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DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

NATIONAL INSTITUTE OF TECHNOLOGY, ROURKELA


M.Tech., 1st Semester, Mid-Sem. Examination, Autumn 2019-20
Subject Code: EC-6601 Subject: Advanced DSP
Maximum Marks: 30 Time: 2 hours

Attempt all questions.


All parts of a question should be answered at one place.

1 (a) The response of a filter with an input x[n] is described as

M 1
1
y[n] 
M
 x[n  k ].
k 0

Show that the transfer function of the filter is a bounded-real function.

(b) A Type 2 real-coefficient FIR filter with a transfer function H ( z ) has the following zeros:
z1  3.1, z2  2  j 4, and z3  0.8  j 0.4 .
(i) Determine the locations of the remaining zeros of H ( z ) having the lowest order.

(ii) Determine the transfer function H ( z ) of the filter. 2+3


2 Consider the first-order causal and stable all-pass transfer function given by

1  d1 z
A1 ( z )  .
z  d1
Determine the expression for (1  A1 ( z) ) , and then show that
2

 0, for z  1,
2

2 

(1  A1 ( z ) )  0, for z  1,
2


 0, for z  1.
2

3 Test the BIBO stability of the following causal IIR transfer functions using Schur-Cohn algebraic
stability test method:

2.2 z 3  1.104 z 2  2.04 z  4


(a) H ( z )  ,
z 3  0.24 z 2  0.436 z  0.3
1
(b) H ( z )  .
6  5 z  4 z  3 z 3  2 z 4  z 5
1 2
5

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4 Realize the following IIR transfer function in the Gray-Markel (Lattice – Tapping) form:

(1  0.7 z 1 )(2  0.5 z 1  0.4 z 2 )


H ( z)  .
(1  0.3z 1 )(3  1.05 z 1  0.3z 2 ) 5
5 (a) Develop a cascade lattice realization of the following linear-phase FIR transfer function with
the reduced number of multipliers:

H ( z )  1  0.16 z 1  0.1z 2  0.16 z 3  z 4


(b) Consider an FIR transfer function given by

H ( z )  0.5  2 z 1  0.5 z 2  z 3  3.1z 5  0.3z 6  z 7  0.5 z 8


Determine the polyphase transfer functions and develop a four-branch polyphase realization. 5
6 Design a Butterworth low-pass filter with the following specifications: passband edge frequency (
f p ) = 1 kHz, stopband edge frequency ( f s ) = 2.2 kHz, minimum stopband attenuation ( As ) = 30
dB, and passband ripple ( Ap )= 1 dB.
5

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DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING
NATIONAL INSTITUTE OF TECHNOLOGY, ROURKELA
M. Tech, 1st Semester, End - Semester Examination, Autumn 2019-20
Subject: Advanced Digital Signal Processing
Maximum Marks: 50 Subject Code: EC-6601 Time: 3 hours
Instructions:
 Answer all questions.
 All parts of a question should be written in single continuous place.
 Steps are essential for mathematical/numerical calculations.

1. Determine the frequency response H (e j ) of a Type-I FIR filter. Show that the amplitude
(magnitude) response of a Type-I linear-phase FIR transfer function is a periodic function of
 with a period of 2 . [5]
2. A first-order all pass transfer function is given by
2  z 1
A( z ) 
1  2 z 1
Realize A( z ) with one multiplier and one delay using the multiplier extraction approach
(mention all steps). [5]
3. Design an IIR digital low-pass filter by transforming an analog Butterworth low-pass filter
using the impulse invariance transformation method. The low-pass filter specifications are
as follows: passband edge frequency (  p ) = 0.1 , stopband edge frequency (  s ) =
0.3 , minimum stopband attenuation (  s ) = 0.2, and passband ripple (  p )= 0.8 and
sampling interval (T) = 0.25 ms.

 b Te  aT sin bT z 1 
Use the transformation:   .
2 aT 2 
  s  a   b
 aT 1
2 2
1  2e cos bT z  e z  [5]
4. Design an IIR digital filter using the bilinear transformation method for the specifications
mentioned in the figure below.

