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0 0 1 0
1
(a) Using the same definition of differentiation operator as in Exercice 1.1,
write out the matrix form of the differentiation operator in C4 .
(b) Consider the following matrix:
1 0 0 0
1 1 0 0
A=
1
.
1 1 0
1 1 1 1
(a)
1 0 0 −1
−1 1 0 0
∆=I−D= .
0 −1 1 0
0 0 −1 1
1
P∞
(a) Let s[n] := + j 31n . Compute n=1 s[n].
2n
n
(b) Same question with s[n] := 3j .
(c) Characterize the set of complex numbers satisfying z ∗ = z −1 .
(d) Find 3 complex numbers {z0 , z1 , z2 } which satisfy zi3 = 1, i = 1, 2, 3.
Q∞ n
(e) What is the following infinite product n=1 ejπ/2 ?
s = 1 + z + z2 + . . . + zN ,
−zs = −z − z 2 − . . . − z N − z N +1 .
2
Summing the above two equations gives
1 − z N +1
(1 − z)s = 1 − z N +1 ⇒ s = .
1−z
Similarly
N2 2 −N1
NX
X z N1 − z N2 +1
z k = z N1 zk = .
1−z
k=N1 k=0
We have
N
X N
X N
X
s[n] = 2−n + j 3−n
n=1 n=1 n=1
1 1 − 2−N 1 1 − 3−N 1
= · −1
+ j · −1
= (1 − 2−N ) + j (1 − 3−N ) .
2 1−2 3 1−3 2
Now,
lim 2−N = lim 3−N = 0 .
N →∞ N →∞
Therefore,
∞
X 1
s[n] = 1 + j .
n=1
2
Since | 3j | = 1
3 < 1, we have limN →∞ (j/3)N = 0. Therefore,
∞
X j j(3 + j) 1 3
s[k] = = =− +j· .
3−j 10 10 10
k=1
zz ∗ = 1, ∀ z 6= 0.
3
(e) We have
N
1 1−2−N
2−n
π
PN
= ejπ 2 ·
Y
ej 2n = ejπ n=1 1−1/2 .
n=1
(c) Show that A = VDVT where the columns of V correspond to the eigen-
vectors of A and D is a diagonal matrix whose main diagonal corresponds
to the eigenvalues of A.
Ax = −x .
4
(b)
det (λI − B) = (λ − (α + β)) (λ − (α − β)) .
The eigenvalues are α − β and α + β. The eigenvectors are then
1 1
x= √ for λ = α − β
2 −1
1 1
x= √ for λ = α + β.
2 1
Remark that if β = 0, there is only one eigenvalue α and any non zero
vector is an eigenvector.
(c)
T 1 1 1 −1 0 1 1 −1 1 2 4
VDV = √ · ·√ = =A
2 −1 1 0 3 2 1 1 2 4 2
.
5
Module 2
(a) Recall one of the most important properties for finite dimensional sub-
spaces: The set of N non-zero orthogonal vectors in an N -dimensional
subspace is a basis for the subspace. Therefore, it is sufficient to just
prove the orthogonality of the vectors {w(k) }k=0,...,N −1 . Let us compute
6
the inner product, that is:
N −1 N −1
2π 2π
X X
hw(k) , w(h) i = w∗(k) [n]w(h) [n] = ej N nk e−j N nh
n=0 n=0
N −1
X 2π N if k = h
= e−j N n(h−k) =
0 otherwise.
n=0
Since the inner product of the vectors is equal to zero, we conclude that
they are orthogonal. However, they do not have a unit norm and therefore
are not the orthonormal vectors.
(b) In order to obtain
√ the orthonormal basis we normalize the vectors with
the factor 1/ N , having:
N −1
(k) (h)
X 1 2π 1 2π
hwnorm , wnorm i = √ ej N nk √ e−j N nh
n=0
N N
N −1
1 X −j 2π n(h−k) 1 if k = h
= e N =
N n=0 0 otherwise.
(a) Show that the set of all ordered n-tuples [a1 , a2 , . . . , an ] with the natural
definition for the sum:
(a) It is straight forward to verify that the set of all ordered n-tuples [a1 , a2 , . . . , an ]
with
• [a1 , a2 , . . . , an ] + [b1 , b2 , . . . , bn ] = [a1 + b1 , a2 + b2 , . . . , an + bn ]
7
• α[a1 , a2 , . . . , an ] = [αa1 , αa2 , . . . , αan ]
satisfies all the properties of a vector space, which are
(i) Addition is commutative.
(ii) Addition is associative.
(iii) Scalar multiplication is distributive.
(b) The set of signals of the form y(x) = a cos(x) + b sin(x) (for arbitrary a,
b) with the usual addition and multiplication by a scalar form a vector
space:
8
(a) The eight vertices of the cube can be represented by the following four
vectors:
Two vectors are orthogonal if their inner product is zero. In this case,
Therefore, the four diagonals of a cube are not orthogonal. Remark also
that it is not possible to have 4 orthogonal vectors in a space of dimension
3. So this also explains that the 4 diagonals cannot be orthogonal.
(b)
δ[n] = u[n] − u[n − 1]
u[n] = Σ∞
k=0 δ[n − k]
.
