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CLASS: V SEM IT
UNIT – I
A signal is defined as any physical quantity that varies with time, space, or any
other independent variable.
1
3. Define discrete system.
Continuous time signal: It is also referred as analog signal i.e., the signal is
represented continuously in time.
Discrete time signal : Signals are represented as sequence at discrete time intervals .
2
9. Define (a) periodic signal (b) non – periodic signal.
Periodic signal: A periodic signal is defined as the signal x(n) is said to be periodic
with period N if and only if x( n + N )=x(n) for all n.
Symmetric signal: A real valued signal x(n) is called symmetric if x(−n) = x(n).
Energy signal:
2
The energy of a discrete time signal x(n) is defined as E = ∑n =−∞ x( n)
∞
A signal x(n) is called an energy signal if and only if the energy obeys the relation
0 < E < ∞ . For an energy signal P = 0.
Power signal:
The average power of a discrete time signal x(n) is defined as
N 2
1
P = lim
N →∞ 2 N +1
∑ x(n) .
n =−N
A signal x(n) is called power signal if and only if the average power P satisfies the
condition 0 < P < ∞ .
13. What are the different types of operations performed on discrete – time
signals?
3
(3) Folding or Reflection of a signal
(4) Time scaling
(5) Amplitude scaling
(6) Addition of signals
(7) Multiplication of signals.
Given
x(n) = { 1, 3, −1, − 4 }
↑
We can write
x( n) = ∑−1 x (k )δ (n − k )
2
A discrete –time system is called static or “memory less” if its output at any
instants ‘n’ depends on the input samples at the same time, but not an past or future
samples of the input.
4
A system is said to be causal if the output of the system at any time n depends
only on present and past inputs, but does not depend on future inputs.
The unit sample response is defined as the output signal designated as h(n),
obtained from a discrete – time system when the input signal is a unit sample
sequence (unit impulse).
The output y(n) of an LTI system for an input signal x(n) can be obtained by
convolution of the impulse response h(n) and the input signal x(n).
y ( n ) = x ( n) ∗ h( n )
= ∑k =∞ x ( k )h(n − k )
∞
The necessary and sufficient condition for causality of an LTI system is, its unit
sample response h(n) = 0 for negative values of n i.e.,
22. What is the necessary and sufficient condition on the impulse response for
stability?
5
The necessary and sufficient condition guaranteeing the stability of a linear time-
invariant system is that its impulse response is absolutely summable.
∞
∑ h(k ) < ∞ .
−∞
(or )
∞
y ( n) = ∑h(k ) x(n − k )
k =−
∞
Sampling is the conversion of a continuous – time signal (or analog signal) into a
discrete – time signal obtained by taking samples of the continuous time signal (or
analog signal) at discrete time instants.
6
A sampling theorem states that in the band limited continuous time signal with
highest frequency (band width) fm in hertz , can be uniquely recovered from its
samples provided that the sampling rate fs is greater than or equal to 2fm samples per
second.
The Nyquist rate is defined as the frequency 2fm , which, under sampling theorem,
must be exceeded by the sampling frequency.
To avoid aliasing the sampling frequency must be greater than twice the
highest frequency present in the signal.
If the sampling frequency is exactly equal to the Nyquist rate then the
sampling frequency is known as critical sampling.
7
Quantization error is the difference between the quantized value and actual
sample value.
eq(n) = xq(n) - x(n)
Quantization level is the value that allows in a digital signal is called the
quantization level.
The quality of the output of the A/D converter is usually measured by the
Signal to Quantization Noise Ratio (SQNR) , which is ratio of the signal power to
noise power.
Pav
SQNR =
Pq
38. What is the use of a Sample and Hold circuit?
The sample and hold circuit is used to hold the sample the analog signal
and hold the sampled value constant as long as the A/D converter takes time for
accurate conversion.
Conversion time may be defined as the time taken by an ADC for converting a
given amplitude, expressed in decimal value, of a quantized analog signal applied
across its input terminals into corresponding binary – equivalent value.
8
Percentage resolution is defined as
1
%resolution = ×100
2n
where
n – number of bits
42. What are the advantages and disadvantages of counter – ramp type ADCs?
n
Conversion time =
f
where
n – number of bits and
9
f – clock frequency
10
The Region Of Convergence (ROC) of X(z) is the set of all values of z for
which X(z) attains a finite value.
