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DIGITAL SIGNAL PROCESSING
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IT 1252 /CS2403/CS 73
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E= [Q 2
n=-
If E is finite i.e. 0<E<WKHQ[QLVFDOOHGHQHUJ\VLJQDO
If P is finite in the expression P=Lt (1/2N+1) EN, the signal is called a power signal.
N->
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i.e. \Q KN
k= -
17.what are the properties of convolution sum
The properties of convolution sum are
Commutative property.
Associative law.
Distributive law.
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21.Define Z-transform
Z- transform can be defined as
X(Z)=[Q]-n
n=-
22.Define Region of convergence
The region of convergence (ROC) of X(Z) the set of all values of Z for which X(Z)
attain final value.
23.State properties of ROC.
The ROC does not contain any poles.
When x(n) is of finite duration then ROC is entire Z-plane except Z=0 or Z=
If X(Z) is causal,then ROC includes Z=
If X(Z) is anticasual,then ROC includes Z=0.
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ii)Time shifting
Z
if x(n) ;=
then
Z
x(n-k) =-KX(Z)
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iii)Scaling in Z-domain
Z
if x(n) ;=
Z
then anx(n) ;D-1Z)
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v)Differtiation in Z domain
Z
nx(n) -Zdz X(Z)
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iv)Time reversal
Z
if x(n) ;=
Z
then x(-n) ;=-1)
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vii)correlation
Z
Z
if x1(n) ;=DQG[Q ;=
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then
=
rx1x2(l=[Q[QO 5x1x2(Z)=X1(Z) .X2(Z-1)
n=-
26.Define DFT and IDFT (or) What are the analysis and synthesis equations of DFT?
DFT(Analysis Equation)
N-1
nk
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X(k)= [Q:N ,
n=0
WN = e-j2
N
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k=0
WN = e-j2
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N-1
IDFT(Synthesis Equation)
N-1
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m=0
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29.How to obtain the output sequence of linear convolution through circular convolution?
Consider two finite duration sequences x(n) and h(n) of duration L samples and M
samples. The linear convolution of these two sequences produces an output sequence of
duration L+M-1 samples, whereas, the circular convolution of x(n) and h(n) give N
samples where N=max(L,M).In order to obtain the number of samples in circular
convolution equal to L+M-1, both x(n) and h(n) must be appended with appropriate
number of zero valued samples. In other words by increasing the length of the sequences
x (n) and h(n) to L+M-1 points and then circularly convolving the resulting sequences we
obtain the same result as that of linear convolution.
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32.What are the two methods used for the sectional convolution?
The two methods used for the sectional convolution are
1)the overlap-add method and 2)overlap-save method.
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36.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
computation of DFT of a sequence of length N into successively smaller DFTs.
37.How many multiplications and additions are required to compute N point DFT using
redix-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,.
38.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.
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Circular convolution
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No.
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No
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Overlap-add method
47.What are the differences and similarities between DIF and DIT algorithms?
Differences:
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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.
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.
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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 computation.
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48. What are the different types of filters based on impulse response?
Based on impulse response the filters are of two types
1. IIR filter
2. FIR filter
The IIR filters are of recursive type, whereby the present output sample
depends on the present input, past input samples and output samples.
The FIR filters are of non recursive type, whereby the present output
sample depends on the present input sample and previous input samples.
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49. What are the different types of filters based on frequency response?
Based on frequency response the filters can be classified as
1. Lowpass filter
2. Highpass filter
3. Bandpass filter
4. Bandreject filter
50. What are the advantages and disadvantages of FIR filters?
Advantages:
1. FIR filters have exact linear phase.
2. FIR filters are always stable.
3. FIR filters can be realized in both recursive and non recursive
structure.
4. Filters with any arbitrary magnitude response can be tackled using
FIR sequence.
Disadvantages:
1. For the same filter specifications the order of FIR filter design can
be as high as 5 to 10 times that in an IIR design.
2. Large storage requirement is requirement
3. Powerful computational facilities required for the implementation.
4. 51.Distinguish between FIR filters and IIR filters.
FIR filter
IIR filter
1. These filters can be easily designed to
These filters do not have linear phase.
have perfectly linear phase.
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IIR filters are easily realized recursively.
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54. List the steps involved in the design of FIR filters using windows.
1.For the desired frequency response Hd(w), find the impulse response
hd(n) using Equation
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hd(n)=1/2
Hd(w)ejwndw
2.Multiply the infinite impulse response with a chosen window sequence
w(n) of length N to obtain filter coefficients h(n),i.e.,
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57. What is the necessary and sufficient condition for linear phase characteristic in FIR
filter?
The necessary and sufficient condition for linear phase characteristic in
FIR filter is, the impulse response h(n) of the system should have the symmetry
property i.e.,
H(n) = h(N-1-n)
where N is the duration of the sequence.