(c) By solving the equations,
(
f (t) = fo (t) + fe (t)
f (−t) = fo (−t) + fe (−t) = −fo (t) + fe (t)
We have
f (t) − f (−t)
fo (t) =
2
and
f (t) + f (−t)
fe (t) =
2
.
9
Module 3
Exercise 3.1
Derive a simple expression for the DFT of the time-reversed signal
in terms of the DFT X of the signal x. Hint: you may find it useful to remark
−(N −k)
that WNk = WN .
−(N −n)
By replacing WNn with WN , we get
N −1 N −1
−(N −n)k −(n+1)k
X X
Xr [k] = x[N − 1 − n]WN = x[n]WN
n=0 n=0
N −1 N −1
n(N −k)
X X
= WN−k x[n]WN−nk = WN−k x[n]WN
n=0 n=0
Consider now the length-2N signal obtained by interleaving the values of x with
zeros
x2 = [x[0] 0 x[1] 0 x[2] 0 ... x[N − 1] 0]T
10
.
Express X2 (the 2N -point DFT of x2 ) in terms of X.
Knowing that
WNnk , 0≤k<N
2π 2π
2nk
W2N = e−j 2N 2nk = e−j N nk = n(k−N )
WN , N ≤ k < 2N
we get
( P
N −1 nk
n=0 x[n]WN = X[k], 0≤k<N
X2 [k] = PN −1 n(k−N )
n=0 x[n]WN = X[(k − N )], N ≤ k < 2N
Exercise 3.3
Compute the DFT of the length-4 real signal x = [a, b, c, d]T . For which values
of a, b, c, d ∈ R is the DFT real?
2π 2π 2π
X[k] = a + be−j 4 k + ce−j 4 2k + de−j 4 3k
11
Solution of Exercise 3.4
Now,
N −1
X 2π N for (i + n) = 0, N, 2N, 3N, . . .
e−j N (i+n)k = = N δ[(i+n) mod N ]
0 otherwise
k=0
so that
N
X −1
y[n] = x[i]N δ[(i + n) mod N ]
i=0
N x[0] for n = 0
=
N x[N − n] otherwise.
12
N −1
1 X 2π
IDF T (X(k)X ∗ (k)) [m] = X(k)X ∗ (k)ej N km
N
k=0
N −1 N −1 N −1
1 X X 2π
X 2π 2π
= [ x(n)e−j N kn ][ x∗ (l)ej N lk ]ej N km
N
k=0 n=0 l=0
N −1 N −1 N −1
1 X X X 2π
= x(n) x∗ (l) ej N k(l+m−n)
N n=0 l=0 k=0
N
X −1
= x(n)x∗ (n − m mod N )
n=0
= rxx (m)
(a) Express the DFT and inverse DFT (IDFT) formulas (analysis and syn-
thesis) as a matrix - vector multiplication.
(b) Is the DFT matrix Hermitian?
X = Wx.
WWH = WH W = N I
13
we obtain the synthesis formula in matrix-vector multiplication form:
1
x = WH X.
N
A = AH
Wnk = W∗kn
for all n, k ∈ {0, 1, . . . N − 1}, which is generally not the case. Consider,
for example, the case when n = k = 1. We would need to have:
2π 2π
e−j N = ej N
which is equivalent to
4π
ej N = 1.
N
Given Y (k) = X(2k), y[n], the 2 point inverse of Y (k) can be expressed as
N
2 −1
1 X 2π
y[n] = Y (l)ej N/2 nl
N/2
l=0
N
2 −1
2 X 2π
= X(2l)ej N 2nl
N
l=0
14
Now we note that (
n 2 if n is even
1 + (−1) =
0 if n is odd
y[n] can now be rewritten in terms of X(k) as
N −1
1 X 2π
y[n] = (1 + (−1)k )X(k)ej N nk
N
k=0
N −1 N −1
1 X 2π 1 X 2π
= X(k)ej N nk + X(k)ej N nk ejπk
N N
k=0 k=0
N −1 N −1
1 X 2π 1 X 2π N
= X(k)ej N nk + X(k)ej N k(n+ 2 )
N N
k=0 k=0
N N
= x[n] + x[n + ] n = 0...( − 1)
2 2
(a) Consider the signal x(n) = cos (2πf0 n). Use the ’fft’ function to compute
and draw the DFT of the signal in N = 128 points, for: f0 = 21/128 and
f0 = 21/127.
Explain the differences that we can see in these two signal spectra.
(b) Repeat the process, this time using the ’dftmtrx’ function and check that
the results are the same. What is the preferred option between the two?
(a) The spectrum of the signal x[n] is obtained using the Matlab commands
given below.
N=128;fo1=21/128;fo2=21/127;
n=0:N-1;
x1=cos(2*pi*fo1*n);x2=cos(2*pi*fo2*n);
X1=fft(x1);X2=fft(x2);
subplot(223),stem(n-N/2,fftshift(abs(X1)))
subplot(224),stem(n-N/2,fftshift(abs(X2)))
Note that we would expect to see just one sample at the frequency of the
signal.
In Figure 3.1, on the left, our prediction is satisfied and the 21st DFT
coefficient represents the exact signal frequency. In Figure 3.1 on the
15
70 70
X [k] X2[k]
1
60 60
50 50
40 40
30 30
20 20
10 10
0 0
−50 0 50 −50 0 50
16
right, the frequency of the signal f0 = 21/127 does not coincide with any
DFT frequency component. The signal energy is spread over all the DFT
components. This is called frequency leakage.