[6]
5. Prove the cascade equivalence:

[4]

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6. Design a rational sampling rate converter using polyphase decomposition that can convert a
digital audio signal of 48 kHz sampling rate to one of 72 kHz sampling rate. [5]
7. Explain the polyphase implementation of M-band uniform analysis filter banks.
[5]
8. Explain the two-channel quadrature-mirror filter (QMF) bank. Evaluate the conditions for
aliasing cancellation and perfect reconstruction in the two-channel QMF bank. [5]
9. A statistical filtering problem is defined as follows.

Prove that the estimate of the desired response d (n) defined by the filter output y(n) and the
corresponding estimation error e(n) are orthogonal to each other when the filter operates in
its optimum condition (in minimum mean-square error sense). [5]
10. Explain least-mean-square (LMS) adaptive algorithm and derive its filter weight update
[5]
equation.

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DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING
NATIONAL INSTITUTE OF TECHNOLOGY, ROURKELA
M. Tech, 1st Semester, End - Semester Examination, Autumn 2020-21
Subject: Advanced Digital Signal Processing
Maximum Marks: 50 Subject Code: EC-6601 Time: 2 hours
Instructions:
 Answer all questions.
 All parts of a question should be written in single continuous place.
 Steps are essential for mathematical/numerical calculations.

1. (a) Consider the structure as shown below where the transfer functions G ( z ) and H ( z )
satisfy the condition

1 L 1  1/ L  j L2 k   1/ L  j L2 k 
 G  Z e H  Z e   1.
L k 0    

Show that v(n)  u (n) .

(b) A multirate structure is given below in the figure. Determine its input – output relations
in both z-domain and time domain.

[5+5]

2. (a) What is a uniform filter bank? Explain polyphase implementations of 5-band uniform
filter bank.
(b) A rational sampling rate interpolator is needed in the conversion of a digital audio signal
of 12 kHz sampling rate to one of 18 kHz sampling rate. Design this interpolator using
the polyphase decomposition (mention all steps). [5+5]
3. (a) The transfer functions of the low-pass and high-pass analysis filters of a two-channel
quadrature-mirror filter (QMF) bank of figure shown below are given by
H0 ( z)  a  bz 1 and H1 ( z)  c  dz 1 . Determine the expressions for the transfer
functions of the low-pass and high-pass synthesis filters, G0 ( z ) and G1 ( z ) , so that the
two-channel QMF bank is a perfect reconstruction system.

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(b) Consider a Wiener filtering problem characterized as follows: The correlation matrix R
of the tap-input vector u(n) is

 1 0.8
R .
0.8 1 
The cross correlation vector between the tap-input vector u(n) and the desired response
d (n) is

p   0.5 0.2 .
T

Compute the tap weights of the Wiener filter. What is the minimum mean-square error
produced by this Wiener filter if the variance of the desired signal d (n) is 0.562. [5+5]
4. (a) In a statistical filtering problem, the estimation error is described as

e(n)  d (n)  w H (n) u( n),


where d (n) is the desired response, u(n) is the tap-input vector, and w ( n) is the
weight vector of the filter. Show that the gradient of the instantaneous square error
J (n)  e(n) equals
2

J (n)  2u(n)d * (n)  2u( n) u H ( n) w ( n).


(b) For a statistical filtering problem, explain the method of steepest-descent to make
adjustments to weights of the filter. [5+5]
5. (a) Suppose you are receiving a signal denoted as s(n)  i(n)  v(n) through wireless
medium where i ( n) and v(n) denote the information signal and the channel noise,
respectively, both of zero mean. It is desired to filter out the information signal i ( n)
using the least-mean square (LMS) adaptive filter. Explain this filtering process in detail.
(Some assumptions can be made if needed.)
(b) Explain the importance of the forgetting factor in the recursive least-squares (RLS)
adaptive filters. Consider a correlation matrix  (n)  u( n) u H ( n)   I , where u(n) is
a tap-input vector,  is a small positive constant, and I is identity matrix. Use the matrix
inversion lemma to evaluate P (n)   ( n) . [5+5]
1

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DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING
NATIONAL INSTITUTE OF TECHNOLOGY, ROURKELA
M.Tech., 1st Semester, Mid-Sem. Examination, Autumn 2021-22
Subject Code: EC6601 Subject: Advanced DSP
Maximum Marks: 30 Time: 2 hours

Attempt all questions.


