(Solutions Manual) Oppenheim, Discrete Time Signal Processing Text
(Solutions Manual) Oppenheim, Discrete Time Signal Processing Text
Solutions - Chapter 2
-....
n
T(z(n -no]) = L z[k-no]
n -n,
= L z[k]
•-fto
'F 11(n - no] = L z[I:]
.......
The system is not Tl.
• Not Memoryless: Values of y(n) depend on past values for n > no, 10 tbis is not memoryless.
(c) T(z[n]) r:::~... z(k]
• Stable: IT(z[n])I $ r;::n~...
lz(k)I $ I;;!,.~ ... z(k]M $ l2no + llM for lz[n]I $ M, so it is
stable.
• Not Causal: T(z[n]) depends on future values of z(n], so it is not causal.
4
• Linear:
n+no
T(az,[n] + bz2[n]) = L az1[k] + bz2[k]
n+no n+no
= a L :t1[k] + b L :t2[k] = aT(z,[n]) + bT(:t2[n])
This is linear.
• TI:
n+no
T(:1:[n - no] = I:
.t=n-ne
:1:[k - nol
ft
= I:
t=n-no
:t[k]
= 11[n - no]
This is TI.
• Not memoryless: The values of 11[n] depend on 2no other values of :t, not memoryless.
(d) T(:t[n]) = :1:[n - no]
• Stable: IT(:1:[n])I = l:1:[n - no]!$ M if [:1:[n] $ M, so stable.
• Causality: If no ~ 0, this is causal, otherwise it is not causal.
• Linear:
T(azt[n] + bz2[n]) = azt[n - no]+ b:t2[n - no]
= aT(:tt[n]) + bT(x2 [n])
This is linear.
• TI: T(:t[n - n•J = :1:[n - no - n•J = 11[n - n•I• This is TI.
• Not memoryless: Unless no = 0, this is not memoryless.
(e) T(:t[n]) = e•l•l
• Stable: lz[n]I $ M, IT(x[n])I = [e•l•lj $ ,1•1•11 $ eM, this is stable.
• Causal: It doesn't use future values of :1:[n], so it causal.
• Not linear:
T(az,[n] + b:t2[n]) = ,u,{nJ+1,z,{n)
= eAZ1(n)eb2[n]
# aT(:t1(n]) + bT(z2[n])
This is not linear.
= =
• TI: T(:t[n - no]) e•l•-nol 11[n - no], so this is TI.
• Memoryless: 11[n] depends on the n••
value of :t only, so it is memoryless.
=
(f) T(:t[n]) az[n] + b
• Stable: IT(z[n])I = !az[n] + bl $ c>[MI + !bl, which is stable for finite a and b.
• Causal: This doesn't use future values of z[n], so it is causal.
• Not linear:
T(e:t1[n] + d:t2[n]) = ae:t1[n] + c>d:t2[n] + b
# cT(:t1[n]) +dT(:1:2[n])
This is not linear.
5
11(n] = L h(k]:z:[n - k]
1--oo
Note that the minimum value of (n - l:) is N2 • Thus, the lower bound on n, which occurs for
=
k No is
(b) H :r[n] # 0, for some n 0 $ n $ (n0 + N - 1), and h[n] # 0, for some n, $ n $ (n, +M -1), the
results of part (a) imply that the output is nonzero for:
(n0 + n 1 ) $ n $ (n0 + n 1 + M + N - 2)
So the output sequence is M + N - 1 samples long. This is an important quality of the convolution
for finite length sequences as we shall see in Chapter 8.
y[n] = }: h[k]:r[n - k]
The step response results when the input is the unit step:
forne:0
:r[n] = u[n) = { 0,l, for n < 0
Forn$0:
I: . -•
00
y[n] =
1=-CX)
00
= I: ...
c=-n
a-n
= 1-4
Forn>0:
•
y[n] = I: . -•
.. i=-m
= r; ..•
bO
1
= 1-a
7
The impulse response (for :r[n] = 6[n]) is the inverse Fourier transform of H(eiw).
H(eiw) _ -8 + 8
- 1 + le-J'"' I - le-Jw
• 2
Thus,
Y(z)
H(z) = X(z)
= l - Sz- 1 + 6z- 2
-2 2
= 1 - 2z- 1 + I - 3z-l'
where the region of convergence is outside the outermost pole, because the system is causal. Hence
the ROC is jz I > 3. Taking the inverse z-transform, the impulse response is
Y(eiw)[l - !e-jw]
2
= X(eiw)[l + 2e-)w + .-;:a..,].
Hence, the frequency response is
l + 2e-;.., + •-;:a..,
= 1- ½e-jw
cross multiplying,
=e"(;r,)(n+N) =e'"(7,n+2••J
e'"(7,n)
= 2.-k = ,/2N,for
" .mtegers k,N
Y(eJW)
H(e'w) = X(eiw)
le-2;...,
= 1 - !e-Jw
3
+ le-2Jw
• •
Now we take the ;,;...,rse Fourier transform to find the impulse response:
H(eP") =
h[n]
10
s[n] = L h[k]u[n - k]
l=-oo
= L• h(k]
l:=-00
1 - (1/3)•+• 1 - (l/2)n+ 1
= -2 1 - 1/3 u[n] + 2 1 - 1/2 u[n]
1 1
= (1 + ( )"-2( )")u[n]
3 2
(b) The homogeneous solution ll>[n] solves the difference equation when :i:[n] = O. It is in the form
ll•ln] = E A(cr' where the e's solve the quadrMic equation
5 1
c2--c+-=0
6 6
So for c = 1/2 and c = 1/3, the general form for the homogeneous solution is:
2.10. (a)
= L
lt~.
a•u[-k - l]u[n - k]
lo=-oo
n $-1
lo=-oo
= -I
:E ..., n>-1
l=-oo
a•
nS-1
= { 1-1/a'
1/a
1- 1/a'
n > -1
11
(b) First, let us define u(n] = 2nu(-n - l]. Tben, from part (a), we know that
2n+l n < -1
w(n] = u(n] • v(n] ={ l, ' n ; _1
Now,
n>3
(c) Given the same definitions for u(n] and w(n] from part(b), we use the fact that h(n] = 2n- 1 u[-(n-
l) - l] = u[n -1] to reduce our work:
y[n] = r[n] • h[n]
= r[n] • u[n - l]
= w[n-1]
{ 2n, n$0
= 1, n>0
We get:
2,J2e-;•/•,j•n/< - 2,,T2,1•l•e-i•n/<
y[n] =
2j
= 2J2sin(=/4 - -rr/4).
12
:(n] = 6[n - l]
the recursion yields
11[n] = 0, for n < 0
y(0] =0
11(1] = 1
11(2] =2
11(3] =6
11[4] = 24
Using h[n] from part (a),
2.13. Eigenfunctions of LTI systems are of the form an, so functions (a), (b), and (e) are eigenfunctions.
Notice that part (d), cos(wt,n) = .S(ei"'•n + e-;..,n) is a sum of two a" functions, and is therefore not
an eigenfunction itself.
2.14. (a) The information given shows that the system satisfies the eigenfunction property of exponential
sequences for LTI systems for one particular eigenfunction input. However, we do not know the
system response for any other eigenfunction. Hence, - can say that the system may be LT!, but
we cannot uniquely determine it. ==> (iv).
(b) If the system were LTI, the output should be in the form of A(l/2)", since (1/2)" would have been
an eigenfunction of the system. Since this is not true, the system cannot be LTI. ==> (i).
(c) Given the information, the system may be LTI, but does not have to be. For example, for any
input other than the given one, the system may output 0, making this system non-LTI. ==> (iii).
If it were LTI, its system function can be found by using the DTIT:
Y(ei"')
H(ei"') = X(e;"')
1
= 1 - le-;1o1
2
h(n] = (21 )"u(n]
2.15. (a) No. Consider the following input/outputs:
1
2:,(n] = o(n] ==> y,(n] = ( )"u(n]
.
4
1
>:2(n] = 6[n - l] ==> 112(n] = ( )"- 1 u(n]
4
Even though 2:2[n] = 2:,(n - l], 112[n] "# 11,[n - l] = (¾)"- u[n - l] 1
(b) No. Consider the input/output pair ,:2 (n] and 112[n] above. 2:2(n] = 0 for n < 1, but 112(0]-# 0.
(c) Yes. Since h[n] is stable and multiplication with u[n] will not cause any sequences to become
unbounded, the entire system is stable.
t.16. (a) The homogeneous solution y,[n] solves the difference equation when ,:(n] = 0. It is in the form
Y•(n] = L A(c)", where the e's solve the quadratic equation
l 1
c2- -c+
4
- =0
8
So for c = 1/2 and c = -1/4, the general form for the homogeneous solution is:
y,(n] =A,(.!)"+ A2(-.! )"
2 4
(b) Ta.king the z-transform of both sides, we find that
(b) We have
- { ½(1 + co&(1r;), for O $ n $ M
[ l-
wn 0, othenose
.
= R(eJw) • f:
n=-00
i(l + i,;'ff + ;,-;'if),-;.,
1 1 2,rl 2.-
= R(ei")•(i(w)+:i6(w+ M)+:i6(w- M))
15
(c)
IR(eiro )I
It (I)
-Jt
2.18. h[n] is causal if h[n] =0 for n < 0. Hence, (a) and (b) are causal, while (c), (d), and (e) are not.
2.19. h[n] is stable if it is absolutely summable.
(a) Not stable because h[n] goes to oo as n goes to oo.
(b) Stable, because h[n] is non-zero only for O $ n $ 9.
(c) Stable.
_, 00
So I: lh[n]I = 15.
2.20. (a) Taking the difference equation y[n] = (1/a)y[n - l] + :i:[n - l] and assuming h(O] = 0 for n < 0:
h(O] = 0
h[l] = 1
h(2] = 1/4
h[3] = (l/a) 2
y[n] = T{x[n]},
Let x[n] = 0 for all n.
11[n] = T{x[n]}
For some arbitrary x 1 [n], we have
11,[n] = T{x1 [n]}
Using the linearity of the system:
= L x[k]h[n - k]
k=-oo
• • • I2 Tl ••n
0 2 3
0 2 n
-1
~2
(c) y[n] =:t[n] • h[n]
5 5 5 s
4
4 4 4
3 3 3
3
2 2 2
2
O I 2 3 4 5 6 7 8 9 10 II 12 13 14 15 16 17 18 19 20 n
(d) y[n] = :r[n] • h[n]
17
-2 -1 0 I 2 3 4 5 6 n
-1
by linearity:
1 ,.,,
T{az,[n] + b:z:,[n]} = M, + M, + 1 L
lt=-M1
(a2:,[n] + b,:2[n])
l lrl2 l M2
= M, + M2 + 1
~-~
L a2:,[n] + M, + M, + 1 L
k=-M1
bx,[n]
= oy, [n] + by,[n]
= L 1=-oo
u[k - 4]h[n - k]
00
y[n] = L h[n - k]
t=4
Evaluating the above summation:
Forn<4: y(n]=O
For n = 4: 11[n] = h[O] = 1
For n = 5: 11[n] = h[l] + h[O] 2 =
For-n = 6: 11[n] = h{2] + h[l] + h[O] = 3
For n = 7: 11[n] = h[3] + h[2] + h[l] + h[O] = 4
For n = 8: y[n] = h[4] + h[3] + h{2] + h[l] + h[O] = 2
For n? 9: y[n] = h[5] + h[4] + h[3] + h[2] + h[l] + h[O] =0
18
L o:[k]h[n -
= A:=-co k]
00
= L 1:=-CIO
o:[k]u[n-k]
The convolution may be broken into five regions over the range of n:
11[n]
,.,,
11lnJ = I;a•
k=O
} _ 4 (N1+l)
= c__::__ _ '
1-a
for N 1 < n < N2
N, n
y[n] = L a.1i: + ~ a<'=-N2)
k=O l:=N2
J_ 4 (N1+l) 1_ 4 (n+l)
= 1-a + 1-a
2 - 4(N1+l) - a<n+l)
= l-a , for N2 $n $ (N, +N2)
N1 N1+N2
y[n] = I;a• + L a<•-N,l
b:::O k=-N2
,.,,
= I:a• + L N,a"'
2.26. Recall that an eigenfunction of a system is an input signal which appears at the output of the syste,
sea.led by a complex constant.
19
00
11[n) = 1: h(k)z[n-k)
t=-oo
00
= L h[k)5(n-•lu[n - k)
t=-oo
n
= 5• L h[kJ5-•
t=-oo
= .,,...,•. H(e':z.,)
YES, EIGENFUNCTION.
(c) ejwn + ,,;:z..n,
00 00
= ,,;wn L 00
h(k],-jwt + .,...,. L 00
h[k]e-j:Z.,t
Since the input cannot be extracted from the above expression, the sum of complex exponentials
is NOT AN EIGENFUNCTION. (Although, separately the inputs are eigenfunctions. In general,
complex exponential signals are always eigenfunctions of LTI systems.)
(d) z[n) = 5":
00
y(n] =
t=-oo
1: h[kJ5<•-•l
00
= 5• L h(k]5-•
k=-00
YES, EIGENFUNCTION.
(e) z[n] = 5"e'...,":
L
00
11[n) = h(kJ5<•-•l.,i:Z..(n-t)
=
k=-oo
5•,,i:Z..n L
.. h(k)5-•,-,2w•
t=-00
YES, EIGENFUNCTION.
20
2.27. • System A:
1
:i:[nl = C2l"
This input is an eigenfunction of an LT! system. That is, if the system is linear, the output will
be a replica of the input, scaled by a complex constant.
Since 11(n] =(¾)",System A is NOT Ln
• System B:
:i:(n] = e,i•l•u[n]
The Fourier transform of :i:(n] is
L
00
Hence, the system is a linear amplliier. We conclude that System B is LTI, and unique.
• System C: Since :i:(n] = ,i•I• is an eigenfunction of an LTI system, we would expect the output to
be given by
where -y is some complex constant, if System C were indeed LTI. The given output, y[n] =2ei•I•,
indicates that this is so.
Hence, System C is LTI. However, it is not unique, since the only constraint is that
2.28. z[n] is periodic with period N if :i:(n] = :i:[n + NJ for some integer N.
(a) :i:(n] is periodic with period 5:
= 2.-k =
2..
5
.
N,for mtegers k,N
(c) This is not periodic because the linear term n is not periodic.
(d) This is again not periodic. ,;w is periodic over period 2.-, so we have to find k, N such that
(a)
(b)
(c)
(d)
(e)
••••
2.30. (a) Since cos(.-n) only takes on values of +l or -1, this transformation outputs the current value of
:r[n] multiplied by eitber ±1. T(:i:[n]) = (-l)n:r:[n].
• Hence, it is stable, because it doesn't change the magnitude of :r:[n] and hence satisfies bounded-
in /bounded-<>ut stability.
• It is causal, because each output depends only on the current value of :r:[n].
• It is linear. Let 11,[n] = T(:r: 1 [n]) = cos(.-n):r: 1 [n], and y2[n] = T(:r: 2[n]) = cos(,m):r:2[n]. Now
Lo[n-k] =u[n]
>=O
So T(:r[n]) = :z:[n]u[n]. This transformation is therefore stable, causal, linear, but not time-
invariant.
To see that it is not time invariant, notice that T(o[n]) = o[n], but T(o[n + l]) = O.
(d) T(:r(n]) = I;:.,._, :r(k]
• This is not stable. For example, T(u[n]) = oo for all n i. 1.
• Ii is not causal, since it sums /OMJJard in time.
• It is linear, since
00 00 00
• It is time-invariant. Let
00
2.31. (a) The homogeneous solution y,[n] solves the difference equation when :r(n] = 0. It is in the form
y,(n] = I; A(c)", where the e's solve the quadratic equation
1 2
c2+-c-
15
-5 =0
So for c = 1/3 and c = -2/5, the general form for the homogeneous solution is:
h.[n] = .!.(!)"u[n]+
113
.!.c-!ru[n]
11 5
h [n] = _.!.(!)"u[-n-1]- .!.(-~)"u[-n -1]
""113 115
(c) Since h,{n] is causal, and the two exponential bases in hc[n] are both less than !, it is absolutely
summable. hoc[n] grows without bounds as n approaches -oo.
(d)
Y(z) =
X(z)B(z)
1 1
=
1-fz-1 {l-tz-1)(1 + fz-')
-25/44 55/12 27 /20
= 1 - 11az-1 + 1 + 2/sz- 1 + 1 - 3/sz-•
55 27
y(n] = ~(!)"u(n]+ (-!)"u[n]+ (~)"u[n]
44 3 12 5 20 5
23
111 [n]
G(,-ii)
Therefore,
and
y[n] = .,.,.y,[n] = w•t• cos(~n-
2
~)
2
2.33. Since H(,-iw) = H"(eN), we can apply the results of Example 2.13 from the text,
To find H(ei'f), we use the fact that H(eiw) is periodic over 2,r, so
Therefore,
3,, ~ 2,r 3,, 11,r
11[n] = cos(-n + - + -) = cos(-n + - )
2 4 3 2 12
2.34. (a) Notice that
r[n] =:to[n - 2] + 2.:o [n - 4] + :to[n - 6]
Since the system is LTI,
ll 5 6 7
-2
24
(bl Since
!lo[n] = -1:1:o[n + 1] + zo[n -1] = 3:o[n] • (-o[n + 1] + o[n - !]),
h[n] = -6[n + !] + 6[n - 1]
2.35. (a) Notice that zi[n] = z 2[n] + z 3[n + 4], so if T{-} is lillear,
T{z [n]} = T{:i:2[n]} + T{z3[n + 41}
1
= 112[n] + y3[n + 4]
From Fig P2.4, the above equality is not true. Hence, the system is NOT LINEAR.
(b) To find the impulse response of the system, we note that
6[n] = ,:3[n + 4]
Therefore,
T{6[n]} = y3 [n+4]
= 36[n + 6].+ 26[n + 5]
( c) Since the system is known to be time-invariant and not linear, we cannot use choices such as:
and l
6[n] = :r,[n + !]
2
to determine the in,pulse response. With the given information, we can only use shifted inputs.
Since,
L{o[n)} 'F L{o[n -1]}
The system is NOT TIME INVARIANT.
(b) An in,pulse may be formed:
-2 -1 0 I 2 n
-2
2.37. For an LTI system, we use the convolution equation to obtain the output:
00
Let n = m+ N:
00
= L :r[(m - k) + NJh[k]
t=-oo
= y[m]
So, the output must also be periodic with period N.
2.38. (a) The homogeneous solution to the second order difference equation,
3 l
11[n] - l] + 11[n - 2] = 2:r[n - l],
411[n - 8
is obtained by setting the input (forcing term) to zero.
and
A2 = 1/2.
(c) Homogeneous equation:
1
y[n] - y(n - 1] + -y(n - 2]
4
=0
Solving,
1- z-• + !z-
4
2
=0'
(l - iz- 1
)(1- iz- 1
) = 0,
and the homogeneous solution takes the form
ll•fn] = A,(½)" -
Invoking the intial conditions, we have
ll•l-1] = 2A1 =1
ll•I0] =A,= 0
Evident from the above contradiction, the initial conditions cannot be met.
(d) The homogeneous difference equation:
1
y(n] - y(n - 1] + y[n - 2]
4 =0
Suppose the homogeneous solution is of the form
Hence, if the input is doubled, the output must also double at each value of n.
Because y[0] = 1, always, the system is NOT LINEAR.
(c) Let :r3 = a:r.[n] + ,8:r2[n].
For n?: o,
ll>[n] = 2:3[n] + ay3[n - l]
• = a:r 1 [n] + ,8:r2[n] + a(:r3[n - 1] + 11,[n - 2])
n-1 n-1
= 0 I:>•:r,[n - k] + ,8 L a•2:2[n - k]
•=O 1=0
= o(h[n] • :r1 [n]) + ,8(h[n] • :r2[n))
= 01/l[n] + ,8112[n].
For n < 0:
113[n] =
=
a- 1 (113[n + l] - 2:3[n])
-Q L
.
a•:r,[n - k] - ,8 L
. a•z,[n - k]
•=-1 ~-1
= 011,(n] + .8112[n].
Forn=0:
113(n] = 1/l[n] = l/2[n] = o.
Conclude,
113[n] = 011,ln] + .81/2[n], for all n.
Therefore, the system is LINEAR. The system is nill NOT TIME INVARlANT.
28
2.40. For the input
:t[n] = cos(.-n)u(nJ
= (-l)"u{n],
the output is
"'
y[n] L (j /2)•u[k](-1J<"-•>u[n -
= f:=-oo k)
"
1-1r Lu12>•1-1i-•
= .....,
"
= 1-1>" Lc-;12>1
= (-l)
n c-
I,${)
(-j /2)(n+l))
l+j/2
X(e'w) = "' }:
}: "' cl[n + 16k]e-iw"
fto;;::.-00 lt=-00
1 "' 2,rk
= 16 L
,k::;:-00
0<w + T6 >·
Therefore, the frequency representation of the illput is also a periodic impulse train. There are
fr"')uency impulses in the range $ w $ 1r.
-,r
IX ,jw )I
29
From the sketch, we observe that the LTI system is a lowpass filter which removes all but three of tbe
frequency impulses. To these, it multiplies a phase factor ,-,...,.
The Fourier transform of the output is
Note that the Fourier transform of a"u[n] is well known, and t.he second term of h[n] (see part (a))
is just a scaled and shifted version of a"u[n]. So, we could have used the properties of the Fourier
transform to reduce the algebra.
(c) We have
30
cross multiplying,
Y(.,;-,)[l - a,-;"']= X(ei"')[l + 13,-;"']
= l - 2acos(w) + a2
(c) magnitude:
IX(e'"')I = [X(ei"')X"(ei"'JJ½
= (1-2a~(w)+a2 )½
(d) phase:
LX(ei"') = arctan ( l -asin(w)
- acos(w)
)
:i:[n)•-;"'"l..=0
= L
.. :1n1
=
(b) X(.,;-,)1.,=• =
6
..
L :i:(nJ•-;•n
00
= L
n=-oo
:r1n1c-1>"
= 2
31
(c) Because z[n] is symmetric about n = 2 this signal has linear phase.
X(.,;"') = A(w),-;:i.,
forn=O:
1_: X(_,;"')dw =2.-z(0] =4.-
(e) Let y[n] be the unknown sequence. Then
Y(e-i"') = xc,-;.. l
= L z[n]e-i"'"
n
= L z[-n],-;..
n
n
= I:;11[n],-;..n
n
•• l • T T• I o I
-1
-4
I I
(f) We have determined that:
XR(_,;..) = 1'.{X(e-i"')}
= A(w) cos(2w)
= !A(w) (e-i:i., + ,-;:i.,)
2
• •
-1/2
• -4
I
• T0 I • • I T ••••
I
4
1/2
.:J/2
32
2.45. Let 2:{n) = •In), then X(e'"') =1
The output of the ideal lowpass filter:
W(e'"') = X(e'"')H(e'"') = H(e'"')
The multiplier:
=
1
. + 1 ae'"'
1 - M-1 111
.
- ae,J"'
= anu[nj + a-nu[-n - l]
= aln\
(b)
1·
1,, -• X (e'"'} cos(w)dw = -
1 1• ;w
X(e'"')' +e
-jw
dw
2,r -• 2
2
= ! ( 2-
2 211' -·
1• X(e"")e""dw+ 1-_ 1•
271' -•
X(e'"')e-'"'tk>)
2.47. (a)
zln] + 2zln - l) + :i:{n - 2)
11[n) =
=
zlnJ • h[n]
:i:[nl • (6{n] + 26[n - l] + o[n - 2!)
=
hln] = 6\nJ + 'loin - l) + .S[n - 2j
33
(c)
H(e'w) = + 2e-Jw + 0 -2jw
l
= 2e-Jw(,!_e'w + l + ,!_ 0 -Jw)
2 2
= 2,-,w(cos(w) + 1)
(d)
IH(e'w)I = 2(cos(w) + 1)
LH(e'w) = -w
Magnitude
-lt lt (I)
-TC
-TC lt (I)
(e)
= ...!...1
2,r <2r>
H(e'(w+•)e'w"dw
= _.!._ 1 H(e'(w)e'(w-•l•tJw
2,r <2•>
= .-,,n_.!._ 1 H(e'(w)e'w"dw
2,r <b>
= -l"h(n]
= 6(n] - 26[n - l] + 6[n - 2]
..'1 ,1 JJ.
0 2 4 6
(b) Since y(n] = z[n]s[n],
34
= 2-1•
2,r -,r
S(d1 )X{d(w-l))dw
= X(d"') + X(ei<w-•l)
So, for a > 2, Y(e;"') contains two noIH>verlapping replications of X(&"'), whereas for a < 2,
"aliasing" occurs. When there is aliasing, W(&"') is not at all close to X(d"'). Hence, a must be
greater than 2 for w[n] to be "close" to :r[n].
