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N
1
Power lim
2
x(n) (for aperiodic signal)
N 2 N 1
n N
1 N 1
Power x(n) (for periodic signal)
2
N n 0
Energy
2
x ( n)
n
4.Find the convolution of the input signal {1, 2, 1} and its impulse response {1, 1, 1}
using Z transform.
x n 1,2,1 h n 1,1,1
2
X z x n Z n 1 2z 1 z 2
n 0
2
H z h n Z n 1 z 1 z 2
n 0
Y z X z H z 1 2z 1 z 2 1 z 1 z 2
1 2z 1 z 2 z 1 2z 2 z 3 z 2 2z 3 z 4
1 3z 1 4z 2 3z 3 z 4
y n 1,3,4,3,1
5.Define twiddle factor. Write its magnitude and phase angle.
The N point DFT is given by,
N 1 j 2 kn
X (K ) x (n )e N , k=0,1,2……..,N-1
n 0
j 2
Let wN e N which is called as twiddle factor.
N 1
X (K ) x (n )wN kn
n 0
2
Its magnitude is one and phase angle is N
6.Compute the number of multiplications and additions for 32 point DFT and FFT.
DFT FFT
PART B – (5 X 13 = 65 marks)
11.(a)Determine the following systems are linear, stability and time invariance of the
system (i) y(n) = x(2n), (ii) y(n) = cos x(n), (iii) y(n) = x(n) + n x(n+1) (13)
Linearity:
i) y[n] = x[2n]
The output y(n) for an arbitrary input x1(n) is, y1 (n) x1 (2n)
The output y(n) for an arbitrary input x2(n) is, y2 (n) x2 (2n)
Thus the sum of the individual response is,
y1 (n) y2 (n) x1 (2n) x2 (2n) 1
When the inputs are applied simultaneously then the response due to sum of inputs
is,
y1 (n) y2 (n) x1 (2n) x2 (2n) 2
As an equation (1) is equal to the equation (2) the given system is linear.
ii) y[n] = Cosx[n]
The output y(n) for an arbitrary input x1(n) is, y1 (n) Cos x1 (n)
The output y(n) for an arbitrary input x2(n) is, y2 (n) Cos x2 (n)
Thus the sum of the individual response is,
y1 (n) y2 (n) Cos x1 (n) Cos x2 (n) 1
When the inputs are applied simultaneously then the response due to sum of inputs is,
y1 (n) y2 (n) Cos x1 (n) x2 (n) 2
As an equation (1) is not equal to the equation (2) the given system is non linear.
iii) y[n] = x[n]+ n x[n+1]
The output y(n) for an arbitrary input x1(n) is, y1 (n) x1 (n) nx1 (n 1)
The output y(n) for an arbitrary input x2(n) is, y2 (n) x2 (n) nx2 (n 1)
Thus the sum of the individual response is,
y1 (n) y2 (n) x1 (n) nx1 (n 1) x2 (n) nx2 (n 1) 1
When the inputs are applied simultaneously then the response due to sum of inputs
is,
y1 (n) y2 (n) x1 (n) x2 (n) n x1 (n 1) x2 (n 1) 2
As an equation (1) is equal to the equation (2) the given system is linear.
Time-invariant vs. time-variant:
i) y[n] = x[2n]
if x1[n] = x[n − n0]
The response due to delayed input is, y1[n] = x1[2n] = x[2(n − n0)]
while the delayed response is, y[n − n0] = x[2n − n0]
So the system is time variant.
ii) y[n] = Cos x[n]
if x1[n] = x[n − n0]
The response due to delayed input is, y1[n] = Cos x[n − n0]
While the delayed response is, y[n − n0] = Cos x[n − n0]
So the system is time-invariant.
iii) y[n] = x[n]+ n x[n+1]
if x1[n] = x[n − n0]
The response due to delayed input is, y1[n] = x[(n − n0)]+ n x[(n − n0)+1]
while the delayed response is, y[n − n0] = x[(n − n0)]+ [(n − n0) x[(n − n0)+1]
So the system is time-variant.
