Name :

What & Why ?

Sr No TOPIC PAGE
[

1 Basic Concepts 2

2 Discrete Time signals 3

Discrete Fourier Transform and Fast Fourier

3 7
Transform

4 Analysis of DT system using Z-Transform 19

5 Digital Filters 26

When you study DSP, so many questions come in mind. The more you study
more questions come. Those question needs to be resolved, I have made
the list of few FAQ questions. After you study DSP, you should be in a
position to answer these questions.

You need to Study.

☯  * 2009-2010 * Kiran Talele

 1. BASIC CONCEPTS

” What & Why V 

(1) What is DSP?

Ans : Digital Signal Processing is a technique that converts signals from real world sources
(usually in analog form) into digital data that can then be analyzed. Analysis is performed
in digital form because once a signal has been reduced to numbers, its components can be
isolated, analyzed and rearranged more easily than in analog form.

Eventually, when the DSP has finished its work, the digital data can be turned back into an
analog signal, with improved quality. For example, a DSP can filter noise from a signal,
remove interference, amplify frequencies and suppress others, encrypt information, or
analyze a complex waveform into its spectral components. This process must be handled in
real-time - which is often very quickly. For instance, stereo equipment handles sound
signals of up to 20 kilohertz (20,000 cycles per second), requiring a DSP to perform
hundreds of millions of operations per second.

(2) What are the applications of DSP ?

Ans :
Speech coding & Decoding
Speech encryption and decryption
Speech recognition
Speech Synthesis
Speaker identification
Hi-fi audio encoding & decoding
Modern algorithms
Noise cancellation
Audio equalization
Audio mixing & editing
Vision
Image compression & decompression
Image compositing
Echo cancellation
Spectral estimation

(3) What do you mean by real time signal ? Give example.

Ans : Signal is processed with the same speed it is captured. Signal is captured, sampled and processed
with the same speed. Signal is not stored before processing. Entire input signal never available
before processing. Processed signal can be stored.
For example, in digital telephone system, Signal is captured, Sampled, Processed , Transmitted and
Made it available to the end user. Real Time Processing is Online Processing.

2
Kiran Talele ( talelesir@yahoo.com 9987030881 )
(4) Give one Real Time practical example of DSP system.
Ans : Digital Telephone System.

(5) What do you know about Analog Signal, Digital Signal, CT signal, DT Signal ?
Ans : i] Analog Signal : Signal value can be anything. NO fixed signal level.
Eg x(t) =cos(100π t). continuous Sinusoidal signal
x[n] = { 10.5, 4.7, 3.5, 5.7, 3.8 } sampled signal

ii] Digital Signal : Only two levels +5v and 0. ie. Logically High and Low.
Eg. Binary data

iii] Continuous Time Signal : Signal is defined for every value of time. Signal value can be
anything.
Eg. x(t) =cos(100π t). continuous Sinusoidal signal
Bilevel Signal

iv] Discrete Time Signal : Signal is defined for Discrete instant of Time. NOT for every
value of time. Signal value can be anything.

Eg. x[n] = { 10.5, 4.7, 3.5, 5.7, 3.8 } sampled signal

(6) How Discrete Time signal is obtained ?

Ans : DT signal is obtained by sampling CT signal at regular intervals of time.

x ( t ) t = nTs = x [ nTs ] = x [ n ]

In practical application sampling is implemented using S/H circuit.

(7) What is antialising filter? Can it be Digital filter ? justify.

Ans : When processing the analog signal using DSP System, it is sampled at some rate depending
upon the bandwidth. The rate of sampling is decided by the Nyquist criterion. However,
signals that are found in physical systems will never be strictly bandlimited. To eliminate
signal content beyond the desired bandwidth, antialiasing filter is used.
The filter cannot be a digital filter. This is because antialias filtering is required to be
performed in the analog domain prior to applying the signal to A/D converter where
aliasing would take place.

(8) Let x[t] = 10 cos(100π) + 20 cos( 120πt)-5 sin(50πt). If x(t) is sampled with sampling
frequency Fs = 200 Hz. What will be Discrete Time Signal x[n] at n=0 ?
Ans : 30

3
Kiran Talele ( talelesir@yahoo.com 9987030881 )
 2. DISCRETE TIME SIGNALS

” What & Why V 

(9) What are the classification of signals ?

Ans : DT signal are classified as
(i) Causal Signal, Anti-causal Signal, Bothsided Signal.
(ii) Even Signal, Odd Signal
(iii) Energy Signal, Power Signal
(iv) Periodic Signal, Non periodic Signal
(v) Symmetric, Anti-symmetric
(vi)Finite Length Signal, Infinite Length Signal

(10) What do you mean by Causal signal, Anti-causal Signal and Both-sided signal ?

Ans : If x[n] = 0 for all n < 0

Then x[n] is causal signal.

If x[n] = 0 for all n ≥ 0

Then x[n] is anticausal signal.

If x[n] is neither causal nor anticausal

Then x[n] is bothsided signal.

(11) Give one example of Causal, Anticausal and Bothesided signal.

Ans : Examples :
(i) Causal signal : x1[n] = u[n] x2[n] =( ½ )n u[n]
(ii) Anti-causal signal : x1[n] = u[-n-1] x2[n] =( ½ )n u[–n–1]
(iii) Both sided signal : x1[n] = u[n] + u[-n-1] x2[n] = (2)n u[n] + (3)n u[–n–1]

(12) What is an energy signal ? Give example.

N −1

E=∑
2
Ans : Energy of signal is defined as, x[n]
n =0

If Energy of DT signal is finite (0<E<∞ ) then x[n] is an energy signal.

{ }
Ex : x[n] = ( ½ )n u[n] E = 2 ( finite)
x [n]= 1 2 3 4 E = 30 (finite)
↑

(13) Consider x1[n] is periodic with period = 4 and x2[n] is periodic with period = 6 .
Let x[n] = x1[n] + x2[n]. What will be the period of x[n] ?