(b) We can achieve the same results using the dftmtx function instead, as
illustrated below.
W=dftmtx(N);
X3=W*x1;X4=W*x2;
norm(X1-X3)
norm(X2-X4)
subplot(223),stem(n-N/2,fftshift(abs(X3)))
subplot(224),stem(n-N/2,fftshift(abs(X4)))
(a) First, we note that x[n] can also be expressed as: x[n] = u[n] − u[n − a].
Using the DTFT shift property,
1 1 e−jaw e−jaw
X(ejw ) = + δ̃(w) − − δ̃(w).
1 − e−jw 2 1 − e−jw 2
17
20
15
|X(ejw|
10
0
0 1 2 3 4 5 6 7
w
Figure 3.2: DTFT of x[n].
1 − e−jaw
X(ejw ) = .
1 − e−jw
(b) We visualize the magnitude of X(ejw ) using Matlab with the following
code:
n=1:10000;
w=(n.*2*pi/max(n));
X=((1-exp(-j.*w.*20))./(1-exp(-j.*w)));
plot(w,abs(X));
N=30;
x1=[ones(1,20),zeros(1,N-20)];
X1=fft(x1);
plot(abs(X1));
18
20
15
|X1[k]|
10
0
0 5 10 15 20 25 30
w
Figure 3.3: DFT of x[n] for N = 30.
(d) The DFT of x[n] for various values of N is represented in Figure 3.4.
As we increase N , the DFT becomes closer and closer to the DTFT of
x[n]. We know that the DFT and the DFS are formally identical, and
as N grows, the DFS converges to the DTFT. We can use Matlab to
approximate the DTFT of any signal.
19
20 20
15 15
|X1[k]|
|X1[k]|
10 10
5 5
0 0
0 5 10 15 20 25 30 0 10 20 30 40 50
w w
20 20
15 15
|X1[k]|
10 |X1[k]| 10
5 5
0 0
0 20 40 60 80 100 0 200 400 600 800 1000
w w
Thus,
Z π Z π
1 1 ∗
X ∗ (ejw )Y (ejw )dw = Σn x[n]e−jwn Σm y[m]e−jwm dw
2π −π 2π −π
Z π
(1) 1
= Σn x∗ [n]ejwn Σm y[m]e−jwm dw
2π −π
Z π
1
= Σn Σm x∗ [n]y[m]ejw(n−m) dw
2π −π
Z π
(2) 1
= Σn Σm x∗ [n]y[m] ejw(n−m) dw
2π −π
(3)
= Σn x∗ [n]y[n],
20
where (1) follows from the properties of the complex conjugate, (2) from
swapping the integral and the sums and (3) from the fact that
Z π
1 jw(n−m) 1 if m = n
e dw = .
2π −π 0 if m 6= n
(b) hx[n], x[n]i corresponds to the energy of the signal in the time domain;
hX(ejw ), X(ejw )i corresponds to the energy of the signal in the frequency
domain. The Plancherel-Parseval equality illustrates an energy conser-
vation property from the time domain to the frequency domain. This
property is known as the Parseval’s theorem.
x[n] ↔ DT F T X(ejω ).
We have:
∞
X ∞
X
x[−n]e−jωn = x[m]ejωm
n=−∞ m=−∞
∞
X
= x[m]e−j(−ω)m
m=−∞
−jω
= X(e )
with the change of variable m = −n. Hence, we obtain that the DTFT of the
time-reversed sequence x[−n] is:
x[−n] ↔ DT F T X(e−jω )
Similarly:
∞
X ∞
X
x[n − n0 ]e−jωn = x[m]e−jω(m+n0 )
n=−∞ m=−∞
∞
X
= e−jωn0 x[m]e−jωm
m=−∞
−jωn0 jω
= e X(e )
21
with the change of variable m = n − n0 . We, therefore, obtain the DTFT of the
time-shifted sequence x[n − n0 ]:
P∞
(a) Using the definition of DTFT, X(ω) = n=−∞ x(n)e−iωn .
Since X(ω) is periodic in 2π, the discretization with N samples is obtained
with ω = 2πk/N
∞
2π X
X k = x(n)e−i2πkn/N k = 0, 1..N − 1
N n=−∞
∞ (r+1)N −1
X X
= x(n)e−i2πkn/N
r=−∞ n=rN
N
X −1 ∞
X
= x(n + rN )e−i2πkn/N
n=0 r=−∞
(b) Let,
∞
X
xp (n) = x(n + rN )
r=−∞
22
is the DFS of the periodic sequence. Applying the inverse transform, we
get the signal xp (n). The sequence of interest, x(n) corresponds to one
period of the periodic signal, xp (n) when N ≥ L
(
xp (n) if 0 ≤ n ≤ N − 1
x(n) =
0 otherwise
When N < L,
∞
X
xp (n) = x(n + rN )
r=−∞
xp (n) is no longer the exact periodic repition of x(n). It is now the sum of
N-shifted versions of the signal x(n) and since N < L, each N-tap signal,
is corrupted by a shifted version of itself.