Stability:
i) y[n] = x[2n]
The impulse response is, h(n) (2n)
n
h(n) ... 0 0 1 0 0 0 ... 1
n
h(n) ... 1 1 Cos1 1 1 1 ...
n
h(n) ... 0 1 1 0 0 0 ... 2
Let xq(n) denote the sequence of quantized samples at the output of the quantizer.
Hence xq(n) = Q[x(n)]. Then the quantization error is a sequence eq(n) defined as the
difference between the quantized value and the actual sample value.
(ii) Rounding (rounded off to next integer level). The distance between
two successive quantization levels is called the quantization step size (or) resolution.
The quantization error eq(n) in rounding is limited to the range of 2
to 2
. In other
words, the instantaneous quantization error cannot exceed half of the quantization
step. If x min & x max represent the minimum and maximum values of x(n) & the
number of quantization levels, then
x max x min
L1
The quantization error decreases and the accuracy quantizer increases as the number
of quantization level increases. The quantization error eq(t) = xa(t) – xq(t)
Let denotes the time that xa(t) stays within the quantization level. The mean-
square error power Pq is,
1 1
Pq
2 e q (t ) dt e2q (t ) dt
2
0
eq ( t ) t
2
2
1 2
Pq
t dt
2 12
0
If the quantizer has ‘b’ bits of accuracy and the quantizer covers the entire range 2A,
2A 4 A2 A2
the quantization step is 2b
. Hence, Pq 22 b 12 22 b 3
3t
11.(b) (ii) Compute the Nyquist sampling frequency of the signal x (t ) 4 Sinc (3)
3t
Sin
3t 4Sin(3t)
x (t ) 4 Sinc 4 3t
3t
The maximum frequency component is,
2 f m 3 f m 3/ 2
Nyquist rate is, Fn 2 f m 3/ Hz
12.(a)(i) State and prove convolution and Parseval’s theorem of Z transform. (6)
Convolution theorem
If x1 (n) & x2 (n) are two sequences
x1 (n)
z
X 1 ( z ) with ROC : R1 &
x2 (n)
z
X 2 ( z ) with ROC : R2
x (n) x2 (n) X1 ( z) X 2 ( z) with ROC : R1 R2
Then 1 z
Proof:
Z .T x1 (n) x2 (n) x (k ) x (n k ) z
1 2
n
x (k ) x (n k ) z
1 2
nk
Hence proved.
Parseval’s theorem:
If x1 (n) & x2 (n) are two complex valued sequences
x1 (n)
z
X 1 ( z ) with ROC : R1 &
x2 (n)
z
X 2 ( z ) with ROC : R2
N 1
1
Then Z .T x1 (n) x2 (n)
N
X (k )Y
K 0
*
(k )
Proof:
12.(a). (ii) Find the Z transform of the system x(n ) Cos(n )u(n ) (7)
Solution:
e j n e j n
Cos (n )u (n) for n 0
2
Cos (n )u (n) 0.5 e j n z n e j n z n
n 0 n 0
e j n e j n
Cos (n )u (n) 0.5
n 0 z n 0 z
z z 2z e e
j j
z
0.5 2
z z e e 1
j j
z e z e 2 j j
z 2 z cos
Cos (n )u (n)
z 2 2 z cos 1
12. (b).Find the inverse Z transform of X(z) = (z+1) / (z+0.2) (z-1), z 1
Using residue method. (13)
Solution:
1
2 j
n 1
x ( n) X ( z )z dz
C
10 5
5 0
3 3
i.e, x(0) 0
for n 1
( z 1) z n 1
x(n) residues of at poles z 0.2 and z 1
( z 0.2)( z 1)
( z 1) z n 1 ( z 1) z n 1
residue of at z 0.2 residue of at z 1
( z 0.2)( z 1) ( z 0.2)( z 1)
( z 1) ( z 1)
( z 0.2) ( z 1)
( z 0.2)( z 1) z 0.2 ( z 0.2)( z 1) z 1
2 5
(0.2) n 1
3 3
2 5
Therefore, x(n) (0.2) n1 u(n 1) u (n 1)
3 3
