbered samples
Splitting operation is done on frequency
domain sequence.
In DIFFFT, input sequence is in natural
order. And DFT should be read in bit
reversed order.

11. What is use of Goertzel Algorithm.

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 If DFT is to be calculated at selected points only then, Goertzel algorithms are used.
Goertzel algorithms are used to calculated DFT as linear filtering 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–jω, then Z transform is reduced to Fourier 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 their
DFT's.

17. Circular Convolution can be obtained from linear convolution but vice-versa
is not possible. Discuss this statement with an example.

18. State any two applications of A) Linear convolution B)Circular Convolution

C) DFT D) FFT

19. What is use of bit reversal technique. Where it is used.

Memory Address x(n) in Memory Address in bit New Address in

Decimal
binary (Natural Order) reversed order 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 binary.
Third column indicates bit reverse values. As FFT is to be implemented on digital computer
simple integer division by 2 method is used for implementing bit reversal 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
computed, there is no need to store a and b. Thus same memory locations can be used to
store A and B where a and b were stored hence called as In place computation. The
advantage of in place computation is that it reduces memory requirement. Thus for
computation of one butterfly, four memory locations are required for storing two complex
numbers A and B.

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21. Can FFT Algorithms are applicable for the values of N which are not power
of 2. Example N=12.
Yes, In such cases sequence is padded with sufficient number of zeros such 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 will 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.

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 UNIT V  DIGITAL FILTER
1. What is Sinc function.
Sinc pulse represents impulse response of ideal LPF while impulse 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 are cascaded,
output is same as input. Thus the condition for system to be invertible in terms of impulse
response is h(n)*h-1(n) = δ(n).

3. Difference between analog and digital filter.

Analog filters are used for filtering analog signals while digital filters are used for
digital signals. Analog filters are designed with various components 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 changes, 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 passband.
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 ideally perfect
nulls in its frequency response characteristic. Notch filters are useful in many applications
where specific frequency components must be eliminated. Example Instrumentation and
recording systems required that the power-line frequency 60Hz and its harmonics be
eliminated.
comb filters are similar to notch filters in which the nulls occur periodically across
the frequency band similar with periodically spaced teeth.
Frequency response characteristic of notch filter |H(ω)| is as shown

ωo ω1 ω

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, including 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
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FIR systems are non recursive. Thus output of FIR filter depends upon present and 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 stability 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 distortion is
introduced in the signal by the filter. Hence FIR filters are generally used if no phase
distortion is desired. Example: Data Transmission over a long 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 required. Also
Analogue filters can be easily and readily transformed into equivalent IIR digital filter.

10. How the stable filters can be designed.

All poles should be placed inside the unit circle on order for the filter 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 sample
response h(n) that is sampled version of the impulse response of the analog filter. Hence
small value of T is selected to minimize the effect of aliasing. Frequency 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 frequency is many to one.

Bilinear transformation Method: 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
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.

-1
ω 2 tan (ΩT/2)

ΩT

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13. What is frequency warping. Why it is used in filter design.
In BZT Frequency relationship is non-linear. Frequency warping or frequency
compression is due to non-linearity. Frequency warping means amplitude response of
digital filter is expanded at the lower frequencies and compressed at the higher frequencies
in comparison of the analog filter. But the main disadvantage of frequency warping is that
it does change the shape of the desired filter frequency response.

14. What are different approximation. how it is useful in filter design.

No Practical filters can provide the ideal characteristic. Hence approximation of the
ideal characteristic are used. Such approximations are standard and used for filter design.
Such three approximations are regularly used. Butterworth Filter Approximation,
Chebyshev Filter Approximation and Elliptic Filter Approximation
Butterworth filters are defined by the property that the magnitude response is
maximally flat in the passband.

15. State the mapping between Z Plane and S plane in Impulse Invariance
method 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) jΩ axis is in s-plane is mapped on the unit circle.

Im[z] jΩ
1

Re(z) 3 σ

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(ω)| = 1 for 0 ≤ ω < ∏
The simplest example of an all pass filter is a pure delay system with system function
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 cascade 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 therefore 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 bounded
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

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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 -∞ to ∞. But filter must have
finite terms. Hence limit of summation index change to -Q to Q where Q is some finite
integer. But this type of truncation may result in poor convergence of the series. Abrupt
truncation of infinite series is equivalent to multiplying infinite series with rectangular
sequence. i.e at the point of discontinuity some oscillation may be observed in resultant
series.
Consider the example of LPF having desired frequency response Hd (ω) as shown in
figure. The oscillations or ringing takes place near band-edge of the filter. This oscillation
or ringing is generated because of side lobes in the frequency response W(ω) 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 -∞ to ∞. Hence h(n) ≠0 for n<0. This means causality
condition is not satisfied by the ideal low pass filter. Hence ideal filters are anti-causal and
thus are not physically realizable.