0.5
0 oL---'--------''---
-2 0 2 -2 0 2
3 " "
5,--------------,
2.5
{ 2
2::
1.5 f -
oL--~'------'u...---"
6
-2
.
0 2 -2
.
0 2
~·
10
2:2
0
-2
.,0 2
(b) From part (a) we know that h[n] is length 3 with even symmetry around h[l]. Let h[0] = h[2] = a
and h[l] = b, from (iv) and using Parseval's theorem, we have
242 +b2 = 2.
From (v), we also have
2a-b= 0.
Solving the above equations, we get
1
h[0] = v'3
2
h[l] = v'3
1
h[2] = v'3
or
l·
h[0] = - v'3
2
h[l] = - v'3
1
h[2] = - v13·
2.50. (a) Carrying out the convolution sum, we get the following sequence q[n]:
4 4
3 3 3 3
q[n]
1 1 1 1
n
012345678910
(b) Again carrying out the convolution sum, we get the following sequence r[n]:
444444 r[n]
3 3
1 1
_ __,.,__.__.___._..._......____._.._......__.__._..,..........__,,-..,........,..-to--n
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16
-4
-8
-12
-16
-20
36
+oo
= L v[-k]w[k - n]
+oo
= L v[r]w[-n - r] where r = -k
= q[-n].
= !2 · [X(ei"') + X 0
(el"')]
l - acos(w)
= l - 2a cos(w) + a2
(b) imaginary part:
(c) m~tude:
IX(ei"')I = [X(ei"')X"(ei"'JJ½
= (1 - 2aco!(w) + a2) ½
(d) phase:
LX(ei"') = arctan ( -asin(w) )
1- acos(w)
5,m
:r[n] = 2 )
cos(
= cos( 2,m )
ei'i- .-;y
= -2-+-2-·
We now use the fact that complex exponentials are eigenfunctions of LTI systems, we get:
. 37
11(n]
2.53. First z[n] goes through a 1owpass filter with cutoff frequency 0.5.-. Since the cosine bas a frequency of
0.6,r, it will be filtered out. The delayed impulse will be filtered to a delayed sine and the constant will
remain unchanged. We thus get:
- nn(0.5,r(n - 5))
wInl - 3 --'--'---""- + 2.
r(n - 5)
Using the fact that complex exponentials are eigenfunctions of LTI systems, we get:
2.55. Since system l is memoryless, it is time invariant. The input, z[n] is periodic in w, therefore w[n] will
also be periodic in w. Al!. a consequence, y[n] is periodic in w and so is A.
2.56. (a)
where 11,[n] and 112[n] are the responses to :r 1 [n] and :r2[n] respectively. We thus conclude that
system S is linear.
(b) Let :r 2[n] =:r[n - no], then:
112[n] = h[n]
+co
• (,-;w,n:t2[n]l
= L ,-;wo(n-•>:r,[n - k]h[k]
••-oo
+co
L ,-;wo(n-•>:r[n - no - k]h[k]
= >=-co
"F 11[n - no].
11[n] = h[n]
+co
• (,-;wonz[n])
L .,-;wo(n-•>:r[n -
= k=-oo k]h[k]
+co
= L k=-oo
.,-;won_,jwo•z[n - k]h(k]
+co
= e-;won
E
l:=-oo
_,;...•x[n - k]h[k].
H,(_,iw) = H(_,i(w-•>J.
We thus have:
39
w
• _,, -0. 7.- -0.3,r O 0.3.- o. 7,r .-
(c) H 3 (e'w) corresponds to a periodic convolution of H,,(dw) with another lowpass filter, specifically:
------~-~-----w 0
"
2.59. (a) Using the change of variable: r = -k, we can rewrite /4[n] as:
00
We therefore have:
g[n] = z"[-n].
(b) The Fourier transform of z"[-n] is X"(~). therefore:
]4(ei"') = X"(eiw)X(eiw) = IX(ei"')l 2 -
2.60. (a) Note that z 2 [n] = - I:!~ :r[n - k]. Since the system is LTI, we haw:
v,[n] = -
....L k] .
..., y[n -
41
-1, n = O,n = 2
h[n] = -~ , n =I
{ o.w.
2.61. The system is not stable, any bounded inpnt that excites the zero input response will result in an
unbounded output.
The solution to the difference equation is given by:
Note that the first summation represents a weighted sum of future values of the input. Thus, if the
system is causal, _,
'E h{k)z[n - k) = 0.
This can only be guaranteed if h[k)= 0 for n < O.
Using reverse logic, we can show that if h(n) = 0 for n < 0,
-
00
h[n]
2.64. Let the input be z(n) = o[n - 1), if the system is causal then the output, r,[n), should be zero for n < 1.
Let's naluate y[O):
= _!_ 1+•
271' -•
e-;i.1e-J1,1/2 dw
2
= -3,.
-# o.
81(w)
.
•2
w
0
-j
-,r
"
0
-j
-,r
2.66. (a)
E(ei"') = H,(ei"')X(ei"')
F(ei"') = E(e-;"')
= H,<,-i"')X(,-;"')
G(ei"') = H 1 (ei"' )F(ei"')
= H, (ei"')H, (,-;"')X(,-;"')
Y(e'"') = G(,-'"')
= H, (,-1"')H1 (e'"')X(ei"').
(b) Since:
We get:
2.67. (a) Using the properties of the Fourier transform and the fact that (-1)" = e''", we get:
V(ei"') = X(ei<.,+<>)
W(ei"') = H 1 (ei"')V(ei"')
= Hi(e'"')X(ei<.,+•)
Y(ei"') = W(e'<w-•>)
= H1 (ei<w-•l)X (ei")
(b)
44
H(e'"')
w
0
2.68. If z, [n] =_z2{n], 1111 [n] and w,{n] will not he necessarily equal.
w,[n] = 2:1{-n - 2]
w,{n] = 2:2{-n + 2]
'F 2:2{-n - 2]
w,[n] = o{n + 2)
w,{n] = o[n - 2)-
2.69. (a) The overall system is not guar~ to he an LTI system. A simple counterexample is:
y,[n] = z[n]
112[n] = z(n]
y[n] = lit [n]112[n] = z 2(n]
Y,(,;-,) =
H,(e""')X(,;-,)
Y2 (el"') = H,(,;-,)X(e""')
Y(ei"') = Y,(ei"') • Y,(,;-,).
!
(a) To determine if the system is linear:
(c) Ta.king the Fourier transform of the result of part (b ), we find that the system function is
Thus,
LH(e'w) = arctan ( sin(w) )
. 1 - cos(w)
46
---......-----1----"----
0 ..
w
(d) In general,
V(z[n]) = z[n] • (o[n] - o[n - 1])
= z[n] - z[n - 1].
So,
and
h.[n] = u[n].
Hence, the unit step is the inverse system for the first difference.
2. 71. For impulse response h[n], the frequency response of an LTI system is given by
H(eiw) = L00
h[n]e-;wn
-=-oo
(a) Suppose the impulse response is h•[n],
= (J 00
h(n]eiwn) •
47
(b) We have
= L h•[n]ei=.
n=-oo
If h[n] is real,
00
H"(ei"') = L h[n]ei"'n
X(ei"') = L z[n],-iwn
n=-00
n=-00
L = :E
00
z"[-n],-jwn .,·[1Jei"''
n=-00 l=-ao
= (t.. z(I]•-;"'').
2.73. From property 1: 00
x·c,-;"'> = :E z[n1,-,.. n
= xc,,;-,i.
Thus, the Fourier tr~orm of a real input is conjugate symmetric.
X(ei"') = XR(ei"') + jXr(~"')
X•(,-i"') = XR(•-;"') - jX,(e;"')
48
We may infer
property 8: XR(e'"') = XR(e-i"')
property 9: X1(ei"') = -X1(e-i"')
X(ei"')= iX(e'"')le'LX(<'")
X"(e-;"') = IX(•-'"')1,-;Lx(,-;.l
From property 7:
So,
property 10: IX(ei"')I = IX(e-'"')I
property 11: LX(e'"') = -LX(e-1"').
2. 7 4. Theorem 1:
L L
00 00
00
L (cu,[n] + b:r2[n])e-'"'" =
n=-00
a.,:,[n]e-;wn +
n.=-00
b:r2[n],-;wn
n=-00
Theorem 2:
= 00
:,:[l),-jw(t-n,)
I:
n.=-00
:,:[n - n•J•-;wn = :E
l=-00
00
= ~"1ft4
I:
l=-oo
x[l]e- 1"''
= e'"'"' X(e'"')
Theorem 3:
L00
z[n]e'"'O"'e-jwn = L00
:t[n],-;(w-w,)n
n.=-00 n.=-00
Theorem 4:
00
L :,:[-n]e-;wn =
Theorem 5:
L00
n:&[n]e-;wn
n.=-00
49
= j: (=f;, kj,-1"")
00
z[k]
00
-h[n-
= Joo (Joo
z[k],-iw> h[n - k]e-iw(n->))
Hence,
2. 77. (a) The Fourier transform of 11•[-n] is Y"(e-'"), and X(e'")Y(e-'") forms a transform pair with z[n] •
y[n]. So
and
g[n] = z[n] • 11"[-n]
form a transform pair.
(b)
00
00 00
= L L z[k]y"[k - n]e-i""
a=-cob-00
forn=O:
50
.--+----. ½
-+-----'---'---'------w
• 0 ••
-,r
_z. ,r
Y(d"')
l
•
-,r
I
_ _ _ _ __JL-.l.......L---------"'
-i 0 i
Substituting into Eq. (P2.77-l):
00
I:
n=-oo
:[n]y" [n] = 2.._
211"
1•
-,r
X(e-i"')Y" (e-i"')dw
= 2... [<!)(!)(2")]
2.- 2 5 6
1
= 60
nL -,r O ,r
• w
(a)
,(n] { :[n), n even
11 = 0, nodd
= ~ (1 + e'"') :[n)
!
51
which transforms to
i
Y,(d"') = [x(d"') + X(d<..,+•>J]
-.-
[''~'
_ _,~<--'--~'----""'-.,-C.-""'----'--•
0 ..
w
(b)
Y•ln] =:r[2n]
Y.(d"') = i [x(d't) + X(d<'t+->i]
= Y,(d't)
(c)
2.79. (a)
00
*,(-N,-w) = L :r(n-N):r'[n+N]d"'n.
n=-oo
00
=
..
L (:,:(n + N]:r*(n - N],-;wn)*
=
..
L :,:*(n + N]:r(n - N]e.iwn
n=-oo
= +.(-N,-w).
(b)
+.(N,w) = L
.. Aan+Nu[n + N]Aan-Nu[n-NJ•-;wn
=
n=-oo
L ..
A2
_,, 0 2ne-jvln
=
..
A· L (a•,-;wr
n=N
= A2 (a•,-;w)N
l - a2e-Jw
= 0 2Ne-iwN
A 2 ..:..__;;___
1 - c 2 e-iwJ ·
(c)
X(e.i<•+(w/2))) =
..
L :r(n],-j(v+w/2)n.
..
n=-00
I:
x•c,;<•-<w121>i =
,.,._ .. :r·1n1,;<•-w/2)n_
Let S = ..L
21f
f'
-11
X(e;Cv+(w/2ll)X*(ei<•-Cw/ 2ll),i2•1' dv ' tben·.
s = -2l 1• L
"' -• n=-oo
oo :,:(n],-;c,+w/2)n
k=-oo
L"" :r*(k]e.i<•-w/2)•,_;2,N dv
= l
2.- Lco Lco :r(n]:r*(k]e-;•I•.♦'' 1• ,;c•-n+2N) dv
n=-oo.t=-oo -r
=
2,r
L
..n=-oo >=-co
z(n]:r*(n - 2N],-;-'..;"''
= L
.. :r(n + N):r*(n - N],-;wn
2.80.
(c)
</>,,[m] = E(:z:[n]s[n + ml)
= E((s[n] + e[n]ls[n + m]l
= <J,.,[m] + E(e[n])E(s[n]) since s(n] and e[n] are independent and stationary.
= <J,.,[m] where we assumed e(n] has zero mean.
Taking the Fourier transform of the above equation, we get:
••• (.,-"') = +,.(.,-"').
2.83. (Throughout this problem, we will assume lol < 1.)
(a)
<i>••lm] = h[m] • h[-m].
IB(.,-"')12 = H(.,;"')B'(e'"')
= H(.,;"')H(e-i"') since h[n] is real
= ••• (.,-"')
1 1
-L
n=-oc
lh[n]l2
=
+oc
= :E<101 2 in
n=O
1
= 1- 1012.
2.84. The first-had:ward-dilference system is given by:
---,~---=----r-:--m
-1 0 1
To get the power spectrum, we take the Fourier transform of the autocorrelation function:
·-
(b) The average power of the output of the system is given by tl>.. [O]:
2.85. (a)
..E
E{z(n]y(nl} = E{z(n]
..,_.., h[n]z(n - kl}
=
..
L h(k]E{z(n]z(n - kl}
=
..
i:=-00
L h(k]t/>.. [k]
>=-co
Because z[n] is a real, stationa,y white noise process:
</>•• [n] = o!o(n].
Therefore,
E{z[n]y(n]}
..
= a! L h[k]o[k]
t=-oo
= o!h[O].
(b) The variance of the output:
o-; = E{(y[n] - "'1,) 2 }
= E{y2 [n]} - m;.
When a zero-mean random process is input to a determistic LTI system, the output is also zero-
mean:
y(n] =
=
..
z[n] • h[n]
L :,;[k]h[n - k].
1=-00
"'1, = L
.. E{z[n]}h[n - k]
ts-oc
m, = 0, ifm. = o.
So,
o-; = E{y2 [n]}
=
..E L.. h[m]h[k]E{z[n - m]z[n - kl}
= a! E
.. ..
,as:-ooi=-oo
L h[m]h(k]o[m - k]
.. m=-005=-oo
,,.• = a! E "'lmJ.
• ffl,a:-00
!
57
=
.(1 ~~a{:;~;')
bn
n
Ea•bn-•,
..,
n?O
u[n]
w[n] = o:[n] • h(n].
Since o:[n] is zero-mean, mw = 0 also.
o-~ = E{w2 [n]}
= L h[k]E{z[n - kl}
=
{m,o, ~~-oo
11:=-00
h[k], n ;;::o
n<O
(b)
t/>,.[n1, n2] = E{11[n1]11(n2]}.
= E {J 00
h{A:]z[n, - k] J.., h(m]z[n2 - m]} , n ;;:: O
= 1: :E
A:s-00 m=-oo
h[A:]h[m]E{z[n, - A:]o:[n2 - ml}
L L
111 n2
(c)
co
= m, L h[k]
lim
n1,n2-oo
~
L-1 ~
L-1 h[k]h[m]t/>.,[n 1 - k, n 2 - m] =~
~
~
L.J h[k]h[m]4>.. [k, m].
t=-oo m=-oo i=-oo m--00
(d)
h[n] =anu[n]
co
E{y[n]} = m, L anu[n]
m,
= 1-a
2.88. (a) No, the system is not linear. In the expression of y[n], we have nonlinear terms such as :r2 [n] and
divisions by :r[n], :r[n - l] and :r[n + l].
(b) Yes, the system is shift inv.aria.nt. If we shift the input by no, m,[n] shifts by no as well as ~[n]
and c,;[n], therefore 11[n] shifts by no and the system is thus shift invariant.
(c) If :r[n] is bounded, m.[n] is bounded so is u!(n] and <>;[n]. ~ a result, y(n] is bounded and
therefore the system is stable.
(d) No, the system is not causal. Values of the output at time n depend on values of the input at time
n + I (through u!(n] and m,(n]). Since present values of the ouput depend of future values of the
input, the system cannot be causal.
(e) Wben U:,(n] is very large, u;[n] is zero, therefore:
11[n] = m,[n]
l n+1
= 3 L
i:=n-l
:r(k]
which is the average of the previous, present and next value of the input.
y(n] = :r(n].
y(n] makes sense for these extreme cases, because in very small noise power, the ouput is equal to
the input since the noise is negligible. On the other hand, in very large noise power, the input is
too noisy and so the output is an average of the input.
2.89. (a)
E{ :r[n]:r[n]} = q,.,(O].
(b)
+.,(ei"') = X(ei"')X.(elw)
= W(elw)H(elwJw·(elWJH·(elw)
= +_(ei")IH(e"')I'
= ~ 1
• 1 - cos(w) + 1/4 ·
59
(c)
= E {=too h[r]z[k - r) J 00
h(m)v[k - n - m)}
+11(eiw) = H1(e'w)H1(e-iw)+.,(e'w)
= (1 - e-iw)(l - e'w)cr:
= cr:(2 - ,1w - e-iw)
= ~(2 - 2 cos(w)).
60
(b) <P11{m] is the inverse Fourier transform of +11(e;.,)_ Using part (a), we get:
+.,(ei"') = H2(ei"')H2(•-;"')+11(ei"')
= { ,r,(2 - 2cos(w)) , lwl < We
o , w, < lwl s .-.
(d)
•
61
Solutions - Chapter 3
The z-Transform
63
3.1. (a)
= ~ (l)n
2 z-ft = ~ 2z1)n = 1 - 1
00
00 1
Z [( 2
l)ft
u[n]
] (
½z-1 l•I > 2
(b)
= - L
-1 (l)n oo
2 z-ft = - L(2z)"
n=-oo n=l
2z 1 1
= -1-
- -2z= 1- 1 ,- 1 l•I < 2
2
(c)
(d)
Z[c5[n]] = ,o = 1 all z
(e)
Z[c5[n - ll] = ,- 1 l•I > 0
(f)
(g)
•
Z [ (D ft (u[n] - u[n - 10])] = t( 1
2,) ft
1 - (2zJ- 10
1- (2zJ-•
l•I > o
3.2.
:r[n] = { nJV, 0<n<N-1
N-$ n- =n u[n]- (n - N)u[n - NJ
d d 1
n :r[n] ~ -zdzX(z)=:-nu[n]~-zdzl-z-l 1•1 > 1
,-1
n u[n)
-N-1
:r[n - no] ~ X(z) · z-... =:, (n - N)u[n - NJ~ (1z_ ,-1)2 lzl > I
therefore
64
3.3. (a)
-1 00
= La"z" + Lo"z-"
a=l n=O
az l z(l - a 2 ) l
= - - + - - -
l - az 1- az-1 = --'-----'-'
(l - az)(z - a) lal < lzl < lal
(b)
l, 0$ n$ N- l N-1 l _ z-N zN _ l
%11 ={ 0,
<
N <n
n 0
· ~ X,(z) L z-n = .,,---,-
= n=O l - z-1
= _.:.__:.._
zN-l(z - l)
z;<O
0,
(c)
x<(n] = z,[n -1] • z,[n] ~ X,(z) = z- 1X,(z) · X,(z)
z i 0,1
pole zero
---41}---~~-..fi;._:,,,,,.-=- cancel
65
X(z}
unit circle
-1 1/3 2 3
(a) For the Fourier transform of :z:(n] to exist, the z-transform of :z:(n] must have an ROC which includes
the unit circle, therefore, l½I < lzl < 121-
Since this ROC lies outside ½, this pole contributes a right-sided sequence. Since the ROC lies
inside 2 and 3, these poles contribute left-sided sequences. The overall :z:(n] is therefore two-sided.
(b) Two-sided sequences have ROC's which look like washers. There are two possibilities. The ROC's
corresponding to these are: l½I < lzl < 121 and 121 < lzl < 131.
(c) The ROC must be a connected region. For stability, the ROC must contain the unit circle. For
causality the ROC must be outside the outermost pole. These conditions cannot be met by any of
the possible RO C's oft his pole-zero plot.
3.5.
= L :z:(n]z-n
--00
Therefore,
3.6. (a)
I l
X (z) = -1-+~l-z--~, l•I >
2
2
+ J'
1 -•
(b)
1 1
X (z) = _l_+_l~z---, izl <2
2
Long division:
2z -4z2 +8z3 + ...
½z- 1
+1 I
1 + 2z
- 2z
-2z -4z2
+ 4z2
+4z2 +8z3
=>:i:[n)=-(-½)" u[-n-1)
(c)
I
izl >2
Partial Fractions:
-3 1
X(z) lzl >
2
:i:[n) =
Long division:
+ ...
(-¾-½Jz-• - iz-2
(-J- ½Jz-• + ¾(-J - ½Jz-•
(d)
I - lz- 1 I
2
X(z) -- 1- lz-2 izl >2
•
Partial Fractions:
I
X(z) l•I> 2
z[n]
67
Long division: see part (i) above.
(e)
Partial Fractions:
X(z) = -a - a-'(l - a•)
1- a-lz-1
lzl > la- 11
3.7. (a)
-1 1 1
X (z) = -1---z-_..,., + -1-_-lrz--"'",
2
2 < izl < 1
Now to find H(z) we simply use H(z) =Y(z)/X(z); i.e.,
H(z)
Y(z)
= -X(z) -- - - ~-½z--
1
- - · (1- z- 1 )(1- ½z- 1 )
---~~-- 1- ,- 1
- -1
(1 - ½z-1)(1 + z-1) -½z-' - 1 + z-
H(z) causal~ ROC lzl > 1.
(b) Since one of the poles of X(z), which limited the ROC of X(z) to be less than 1, is cancelled by
the zero of H(z), the ROC of Y(z) is the region in the z-plane that satisfi.. the remaining two
constraints lzl > ½and lzl > 1. Hence Y(z) converg.. on lzl > 1.
(c)
1 I
Y(z) = 1 - --
lz- 1
+ 3 1
1 + z-
lzl >1
2
Therefore,
y[n] = --31 (l)n
-2 u[n] + -(-ltu[n]
1
3
3.8. The causal system has system function
1- ,-,
H(z)- --.--:-
- l+ l,-1
•
and the input is :[n] = {lt u[n] + u[-n - 1). Therefore the z-transform of the input is
1 1 _ 1 -Jz- 1
X(z) = 1- lz-l
3
1
1 - z- - (1-"- ½z-1)(! - z-1) 3 < l•I < 1
68
(b)
3
Y(z}
4 < lzl
-1 i
Y(z) = l + lz-
3 + 3
1 1- 2z- 1
• l + z- 1 1
=
(1 + ¼z- 1 )(1- 2z- 1 ) 4 < 1•1 < 2
Y(z) (1- lz- 1)
X(z) = --=
H(z)
2
(1- 2z-1) 1•1 < 2
z[n] = -(2)nu[-n - 1] + !(2)n-1u[-n]
2
(d)
3.10. (a)
z[n] = { 1, -10 :S _n :S 10
0, otherwJSe
Finite length but has positive and negative powers at z in its X (z ). Therefore the ROC is O <
l•I < oo.
(c)
(d)
z[n) is right-sided, so its ROC extends outward from the outermost pole eJ•/ 3 _ But since it is
non-zero at n = -1, the ROC does not include oo. So the ROC is 1 < izl < oo.
(e)
z[n) is finite-length and has only positive powers of z in its X(z). So the ROC is izl < oo.
(f)
z[n) is two-sided, with two poles. Its ROC is the ring between the two poles: ½< lzl < j2)3; j, or
½ < lzl < *,-
3.11.
00
· which means this summation will include no positive powers of z. This means that the closed form of
= =
X(z) must converge at z oo, i.e., z oo must be in the ROC of X(z), or lim,~00 X(z) -I oo.
(a)
could be causal
70
(b)
(c)
(z ')•
lim -• =O could be causal
•-+oo(z - ½)•
(d)
(z - ')'
lim i = 00 could J>Ot be causal
•-+co (z - ½)•
3.12. (a)
1- lz-1
2
X 1 (z) -
- l +2z- 1
The pole is at -2, and the zero is at 1/2.
(b)
1 - lz-1
X2(z) = (l + ½z-')(l- jz-')
The poles are at -1/2 and 2/3, and the zero is at 1/3. Since :r2 [n] is causal, the ROC is extends
from the outermost pole: l•I > 2/3.
1/2
(c)
1 + z- 1 -2z- 2
X3(z) = 1- llz-1 + z-2
•
The poles are at 3/2 a.od 2/3, a.od the zeros are at 1 and -2. Since :rs[n] is absolutely summable,
the ROC must include the unit circle: 2/3 < izl < 3/2.
3/2
71
3.13.
g[ll] is simply the coefficient in front of ,-11 in this power series expansion of G(z):
3.14.
1
B(z) =
1 .
