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X N X Ne: Iii Ece/Digital Signal Processing/Two Marks Questions & Answers 1
X N X Ne: Iii Ece/Digital Signal Processing/Two Marks Questions & Answers 1
UNIT I
1. If H(k) is the N-point DFT of a sequence h(n), Prove that H(k) and H(N-K) are complex
conjugates.
(Nov2008)
If DFT[x(n)]=X(k)
Then DFT[x*(n)]=X*(N-k)=X*((-k))N
Proof:
*
DFT [x (n ) ] =
j2 kn
*
x (n ) e
n =0
1
j2 kn
x(n )e
n =0
j2 n ( N
k)
x(n )e
=X (N
k)
n =0
DFT[x*(N-n)] = X*(k)
Proof:
1
*
IDFT [X (k ) ] =
N
=
1
N
j2 kn
*
X (k ) e
k =0
j2 k ( N
X (k ) e
n)
1
=
N
j2 kn
X (k ) e
k =0
k =0
= x*(N-n)
Therefore DFT[x*(N-n)] = X*(k)
2. What are the differences and similarities between DIF and DIT algorithms? (Nov2008)
Sl.No
DIT FFT
DIF FFT
Input sequence is to be given in bit The DFT at the output is in bit reversed
reversed order.
order.
First calculate 2-point DFTs and Decimates the sequence step by step to
combines them
2-point sequence and calculate DFT.
A=a+Wnb
B =a Wn b
b Wn
-1
n =0
1N 1
*
x(n )y (n ) = X(k )Y (k )
N k =0
*
Proof:
N
x(n )y (n ) =
n =0
1
N
x(n )
n =0
1N 1
= x(n )
N n =0
k =0
j2 kn
Y(k ) e
k =0
j2 kn
*
Y (k ) e
1N 1 *
=
Y (k )
N k =0
j2 kn
x(n ) e
n =0
1N 1
*
X(k )Y (k )
N k =0
Hence proved
6. What do you mean by the term bit reversal as applied to FFT
In DIT algorithm we can find that for the output sequence to be in a natural order (i.e., X(k) , k=0,1,2,.N-1)
the input sequence has to be stored in a shuffled order. For an 8-point DIT algorithm the input sequence is in the order
x(0), x(4), x(2),x(6),x(1),x(5),x(3) and x(7). We can see that when N is a power of 2 , the input sequence must be
stored in bit-reversal order for the output to be computed in a natural order. For N = 8 the bit-reversal process is
shown in table.
Binary
representation
000
001
010
011
100
101
110
111
Bit reversed
binary
000
100
010
110
001
101
011
111
7. What are the advantages of FFT algorithm over direct computation of DFT?
(May 2007)
The complex multiplication in the FFT algorithm is reduced by (N/2) log2N times.
Processing speed is very high compared to the direct computation of DFT.
8. The first five DFT coefficients of a sequence x(n) are x(0) = 20, x(1) = 5+j2, x(2) = 0, x(3)=0.2+j0.4, X(4) = 0.
Determine the remaining DFT coefficients.
(May 2007)
By complex conjugate property
x(5)=0.2-j0.4,x(6)=0,x(7)=5-j2
9. Define symmetric and Anti symmetric signals. How do you prevent aliasing while sampling a CT signal?
(May 2007)
A real valued signal x(n) is called symmetric if
X (n) = X (-n)
On the other hand, a signal x(n) is called antisymmetric
X (-n) = -X (n)
10. what is the necessary and sufficient condition on the impulse response for stability? (May 2007)
The necessary and sufficient condition for the impulse response is given by
+
|h (n)|<
n=-
11. Define Complex Conjugate of DFT property.
(May 2007)
DFT
If
x(n)X(k) then
N
X*(n)(X*(-k))N = X*(N-K)
12. What is FFT?
(Nov 2006)
The fast Fourier transform is an algorithm is used to calculate the DFT. It is based on fundamental principal of
decomposing the computation of DFT of a sequence of the length N in to successively smaller discrete Fourier
Transforms. The FFT algorithm provides speed increase factor when compared with direct computation of the DFT.
(Nov 2006)
Sampling is the process to convert analog time domain continuous signal into discrete time domain
signal. But it is the process of converting only time domain not in amplitude domain.