21. FIR Filters always provides linear phase response. Explain.

The phase or angle of H(ω) is given as

-ω M-1 for |H (ω)| > 0

2
Angle H(ω) =
-ω M-1 +∏ for |H (ω)| < 0
2

In above equations M is constant. Hence Phase of H(ω) is linear function of ω. That is

phase is linearly proportional to frequency. When |H(ω)} changes sign, phase 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
preferred.
FIR filter always provides linear phase response. This specifies that the 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

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performance. In various applications like speech processing, 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 filters
1 Unit sample response of FIR filters is symmetric if it satisfies following 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 following
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
IIR filters.
Impulse response of ideal low pass filter is as shown in fig. In order to have finite
terms we will multiply this infinite series with rectangular window which will generate
desired frequency response. But some oscillation or ringing effect will be observed at the
point of truncation. This effect is known as Gibbs Phenomenon.
As M increases this side lobes becomes narrow and oscillatory behavior decreases.
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 types of windows like rectangular,
triangular, hamming, Hanning window etc.
The windows are selected depending upon the transition width of main lobe and amplitudes
of sidelobes. The windows are selected such that Gibb's phenomenon is reduced.
The particular window is selected depending upon minimum stop band attenuation.

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 and 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 lobe. In
Kaiser window has reduced side lobes and transition band is narrow and hence
mostly used.

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27. What are the constraints to be imposed while designing filters from it pole
zero plot.
Filters can be designed from its pole zero plot. Following two constraints should be imposed
while designing the filters.
1. All poles should be placed inside the unit circle on order for the filter 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 for the filter
coefficients to be real.
In the design of low pass filters, the poles should be placed near the unit circle at points
corresponding to low frequencies ( near ω=0)and zeros should be placed near or on unit
circle at points corresponding to high frequencies (near ω=∏). 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 ideal case.

29. What are frequency transformation techniques. Why they are used.
Frequency transformation techniques are used to generate High pass filter, Bandpass and
bandstop filter from the lowpass filter system function.
Sr Type of transformation Transformation ( Replace s by)
No
ωhp
1 High Pass s
ωhp = Password edge frequency of HPF
(s2 + ωl ωh )
s (ωh - ωl )
2 Band Pass
ωh - higher band edge frequency
ωl - Lower band edge frequency
s (ωh - ωl)
s2+ ωh ωl
3 Band Stop
ωh - higher band edge frequency
ωl - Lower band edge frequency

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 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 called 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 other DSP
applications. A DSP is optimized to perform repetitive mathematical operations such as the
dot product. There are four basic requirements of DSP processor to optimize the
performance They are
1) Fast arithmetic
2) Fast Execution - Dual operand fetch
3) Fast data exchange
4) Circular buffering

Sr Requirements Features of DSP processor

No
1 Fast Arithmetic
2 Fast Execution
3 Fast data Exchange
4 Circular shift operations
Faster MACs means higher bandwidth.
Able to support general purpose math functions, should
have ALU and a programmable shifter function for bit
manipulation.
Powerful interrupt structure and timers
Parallel Execution is required in place of sequential.
Instructions are executed in single cycle of clock called as
True instruction cycle 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.
Multiple registers, Separate program and data memory and
Multiple operands fetch capacity
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

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simultaneously access two memory blanks using two independent sets of buses allowing
operands 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
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 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 are
available for ADSP-21xx family.
The system builder is the software development tool for describing the
configuration of the target system's memory and I/O. The ranges for program
memory(PM) and data memory(DM) are described.
The assembler translated source code into object code modules. The source code is
written in assembly language file (.DSP) Assembler reads .DSP file and generates four
output filed with the same root name. Object file(.OBJ), Code File(.CDE), Initialization File
(.INT), List File(.LST) etc.
The linker is a program used to join together object files into one large object file.
The linker produces a link file which contains the binary codes for all the combined
modules.
A debugger is a program which allows user to load object code program into
system memory, execute the program and debug it.

Difference between DSP and General Purpose Processor

Sr Parameter General Purpose DSP Processor

Processor
1 Instruction Cycle Multiple clock cycles Single cycle of the clock is
required for execution of needed.
one instruction
2 Instruction Sequential Execution Parallel execution - Pipelining
execution involved
3 Operand fetch Sequential Multiple operand fetch
from the memory capability
4 Memories No separate memory Separate program and data
memory
5 Instruction set Mostly Contains Data Contains complex addition,
movement instructions multiplication & shifting
instructions
6 Address PC is used DAG and Program Sequencer
generation
7 On chip address Single pair of buses PMA,DMA, PMD and DMD
and data buses
8 Computational ALU ALU, MAC and Shifter
Units

5. What are the different functions used in MATLAB related with DSP.

Sr Function Application
No
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
convolution.

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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

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