So,
1 1
A1 = 2; 02 =--;
2
3.15. Using long division, we get
1 -10
I - 1014z
B(z) = l - ~:-l
=•
= I;c!i••-•
n=O 2
5 5
Y(z) = l -½z-1 1- lz-1
3
-Jz-1 2
= (1 - ½z-1)(1 - }z-1)'
lzl >
3
Now
Y(z)
H(z) = X(z)
1 - 2z-l 2
lzl >
3
The pole-zero plot of H(z) is plotted below.
H(z)
2/3 2
h[n]
(c) Since
1
H(z) = Y(z) = 1 - 2z- ,
• X(z) 1- Jz- 1
we can write
Y(z)(l - ~z- 1) = X(z)(l - 2z- 1),
whose inverse z-transform leads to
2
y[n] - -y[n - I]
3
=:t[n] - 2:t[n - I]
(d) The system is stable because the ROC includes the unit circle. It is also causal since the impulse
response h[n] = 0 for n < 0.
3.17. We solve this problem by finding the system function H(z) of the system, and tben"loolcing at the
different impulse responses which can result &om our choice of the ROC.
Taking the z..transform of the difference eqnation, we get
and thus
Y(z) 1- z- 1
H(z)
= X(z) = 1- !z-1 + z- 2
73
lithe ROCis
(a) l•I < ½:
3.18. (a)
+ 2'-1 + ,-2
l
H(z) = (1 + ½z-')(1- z- 1 )
-,' !
= -2+ lT
'
.!z- 1
+ --"-~
1- z- 1
2
1
Y(z) = (1 + ½z- 1 )(1- ¼z- 1 )
The intersection of ROCs of H(z) and X(z) is izl > ½- So the ROC of Y(z) is lzl > ½-
(b) The ROC of Y(z) is exactly the intersection of ROCs of H(z) and X(z): ½< l•I < 2.
(c)
(b)
H(z)
..
= La"z-"= 1 - 1 lzl > lal
az-l
n=O
N-1 1-z-N
X(z) = I;,-n -
- 1- z- 1 lzl > o
n=O
Therefore,
1 r¼-r 1 l~• 1 ( 1 ) ( 1 Cl )
(1-az- 1 )(1-z- 1) = l-az- + l-z- = ~ 1-z- 1 -1-aZ- 1
!
So
J-a•+l
11[n] = ·-•
3.22. (a)
~
11[n] = L h[k]:r[n - k]
i=-00
(b)
Y(z) = H(z)X(z)
3 1
= 1 + l,-1 1 - z-1
3
• l !
= 1 + ½z-' + 1- •z-1
= ~ ( 1 + ~ ( - ~) ") u[n]
1
9 ( 1- ( - 1)•+ ) u[n]
= 4 3
3.23. (a)
2
1- lz-2
H(z) = (1- ½z-')(1- ¼z-1)
5+ lz-1
= -4+ 1-lz-l + lz-2
2
• •
7
= -4- 1- lz-1 + 1 _ l,-1
2 •
(b)
3 l l
y(n] - -11[n - l] + -11(n - 2]
4 8
= z(n] - -z(n - 2]
2
3.24. The plots of the sequences are shown below.
(a) Let
co
a(n] = L 6(n - 4k],
•=-00
Then
co
A(z) =L z-•n
l=-oo
(b)
b(n]
B(z)
l•I > l
. .
' .
... ...
.•
•
• "
,.
c
z-
o.sL • ••
• _,. . . . . •
l
• • •
i I
r·
n "
3.25.
z• z2
X(z) = -(z___a_)(_z b) = -z•-_-(_a_+_b_)z_+_ab_
___
1
z 2
- (a+ b)z + ab z2
z2 - (c + b)z + ab
(c+ b)z - ab
X(z)
z(n]
Therefore,
78
(d)
1
X(z) = 1-,z-
' • [z[ > (3)-l => causal
By long division:
1 + ½z- 3 + lz-• + ...
3
1- ½z- 1
1 - ½z-3
+ !z-3
3
+ ½z-3 - iz-6
+ lz-•
•
n = 0,3,6, ...
z(n]= {
otherwise
3.27. (a)
1 1
X(z) =
(I+ ½z- 1 ) 2 (1 - 2z- 1 )(I - 3z- 1) 2 < [z[ < 2
Therefore,
Therefore,
z(n] = o[n + 2] + U(n + l] - 2(2)"u[-n - l]
3.28. (a)
d
nz[n] ~ -z dz X(z)
X(z) = (1-3z-•
¼z-1)2 = 12z
-2 [ d ( 1
-z dz 1- ¼z-'
)]
(b)
X(z) °"
00 (-1)•
=sin(z) = {;;:, _,_..::.,.__z2l+l
(2k+ l)!
ROC includes \zl = l
Therefore,
Which is stable.
(c)
z 27 - l
X(z) = 1- z- 7 =z
7
- 1- z- 7 izl > l
00
X(z) = z7 - L z- n 7
n=O
Therefore, 00
3.29.
X(z) = e' +e11' z el 0
3.30.
l
X(z) = log2 (l2- - z) lzl < 2
(a.)
-l (I)'
-2 z_'
I
Therefore,
:i:[n] = ;; l (l)n
2 u[-n - l]
(bl
:i:[n] = l (l)n
;;
2 u[-n - l]
80
3.31. (a)
Poles: a, b, c 1
Zeros: z1, z2, ex> where z1 and z2 are roots of numerator quadratic.
pole-zero pl t (a) pole-zero plot (b)
~~er~
~
(b)
:z:(n] = n 2 cnu(n]
l
a:,(n] = anu(n].,; X1(z) = 1- az-1 izl > a
1
a: 2 (n] =na:,(n] = d
nanu(n] ..,; X2(z) = -z dz X, (z) = (l _az-
az-• )• lzl >"
2 n [ ] d ( d ( cz-• )
a:[n] = na:,(n] = n" un.,; -z dzX2 z) = -z dz (l -cz-1)2 lzl > a
-az- 1(1 + iu- 1)
X(z) = (1- az- 1)3 lzl >"
(c)
!
81
Poles at 0, -f 1 l ± j zeros at oo
1
zeros
poles
Y(z)
(b)
poles at ±½i
zeros at ±1
82
Y(z)
3.34.
3-7z- 1 +sz- 2 I 3
H(z) = -'------'c_:.:- =5+---1
1 - f z- + z-
1 2 1 - 2z-
(a)
n
y[n] = h[n] • z[n] = L h[l:]
n
-L 2• = -2n+I n<O
t-=-oo
{b)
1 1 1 1 1
Y(z) = 1- z-1 H(z) = -21 - z-1 + 21- 2z-l + 31 - !z-1 ,
2
2 < izl < 2
y[n] = -2u[n] - 2(2ru[-n - l] + 3 G) n u[n]
3.35.
H(z) = 1 - z•
---z
1- z• -
-1
- -
1- ,-•
c-,-•) lzl > 1
1 z
u[n] ~ =
1-z-l z-1 - - 1•1 > 1
z- 1 - z-4
U(z)H(z) = (1- z-<)(1 - z- 1 )
.-• .--
u[n] • h[n]
=
..
1- z- 1 -1-z-<
bO
83
3.36.
1
z[n] = u[n] ¢> X(z) = 1-z-1 l•I > 1
~
(b)
4z 4 1
H(z) = 1 - lz-l I - lz-1 lzl > 2
2 2
= 4o[n + 1] - 2 ( D n u[n]
=
bas poles at z ½and z 2. =
Since the unit circle is in the region of oonvergence X(z) and z[n] have both a causal and an anticausal
part. The causal part is "outside" the pole at ½- The anticausal part is "inside" the pole at 2, therefore,
z[O] is the sum of the two parts
1 lz 1 1
z(O]= lim +lim_!.::_=- +0=-
&-+OO t- 21 z-l z-Oz-2 3 3
3.38.
l•I > 1
84
Stable => ROC includes izl = l. Therefore, the ROC is ½ < izl < j.
(b) :r[-8] = E[residues of X(z)z-• inside C], where C is contour in ROC (say the unit circle).
:r[S] = E [residues of (z - 1
;Hz - 23
)
10 (z
z 3 )2 (
+ 2 z + 25 )(z +;7
) inside unit circle]
First order pole at z = ½is only one inside the unit circle. Therefore
(b)
,-,
H(z) r=,=-r -1
H1(z) = l +
H()=
z
,-,
l + 1 _,_ 1
=z
(c) H(z) is not stable due to its pole at z =l, but H,(z) and H2{z) are.
3.41. (a) Yes, h{n] is BIBO stable if its ROC includes the unit circle. Hence, tbe system is stable if rm,n < l
and Tm.cz > 1.
(b) Let's consider the system step by step.
=
(i) First, v[n] o-•x[n]. By t.aking the z-trausform of both sides, V{z) = X(oz).
=
(ii) Second, v[nJ is filtered to get w(n]. So W(z) H(z)V(z) H(z)X(oz). =
=
(iii) Finally, y(n] o"w(n]. In the z-trausform domain, Y(z) = W(z/o) = H(z/o)X(z).
In conclusion, the system is LTI, with system function G(z) = H(z/o) and g(n] = o"h[n].
(c) The ROC of G(z) is or,.,.< l•I < ar...,. We want r,.,. < 1/a and'=• > 1/a for the system
to he stable. -
3.42. (a) h(n] is the response of the system when z(n] = <l(n]. Hence,
10
h(n] + L a1h[n - .I:]= <l(n] + tl<l[n - l],
l<=l
(b) Atn=l,
h(l) + a,h(0) = o{l] + Po(0]
(c) How can we extend h(n) for n > 10 and still have it compatible with the difference equation for S?
Note that tbe difference equation can describe systems up to order 10. If we choose
.
h(n] = (0.9)" cos( n)u[n],
4
we only need a second order difference equation:
H(z)
,
,,
'
3.43. (a)
l l
X(z)=1-!,-1
2
1-2z-1• 2 < 1•1 < 2
6 6 3
Y(z) = 1- !,-1 -1- !,-1• l•I > 4
2 •
H(z)
86
~-olH(zt
u
...
l
~
t... 2
_,
.u _,
... -. ...
0 1 u 2
(b)
(c)
3
y[n] - I] = :i:[n] - 2:i:[n - 1]
4y(n -
(d) The system is stable because the ROC includes the unit circle. It is also causal since h[nj = 0 for
n < 0.
3.44. (a)
,.,,----...-------,
_,
(c)
-u,c_,-
1 0
Y(z)
05
--
---:,--!,---,,,....-,,--,,,.,--=',
1
H(z)
= X(z)
(1-z-l)(l+ ,.-i)(1-2z-i)
= (l-l2z-l)(1
l
2z-l)
87
=
ny[nj z[n]
dY(z)
-•~ = X(z)
(b) To apply the results of part (a), we Jet z[n] = u[n - l], and w[n] = y[n].
W(z) /
= - ,-1 I -,-•z-' dz
= - / z(z l- I) dz
= - 1-l
- + - -l d z
z z-1
= ln(z) - ln(z - I)
3.46. (a) Since y[n] is stable, its ROC contains the unit-circle. Hence, Y(z) converges for ½< l•I < 2.
• (b) Since the ROC is a ring on the z-plane, y[n] is a two-sided sequence.
(c) z[n] is stable, so its ROC contains the unit-circle. Also, it has a zero at ex, so the ROC includes
oo. ROC: lzl > ¾-
(d) Since the ROC of z[n] includes oo, X(z) contains no positive powers of z, and so z[n] = 0 for
n < 0. Therefore z[n] is causal.
(e)
z[0] = X(z)I,=~
¼z- 1 )
A(l -
= (I+ ¾z-1 )(1- ½z- 1 ) 1•=~
= 0
(f) H(z) has zeros at -.75 and 0, and poles at 2 and co. Its ROC is izl < 2.
88
ROC:ldc2
_,
-...
-u,,_. ,-.._.~-,._-.,--,
. -,s~-',
(g) Since the ROC of h[n] includes 0, H(z) contains no negative powers of z, which implies that
=
h[n] 0 for n > 0. Therefore h[n] is anti-causal.
3.47. (a)
00
X(z) = I:;z[n]z-n
n=()
00
Therefore, X(oo) =
z[O] # 0 and finite by assumption. Thus, X(z) has neither a pole nor a :zero
=
at z oo.
(b) Suppose X(z) has finite numbers of poles and zeros in the finite z-plane. Then the most general
form for X(z) is
,,, ·
00 nc,-c.)
X(z) = I:;z(n]z-n = KzL.=•=;.:..'---
n=O IT (z - d•)
Kzl
where K is a constant and M and N are finite positive integen and Lis a finite positive or negative
integer representing the net number of poles (L < 0) or zeros (L > 0) at z =
O. Clearly, since
=
X{oo) z[O] # 0 and < oo we must have L + M = N; i.e., the total number of :zeros in the finite
z-plane must equal the total number of poles in the finite z-plane.
3.48.
X( z )
= P(z)
Q(z)
where P(z) and Q(z) are polynomials in z. Sequence is absolutely summable=> ROC contains jzj = 1
and roots of Q(z) inside l•I 1. =
These conditions do not necessarily imply that z[n] is causal. A shift of a causal sequence would only
=
add more zeros at z 0 to P(z ). For example, consider
.. 1
X(z) = --,
z-2
l•l>-2
z 1
= --1
z- i
=z• l
1- 2z-l
Therefore,
Z[:i:"ln]] = =~
00
X"[n],-n = (
n~
00
:i:[n](z"}-n) " = X"(z"}
00 00
(b}
00 00
(c}
3.52.
00
+
2
:i:(2] ( tan 8. ( ; ) cos•;+ sin 4;)] = 0
l
l + :i:[O] (:i:[l] · l + :i:(2] · -1) = 0
:i:(2]
= 2:r[O]
= 3:i:(O]
Therefore
:i:[n] = :r(0](6(n] + U[n - l] + 3o[n - 2]}
where :r(O] is undetermined.
a.ss. (a)
m m
Cs,(n] = L :r:[A:J:r:(n + A:] = L
.__m
:r:(-i;]:r:(n - A:] =:r:[-n] • :r:[n]
X(z) has ROC: r1t < lzl < r, and therefore X(z- 1) has ROC: ri; 1 < lzl < rji 1 • Therefore C,,(z)
has ROC: maz{ri; 1 ,r1t) < lzl < miD[rji 1,r,)
(b) :r:[n] = a"u[n) is st&ble if lol < 1. In this case
1 1
X(z)
1- oz-1
lol < lzl and X(z- 1 ) = -1-az-
Therefore
1 1
c•• (z) =
~
= 1 - az- 1 - l - o- 1 z- 1
This implies th&t
c.,[n] = :
42
(a"u[n] + ,.-"u(-n - 11)
1
Thus, in summ&ry, the poles &re &ta and ,.- 1 ; the zeros &re &t O and 00; and tbe ROC of c•• (z)
is lal < lzl < 1a- J.
1
(c) Cleuly, :i:1[n) = :i:[-n) will have the same autocorrelation functi011. For ex.ample,
l
X,(z)=-- lzl < ,..-,, ~ c.,•• (z) = l l l l -I
-a.z
=c•• (z)
1-oz -GZ
(d) Also, any delayed wnioll of :i:[nj will have the same autoconela.tion functi011; e.g., :t2(n) = :[n- m)
implies
z-• z•
lol < l•I ~ c...,(z) = l -az
-I
1-oz =c•• (z)
92
S.56. In order to be a z-transform, X(z) must be analytic in some aDDular region afthe z-plane. To determine
=
if X(z) z• is ~alytic we examine the existence oI X'(z) by the Canchy Riemann conditions. If
In our cue,
and thus,
8u a,,
B:r =l 'F 8,, = -1
unless"' and I/ are zero. Thus, X'(z) exists only at z = 0. X(z) is not aulytic anywhere. Therefore,
S.57. If X(z) has a pole at z = Zo then A(z) can be expressed as a Taylor's series about z = zo.
00
B(Zo)
= A'(Zo)
!
93
Solutions - Chapter 4
z[n] = z,(nT)
= sin ( 21r(l00)11 :io)
= sin (in)
z,(t) = ms(o.,tj.
Since w = OT and T = 1/1000 seconds, the signal &equeiicy could be:
.
0., = - - 1000 =
4
25(),r
or possibly:
,r
I n. = (21r + 4). 1000 = 2250,r.
4.3. (a) Since :r[n] = :r,(nT),
.-n
3 = 40()/)r,nT
l
T = 12000
(b) No. For example, since
T can be 7/12000.
4.4. (a) Letting T = 1/100 gives
:r{n] = :z:,(nT)
= sin (20.-n ~) + cos ( 40.-n 1~)
1
= sin('";) +cns(
2
;')
• •(I)
(a)
z.(t) = 0,
, 1n12: 2.- · 5000
The Nyquist rate is 2 times the highest frequency. ,. T = ,o.:..O sec. This avoids all aliasing in
the C/D convener.
(b)
1
= lOkHz
T..,
.
8
= TO
1
= w,ooii{l<
n. = 2.- · 625rad/sec
le = 625Hz
(c)
1
T..,= 20kHz
= TO
,r 1
8
= 20,000°'
n. = 2" • 1250rad/sec
le = 1250Hz
4.6. (a) The Fourier transform of the filter impulse response
• H.(jn) = L 11.(t)e-;n, dt
= f."° a-a.ie-;nt dt
1
= a+ ;n
So, we take the magnitude
llfcUOll
l/a
0
97
(b) Sampling the filter impulse response in (a), the cliscrete-time &I.er la described by
h4(n] = T•-•"T u[n]
H4(eiw) =
..
L Te-•"T.-jwn
....,
Taking the magnitude of this response
. T
IH4(e'w)I = (1- 2e-aT cos(w) + e-2oT)i.
Note that the frequency response of the discrete-time filter is periodic, with period 2.-.
'%(• jCD)I
(c) The minimum occurs at w = .-. The corresponding value of the frequency response magnitude is
T
(1 + 2e-•T + e-••T) ½
T
=
'%(e jO>)I
T -----------------------
1/2
1/a 2/a 3/a
4. 7. The continuous-time signal contains an attenuated replica of the original signal with a delay of,,.•.
Zc(t) =Sc(t) + asc(t - 'rd)
(a) Talling the Fourier transform of the analog signal:
Xc(;O) =Sc(;O) · (1 + ae-;,•n)
Note that Xc(j!l) is zero for 101 > .-JT. Sampling the continuous-time signal yields the discrel<
time sequence, z(n]. The Fourier transform of the sequence is
1 ~ ;w + 1-y)
X(~) = T
..
L, Sc(T
,,._
.hr
98
== 1.1•
2,r -•
(l + ...-;!'.f').,;- d,.i
-2xx10 4
(a) For :,(1) to be recoverable from :{n], the transform of the disaete signal must ha.,., no aliasing.
When sampling, the radian frequency is rewed to the analog frequency by
w = OT.
No aliasing will occur if the sampling iDtenal sat.is&es the Nyquist Criterion. Tbus, for the band-
limited signal, :,(I), we should select T aa: ·
l
T:S2x1o<·
!
99
(b) Assuming that the system is linear and time-invariant, tbe convolution 111111 describes the input-
output relationship.
11[n) = L
.. :r[k]hjn - k)
We are given
---..
.
11(n) = T 'E :,;(l:)
..
•=-ao
T
..
= ...L_ :r[l:]u(n - l:)
= T·
n-ao
L
.. £...
t-=-ao
:,;[k]
ts-oo
X(~w) =
..
L :,;[n]e-jwn
n.=-co
Hence,
L..
---co
;2,rr
X,(T) = [
-co
z.(t) cit
= x.(;ll)ln..o
For the &nal equality to be true, there mast be 110 contribution &om tbe t.erms for which r f, O.
That is, ft require IIO aliasing at (l =
0. Since ft are only imereRed in preserving the spectral
component at ll = 0, f t may sample at a rate which is lower than the Nyquist rate. Tbe maximum
-.alue ol T to satisfy ~ Clllllditiom is
l
T<-.
- l x 10"
100
4.9. (a) Since X(.,i") = X(.,i<w-•>), X(ei-'} is periodic with period ,r.
(b) Using the in'nl?le DTFT,
z(n] = 2.-
1
..!._
(h)
X(.-i"')ei-"'dw
= 2...1
2.- (h)
X(.-i<w-•IJ.-i""'dw
= 2...1
20' (h)
X(.-i"'}.,i(w+s)ndw
= ..!...,;•n
2,r (h)
1
X(.-i"').-iwndw
= (-1rz[n].
All odd samples of z[n] =0, because ~[n] =-z[n]. Heoce z[3] =0.
(c) Yes, y[n] cootains all even samples of z[n], aod all odd samples of z[n] are 0.
There are other choices. For example, by realizing that sin(O"n/4) = sin(9'm/4), we find T. = 9/40.
(b) Choose T = l/20 to mab, :z:[n] = zc(nT). This is Wlique.
4.13. (a)
:i:,(t) = sin(;t)
=
20
. .
sin(-t - -)
20 4
.-n ..
11[nJ = sin(- - -)
2 4
(b) We get the same result as before:
:i:,(t) = sin(;0 1J
11,(t) = sin( :0 (t - 2.5))
= sin(-t
10
. .
- -)
4
. .-n 1(
11[n] = sm(- - -)
2 4
(c) The samplillg period Tis not limited by the continuous time system h,(t).
4.14. There is no loss of information if X(,.;-,1 2 ) and X(,i!w/Z-•l) do not overlap. Tbis is true for (b), (d),
(e).
4.15. The output :i:,[n] = :i:[n] if nn aliasing -:,ccun as result of dowmampling. That is, X(e'w) = 0 for
,r/3 ~ ,..,, ~ ...
(a) r[n] = cos(.-n/4). X(eiw) has impulses 81 w = ±.-/4, so there is DO aliasi.Dg. r,[n] = r[n].
(b) r[n] = cos(.-n/2). X(eiw} has impulses 81"' = ±r/2, so there is aliasi.Dg. r,[n] # r[n].
(c) A sketch of X(eiw) is shown below. Clearly there will be no aliasi.Dg and r,[n] = r[n].
X(ej<il )
-1t/4 !t/4
•
,r;"• ~ . , r·;~· .
-lt/3 lt/3 • 0> --""'S-:f-al-:-6::---'--s""JCl'-:6-::--0>
Tbis is UDique.
102
(b) One choice is
M 1t/2 2
r:;=3.,;4=3
However, this is DOt llllique. We can also write i 4 [n] = eos('fn), 10 another choice is
which leads to
:z:[n] = ~ sin(ffR/2)
3 .-n
(b) Upsampling by 3 and low-pass filtering :z:[n] = sin(3.-n/4) results ill sill(.-n/4). Downsampling by
= =
5 gives us i4[n] sin(51tR/4) -sin(3=/4).
4.18. For the condition to be satisfied, we have to ensure that Wo/L S mm(1t/L,1t/M), so that the lowpass
filtering does 110t cut out part of the spectrum.
(b) A straight-forward application of the Nyquist criteri011 would lead to an incorrect conclusion that
the sampling rate is at least twice the mazimum frequency of :z:,(I), or 202 • However, since the
spectrum is bandpass, we only need to ensure that the replicatiOIIS ill frequency which occur as a
result of sampling do not overlap with the origillal. (See the following figure of X,(;O).) I'herdoR,
we only need to ensure
2,r
n2 - -T < n, T < -.0.0 = 2ff
i03
(c) The block diagram along with the frequency response of h(t) is shOWD here:
4.22. (a)
• lD 'h
w == !lT,
i ~
-It
~
(b) To recover simply filter out the undesired parts of X(e'w).
I,
It CD
(c)
. -2xrr
I -ltff
I rI 1fl
I
21tff -zi
T<~
- !lo
4.23. In tbe frequency domain, we have
104
Zc(I) = 0, 101 ~ ;,
Therefore, since we ue sampling this s.(t) at the Nyqum frequency s(n) will be full band and uu•li•eed
s(n) = z 0 (nT1)
1/c(I) is a band-limned iuterpolatiou of z[n] at a di!'erent period. Since no aliasing occun at z[n), the
spectrum of 110 (1) will be a frequency axissc.aling of the spectrum of Zc(I) for T, > T2 or T, < T2. As
we show iD the ficu,e,
(a)
Q)
-lt X 5 X 1<>3
(b)
I
Q)
-2 lt X 5 X 1<>3 21< X 5 X 1<>3 {l
(c)
II XS X l<f {l
(d)
-1t Q)
-2 ,. • s. 103 211 • s. ,o3 °
4.25. (a) z,(I) = :1:c(l)s(t) ~ X,UO) • •Ull)
0
106
(b) Since H,(e'"') is an ideal lowpass filter with"'• = i• we don't care about any signal aliasing that
occurs in the regi011 f !', .., !', 11. We require:
211 1l
- - 211 · 10000 ,!:
T 4T
1 8
,!: - -10000
"f 7
T 7 X 10-•MC
!', ·-
8
Abo, 011ce all of the signal lies in the range lwl !', f, the filter will he inelfective, i.e., f :$ T(211xl04).