58.What are the advantages of Kaiser window?
o It provides flexibility for the designer to select the side lobe level
and N
o It has the attractive property that the side lobe level can be varied
continuously from the low value in the Blackman window to the
high value in the rectangular window
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59. What is the principle of designing FIR filter using frequency sampling method?
In frequency sampling method the desired magnitude response is
sampled and a linear phase response is specified .The samples of desired
frequency response are identified as DFT coefficients. The filter coefficients are
then determined as the IDFT of this set of samples.
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62. Draw the direct form realization of a linear Phase FIR system for N even.
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63.Draw the direct form realization of a linear Phase FIR system for N odd
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65. State the equations used to convert the lattice filter coefficients to direct form FIR
Filter coefficient.
m(0) = 1
m(m) = km
mN m-1N m(m) m-1(m-k)
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66. State the equations used to convert the FIR filter coefficients to the lattice filter
Coefficient.
)RUDQ0BVWDJHILOWHU m-1(0) =1 and km m(m)
m-1N
m(k)
-
1-
m(m) m(m-k)
2
m
(m)
, 1NP-1
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68. State the advantage of direct form structure over direct form structure.
In direct form structure, the number of memory locations required is less than
that of direct form structure.
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69. How one can design digital filters from analog filters?
Map the desired digital filter specifications into those for an equivalent analog
filter.
Derive the analog transfer function for the analog prototype.
Transform the transfer function of the analog prototype into an equivalent digital
filter transfer function.
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70. Mention the procedures for digitizing the transfer function of an analog filter.
The two important procedures for digitizing the transfer function of an analog
filter are
Impulse invariance method.
Bilinear transformation method.
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72. What is the mapping procedure between S-plane & Z-plane in the method of mapping
differentials? What are its characteristics?
The mapping procedure between S-plane & Z-plane in the method of mapping of
differentials is given by
H(Z) =H(S)|S=(1-Z-1)/T
The above mapping has the following characteristics
The left half of S-plane maps inside a circle of radius centered at Z= in the Zplane.
The right half of S-plane maps into the region outside the circle of radius in the
Z-plane.
The j -axis maps onto the perimeter of the circle of radius in the Z-plane.
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74. Give the bilinear transform equation between S-plane & Z-plane.
S=2/T(1-Z-1/1+Z-1)
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Disadvantage:
The mapping is highly non-linear producing frequency, compression at high
frequencies.
Neither the impulse response nor the phase response of the analog filter is
preserved in a digital filter obtained by bilinear transformation.
79. What is the advantage of cascade realization?
Quantization errors can be minimized if we realize an LTI system in cascade form.
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1.Input quantization errors2.Coefficient quantization errors3.Product quantization errors
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The filter coefficients are computed to infinite precision in theory. But in digital computation the
filter coefficients are represented in binary and are stored in registers. If a b bit register is used the
filter coefficients must be rounded or truncated to b bits ,which produces an error.
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95.What is truncation?
Truncation is a process of discarding all bits less significant than LSB that is retained
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PART B QUESTIONS
1.Describe the concept of frequency in continuous time and discrete time sinusoidal signals
Ans: Waveforms and Derivation
2.State and prove sampling theorem .
Ans: Statement, Derivation, Reconstrution
3.Explain the classification of DT signals.
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Ans: maping from S plane to Z plane.
14. Derive the equation for designing IIR filter using bilinear transformation.
Ans: maping from S plane to Z plane.
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13. Derive the equation for designing IIR filter using impulse invariant method.
Ans: Maping from S plane to Z plane.
Impulse invariant equation
d/dt y(t)=y(nT)-y(nT-T)/T
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16.Explain the steps involved in the design of FIR filter using window technique.
Ans: Specify the equation of window function
Find h(n)
Find hd(n)=h(n)w(n)
Find H(
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17. Explain the steps involved in the design of FIR filter using frequency sampling technique.
Ans: Find G(k) from given Hr(k)
Find H(k)
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18. Explain the steps involved in the design of FIR filter using Kaiser window .
Ans: Determine hd(n)
&KRRVH
&DOFXODWH
'HWHUPLQHWKHSDUDPHWHU s
Choose the parameter
Find N
Calculate yhe window function wk(n)
Find h(n)=wk(n) hd(n)
19. Draw the structures of FIR filters.
Ans: Direct form
Cascade structure
Parallel structure
Lattice structure.
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25.Explain the construction and operation of channel vocoder with block diagram.
Ans : Block diagram
Explanation.
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24.Explain the different types of limit cycle oscillations and also the solutions
Ans: Zero input limit cycle oscillations
Overflow input limit cycle oscillations
Solutipn:scaling