Nyquist criteria:
We sample the signal based on the following condition i.e.,
fs 2fm
||h||1 = |h(n)|
n=-
20. Define DFT pair?
x(n)e-j2kn/N
x(n) =
X(k)ej2kn/N
(Nov 2003)
If we operate the sampler at fx < fm, the frequency components of the frequency spectrum will overlap with
each other i.e., the lower frequency of the second frequency component will overlap with higher frequency of the first
frequency component. This overlapping effect is called as Aliasing effect. For avoiding overlapping of high and low
frequency components, we have to use low-pass filter to cut the unwanted high frequency components.
22. Give any two properties of DFT
a)Periodicity x(k+n)=x(k)
b)Linearity DFT{a1x1(n)+a2x2(n)}=a1x1(k)+a2x2(k).
23. Explain Linearity property of DFT
DFT{x (n)}=x (k)&DFT{y (n)}=Y (k)
For any real valued constant a &b .
DFT{a x (n)+b y (n)=a X (k)+a Y (k)
1
2
3
1
1
1
2
3
3
1
1
2
2
3
1
1
4
3
2
1
4+3+6+2
8+3+2+3
12 + 6 + 2 + 1
4+9+4+1
15
16
21
18
X3(n) = {15,16,21,18}
30. Find the linear convolution of {1,0,1} and {2,0,2}. (N/D 2007)
Y(n) = {2,0,4,0,2}
PREPARED BY : M.MANIKANDAN.,M.E.,(Ph.d),ECE DEPT./AMSEC
Circular convolution
Overlap-save method
In this method the size of the input data block
is N=L+M-1
Each data block consists of the last M-1 data
points of the previous data block followed by
L new data points.
In each output block M-1 points are corrupted
due to aliasing, as circular convolution is
employed.
To form the output sequence the first M-1
data points are discarded in each output block
and the remaining data are fitted together.
Overlap-add method
In this method the size of the input
data block is L.
Each data block is L points and we
append M-1 zeros to compute N-point
DFT.
In this no corruption due to aliasing, as
linear convolution is performed using
circular convolution.
To form the output sequence, the last
M-1 points from each output block is
added to the first (m-1) points of the
succeeding block.
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.
39. 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(n) can be obtained by circular convolution defined as
The direct evaluation DFT requires N2 complex multiplications and N2 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. So we use FFT.
41. Why
Once the butterfly operation is performed on a pair of complex numbers (a,b) to produce (A,B), there
is no need to save the input pair. We can store the result (A,B) in the same locations as (a,b). Since the same
storage locations are used troughout the computation we say that the computations are done in place.
42. 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 computation.
PREPARED BY : M.MANIKANDAN.,M.E.,(Ph.d),ECE DEPT./AMSEC
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.
44. What
is overlap-save method?
In this method the data sequence is divided into N point sections xi(n).Each section contains the last
M-1 data points of the previous section followed by L new data points to form a data sequence of length
N=L+M-1.In circular convolution of xi(n) with h(n) the first M-1 points will not agree with the linear
convolution of xi(n) and h(n) because of aliasing, the remaining points will agree with linear convolution.
Hence we discard the first (M-1) points of filtered section xi(n) N h(n). This process is repeated for all
sections and the filtered sections are abutted together.
45. State
46. Define
DFT and IDFT (or) What are the analysis and synthesis equations of DFT?
DFT(Analysis Equation)
N-1
nk
IDFT(Synthesis Equation)
X(k) =
xne
2 nk
j
N
n 0
10
1. Show that the filter with h (n) = [-1, 0, 1] is a linear phase filter. (Nov 2008, May 2007)
H (e
)=
n=
j
=e
h(n )e
[ e
jn
+e
= 1+e
j
] =e
j2
( 2jsin ) = 2je
sin
From the above equation we can find () = - which is the proportional to . Hence the filter h(n) is a linear
Phase filter.
2. What are the merits and demerits of FIR filters? (Nov 2005 & April 2008)
FIR filters that have ideal linear phase characteristics can be easily designed.
FIR filters realized non-recursively are always stable.
