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Digital Signal Processing Qn.s and Answers

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0% found this document useful (0 votes)
53 views5 pages

Digital Signal Processing Qn.s and Answers

And

Uploaded by

polikkgate
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Digital Signal Processing

Define DFT of a discrete time sequence.


The DFT is used to convert a finite discrete time sequence x(n) to an N point
frequency domain sequence X(k).The N point DFT of a finite sequence x(n) of length
L,(L<N) is defined as

List any four properties of DFT


A). Periodicity B).Linearity
C). Time reversal D).Circular time shift
E). Duality F).Circular convolution
G). Symmetry H).Circular symmetry
State periodicity property with respect to DFT.
If x(k) is N-point DFT of a finite duration sequence x(n), then x(n+N) = x(n) for all n.
X (k+N) = X(k) for all k.
State periodicity property with respect to DFT.
If X1(k) and X2(k) are N-point DFTs of finite duration sequences x1(n) and x2(n),
then DFT [a X1(n) +
b X2(n)] = a X1(k) + b X2(k), a, b are constants.
State time reversal property with respect to DFT.
If DFT[x(n)] =X(k), then
DFT[x((-n))N] = DFT[x(N-n)] = X((-k))N = X(N-k)
Define circular convolution.
Let x1(n) and x2(n) are finite duration sequences both of length n with DFTs x1(k)
and x2(k). If X3(k) =
X1(k) X2(k), then the sequence X3(k) can be obtained by circular convolution.
What is the need for DFT?
DFT is used for analysis the both periodic and a periodic signals.
What is zero padding? What are its uses?
Let the sequence x (n) has a length L. If we want to find the N-point DFT (N>L) of the
sequence x(n), we have to add (N-L) zeros to the sequence x(n). This is known as
Zero padding.
The uses of zero padding are
1) We can get better display of the frequency spectrum.
2) With zero padding the DFT can be used in linear filtering.
Why FFT is needed?
The direct evaluation of DFT requires complex multiplications and N - N complex
additions. Thus for large values of N direct evaluation of the DFT is difficult. By using
FFT algorithm the number of complex computations can be reduced.
What is FFT?
The Fast Fourier Transform is an algorithm used to compute the DFT. It makes use
of the symmetry and periodicity properties of twiddle factor to effectively reduce the
DFT computation time. It is based on the fundamental principle of decomposing the
mutation of DFT of a sequence of length N into successively smaller DFTs.
How many multiplications and additions are required to compute N point DFT
Using Radix-2 FFT?
The number of multiplications and additions required to compute N point DFT Using
radix-2 FFT are N log2 N and N/2 log2 N respectively.
What is meant by radix-2 FFT?
The FFT algorithm is most efficient in calculating N point DFT. If the number of
output points N can be expressed as a power of 2 that is N=2M, where M is an
integer, then this algorithm is known as radix-2 algorithm.
What is DIT algorithm?
Decimation-In-Time algorithm is used to calculate the DFT of a N point sequence.
The idea is to break the N point sequence into two sequences, the DFTs of which
can be combined to give the DFT of the original N point sequence. This algorithm is
called DIT because the sequence x(n) is often spitted into smaller sub- sequences.
What DIF algorithm?
It is a popular form of the FFT algorithm. In this the output sequence X(k) is divided
into smaller and smaller sub-sequences , So it is called the name is “Decimation In
Frequency”.
What are the applications of FFT algorithm?
The applications of FFT algorithm includes
1) Linear filtering
2) Correlation
3) Spectrum analysis
Distinguish between linear convolution and circular convolution of two
sequences.

What are the differences and similarities between DIF and DIT algorithms?
Differences:
1) The input is bit reversed while the output is in natural order for DIT, whereas for
DIF the output is bit reversed while the input is in natural order.
2) The DIF butterfly is slightly different from the DIT butterfly, the difference being
that the complex multiplication takes place after the add-subtract operation in DIF.
Similarities:
Both algorithms require same number of operations to compute the DFT. Both
algorithms can be done in place and both need to perform bit reversal at some
Place during the computa
What is meant by radix-2 FFT?
If the number of output points N can be expressed as a power of 2, i.e., N = 2M
Where M is an integer then
this algorithm is known as radix-2 algorithm.
What is DIT radix-2 algorithm?
The radix 2 DIT FFT is an efficient algorithm for computing DFT. The idea is to break
N point sequence
in to two sequences, the DFT of which can be combined to give DFT of the original
N-point sequence. Initially
the N point sequence is divided in to two N/2 point sequences, on the basis of odd
and even and the DFTs of them
are evaluated and combined to give N-point sequence. Similarly the N/2 DFT s are
divided and expressed into the combination of N/4 point DFTs. This process is
continued until we left with 2-point DFT‟s.
What is DIF radix-2 algorithm?
1. The radix 2 DIF FFT is an efficient algorithm for computing DFT in this the out put
sequence x(k) is divided into smaller and smaller.
2. The idea is to break N point sequence in to two sequences ,x1(n) and x2(n)
consisting of the first N/2 points of x(n)and last N/2 points of x(n) respectively. Then
we find N/2 point sequences f(n) and g(n Similarly).
3. The N/2 DFT s are divided and expressed in to the combination of N/4 point DFT‟
s. This process is continued
until we left with 2-point DFT‟s.
What are the differences between DIT and DIF algorithms?
* For DIT the input is bit reversed and the output is in natural order, and in DIF the
input is in natural order and output is bit reversed.
* In butterfly the phase factor is multiplied before the add and subtract operation but
in DIF it is multiplied after add-subtract operation.
1).DIT – Time is decimated and input is bi reversed format output in natural order
2).DIF – Frequency is decimated and input is natural order output is bit reversed
Format
What is meant by in place in DIT and DFT algorithm?
An algorithm that uses the same location to store both the input and output
sequence is called in-place algorithm.
How many stages are there for 8 point DFT
3 stages
How many multiplication terms are required for doing DFT by expressional?
Expression Method and FFT method
Expression –N2 FFT - N /2 logN

How will you perform linear convolution from circular convolution?