4
Kiran Talele ( talelesir@yahoo.com 9987030881 )
(14) What is power signal ? Give example.

N −1
1
Ans : The average power of the x[n] is given as P = Nlim
→∞ N
∑ x[n]
n =0
2

If P is finite and nonzero then x[n ] is a power signal.

Ex x[n] = u[n]

(15) What is symmetric signal ? Give example.

Ans : If x [ n ] = x [ N-1-n ]
Then x[n] is causal symmetric
Ex. Causal Symmetric signal : x [n] = ⎧⎨ − 1 − 2 3 − 2 − 1 ⎫⎬
⎩ ↑ ⎭
(16) What is Anti-symmetric signal ? Give example.
Ans : If x [ n ] = - x [ N-1-n ]
Then x[n] is causal anti-symmetric.
Ex. Causal Anti-Symmetric signal : x [n] = ⎧⎨ − 1 − 2 0 2 1 ⎫⎬
⎩ ↑ ⎭
(17) What is an Even signal ? Give example.
Ans :
If x[ n ] = x [-n ]
Then x [n ] is even signal.

Ex. Even signal : x [n] = ⎧⎨ − 1 − 2 3 − 2 − 1 ⎫⎬ Nonperiodic

⎩ ↑ ⎭

⎧ ⎫ Periodic
x [n ] = ⎨ − 1 −2 3 3 −2 ⎬
⎩ ↑ ⎭

(18) What is an odd signal ? Give example.

Ans :
If x[ n ] = – x [–n ]
Then x [n ] is odd signal

Ex. Odd signal : x [n] = ⎧⎨ − 1 − 2 0 2 1 ⎫⎬ Nonperiodic

⎩ ↑ ⎭

⎧ ⎫ Periodic
x p [n ] = ⎨ 0 − 2 3 − 3 2 ⎬
⎩ ↑ ⎭

(19) What is the sum of odd signal values ?

Ans : Sum of odd signal value is 0.
Ex ⎧ ⎫ Sum = 0
x [n ] = ⎨ − 1 − 2 0 2 1 ⎬
⎩ ↑ ⎭
⎧ ⎫
x p [n ] = ⎨ 0 − 2 3 − 3 2 ⎬ Sum = 0
⎩ ↑ ⎭
5
Kiran Talele ( talelesir@yahoo.com 9987030881 )
(20) How to check whether the given signal is periodic or not ?
Ans : If the digital frequency of the signal is rational number then the signal
is periodic. Otherwise signal is nonperiodic.
3
Ex x[n] = cos (0.6 π n) where w = 0.6π and f = 0.3 =
10
Here f is a rational number, so x[n] is periodic signal with period = 10

(21) What is the concept of digital frequency f ?

F
Ans : Digital Frequency is ratio of Analog Frequency to Sampling frequency. i.e. f =
Fs
(22) What is the range of w and f ?
Ans : Range of Digital frequency ω is ( –π , π ]

⎛ −1 1 ⎤
⎜⎜
2 ⎥⎦
Range of Digital frequency f is ,
⎝ 2
(23) What is the unit of digital frequency w and f ?
Ans : Unit of digital frequency w is radians and f is unit less quantity.

(24) Classify the following signal : Finite Length or Infinite Length :-

x[n] = u[n] + 2 u[n-1] – 3 u[n-5]
Ans : Finite length with length N = 5

(25) What is correlation ?

Ans : Correlation gives a measure of similarity between two data sequences. In this process, two signals
are compared and the degree to which the two signals are similar is computed.

(26) What are the applications of Correlation ?

Ans : Typical applications of correlation include speech processing, image processing and radar systems.
In a radar system, the transmitted signal is correlated with the echo signal to locate the position of
the target. Similarly, in speech processing systems, different waveforms are compared for voice
recognition.
(27) What is the application of Convolution ?
Ans : Application of Convolution is to find output of Digital Filter for any given input signal.
Output of Digital filter y[n] is linear Convolution of input signal x[n] and impulse response of the
filter h[n].

(28) What are the properties of Convolution ?

Ans :
i) Commutative
x [n] * h[n] = h[n] * x[n]
ii) Associative
( x [n] * h1[n] * h2[n] ) = ( x [n] * h1[n] ) * h2[n]
iii) Distributive
x[n] * [h 1[n ] + h 2 [n ]] = x [n] * h1[n] + x[n] * h2[n].

6
Kiran Talele ( talelesir@yahoo.com 9987030881 )
(29) Consider x1[n] is periodic with period N1 = 4 and x2[n] is periodic with period
N2 = 6 . Let x[n] = x1[n] + x2[n]. What will be the period of x[n] ?
Ans : Period N = LCM { N1, N2 } = 12

(30) Let x[n] = δ[n] + 2 u[n] – 2 u[n-4] . Determine which of the following classification is true for x[n].
(a) Periodic, Finite length (b) Periodic, Infinite length
(c) Non periodic, Finite length (d) Non-periodic, Infinite length

Ans : Non periodic, finite length

NOTE :
Linear Shifting of NON-Periodic DT Signals

1) x[ n ] = ⎧ ⎫
⎨ 1 2 3 4 ⎬
⎩ ↑ ⎭

⎧ ⎫
2) x[ n –1 ] = ⎨ 0 1 2 3 4 ⎬
⎩ ↑ ⎭

⎧ ⎫
3) x[ n + 1 ] = ⎨ 1 2 3 4 ⎬
⎩ ↑ ⎭

⎧⎪ ⎫⎪
4) x[ –n ] = ⎨ 4 3 2 1 ⎬
⎪⎩ ↑ ⎪⎭

⎧⎪ ⎫⎪
5) x[ –n + 1 ] = ⎨ 4 3 2 1 ⎬
⎪⎩ ↑ ⎪⎭

⎧ ⎫
6) x[ –n – 1 ] = ⎪ ⎪
⎨ 4 3 2 1 0 ⎬
⎪ ↑ ⎪
⎩ ⎭

----------------------------------------------------------------------------------------------------
Circular Shifting of Periodic DT Signals
1) x[ n ] = { 1, 2, 3, 4 }
2) x[ n-1 ] = { 4, 1, 2, 3 }
3) x[ n+1 ] = { 2, 3, 4, 1 }
4) x[ –n ] = { 1, 4, 3, 2 }
5) x[ –n+1 ] = { 2, 1, 4, 3 }
6) x[ –n–1 ] = { 4, 3, 2, 1 }
-----------------------------------------------------------------------------------------------------------

7
Kiran Talele ( talelesir@yahoo.com 9987030881 )
 3. DFT and FFT

” What & Why V 

(31) Define Discrete Fourier Transform of x[n].