(c) Consider the previous result for N > L,
N −1
2π X
X k = xp (n)e−i2πkn/N ,
N n=0
where we note that xp (n) = x(n), for n ∈ [0, N − 1]. Then, (??) can be
rewritten as
NX −1
2π
X k = x(n)e−i2πkn/N .
N n=0
The right hand side of the above equation is the DFT sum, while the left
hand side is one period of the uniformly sampled version of the DTFT,
sampled such that N > L.
23
Consider a lowpass filter hlp [n] with bandwidth ωb . If we consider the sequence
24
Module 4
The system is not time-invariant. To see this consider the following signals
x[n] = δ[n]
y[n] = δ[n − 1]
(b) Using the results found in (a), compute the DTFT of x[n].
(a) x[n] can be written as the convolution of x1 [n] and x2 [n] defined as
1 −(M − 1)/2 ≤ n ≤ (M − 1)/2
x1 [n] = x2 [n] =
0 otherwise.
= u[n + (M − 1)/2] − u[n − (M + 1)/2].
25
Then,
(1) follows from the fact that x1 [n] = x2 [n] and the symmetry of x1 [n]. (2)
follows by noticing that the sum corresponds to the size of the overlapping
area between x1 [k] and its n-shifted version x1 [k − n].
(b)
(1) 1 1 jω(M −1)/2 −jω(M +1)/2
X1 (ejω ) = + δ̃(ω) e − e
1 − e−jω 2
(2)ejω(M −1)/2 − e−jω(M +1)/2 e−jω/2 (ejωM/2 − e−jωM/2 )
= −jω
=
1−e e−jω/2 (ejω/2 − e−jω/2 )
sin(ωM/2)
=
sin(ω/2)
(1) follows from the DTFT of u[n] (2) follows from ejw(M −1)/2 δ̃(w) =
e−jw(M +1)/2 δ̃(w) = δ̃(w). Now, using the convolution theorem, we can
write
(a) The impulse response of an LTI system is shown in Figure 4.1. Determine
and carefully sketch the response of this system to the input x[n] = u[n−4].
(b) Calculate the impulse response of the system of Figure 4.2 given the im-
pulse responses of the separate blocks
• h1 [n] = 3(−1)n ( 14 )n u[n − 2]
• h2 [n] = h3 [n] = u[n + 2]
• h4 [n] = δ[n − 1]
26
1
0.5
0
h[n]
−0.5
−1
−1.5
−2
−10 −5 0 5 10
n
which means the system is BIBO stable. The system is causal because
h[n] = 0 for n < 0.
27
Figure 4.2: Impulse response block schema
(a) • H is linear:
H{ax1 [n] + bx2 [n]} = ax1 [−n] + bx2 [−n] = aH{x1 [n]} + bH{x2 [n]}.
• H is BIBO stable:
|x[n]| ≤ M ⇒ |H{x[n]}| ≤ M.
• H is not causal.
• H is not LTI, therefore h[n] does not characterize the system.
28
4
3
h[n]
0
−10 −5 0 5 10
n
(b) • H is linear:
H{ax1 [n]+bx2 [n]} = e−jωn (ax1 [n]+bx2 [n]) = aH{x1 [n]}+bH{x2 [n]}.
• H is BIBO stable:
• H is causal.
• H is not LTI, therefore h[n] does not characterize the system.
(c) • H is linear:
n+n
X0
H{ax1 [n]+bx2 [n]} = (ax1 [k]+bx2 [k]) = aH{x1 [n]}+bH{x2 [n]}.
k=n−n0
• H is time invariant:
n+n
X0 n
X
H{x[n − n0 ]} = x[k − n0 ] = x[k] = y[n − n0 ].
k=n−n0 k=n−2n0
• H is BIBO stable:
• H is not causal.
29
• The system impulse response is
(
1 if |n| ≤ |n0 |,
h[n] =
0 otherwise.
(d) • Note that all inputs x[n] can be expressed as a linear combination of
delayed impulses:
∞
X
x[n] = x[k]δ[n − k].
k=−∞
y[n] = (a + b)n!u[n].
R{x[n]} = x[−n]
Let H be a linear time invariant system. What can you say about the following
system?
y[n] = R{H{R{H{x[n]}}}}
Let X(ejω ) be the DTFT of the sequence x(n) and H(ejω ) be the frequency
response of the LTI operator H. By the time reversal property of DTFT, the
DTFT of x(−n) is X(e−jω ). Since x(n) is real, the DTFT of x(−n) is X ∗ (ejω )
30
H{x(n)} = H(ejω )X(ejω )
R{H{x(n)}} = H ∗ (ejω )X ∗ (ejω )
H{R{H{x(n)}}} = H(ejω )H ∗ (ejω )X ∗ (ejω )
R{H{R{H{x(n)}}} = H ∗ (ejω )H(ejω )X(ejω )
= |H(ejω )|2 X(ejω )
The above system is a linear, time invariant system with zero phase delay.
By using the modulation theorem we can see that a high-pass filter with cut-off
frequency at π − ωc can be written as
π
hhp (n) = (2(cos( n))2 − 1)hlp (n)
2
Express the Fourier transform WT of wT [n] with respect to the Fourier transform
WR of wR [n].