Errors arising from quantization of signals and finite word length effects are usually less critical for FIR filter
designs as these realization do not have feedback FIR filters are implemented through FFT algorithms, which greatly
reduced its processing time.
3. In the design of FIR digital filters, how is Kaiser window different from other windows? (Nov 2007)
It provides flexibility for the designer to select the side lobe level and N. 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.
4. State the condition for a digital filter to be causal and stable.
(May 2007)
The response of the causal system to an input does not depend on future values of that input, but depends only on
the present and/or past values of the input. A filter is said to be stable, bounded-input bounded output stable, if every
bounded input produces a bounded output. A bounded signal has amplitude that remains finite.
5. What is the condition satisfied by linear phase FIR filter? (Nov/Dec 2003 & May 2007)
Linear phase is of the form
() = k
Here k is constant. Thus phase shift is linearly proportional to frequency. For linear phase, the impulse response
should satisfy following condition.
h (n) = h (M-1-n)
6. Give any two properties of Butterworth filter and chebyshev filter. (Nov/Dec 2006, May/June 2006, Apr 2005 &
Nov 2004)
a. The magnitude response of the Butterworth filter decreases monotonically as the frequency increases ()
from 0 to .
b. The magnitude response of the Butterworth filter closely approximates the ideal response as the order N
increases.
c. The poles on the Butterworth filter lies on the circle.
d. The magnitude response of the chebyshev type-I filter exhibits ripple in the pass band.
e. The poles of the Chebyshev type-I filter lies on an ellipse.
7. What are the desirable and undesirable features of FIR Filters? (May2006)
The width of the main lobe should be small and it should contain as much of total energy as possible.
The side lobes should decrease in energy rapidly as w tends to
11
Otherwise
Otherwise
The width of the main lobe is approximately 8/M and the peak of the first side lobe is at -32dB.
The window function of a causal Blackman window is expressed by
WB(n) = 0.42 0.5 cos2n/ (M-1) +0.08 cos4n/(M-1), 0nM-1
= 0,
otherwise
otherwise
The width of the main lobe is approximately 12/M and the peak of the first side lobe is at -58dB.
9. Write the magnitude function of Butterworth filter. What is the effect of varying order of N on magnitude and
phase response?
(Nov 2005)
|H(j)|2 = 1 / [ 1 + (/C)2N] where N= 1,2,3,.
10. Mention the necessary and sufficient condition for linear phase characteristics in FIR filter. (Nov 2005)
The necessary and sufficient conditions is that the phase function should be linear function w, which in turn
requires constant phase delay (or) constant phase and group delay i.e., Q(w) w
Q(w) = - w
-w
11. What is linear phase? What is the condition to be satisfied by the impulse response in order to have a linear
phase?
(Apr 2005 & Nov 2003)
For a filter to have linear phase the phase response (w) w is the angular frequency.
The linear phase filter does not alter the shape of the signal. The necessary and sufficient condition for a filter to
have linear phase.
h(n) = h(N-1-n); 0 n N-1
12. List the characteristics of FIR filters designed using window functions. (Nov 2004)
the Fourier transform of the window function W(ejw) should have a small width of main lobe
containing as much of the total energy as possible
the fourier transform of the window function W(ejw) should have side lobes that decrease in
energy rapidly as w to . Some of the most frequently used window functions are described in
the following sections.
PREPARED BY : M.MANIKANDAN.,M.E.,(Ph.d),ECE DEPT./AMSEC
(Apr 2004)
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.
17. What are the design techniques of designing FIR filters?
There are three well known methods for designing FIR filters with linear phase .They are
(1.)Window method
(2.)Frequency sampling method
(3.)Optimal or minimax design.
18. What is Gibbs phenomenon?
One possible way of finding an FIR filter that approximates H(ejw) would be to truncate the infinite
Fourier series at n=(N-1/2).Direct truncation of the series will lead to fixed percentage overshoots and
undershoots before and after an approximated discontinuity in the frequency response.
PREPARED BY : M.MANIKANDAN.,M.E.,(Ph.d),ECE DEPT./AMSEC
12
20.
List the steps involved in the design of FIR filters using windows.
21. What
22.
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.
13
25. When
The cascade form realization is preferred when complex zeros with absolute magnitude is less than one.