N −1
Ans : X [k ] = ∑n = 0
x[n ] W n
nk

(32) What is the interpretations of DFT coefficients ?

2πk
Ans : DFT gives N values of Fourier Transforms of DT signal x[n] at w = for k = 0,1, 2, ......N −1.
N
2π
They are equally spaced with frequency spacing of
N
(33) How many complex multiplications and additions are required to find DFT ?
Ans : By DFT
(i) Complex Multiplications = N 2
(ii) Complex Additions = N ( N − 1)

(34) How many real multiplications and additions are required to find DFT.
Ans :
Let P = a + j b and Q = c + j d

(1) P X Q = (a+jb) (c+jd)

= (ab – bd )+ j ( bc + ad ) 4 Real Multiplications and 2 Real Additions
For 1 Complex Multiplications we require 4 Real Multiplications. and 2 Real Additions

(2) P+Q = (a+jb)+(c+jd)

= (a + c )+ j ( b + d ) 2 Real Additions

For 1 Complex Addition we require 2 Real Additions

Now, In DFT, Total Complex Multiplications = N2 and Total Complex Additions = N ( N − 1)

So, For N2 Complex Multi. we require 4 N2 Real Multi. and 2 N2 Real Additions
For (N2-N) Complex Additions we require 2( N2-N) Real Additions.
In suumary, Total Real Mulitplications = 4 N2
Total Real Additions = 2 N2 + 2(N2-N) == 4N2– 2N

(35) How many real multiplications and additions are required to find DFT of 32 point signal.?
Ans : By DFT
(i) Real Multiplications = 4 N 2 = 4(32) 2 = 4096
(i) Real Additions = 4 N 2 − 2 N = 40321

8
Kiran Talele ( talelesir@yahoo.com 9987030881 )
(36) How many complex multiplications and additions are required to find FFT ?
Ans : By DFT
N
(i) Complex Multiplications = log 2 N
2
(ii) Complex Additions = N log 2 N

(37) How many real multiplications and additions are required to find DFT of 32 point signal using FFT
algorithm?
Ans : By FFT
(i) Real Multiplications = 2 N log 2 N = 320
(ii) Real Additions = 3 N log 2 N = 480

(38) What is Scaling and Linearity property of DFT ?

Ans : Scaling Property : If signal is multiplied by constant Then DFT is also multiplied by the same
constant. i.e. DFT { a x 1 [n] } = a X 1 [k]

Linearity Property : If signals are added, Then DFT’s are also added.
i.e. DFT { a x 1 [n] + b x 2 [n] } = a X 1 [k] + b X 2 [k

(39) What is the DFT of δ[n] ?

Ans : DFT { δ[n] } = 1

(40) What is the DFT of N pt signal u[n] ?

Ans : DFT {u[n] } = N δ[k]

(41) What is the DFT of 4 pt x[n] where x[n] = δ[n] + u[n] ?

Ans : X[k] = 1+ 4 δ[k]
= { 5, 1, 1, 1 }

(42) What is periodicity property of DFT ?

Ans : DFT equation produces periodic results with period = N
i.e. X[k] = X[k+N] = X[k MOD N] = X[((k))]
Inverse DFT equation produces periodic results with period = N
i.e. x[n] = x[n+N] = x[n MOD N] = x[((n))]

(43) Why DFT results are periodic ?

Ans : DFT results are periodic because twiddle factor is periodic with period = N

(44) DFT gives discrete spectrum or continuous spectrum ? Justify ?

Ans : DFT gives discrete spectrum.
If the signal is periodic then spectrum is discrete and if the signal is non-periodic then spectrum is
continuous. DFT assumes that input signal is periodic and therefore DFT gives discrete spectrum.
9
Kiran Talele ( talelesir@yahoo.com 9987030881 )
(45) What do you mean by spectrum is Discrete or continuous?
Ans : Continuous spectrum is defined for every value of frequency. Discrete spectrum is
defined only at discrete values of frequencies ie. Not defined for every value of
frequency.

(46) Find DFT of x[n] where x[n] = u[n] + 2 u[n-2] – 3 u[n-4]

Ans : Here x[n] = { 1, 1, 3, 3 } By DFT X[k] = {8, -2+2j, 2, -2-2j }

(47) Find DFT of 10 pt x[n] where x[n] = δ[n] + δ[n-5] ?

Ans : X [k ] = 1 + W N5k = 1 + (−1) k

(48) What is Time shift and frequency shift property of DFT ?

Ans : DFT {x [n − m] } = W Nmk X [k ]

{ }
DFT W N− mn x [n] = X [k − m]

(49) What is symmetry property of DFT ?

Ans : If x[n] X[k] Then X[k] = X*[-k].
i.e. If x[n] is real valued signal, then real part of X[k] is symmetric about
k = N/2 and Imaginary part of X[k] is Anti-symmetric about k = N/2.

(50) What is DFT property of EVEN signal ?

Ans : If x[n] is Even , Then X[k] is also Even
i.e.. If x[n] = x[–n] Then X[k] = X[–k]

(51) What is the DFT of real and even signal.?