31
Solution of Exercise 4.7
We observe that
ŵT [n − 1] = wR [n] ? wR [n]
, which becomes in the Fourier domain
using the convolution property. We also have that wT [n] = 12 ŵT [2n], which
becomes in the Fourier domain, (see for example the proof in Chapter 11 of the
book, the formula for downsampling by 2)
1 1
WT (ejω ) = ŴT (ejω/2 ) + ŴT (ejω/2−π ).
4 4
Combining these two relations, we obtain
1 jω/2 2 jω/2
WT (ejω ) = e WR (e ) + WR2 (ejω/2−π ) .
4
32
Consider a causal LTI system with the following transfer function
3 + 4.5z −1 2
H(z) = −1
−
1 + 1.5z 1 − 0.5z −1
Sketch the pole-zero plot of the transfer function and specify its region of con-
vergence. Is the system stable?
We have to factorize the filter expression to make explicit the numerator and
denominator roots
3 + 4.5z −1 2 (z − 1.5)(z + 1.5)
H(z) = −1
− −1
=
1 + 1.5z 1 − 0.5z (z − 0.5)(z + 1.5)
In this way we see that the zeros of this system are in z01 = 1.5 and z02 = −1.5
and the poles in zp1 = 0.5 and zp2 = −1.5. Now we can draw the pole-zero plot
represented in Figure 4.4
0.5
Imaginary Part
−0.5
−1
In theory, the zero z01 = −1.5 cancels out the pole zp2 = −1.5. As our system is
causal, the ROC extends outward from the remaining outermost pole zp1 = 0.5.
As the unit circle is included in this ROC, the system is stable. However, in
33
practice, the stabilization of a system with this technique is a risky enterprise
since the exact cancellation of the pole outside the unit circle will be extremely
sensitive to numerical precision issues. If the coefficients of the filter (or the
internal accumulators) are subject to truncation or rounding, the implicit posi-
tion of the zero may drift ever so slightly from the implicit position of the pole
and the system will no longer be stable.
(a) Compute the transfer function and check the stability of this system both
analytically and graphically.
(b) If the input signal is x[n] = δ[n] − 3δ[n − 1], compute the z-transform of
the output signal and discuss the stability.
(c) Take an arbitrary input signal that does not cancel the unstable pole of
the transfer function. Compute the z-transform of the output signal and
discuss the stability.
and we obtain
Y (z) z −1 + 3z −2
H(z) = = =
X(z) 1 − 3.25z −1 + 0.75z −2
z −1 (1 + 3z −1 ) z+3
= −1 −1
= .
(1 − 0.25z )(1 − 3z ) (z − 0.25)(z − 3)
Since the system is causal, the convergence region is |z| > 3. We can see
that there is the pole z = 3 that is out of the unit circle and therefore the
system is unstable. See Figure 4.5.
(b) Z-transform of the output signal is
z −1 (1 + 3z −1 )
Y (z) = H(z)X(z) = (1 − 3z −1 )
(1 − 0.25z −1 )(1 − 3z −1 )
z −1 + 3z −2
= .
1 − 0.25z −1
34
2.5
1.5
0.5
Imaginary Part
−0.5
−1
−1.5
−2
−2.5
−3 −2 −1 0 1 2 3
Real Part
From Y (z) we can see that the unstable pole z = 3 is canceled and only
the pole z = 0.25 of Y (z) is left. Since the system is causal, even from
the unstable system we can get the stable output if the unstable pole is
canceled by the input signal.
(c) See for example the input and output signals depicted in Figure 4.6.
Note that the unstable pole is not canceled and the output signal is there-
fore Y (z) is unstable function.
Exercise 4.11 Properties of the z-transform
Let x[n] be a discrete-time sequence and X(z) its corresponding z-transform
with appropriate ROC.
(a) Prove that the following relation holds:
Z d
nx[n] ←→ −z X(z).
dz
(b) Show that
Z 1
(n + 1)αn u[n] ←→ , |z| > |α|.
(1 − αz −1 )2
35
input signal x[n]
2
−1
−2
−3
0 5 10 15 20 25 30
0
0 5 10 15 20 25 30
(c) Suppose that the above expression corresponds to the impulse response of
an LTI system. What can you say about the causality of such a system?
About its stability?
(d) Let α = 0.8, what is the spectral behavior of the corresponding filter?
What if α = −0.8?
d d
Σn x[n]z −n
X(z) =
dz dz
= Σn (−n)x[n]z −n−1
= −z −1 Σn nx[n]z −n
36
(b) We have that
Z 1
αn u[n] ←→ .
1 − αz −1
Using the previous result, we find
αz −1
n Z d 1
nα u[n] ←→ −z = .
dz 1 − αz −1 (1 − αz −1 )2
Thus,
Z αz −1
(n + 1)αn+1 u[n + 1] ←→ z
(1 − αz −1 )2
and
Z 1
(n + 1)αn u[n + 1] ←→ .
(1 − αz −1 )2
The relation follows by noticing that
(c) The system is causal since the ROC corresponds to the outside of a circle
of radius α (or equivalently since the impulse response is zero when n < 0).
The system is stable when the unit circle lies inside the ROC, i.e. when
|α| ≤ 1.