26. State the equations used to convert the lattice filter coefficients to direct form FIR Filter
coefficient.
27. Draw
the direct form realization of a linear Phase FIR system for N even.
14
30. State the equations used to convert the FIR filter coefficients to the lattice filter Coefficient.
15
(May 2007)
Sl.No
IIR
FIR
More
16
Less complicated
10
Design methods:
Design methods:
17
1. Bilinear Transform
1. Windowing
2. Impulse invariance.
2. Frequency sampling
Can be used where sharp cutoff characteristics
Used where linear phase characteristic is
with minimum order are required
essential.
11
UNIT III
1. What is prewarping?
(Nov 2003,2008)
When bilinear transformation is applied, the discrete time frequency is related continuous time frequency as,
= 2tan-1T/2
This equation shows that frequency relationship is highly nonlinear. It is also called frequency warping. This effect can
be nullified by applying prewarping. The specifications of equivalent analog filter are obtained by following relationship,
= 2/T tan /2
This is called prewarping relationship.
2. What is the relation betweeen analog and digital frequency in impulse invariant transformation?(April 2008)
T=
3. State the condition for a digital filter to be causal and stable.
(May 2007)
The response of the causal system to an input does not depend on future values of that input, but depends only on the
present and/or past values of the input.
A filter is said to be stable, bounded-input bounded output stable, if every bounded input produces a bounded output.
A bounded signal has amplitude that remains finite.
4. Find the digital transfer function H (z) by using impulse invariant method for the analog transfer function H(s) =
1/(S+2). Assume T=0.5sec.
H(s) = 1/(s+2)
The system function of the digital filter is obtained by
H (z) = 1/ (1-e-2Tz-1)
Since T=o.5 sec
H (z) = 1/ (1-.067Z-1)
5. Mention any two procedures for digitizing the transfer function of an analog filter.
1.
2.
(Nov 2006)
(or)
18
Give the equation for the order N, major, minor and axis of an ellipse in case of chebyshev filter. (Nov 2005)
N cosh-1 (/) / cosh-1(S/ P)
Where = (100.1s 1)
= (100.1p 1)
7. What are the advantages and disadvantages of bilinear transformation?
(May 2006)
Advantages:
The bilinear transformation provides one-to-one mapping.
Stable continuous systems can be mapped into realizable, stable digital systems.
There is no aliasing.
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.
8. What is impulse invariant mapping? What is its limitation?
(Apr/May 2005)
The philosophy of this technique is to transform an analog prototype filter into an IIR discrete time filter whose
impulse response [h(n)] is a sampled version of the analog filters impulse response, multiplied by T.
This procedure involves choosing the response of the digital filter as an equi-spaced sampled version of the analog
filter.
9. What is frequency warping?
The bilinear transform is a method of compressing the infinite, straight analog frequency axis to a finite one long
enough to wrap around the unit circle only once. This is also sometimes called frequency warping. This introduces a
distortion in the frequency. This is undone by pre-warping the critical frequencies of the analog filter (cutoff frequency,
center frequency) such that when the analog filter is transformed into the digital filter, the designed digital filter will meet
the desired specifications.
10. What are the limitations of impulse invariant mapping technique? (Apr2004)
The impulse invariance technique is appropriate only for band limited filter like low pass filter. Impulse invariance
design for high pass or band stop continuous-time filters, require additional band limiting to avoid severe aliasing
distortion, if impulse designed is used. Thus this method is not preferred in the design of IIR filters other than low-pass
filters.
11. Give the transform relation for converting low pass to band pass in digital domain.
(Apr 2004)
Low pass with cut off frequency C to band pass with lower cut-off frequency 1 and higher cut-off
frequency 2:
S ------------- C ( s2 + 1 2) / s (2 - 1)
19
(Nov2003)
The bilinear transformation method overcomes the effect of aliasing that is caused due to the analog frequency
response containing components at or beyond the nyquist frequency. The bilinear transform is a method of compressing
the infinite, straight analog frequency axis to a finite one long enough to wrap around the unit circle only once.