Ans : If x[n] is Real and Even, Then X[k] is also Real and Even
Eg. x[n] = { 1, 2, 3, 2 }
X[k] = { 8, -2, 0, -2 }

(52) What is the DFT of Imaginary and Even signal ?

Ans : If x[n] is Imaginary and Even
Then X[k] is also Imaginary and Even
Eg. x[n] = { j, 2j, 3j, 2j }
X[k] = { 8j, -2j, 0, -2j }

(53) What is DFT property of ODD signal ?

Ans : If x[n] = – x[–n] Then X[k] = – X[–k]
i.e. If x[n] is Odd , Then X[k] is also Odd.

(54) What is the DFT of real and Odd signal ?

Ans : If x[n] is Real and Odd, Then X[k] is also Imaginary and Odd
Eg. x[n] ={ 0, 2, 0, –2 }
X[k] = { 0, – 4j, 0, 4j }

10
Kiran Talele ( talelesir@yahoo.com 9987030881 )
(55) What is the DFT of Imaginary and Odd signal ?
Ans : If x[n] is Imaginary and Odd
Then X[k] is also Real and Odd
Eg. x[n]={ 0, 2j, 0, – 2j }
X[k] = { 0, 4, 0, – 4 }

(56) If DT signal is expanded in time domain what will be the effect in frequency domain?
Ans : Expansion in time domain corresponds to Compression in frequency domain.
Eg. x[n] = {1,2,3,2 } X[k] = { 8, –2, 0, –2}
Let p[n] = {1, 0, 2, 0, 3, 0, 2,0 } Then P[k] = { 8, –2, 0, –2, 8, –2, 0, –2}

|X[k]| |P[k]|

0 0.5π π 1.5π 2π 0 0.5π π 1.5π 2π

(57) If DT signal is compressed in time domain what will be the effect in frequency domain?
Ans : Compression in time domain corresponds to Expansion in frequency domain.
Eg. x[n] = {1, 0, 2, 0, 3, 0, 2,0 } X[k] = { 8, –2, 0, –2, 8, –2, 0, –2}
Let p[n] = {1,2,3,2 } Then P[k] = { 8, –2, 0, –2 }

|X[k] |P[k]|
|

0 0.5π π 1.5π 2π 0 0.5π π 1.5π 2π

(58) If DT signal is appended by zeros in time domain what will be the effect in frequency domain?
Ans : Eg. x[n] = {1,2,3,2 } X[k] = { 8, –2, 0, –2} N=4 pt
Let p[n] = {1, 2, 3, 2, 0, 0, 0, 0 } N=8 pt

|X[k]| |P[k]|

0 0.5π π 1.5π 2π 0 0.5π π 1.5π 2π

As the length of signal increases, the frequency spacing decreases.

The number of points per unit length i.e. resolution of the spectrum increases.
Therefore the approximation error in the representation of the spectrum decreases.
11
Kiran Talele ( talelesir@yahoo.com 9987030881 )
(59) What is convolution property of DFT ?
Ans : Convolution in time domain corresponds to multiplication in frequency domain.
If x[n] X[k] and
h[n] H[k]
Then
DFT { x[n] ⊗ h[n] } = X[k] H[k]
(60) What is correlation property of DFT ?
Ans : If x[n] X[k] and
h[n] H[k]
Then
DFT { x[n] o h[n] } = X[k] H * [k]

(61) How to find energy of signal from its DFT ?

Ans : According to parseval’s energy theorem, Energy of the signal is given by,
1 N −1
E= ∑
N k =0
| X [k ] | 2

(62) Are FFT's limited to sizes that are powers of 2?

Ans : No. The most common and familiar FFT's are "radix 2". However, other radices are
sometimes used, which are usually small numbers less than 10. For example, radix-4 is
especially attractive because the "twiddle factors" are all 1, -1, j, or -j, which can be
applied without any multiplications at all.

Also, "mixed radix" FFT's also can be done on "composite" sizes. In this case, you break a
non-prime size down into its prime factors, and do an FFT whose stages use those
factors. For example, an FFT of siz 1000 might be done in six stages using radices of 2
and 5, since 1000 = 2 * 2 * 2 * 5 * 5 * 5. It might also be done in three stages using radix
10, since 1000 = 10 * 10 * 10.

(63) What is an FFT "radix"?

Ans : The "radix" is the size of an FFT decomposition. In the example above, the radix was 2.
For single-radix FFT's, the transform size must be a power of the radix. In the example
above, the size was 32, which is 2 to the 5th power.
(64) What is an "in place" FFT?

Ans : An "in place" FFT is simply an FFT that is calculated entirely inside its original sample
memory. In other words, calculating an "in place" FFT does not require additional buffer
memory (as some FFT's do.)

(65) What is "bit reversal"?

Ans : "Bit reversal" is just what it sounds like: reversing the bits in a binary word from left to
write. Therefore the MSB's become LSB's and the LSB's become MSB's. But what does
that have to do with FFT's? Well, the data ordering required by radix-2 FFT's turns out to
be in "bit reversed" order, so bit-reversed indexes are used to find order of input and
output.
It is possible (but slow) to calculate these bit-reversed indices in software; however, bit reversals are trivial
when implemented in hardware. Therefore, almost all DSP processors include a hardware bit-reversal
indexing capability (which is one of the things that distinguishes them from other microprocessors.)
12
Kiran Talele ( talelesir@yahoo.com 9987030881 )
(66) How efficient is the FFT?
Ans : The DFT takes N2 operations for N points. Since at any stage the computation
required to combine smaller DFTs into larger DFTs is proportional to N, and
there are log2(N) stages (for radix 2), the total computation is proportional to N log2(N).
Therefore, the ratio between a DFT computation and an FFT computation for the same N
is proportional to N / log2(N). In cases where N is small this ratio is not very significant,
but when N becomes large, this ratio gets very large. (Every time you double N, the
numerator doubles, but the denominator only increases by 1.)

(67) FFT is faster than DFT. Justify.