(d) When α = 0.8, the angular frequency of the pole is ω = 0. Thus the filter
is lowpass. When α = −0.8, ω = π and the filter is highpass.
x[n] = . . . , h[−3], g[−3], h[−2], g[−2], h[−1], g[−1], h[0], g[0], h[1], g[1], h[2], g[2], h[3], g[3], . . .
(a) Express the z-transform of x[n] in terms of the z-transforms of h[n] and
g[n].
(b) Assume that the ROC of H(z) is 0.64 < |z| < 4 and that the ROC of G(z)
is 0.25 < |z| < 9. What is the ROC of X(z)?
37
1 1 1 3
Imaginary Part
Imaginary Part
0.2 0.2 0.2
3 3
0 0 0
= H(z 2 ) + z −1 G(z 2 )
(b) The ROC is determined by the poles of the transform. Since the sequence
is two sided, the ROC is a ring bounded by two poles zL and zR such
that |zL | < |zR | and no other pole has magnitude between |zL | and |zR |.
1/2
Consider H(z); if z0 is a pole of H(z), H(z 2 ) will have two poles at ±z0 ;
however, the square root preserves the monotonicity of the magnitude p and
therefore
p no new poles will appear between the circles |z|
p = |z L | and
2
|z|
p = |zR |. Therefore the ROC for H(z ) is the ring |zL | < |z| <
2 −1 2
|zR |. The ROC of the sum H(z ) + z G(z ) is the intersection of the
ROCs, and so
ROC = 0.8 < |z| < 2.
38
1 1 1
|H(ejW)|
|H(ejW)|
0.5 0.5 0.5
0 0 0
0 0.5 1 1.5 2 2.5 3 3.5 0 0.5 1 1.5 2 2.5 3 3.5 0 0.5 1 1.5 2 2.5 3 3.5
W W W
The first filter is a low-pass filter. Note that there are three poles located in low
frequency (near ω = 0), while there is a zero located in high frequency (ω = π).
The second filter is just the opposite. The zero is located in low frequency,
while the influence of the three poles is maximum in high frequency (ω = π).
Therefore, it is a high-pass filter. In the third system, there are poles which
affect low and high frequency and two zeros close to w = π/2. Therefore, this
system is a stop-band filter.
39
Module 5
i.e. the signal interpolated with a zero-centered zero-order hold and Ts = 1 sec.
(a) Express X0 (jΩ) (the spectrum of x0 (t)) in terms of X(ejω ).
(b) Compare X0 (jΩ) to X(jΩ). We can look at X(jΩ) as the Fourier trans-
form of the signal obtained from the sinc interpolation of x[n] (always with
Ts = 1):
X
x(t) = x[n]sinc(t − n)
n∈Z
Comment on the result: you should point out two major problems.
(c) The signal x(t) can be obtained from the zero-order hold interpolation
x0 (t) as x(t) = x0 (t) ∗ g(t) for some filter g(t). Sketch the frequency
response of g(t)
(d) Propose two solutions (one in the continuous-time domain, and another
in the discrete-time domain) to eliminate or attenuate the distortion due
to the zero-order hold. Discuss the advantages and disadvantages of each.
(a)
Z ∞
X0 (jΩ) = x0 (t)e−jΩt dt
−∞
Z ∞ ∞
X
= x[n]rect (t − n)e−jΩt dt
−∞ n=−∞
∞
X Z ∞
= x[n] rect (t − n)e−jΩt dt
n=−∞ −∞
∞ Z 1/2
X sin(Ω/2)
= x[n]e−jΩn e−jΩτ dτ = X(ejΩ )
n=−∞ −1/2 Ω/2
= sinc(Ω/2π) X(ejΩ ).
40
(b) Take for instance a discrete-time signal with a triangular spectrum repre-
sented in Figure (b).
41
Conversely, the spectrum of the continuous-time signal interpolated by
the zero-order hold looks like the on in Figure 5.3
There are two main problems in the zero-order hold interpolation as com-
pared to the sinc interpolation:
• The zero-order hold interpolation is NOT bandlimited: the 2π-periodic
replicas of the digital spectrum leak through in the continuous-time
signal as high frequency components. This is due to the sidelobes
of the interpolation function in the frequency domain (rect in time
↔ sinc in frequency) and it represents an undesirable high-frequency
content which is typical of all local interpolation schemes.
• There is a distortion in the main portion of the spectrum (that be-
tween −ΩN and ΩN , with ΩN = π) due to the non-flat frequency
response of the interpolation function. It can be seen in the zoom in
version of the main portion of the spectrum in the next slide.
(c) See Figure 5.4.
X(ejΩ ) if Ω ∈ [−π, π]
X(jΩ) =
0 otherwise,
42
where X(ejΩ ) is the DTFT of the sequence x[n] evaluated at ω = Ω. So
Ω Ω Ω
X(jΩ) = X(ejΩ )rect = X0 (jΩ)sinc−1 rect .
2π 2π 2π
Hence
−1 Ω Ω
G(jΩ) = sinc rect ,
2π 2π
This is a triangular characteristic with the same support as the zero-order hold.
If we pick an interpolation interval Ts and interpolate a given discrete-time
signal x[n] with I(t), we obtain a continuous-time signal:
X t − nTs
x(t) = x[n]I( )
n
Ts
(a) Compute and sketch the Fourier transform I(jΩ) of the interpolating func-
tion I(t) – Recall that the triangular function can be expresses as the
convolution of rect(2t) with itself.