S = (2/T) (Z-1) (Z+1)
13. State the structure of IIR filter?
IIR filters are of recursive type whereby the present o/p sample depends on present i/p, past i/p
samples and o/p samples. The design of IIR filter is realizable and stable. The impulse response h(n) for
a realizable filter is h(n)=0 for n 0
14. State the advantage of direct form II structure over direct form I structure.
In direct form II structure, the number of memory locations required is less than that of direct form I
structure.
15. 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.
16. What do you understand by backward difference?
One of the simplest method for converting an analog filter into a digital filter is to approximate the
differential equation by an equivalent difference equation.
d/dt y(t)=y(nT)-y(nT-T)/T
The above equation is called backward difference equation.
17. 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.
18. What is meant by impulse invariant method of designing IIR filter?
In this method of digitizing an analog filter, the impulse response of resulting digital filter is a sampled
version of the impulse response of the analog filter. The transfer function of analog filter in partial
fraction form.
PREPARED BY : M.MANIKANDAN.,M.E.,(Ph.d),ECE DEPT./AMSEC
20. What
The mapping for the bilinear transformation is a one-to-one mapping that is for every point Z, there is
exactly one corresponding point S, and vice-versa.
The j-axis maps on to the unit circle |z|=1,the left half of the s-plane maps to the interior of the unit
circle |z|=1 and the half of the s-plane maps on to the exterior of the unit circle |z|=1.
21. Define an IIR filter.
The filter designed by considering all the infinite samples of impulse response are called IIR filters.
The impulse response is obtained by taking inverse fourier transform of ideal frequency response.
22. Distinguish between IIR and FIR filters.
The filter design starts from ideal frequency response. By taking inverse fourier transform of ideal
frequency response,the desired impulse response is obtained, which consists of infinite number of samples.
The digital filters designed by selecting only N samples of the impulse response are called FIR filters.
The digital filters designed by selecting all the infinite samples of impulse response are called IIR filters.
23. What are the requirements for an analog filter to be stable and causal?
(i) The analog filter transfer function Ha(s) should be a rational function os s and the coefficients of
s should be real.
(ii) The poles should lie on the left half of s-plane.
(iii) The number of zeros should be less than or equal to number of poles.
24. Write a brief note on the design of IIR filter. (OR) How a digital IIR filter is designed?
For designing a digital IIR filter, first an equivalent analog filter is designed using any one of the
approximation technique and given specifications. The result of the analog filter design will be an analog
filter transfer function Ha(s). The analog filter transfer function is transformed to digital filter transfer
function H(z) using either Bilinear or Impulse invariant transformation.
25. Mention the important features of IIR filters.
(i) The physically realizable IIR filters does not have linear phase.
(ii) The IIR filter specifications includes the desired characteristics for the magnitude response only.
26. What are the advantages and disadvantages of digital filters?
Advantages of digital filters:
(i) High thermal stability due to absence of resistors, inductors and capacitors.
(ii) The performance characteristics like accuracy, dynamic range, stability and tolerance can be
enhanced by increasing the length of the registers.
(iii) The digital filters are programmable.
(iv) Multiplexing and adaptive filtering are possible.
PREPARED BY : M.MANIKANDAN.,M.E.,(Ph.d),ECE DEPT./AMSEC
20
Bilinear transformation
(i) It is one-to-one mapping.
(ii) The relation between analog and digital
frequency is nonlinear.
(iii) There is no problem of aliasing and so the
analog filter need not be bandlimited.
(iv) Due to the effect of warping, the phase
response of analog filter cannot be preserved.
But the magnitude response can be preserved
by prewarping.
21
Chebyshev Type-I
(i) All pole design.
(ii) The poles lie on an ellipse in s-plane.
(iii) The magnitude response is equiripple in
passband and monotonically decreasing in
the stopband.
(iv) The normalized magnitude response has a
value of 1/(1+2) at the cutoff frequency c.
22
1. Express the fraction (-9/32) in sign magnitude, 2s complement notations using 6 bits.
(Nov 2008)
4. What is meant by limit cycle oscillations? ((May 2006,Apr 2005 May 2007, Nov 2007 & Apr 2008)
In fixed point addition, overflow occurs due to excess of results bit, which are stored at the registers. Due
to this overflow, oscillation will occur in the system. Thus oscillation is called as an overflow limit cycle
oscillation.