So, T :!: 12.5,-:.
(c)
8nx10 4 1rr
4.26. First we show that X,(ei"') is just a sum of shifted versio11S of X(e'"'):
r
:r,[n]
0, otherwise
= (! ei(2de/M)) :r[n)
X,(e'"') =
00 1 Jl-1
= L M L :r[n]eil...,./M)•-;.,.
-=-oo '-0
= ! L .--oo
M'-1
L
l:=-0
oo
:r{n)e-,l.,-(2d/Mll•
1 Jl-1
= M ~ X (ei1.,-(2d/Jtll)
X,(e"") = ___
....
Additionally, X4 (e'"') is simply X,(e"") with the &equency am expanded by a factor of M:
L X,(Mn)•-;,.,•
=
..
L :r,(l]e-J(w/Jl)l
,__
107
(a) (i) X.{ei'-) u,d X,(ei'-) are sketched below for M = 3, WR= r/2.
Xs(•jw }
1/3
'' ,, ''
.,
,, ''
K (I)
,, '
' ,, ''
<
,, ''
-21t --1t 1t 21t (I)
{ii) X.(.,;"') and X,(ei'-) are sketched below for M =3, WH = "/4.
X 5 (eiW )
1/3
It (I)
(I)
(b) From the definition of X,(ei"'), we see that there will be no aliasing if the signal is bandlimited to
1r / M. In this problem, M = 3. Thus the maximum value of "'H is 1r /3.
When we upsample, the added samples are zeros, so the apsampled signal z.[n] has the same energy as
the original z(n]:
..L Jz(n]l 2
..
=L 1:r.(n]l2 ,
-=-oo --00
and by Paneval's theorem:
4.28. (a) Yes, the l}'flelll is liDear because ea.ch of the subblocks is liDear. The C/D step is defined by
z[n] = z.(nT), which is clearly linear. The DT system is an LTI l)'SUm. The D/C atep consistS
of converting the aequeace to impttlsa and of CT LTI filtering, both of which are linear.
(b) No, the system is not time-iD...-iaat.
For exu>ple, suppooe that h[n] = 6{n], T = 5 and z,(t) = 1 for -1 !, t ~ 1. Such a sysUm would
=
rault in z[n] 6[n] and 11c(t) = linc(r/5). Now suppose we delay the input to be z,(t - 2). Now
z[n] = 0 and 11,(I) = 0.
4.29. We can analyze the system in the frequency domain:
Since only half the frequency band of X,(jO) is needed, we can alias everything past O = 2000,r. Hence,
T =1/3000 s.
Now that T is set, figure out H (ei") band edges.
IH(e jm )I
arg(H(e jO> ))
(I)
-:ZX 1t 211:
-lt
I:..,_, . ~· ..
... ... ...__... ..._..,.,.__1,
0
.
••
.
,.
I:
i ..,
(b)
_..,_,
... ... .... __ • .
... -..cy,..,.__,,~· .. ...
(c) To filter the 60Hz out,
1 31t
"'°=T!l=---2.--60=-
10,000 250
4.33.
11(n] = r(n)
Y(ei"'J = X(~) • X(ei"'J
therefore, Y(~) will occupy twice the frequency bud thai X(~) does if rn aliasing occurs.
U Y(e'"') 'I 0, -1t < w < 1t, then X(~) 'I 0, -j < w < j and so X(jSl) = 0, (!ll ~ 21t(IOOO).
Sincew=,OT,
11'
2 ~ T · 21t(lOOO)
1
T :5
4000
110
4.S4. (a) Since there is no aliasing involved in this process,"" may choose T to be any value. Choose T = l
for simplicitf· X,UO) = 0,101 2: 1t/T. Since Y,(jO) = H,(jO)X,(jO), Y,(jO) = 0,101 2: 1t/T.
Therefore, there trill be no aliasing problems in going &om 11,(t) to 11[n].
Recall the relationship"'= OT. We can simply use this in our system oonversion:
H(el"') =
__,.,,.
H(jO) = .-;art•
= ,,-;n;2, T=l
5
cos ( ; n - i) = ½[ei<'fn-t) + .-j('fn-t>]
= !.-l(•/4)e-f(../2)n + !e-f(•/4) 0-;(.. /2)n
2 2
Since H(ei-') is an LTI syste,n, ""can find the respoDSe to each oftbe two eigenfunctions separately.
Since H(e'"') is defined for O 5 lwl 5 " we must evaluate the frequency at the baseband, i.e.,
=
5,r /2 => 51' /2 - 2,r ,r /2. Therefore,
= cos (5ft2 n - ~)
2 ·
.
y[n)
0 n
.(
1
H(ei"') = (10jw) 2 + 4{10jw) +3
1
= -loo...'+ 3 + 40jw
!
111
(b) The downsampler bas M = 2. Since 2:[n) is bandlimited to Ti, there will be 110 aliasing. The
frequency axis simply expands by a fador of 2.
*
For 11.(1) = 2:.(1) Y.(;O) = Xc(;O).
Therefore OT' ~ 2,r · lOOT' ~ T' = •·
4.37. In both systems, the speech was fihered first 10 that the subsequent sampling results ill 110 aliasillg.
Therefore, going •In) to ,i[n) basically requitt< ch•ngi11& the sampling rate by a factor of 3kHz/5kllz =
3/5. This is clone with the followiDg syatem:
1----,.,""'. cutoff
Igain,.3
LPFI
Digital=1tl3 I
1 - -- • . . ,
.
~S
•
::tc{I) is sampled at sampling period T, so there is 1>n aliasing ill ::t[n).
.A -Jt
A'A. rr
•
Inserting L - l zeros between samples compresses the frequency axis.
(a)
The
~- -w/L wJL
filter H(ei"') removes frequency components between ,r / L and ,r.
. IO
. I\ -s/ L
&T /\
-~-;-)
wJL ct
The multiplie&ti011 by ( -1 )• shifts the center of the frequency band from 0 to ., .
• _,. " /. IO
112
The D/C conversion maps the range-.- to.- to the range -.-JT to .-JT.
4.S9. (a)
..... 0
If n = mL (m an integer), then w,, don't bave any multiplications since h[O] = 1 and the other
non-zero samples of v[A:] hit at tbe zero,; h[n]. Otbenrise the impulse response spans 2RL - 1
samples of v{n], but OA!y 2R of tb- are DOD-zero. Tberefott, there a... 2R multiplies.
4.40. Split H(eJw) into a lowpasg and a delay.
x, (I)
CID
x[n)
I
tL - Hu,<i"' )
I
e
-j., v[n)
♦ L ..!...i
I)
of I instead of L
wvfNv
-Slt/4 -311:14 -11:14 11:/4
YJei"'J
311:/4 511:/4 o,
4/T
(I)
4.42. (a) The Nyquist criterion states that :r0 (t) can be ncooaed,.. long as
2.- 1
T ~ 2 x 2.-(250) ~ T :S 500·
In this case, T = 1/500, so the Nyquist crneriOD is satisfied, and :r0 (t) can be recovered.
(b) Yes. A delay ill t.ime does llOt c:bange the bandwidth ol the signal. Hence, 11.(t) has the sam
bandwidth and same Nyqwst sampling rate as :r0 (t).
(c) Consider first the follcrwing a::preaiolll far X(ei"') and Y(ei"'):
Hence, we let
B(ei") = { 2eo, -;.. • lwl < j
otherwise
Then, iD the following figure,
""' < j
otherwise
x[n] y[n)
H
For the given T = 1/800, there is no aliasing &om the C/D conversion. Hence, the equivalent CT
transfer function H.(jO) can be written as
,
' ''
, ,, ' ' ,,
,
'
,, '
'I
I I
I ,.
, '
''
21t
115
2...
-
T - 800,t >
- 400,r = -T >-
2..-
1200,t
X(c ID)
II)
(b) For this to be true, H(ei"') needs to filter out X(ei"') for ..-/3 :S lwl :Sr. Hence let wo = r/3.
Furthermore, we want
•/2
T =2..-(1000) =- T2 = 1/6000
2
(c) Matching the following figure of S(ei"') with the figure for R.,(;O), and remembering that O = w/T,
we get T3 = =
(2r/3)/(2000r) 1/3000.
II)
A= £ 10
#ic(t)dt = J: :r.(t)dt = X,(;O)Jo-o.
To estimak X,U · O) by DT pro •sing. "" need to sample only fast enollgh so that X.(; · 0) is not
aliued. Hence, f t pick
116
Further,
X(ei"') =
..l:: z(n].
--oo
=
Therefore, - pick h(n] Tu(n], whicb males the system an accumul&tor. Our es\illlate A is the output
11(n] at n = 10/(10--<) =
lo", when all rJ the noa-zero Ample& of z(n] haft heeD added-up. This is
an C4Ct estimate giftD out aasumptiaa al both band- and time-limitedness. Since the aasumptiou c:an
nner he exactly satisfied, however, this method only giYeS an approximale estimate for actual signals.
The overall system is as follows:
T
f
= 1/10000
4.46. (a) Notice that
11o[n] = z[3nJ
111 [n] = z(3n + 1]
112(n] = z[3n + 2],
and therefore,
11o[n/3], n = 3k
z[n] ={ w,[(n -1)/3], n = 3k + 1
112[(n - 2)/3], n = 3k + 2
(b) Yes. Since the bandwidth of the tilters are 2,r /3, there is no aliasing introduced by down.sampling.
H;nce to reconstruct z[n], - need the system shown in the following figure:
YnlnJ., f3 •I E 0 z>I l
YJ [n].,, t 3 I
1----'• E 1(2)~~1
In the following discussion, let "'• [n] denote the even samples of :r[n], and :r.(n] denote the odd
samples ot :i:(n]:
= { 0,:i:(n], ne-.en
n odd
n even
= { ~in], nodd
{ 11,(n/2], n even
11,(n] = 0, n odd
{ w 4 (n), n even
= 0, n odd
{ (:r • h.)[n], n even
= 0, nodd
= :i:0 [n] • "4[n]
where the last equali~ follows from the fact that h.[n] is non-zero only in the odd samples.
Now, s[n] = 114 {n]•"4(n] = :i:0 (n]•h.(n]•"4(n] = .rrlnl. and since :r[n] = :r,[n]+:r0 [n], •ln]+113{n] =
:r[n].
4.47. Sampling random processes
(a)
1 ~
P..(w)=T L.J P•••• ("'
T+T2rk)
K=-oo
(c) If
P•••• =o, for lwl 2: r
then
118
-t.48. {a)
2,rr)
Therefore, we require that f
P..(w)=T1 LP,,.,
~ Clc,.
... ("'
T+T
--
(c) For the spectrum of F"ig P3.&-2 it is dear \hat if T = ii; then the discrete-time power spectrum
will be wbi1e, as shown ill the figure above.
pxx ( m)
I I I
I
I \
'' I
I
'' I '' I '' I ''
I
I
>
I
'
I
V
\
I
'<
I
I
\
'<
I
I
''
I I \
I
'' I
'' I I '' '
I I
I
I I '
'
I
' ''
-4,c -2,c 2,c It (I)
(d) For white ~ t i m e signal:> 4>,,[m] = 0, m-# 0 but 4>,,[m] 4>,,,.(mT). Therefore, any =
analog signal whose autocorrelation func:t.ion bas zeros equally spaced at intervals of T will yield a
white discrete-time sequence is sampled with sampling period T. For example, for Fig P3.&-l:
sill Clc,T Clc,mT
4>...,(.-) =-y;- * 4>.,(m) = sillrmT
sill,rm
ifT= ~ 4>.,[m] = rm/n. = o, m-#0
'b
Sysran I: Sy11e1112:
-
- ~.
/K.
;:vK%\.
•
-
-&fm
m) m "b •
119
=
l,11(t) 112(!): Convolution is a linear process. Aliasing is a linear process. Periodic convolution is
equivalent to convolution followed by aliasing.
¥1(t) 'F r(t): System 2 at Siep l shows X;UO). This is clearly not Y1UO). Y,(jO) is an aliased
version of X,(jO)
(b) Now,
(c)
:(t) = A cos(30,rt)
3 l
4Acos(30.-t) + 4Acos(3. 30,-t),
:3(t) =
Ii H
-
(ei") = sin(wL/2) -i!L-t,._/2
sin(w/2) e
(b) The impu!,e response h,;,. (n] com!SpODds to tbe eo11volution of two rectaDgular sequences, as shown
below.
• -L
A' L •n
= 1/L.
• _j,d
2
II
J,d
2
1
*
•n •
- J,d
2
II
J,d
2
I
•n
120
2
H (ei") _ .! (siD(wL/2))
Im - L sill(w/2)
(c) The frequency response of-...orcier-hold is llatter ill the region [-.,/L,.,/L], but achieves less
out.<lf-balld attenuation.
4.51.
Tb• bandwidth of +,.(ei"') i.s no larger than the bandwidth of X(ei"). Therefore, the outputs of the
sys\ems will be the same if H2(ei") is an ideal lowpau 6her with a cutoff offr/L.
4.52. The idea here i.s to aploit the fact that every other sample supplied to h(n] ill Fig 3.27-1 is zero. That
is,
w 1 (n]
= { hi(n/2) • ,:(n/2], n ewn
0, n odd
= { h [0),:(n/2] + h1(1],:{(n/2) -1) + h (2J,:((n/2) -
1 1 2), newn
0, n odd
The down.sampler expands the frequency axis. Since Ro(e'"') is bandlimited to i;, no aliasing
occurs.
(c)
Yo(.,;,,) = ½Ho(.,;,,) (X(.,;w)Bo(.,;w) + X(eA-tµ,.{.,;r,-+"'l})
Yi(.,;,,)
-
½s,c.,;wi (xc.,;,,JH,<.,;-,1 + X(~)Jr.~l)
Y(.,;w) = Yo(.,;,,) - Y,(.,;w)
= ½xc.,;-,1 [S:<.,;,,1-a:c.,;-,11
+ ½X (ei<-l) [s.,(.,;w )Ho(ei<__,, _ .....)Bi{~)]
..,
n,,. 1li1sing terms always C111cei. Y(ei-') is proportkmal•X(«>");fi14<e""'}-JP.(.,-..X is a
amstaDt.
=
xc.,;w) o, ... /3 :S lwl :S .... :r:[n] cu be tlioagbt <i M aa Ci ,ilal...,L De -.,,nwh is to
delermine whether no is odd or ewn, ti.m sample ., u,ai no is - IA, • • : • • ilur; alter.
Thii reco,ers a.[no].
4.54. (a) In the cue where no is not lmcwn, we "-mine whether ii • - s oliii a ~
s.e.-en
u,(n] = { :r::r1 [n/2],
2 ((n - 1)/2], ir.....W
:r, [n] = u,(2n]
:r,(n] = w[2n + 1]
~2 tp]
w[n]
~n+I]
The system is linear, time-..a,ying (due to clcwnAmpling), ..w-- rf (._ to I(&+ ID, ad llla!lle.
(b)
!
123
_',,/&/. .vfv.
-lt II (I) -a: 11 (I)
--KIL
~ 11 IL
'"' "
i~ -1t1L 11 IL
O
"'
AhrLPF
•' Ar ' , tk
-11 -KIL II IL II (I) -r. -xlL "IL
LB/2T
"
I •
(I)
After cosine
modulalion
.. /\i'.A .. .. Ll 1Ll ..
-II -(I) I "' I a: "' -It ~ "'2
LB/2T
1t "'
AhrHPF
.,;J
-a --(1)1
i:h.'
"' I 11 "'
,./l-It ~
1 [\,_
"'2 1t "'
I l:ur.
I fv1
(d) To geaeralize for M clwmels, - would ...., the same modalalors, but - would choooe a larger
value of L t.o make room for additional spectra above the lower frequency boUDd. If the • -
124
=
bound remained 2,r · 10", L would become L 20 + M for M channels.
A branch al the TDM demultiplexing system would be:
w[n]
li[n+k]
4.56. Since we want W(ei-') kl eqaal X(eiw), Ulen H(ei-') mun compens,ue for die drop offs ill H .. Ufl).
4.57. (a)
E(e[m])E(e[n]l, m ¥- n
r[m, n) = E(e[m)e[n]) = { E(e'[n]l, m =n
t;.2
r[n,m) = r[n - m] = 12 o[n - m)
(b)
e,[n] = L h[k)e[n - k)
•
The variance of :[n] is .,..;ghted similarly so die SNR does not change. SNRout =12~.•
125
I m
·I n
w I [n)
•
I ·I
x, [n) • I
L= 11 Cllc= 11/ll
I M= 10
L=ll Cllc=ll/11
4.61. (a)
V(z) = H 1 (z)(X(z)- Y(z))
U(z) = H2(z)(V(z) - Y(z))
Y(z) = U(z) + E(z)
H 1 (z)H2(z) X l E( )
= 1 + H2(z)(l + H,(z)) (z) + 1 + H2(z)(l + H,(z)) z
Substituting H,(z) =1/(1 - .. - 1 ) and H2 (z) =z- /(1 -
1 z- 1 ), we find
H.,(z) = z-•
H.,(z) = (1- z- 1 ) 2
P11(ei") = .,:1s,,(ei")I'
= ..:1<1 - .-;..,i•,·
= ..:(1 - .-;")'(l - e'")'
= ~(2 - 2cos(w)) 2
= o-!(4sin2 (w/2)) 2
= 16o-! sin4 (w/2)
The total noise power cry is the autocorrelation of /[n] evaluated at 0:
ee
K CO
127
u: = 2i
1 1·'.,,,,"' .,:c2uw/2)'""
"' 2 1·'"'
1 r,(?.w/2)'""
,r s/11
~"'" I·'"'
= 2.- 5 __ ,,,,
= SM• -·-
~11•
I
4.62. (a) (i) The transfer function from :i:[n] to y.(n] is
H,,(z) = l _, = 1-,-1
1 + 1!,-1
So
P,. (w) = P,(w)H,,(_,;..)H.,•-;w
= ~(l - •-;.,)(l - _,;..)
= ~(2 - 2cos(w))
(b) (i) :i:[n] contributes ODly to 1/1 [n], but not Wz[n]. Therefore
111s[n] =
:i:[n - 1]
r,(n] = :i:[n -2]
128
(ii) In pan(a), the dilference equation desaibing the sigma-delta noise-shaper is
!
129
Solutions - Chapter 5
Transform Analysis of Linear Time-Invariant Systems
131
5.1.
- { 1, 0 $ n $ 10,
II [n 1- 0, otherwise
Therefore,
This Y(el-') is full band. Therefore, since Y(~") = X(~")H(~"), the Oll!y possible z}n) and..,,. that
could produce 11(n] is z[n] = 11[n] and"'• = 11.
5.2. We haft 11[n - l] - Jf11[n] + 11[n + 1] =z[n] or ,- 1Y(z) - ¥Y(z) + zY(z) =X(z). So,
1
H(z) =
z- 1 - !p + z
z
= (z - !Hz - 3)
_! !
= _.L+....L.
z-! z-3
(a)
Im
1 zeroatz=-
113 Re 3
• (b)
-tz-1 !z-1
H(z) = _l__..,l_z-_• + _l.._~...,3-,-...,,
3
5.3.
1
11[n - l] + 11[n - 2] = z[n]
3
,-
1
Y(z) + ~z- 2Y(z) = X(z)
Y(z) 1
H(z) = X(z) =
z-1+ 1.-•
z
H(z) =
l + 1,-1
132
5.4. (a)
(b}
l 2,- 1 3
H(z) = 1- 1,-1 1- !,-1' 1•1 > 4
• •
h(n] = (43)n u(n) - 2 (3)n-l
4 u(n - 1)
(c)
1 1 I
Y ( z ) = - ~ - + - ~ - + -1 lzl>l
I- tz-•
1 - ¼z-1 I - z- '
:r(n) = u[n]
1 .
X(z) = I - z-1' lzl > I
H(z) - Y(z) - 3 - 11-,-1 + J,-2 lzl > 3
I
- X(z) - 1- f.z- 1 + f,;z- 2 '
(a) Cross multiplying and equating ,-1 with a delay in ti?>•=
7 I 19 2
11[n - I]+ 11[n - 2] = 3:r[n] -
11(n] -
12 12 6 :r[n - I]+ 3:r[n - 2]
(b) Using partial fractions on H(z) we get:
I ,- 1 I ,- 1 I
H(z) = 1 _ 1,-1 - I - 1,-1 + I_ 1,-1 - 1 _ 1,-1 + I, l•I > 3
3 3 4 4
So,
= (3I)" u[nJ - (l)n-l (I)" (1)•- 1
h[n] u[n - I]+ u[n] - u[n - I]+ 6[nJ
3 4 4
(c) Since the ROC of H(z) includes Jzl = I the system is stable.
5.6. (a)
l I
2 < fzl < 2
(b)
1 -.,.,..,...--2- - - , -
Y(z) - -,--..---
. - (I - ½z- 1)(I - 2z-1 )
This has the same poles as the input, then!l'ore the ROC is still ½< lzl < 2.
(c)
Y(z) _2
H(z) = X(z) = 1 - z .,. h(n] = 6[n] - cf(n - 2]
134
5. 7. (a)
5
:z(n] = 5u(n] * X(z) = _ z-l, lzl > l
1
112 Re
(b)
1-z-1 -1 ! 3
H( z) = -,--,-----e---,------.--,,.. = _ _,.._,.. + • l•I > 4
(1- ½z- 1 )(1 + ¾z- 1 ) (l - }z- 1 ) (1 + }•-•)'
h[n] = - 2(l)"
5 2 u(n] + 5 - 4 u(n]
7( 3)"
(c)
Im
1 zeroalz•-
-1/2 Re 2
!
135
(b)
izl > 2
Y(z)
H(z) = X(z)
z-•
= (l -2z- 1 )(1- ½z-')
a
= 1-2,-1
Im
1 zeroatzs-
112 Re 2
2
h[n] = --(2)"u[-n - l] - - -
3 3 2
2(l)" u[n]
Includes l•I = l, so this is st.able.
136
5.11. (a) It connot be detet-rn.iMd. The ROC might or might not include the wut circle.
(b) It connot be determined. The ROC might or might not include z = 00.
(c) Foue. Given that the system is causal, we know that the ROC must be outside the outermost pole.
Since the outermost pole is outside the unit circle, th~ ROC will not include the wut circle, and
thus the system is DOI stable.
(d) True. H the system is stable, the ROC must include the wut circle. Because there are poles both
inside and outside the unit circle, any ROC including the wut circle must be a ring. A ring-shaped
ROC means that we have a two-sided system.
5.12. (a) Yes. The poles z = :!:j(0.9) are inside the wut circle so the system is stable.
(b) First, factor H(z) into two parts. The first should be minimum phase and therefore have all its
poles and zeros inside the wut circle. The second part should contain the remaining poles and
zeros.
l +0.2z- 1 1- 9,- 2
H(z) = 1 + 0.81.- 2 I
minimum phase poles & zeros
outside llllil circle
Allpass systems have poles and zeros that occur in conjugate reciprocal pairs. H we include the
factor (l - ,,- 2 ) in both parts of the equation above the first part will remain minimum phase
and the second will become allpass.
H(z)
= (l + 0.2z- 1 )(1 - tz- 2 ) . 1- 9z- 2
I+ O.Slz- 2 1- l•-•
= H1(z)H,.(z)
5.13. An a.rid,,: Technically, tms problem is not well defined, since a pole/zero plot does not wuquely
determine a system. That is, many system functions can have the same pole/zero plot. For example,
consider the systems
H1(z) = •-•
Hz(z) = 2.r- 1
Both of these syswms haft the same pole/zero plot, umely a pole at zero and a zero at iDfinity.
Clearly, the system H1 (z) is allpass, as it passes all frequencies with wuty gain (it is simply a wut
delay). However, one could ask whether H 2 (z) is allpass. Looking at the standard definition of an
137
allpaas system provided ID this chapter, the answer would be no, since the system does not pass all
&equencies with •nit11 gain.
A broader definition of an allpass system would be a l)'llem for wbic:b the system magnitude respome
IH(el-')j = o, where a is a real constant. Such a system would pass all &equencies, and scale the output
by a constant factor a. Ill a practical setting, this definition of an allpass system is satisfactary. Under
this delinitiOD, both systems H 1(z) and B 2(z) woaJd be considered allpua.
For this problem, it is assumed that none of the poles or seros shown in the pole/sero plots are scaled,
10 this issue of uiDg the proper definition of an allpass l)'llem does not apply. Tbe standard definition
of an allpaas system is nsed.