* Calculate the value of „N‟, that means number of samples contained in linear
convolution.
* By doing zero padding make the length of every sequence equal to number of
samples contained in
linear convolution.
* Perform the circular convolution. The result of linear and circular convolution will be
same.
What methods are used to do linear filtering of long data sequences?
* Overlap save method. * Overlap adds method.
What is the disadvantage of direct computation of DFT?
For the computation of N-point DFT, N2 complex multiplication and N2 – N complex
additions are
required. If the value of N is large then the number of computations will go into lakhs.
This proves inefficiency of
direct DFT computation.
What is the way to reduce number of arithmetic operations during DFT
computation?
Numbers of arithmetic operations involved in the computation of DFT are greatly
reduced by using
different FFT algorithms as follows,
1). Radix-2 FFT algorithm.
A).Radix-2 Decimation In Time (DIT) algorithm.
B).Radix-2 Decimation In Frequency (DIF) algorithm.

For IIR filter realization what is required?


Present, past, future samples of input and past values of output are required.
Why IIR systems are called recursive systems?
Because the feedback connection is present from output side to input
Which types of structures are used to realize IIR systems?
1). Direct form structure 2).Cascade form structure 3).Parallel form structure
Why direct form-II structure is preferred most and why?
The numbers of delay elements are reduced in direct form-II structure compared to
direct form-I
structure. That means the memory locations are reduced in direct form-II structure.
. Why direct form-I and direct form-II are called as direct form structures?
The direct form-I and direct form-II structures are obtained directly from the
corresponding transfer
function without any rearrangements. So these structures are called as direct form
structures.
What is advantage of direct form structure?
Implementation of direct form is very easy.
Give the disadvantage of direct form structure?
Both direct form structures are sensitive to the effects of quantization errors in the
coefficients. So
practically not preferred
What is the use of transpose operation?
If two digital structures have the same transfer function then they are called as
equivalent structures. By
using the transpose operation, we can obtain equivalent structure from a given
realization structure.
What is transposition or flow graph reversal theorem?
If we reverse the directions of all branch transmittances and interchange input and
output in the flow graph
then the system transfer function remains unchanged.
How a transposed structure is obtained?
1). Reverse all signal flow graph directions.
2). Change branching nodes into adders and vice-versa.
3). Interchange input and output.
Why feed back is required in IIR systems?
It is required to generate infinitely long impulse response in IIR systems.
What are the factors that influence selection of DSPs?
1). Architectural features
2).Execution speed
3).Type of arithmetic
4).Word length
What are the classification digital signal processors?
The digital signal processors are classified as
(i) General purpose digital signal processors. (ii) Special purpose digital signal
processors.
What are the applications of PDSPs?
Digital cell phones, automated inspection, voicemail, motor control, video
conferencing, noise cancellation, medical imaging, speech synthesis, satellite
communication etc.
Give some examples for fixed point DSPs.
TM32OC50, TMS320C54, TMS320C55, ADSP-219x, ADSP-219xx..

What are the significance of chebyshev filter?


1. The magnitude response of the chebyshev filter exhibits ripple either in the stop
band or the pass band.
2. The poles of this filter lies on the ellipse.
Give the Butterworth filter transfer function and its magnitude characteristics
for Different orders of filter.

How can you design a digital filter from analog filter?


Digital filter can de designed from analog filter using the following methods
1. Approximation of derivatives
2. Impulse invariant method (IIM)
3. Bilinear transformation (BLT)
List the Butterworth polynomial for various orders.
N Denominator polynomial
1).S+1
2).S2+.707s+1
3). (s+1)(s2+s+1)
4). (s2+.7653s+1)(s2+1.84s+1)
5). (s+1)(s2+.6183s+1)(s2+1.618s+1)
6). (s2+1.93s+1)(s2+.707s+1)(s2+.5s+1)
7). (s+1)(s2+1.809s+1)(s2+1.24s+1)(s2+.48s+1)
Differentiate Butterworth and Chebyshev filter.
Butterworth damping factor 1.44 and chebyshev is 1.06.Butterworth is flat response
.but chebyshev is damped response.

What is the condition for linear phase FIR filter?


Linear phase is a property of a filter, where the phase response of the filter is a linear
function of frequency. The result is that all frequency components of the input signal
are shifted in time (usually delayed) by the same constant amount, which is referred
to as the phase delay. And consequently, there is no phase distortion due to the time
delay of frequencies relative to one another.
A filter with linear phase may be achieved by an FIR filter which is either symmetric
or antisymmetric.

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