All parts of a question should be answered at one place.

1 (i) Determine spectrum of the output of an up-sampler (L = 3) with an input signal of


frequency response as shown below. (Mention frequencies in the plot corresponding to
the end-points and center point of spectral bands properly)

(ii) Show that the up-sampler is a linear and time-varying discrete-time system.
3+3
2 (i) Express the time-domain output y[ n] for the multirate structure of figure below as a
function of the input x[n] .

(ii) Determine the output sequences y1[n] and y2 [n] of the structure given below for the
causal input sequence x1[n]  [0 1 0 4 5 3] and x2 [n]  [2 0 0 1 3 3] .

3+3

3 Design a rational sampling rate converter using polyphase decomposition that can
convert a digital audio signal of 30 kHz sampling rate to one of 12 kHz sampling rate
6
(mention all steps).
4 (a) The prototype filter in a four-channel uniform DFT filter bank is characterized by
the transfer function H 0 ( z )  1  z 2  2 z 3  4 z 4 . Determine the transfer functions
of the filters H1 ( z ) , H 2 ( z ) , and H3 ( z ) .

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(b) Prove the cascade equivalence:

3+3
5 (a) Explain 3-channel quadrature mirror filter (QMF) banks.

(b) Suppose that low-pass filter of the analysis filter bank is defined as H 0 ( z )  1  z 2  z 3 ,
determine transfer functions of the high-pass filter H1 ( z ) of the analysis filter bank, and low-
pass filter G0 ( z ) and high-pass filter G1 ( z ) of the synthesis filter bank such that this 2-
channel QMF is a perfect reconstruction system. 6

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DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING
NATIONAL INSTITUTE OF TECHNOLOGY, ROURKELA
M.Tech., 1st Semester, End-Sem. Examination, Autumn 2021-22
Subject Code: EC6601 Subject: Advanced DSP
Maximum Marks: 50 Time: 2 hours
Attempt all questions.
All parts of a question should be answered at one place.

1. (a) Develop (and draw) a computationally efficient realization of a factor-of-3 decimator


employing an FIR filter

H ( z)  1  2 z 1  3z 4  2 z 5  1.5z 7  z 8.
Also, compute the number of multiplications required to produce an output of this
computationally efficient realization.
(b) Show that the multirate system of figure below is time-invariant and determine its
transfer function.

5+5
2.
(a) Prove the identity

L 1
where the transfer function H ( z )  z
k 0
k
Ek ( z L ).

(b) Discuss the impact of weighting factor and regularization term in the cost function of
recursive least-squares (RLS) algorithm.
(c) Discuss the impact of a negative value of the step size parameter on performance of the
steepest-descent algorithm. 5+3+2
3. (a) Consider a Wiener filtering problem characterized using the autocorrelation
matrix and the cross-correlation matrix of order four ( M  4 ) is as follows:

 1.1 0.5 0.1 0.1


 0.5 1.1 0.5 0.1 
R , P   0.5 0.4 0.2 0.1 .
T

 0.1 0.5 1.1 0.5 


 
 0.1 0.1 0.5 1.1 

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The variance of the desired signal d (n) is 1.0. Compute the mean-square error produced
by the Wiener filter for M  1, 2 .