5. Express the fraction(-7/32) in signed magnitude and twos complement notations using 6 bits. (Nov
2007)
Sign magnitude : 1.00111
2s complement : 1.11001
6. Express the fraction 7/8 and -7/8 in sign magnitude, 2s complement and 1s complement.
7/8
-7/8
1.111
1s complement : 0.000
1.000
2s complement : 0.001
1.001
(Nov 2006)
(May 2007)
Sampling rate conversion is the process of converting a signal from one sampling rate to another,
while changing the information carried by the signal as little as possible.
Sample rate conversion needed because different systems use different sampling rates.
8. Convert the number 0.21 into equivalent 6-bit fixed point number. (May 2007)
0.001101
PREPARED BY : M.MANIKANDAN.,M.E.,(Ph.d),ECE DEPT./AMSEC
23
(Nov 2006)
amplitude quantization
vector quantization
scalar quantization
11. What is zero padding? Does zero padding improve the frequency resolution in the spectral estimate?
(Nov 2006)
The process of lengthening a sequence by adding zerovalued samples is called appending with
zeros or zero padding.
12. List the advantages of floating point arithmetic.
(Nov 2006)
13. Give the expression for signal to quantization noise ratio and calculate the improvement with an
increase of 2 bits to the existing bit.(Nov2006,Nov2005)
SNRA / D = 16.81+6.02b-20log10 (RFS /x) dB.
With b= 2 bits increase, the signal to noise ratio will increase by 6.02 X 2 = 12dB.
14. Draw the probability density function for rounding.
(Nov 2005)
(May/Jun 2006)
Hardware implementation is
costlier and difficult to design
24
Overflow
is
phenomenon
rare
Overflow
is
phenomenon
range
(Nov 2004)
In a limit cycle the amplitude of the output are confined to a range of value, which is called dead band.
17. How can overflow limit cycles be eliminated?
(Nov 2004)
Saturation Arithmetic
Scaling
(Apr 2004)
2
2
P = e .1/2 |H()| dw
-
2
Here e is the variance of input error signal.
e2 = 2-2LRFS2 /48
2
2
-2L
v = 2 RFS /48 X |H ()|2 dw
-
This equation gives steady state noise power due to quantization.
20. What is meant by finite word length effects in digital filters? (Nov 2003)
The digital implementation of the filter has finite accuracy. When numbers are represented in digital
form, errors are introduced due to their finite accuracy. These errors generate finite precision effects or finite
word length effects.
When multiplication or addition is performed in digital filter, the result is to be represented by finite
word length (bits). Therefore the result is quantized so that it can be represented by finite word register. This
quantization error can create noise or oscillations in the output. These effects are called finite word length
effects.
21. What is round-off noise error?
Rounding operation is performed only on magnitude of the number. Hence round-off noise error is
independent of type of fixed point representation. If the number is represented by bu bits before quantization
and b bits after quantization, then maximum round-off error will be (2_b-2-bu)/2. It is symmetric about zero.
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There are three types of arithmetic used in digital systems. They are fixed point arithmetic, floating
point ,block floating point arithmetic.
24. What are the different types of fixed point arithmetic?
Depending on the negative numbers are represented there are three forms of fixed point arithmetic.
They are sign magnitude,1s complement,2s complement
25. What is meant by sign magnitude representation?
For sign magnitude representation the leading binary digit is used to represent the sign. If it is equal
to 1 the number is negative, otherwise it is positive.
26. What is meant by 1s complement form?
In 1,s complement form the positive number is represented as in the sign magnitude form. To obtain
the negative of the positive number ,complement all the bits of the positive number.
27. What is meant by 2s complement form?
In 2s complement form the positive number is represented as in the sign magnitude form. To obtain
the negative of the positive number ,complement all the bits of the positive number and add 1 to the LSB.
28. What is meant by floating pint representation?
In floating point form the positive number is represented as F =2CM,where is mantissa, is a fraction
such that1/2<M<1and C the exponent can be either positive or negative.
29. What are the advantages of floating pint representation?
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The addition of two fixed point arithmetic numbers cause overflow when the sum exceeds the word size available to
store the sum. This overflow caused by adder make the filter output to oscillate between maximum amplitude limits. Such
limit cycles have been referred to as overflow oscillations.