1. Linearity: Z [ a1 x1 ( n) + a 2 x 2 ( n)] = a1 X 1 ( Z ) + a 2 X 2 ( Z )
m m −i
m −1
2. Shifting: (a) Z [ x( n + m)] = z X ( Z ) − ∑ x(i ) Z
i =0
(b) Z [ x(n − m)] = Z X ( Z )
−m
3. Multiplication: Z [n m x(n)] = − Z
d
X (Z )
dz
[
4. Scaling in z- domain: Z a n x ( n) = X ( a −1 Z ) ]
5. Time reversal : Z [ x (−n)] = X ( Z −1 )
[ ]
6. Conjugation: Z x ∗ (n) = X ∗ ( Z ∗ )
n
7. Convolution: Z ∑ h(n − m)r ( m) = H ( Z ) R( Z )
m =0
8. Initial value: Z [ x(0)] = Lt X ( Z ) z →∞
11
where z = re jω
Substituting z value in equation (1) we get,
X ( re jω
) = ∑x( n) r −n e − jωn …………………..(2)
{ }
x(n) = 1,1,1 ; h (n ) = 1, 2,1
↑
{ ↑
}
Solution:
Z { x(n) ∗ h(n)} = X ( z ) H ( z )
X ( z )=(1 + z −1 + z −2 )
H ( z ) = (1 + 2 z −1 + z −2 )
X ( z ) H ( z ) = (1 + 3 z −1 + 4 z −2 + 3 z −3 + z −4 )
x(n) ∗ h(n) = { 1,3, 4,3,1 }
Y ( z)
H ( z) =
X ( z)
where,
12
Y(z) is the z – transform of the output signal y(n)
X(z) is the z – transform of the input signal x(n)
UNIT – II
∞
F { x( n)} = X (ω) = ∑ x ( n) e − jωn
n =−∞
3. List the difference between Fourier transform of discrete time signal and
analog signal.
π
1
F −1 { X (ω)} = x( n) = ∫π X (ω)e
jωn
dω
2π −
13
1. The frequency response of LTI system is given by the Fourier transform of
the impulse response of the system.
The Fourier transform of the impulse response h(n) of the system is called
frequency response of the system. It is denoted by H(ω ).
iii) If h(n) is complex then the real part of H (ω ) is antisymmteric over the
interval 0 ≤ ω ≤ 2π .
y ( n) = x( n) + ay (n −1)
The frequency response of first order system depends on the co efficient “a”
in the difference equation governing the LTI system. When the value of “a|” is in
the range of 0<a<1, the first order system behave as a low pass filter. When the
value of “a” is in the range –1<a<0, the first order system behave as a high pass
filter.
14
y ( n) = 2r cos ω0 y ( n −1) − r 2 y (n − 2) + x(n) − r cos ω0 x(n −1)
The frequency response of second order system depends on the parameters
“r” and “ ω 0 ” in the difference equation the LTI system. When the value of r is in
the range of 0<r<1, the second order system behave as a resonant filter with
center frequency ω 0 . When the value of r is varied from 0 to 1 , the sharpness of
resonant peak increases.
∞ 2
X (ω) = ∑x ( n ) e
n =−∞
− jωn
= ∑e
n =−2
− jωn
= e j 2ω + e jω + 1 + e − jω + e − j 2ω
=1 + 2 cos ω + 2 cos 2ω
15
The N-point IDFT of a sequence X(k) is
1 N −1
x ( n) = ∑
N K =0
X (k )e j 2πk / N n= 0, 1 , 2 …..N-1 .
(a) Periodicity
(b) Linearity
If X1(k) = DFT[x1(n)] and
X2(k) = DFT[x2(n)]
then
DFT[a1x1(n)+a2x2(n)]=a1X1(k)+a2x2(k)
(c) Time reversal of a sequence
If DFT {x(n)} = X(k),
then
DFT{x((-n))N} = DFT{x(N-n)} = X((-k))N = X(N-k)
42. IF N-point sequence x(n) has N- point DFT X(k) then what is the DFT of
the following?