Ans : FFT produces fast results because calculations are reduced by decomposition technique.
In FFT, N pt DFT is decomposed into two N/2 pt DFT’s, N/2 pt DFT is decomposed into N/4 pt
DFT’s and so on… Decomposition reduces calculations. FFT algorithms are implemented using
parallel processing techniques. Because calculations are done in parallel, FFT produces fast
results.

Complex Multiplications : DFT FFT

N N
N2 log 2 N
2
16 256 32
32 1,024 80
64 4,096 192
256 65,536 1,024
512 2,62,144 2,304
1024 10,48,576 5,120

(68) What do you mean by Decimation ?

Ans : Decimation means Sampling.

(69) Why Radix-2 algorithms are fast compared to radix-3 algorithms. ?

Ans : In FFT, N pt DFT is decomposed into two N/2 pt DFT’s, N/2 pt DFT is decomposed into N/4 pt
DFT’s and so on… Decomposition reduces calculations This process continues till further
decomposition is not possible. In radix-2 last level of decomposition is when the length of signal
becomes 2 pt.
For minimum calculations there must be maximum levels of decompositions. In Radix-2
algorithms, we get maximum levels of decompositions and therefore radix-2 algorithms requires less
calculations. Radix-2 algorithms are fast algorithms.

(70) Which algorithm is more powerful : DIT-FFT or DIF-FFT ?

Ans : Computationally, both the algorithms are exactly same.

(71) What is the order of input and output sequence in 8 pt DIT-FFT ?

Ans : x[n] = { x[0], x[4], x[2], x[6], x[1], x[5], x[3], x[7] }
X[k] = { X[0], X[1], X[2], X[3], X[4], X[5], X[6], X[7] }
13
Kiran Talele ( talelesir@yahoo.com 9987030881 )
(72) What is the order of input and output sequence in 8 pt DIF-FFT?
Ans : x[n] = { x[0], x[1], x[2], x[3], x[4], x[5], x[6], x[7] }
X[k] = { X[0], X[4], X[2], X[6], X[1], X[5], X[3], X[7] }

(73) How to find CC using DFT ?

Ans : To find CC of x[n] and h[n] using DFT,
(i) Select N
Let N = max(L,M) where L is the length of x[n] and M is length of h[n],
(ii) Append x[n] by (N-L) zeros and Append h[n] by (N-M) zeros
N −1
(iii) Find X[k] where X [k ] = ∑ x[n] W Nnk
n= 0
N −1
(iv) Find H[k] where H [k ] = ∑ h[n] W Nnk
n= 0
(v) Let Y[k] = X[k] H[k].
N −1
∑ Y [k ] WN− nk
i
(vi) Find y[n] where y[n] =
N
K =0

Always explain wrt diagram

x[n] DFT/FFT
X[k]
Y[k] y[n]
× iDFT /

h[n] DFT/FFT
H[k]

(74) How to find CC using FFT ?

Ans : To find CC of x[n] and h[n] using FFT,
(i) Select N
Let N = max(L,M) where L is the length of x[n] and M is length of h[n],
(ii) Append x[n] by (N-L) zeros and Append h[n] by (N-M) zeros

(iii) Find X[k] by using N point DIT-FFT / DIF-FFT flowgraph

(iv) Find H[k] by using N point DIT-FFT / DIF-FFT flowgraph
(v) Let Y[k] = X[k] H[k].
(vi) Find y[n] by Inverse FFT.

By Inverse FFT, y [n] =

N
1
( FFT {Y * [k ]} )
*

Always explain wrt diagram.

14
Kiran Talele ( talelesir@yahoo.com 9987030881 )
(75) How to find LC using CC ?
Ans : To find LC of x[n] and h[n] using CC,
(i) Select N:
Let N ≥ L + M – 1 where L is the length of x[n] and M is length of h[n],
(ii) Append x[n] by (N-L) zeros and Append h[n] by (N-M) zeros
(iii) Perform N point Circular convolution of x[n] and h[n]

(76) How to find LC using DFT /FFT ?

Ans : : To find LC of x[n] and h[n] using DFT/FFT,
(i) Select N
Let N ≥ L + M – 1 where L is the length of x[n] and M is length of h[n],
(ii) Append x[n] by (N-L) zeros and Append h[n] by (N-M) zeros
(iii) Perform N point Circular convolution of x[n] and h[n] using DFT/FFT.
Find N point X[k] and H[k]
Let Y[k] = X[k] H[k].
Find y[n] by Inverse DFT/FFT.

Always explain wrt diagram.

(77) What are the applications of FFT. ?

Ans : (i) Linear Filtering i.e. to find output of digital filter for any given input sequence
(ii) Spectral Analysis i.e. to find magnitude spectrum and phase spectrum
(iii) Circular Correlation ie to find degree of similarity between two signals.

(78) How to find output of the filter using DFT ?

Ans : Output of the filter is Linear convolution of impulse response with the input of the signal.
To find output means to find LC by DFT.

(79) How to find output of the FIR filter using FFT ?

Ans : In FIR filter length of h[n] is finite. Output of the filter is always Linear convolution of impulse
response with the input of the signal. To find output i.e. to find LC by FFT
(i) Select N
Let N ≥ L + M – 1 where L is the length of x[n] and M is length of h[n],
(ii) Append x[n] by (N-L) zeros and Append h[n] by (N-M) zeros
(iii) Perform N point Circular convolution of x[n] and h[n] using FFT.
* Find N point X[k] and H[k] by using FFT flowgraph.
* Let Y[k] = X[k] H[k].

* Find y[n] by Inverse FFT. y[n] =

1
N
( { })
FFT Y * [k ]
*

Always explain wrt diagram.

15
Kiran Talele ( talelesir@yahoo.com 9987030881 )
(80) How to find output of the IIR filter using DFT / FFT?
Ans : Output of the filter is Linear convolution of impulse response with the input of the
signal.
To find output means to find LC by DFT/FFT. Length of h[n] in IIR filter is infinite.
So, DFT/FFT implementation of infinite length signals is not possible.