43
(b) Sketch the Fourier transform X(jΩ) of the interpolated signal x(t); in
particular, clearly mark the Nyquist frequency ΩN = π/Ts .
(c) The use of I(t) instead of a sinc interpolator introduces two types of errors:
briefly describe them.
(d) To eliminate the error in the baseband [−ΩN , ΩN ] we can pre-filter the
signal x[n] before interpolating with I(t). Write the frequency response of
the discrete-time filter H(ejω ).
(a) From last exercise, we know that the Fourier Transform of rect(t) is
sin(Ω/2)
= sinc(Ω/2π)
Ω/2
In our case,
I(t) = 2 rect(2t) ∗ rect(2t)
so 2
1 sin(Ω/4) 1
I(jΩ) = 2· = sinc2 (Ω/4π)
2 Ω/4 2
which is represented in Figure 5.5
44
(b)
X t − nTs
x(t) = x[n]I( )
n
Ts
Then,
x[n]FT{I( t−nT
P
X(jΩ) = n Ts )} =
s
45
If we use H(ejω )X(ejω ) in the interpolation formula, we have
Ts ΩTs
X(jΩ) = H(ejΩTs )X(ejΩTs ) sinc2
4 4π
so that
Ts ΩTs −1
H(e jΩTs
) = [ sinc2 ] .
4 4π
Therefore, the frequency response of the digital filter will be
4 ω
H(ejω ) = sinc−2 , −π ≤ ω ≤ π
Ts 4π
prolonged by 2π-periodicity over the entire frequency axis.
Φ∗ = W
c.
For the interpolation we use the IDFT matrix with the columns pruned accord-
ing to the indices satisfying the aforementioned relationship,
c −1 = W
Φ = W c∗.
Note now that we can also multiply Φ∗ by any unitary matrix to get a plausible
sampling operator. That is U Φ∗ and ΦU ∗ are still ideally matched since U U ∗ =
U ∗ U = I.
46
Exercise 5.4 Another view of Sampling
One of the standard ways of describing the sampling operation relies on the
concept of “modulation by a pulse train”. Choose a sampling interval Ts and
define a continuous-time pulse train p(t) as:
∞
X
p(t) = δ(t − kTs ).
k=−∞
This is tricky to show, so just take the result as is. The ”sampled” signal is
simply the modulation of an arbitrary-continuous time signal x(t) by the pulse
train:
xs (t) = p(t) x(t).
Derive the Fourier transform of xs (t) and show that if x(t) is bandlimited to
π/Ts then we can reconstruct x(t) from xs (t).
In other words, the spectrum of the delta-modulated signal is just the periodic
repetition (with period (2π/Ts )) of the original spectrum. If the latter is ban-
dlimited to (π/Ts ) there will be no overlap and therefore x(t) can be obtained
simply by lowpass filtering xs (t) (in the continuous-time domain).
47
Figure 5.7: Spectrum of xc (t)
(a) What is the bandwidth of the signal? What is the minimum sampling
period in order to satisfy the sampling theorem?
(b) Take a sampling period Ts = Ωπ0 ; clearly, with this sampling period,
there will be aliasing. Plot the DTFT of the discrete-time signal xa [n] =
xc (nTs ).
(c) Suggest a block diagram to reconstruct xc (t) from xa [n].
(d) Therefore, we can exploit aliasing to reduce the sampling frequency nec-
essary to sample a bandpass signal. What is the minimum sampling fre-
quency to be able to reconstruct with the above strategy a real signal
whose frequency support on the positive axis is [Ω0 , Ω1 ]?
(a) The highest nonzero frequency is 2Ω0 and therefore xc (t) is 2Ω0 -bandlimited
for a total bandwidth of 4Ω0 . The maximum sampling period (i.e. the in-
verse of the minimum sampling frequency) which satisfies the sampling
theorem is Ts = π/(2Ω0 ). Note however that the total support over which
the (positive) spectrum is nonzero is the interval [Ω0 , 2Ω0 ] so that one
could say that the total effective positive bandwidth of the signal is just
Ω0 .
(b) The digital spectrum will be the rescaled version of the periodized continuous-
time spectrum
X∞
X̃c (jΩ) = Xc (j(Ω − 2kΩ0 )).
k=−∞
The general term Xc (jΩ − j2kΩ0 ) is nonzero for Ω0 ≤ |Ω − 2kΩ0 | ≤
2Ω0 for k ∈ Z. or equivalently
(2k + 1)Ω0 ≤Ω≤ (2k + 2)Ω0
(2k − 2)Ω0 ≤Ω≤ (2k − 1)Ω0
48
X a (e j ω )
π π
)= X c (j (Ω − 2k Ω∆ )).
49
k =−∞
Ω0 ≤ |Ω − 2kΩ∆ | ≤ Ω1 for k ∈ Z.
Exercise 5.6
Consider a bandlimited continuous-time signal x(t) with a spectrum X(jΩ) as
sketched in Figure 5.9
Sketch the DTFT of the output signal for each of the following systems, where
the frequency of the raw sampler is indicated on each line.