(i) x ∗ (n) (ii) x ∗ ( N − n) (iii) x((n − l )) N (iv) x(n)e j 2π ln/ N
Solution:
(i ) DFT {x ∗ ( n)} = X ∗ ( N − k )
(ii ) DFT { x ∗ ( N − n)} = X ∗ ( k )
(iii ) DFT { x(( n − l )) N } = X ( k )e − j 2πkl / N
(iv ) DFT {x (n)e j 2π ln/ N } = X (( k − l )) N
n
1
43. Calculate the DFT of the sequence x(n) = forN = 16
4
Solution:
16
N −1
X ( k ) = ∑x(n)e − j 2πkn / N K=0, 1 , 2…..N-1
n =0
n
15
1
= ∑ e − j 2πkn / 16
n =0 4
n
15
1
= ∑ e − jπk / 8
n =0 4
16
1
1 − e − j 2πk
= 4
1
1 − e − jπk / 8
4
If DFT[x(n)] = X(k),
then
DFT[x((n-m))N] = e − j 2πkn / N X (k )
Solution:
N −1
X (k ) = ∑x( n)e − j 2πkn / N ; k = 0,1,2,…..N-1.
n =0
3
= ∑ x(n)e − j 2π kn /4 k = 0,1, 2 , 3.
n =0
3
X (0) = ∑ x(n) = {1 + 1 + 0 + 0} = 2
n =0
3
X (1) = ∑ x( n)e − jπ n /2 = {1 − j + 0 + 0 } =1 − j
n =0
3
X (2) = ∑ x(n)e − jπ n = {1 − 1 + 0 + 0 } = 0
n =0
3
X (3) = ∑ x( n)e − j 3π n /2 = {1 + j + 0 + 0 } =1 + j
n =0
X (k ) = {2,1 − j , 0,1 + j }
17
20. When the DFT X(k) of a sequence x(n) is imaginary?
If the sequence x(n) is real and odd (or) imaginary and even, then X(k) is purely
imaginary.
If the sequence x(n) is real and even (or) imaginary and odd , then X(k) is purely
real.
If DFT[x(n)]=X(k),
Then
j 2π ln/ N
DFT[x(n) e ] = X (( k − l )) N
USES:
(i) We can get “better display” of the frequency spectrum.
(ii) With zero padding, the DFT can be used in linear filtering.
24. What do you understand by periodic convolution?
Let x1 p ( n) and x 2 p (n) be two periodic sequences each with period N with
[ ]
DFS x1 p (n) = X 1 p (k ) and
DFS [ x 2p ( n) ] = X 2p (k ) ……………………………(1)
If X 3 p ( k ) = X 1 p (k ) X 2 p ( k )
then the periodic sequence x3 p (n) with Fourier series coefficients X 3 p (k ) can be
obtained by periodic convolution, defined as
N −1
x3 p (n) = ∑ x1 p (m) x 2 p ( n − m) ………………………………(2)
n =0
The convolution in the form of equation (2) is known as periodic convolution, as the
sequences in equation (2) are all periodic with period N, and the summation is over
one period.
18
25. Define circular convolution.
The circular convolution involves basically four steps as the ordinary linear
convolution. These are
19
1. Folding the sequence
2. Circular time shifting the folded sequence
3. Multiplying the two sequences to obtain the product sequence.
4. Summing the values of product sequence.
1. Graphical method
2. Stockhman’s method
3. Tabular array method
4. Matrix method.
Solution:
The circular convolution of the above sequences can be obtained by using matrix
method.
31. How will you obtain linear convolution from circular convolution.
Consider two finite duration sequences x9n) and h(n0 of duration L samples and
N samples respectively. 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.
20
If the data sequence x(n) is of long duration , it is very difficult to obtain the
output sequence y(n) due to limited memory of a digital computer. Therefore, the data
sequence is divided up into smaller section. These sections are processed separately
one at a time and combined later to get the output.
33. What are the different methods used for the sectioned convolution?
The two methods used for the sectioned convolution are (i)the overlap-add
method (ii)Over lap-save method.
The term Fast Fourier Transform (FFT) usually refers to a class of algorithms for
efficiently computing the DFT.It makes use of the symmetry and periodicity
properties of twiddle factor W NK 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 discrete Fourier transforms. The FFT
algorithm provides speed increase factors, when compared with direct computation of
21
the DFT, of approximately 64 and 205 for 256 points and 1024 – point transforms
respectively.
37. How many multiplications and additions are required to compute N-point
DFT using radix-2 FFT?
38. How many multiplications and additions are required to compute N-point
DFT directly?
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 = 2 m , where m is an
integer, and then this algorithm is known as radix – 2 FFT algorithms.
The computation of 8 – point DFT using radix-2 FFT, involves three stages of
computations. Here N=8=23, therefore r=2 and m=3.
The given 8 – point sequence is decimated to 2- point sequences. For each
2 – point sequence, the 2-popint DFT is computed. From the result of 2 – point DFT
22
the 4 – point DFT can be computed. From the result of 4-point DFT, the 8 – point
DFT can be computed.