(81) What is the length of linearly convolved signals ?

Ans : Length of linearly convolved signal is always equal to N = L + M – 1 where L is length of first
signal and M is length of second signal.

(82) What is periodic convolution ?

Ans : Periodic convolution is convolution of two periodic signals of the same period. When two
periodic signals are periodic with common period, periodic convolution is similar to circular
convolution.

(83) What is the difference between circular convolution and periodic convolution ?
Ans : In periodic convolution input signals are originally periodic with common value of period.
In circular convolution, if input signals are not periodic then they are assumed to be periodic
with period = N where N = max(L,M) where L is the length of first signal and M is length of
second signal.

(84) What do you mean by aliasing in circular convolution ?

Ans : In circular convolution if value of N < L+M-1 then last M-1 values of y[n] wraps around gets
added with first M-1 values of y[n]. This is called aliasing.

(85) Why FFT is used to find output of FIR filter ? Justify.

Ans : FFT produces fast results because in practical applications FFT algorithms are implemented
using parallel processing techniques. Because in FFT calculations are done in parallel, FFT
produces fast results.

(86) What are the limitations of filtering by FFT algorithms? Justify.

Ans : (i) NOT suitable for real time applications :
FFT algorithms are implemented using parallel processing techniques. When FFT is used
input is applied in parallel i.e simultaneously. For real time applications entire input signal is
not available. So FFT algorithms can not be used.
(ii) NOT suitable for Long Data Sequence.
As the length of the input sequence increases, the no of stages in FFT will also increase
proportionally and so the delay increases, processing time at each stage increases.

(87) How to find output of FIR filter for long input sequence.
Ans : In FIR filter length of h[n] is finite. Output of the filter is always Linear Convolution of impulse
response with the input of the signal. To find output of digital FIR filter FFT technique is used. But
for Long data sequence, direct FFT technique is not suitable.
For long data sequence, Overlap Add Method using FFT or Overlap Save Method using FFT is
used.

(88) What is Overlap Add Method?

(89) What is Overlap Save Method?
16
Kiran Talele ( talelesir@yahoo.com 9987030881 )
(90) How to find output of FIR filter for real time input signal.?
Ans : In real time application entire input is not available and input signal has to be
processed online. Length of input signal depends on application. It can be long
sequence also.
In FIR filter length of h[n] is finite. Output of the filter is always Linear Convolution of impulse
response with the input of the signal.
To find output of digital FIR filter, Overlap Add Method using FFT or Overlap Save Method using
FFT is used.

(91) How to find output of IIR filter for real time input signal.?
Ans : In real time application entire input is not available and input signal has to be processed online.
Length of input signal depends on application. It can be long sequence also.
In IIR filter length of h[n] is infinite. Output of the filter is always Linear Convolution of impulse
response with the input of the signal. To find output of digital IIR filter, Overlap Add Method
using FFT or Overlap Save Method using FFT can not be used.
Output of digital IIR filter is calculated using difference equation recursively.

(92) How to find output of IIR filter for long input sequence.?
Ans : In IIR filter length of h[n] is infinite. Output of the filter is always Linear Convolution of impulse
response with the input of the signal. To find output of digital IIR filter, Overlap Add Method
using FFT or Overlap Save Method using FFT can not be used.
Output of digital IIR filter is calculated using difference equation recursively.

(93) What is DTFT ?

Ans : DTFT is Fourier Transform of DT signal that converts the sampled DT signal from time domain
to frequency domain. Frequency domain representation parameters are magnitude and phase.
DTFT gives frequency response that includes magnitude response and phase response.

(94) If DTFT is Fourier Transform of DT signal then What is DFT ?

Ans : DFT is frequency sampling of DTFT. When DTFT is sampled in frequency domain we get DFT.

(95) Describe the relation between DFT and DTFT.

Ans : DFT is frequency sampling of DTFT. When DTFT is sampled in frequency domain with frequency
2π
spacing of w = we get DFT coefficients. X [ k ] = X ( w )
N 2π k
w=
N

(96) Derive DFT equation . [ Refer note book ]

(97) Why DFT ? What is need of Sampling DTFT ?

Ans : In digital domain for processing, input has to be discrete. For frequency domain analysis, DT
signal is converted to frequency domain. Frequency domain representation of DT signal is
continuous, NOT discrete. For processing in digital domain we need to take sampled values. The
frequency samples thus obtained are called DFT coefficient. That is what DFT is.

17
Kiran Talele ( talelesir@yahoo.com 9987030881 )
(98) How to find DFT of infinite length sequence ?
Ans : To find DFT of infinite length sequence x[n]:
∞
(i) Find DTFT of x[n] i.e. X ( w) = ∑ x[n] e − jnw
n = −∞

(ii) Find DFT by frequency sampling DTFT. i.e. X [ k ] = X ( w) 2πk

w=
N
DFT coefficients can be obtained by evaluating DFT equation.

(99) What is Power Density Spectrum of Periodic DT Signals ?

Ans :
N −1
1
∑
2
The average power of periodic DT signal is given by P = x [n]
N n=0
2
1 N −1 N −1
∑ x [ n] = ∑ C k
2
According to Parseval’s theorem, P=
N n=0 k= 0
2
The coefficients Ck for k =0, 1, 2…..N-1 is the distribution of power as a function of
frequency is called the power density spectrum of the DT periodic signal

(100) What is Energy Density Spectrum of DT Aperiodic Signals

Ans :
∞
∑
2
The energy of DT signal x[n] is E = x [ n]
n = −∞
π 2
∞
1
∑ ∫
2
According to parseval’s theorem, E = x [n] = X ( w) dw
−∞
2π
−π
2 *
Let S x ( w) = X ( w) = X ( w) X ( w)
Sx (w) is the function of frequency and it is called energy density spectrum of x [n].
∞ π
1
E= ∑ = = ∫
2
x [n] Sx ( w) dw.
−∞ 2π
−π

(101) Find DTFT and Energy Density Spectrum of x[n] = u[n].