13.(a)Determine the 8 point DFT of the sequence x(n) = {1, 1, 1, 1, 1, 0, 0,0} (13)
N 1 j 2 kn 7 j 2 kn
( ) ( )
X k x n e N x n e 4 where N 8 and k 0,1, 2, 3, 4, 5, 6, 7
n 0 n 0
j k
j 3 k j 5 k j 7 k
j k
X k x (0) x (1)e 2 x (2)e x (3)e 2
x (4)e j 2 k x (5)e 2
x (6)e j 3 k x (7)e 2
j 12
x (2)e j 4
j 2
X (4) x (0) x (1)e x (3)e 2
x (4)e j 8
x (5)e j10 x (6)e j12 x (7)e j14 1
j 5
j15
X (5) x (0) x (1)e 2 x (2)e j 5 x (3)e 2
x (4)e j10
j 25 j 35
x (5)e 2
x (6)e j15 x (7)e 2
j 0.414
j 6
j 18
j 6
X (6) x (0) x (1)e 2 x (2)e x (3)e 2
x (4)e j12
j 30 j 42
x (5)e 2
x (6)e j18 x (7)e 2
1
j 7
j 21
j 7
X (7) x (0) x (1)e 2 x (2)e x (3)e 2
x (4)e j14
j 35 j 49
x (5)e 2
x (6)e j 21 x (7)e 2
j 2.414
X k 5, j 2.414,1, j 0.414,1, j 0.414,1, j 2.414
13. (b)Compute 8 point DFT of the given sequence using DIT algorithm
n n 7
x (n )
0 otherwise (13)
2 1
j
w41 w82 j; w83 e 8
0.707 j 0.707
X ( K ) 28, 4 j9.6565, 4 j 4, 4 j1.656, 4, 4 j1.656, 4 j 4, 4 j9.6565
14.(a)Design a 15 tap linear phase filter using frequency sampling method to the
1 0 k 3
2 k
H 0.4 k 4
following discrete frequency response 15
0 k 5,6,7
2 pT
2 2000 2 104
pH tan 4
tan 7265 rad / sec
T 2 2 10 2
T 700 2 10
4
2 2
SH tan s 4
tan 2235 rad / sec
T 2 2 10 2
The characteristics are monotonic in both pass band and stop band. Therefore, the filter is
Butterworth filter.
Design of prototype LPF:
pL 1
pH
SL 3.25
SH
Determination of order:
100.1 s 1 100.1(10) 1
log 0.1 p log
10 1 100.1(3) 1 0.932
N
log SL log 3.25
PL
Choose N=1
Determination of cutoff frequency:
C pH 7265rad / sec
C
The transfer function of highpass filter is obtained using the transformation S i.e.,
S
7265
S
S
1 S
H (S )
1 S S
7265 S 7265
S
1 z 1
10000
S 1 z 1
S 7265 S 2 1 z 1 1 z 1
210 4
1
1 z 10000 1
7265
1 z
0.5792(1 z 1 )
H ( z)
1 0.1584 z 1
15.(a)Discuss the features and architecture of TMS 320C50 processor. (13)
15. (b)Explain the addressing modes and registers of DSP processors. (13)
Refer Q.No. 15. (b) May 2015 Answer Key for addressing modes
PART C – (1 X 15 = 15 marks)
16.(a)The analog signal has a bandwidth of 4KHz. If we use N point DFT with N =
2m (m is an integer) to compute the spectrum of the signal with resolution less than
or equal to 25 Hz. Determine the minimum sampling rate, minimum number of
required samples and minimum length of the analog signal. What is the step size
required to quantize this signal. (15)
N=8000/25 =320
or
1 1 8000
25 L L 320 samples
LT 25T 25
1
LT 320 0.04sec
8000
16.(b)Convert the single pole low pass filter with system function
0.5(1 Z 1 )
H (z ) into band pass filter with upper and lower cutoff
1 0.302Z 1
frequencies u and l respectively. The lowpass filter has 3dB bandwidth and
p /6 and u 3 /6 , l /4 and draw its realization in direct form II.
[15]
a1
2 K
& a2
K 1
K 1 K 1
l p
K Cot u tan
2 2
u l u l
Cos /Cos
2 2
p /6 and u 3 /6 , l /4
We have,
We find, 0.414, K 1.4,a1 0.483& a2 0.167