It is the popular form of the FFT algorithm. In this the output sequence X(k) is
divided into smaller and smaller subsequences, that is why the name decimation in
frequency.
43. What are the difference between and similarities between DIT and DIF
algorithms?
23
UNIT – III
1. What are the different types of structures for realization of IIR systems?
24
(a) The Direct form – I realization requires M+N+1 multiplications, M+N
additions and M + N + 1 memory locations.
The Direct form –II realization requires minimum number of delays for
the realization of the system. Hence it is called as “Canonic form” structure.
25
9. What is the advantage of cascade realization?
10. What are the different types of filters based on impulse response?
1. IIR filter
2. FIR filter
The IIR filters are of recursive type, whereby the present output sample
depends on the present input, past inputs samples and output samples.
The FIR filters are of non recursive type whereby the present output
sample is depends on the present input sample and previous input samples.
∑b z k
−k
H (z) = k =0
N
1 + ∑ ak
k =1
12. Give the magnitude of Butterworth filter. What is the effect of varying order
of N on magnitude and phase response?
1
H ( jΩ) = 1
N = 1, 2 , 3.......... ....
Ω
2N
2
1 +
Ω
c
Where N is the order of the filter and Ωc is the cut off frequency. The magnitude
response of the Butterworth filter closely approximates the ideal response as the order
N increases. The phase response becomes more non-linear as N increases.
26
1. The magnitude response of the Butterworth filter decreases monotonically as
the frequency Ω increases from 0 to α
2. The magnitude response of the Butterworth filter closely approximates the
ideal response as the order N increases
3. The Butterworth filters are all pole designs
4. The poles of the Butterworth filter lies on a circle
5. At the cut off frequency Ωc , the magnitude of normalized Butterworth filter
1
is
2
15. How the poles of Butterworth transfer function are located in s- plane?
In type –1 Chebyshev approximation, the error function is selected such that, the
magnitude response is equiripple in the pass band and monotonic in the stop band.
In type –2 Chebyshev approximation, the error function is selected such that, the
magnitude response is monotonic in pass band and equiripple in the stop band. The
Type -2 magnitude response is called inverse Chebyshev response.
27
1
H a ( Ω) =
Ω
1 + ε 2 C N2
Ωc
where
ε is attenuation constant and
Ω
CN
Ω
is the Chebyshev polynomial of the first kind of degree N
c
20. How the order of the filter affects the frequency response of Chebyshev filter.
From the magnitude response of Type -1 Chebyshev filter it can be observed that
the magnitude response approaches the ideal response as the order of the filter is
increased.
ε = 10 0.1α −1
s
28
23. Compare the Butterworth and Chebyshev Type -1 filter.
24. What are the different types of filters based on the frequency response?
These are three well-known methods for designing FIR filters with linear
phase. These are (1) Windows method (2) Frequency sampling method (3) Optimal
or minimax design.
29
27. What do you understand by linear phase response?
For a linear phase filter θ (ω) α ω . The linear phase filter did not alter the
shape of the original signal. If the phase response of the filter is non linear the output
signal may be distorted one. In many cases a linear phase characteristic is required
throughout the passband of the filter to preserve the shape of a given signal with in
the pass band. IIR filter cannot produce a linear phase. The FIR filter can give linear
phase, when the impulse response of the filter is symmetric about its mid point.
28. For what kind of application, the antisymmetrical impulse response can be
used?
The antisymmetrical impulse response can be used to design Hilbert
transformers and differentiators.
29. For what kind of application, the symmetrical impulse response can be used?
30. How can you design digital filters from the analog filters?
1. Map the desired digital filter specifications into those for an equivalent
analog filter
2. Derive the analog transfer function for the analog prototype
3. Transform the transfer function of the analog prototype into an equivalent
digital filter transfer function
31. Mention any two procedures for digitizing the transfer function of an analog
filter.
The two important procedures for digitizing the transfer function of an
analog filter are
30
32. What are the requirements for a digital filter to be stable and causal?
i. The digital transfer function H(z) should be a rational function of z and the
co-efficient of z should be real
ii. The poles should lie inside the unit circle in z-plane
iii. The number of zeros should be less than or equal to number of poles
33. What are the requirements for a analog filter to be stable and causal?
31
transformation the impulse response of the digital filter will be sampled version of the
impulse response of the analog filter.)
37. Write the impulse invariant transformation used to transform real poles with
and without multiplicity.
38. What is the relation between digital and analog frequency in impulse
invariant transformation?