34. What are the two kinds of limit cycle behavior in DSP?
(i) Zero limit cycle oscillations
(ii) Overflow limit cycle oscillations
35. What is meant by quantization step size?
Let us assume a sinusoidal signal varying between +1 and -1 having a dynamic range 2. If ADC used to convert the
sinusoidal signal employs b+1 bits including sign bit, the number levels available for quantizing x(n) is 2b+1. Thus the
interval between successive levels. q= 2/(2b+1) = 2-b . where q is known as quantization step size.
36.Explain briefly the need for scaling in the digital filter implementation.
To prevent overflow, the signal level at certain points in the digital filters must be scaled so that no overflow occurs in
the adder.
37. Why the limit cycle problem does not exist when FIR digital filter is realized in direct form or cascade form?
In the case of FIR filters, there are no limit cycle oscillations, if the filter is realized in direct form or cascade form
since these structures have no feedback.
38. Why rounding is preferred to truncation in realizing digital filter?
(i) The quantization error due to rounding is independent of the type arithmetic.
(ii) The mean of rounding error is zero.
(iii) The variance of the rounding error signal is low.
39. What is truncation?
Truncation is a process of discarding all bits less significant than LSB that is retained
40. What is Rounding?
Rounding a number to b bits is accomplished by choosing a rounded result as the b bit number closest number
being unrounded.
41. State some applications of DSP?
Speech processing ,Image processing, Radar signal processing.
42. what is meant by A/D conversion noise?
A DSP contains a device, A/D converter that operates on the analog input x(t) to produce xq(t) which
is binary sequence of 0s and 1s. At first the signal x(t) is sampled at regular intervals to produce a
sequence x(n) is of infinite precision. Each sample x(n) is expressed in terms of a finite number of bits
given the sequence xq(n). The difference signal e(n)=xq(n)-x(n) is called A/D conversion noise.
PREPARED BY : M.MANIKANDAN.,M.E.,(Ph.d),ECE DEPT./AMSEC
Harvard Architecture
Separate
memories
for
program and data.
The speed of execution in
Harvard architecture is high
In this architecture having a
common interval address and
data bus.
It is not suitable for DSP
processors.
(May 2007)
Von-Neumann Architecture
It shares same memory for program
and data.
The speed of execution is increased
by pipelining
It is having a separate interval address
and data bus.
It is normally used for Harvard
architecture.
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DSP processors should have circular buffers to support circular shift operations.
The DSP processor should be able to perform multiply and accumulate operations very fast.
DSP processors should have multiple pointers to support multiple operands jumps and shifts.
7. What is the advantage of Harvard architecture of TMS 320 series? (Nov 2006)
DSP processors should have multiple registers so that data exchange from register to register is fast.
DSP operations require multiple operands simultaneously. Hence DSP processor should have multiple
operand fetch capacity.
DSP processors should have circular buffers to support circular shift operations.
The DSP processor should be able to perform multiply and accumulate operations very fast.
DSP processors should have multiple pointers to support multiple operands jumps and shifts.
Multi processing ability.
29
The source statement can contain following four ordered fields. i.e.,
[Label][:] mnemonic [operand list] [; comment]
The source statement follows following guidelines
All the statements must begin with a label, a blank, an asterisk or a semicolon.
Labels may be placed before the instruction mnemonic on the same line or on the proceeding line
in the first column.
Each field must be separated with blanks.
If comment begins in column 1 it must have semicolon or asterisk at its beginning. In other
columns, comments can begin with semicolon.
PREPARED BY : M.MANIKANDAN.,M.E.,(Ph.d),ECE DEPT./AMSEC
30
Cost saving
Smaller size
Low power consumption
Processing of many high frequency signals in real-time
31
Increased performance
Better compiler targets
Potentially scalable
Disadvantages of VLIW architectures
o Increased memory use
o High power consumption
o Misleading MIPS ratings
Auto increment
Auto decrement
Post indexing by adding the contents of AR0
Post indexing by subtracting the contents of AR0
Single indirect addressing with no increment
Single indirect addressing with no decrement
Bit reversed addressing
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33
One-operand instructions
Two-operand instructions