Solution:
5.14. (a) By the symmetry of z 1 [n] we kaow it has linear phase. Tbe symmetry is around n = 5 so the
continuous phase of X 1 (e''") is arg[X1 (~w)] = -S.,. Thus,
. d { . } d
grd[X,(e'w)] = - dw arg[X,(e'w)] = - dw (-S.,} = 5
3 x,[nJ
2
1
1234.56789 n
(b) By the symmetry of z2[n] we know it has liaear phase. Tbe symmetry is around n = 1/2 so we
kaow the phase of X 2(~w) is arg[X2(~] -w/2. Thus, =
grd[X2(~w)] = -~ {arg[X2(~w)]} = -~ {-~}
dw dw 2
= !2
312
3/4
3/8
••• • ••
-2 -1 0 1 2 3 n
5.15. (a) h[n] is symmetric about n = 1.
B(~) = 2+e-""+2e-2;w
= e-""(2e'W +I+ 2e-iw)
= (I+ 4cosw)e-;w
138
H(e'"') = 1 - •- 2'"'
= •-'"'(e'"' - .-,.. )
= e-1""2i sin""
= (2siDw)e-i"'+if
A(w)
.
= 2S1Dw, a= 1, /J =
.
2
Generalized Linear Phase but Doi Linear Phase since A(w) is not always positive.
5.16. The causality of the syslem caDDot be determined &om the figure. A causal system h 1 (n] that has a
linear phase respoDSe LH(ei•) = -av,, is:
h 1 (n] = .S(n] + 26(n - l] + 6(n - 2]
H1(e'"') = 1 + 2e-;., + .-;:i.,
= .-,.. (~ + 2 + .-,.. )
= e-1"'(2 + 2cos(w))
IH,(e'"')I = 2+ 2cos(w)
LH1(e'"') = -w
AD example o( a non-ausal system with the same phase response is:
h2(n] = 6[n + l] + 6[n] + 46(n - 1) + 6(n - 2] + 6[n - 3)
H2(e'"') = ~ + 1 + 4e-;., + .-J2w + .-;s.,
= •-;.(.;:i.., + ~ + 4 + •-;w + .-,-...)
= •-'"'(4 +2coo(w) + 2coo(2w))
-
IH2(~)I = 4+ 2CXll(w) + 2coo(2w)
LH2(e'"') =
Thus, both the causal sequence h,(n] ud the DOD-caasal sequace h2(n] have a liDear phase respoDSe
LH(e'"') = -aw, where a= I.
139
5.17. A minimum phase system Is ooe which has all its poles and zeros illside the wlit circle. It is causal,
stable, and has a causal and stable in-.e.
(a) H, (z) has a zero outside the llllit circle at z =2 ID it ii DOI minimum phase.
(b) H2(z) is minimum phue lince its poles and sero. are inside the unit circk.
(c) Bs(z) is minimum phue lince its poles and seros are inside the UDit circle.
(d) H,(z) bas a zero outside the UDit circle at z =
00 ID it la DOI millimum phase. Moreover, the
ilrRr.se system would no& be causal due to the pole at iwity.
5.18. A minimum phase system with an eqail'llleut magnitude spectrum ean be found by analyzing the system
fmictiOll, and rellectin,g all poles are sero. dial are outside the wlit circle to their COlljugate reciprocal
locat.i0111. Tim will lDOft them illlide the Wlit circle. Then, all poles and uras for H ,..,.(z) will be
inside the llllit circle. Note dial a ICAle factor may be introduced wh• the pole or zero is rededed
inside the UDil circle.
(a) Simply rellect the zero at z = 2 to Its oanjugate reciprocal location at z = ½- Tben, determine the
scale factor.
H-(z) =2 G: t:=:)
(b) First, simply reflect the zero at z =
-3 to its conjugate reciprocal location at z = -!- Then,
determine the sea.le faaor. Tbis NSlllts in
Note that the term ;h does not alfect the frequency response magnitude or the system. Con-
sequently, it can be remo,ed. Thus, th~ remaining t"'1l! has a zero inside the unit circle, and is
therefore minimum phase. Aa a result, - are left with the system
(c) Simply reflect the zero at 3 to its conjugate reciprocal location at } and reflect the pole at f to its
conjugate reciprocal locatioll at ¾. Then, determine the scale factor.
To find each system's group delay - need only find the point or symmetry n 0 in each system's impulse
response.
140
5.20. (a) Ye.s. The system function could be a generalized linear phue system implemenled by a linear
constaDkoellicient cWferential equaiion (LCCDE) with real coefficients. The seros come in a
set "' four: a r.ero, iu CODjugate, and the WO COlljugate reciprocals. The pole-zero plot could
correspond to a Type I FIR linear phue system.
(b) No. This system function could - be a generalized linear phue system implemenled by a LCCDE
with real coelficiems. Since the LCCDE has real c:oellicients, its poles aad seros mmt come in
conjugate pairs. Howe-,er, the seros in this po1e-sero plot do not have corresponding conjugate
(c) Yu. The system function could be a generalized linear phase system implemenled by a LCCDE
· with real coefficients. The pole-zero plot could correspond to a Type II FIR linear phue system.
1
.... . ..
-II -1114 0 ffl4 II II)
(b) z(n] is first modulaled by ,r, 1owpass filtered, and demodulated by .-. Therefore, H,,(ei'"') filters
the high &equency components cl X(e-i").
This is a highpass filter.
1
••• •••
. 0
-II --311/4 II QI
(c) h1,(2n] is a da,rnsampled version of the filter. Therefore, the &equency respoase will be "spread
out" by a factor of two, with a gain of ½-
This is a lowpass filter.
!
141
••• 112 • ••
0 K (I)
(d) This system apsamples lli,.!n] by a factor ol two. Therefore, the frequency ui5 will be compressed
by a faaor of two.
Thia is a bandstop lilter.
•••
- 1 ..--
••
.
-a-7Kl8 lt/8 0 a/8 7Jt/8 K (I)
(e) This system upsamples the input before passing it through h,,[nj. This effectively doubles the
frequency bandwidth ol H,,(.,;w).
This is a lowpass filter.
••• 112 • ••
-1112 0 ll (I)
5.22.
I - a- 1 z- 1 Y(z)
H(z) = 1
_ az-• = X(z), causal, so ROC is [zi > a
(a) Cross multiplyuig and taking the inverse transform
l
1,t(n] - a1,1(n - l] =z[n] - -z(n - I]
a
(b) Since H(z) is causal,.., blow that the ROC is lzl > a. For stability, the ROC must include the
unit circle. So, H(z) is stable for !al < 1..
(c) a=½ ·
.-112 Re 2
142
(d)
1 -,.-1
..
B(•l= _ ... _,- _ ...-,, l•l>ca
1 1
1+:!,-1C()I..,)½
1
IB(ei")I = '
( 1 + a2 - 2a COS"1
= ! ("' + 1- 2aC()lw) ½
G 1 + a2 - 2a C()IW
1
= .
5.23. (a) Type I:
Jl/2
C()I0
Type II:
A(w)
(Jl+l)/2 l
A(w) L
= _, b{n)cosw (n- )
2
cooO = 1, coo (n.- - j) = 0. So H(ei•) = O.
Type ill:
-
Jl/2
A(w) = L c(n) sillwn
siDO = 0, sinn.- = 0, so B(ei") = B(ei•) = O.
Type IV:
(Jl+l)/2 1
A(w)= L
_, d(n)sillw(n- )
2
sinO = 0, sin (n.- - j) ,f, 0, so just B(ei") = 0.
B(z) - Y(z) - -¼
+ ,-•
- X(z) - 1- ¼z- 2
Since the poles and lel'0II {2 poles at z = ±1/2, 2 lel'0II at z = ±2} occur in conjugate reciprocal
pairs the system is allpw. This property is wy to r-cogniu mice, as in the system above,
the c:oellicients ol the nlllDf!rator and denominator z-polynomials get reversed (and in general
conjugated).
(b) It is a property of alJpw systems that the output energy is equal to the input energy. Here i.s the
proof.
..
-
N-1
:E 11,[nJ1 2
=
--
= :E
..1•
..!._
2,r -•
i11[n]l 2
= 1-:2~ IH(eJW)X(,:iWJl dw
2
= 1-:2~ IX(e1W)l dw
2
(IH(,:iw)I' = I since h[n] is allpass)
=
..
L lz[n]l
2
(by Parseval's theorem)
fto:=:-00
N-1
-
L 2
= lz[n]l
= 5
5.25. The statement is /out. A non-causal system can indeed have a positive collStallt group delay. For
example, collSider the non-ausal system
.
X
Re
144
rz- 1 rz- 1
B(z) =,,......,.,,.--,--,---:---:,
1- {2r a,s'"'°)z-' + r2z- 2 = {l - rz-
')'' lzl > r
Again, using a table loolcup g;...,. us
h(n) = nr"u[n)
Im
1 zeroalZ=•
5. 27. Making use of some DTFT properties can aide in the solution of this problem. First, note that
h 2 (n) = (-l)"h1 [n)
h2(n) = e-i•"hi[n]
Using the DTFT property that states that modulation in tbe time domain corresponds to a shift in the
&equency domain,
B2(~) = H, (eJ<w+•l)
Consequently, B 2 ( ~ ) is simply B 1 (ei'-') shifted by .-. Tbe ideal low pass filter has now become the
ideal high pass filter, as shown below.
-7"2 0 "'2
-lt -7"2 0
145
5.28. (a)
A 1
H(z) = (l- ½z-l)(l + ½z-l)' lzl > 2 h(n] ausal
H(l) =6~A =4
(b)
4 1
H(z) = (1- ½z- 1)(1 + tz- 1)'
lzl> 2
= (1/) + <fl
1 - ½z-1 1 + 1,-1
h(n] = -12
15
(1)''
-
2
u(n] + -8 ( - -1)• u(n]
5 3
(c) (i)
1 1 - 1,-1
:r:(n] = u(nJ - 2u(n -1] # X(z) =
1
_ •,_ 1 , lzl > 1
Y(z) = X(z)H(z)
1 -21,-1
_ ____ 4
= , -· l•I > 1
. _: - j•-')(l+½z- 1 )'
4
=
(1- z- 1 )(1 + tr')
3 1
= 1- z-1 + 1 + 1,-1
3
5.29.
21
H(z) = (1 - ½.-• )(1 - 2.-• )(1 - 4z-1)
l 28 48
= l - 1,-1
-- --+
l - 2.-1 l - 4z-1
2
1 28
H,(z) =I- ½z-' - 1- 2.- 1
48
H,(z) = I - 4z-1
5.30. (a)
M
M-2 M-1 n
-1/4
(b)
1
111(n] = z[n - (M - 2)] - M]
4z[n -
1
v[n] = w(2n] = z[2n - (M - 2)] -
4z(2n - M]
H(z) = (l _
z-•
½•-• ){l _ 3,_ 1 ), stable, so the ROC is ½< lzl < 3
!
147
l
:[n] = u[n] ~ X(z) = _ z-l, lzl > l
1
1 l l
Y(z) = X(z)H(z) = 1 _ ½z-• + 1 _ Jz-• - l _ .-•, l < lzl < 3
y[n) = 54(l)" l
2 u[n] - 5(3)"u[-n -1)- u[n)
(b) ROC includes z = 00 so h{n] is caasal. Since both h[n) and :[n] are 0 for n < 0, we know that y[n)
isalso0forn<0
H z - Y(z) - z-2
( ) - X(z) - l - ;.-• + Jz-2
l
H,(z) = H(z) =z - 27 z + 32,
2
ROC: entire z..plane
7 3
h;(n] = o[n + 2) - 2o[n + 1) + 26[n)
5.32. Since H(e"') has a zero on the unit circle, its inverse system will have a pole on the unit circle and
thus is not stable.
5.33. (a)
H1(z) = 1- •-•o,-1
There are 8 zeros at z =.- 0
,; t• for k =0, ... , 1 and 8 poles at the origin.
Im
lzl>O
0 0
11u Re
0
ftlordorpolo
(b)
(c) Only the causal ho(n] is stable, therefore only it can be used to recover •In].
•-°", n = 0,8,16, ...
hin 1={ 0, otherwise
(d)
1 1 2 - lz- 1 1
H(z) =
1-
½z-1 + 1- ½z-' = •
1- iz-' + 11 z-•
, lzl > -2
Since h[n], z(n] = 0 for n < 0 we can assume initial rest conditions.
5 l 5
11(n] = 11[n - l] - 2] + 2>:[n] -
6 611[n - 6z[n - l]
(b)
N-1
11(n] L h[m]z(n - m]
=.....,
(d) For llR., we have 4 multiplies and 3 adds per ontput point. This gnes us a total of 4N multiplies
and 3N adds. So, IIR grows with order N. For FIR, we have N multiplies and N - 1 adds for the
n" output point, so this configuration bas order N2. ·
5.35. (a)
Im
X
Re
X
H(z)
= (1- 2z- 1 )(1 + 1,-
1 )(1 + 0.9,- 1 )
(b)
Im
Re
I H(el°') I
20~-------~~--------,
l
f
0
-It 0 1t
I
(I)
!t
(d) (i) The system is not stable since the ROC does not include i•I = l.
'
(ii) Because h(n] is not stable, h(n] does not approach a coDStant as n -+ ex,.
(iii) We can see peaks at"'= ±f in the graph of IH(e'w)I shoWD iD part (c), so this is false.·
(iv) Swapping poles and zeros gives:
Im
Re
There is a ROC that includes the unit circle (0.9 < lzl < 2). However, this stable system
would be two sided, so - must conclude the statement is false.
5.37.
(l - tz- 1 )(1- ¼z- 1 )(1- lz) 6 (l - ½z- 1 )(1- lz- 1 )(1- 5,- 1 )
X(z)= (l-tz) • =s {l-~z-•)
Therefore, a•:[n] is real and minimum phase ilf a is real and lal < l·
5.38. (a) The causal systems haw conjugate zero pain inside or outside the UDit circle. Therefore
(1 - i.25e-JO-k z-l)
H,(z) = (1.25)2(1 - o.!lf:IO·••,.- 1 )(1 -o.9e-i0-•• .. - 1 )(1 - o.Se'°·'• ,.-•> •
(1 - o.ae-;o·•• ,.-•i
H2(.r) baa all its r.eros outside the UDit circle, and is a maximum phase sequence. Bs (z) has all its
r.eros inside the UDit circle, and thus is a minimum phase sequence.
(b)
7.1
I.I
012345n 012345n
27.8 27.1 27.1 27.8 27.I
EJ_n) ~ ·• EJ_n) 22.3
17
10.8
9;1
~
J
012345n 012345n
152
Th• plot ol E,[n] conesponds io the minimum phase sequence.·
5.S9. All zeros inside the llllit circle means the sequence is minimum phase. Since
Ill Ill
= O aad just compute h2 [0]. The largest result will he the minimum
TheansnrisF.
5.40.
(i) A zero phase ,equence has all its poles and zeros ill conjugate reciprocal pairs. Generaliz.ed
linear phase sySlemS an zero phase sySlemS with additioul poles or seroo at z = 0, 00, l or
-1.
(ii) A stable system's ROC includes the UDit circle.
(a) The poles aN not ill conjugau reciprocal pairs, 10 this does not have zero or generalized linear
phase. H,(z) has a pole at z = o aad perhaps z = 00. Therefore, the ROC is O < l•I < 00, which
means the iDvene is stable. If the ROC includes • = 00, the inverse will also he causal.
(b) Sillce the poles a,e not conjugate reciprocal pairs, this does not have sero or generalized linear
phase either. H,(z) has poles illside the wilt circle, 10 ROC is 1•1 > j io match the ROC of H(z).
Therefore, the illvene is both stable and causal.
(c) The seroo occur ill conjugate reciprocal pairs, 10 this is a zero phase system. Th• inverse has poles
both inside and outside the UDit circle. Therefore, a stable non-causal inverse emu.
(d) The r.eros
occur ill conjugate reciprocal pairs, 10 this is a zero phase system. Since the poles of the
inverse system are on the llllit circle_& stable illvene does not exist.
5.41. Convolving two symmetric sequences yields &DOtber symmetric sequence. A symmetric sequence con-
volved with an antisymmetric sequence gives an antisymmetric sequence. If you convolve two antisym-
metric sequences, you will pt a symmetric sequence.
!
153
all
••• •••
& (I)
••• • ••
-all ........... .
0 & GI
(b)
•••
a=3
•
0 ,• 2• l
3 •
4
e
5
e
6
...
n
III
(ls 3.5
••• 0 5 y •••
l, 2
b
I 3 4 6 n
•••
(l:
0
6
3.25
, l
3
T
4
5
.&
'i'
6
(c) U a is an integer, then h(n] is symmetric about about the point n = a. U a = ':, where Mis odd,
•••
n
=
-
L h[n]•-;..n
(N-2)/2
L
...0
h[n]•--
.-(Jl+2)/2
+ L
N
h(nje-;,,n + h[M/2]•-;w(N/2)
(N-2)/2 (N-2)/2
= L h[n]•-;wn + L
h[M - m]•-jw(N-,nJ + h{M/2]•-;w(N/2)
n=O ......
154
=
e-;w(Jl/2)
(
-
(Jl-2)/2
L h(m)e'"((Jl/2)-"') +
= e-,w(JI/Jl (~2h((M/2)-n)coswn+h(M/2))
Let
ca{n) = { h(M/2), n 0 =
2h[(M/2)-n), n= l, ... ,M/2
Then
...
Jl/2
B(e"') =e-,w(JI/J) L o(n) coswn
and 'ft have
Jl/2
-
M
.A(w) = L o(n) cos(wn), 0=2, /3=0
-
JI
H(e'") = L h(n)•-;""
(Jl-1)/2 JI
= L h(n]•-'"'" + L h(n]e-;""
..... na(Jl+l)/2
=
(Jl-1)/2
L
.....
= ,-,w(Jl/2)
h[nJ•-- +
(
f
(Jl-1)/2
-
(Jl-1)/2
L h(M - m),-;w(Jl-m)
h(m)e"'((Jl/2)-"'l +
(Jl-1)/2
l;, h(m)e-;"'((M/2)-ml
)
Let
6(n) = 2h[(M + 't)/2 - n), n = 1, ... , (M + 1)/2
Then (Jl+l)/2
B(e"') = e-,w(JI/Jl L
_, 6(n) cosw(n - (1/2))
and.., have
(Jl+l)/2
M
.A(w) = L
_, b[n) cosw(n - (1/2)), 0=2, /3=0
155
B(ei°')
=
-
= L h(n)e-;....
(1,/-2)/2
L
..0
h(n)e-i""' + O+
(l,/+2)/2
I,/
L
h(n)e-;....
-
(1,/-2)/2 (1,/-2)/2
=
=
-L h{n)e-;.," +
•-j.. (1,//2)
(
(1,/-2)/2 ·
L
....0
L
h[M - m)e-i"'(l,/~m)
h(m]ei-'((1,//2)-m) _
(1,/-2)/2
L
h[mJ•-;.,((M/2)-m)
m=O
)
= •-;.,(M/2) (i 2
(Mj;/ 2h[m]sinw((M/2)-m))
Let
c(n) = h((M/2)-n), n = l, ... ,M/2
Then
1,//2
B(ei"') = •-;.,(M/2lei<•J2) L c(n)sinwn
n=l
and we have
M/2
M
A("') = L c(n] sin(wn), 0=2· fJ=-"2
....,
• Type IV: Antisymmetric, M Odd, Even Length
M
B(ei"') = L h(n)e-;....
n=O
(M-1)/2 M
= L h[n]e-i"'" + L h[nJ•-'"'"
n=O n=(M'+l)/2
(M-1)/2 (M-1)/2
= L h(n]e-;""' + L h{M - mJ•-;.,(M-m)
...0 -
(M-1)/2 (1,/-1)/2 · )
= •-;.,(M/2) ~ h[m]ei"'((l,//2)-m) _ ~ h[mJ•-;..((M/2)-m)
(
2
= •-;.,(M/2) (; (Mj;/ 2hfm)sin"1((M/2) - m))
(Jt+l)/2
= •-;..(M/2lei1•/2l
....L, 2h[(M + 1)/2 - n] sinw(n - (1/2))
156
Let
d[n] = 2h[(M + 1)/2- n], n = 1, ... ,(M + 1)/2
Then (Af+l)/2
and we have
B(ei"') = e-;.,(J1/•let(•/2l
....L d[n] sm"'(n - {1/2))
(Jl+l)/2 M ,,.
A("1) = L 2. ,.,"'=-2
d[n] llill"'(n - (1/2)), a- - -
5.44. Filter Types II and m CaDDOt be bighp•ss filters siDce they both must haw a wo at z =1.
Type I -+ Type I could be bighpus:
l IIII ➔
'
IIIIII ➔
'
➔
157
5.45.
H(z) = (l -O.sz- 1 )(1 + 2jz-1 )(1- 2jz- 1 )
(1 - 0.8z- 1)(1 + 0.8z- 1 )
2j
Im
112"'5 Re
-21
(a) A minimum phase syslelD has all poles and zeros inside l•I = 1
H ( J = (1 - o.sz- )(1 + ¼z- 2 )
1
I z (1 - Q.64z- 2 )
Im
(1l2)j
112 "'5 Re
-(112~
Im
(112¥
Re
-(1/2)j
158
=
(bl A generalizedlinearphue system has zeros and poles at z 1, -1,0or oo or in conjugate reciprocal
pain.
· (1 - o.sz- 1 )
Im
(112M
112 4/S Re
-(1J2M
3rd--
2j
Im
Re
-2i
5.46. (a) Minimum phue systems ha..., all poles and zeros inside izl =1. Allpass systems have pole-zero
pairs at conjugate reciprocal locations. Generalized linear phue systems ha..., pole pain and zero
pain in conjugate reciprocal locations and at z = 0, 1, -1 and oo. This implies that all the poles
and zeros of B.,,n(•l are seconCH>rder. When the allpass filter llips a pole or zero outside the unit
circle, one is left in the conjugate reciprocal locati011, giving us linear pha&e.
(b) We lcnow that h[n] is length 8 and theraore has 7 zeros. Since it is an even length generalized
linear phue filter with real coefficients and odd sy,nmetry it must he a Type IV filter. It therefore
has the property that its zeros come in coajugate reciprocal pain stated mathematically as B(z) =
B(l/z"). The zero at z = =
-2 implies a zero at z =-½,while the zero at z o.s.i<•l•l implies
zeros at z = O.&-i(•I•>, z = l.:zsei<•l•l and z = 1.25e-i(•/4 l Bec•o,e it is a IV filter, it also must
have a zero at z = 1. Putting all this together g;...,. us
B(z) = (1 + 2z-1)(1 + O.s,- 1)(1- O.Sei<•l•lz- 1)(1 - o.s.,-il•l•J z- 1).
(1-1.:zseil•/•lz-')(l - l.2Se-i(•/<lz-•)(1 - ,-1)
!
159
X(el°')
'•(10lt) (10lt)
5
5.48. (a)
Property Applies? Comments
Stable No For a stable, causal system, all poles must be
inside the unit circle.
IIR Yes The system has poles at locatiollS other than
.r=Oorz=oo.
FIR No n.K systems can ollly have poles at • = 0 or
z=c:o.
Minimum No Minimum pDMe sysiems have all poles and r.eros
Phase located inside tbe unit circle.
Allpass No Allpua systems have poles and zeros in conjugate
reciprocal pain.
Generalized Linear Phase No Tbe causal generalised linear phase systems
presented in this chapter are FIR.
Positive Group Delay for all w No This system is not in the appropriate form.
160
(b)
rl"P,-ro""pe.=t=-,--------,I-A,...p-p7eli-es""•"Tl""'Coo-111111e11--ts-----------
St&ble Yes The ROC for this system fwlcti011,
lzl > 0, cocwm the amt circle.
(Note there is 7th order pole at z = O).
IIR No The system has ........ only at z = o.
FIR les The s,stem has poles Ollly at • = O.
Minimum No By de6mtioa., a minimum phase system must
Phase haft all its poles uad _,. locued
inaib die amt cirde.
Allpus No Note that die 1G01 Oil die amt circle will
came the mapitude spectrum to drop r.ero at
certain frequencies. Clearly, tbis system is
not allpass.
Generalized Linear Phase Yes n
This is the pole/zero plot of a type FIR
linear phase system.
Positift Group Delay for all w Yes This system is causal uad linear phase.
Comequently, its group delay is a positift
COIIStaDt.
(c) .
Property Applies? L;OmmeDtl
Stable Yes All poles are iDside the Wlit circle. Since
the system is causal, the ROC includes the
Wlit circle.
IIR Yes The s,stem has poles at locations other than
• =O or•= 00.
Fffi No FIR 1JS1e1DS CUI only haft poles at z = 0 or
Z = 00.
Minimum No Minimum phase systems haft all poles uad zeros
Phase locued inside the wut circle.