Digital frequency, ω = ΩT
Where,
Ω - Analog frequency and
T - Sampling time period
32
2 1 − z −1
s=
T 1 + z1
40. What is the relation between digital and analog frequency in Bilinear
transformation?
In Bilinear transformation , the digital frequency and analog frequency are
related by the equation,
ΩT
Digital frequency, ω = 2 tan −1 or
2
2 ω
Analog frequency Ω = tan
T 2
where,
Ω - Analog frequency
33
2. The relation between analog and The relation between analog and digital
digital frequency is linear. frequency is nonlinear.
3. To prevent the problem of aliasing There is no problem of aliasing and so
the analog filters should be band the analog filter need not be band
limited. limited.
The magnitude and phase response of Due to the effect of warping, the phase
4. analog filter can be preserved by response of analog filter cannot be
choosing low sampling time or high preserved. But the magnitude response
sampling frequency. can be preserved by prewarping.
UNIT – IV
For linear phase FIR filter to have both constant group delay and constant
phase delay.
θ (ω ) = − α ω −π ≤ω ≤π
For satisfying above condition
h(n) = h( N −1 − n)
N −1
that is the impulse response must be symmetrical about n =
2
If one constant group delay is desired then
θ (ω ) = β − α ω
For satisfying the above condition
h( n) = −h( N −1 − n)
N −1
that is the impulse response must be antisymmetrical about n =
2
1. FIR filter is always stable because all its poles are at the origin.
34
2. A realizable filter can always be obtained.
3. FIR filter has a linear phase response.
4. What is the necessary and sufficient condition for the linear phase
characteristic of a FIR filter?
The necessary and sufficient condition for the linear phase characteristic
of a FIR filter is that the phase function should be a linear function of ω , which in
turn requires constant phase delay or constant group delay.
5. How the constant group delay and phase delay is achieved in linear phase FIR
filters.
Frequency response of FIR filters with constant group and phase delay
N −1
Phase delay, α = (i.e., phase delay is constant)
2
π
Group delay, β = ± (i.e., group delay is constant)
2
Impulse response, h(n) = - h( N -1 – n ) (i.e., impulse response is anti symmetric)
6. What are the possible types of impulse response for linear phase FIR filters?
There are four types of impulse response for linear phase FIR filters
35
3. Antisymmetric impulse response when N is odd.
4. Antisymmetric impulse response when N is even.
7. List the well known design techniques for linear phase FIR filter.
There are three well known methods of design techniques for linear phase
FIR filters. They are,
i) Fourier series method and window method
ii) Frequency sampling method
iii) Optimal filter design method
8. Write the two concepts that lead to the Fourier series or window method of
designing FIR filters.
The following two concepts lead to the design of FIR filters by Fourier series
method.
9. Write the procedure for designing FIR filter by Fourier series method.
36
N −1 N −1
−
h(0) + 2 h(n)( z n + z −n )
H ( z) = z 2
∑
n =1
In designing FIR filter using Fourier series method the infinite duration impulse
N −1
response is truncated at n= ± . Direct truncation of the series will lead to
2
fixed percentage overshoots and undershoots before and after an approximated
discontinuity in the frequency response.
12. Write the procedure for designing FIR filter using windows.
iii) Choose a window sequence w(n) and multiply the infinite sequence
hd (n) by w(n)to convert the infinite duration impulse response to
finite duration impulse response h(n)
h(n) = hd (n) w( n)
iv) Find the transfer function of the realizable FIR filter
N −1 N −1
−
h(0) + 2 h(n)( z n + z −n )
H ( z) = z 2
∑
n =1
37
The desirable characteristics of the window are
1. The central lobe of the frequency response of the window should contain most
of the energy and should be narrow
2. The highest side lobe level of the frequency response should be small
3. The side lobe of the frequency response should decrease in energy rapidly as
ω tends to π
16. Write the procedure for FIR filter design by frequency sampling method.
The procedures for FIR filter design by frequency sampling method are
38
17. What is meant by Optimum equiripple design criterion? Why it is followed?
20. List the features of FIR filter designed using rectangular window.
i) The width of the transition region is related to the width of the mainlobe of
window spectrum
ii) Gibb’s oscillations are noticed in the passband and stop band
iii) The attenuation in the stopband is constant and cannot be varied
39
22. Write the equation specifying Hamming windows.
2n ( N −1) ( N −1)
wT ( n) =1 − for − ≤ n≤
N −1 2 2
=0 Otherwise.
2n
α 1 −
N −1 ( N −1)
wk ( n ) = I 0
I 0 (α) for n ≤
2
=0 Otherwise.