Ans : Energy of u[n] is infinite. Therefore u[n] is not energy signal.

Fourier Transform is defined only for energy signal.

18
Kiran Talele ( talelesir@yahoo.com 9987030881 )
(102) What is the necessary condition to find DTFT of any signal. ?
Ans : To find DTFT of any signal the necessary condition is, signal must be an energy
signal. It must be absolutely summable.

(103) DTFT gives continuous spectra or discrete spectra?.

Ans : When signal is periodic spectrum is Discrete. If the signal is not-
periodic then spectrum is always continuous.
DTFT is fourier transform of Non-periodic signals. Therefore DTFT gives
continuous spectra.

(104) How to use FFT algorithm to find IDFT ?

Ans : By IFFT equation we get, x [n] =
1
N
(
FFT { X * [k ] }
*
)
Algo : (i) Find X*[k]
(ii) Find FFT (X*[k] ) using DIT-FFT/DIF-FFT flowgraph,
Here same flowgraph is required to find FFT {X*[k]}result..
(iii)Find x[n]

(105) What is the difference between DFT and DTFS ? [Refer Notes]
(106) What is the relation between DFT and DTFS ? [Refer Notes]
(107) What is the relation between DFT and DTFT ? [Refer Notes]
(108) What is the relation between DTFT and ZT ? [Refer Notes]
(109) What is the relation between DFT and ZT ? [Refer Notes]
(110) How to find DFT of Two N point Real Sequence using a single N point FFT ?
(111) How to find DFT of 2N point DFT of real valued sequence using a single N point FFT
algorithm? [Refer Notes]

---------------------------------------------------------------------

19
Kiran Talele ( talelesir@yahoo.com 9987030881 )
 4. Z-Transform

” What & Why V 

(112) Why ZT is used for frequency domain analysis of DT systems

instead of DTFT ?
Ans : DTFT of every input signal is not possible. DTFT of u[n] is not possible because
u[n] is not an energy signal. However ZT of u[n] is possible. Therefore ZT is used
for analysis.
(113) What is the ZT of δ [n] and u[n]
Ans : ZT {δ [n]}=1 and ZT{u[n]} = z/(z-1)

(114) What is the ZT of x[n] = (2) n u[n] ⎧ z ⎫

Ans : X(z) = z/(z-2) ROC : |z| > 2 ( ) ⎪
ZT a n u[ n] = ⎨ z − a
z 〉 ⎪
⎬
a
⎪⎩ 0 Otherwise ⎪⎭
(115) Let x[n] = (4) n u[n] ⎧ −z ⎫
What is X(z) at z = 6 and z = 2 ?
( ) ⎪
ZT a n u[ −n − 1] = ⎨ z − a
z 〈 a ⎪
⎬
⎪⎩ 0 Otherwise ⎪⎭
Ans : X(z) = z/(z-4) ROC : |z| > 4
(i) At z = 6 X(z) = 6/2 = 3
(ii) At z = 2 X(z) = ∞

(116) What is the concept of ROC ?

Ans : ROC gives the set of values of Z for which X(z) is finite. Every value of Z in the
ROC gives X(z) finite.
(117) What is the ROC condition for causal signal. ? Why ? Justify with
example.
Ans : ROC is |z| > | Largest value of POLE |
Ex x[n] = (2)n u[n] + (3)n u[n]

NOTE : If x[n] is right handed sequence, the ROC extends outward from the
outermost finite pole in X ( z ) to z = ∞

Sequence ROC
1 x[n] = Entire Z-plane

2 x[n] = |Z| > 0

3 x[n] = an u[n] |Z| > |a|
4 x[n] = an u[n] + bn u[n] |Z|> max{ |a |,|b| }
5 x[n] = (-3)n u[n] + (2)n u[n] |Z| > 3

20
Kiran Talele ( talelesir@yahoo.com 9987030881 )
(118) What is the ROC condition for Anti-causal signal? Why ?
Justify with example.
Ans : ROC is |z| < | Lowest value of POLE |
Ex x[n] = (2)n u[-n–1] + (3)n u[-n–1]

NOTE : If x[n] is Left handed sequence, the ROC extends inward from the
innermost finite pole in X(z) to z = 0

Sequence ROC
1 x[n] = { 1, 2, 3, 0 } |Z| < ∞

2 x[n] = an u[-n-1] |Z| < |a|

3 x[n] = an u[-n-1] + bn u[-n-1] |Z| < min { |a|, |b| }
4 x[n] = (-3)n u[-n-1] + (2)n u[-n-1] |Z| < 2

(119) What is the ROC condition for Both-sided signal. ? Why ? Justify
with example.
Ans : ROC condition for both sided signal is bounded between two POLES.
Ex x[n] = (2)n u[n] + (3)n u[-n]

NOTE : If x[n] is two sided sequence, the ROC consist of a ring in the Z plane,
bounded by interior and exterior pole.]
Sequence ROC
1 x[n] = an u[n] + bn u[-n-1] |b| > |z| > |a|
2 x[n] = (2)n u[n] + (3)n u[-n-1] 3 > |z| > 2

3 x[n] = (3)n u[n] + (2)n u[-n-1] Not possible

x[n] = (2)nu[n] + (3)nu[n] +

4 4 > |z| > 3
(–4)nu[-n-1] + (5)nu[-n-1]

(120) What is DT sy stem ?

Ans : A DT system is a device or algorithm that operates on a DT signal according to some well defined
rule, to produce another DT signal. In general a DT system can be thought as a set of operations
performed on the input signal x[n] to produce the output signal y[n].

(121) What are the classification of DT systems ?