Allpass Yes The poles inside the amt circle have
wmspODdillg 1er01 locued at cocjugate ·
reciprocal locatioDS.
Generalized Linear Phase No ·ue causal generalized linear phase systems
presented ill this cbapler are FIR.
Positive Group Delay for all w Yes Stable allpass systems have positive group delay
for all v,.
5.49. (a) Yu. By the region of wnvergence we know there are DO poles at z = 00 uad it therefore must be
causal. Another way to Me this is to use long divisiOll to write H1(z) as
(b) h 1 [n] is a causal rectUlglll&r pu!,e of length 5. If we a111'10lve h,[n] with UlCllher causal rectangular
pulse of length N we will get a triUlgular pulse of leDgth N + 5 - 1 = N + 4. The trw,gular pulse
is symmetric aro1111d its apa uad thus has linear phase. To mw the triangular pulse g[n] have at
least 9 llOllzerO aamples we CUI chome N = 5 « let h,(n] = h1 (n].
Proof:
161
(c) The required values for h 3 [n] can intuitively be worked out using the flip and slide idea of conv~
lution. Here is a second way to get the answer. Pick h3 (n] to be the inverse system for hi(n] and
then simplify using the geometric series as follows.
l-z- 1
H,(z) = 1- z-•
= (1 - z- 1 ) [1 + z-• + z-• 0 + z-» + .. -)
= I - z-1 + z-5 - z-6 + z-10 - z-11 + z-15 - z-16 + ...
This choice for h3 (n] will make q[n] = o[n] for all n. However, since we only need equality for
0 S n S 19 truncating the infinite series will give us the desired result. The final answer is shown
below.
1 ' hJnJ
- - - 10
---
0 5 15 n
-1
'
5.50. (a) This system does not necessarily have generalized linear phase. The phase response,
G,(,;,-) = H,(,;,-)H2(,;,-)
jG2(&w)j = jH,(&w)j jH2(,;,-)j
LG2(&W) = LB,(,;,-)+ LH2(,;,-)
The sum of two linear phase responses is also a linear phase response.
162
(c) This system does not necessarily have linear phase. Using properties t: ~ DTIT, the circular
convolution of H,(.,;"') and H 2(_,j"') is related to the product of h 1 [n] and h2[n]. Consider the
systems
Proof: Although the group delay is constant ( grd (H(e'"')] = 4.3) the resulting Mis not an integer.
h[n] = ±h[M-n]
H(e'"') = ±ei"'"' H(e-i"')
e-;4.3111
= :1:eJ(M'+4.3)1o1 I [wl <w,
M = -8.6
5.52. The type II FIR system H 11 (z) has generalized linear phase. Therefore, it can be written in the form
where M is an odd integer and A.(_,j"') is a real, even, periodic function of w. Note that the system
=
G(z) (1- ,- 1 ) is a type rv generalized linear phase system.
G(ei"') = 1 - e-jw
,-,w/2(,i"'/2 _ ,-;w/2)
=
= ,-jw/2(2j sin(wf.2))
2 sin(w /2)e-jw/2+jw 12
=
A.,(e'"')•-jw/2+jw /2
=
A.(ei"')
LG(e'"')
=
= --+-
2 2
.
2 sin(w/2)
w
The cascade of Hu(z) with G(z) results in a generalized linear phase system H(z).
H(ei"') = A,(e'"')A.,(ei")e-j"'M/2,-jw/2+jw/2
= A'.(ei"')e'"'"'' /2+;w/2
where A' 0 (e'"') is a real, odd, periodic function of wand M' is an even integer.
Thus, the resulting system H(e'"') has the form of a type m FIR generalized linear phase system. It is
antisymmetric, has odd length (M is even), and has generalized linear phase.
163
5.53. For all of the following we know that the poles and zeros are real or occur in complex conjugate pairs
since each impulse response is real. Since they are causal we also know that none have poles at infinity.
(a) • Since h1[n] is real there are complex conjugate poles at z = o.9e:;•/3 •
• If z[n] = u[n]
Y(z) = H 1 (z)X(z) = 1H_1 ,-,
(z)
We can perform a partial fraction expansion on Y(z) and find a term (l)nu[n] due to the pole
at z = 1. Since 11[n] eventually decays to zero this term must be cancelled by a zero. Thus,
the filter must have a zero at z = 1.
• The length of the impulse response is infinite.
(b) • Linear phase and a real impulse response implies that zeros occur at conjugate reciprocal
locations so there are zeros at z = z 1 , 1/ z 1 , zi, 1/ ~i where z1 = 0.8e' .. / 4 •
• Since h2(n] is both causal and linear phase it must be a Type I, Il, m, or IV Fm filter.
Therefore the filter's poles only occur at z = 0.
• Since the arg {H2(&")} = -2.Sw we can narrow down the filter to a Type II or Type IV filter.
This also tells us that the length of the impulse response is 6 and that there are 5 zeros. Since
the number of poles always equal the number of zeros, we have 5 poles at z = O.
=
• Since 20log jH2 (&0 )j = -ex, we must have a zero at z 1. This narrows down the filter type
even more from a Type II or Type IV filter to just a Type IV filter.
With all the information above we can determine H2(z) completely (up to a scale factor)
(c) Since H,(z) is allpass we know the poles and zeros occur in conjugate reciprocal locations. The
impulse response is infinite and in general looks like
M
II <1- c.,-•i
X(z} = Go
bo =•cc=.,_!----
N
II (1-d·•-'l
•=1
Because x[n] is real, its zeros must appear in conjugate pairs. Consequently, there are two more
zeros, at z=½•-i•I•, and z = ½•-,"I•. Since x(n] is zero outside O:,;
n:,; 4, there are only four
zeros (and poles) in the system function. Therefore, the system function can be written as
Im
X(zj
0 0
Re
0 0
4lhordlfpole
(c) A sketch of the pole-zero plot for Y(z) is shown below. Note that the ROC for Y(z) is lzl > ½-
Im
X X
Re
X X
5.55. • Since z[n] is real the poles & zeros come in complex conjugate pairs.
• From ( 1) we know there are no poles except at zero or infinity.
• From (3) and the fact that z[n] is finite we know that the signal has generalized linear phase.
• From (3) and (4) we have a = 2. This and the fact that there are no poles in the finite plane
except the five at zero (deduced from (1) and (2)) tells us the form of X(z) must be
X(z) =z[-l]z + :t[O] + :t[l]z- 1 + z[2]z-• + z[3]z- 3 + z[4)z 4 + z[S)z-•
The phase changes by .. &t w = 0 and .- so there must be a zero on the unit circle at z = ±1. The
zero &t z = 1 tells us Ez[n] = 0. Tb,e zero &t z = -I tells us E(-l)"z[n] = 0.
We can also conclude z[n] must be a Type m filter since the length of z[n) is odd and there is a
ze~o at both z = ±1. z[n] must therefore be antisymmetric around n = 2 and z[2] = 0.
• From (5) and Parseval's theorem we have E lz[n]l 2 = 28.
• From (6)
y[O] = 2-1•
2,r -r
Y(eiw)dw =4
= z[n] • u[n] ln=0 = :t[-1) + :t[O]
11[!] = 2_
2,r
1•
-r
Y(eiw)ei"'dw =6
= z[n] • u[n] I..,., = z[-1] + :t[O] + z[l]
• The conclusion from (7) that E(-l)"z[n] = 0 we already derived earlier.
• Since the DTIT {z.[n]} = RA, {X(,.;-,)} we have
:r[S] + z[-5] 3
2
= -2
z(S] = -3+:i:[-S]
:t(S] = -3
!
165
:t[n] is easily obtained from solving the equations in the following order: (3),(10),(4),(8),(5),(9), and (6).
3
JC(nj
2
1
3 4 5
-1 0 1 2 n
-1
-2
-3
I 1
= (1- z-• + ¼z-•) = (1- ½z-•)•
H1(z)
This system function has a second order pole at z = ½· (There is also a second order zero at z = 0).
Evaluating this pole-zero plot on the unit circle yields a low pass filter, as the second otder pole
boosts the low frequencies.
Since
H2(e'"') = H,(-e'"')
H 2 (z) = H 1 (-z)
(b) Tbe LTI system S, is characterized as a highpass filter. H,(&"') is the inverse system of H, (e'"'),
since H,(e'"')H,(e'"') = 1. Consequently, H3 (z)H1 (z) = 1.
~ shown in part (a), H 1 (z) has a second order pole at z = ½, and a second order zero at z = 0.
Thus, H,(z) has a second order zero at z = ½, and a second order pole at z = 0. Evaluating this
pole-zero plot on the unit circle yields a high pass filter, as the second order zero attenuates the
low frequencies.
S3 is a minimum phase filter, since its poles and zeros are located inside the unit circle. However,
because the zeros of S. do not occur in conjugate reciprocal pairs, S3 cannot be classified as one
of the four types of FIR filters with generalized linear phase.
(c) First, we compute the system function H4 (z)
11(n] + a,y[n - 1] + a 2 v[n - 2] = Poz(n]
Y(z) + a 1 Y(z)z- 1 + a 2 Y(z)z-2 = A,X(z)
H4 (z) - Po
- 1 + 01z- 1 + o 2 z- 2
s. is a stable and noncausal LTI system. Therefore, its poles must be located 01dsidt the unit
circle, and its ROC must be an interior region that includes the unit circle. We place a second
order pole at z = 2, which is the (conjugate) recip,;ocal location of the second order pole of H 1 (z)
at z = ½- This gives
Po
In order for
jH.(e''")I = IH,(e'•)j
an appropriate value of /3o must be found. Consider the case when z = 1. Then,
The values a, = =
-4, a 2 4, and Po = 4 satisfy the criteria. Note that Po =-4 also is a valid
solution.
(d) If h•[n] • h 1 (n] is FIR, then the poles of H 1 (z) must be cancelled by zeros of Hs(z). Thus, we
1 2
expect a second order zero of H5 (z) at z = ½- Therefore, H.(z) will have the term (1- ½z- ) -
=
In order for the filter hs(n] to be zero phase, it must satisfy the symmetry property h•(n] h,[-n],
which means that H,(z) = H,(z). For this property to be satisfied, we need two more zeros located
at z = 2. In addition, we want these zeros to correspond to a noncausal sequence. Therefore, H•(z)
will also have the term (1 - ½z) 2 •
Combining these two results,
167
N M
11(n] = L "•llln - k] + L o.:z:[n - k], bo = 1
•=0
N M
Y(z) = L"•Y(z)z-• + X(z) + Lb•X(z)z-•
t=l i:=1
M JI
1+ L b•z-• II (1 - c•z- 1 )
H(z) = --•-=~•-- - A=•-::;•~---
N - N
1 - L ... ,-• II (1 - d.,- 1)
t=l A=l
So,
Or equivalently,
n
M
(1 - c.z- 1 )(1- c.,)
r.,, (z) = A2 a-•2 -="=c,.1
N '--------
fi (1- d.,- )(1- d•z)
1
H,.(z)H,.(z-') = H(z)!(z-1)
Therefore,
N
fi (1- d•z- )(1-d•z)1
H,.(z)H,.(z-•) = =";c;'~------
IT (1- c.z- )(1 - c,z)
1
l=l
The poles of H,.(z) are the seros of H(z) and the seros of B,.(z) are the poles of H(z). \_Ve mus\
now decide which N of the 2N zeros of H,.(z)H,.(z- 1 ) to associate with B,.(z). The remaining
N zeros and M poles will be reciprocals and will be uaociated with H,.(z- 1 ). Ia order for H,.(z)
to be stable, we must chose all its poles inside the nnit circle. Thus for a pair c•, c; 1 we chose the
one which is inside the nnit circle.
169
{c) There is no real constraint on the zeros of H,.(z), so we can select either d• or a, 1 • Thus, it is not
unique.
5.59. (a)
•
H,(e'w) = 11-e-jw.M
- ,-,.,
<e> h;[n] = Loo 6[n - kM) - 6[n - kM - 1]
•=<>
1 ,, 1
' 1
•••
1 M+1 2M+1
0 M 2M n
'-1 -1
h;[n] has infinite length, so we can never get a result without infinite sums. Therefore, it is not a
real time filter. We can use the transform approach but we must have all the input data available
to do this.
{b) The proposed system is a windowed version of h;[n]:
q 1-z-qM
h 1 [n] = L 6[n - kM) <e> H 1 (z) = -M
•=<> 1-z
Thus,
1 1-z-M
H>{z) = H(z) 1- ,-,M
Note that
1- ,_,,.,
bas M zeros and qM poles. Since H2(z) is causal, there are no poles at z = oo. If H(z) bas P
poles and Z zeros:
Z+M$P+qM
170
5.60. (a)
1 cz-1 a- z- 1
H(z) = z - -a = -a- = u- 1
Im
, poleatz=-
Re 1/a
·
H(e'"') = e'"'. - -a1 = cosw + j sinw - -a1
·
arg[H(e'"')] sinw )
= tan- 1 ( cosw _ ¼
(b)
Im
a Re
· z 1
G(z)=-=---,-1
z- a 1- az-
G(ei"') = 1 = ___1_ __
l - OL!-;"' I - a cosw + jasinw
5.61. (a) Because h 1 [n], h2 [n] are minimum phase sequences, all pales and urns of their transforms must
be inside the unit circle. ·
h,[n] • h2[n] ++ H1 (z)H2(z)
Since H (z) and H (z) have all their pales and zeros inside the unit circle, their product will also.
1 2
(b)
h 1 (n] + h,(n] ++ H,(z) + H2(z)
!
171
z2(n] =2 - (l)n
2
u(n] +---+ _
1- 1 1
2,z = X2(z)
This is minimum phase, with the same pole and zero as X,(z} and X2(z}.
R(z} = •3
1- lz-1
2
l -2z- 1
-2z-l
= (1 - ½z- )(1 - 2z- 1 1}
1
= - - - ---
(I- ½z- )(1-½z)' 1
ROC: ½< lzl < 2
Im
1 zeroatz=-
112 Re 2
(b)
r(n] = h(n] • h[-n] ~ R(z} = H(z)H(z- 1 )
R(z) =.-~--~- 1
(1 - ½z- )(1- ½z) 1
We have two choices &om H(z). Since h[n] is minimum phase we need the one which has the pole
at z ·= ½, which is inside the llllit circle.
±1
H(z} = (
1- 1 ,z _,)' ROC: lzl > ½
h(n] = ± (D n u[n]
172
Since the poles are outside the unit circle, the only stable system will have a ROC of lzl < min ld•I-
This implies the poles will all contribule to the h(n] with terms of the form -(d.ru[-n-1], which
are anticausal. The zeros are all positive powers of z, which means they are shifting to left, and
h(n] is still anticausal.
(b)
M
Hm,.(z) = hm,.(o] II (1- c,z- 1 )
k=l
(c)
H,....(z) = hm,.[O] ft
Pl
(1 - c.,-,) ft (:=• -,~,)
k=l C1c
M
= hm,.(O] II(,-, - c.)
bl
M
= ,- 11 hm,.(o] II (1- c;z)
i=l
= ,-MHm,n(z- 1 )
(d)
(b) Since
l
H.(z) =H . ( )
min Z
We have
G(z) = H 0 ,(z)
(c)
H,,.;n(z) = (1.44)(1 - O.W°.>• z- 1 )(1 - O.Se-i0•3 • z- 1 )(1 - (5/6),;o•" ,-1 )(1 - (S/6)e-i0• 7• z- 1 )
l
H.(z) = (1.44)(1- O.SeiO-••z-• )(1 -O.Se-i0.3••-• )(1 - (5/6)ei0• 7 •z- 1 )(1 - (5/6)e-i0-7 • z- 1 )
(z- 1 - (5/6)e-i0 "Hz-• - (5/6)ei0·")
G(z) = H.,(z) = (1- (5/6)ei0- 1 •z- 1 )(1- (5/6)e-,0·"z-')
Im
0 0
Re
0
41h order pole
Im
X X
Re
C#'lorderzero
5.65.
z- 1 -a
H(z) = Hm,n(z) , Jal< l
1 -az- 1
Thus,
1 - az- 1
fun Hm,n(z) = fun _1 H(z)
.1 ➔ 00 .1 ➔ 00 z - a
h,,.;n[O] = - !h(O]
a
Therefore, lhm,n[O]I > Jh[O]I since Jal < 1. This process can be repeated if more than one allpass system
is cascaded. In each case, the factor for each will be larger than unity in the limit. ·
5.66. (a) We use the allpass principle and place a pole at z = •• and a zero at z = .J,.
••
.
H(z)
174
(b)
€ = L lh...m[m]l 2
- L lh[m]l 2
ffl=CI ~
n
= L (lq[m]l 2 - z•q[m - ljq*[mj - z;q•[m - ljq[mj + l••l 2 lq[m - 1]1 2 )
n
- L (lq[m - 1]1 2
- z;q•[m - l]q[mj - ••q[m - ljq•[m] + l••l 2 lq[m]l 2 )
n
= (l - 1••12 ) L (lq(m]l 2
- lq(m - 1Jl 2 )
= (l - l••l 2 )lq(n]l 2
(d)
S.67. (a) x[n] is real, minimum phase and x(n] = 0 for n < O. Consider the system:
xfn} .
- H,..,(z) H..,(z) - y[o]
x(n] is the impulse response of a minimum phase system. 11[n] is the impulse response of a system
which has the same frequency response magnitude as that of x[n] but it is DOt minimlllll phase.
Therefore, the equation applies.
n n
L l:r:[kJl2 ~ L 111[1<]1
.... ....
2
Since h0 ,(n] is causal and :r:[n] is causal, 11[n] is also causal, and these sums are meaningful.
175
(b) As discussed in the book, the group delay for a rational allpass system is always positive. That is,
Therefore, filtering a signal z[n] by such a system will delay the energy in the output y(n]. H we
require that z[n] is causal, then 1,'[n] will be causal as well, and the equation
n n
L lz(A:]1 L l11[kJl
2
::'.
2
bO t=O
1,'(n] = g[n] + r[-n] = z{n] • h(n] + z[n] • h(-n] = z(n] • (h[n] + h[-n])
(c)
✓2 ······
31t/4
1 ······----- 0 fl/4
" "'
-✓2 ·····················
0 "14 3fl/4
" "'
In general, method A is preferable since method B causes a magnitude distortion which is a function
of the (possibly non-linear) phase of h[n].
176
1 1
H(z) = (1 - ½z-' )(1 - 2,- 1 ) - 1- !z-1 + z- 2
This system function bas poles at z = 1/2 and z = 2. However, as the following shows it is a generalized
linear phase fiher.
= e;"' _ 1 +e-jw
2
= (2cos~ - ½) e'w
5. 70. (a) Since h[n] is a real causal linear phase filter the zeros must occur in sets of 4. Thus, if z1 is a zero
of H(z) then zi,1/z, and 1/zi must also be zeros. We can use this to find 4 zeros of H(z) from
the given information.
(b) There are 24 zeros so the length of h(n] is 25. Since it is a linear phase filter it bas a delay of
(L - 1)/2 = (25 - 1)/2 = 12 samples. That corresponds to a time delay of
(c) The zero locations used to aeate the following plot were estinlated from the figure using a ruler
"!'d a protractor.
-100
0 0.2 0.4 0.6 0.8 1
n •Tht
5. 71. (a) There are many possible solutions to this problem. The idea behind any IOlution is to have h(n]
be an upsampled (by a factor of 2) version of g(n]. That is,
Thus, h[n] will process only the even-indexed samples. One suC:. system would be described by
h[n) = 1 +o(n -2)
gfn) = l+o(n-1)
H(z) = l+z-•
G(z) = l+z- 1
(b) As in part a, there are many possible solutions to this problem. The idea behind any solution is
to choose an h(n] that cannot be an upsampled (by a factor of 2) version of g[n]. Clearly, choosing
h(n] to filter odd-indexed samples satisfies this criterion. One such h{n] would be
h[n] = 1 + o(n - 1) + o(n - 2)
H(z) = 1 + z- 1 + z-•
(c) In general, the odd-indexed samples of h[n] must be zero, in order for a g(n] to be found for which
=
r{n] 11(n]. Thus, there must not be any odd powers of z- 1 in H(z).
(d) For the conditions determined in part c, g{n] is a downsampled (by a factor of 2) version of h[n].
That is,
g[n) = h{2n)
5.72. (a) No. You cannot uniquely recover h[n] from c,..(1].
c.. [1) = h[~ • h(-1]
c,..c.,;w) = H(.,.iW)H(e-iw) = IH(.,.iw)I'
C,..(z) = H(z)H.(1/z")
Causality and stability put restrictions on the poles of H(z) (they must be inside the unit circle)
but not its zeros. We know the zeros of c.. (z) in general occur in sets of 4. Here is why. A
complex conjugate pair of zeros occur in H(z) due to the fact that h[n] is real. These 2 zeros and
their conjugate reciprocals occur in C.,.(z) due to the formula above for a total of 4. Thus, H(z)
is not uniquely determined since we do not know which 2 out of these 4 zeros to factor into H(z).
This is illustrated with a simple example below.
CM(ZJ Im 01/zl •
Oz1
Re
Oz1•
_,ordo<pole 01/z ,
Let the above be the pole-zero diagram for c.. (z) and
H 1 (z) = (1- z,z- 1 ) (1- z;z- 1)
1
H 2 (z) = (1-:,z- ) (1-:iz-
1
)
Since
C.,.(z) = H 1 (z)Hi(l/z•) = B2(z)H2{1/z•)
we cannot determine whether h 1 (n] or h2 (n] generated c,..[1).
178
(b) Yes. The poles of c..(z) must occur in sets of 4 for the same reasons outlined above for the zeros.
However, since the poles of h[n] must be inside the unit circle to be causal and stable we do not
have any ambiguity in determining which poles to group into h[n]. We always choose the complex
conjugate poles inside the unit circle. Here is an example
CM(r, Im X 1/p •
1
xp,
Re
Xp1 •
4thanle<zero X 1/p
1
Let the above be the zero/pole diagram for c.. (z). Then, if h(n] is to be real, causal, and stable
H(z) must equal
1
H(z) -
- (1-p,z- 1 )(1-pjz- 1 )
5. 73. As shown in the book, the most general form of the system function of an allpass system with a
real-valued impulse response is
l•I ER.
where R. is the ROC which includes the unit circle. Correspondingly, the associated inverse system is
Now send the corrupted signal g[n] through a highpass filter hap/ [n] with a cutof of 111, = ,r /2.
179
112
• • • -5 -1 1 5 • • •
-7 -3 o 3 7 n
The highpa.ss filter completely filters out the lowpass signal :r[n]. The output 11[n] is
H(z) = L h[n]z-n
n=O
N-1
= L h(N - l - n]z-n
n=O
H(z) = L• h(m]zm-N+l
m=N-1
180
N-1
= L h[m]z"'z-N+l
.....0
N-1
= z-N+1 L h[m]z"'
.....0
= z-N+1H(z- 1 )
Solutions - Chapter 6
6.1. We proceed by obtaining the transfer functions for each of the networks. For network 1,
W(z)
z[n]
-rsin8
rsin8
rcos8
y[n)
z-•
then
and
Eliminate W ( z) to get
Y(z) rsin8z- 1
H,(z) = X(z) = l -2rcos8z- 1 +r 2 z- 2
That is:
Y(z)(l + ¼z- 1 - ¾z- 2 ) = X(z)(2 + ¼z- 1 ).
The system function is thus given by:
(b) To get the difference equation, we just inverse Z-transform the equation in a. We get:
1 3 1
11[n] + -11[n - l] - -y[n - 2]
4 8
= 2:[n] + -z[n
4
- 1].
w[n]
z(n] y(n]
(a)
(c) Adds and multiplies are circled above: 4 real adds and 2 real multiplies per output point.
(d) It is not possible to rednce the number of storage registers. Note that implementing H(z) above
in the canonical direct form II (minimum storage registers) also requires 4 registers.
4 4
3 3
1 1 1 1
(a) (b)
0 0
-1 -1
-2 -2
3 3 3
2 2 2 2
1 1 1 1
(c) (d)
0 0
-1 -1 -1 -1
6.7. We have
y(n]
1/4
w,(n] w,(n]
x(n] y[n]
z-•
-1 2 w,(n]
w3(n]
,-1
4
186
Taking the Z-transform of the above eqnations, rearranging and substituting terms, we get:
l + 3z- + z- - s,-
1 2 3
H(z) = ;:_:_..:::...---'-.:..____.:_.
l +z- 1 -Sz- 2
The difference equation is thus given by:
h(O] =l
h(l] =3 - h(O] = 2.
(b) From part (a) we have:
(b) Using the Z-transform of the difference equations in part (a), we get the transfer function:
:i:(n] -·I____._.!·-•----:
l .--..--I.-1.--:______.I,i
-1/2
187
(c) The system function has poles at z = -½ and z = 1. Since the second pole is on the unit circle,
the system is not stable.