(b) In the method of steepest descent, show that the correction applied to the tap-
weight vector after n 1 iterations may be expressed as
 w(n  1)   E u(n)e (n)  , where u(n) is the tap-input vector and e(n) is the

estimation error. Using the concept of the principle of orthogonality, discuss what
happens to this adjustment at the minimum point of the error-performance
surface? 5+5
4. (a) Let uˆ (n)  au(n), where a is a scaling factor. Let wˆ ( n ) and w ( n) denote the tap-
weight vectors computed at iteration n , which respectively correspond to the
input vectors u(n) and uˆ (n) . Show that the equality wˆ (n)  w (n) holds for the
normalized least-mean-squares (NLMS) algorithm when the weight vectors (
wˆ ( n ) and w ( n) ) are initialized with the same values.

(b) A statistic filtering problem is solved using the normalized LMS (NLMS) algorithm,
where the input signal u (n) and the desired signal d (n) are given as
u(n)  [1 2 0 1 1 3 2], d (n)  [0 1 1 0 2 3 2].
If the weight vector of 3rd iteration w (3)  [0.3 -0.1 0] and the step size parameter
T

  0.1 . Determine the weight vectors of 4th and 5th iterations, i.e., w (4) and w(5) ,
5+5
respectively.
5. (a) The M  by M time average autocorrelation matrix  of the tap inputs
u(n), u(n -1),..., u(n - M  1) can be calculated using
N
 (l , k )   u(n  k )u (n  l ), 0  (l , k )  M 1.
nM

Using the above expression show that the autocorrelation matrix  is a product of two
rectangular Toeplitz matrices.
(b) Consider a linear least-squares filter with filter length as 2 and a real-valued input time-
series u (n) consisting 6 samples are given as u (n)  [1 1 0 0 2 1]. Assuming the
observation window duration of 3 samples starting from the sample index n  2 ,
calculate the tap-weight vector and the estimation error vector of the linear least-squares 5+5
filter for the given desired signal d (n)  [1 1 1 0 1 1].

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NATIONAL INSTITUTE OF TECHNOLOGY, ROURKELA
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING
M.Tech., 1st Semester, Mid-Semester Examination, 2022-23 (Autumn)
Subject Code: EC6601 Subject: Advanced Digital Signal Processing
Maximum Marks: 30 Time: 2 hours

Attempt all questions.


All parts of a question should be answered at one place.

1 (i) A type-4 real-coefficient FIR filter (order = 5) with a transfer function H ( z ) has one of the
zeros at z = j 0.5 .

(a) Determine the locations of the remaining zeros of H ( z ) .


(b) Determine the transfer function H ( z ) of the filter.
(ii) The first five impulse response samples of a causal linear-phase FIR filter are given by
{a, b, c, d , e} . Determine the remaining impulse response samples of the transfer function
of lowest order for each type of linear phase FIR filters. 3+3
2 (i) Determine the output sequence y[n] of the structure given below for the causal input
sequence x[n] = 0, 1, 1, 4, 5, 0, 3 . Also, express the time-domain relationship between
x[n] and y[n] .

(ii) Prove the cascade equivalence:

3+3

3 (i) Consider the structure below in figure where the transfer functions G ( z ) and H ( z ) satisfy
the condition
1 L−1
 G ( z1 LWLk )H ( z1 LWLk ) = 1,
L k =0

where, WLk = e− j 2 k L . Show that it is an identity system, i.e., v[n] = u[n] .

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(ii) Prove that the specifications for the lowpass filter to remove the spectral images (i.e.,
interpolation filter) caused by an integer factor-of- L upsampler are given as
 L,   c L
H (e j ) = 
 0, otherwise,
where,  c denotes the highest frequency that needs to be preserved in the signal to be
interpolated. Also, discuss about the filter specifications for fractional sampling rate
converter with a factor L M . 3+3
4 (i) Explain M-band uniform filter bank. Derive the relationship among analysis filters.
(ii) Consider the two-channel analysis filter bank structure with analysis filters H k ( z ), k = 0, 1
, are FIR filters. The impulse response h0 [ n ] corresponding to the prototype filter H 0 ( z ) ,
with a cutoff frequency c =  2 , is given by
h0[n] = 0.0167, 0.4833, 0.4833, 0.0167.
Compute the impulse response h1[n] of the filter H1 ( z ) . Also, plot an interpretative
magnitude spectrum of both the filters. 3+3
5 (i) Develop (and draw) a computationally efficient structure (via polyphase decomposition) for
realizing a factor-of-4 interpolation employing the following linear-phase filter (mention all
the steps)
H ( z ) = -0.0054+0.0101z −1 -0.0966z −3 + 0.5919z −4 + 0.5919z −5 -0.0966z −6 + 0.0101z −8 -0.0054z −9 .