50
Figure 5.9: Spectrum of x(t)
51
−π −π/2 0 π/2 π
(a)
Figure 5.12: DTFT for system (a)
−π −π/2 0 π/2 π
(a)
−π −π/2 0 π/2 π
(b)
−π −π/2 0 π/2 π
(b)
Figure 5.13: DTFT for system (b)
−π −π/2 0 π/2 π
(c)
52
−π −π/2 0 π/2 π
(c)
(a) The solution is represented in Figure 5.12
(b) The solution is represented in Figure 5.13
Exercise 5.7
Consider a stationary i.i.d. random process x[n] whose samples are uniformly
distributed over the [−1, 1] interval. Consider a quantizer Q{·} with the follow-
ing characteristic:
−1 if − 1 ≤ x < −0.5
Q{x} = 0 if − 0.5 ≤ x ≤ 0.5
1 if 0.5 < x ≤ 1
The quantized process y[n] = Q{x[n]} is still i.i.d.; compute its error energy.
Exercise 5.8
Consider a stationary i.i.d. random process x[n] whose samples are uniformly
distributed over the [−1, 2] interval.The process is uniformly quantized with a
1-bit quantizer Q{·} with the following characteristic:
(
−1 if x < 0
Q{x} =
1 if x ≥ 0
3
R1 ∆2
In this quantizer, K = 3, B = −1, A = 2. Thus Pe = 3 0
τ 2 dτ = 3 = (1)/3.
(B−A)2 Px 9
Px = 12 = 0.25. Thus SN R = Pe = 4
53
Module 6
Exercise 6.1
Let x[n] be the discrete-time version of an audio signal, originally bandlimited
to 20KHz and sampled at 40KHz; assume that we can model x[n] as an i.i.d.
process with variance σx2 . The signal is converted to continuous time, sent over
a noisy analog channel and resampled at the receiving end using the following
scheme, where both the ideal interpolator and sampler work at a frequency
Fs = 40KHz. The channel is represented in Figure 6.1
The channel introduces zero-mean, additive white Gaussian noise. At the re-
ceiving end, after the sampler, assume that the effect of the noise introduced
by the channel can be modeled as a zero-mean white Gaussian stochastic signal
η[n] with power spectral density Pη (ejω ) = σ02 .
(a) What is the signal to noise ratio (SNR) of x̂[n], i.e. the ratio of the power
of the “good” signal and the power of the noise?
(b) The SNR obtained with the transmission scheme above is too low for our
purposes. Unfortunately the power constraint of the channel prevents us
from
R π simply amplifying the audio signal (in other words: the total power
jω
P
−π x
(e ) cannot be greater than 2πσx2 ). In order to improve the quality
of the received signal, we modify the transmission scheme as shown in
Figure6.2, by adding pre-processing and post-processing digital blocks at
the transmitting and receiving ends, while Fs is still equal to 40000Hz.
Design the processing blocks A and B so that the signal to noise ratio of
x̂[n] is at least twice that of the simple scheme above. You should use
upsamplers, downsamplers and lowpass filters only.
(c) Using your new scheme, how long does it take to transmit a 3-minute song
signal?
54
Figure 6.2: Modified channel
so that the SNR is simply σx2 /σ02 . Just as in oversampling, the idea is to
send the signal more slowly as to occupy less bandwidth. If the signal has
a smaller bandwidth, we can increase its amplitude without exceeding the
power constraint, which will allow us to have a better SNR over the band
of interest.
(b) Consider the following preprocessing chain depicted in Figure 6.3, where
the lowpass filter has a cutoff frequency π3
#
3
x [n ] 3↑ x ′ [n ]
so that the power constraint is fulfilled. The signal and the noise at the
receiver after the sampler have PSDs represented in Figure 6.4.
At the receiver we can filter out the out-of-band noise with the following
scheme depicted in Figure 6.5, where, once again, the lowpass has cutoff
frequency π/3.
After the filter the psd is represented in Figure 6.6 and after the down-
55
3σx2
2σx2
σx2
σ02
0 π/3 2π/3 π
Figure 6.4: PSD of the signal (black) and noise (gray) after sampler
xˆ′ [n ] 3↓ x̂ [n ]
3σx2
2σx2
σx2
σ02
0 π/3 2π/3 π
Figure 6.6: PSD of the signal (black) and noise (gray) after filtering
56
σx2
σ02 /3
0 π/3 2π/3 π
Figure 6.7: PSD of the signal (black) and noise (gray) after downsampler
(c) 9 minutes
We know that σα2 = |α|2 pa (α). Then, we have to compute |α| to solve
P
α∈A
the exercise.
• Constellation (a):
57
Im Im
x=1 x=1
Re Re
(a) (b)
Figure 6.8: Constellations
√ √
– |α1 | = x2 + x2 = 2x
√
– |α2 | = 1 + 3
• Constellation (b):
√ √
– |α1 | = x2 + x2 = 2x
p √
– |α2 | = x2 + (3x)2 = 10x
Constellation(a):
√ √
4( 2x)2 +4(1+ 3)2
• σα2 = 2
P
α∈A |α| pa (α) = 8 = 4.73
Constellation(b):
√ √
4( 2x)2 +4( 10x)2
• σα2 = |α|2 pa (α) =
P
α∈A 8 =6
58