6.11. (a) H(z) can be rewritten as:
z- 1 - 6z-• + sz-•
H(z) = '
1- 2 z -1 .
1/2
t,;-1
-6
t,;-1
8
{b) To get the trausposed form, we just reverse the arrows and exchange the input and the ouput. The
graph can then be redrawn as:
:[n] y[n]
z-•
1/2
iz-1
-6
Lz-1
6.12. We define the intermediate variables w1 [n], w,[n] and w3 [n] as follows:
-1 W1[n] 2
:[n] ---i----<---11'--+-----ell[n]
188
wi(n] =
-z[n] + w:,[n] + w,[n]
w:,[n] = z[n - l] + 2w,[n]
w,(n] = w,[n - 1] + y[n - 1)
1,1[n] =
2w,(n].
Z -transforming the above equations and rearranging and grouping terms, we get:
6.13.
z[n]
- - y[n]
z-• z-1
1/4
I
•
z-• z-•
1/8
-1/2
6.14.
z[n]
- - y[n]
z-•
1/2 5/6
z-•
1/2 1/6
!
189
6.15.
1 - !z- 1 + !z- 2
H(z) = l +•z-, + !•z-•.
2
To get the transposed direct form Il implementation, we first get the direct form II:
:i:[n] -
- y[n]
z-1
-1 -7/6
z-1
-1/2 1/6
Now, we reverse the arrows and exchange the role of the input and the ouput to get the transposed
direct form II:
:r[n] y[n]
z-1
-7/6 -1
z-1
1/6 -I . ,
~
6.16. (a) We just reverse the arrows and reverse the role of the input and the output, we get:
:i:[n] y[n]
- -
z-1 z-1
-1/2 -2
'
z-1
1/4 3
190
(b) The original system is the cascade of two transposed direct form ll structures, therefore the system
function is given by:
1
H(z) = ( l - 2z- + 3z-• )(l - !,-').
1-!z-2 2
•
Tbe transposed graph, on the other band, is the cascade of two direct form ll structures, therefore
the system function is given by:
Tbis confirms that both graphs have the same system function H(z).
6.17.
6.18. The flow graph is just a cascade of two transposed direct form II structures, the system function is
thus given by:
(1+2z- 1)(1-lz-
3
1)
B(z) -
- (1 + ¼z-1 - }z-•)(1 - cz- 1)
In order to implement this system function with a second-order direct form ll signal flow graph, a
pole-zero cancellation has to occur, this happens if a = J, = = =
a -2 or a O. U a j, the overall system
function is: ·
1
B( ) 1 + 2z-
z = 1 + lz-1 - !z-2.
Ua = -2 , the overall system function is: • •
Now we can draw the Bow graph that implements this system as a parallel combination of first-order
transposed direct form n sections:
-8
:i:(n] -- -- y(n]
z-•
'
1/3
z-•
-2/3
which can be implemented as the following cascade of second-order transposed direct form ll sections:
192
:[n]
-- - y[n]
z-• z-1
2 5/2
z-1 z-1
5/4 -1/4 -1
6.21.
. l Y(z)
h[n] =e1w•"u[n] +-¼ H(z) = . = --.
1 - eJ.,.z- X(z)
1
= =
So y[n] e'w•y[n - I]+ :t[n]. I.et y[n] 11,[n] + i11,[n]. Then 11,[n] + i11,[n] = (cosWo +; sinWQ)(y,[n -
l] + jy,[n - 11) + :r[n]. Separate the real and imaginary parts:
11,[n] = :r[n] + coswo11,[n - l] - sinwoy,[n -1]
y;[n] = sinwoy,[n - l] + coswo11,[n - l].
:r[n] 11,[n]
-sinwo
COSWo
COSWQ
sinwo
6.22.
193
z[n] y[n]
• I
' ' •
I 1/4
I.-: l I 1/2
I.-: I
1 1
( 1 + z- ) ( 1 + z- )
H(z) = 1 - ½z-' 1 - ¼z-' .
z[n] y[n]
• ' ' ' •
I 1/2
I< I I 1/4
I.-: I
Plus 12 systems of this form:
:r[n] y[n]
1/4 1/2
1- lz-1
5
H(z) -
- I- !z-
4
1 + .!.z-
24
2 + .l.z( -
12
3)
so
. 1 1 5 1
bo = 1, bi = --5 and 01 = -4 , a2 = --
24
, 03 = --.
12
194
z[n] 11[n]
- .
z-1 z-'
-5/24 z-'
-1/12
.-
., [n] 11[n]
,-,
1/4 -1/5
,-•
-5/24
,-•
-1/12
(iii) Cascade form using 6.rst and second order direct form IT sections.
195
1 - 1,-1 1
H(z) = ( 1 + L-1 )( 1 - 1,-1 + 1,-2 ).
4 2 3
So
/Jo, = 1 , bu=-¼ , "21 = 0,
/Jo, = 1 , b1, = 0 , "2, = 0 and
au=-¼, a21 = 0, 012 = 022 ½, = -l·
:i:[n] 11[n]
-
.
,-1 ,-1
-1/3
(iv) Parallel form using first and second order direct form II sections.
We can rewrite the transfer function as:
..ll. .!!. - .l!. z-1
H(z) _ 12> + 12• 12•
- 1 + •lz- 1 1- lz-
,
1 - lz-2 ·
3
So
eo1 = 12:s ,eu =O,
36
•- - .!!. e12--125
'"'U2-125, - 1 and
l , . -0
.,_11 - -4 I ,.21 -
11 - n _
I ._.12 - 2l 1a 22 -- l
-3•
196
27/125
-
z-•
-1/4
(n] y[n]
•
-
--
z-•
1/2 -36/125
z-•
1/3
w 1 [n]
.-[n] --
y[n]
z-•
w,[n]
-1/5 1/4
z-•
w3[n]
-5/24
z-•
-1/12
(b) To get the difference equation for the flow graph of part (v) in (a), we first define the intermediate
variables: wi[n] , w,[n] and w3 [n] . We have:
197
1 5 - 1
11(n)- 11[n- l] + 11[n-2) + 11[n - 3)
4 24 12
= z[n)- 51 z(n -1).
Taking the Z-transform of this equation and combining terms, we get the following transfer func-
tion:
·.~•l I
y[n] z[n]
• ' •
(b)
1+ 1 -1
H(z) = 1- •'
½z-'
v[n) z(n]
•
I,
I l~. : I
1/2 1/4
•
198
(c)
y[n]
•
z[n]
•
y[n] = ax[n] + bz[n - l] + c:r[n - 2]
Hr(z) =a+ bz-• + cz- 2 = H(z)
(d)
H rsinllz-•
(z) = l -2rcosllz- 1 + r 2 z- 2
y[n]
,-1
w[n]
-rsinll
rsinll
rcosll z[n]
199
= X +z-1U
V
u = rcos/lV - rsin/lY
w = rsin/lV + rcosllz- 1 W
y = ,-1w
y
=-
X =
Hr(z)
rsinez- 1
= 1 - 2r cosBz-1 + r2z- 2
= H(z).-
6.25. (a)
H(z)
(b)
y[n]
--
.
:r[n] 11[n]
z-1
11/8 9/8
z-t
-5/4 9/8
z-t
7/8 11/8
z-t
7/8
The solution is not unique; the order of the denominator 2ud-order sections may be rearranged.
(b)
1
u[n] = :,:[n] + 2:,:[n - 1] + :r[n - 2] + u[n - l] - u[n - 2]
2
1
v[n] = u[n] - v[n - l] -
2v[n- 2]
w[n] = 11[n] + 211[n - l] + 11[n - 2]
7
11[n] = w[n] + 2w[n - lj + w[n - 2] + 2y[n - l] -
811[n - 2].
,L 1
. '
•
-,r -• • '
.. w
T 2
(b) For H,(z) =H(-z), replace each ,- 1 by -,-1 . Alternatively, replace each coeflicient of an
odd-delayed variable by its negative.
201
(c)
1,1[n]
"' [n] .-•
-1 2
.-•
2 -1
.-•
-2 1
.-•
1 -2
6.28.
y[n]
-
- -
"
,-1
a w[n] -1
(a)
(b)
202
z(n] 11(n]
• •
-b
6.29. (a)
z[n]
• I
11(n]
•
(b) From
it follows that
7
~ n -n
l -as Z -s
L- a z = 1 - az-1 .
n=O
(c)
z[n]
• •
•
11[n]
(a)
·-•,----------"--'----------,
...
--..... 203
-
•• .
... . ..
. .
-
- • •
• •
(b)
1
H(z) = 15
~[l+cos(!;(n-no))],-n
14 14
= L ,-n + ..!._15=<)2
..!._
15n=O
L .!. [&it(n-no) + ,-,ft(n-no)] ,-•
l 1- z- 10 l l •-;ifno[l -(&ifz- 1 ) 15 ]
= 151-z-1 + 152 1-&itz-t
l l e;ffno[l - (e-;if z- 1 ) 15 )
+ 15 2 l - .-Atz-1
l [ l le-;ifno
= -(1 - z- 15 ) .,..----,.1 + ~.........,=--+
15 l-z- 1-e'itz-t
1,;itno
2 ]
l - •-;if z- 1 •
(c)
When no = 15/2, .
204
1 _;.,.+(h/ll) ( - ·15w,,)]
2c-- ;1 1-e'
eiw+(2;/U) _ e-j•+(2;'UJ
½dn sin(l5w/2)]
sin ( w+(2; /15))
....... -.... -
..
.. .. ••
When no= 0,
•-;.,, [sin{l5w/2) ½e-h'< sin(l5w/2)
= - - sin(w /2) +..____,,....;:.'--'--:.,..:.
15 sin { w-(2; /15))
+
½&n sin(ISw/2)]
sin ( 1o1+(2;/15))
.....,.,_.._ ..
The system will have generalized linear phase if the impulse response has even symmetry (note it
cannot have odd symmetry), or alt.ematively, if the frequency response can be expressed as:
H(,;.,) =,-;•7 A.(ei"')
205
where A.(ei"') is a real, even, periodic function in w. We thus conclude that the system will have
generalized linear phase for no = 1f k, where k is an odd integer.
(d) Rewrite H(z) as
15 1
l-z- 1 cos~-cos(:!!c+~)z-
H(z) = ---
15
--- [
1- ,-
+ " 1
1- 2cos + z- 2• 15
z-1
" 2
]
•
15
1/15
:fn]
:,; 111in]
z-15 ,-1
-1
") /3
-1
G u[n]
:,;iTn] l+r -[n]
!/
-r r -1
1-r
.-, wn] z-1
(b)
respectively.
(c) U(z) = ,- 1(GX(z) + W(z)}, W(z) = -rU(z)- (1- r)Y(z), and Y(z) = z- 1((1 +r)U(z)- rY(z))
lead to
+ r)z- 2
H2(z) = 1 +G(l2rz- 1 + z- 2 = z
_2
H 1 (z).
6.32. (a)
(b)
(c)
(a)
207
•
z[n] •
11[n]
b c
H(z) = ai,-1 + d
1- bz- 1
so set b = 0.54, c = -l.852, and d = -0.54.
(b) With quantized coefficients b, c, and d, &i -I l and d -I -bin general, so the resulting system would
DOI be alJpass.
(c)
,-1
•
z[n] •
11[n]
-1
-1 w[n-1] -1
The first delay in the second section has output w[n - l] so we can combine with the second delay
of the first section.-
z-1 -1
•
z[n]
11[n]
•
208
w 1 (n] w,[n]
2 11[n]
•
:i:(n]
•
First, we find the system function, we have:
Taking the Z-transform of the above equations and combining terms, we get:
(1- z- 1 )Y(z) + z- 1 Y(z) = (2 + z- 1 )X(z).
The system function is thus given by:
H z _ Y(z) _ 2 + z- 1
( ) - X(z) - 1 + z-1 - z-2
Since the system function is second order (highest order term is ,-• ), we should be able to im-
plement this system using only 2 delays, this can be done with a direct form II implementation.
Therefore, the minimum number of delays required to implement an equivalent system is 2.
w,(n] w,[n]
2
:i:[n]
Taking the Z-transform of the above equations and combining terms, we get:
Since the transfer function is not the same as the one in part a, we conclude that system B does not
represent the same input-output relationship as system A. This should not be surprising since in
system B we added two unidirectional wires and therefore changed the input-output relationship.
6.35.
z-1/3
-
z[n]
- y[n]
z-1 z-1
'"'
z
From the graph above, it is clear that 2 delays and 2 multipliers are needed.
(bl
1
11[n] = (11[n - l] - :z:[n]) + :z:[n - l]
3
Which can be implemented with the following Bow diagram:
.-1
f,,'
:z:[n] •
I: I
- - - - - - - - - - • 11[n]
I
-1 ,-1
(c)
z- 1 - ! z- 1 -2
H(z) = (l-3z
1 :1 ){ 1 - 2 -1 ).
z
This can be implemented as the cascade of the Bow graph in part (b) with the following Bow graph:
:r[n] •
I, .
.-,
I: I
• 11[n]
-1 ,-1
However the above Bow graph can be redrawn as:
-1 ,-1
:r[n] • I
' ,-•!
: I· I • y[n]
Now cascading the above Bow graph with the one from part (b) and grouping the delay element
we get the following system with two multipliers and three delays:
-1
-1
211
6.36. (a) Transpose = reverse arrows direction and reverse the input/output, we get:
l, .
w,(n]
:r(n] •
l·-:
l :, -,: ,,~
..,, (n]_
• w,(n]
•
z-1
• y[n]
Taking the Z -transform of the above equations, substituting and rearranging terms, we get:
1
y[n] - l] - 2y[n - 2] = :r[n] + 2:r[n - l].
2y[n -
H(z) = 1 + 2,-1 .
1- lz-
2
1 - 2z-2
It has poles at
8 8
- - - and z
1 - -/33
= ----
1 + -/33
which are outside the unit circle, therefore the system is NOT BIBO stable.
(d)
H[O]
,-,
...
H[l]
,-,
z,
1/N
:r[n]
- ''
- y[n]
''
'' '
''
'
' '
H[N-1]
(b) Note that the•• 's are the zeros of (1 - ,-NJ. Then write H(z) over a common denominator:
H(z) =
Therefore, H(z) is the sum of polynomials in ,- 1 with degree :SN -1. Hence, the system impulse
response has length :S N.
(c)
!
213
= (1 - z-N)ii(m)/NI
l - z,,,z- 1
z=.i: ..
d N •
= j!i"{(l - z- )H[m]/N}lz=z-
1;{1 - z,,,z-l }Jz=z-
N z;;.N-l b[m]/N
= z,,,z;;.•
= H(m]z;;.N
= H[m].
(e) If h[n] is real, JH(eiw)I = =
JH(ei<2 •-wl)j, and LH(ei"') -LH(ei< 2•-wl). H(ei 2rtfN) = H[k] =
=
Jii[k]lei'l•j, so Jii[k]j Jii[N - k]j and ii[k] = -1/(N - k], k = 0, 1, ... , N - l.
= (l _ z-N) [H[O]/N
1 - z- 1
+ ii[N/2]/N +
1 + z- 1
t-• 1H[k)/N
L.,, - ztz- 1
+ t-=-• H[N - p]/N]
L., 1 - ZN-.,,z- 1
t=l p=l
= (l - z-N) [ii[O]/N
I - z- 1
+ H[N/2]/N +
1 + z- 1
t-• (1H[k]/N
L.,, - ztz- 1
+ H[N - k)/N)]
1 - z_.z- 1
•=1
= (l - z-N) [H[O]/N + H[N/2]/N +
l-z- 1 l+z- 1
"-'
t=l N l-2cas IV
.ii[oJ/16
-[n]
:,; -
y[n]
-z-16 .-,
llH[l]I
"
.-,
/3 Q
.-•
'Y
.-•
zTn] - --- '
h(O] h(l] h(2] h[lO]
M:!
(b)
oln]
--
:r[n]
M:l
111•n]
-
,-,
l I'2
The total computation can not be reduced because to compute the value of any given output
sample, the previous output value must be known.
216
(d)
nI
- M:l
-
z-1 z-•
7/8 1/2
CJ
u
M:l
.-,
1I12 7I 8
M:l
.-, .-,
1/2
I 7/8
rIVJ
. M:l
-
z-1
7 I8 112
I
Only direct form IT (ii) can be implemented more efliciently by commuting operations with the
downsamplers.
3.4cm - 10-•
3.4~ - 104 - sec
to traverse one section. Since the sampling rate is 20ldh {T, = 0.5 - 10-.sec), it takes two sampling
intervals to traverse a section. The entire system is linear and so the forward going and baclcward going
217
•
:i:(n]
Ps(e)
1/A
-A/2 A/2 e
Ps(e)
1/A
-A
218
6.41. Since the system is linear, 11[n] is the sum or the outputs due to :z:1[n] and :z:2[n]. Therefore
00 00
11[n] = L h,[k]:z:,[n - k] + L
b:-c:c,
h2[k]:z:2[n - k]
= 11,[n] + 112[n].
E{y,(m]112[n]} = E {,f 00
h,[l)z,[m - l) · .f 00 h2[k]:r2[n - k]}
00 00
If :z:,[n] and :z:2[n] are uncorrelated, E{:r 1 [m - l):z:2[n - kl}= O; hence, E{y,[m]y2[n]} = 0. Therefore,
11,[n] and 112[n] are uncorrelated.
6.42. (a) The linear noise model for each system is drawn below.
!
219
(a)
z[n] ho y(n]
(b)
-
b, a
(2a 2 )
(c)
y[n]
l.-:
z[n] ho
•
.,-• j '
I ' ' •
:b,.
l (3<72)
I a
(b) Clearly (a) and (c) are different. Thus tbe answer is either (a) and (b) or (b) and (c). H we take
(b) apart, we get
220
bo '
-- -
,-, ,-,
b, a
'
=
bo
• •
We see that the noise all goes through the poles. Note that the la' source sees a system function
(I - az- 1 )- 1 while the 2<7 2 source sees ,- 1 /(1 - a,- 1 ). However, the delay (z- 1 ) does not affect
the average power. Hence, the answer is (b) and (c).
(c) For network (c),
OJ = 3o-2_I_f l __ I_ dz
2,rj l - az- 1 l - az z
= 3o-2_l_f dz
2,rj (z - a){l - az)
3u2
= 1-a•·
= 2u2+u2(b2 +(abo+b,.)2)·
o l -a2
• Frequency domain calculation:
= _21.
1'J
f H(z)H(,-') dz
z
-b,bo
residue (z = 0) = - a
-
2
.d ( _ ) _ (boa+ b,)(bo + b,a) _ ~a+ /?,a+ b,bo + b1 boa
res, ue z - a - a(l - a•) - o(l - c')
1L (l)'
111n1 = -2i=O - = 11- (!f+l
n
;
4 2 i
For large n, y[n] = (1/2)/(3/4) = 2/3.
(b) Working from the diHerence equation and qoa.ntizating after multiplication, it is easy to see that,
= =
in the quantized case, 11(0] 1/2 and y(n] 5/8 for n 2'. 1. In the unquantized case, the output
monotonically approaches 2/3.
ll
32
..,
li!
'
i.rn.
2048
• 11[n] unqoa.ntized
n
0 1 2 3 4 5
222
! ! 5 5 !
1 8 8 i i I
2 y[n] quantized
I 0
llllI
1 2 3 4 5
n
So
!
Y(e'w) = H(e'w)X(e'w)
·
=1- 1 .
4 e-Jw
=
which implies that y[n] (1/2)(1/4)", which approaches O as n grows large.
To find the quantized output (working from the difference equa.tion): 11[0] 1/2, y[l] = =1/8, and
=
y[n] 0 for n 2.e:
l
'
2 1
fi
_, ...L
128
512
'
y[n] unquantized
n
0 1 2 3 4 5
½ !
•
y[n] quantized
n
0 1 2 3 4 5
6.44. (a.) To check for stability, we look at the poles loca.tion. The poles are located at
Note that
izl 2 " ' o.976 < 1.
The poles are inside the unit circle, therefore the system function is .stable.
223
· ·.-•D-a• · n.-•
a"'-J ·
z[n] -------w~1.-[n_J_ _ _
wi.[n_J_ __.._......._ y[n]
z[n]
• I
y[n]
•
(a) For Network 1, we have:
= z[n] -
w 1 [n] a8 z[n - 8]
w,[n] = ay[n - l] + w,[n]
y[n] = w,[n]
Taking the Z-transform of the above equations and combining terms, we get:
(b) Network 1:
z{n]
Network 2:
z[n]
,-, .-, z-• z-• z-• .-• z-•
1
., a' •• a• a• a• a1
y[n]
,
225
We thus conclude that to avoid overflow in network 1, we need:
1- ial
:z:ffl4% < --a
1 ••
Now, for network 2, the transfer function from input to output is given by o[n] +acS[n-1) +a2 6[n-
2] + ... + a1 o[n- 7], therefore to avoid overflow, we need:
l
:...., < l+ Ial+a2 + ... +,..
,_,,.
~,, <
1- a
1cr:.
That is:
5
!al< 7.
The largest value of lal such that the noise in network l is less than network 2 is therefore J.
227
Solutions - Chapter 7
(a) Therefore,
we get
S _ s+a _ A, A2 A2
,(s)- ,(s+a+jb)(s+a-jb) - s + s+a+jb + s+a-jb
where
a 0.5
A1 = a2 + b2' A2=---
a+ jb
Though the system h 2 [n] is related by step invariance to h,(t), the signal s 2 [n] is related to s,(t) by
impulse invariance. Therefore, we know the poles of the partial fraction expansion of S,(s) above
must transform as z1 = e•• T, and we can find
A, A2 2 A
S.(z) = l _ ,-1 + l _ e-(a+;•)T z-1 + l _ e-<•-;•)T ,-1
Now, since the relationship between the step response and the impulse response is
n oo
•2[n] = L h2[k] = L h2[k]u[n - k] = h2[n] • u[n]
S.(z) = 1-
H2(z)
z- 1
!
230
(c)
=
12 [l _
1=-00
1 - e-(o+;•>T
l:=O
e-(•+i•)(n+l)T
+
1_ e-<•-i•)(n+l)T]
1 - e-<•-i•lT u[n]
= [s, + s,.-,.+;•)Tn + B,,-1•-i•)Tn] u[n]
where
1- ,-•T cosbT e-<o+;•)T
B, = 1 - 2e-•T cos bT + ,-2oT' B, = - I _ e-l•+i•)T
;.From this we can see that
B, B2 Bi
= I_ z-1 + 1_ e-(o+ji)T z-1 + 1_ e-(o-j6')T z-1
I S.(z)
since the partial fraction constants are different. Therefore, s 1 [n] I s2[n), the two step responses
are not equal.
The overall idea this problem illustrates is that a filter designed with impulse invariance is different
from a filter designed with step invariance.
7.2. Recall that n = w/T•·
(a) Then
o.s912s :s IHUO)I :s 1, o :s 1n1 :s 0.2.- /T•
IH(jO)I :S 0.17783, 0.3,r /Td :5 101 :5 fr /Td
The plot of the tolerance scheme is
I H(jO) I
1
o.a912s -J;;:;:;;;;,-s-.a:--i.1-:..=;i:ii
o.1naa
00
231
(b) As in the book's example, since the Butterworth frequency response is monotonic, we can solve
1
IH,(io.2..JT•>I' = >N = (0.89125) 2
l + ( 0.2 .. )
O,Td
IH,(j0.3.-JT.)1 2 = l 2N = (0.17783) 2
l+ (0_3.,)O,T4
to get O,Td = 0.70474 and N = 5.8858. Rounding up to N = 6 yields O,Td = 0.7032 to meet the
specifications.
(c) We see that the poles of the magnitude-squared function are again evenly distributed around a
circle of radius 0.7032. Therefore, H,(s) is formed from the left half-plane poles of the magnitude-
squared function, and the result is the same for any value of Td- Correspondingly, H(z) does not
depend on Td-
cl,
=
=
.