Where
α is an independent parameter.
I 0 (x) is the zeroth order Bessel function of the first kind
2
∞ 1 x k
I 0 ( x) = 1 + ∑
k! 2
k =1
26. Write the characteristics features of Triangular window.
40
The characteristic features of Triangular windows are
8π
i) The main lobe width is equal to
N
ii) The maximum sidelobe magnitude is -25dB
iii) The sidelobe magnitude slightly decreases with increasing ω
27. Why the triangular window is not good a good choice for designing FIR
filters?
41
4. In FIR filter designed using In FIR filter designed using Hanning
rectangular window the minimum window the minimum stopband
stopband attenuation is 22dB. attenuation is 44dB.
42
33. Compare the Hamming and Blackman window.
43
Sl.No Hamming window Kaiser window
1. The width of mainlobe in window The width of mainlobe in window
8π spectrum depends on the values of α and
spectrum is
N N
2. The maximum sidelobe magnitude The maximum sidelobe magnitude with
in window spectrum is -41dB respect to peak of mainlobe is variable
using the parameter α
3. In window spectrum the sidelobe In window spectrum the sidelobe
magnitude remains constant with magnitude decreases with increasing ω
increasing ω
4. In FIR filter designed using In FIR filter designed using Kaiser
Hamming window the minimum window the minimum stopband
stopband attenuation is 51dB attenuation is variable and depends on the
value of α
UNIT –V
FINITE WORD LENGTH EFFECTS
1.Architectural features
2.Execution speed
44
3.Type of arithmetic
4.Word length
1. Increased performance
2. Better compiler targets
3. Potentially scalable
4. Potentially easier to program
5. Can add more execution units, allow more instruction to be
packed into the VLIW instruction.
7. What is pipelining?
TMS320C50 – 4
TM 320C54x – 6
45
The different stages in pipelining are
The program bus carriers the instruction code and immediate operands from
program memory to the CPU.
The program address bus provides address to program memory space for both read
and write.
The data read bus interconnects various elements of the CPU to data memory
spaces.
The data read address bus provides the address to access the data memory spaces.
13. What are the elements used for the control processing unit of ‘c5X?
The elements used for the control processing unit of ‘c5X are
46
2. Parallel logic unit (PLU)
3. Auxiliary register arithmetic unit (ARAU)
4. Memory mapped registers
5. Program controller
The parallel logic unit is second logic units, which execute logic operations on
data without affecting the contents of accumulator.
47
20. What are load/store instructions?
LACB, LACC, LACL, LAMM, LAR, SACB, SACH, SACL and SAR
In fixed-point arithmetic the positions of the binary point is fixed. The bits to
the right represent the fractional part of the number and those to the left represent
the integer part. For example, the binary number 01.1100 has the value 1.75 in
decimal.
23. What is meant by block floating point representation? What are its
advantages?
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5.Overflow occur in addition Overflow does not arise
6.Used in small computers Used in larger, general purpose
computers
26. What are the three quantization errors due to finite word length registers in
digital filters?
The three quantization errors due to finite word length registers in digital filters are
27. How the multiplication and addition are carried out in floating point
arithmetic?
That is, mantissas are multiplied using fixed point arithmetic and the exponents
are added.
The sum of two floating point numbers is carried out by shifting the bits of the
mantissa of the smaller number to the right until the exponents of the two number are
equal and then adding the mantissas.
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.
Due to quantization of coefficients, the frequency response of the filter may differ
appreciably from the desired response and some times the filter may actually fail to
meet the desired response and some times the filter may actually fail to meet the
desired specifications. If the poles of desired filter are close to the unit circle, then
those of the filter with quantized coefficients may lie just outside the unit circle,
leading to unstability
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29. What is product quantization error (or) What is product round off error in
DSP?
In DSP, the continuous time input signals are converted into digital using
a b bit ADC. The representation of continuous signal amplitude by a fixed digit
produce an error, which is known as input quantization error.
Round is the process of reducing the size of a binary number to finite word
size of b bits such that, the rounded b-bit number is closest to the original
unquantized number.
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35. What is zero input limit cycle?
37. How the system output can be brought out of limit cycle?
The system output can be brought out of limit cycle by applying an input
of large magnitude, which is sufficient to drive the system out of limit cycle.
In fixed point addition the overflow occurs when the sum exceeds the
finite word length of the register used to store the sum. The overflow in addition may
lead to oscillations in the output which is called overflow limit cycle.
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