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Adsp QB
Adsp QB
M 1
1
y[n]
M
x[n k ].
k 0
(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.
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:
Page 1 of 2
4 Realize the following IIR transfer function in the Gray-Markel (Lattice – Tapping) form:
Page 2 of 2
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]
Page 1 of 2
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.
Page 2 of 2
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
(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.
Page 1 of 2
(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
Page 2 of 2
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
(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 ) .
Page 1 of 2
(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
Page 2 of 2
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.
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:
Page 1 of 2
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.
nM
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].
Page 2 of 2
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
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 .
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
Page 1 of 2
(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
Page 2 of 2
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
b) Determine the necessary and sufficient condition for the convergence or stability of the steepest
decent algorithm. 5+5
Page 1 of 2
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
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) ni 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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