Ans : Systems are classified as,

(1) Static (Memorylees ) / Dynamic (Memory System) :-

(2) Linear / Non Linear System.
(3) Causal / Non Causal System
(4) Time Invariant / Time Variant System.
(5) Stable / Unstable system
21
Kiran Talele ( talelesir@yahoo.com 9987030881 )
(122) Explain classification of DT system

(1) Static (Memorylees ) / Dynamic (Memory System) :-

A DT system is called static or memoryless if it output at any instant depends on the input
sample at the same time and not on past or future samples of the input. If the system is
not static then it is dynamic.

(2) Linear / Non Linear System.

A system that satisfies the superposition principle is called Linear System.
If a system is Linear then,
T { a . x1[n] + b x2[n] } = a1 T {x1 [n]} + a2 T {x2 [n] }
If a system does not satisfy the superposition principle then it is Non Linear System.

(3) Causal / Non Causal System

A system is said to be causal if the output of the system at any time depends only on
present and past values of input and does not depend on future values of input.
If the system is not causal then it is Non casual. For non causal system output depends on
future values of input.

(4) Time Invariant / Time Variant System.

A system is called Time Invariant if a time shift in the input signal causes a time shift in
the output signal. Otherwise the system is Time Variant System.

(5) Stable / Unstable system.

A system is said to be bounded input, bounded output stable if and only if every bounded
input produces a bounded output.

(123) What is Impulse response ? Step response ?

Ans : Impulse Response is output of the system when input is δ[n].
Step Response is output of the system when input is u[n].

(124) What is zero input response ?

Ans : If the initial state of the system is NOT zero and the input x[n] = 0 to all n, then the output of the
system with zero input is called the zero input response or natural response or free response of the
system.

(125) What is zero state response ?

Ans : If the initial state of the system is zero and the input x[n] ≠ 0 then the output of the system with
non zero input is called the zero state response or forced response of the system.

22
Kiran Talele ( talelesir@yahoo.com 9987030881 )
(126) What is zero state step response ?
Ans : If the initial state of the system is zero and the input x[n]=u[n] then the output of the
system is called zero step response of the system.

(127) What is Transient response ?

Ans : Transient response of the system is the response of the system that decays to zero.

(128) What is Steady State Response ?

Ans : Everlasting response of the system that depends on magnitude response and phase response of the
system is steady state response of the system.

(129) What is Infinite Impulse Response ?

Ans : When length of h[n] is infinite it is called infinite impulse response. E.g. h[n] = ( ½ )n u[n]

(130) What is Finite Impulse Response ?

Ans : When length of h[n] is finite it is called finite impulse response, E.g. h[ n ] = { 1 2, 3, 4 }
↑
(131) What is frequency response ?
Ans : Frequency response means magnitude response and phase response.

(132) What is Magnitude Response ?

Ans : Magnitude Response = Magnitude of Numerator
Magnitude of Denominato r

2 2
Where Magnitude = (Real) + (Imaginary)

(133) What is Phase Response?

Ans : Phase Response = Angle of Numerator – Angle of Denominator
⎡ − 1 ⎛ Imaginary ⎞
⎢ tan ⎜ ⎟ When Real > 0
Where ⎝ Real ⎠
Angle = ⎢
⎢ − 1 ⎛ Imaginary ⎞
⎢180 + tan ⎜ ⎟ When Real < 0
⎣ ⎝ Real ⎠

(134) How to obtain Frequency Response Graphically ?

Ans : In Graphical method, the frequency response at a given frequency w is determined by
the ratio of the product of the zero vectors with the product of pole vectors.

Product of distance from zeros

Magnitude Response =
Product of distance from poles

Phase Response = Summation of angles from ZEROS – Summation of angles from POLES.

23
Kiran Talele ( talelesir@yahoo.com 9987030881 )
(135) Magnitude spectrum is continuous or discrete ?
Ans : If the signal is periodic then magnitude spectrum is discrete and If the signal is not-
periodic then spectrum is continuous function of w.

(136) What is a digital resonator ?

Ans : A digital resonator is essentially a narrowband bandpass filter.

(137) What is eigen value of the system ?

Ans : Eigen-function of a system is an input signal that produces an output that differs
from the input by a constant multiplicative factor. The multiplicative factor is
called an eigen value of the system.

(138) How to find value of DT signal at infinity . ?

⎛ z −1⎞
Ans : By final value theorem we can find x[∞]. x(∞) = lim ⎜ ⎟ X ( z)
z →1 ⎝ z ⎠
(139) What is Transfer function of DT system ?
Ans : The Z – Transform H(z) of an impulse response h[n] is known as the system function or
transfer function of the system

(140) What are different realization methods of digital filters ?

Ans :
IIR FILTER LINEAR PHASE FIR FILTER.
1 Direct Form Realization Direct Form Realization
a) DF-I -DF-I
b) DF-II -DF-II
2 Lattice Realization Lattice Realization
3 Linear Phase Realization
4 Frequency Sampling Realization

(141) Impulse response of Digital Low Pass filter is given by h[n] ={ 3, 2, 1, 2, 3 }. What will
be the output of the filter for any given input x[n] ?

x[n] y[n]
Digital Filter

24
Kiran Talele ( talelesir@yahoo.com 9987030881 )
€ Always remember this →

[i] To find Zero State Response (ZSR)

(i) h[n]
x[n] (ii) H[z] y[n]
(iii) Difference Equation
(iv) Realization Diagram
ZT (v) Pole Zero Plot IZT

X(z) Y(z)
H(z)

[ii] Relationship Diagram

(i) Take ZT
P.Z.
(ii) Group the terms
with Y(z) & X(z)
(iii) Arrange in terms
of Y(z)/X(z)

IZT
D.E. H(z) h[n]

ZT

Put z = ejw

(i) Write H(z) in –ve powers of z

H(ejw) OR H(w)
(ii) Let H(z)=Y(z)/X(z) R.D. Freq. Response
(iii) Cross Multiply i.e. DTFT
(iv) Take IZT
`

25
Kiran Talele ( talelesir@yahoo.com 9987030881 )