(ii) Determine an expression for the output y[n] for the multirate structure below in figure as a
function of the input x[n] .

3+3

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NATIONAL INSTITUTE OF TECHNOLOGY, ROURKELA
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING
M.Tech., 1st Semester, End-Sem. Examination, Autumn 2022-23
Subject Code: EC6601 Subject: Advanced Digital Signal processing
Maximum Marks: 50 Time: 3 hours
Attempt all questions.
All parts of a question should be answered at one place.
1. a) A sequence x(n) is up-sampled by L  2, it passes through an LTI system H1 ( z ) , and then it is
down-sampled by M  2. Can we replace this process with a single LTI system H 2 ( z ) ? If the
answer is positive, determine the system function H 2 ( z ) .

b) Design a rational sampling rate converter using polyphase decomposition that can convert a digital
audio signal of 30 Hz sampling rate to one of 20 Hz sampling rate. 5+5
2. a) Consider a two-channel QMF bank with the prototype lowpass analysis filter H 0 ( z ) is given by

H 0 ( z) 
1
2
1  z 1  .
Compute the highpass analysis filter H1 ( z ) , and the lowpass and highpass synthesis filters G0 ( z )
and G1 ( z ) , respectively, such that the QMF bank is a perfect reconstruction system.
b) Derive the condition for which two different cascade arrangements (given below) of a down-
sampler and an up-sampler are equivalent

5+5
3. a) A statistical filtering problem is defined as follows.

where, u (n) and d (n) are single realizations of jointly wide-sense stationary stochastic processes,
both with zero mean. Derive that the filter operates in its optimum condition (attaining the
minimum value of the mean square error) when

E u(n  k )e (n)   0, k  0,1, 2,...

b) Determine the necessary and sufficient condition for the convergence or stability of the steepest
decent algorithm. 5+5

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4. a) Consider a Wiener filtering problem characterized by the autocorrelation matrix R of the
tap-input vector u(n) as

 1 0.5 0.25 
R   0.5 1 0.5 
0.25 0.5 1 

and the cross-correlation vector p between the tap-input vector u(n) and the desired response

d (n) as p  0.5 0.25 0.125 .


T

1) Evaluate the tap weights of the Wiener filter.


2) What is the minimum mean-square error produced by this Wiener filter if the variance
of the desired response is 0.9?
b) The least-mean squares (LMS) algorithm is used to design an adaptive filter to solve a statistical
filtering problem, where the input signal u (n) and the desired signal d (n) are given as
u(n)  {1, 1, 0.5, 0.5, 1, 0, 1}, d (n)  {1, 1, 0, 0, 1, 1, 1}.
If the initial weight vector w (0)  [0 0 0] and the step size parameter   0.01 . Determine the
T

weight vectors of 1st and 2nd iterations, i.e., w (1) and w (2) , respectively. 5+5
5. (a) Explain the normalized LMS algorithm and derive its filter weight update equation.
(b) How is the recursive least-squares (RLS) algorithm different from the LMS algorithm? Show that
the M  M time-average correlation matrix  (n) of the tap-input vector u(i ) in the RLS
algorithm turns out to be
n
 (n)    ni u(i)uH (i)   n I ,
i 1

where, I is the M  M identity matrix,  is the regularization parameter, and  represents the
weighting factor. 5+5

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