I +01
I+ o,
(b) Solving the equations in Part (a.) for 61 and o,, we find
cl,
o, =
2 - cl,
26,
0, = 2 - cl,
In the example, we were given
6, = 1 - 0.89125 = 0.10875
cl, = 0.17783
Plugging in these values into the equations for o, and o,, we find
o, = 0.0575
a, = 0.1881
232
The filter H'(z) satisfies the discrete-time filter specifications where H'(z) = (I+ o,)H(z) and
H(z) is the filter designed in the example. Thus,
-2.1428 + 1.1455,- 1
H'(z) = 1 0575 [ 0.2871 - 0.4466z-•
· 1 - l.297lz-1 + 0.6949z-• + 1 - l.069lz-1 + 0.3699z- 2
1.8557 - 0.6303z- 1 ]
+ 1 - 0.9972z-1 + 0.2570z-2
0.3036 -0.4723z-• -2.2660 + 1.2114z- 1
= 1 - l.297lz-1 + 0.6949z- 2 + 1 - l.069lz- 1 + 0.3699z-•
+ 1.9624 - 0.6665z-1
1 - 0.9972z-1 + 0.2570z-2
(c) Following the same procedure used in part (b) we find
1. [ 0.0007378(1 + z- 1 ) 1
H'(z) = 0575
(1- l.2686z- 1 + 0.7051z- 2)(1- l.0106z- 1 + 0.3583z- 2)
X1 - 0.9044z-! + 0.2155z- 2]
0.0007802(1 + Z- 1)'
= (1 - l.2686z- + 0.7051z- 2)(1 - l.0106z- 1 + 0.3583z- 2)
1
1
x--------~
1 - 0.9044z-1 + 0.2155z-2
7.4. (a) In the impulse invariance design, the poles transform as z• = , .. T, and we have the relationship
1 Td
- a +-+ ----==,--~
I+- 1 - e-oT-,z-1
Therefore,
H,(s)
= 2/Td _ 1/Td
• + 0.1 s + 0.2
1 0.5
= - ----
•+0.1 •+0.2
The above solution is not unique due to tbe periodicity of z = .;w. A more general answer is
_ 2/Td
H,(s)
- •+(0.1+;~) s+ (0.2+;~)
where k and I are integers.
(b) Using the inverse relationship for the bilinear transform,
1 + (Td/2)s
z=
1- (T./2)s
we get
2 1
H,(s) = 1 - e--<>- 2 ( W,) 1 - e--<>-• ( W,)
2(•+ 1) <• + 1)
= s(l + e--<>.2) + (1 - e--<>·2) a(l + e--<>·•) + (1 - e--<>·•)
6 = O.Ql
l!.w = 0.05.-
A = -20log10 6=40
A-8
M + 1 = 2 _285.<:l.w + 1 = 90.2 -+ 91
/3 = 0.5842(A - 211•·• = 3.395
+ 0.07886(A - 2ll
(b) Since it is a linear phase filter with order 90, it has a delay of 90/2 = 45 samples.
(c)
Hiei"')
1
.
-It -0.6251t -0.31t 0 0.31t 0.6251t It (I)
7.6. (a) The Kaiser formulas say that a discontinuity of height l produces a peak error of 6. If a filter has
a discontinuity of a different height the peak error should be sea.led appropriately. This filter can
be thought of as the sum of two filters. This first is a lowpass filter with a discontinuity of l and
a peak error of 6. The second is a highpass filter with a discontinuity of 2 and a peak error of 26.
lo the region 0.3,r S lwl S 0.475,r, the two peak errors add but must be less or equal to than 0.06.
6 + 26 $ 0.06
6...,. = 0.02
A= -201og(0.02) =33.9794
/3 = 0.5842(33.9794 - 211•·· + 0.07886(33.9794 - 21) = 2.65
(b) The transition width can be
l:.w = 0.3,r - 0.2,r ar
6w = 0.525,r - 0.475,r
= 0.1,r rad = 0.05,r rad
=
We must choose the smallest transition width so l:.w...,. 0.05.- rad. The corresponding value of
Mis
33.9794-8
=
M 2.285(0.05,r) 72.38 -+ 73 =
7.7. Using the relation w = flT, the passband cutoff &eqnency, w9 , and the stopband cutotr frequency, w.,
are found to be
Wp = 2.. (1000110-<
= 0.2.. rad
= 2r(llOO)l0-'
"'•
= 0.22.- rad
234
Therefore, the specifications for the discrete-time frequency r~nse H4(ei•) are
w, = 2tan- 1 (fl;T)
3
= 2 tan-1 (2.-(2000)(~.4 X 10- ))
= 0.7589.- rad
7.11. Using the relation w = flT,
We
fie = T
,r/4
= 0.0001
= 2500,r
= 2.-(1250) rad
s
7.12. Using the bilinear transform frequency mapping equation,
fie = 2
-tan
T
-
2
(w')
= 2 ("'2)
0.001 tan 2
2000 rad
= s
= 2,r(318.3) rad
s
7.13. Using the relation w = flT,
We
T = fie
21t/S
= 2"(4000)
= 50µs
This value of T is unique. Although one can find other values of T that will alias the continuous-time
frequency fie = 2.-(4000) rad/s to the disaete-time frequency We = 2,r /5 rad, the resulting aliased filter
will not be the ideal lowpass filter.
235
= ftan(~')
T 2
= 2.-(300) tan (3,r/5)
- 2- = 1.46 ms
The only ambiguity in the above is the periodicity in w. However, the periodicity of the tangent function
"cancels" the ambiguity and so T is unique.
7.15. This filter requires a maximal passband error of••= 0.05, and a maximal stopband error of 6, = 0.1.
Converting these values to dB gives
op= -26 dB
•• = -20 dB
This requires a window with a peak approximation error less than -26 dB. Looking in Table 7.1, the
Hanning, Hamming, and Blackman windows meet this criterion.
Next, the minimum length L required for each of these filters can be found using the "approximate
width of mainlobe" column in the table since the mainlobe width is about equal to the transition width.
=
Note that the actual length of the filter is L M + 1.
Hanning:
8,r
0.1,r= M
M = 80
Hamming:
8,r
o.i.- =
M
M = 80
Blackman:
12,r
= M
0.171"
M = 120
7.16. Since filters designed by the window method inherently have 6, = cS, we must use the smaller value for
6.
6 = 0.02
A = -20log10 (0.02) = 33.9794
{J =0.5842(33.9794- 21) 0
·• + 0.07886(33.9794- 21) =2.65
A- 8 33.9794 - 8
M = 2.285&, = 2.285(0.65.- - 0.63.-) = 180.95 ➔ 181
7.17. Using the relation w = OT, the specifiation, which should be used to design the prototype continuous-
time filter are
-0.02 < H(jO) < 0.02, 0 :S 101 :S 2r(20)
0.95 < H(jO) < 1.05, 2,r(30) :S 101 :S 2.-(70)
-0.001 < H(jO) < 0.001, 2.-(75) :S 101 :S 21r(lOO)
!
236
Note: Typically, a continuous-time filter's passband tolerance is between 1 and 1 - 6, since historically
most continuous-time filter approximation methods were developed for passive systems which have a
gain less than one. H necessary, specifications using this convention can be obtained &om the above
specifications by scaling the magnitude response by ,.:.S.
7.18. Using the bilinear transform frequency mapping equation,
S - .!.. (1 - ,-iw)
- T• l +e-iw ·
The continuous frequency axis gets warped onto the discrete-time frequency axis, but the magnitude
values do not change. H H(s) is constant for alls, then H(e'") must also be constant.
7.21. (a) Using the bilinear transform frequency mapping equation,
we have
(b)
237
lt ················ . . . . . . . . . . . . . . . . . . . . . ······•··· .. .
"'p
0+---------------------•
0
(c)
w, = 2 tan-• ('\T•)
21.an-• (nf•)
w, =
tow= w, -w, = 2[tan- (n;T•) -1.an- (n;T•)]
1 1
lt
oL-__:================-
o
l•I > 1
which has the impulse response
= T•
2
(l+•-;"')
1- .-,..
=· r. (,;--1• + ,-;..,,)
2 ,;..12 _ ,-;,,,12
T•
= 2
j cot(w/2)
and since the Laplace transform evaluated along the j!l axis is the continous--time Fourier transform
we also have
H,(j!l) = j~
I H(ei"') I
2h
-It 0 7t/2 1t Ol
It It
-It 0 0) -It 0 n
-7t/2 1 - - - - -
In general, we see that we will not be able to approximate the high frequencies, but we can
approximate the lower frequencies if we choose T• 4/,r. =
(d) Applying the bilinear transform yields
G(z)
Jzl > 1
which has the impulse response
g(n]
239
(e) This system does not strictly have a frequency response either, due to the pole on the unit circle.
However, ignoring this fact again we get
G(.,;w) = ;. [! ~ :=::]
= 2 (.,;w/2 _,-;w/2)
Tc1 eiw/2 + e-ii.1/2
i.
2·
= tan(w/2)
G(jO) = ;n
-II 0 1112 II
., -II 0 1112 II n
LG(ei"') LG (jO)
C
1112 1112
-II -II
0 II ., 0 II n
-1112 -1112
Again, we see that we will not be able to approximate the high frequencies, but we can app•oxim•te
the lower frequencies ifwe choose T• 4/1'. =
(f) If the same value of T• is used for each bilinear transform, then the two gystems are inverses of
each other, since then
. =1
IH, (e'w)I ~ He (·"'
,!:.,, J T. + J.2.-k)I
T.
240
1Cht
Then, to get the overall system response we scale the frequency axis by T and bandlimit the result
according to the equation
n
(b) Using the frequency mapping relationships of the bilinear transform,
n = :.tan(i),
w = 2tan- 0{•) ,
1
(
we get
10,t
--
0.9871
-71 71 Cl)
Then, to get the overall system response we scale the frequency axis by T and bandlimit the result
according to the equation
I Helf (jO) I
• 10lt
-9800lt
7.24. (a) Expanding the sum to see things more clearly, we get
r A
= L ( • J• + G.(s)
bl I - lo
A1 A2 A, G( )
= + (S - .. ) 2 + + (1 - .. Jr +
0 0 0
I - .. C$
- d
= (r - l)A, <• - .. i•-2 + (r - 2)A2(• - ..r-• +···+A,_, + 0 + ds I(• - ..re.(,
d'-•
At = (r -1 k)! ( d,•-· [(, - •or Hc(•Jl l,z.. )
(b) Using the following transform pair from a lookup table,
i1-1 1
(k-1)!•-"'u(t)-+ (s+o)•' ?u{s} > -a
we get
242
7.25. (a) Answer: Only the bilinear transform design will guarantee that a minimum phase discrete-time
filter is created &om a minimum phase continuous-time filter. For the following explanations
remember that a discrete-time minimum phase system bas all its poles and zeros inside the
unit circle.
Impulse Invariance: Impulse invariance maps left-half 6-plane poles to the interior of the z-plane
unit circle. However, left.half s-plane zeros will not necusarilN be mapped inside the z-plane
unit circle. Consider:
H,(s)
H(z)
H we define Poly.(z) = Ilf., {I -e•,T•,- 1), we can note that all the roots of Poly.(z) are
j"¢Jc
inside the unit circle. Since the numerator of H(z) is a sum of A•Poly.(z) terms, we see
that there are no guarantees that the roots of the numerator polynomial are inside the unit
circle. In other words, the sum of minimum phase filters is not necessarily minimum phase.
By considering the specific example of
s+ 10
H,(s) = (s + l)(s + 2)'
and using T = l, we can show that a minimum phase filter is transformed into a non-minimum
phase discrete time filter.
Bilinear 'Iransform: The bilinear transform maps a pole or zero at s = So to a pole or zero
(respectively) at Zo = :~f;: .Thus,
l.zol = 11l +
-
f.. I
2 so
Since H,(s) is minimum phase, all the poles of H,(s) are located in the left half of the •-plane.
Therefore, a pole so = a + jfl must have a < 0. Using the relation for , 0, we get
(l + ta) 2 + (tO)•
l.zol = (1- fa) 2 + (tfl)2
< I
Thus, all poles· and zeros will be inside the z-plane unit circle and the discrete-time filter will
be minimum phase as well.
(b) Answer: Only the bilinear transform design will result in an allpass filter.
Impulse Invariance: In the impulse invariance design we have
H(e'w) = .t.., 2
He(;(;.+ ;:))
The aliasing terms can destroy the allpass nature of the continuous-time filter.
243
Bilinear Transform: The bilinear transform only warps the in,quency axis. The magnitude
response is not affected. Therefore, an all pass filter will map to an all pass filter.
(c) Answer: Only the bilinear transform will guarantee
H(e'"')l.,=0 = H,(jO)ln=0
Impulse Invariance: Since impulse invariance may result in aliasing, we see that
if and only if
or equivalently
f: (l;,k)
,.., __ H, =O
....
which is generally not the case.
Bilinear Transform: Since, under the bilinear transformation, 0 = 0 maps tow= 0,
H(e'0 ) = H,(jO)
(f) Answer: The property holds for both impulse invariance and the bilinear transform.
Impulse Invariance:
= H 1 (e'"') + H,(e'"')
244
Bilinear Transform:
H2 (eiw) = f: 2
+ ;.k))
H,, (; (;..
•=-00
We can clearly see that due to the aliasing, the phase relationship is not guaranteed to be
maintained.
Bilinear Transform: By the bilinear transform,
therefore,
and we desire
we see that
H(e'w )lw=0 = ~ (·(w 2.-k)) lw=0 = H,(jO)ln=0
L. H, 1 T• + T, .
lr=-oo
requires
f: (;2;,k) =O.
H,
·---
....,
(b) Since the bilinear transform maps ll = 0 tow = 0, the condition will hold for any choice of H,(jll).
7.27.
245
(a)
h1 [n] = h[2n]
00
H, (e'w) = L h[2n]e'wn
n=-00
= I:
n even
h[n]e"r
= f.
n.=-00
~ [h[n] + (-lth[n]Je'"r
I ¥I(""")
= 2H(e' ) + 2H e3-,-
H, (ei"')
1/2
(b)
h[n/2],-Jwn
H2(e'w) =
I:
n even
00
h[n]e-Jw2n
= I:
n.=-oc
= H (<'2w)
HiCei"'J
- 1
-
-x-7lt/8 -lt/8 0 lt/8 7lt/8 x <1l
(c)
1- z- 1
• = 1 + z- 1
1- e-jw
;n = 1 + e-Jw
ei""/2 _ e-Jw/2
= eiw/2 + e-J,.;/2
n = tan (i)
O.=tan(w;') ........ w,,. = 2ian- (0,) 1
(b)
I+ z- 1
• = 1-z-•
1+ e-jw
;n = 1- e-iw
e31i11/2 + e-iw/2
= eii.1/2 _ e-iw/2
n = -cot (i)
= =(w;.-)
n, = tan ( w"' 2-.-)
(c)
(d)
The even powers of z do not get changed by this transformation, while the coefficients of the odd
powers of z change sign.
Thus, replace A, C, 2 with -A, -C, -2.
(b)
247
H, (ei"')
ec ec
'
A
.
0 7fl2 II (I)
(c)
h[n] +-+ B(,,;')
h1[n] +-+ B (,,;<..,+•>)
In the frequency domain, we first shift by 1r and then we upsample by 2. In the time domain, we
can write that as
h [n]
1
= { (-l)nl2 h[n/2], for n even
O, for n odd
(d) In general, a filter
H(z) = bo + b,z- 1 + 1>,z-• + · ••+ b,.,_,zM-I + b,.,z-M
ao + a1z- 1 + a2z- 2 + · · · + CN-1zN-l + aNz-N
will transform under Hi(z) = H(-z2 ) to
bo - b,z-• + 1>,z-• + · · · - bM_,z2M-• + bMz-•M
H,(z) = -=--=-~-=---'----=--=---'---'=--~
Go - a1z-2 + a2z-t + ... - aM-1z2N-2 + O.Nz-2N
where we are assuming here that M and N are even. All the delay terms increase by a factor of
two, and the sign of the coellicient in front of any odd delay term is negated.
The given difference equations therefore become
g[n] = z[n] + a,g[n - 2] - bif[n - 4]
/[n] = -a2g[n - 2] - l>,/[n - 2]
y[n] = cif[n] + c2g[n - 2]
To avoid any possible confusion please note that the b, and a, in these difference equations are
not the same b, and a, sbown above for the general case.
7.30. We are given·
H(z) = He(•) I -/J[=]
1+ .. - ·
,=
where a is a nonzero integer and /3 is a real number.
• = /3 + z-
z-o
[11- 0
]
s + ,z-0 = /3 - {Jz-o
• -/3 = -/3z-a - sz-o
{J-. = z- (/J + s)
0
z-o = /3-.
fJ +.
zo /3 +.
= fJ-.
!
248
The poles •• of a stable, causal, continuous-time filter satisfy the condition & {s} < 0. We want
these poles to map to the points z•
In the z-plane such that lz• I < 1. With a > 0 it is also true
that if lz•I < I then lzfl < l. Letting •• =,; + jw we see that
< l
lz•I
< l
lzfl
1.B+a+JOI < 1.8- u-JOI
(,8 + a) 2 + 0 2 < (,8- a) 2 +02
2a,8 < -2<7,8
But since the continuous-time filter is stable we have&{,.} < 0 or a < 0. That leads to
-,8 < ,8
This can only be true if ,8 > O.
(b) It is true for ,8 < 0. The proof is similar to the last proof except now we have jz0 j > 1.
(c) We have
z' =
2
:~ ;J.=,n
lz I = l
izl = l
Hence, the jO axis of the s-plane is mapped to the unit circle of z-plane.
(d) First, find the mapping between {l and w.
1- e-i'lw
j{l = 1 + e-i2w
eJw - e-i1,,1
= eJw + e-ir.i
{l = tan(w)
w = tan- 1(0)
Therefore,
7.31. (a)
l + z- 1 s+ I
• = ---,-
1 - z-
+--+ z = - -
1 •-1
Now, we evaluate the above expressions along the jO axis of the •-plane
jO+ l
z =
;n-1
lzl = l
249
1 + e-i""
;o = 1 - e-,w
e'..,/2 + e-JIIJ/2
= eiw/2 _ e-Jw/2
o = - cot(w/2)
10,1 = lcot(,r/6)1=-✓3
10,,, I = Icot(.-/2)1 = o
1n.,1 = lcot(,r/4)1=1
Therefore, the constraints are
<2 = 2..1·
2,r -ir
IE(e'W)l 2 dw
= L""
n.=-00
le[n]l 2
where
(b) Since we only have control over e[n] for 0 $ m $ M, we get that <2 i, minirniwl if h[n] = h,,[n]
for 0 $ n $ M.
(c)
0$n$M,
w[n] = { ~: otherwise.
7.33. (a)
IHd(~w)I = 1, 'lw
;----~'1112
It
-It 0
-'1112 i----....:
(b) A Hilbert transformer of this nature requires the filter lo have a zero at z = 0 which introduces the
180° phase difference at that point. A zero at z = 0 means that the sum of the filter coefficients
equals zero. Thus, only Types ill and IV fulfill the requirements.
(c)
For the -windowed Fm system lo be linear phase it must be antisymmetric about ';. Since the
ideal Hilbert transformer h.i[n] is symmetric about n = T we should choose T = ':.
{d) The delay is M/2 = 21/2 = 10.5 samples. It is therefore a Type IV system. Notice the mandatory
zero at w = 0.
IH(ei°')I
0-1--------------~ It
251
=
(e) The delay is M /2 20/2 = 10 samples. It is therefore a Type ID system. Notice the mandatory
=
zeros at w 0 and ,r.
Oi------------......;
(j) "
7.34. (a) It is well known that convolving two rectangular windows results in a triangular window. Specifi-
cally, to get the (M + 1) point Bartlett window for M even, we can convolve the following rectangular
windows.
= {.f½, - ,-.-
n - 1, ... M-1
••lnJ
0, otherwise
In the frequency domain we have
= {2 sin(w(M + 1)/4) -,w( ¥ )
VM sm(w ;2J
= {2sin(w(M- l)/4) jw(.l!.f-!+t)
VM sm(w/2)
Ws(ei"') = WR,(ei"')WR.(ei"')
= 2. (smlw(M + 1)/21) (sm(w(M - 1)/2]) -jwll/2
M sin(w/2) sin(w/2) e
252
(b)
2
w[n] = [A+ Bcos ( :;') + Ccos ( ;,n)] WR(n]
4
W(,;w) = {2.-A6(w) +Bir [6 (w + !;) +6 (w- !;) ] +Cir [6 (w + ;;) + 6 (w- ;;) ]}
~ { sin(w(M + 1)/2)) e-jwlll/•}
2,r sin(w/2)
where @ denotes periodic convolution.
=
(c) For the Hanning window A= 0.5, B -0.5, and° C =0.
2
Wftanmnglnl = [o.s - o.scos ( :;')] w,[nJ
Wftanning(,;w) = 0.5WR(,;w) -0.25WR(,;w)@ [6 (w + !;) + 6( W - :)]
where
W (,;w) = sin(w(M + 1)/2)) -jwM/2
R sin(w/2) e
Below is a normalized sketch of the magnitude response in dB.
1/2 1/2 - - - - ~
This can be viewed as the sum of two lowpass filters, one of which has been shifted in frequency
(modulation in time-domain) tow= ,r. The linear phase factor adds a delay.
h,[n] = sin(0.3,r(n -24)) + !(-l)(n-••>sin(0.4.-(n -24))
1r(n - 24) 2 .-(n - 24)
253
(c) To find the ripple values, which are all the same in this case since it is a Kaiser window design, we
first need to _determine A. Since we lcnow /j and A are related by
0.1102(A - 8.7), A> 50
{J = 3.68 = { 0.5842(A - 21) 0 ·• + 0.07886(A - 21), 21 $ A $ 50
0, A< 21
we can solve for A in the following manner:
=
1. We lcnow /j 3.68. Therefore, from the formulas above, we see that A 2'. 21.
2. If we assume A > 50 we find,
3.68 = 0.1102(A - 8.7)
A = 42.1
But, this contradicts our assumption that A > 50. Thus, 21 :,; A $ 50.
3. With 21 $ A $ 50 we find,
3.68 = 0.5842(A - 21) 0·• + 0.07886(A- 21)
A = 42.4256
With A, we can now calculate 6.
6 = 10-A/20
= 10-42.425<1/20
= 0.0076
The discontinuity of I in the first passband aeates a ripple of 6. The discontinuity of 1/2 in the
second passband aeates a ripple of 6/2. The total ripple is 36/2 = 0.0114 and we therefore have
61 = 6, = 6, = 0.0114
Now using the relationship between M, A, and t:.w
A-8
M = 2.285!:.w
42.4256- 8
2.285(48) =0.3139 a, 0.I,r
Putting it all together with the information about Hd(eiw) we arrive at our final answer.
o.9886 $ IH(eiw)I $ 1.0114, 0 :S w $ 0.25w
IH(eiw)I :5 0.0114, 0.35w $ w $ 0.55w
o.4886 :5 IH(eiw)I :5 0.5114, 0.65w :,; w $ ff
I H(ei'°) I
1 t 20,
0.5 203 t
i;2
o-l-----___.;-..:..1.--===-;....--;..--------,,--- n
0 0.2511 0.3511 0.5511 0.6511 II
!
254
7.36. (a) Since H(ei0 J t, 0 and H(ei•J t, 0, this must be a Type I filter.
(b) With the weighting in the stopband equal to 1, the weighting in the passband is ~-
W(Ol)
1.6~---------,
(c)
IE(Ol)I
li2
It
0
-li2
(d) An optimal (in the Parks-McClellan sense) Type I lowpass filter can have either L + 2 or L + 3
alternations. The second case is true only when an alternation occurs at all band edges. Since this
filter does not have an alternation at w = .-
it should only have L + 2 alternations. From the figure,
= =
we see that there are 7 alternations so L 5. Thus, the filter length is 2L + 1 11 samples long.
(e) Since the filter is 11 samples long, it has a delay of 5 samples.
(f) Note the zeroes off the unit circle are implied by the dips in the frequency response at the indicated
frequencies.
Re
0
101h order pole
7.37. (a) The most straightforward way to find h,i(n] is to recognize that H,(,;w) is simply the (periodic)
convolution of two ideal lowpass filters with cutoff frequency We= .-/4. That is,
where
255
sin2 (.-n/4)
=
(b) h[n] must have even symmetry around (N - 1)/2. h[n] is a type-I FIR generalized linear phase
system, since N is an odd integer, and H(eiw) ,f- 0 for..,= 0. Type-I FIR generalized linear phase
systems have even symmetry around (N - 1)/2.
(c) Shifting the filter hd[n) by (N - 1)/2 and applying a rectangular window will result in a causal
h[n] th..t roinirni:res the integral squared error •· Consequently,
· 2 ['( N-1))
h[n] = sm • n- 2 w[n)
.-•(n _ N
2 1 )2
where
w[nJ = { l, 0$ n$ N - 1
0, otherwise
(d) The integral squared error <
•= L Ja[n] - hd[n)J
-00
2
Since
By symmetry,
00
• =2 L Jh,,[n)J
2
(N-1)/2+1
7.38. (a) A Type-1 lowpass filter that is optimal in the Parks---McClellan can have either L + 2 or L + 3
alternations. The second case is true only when an alternation occun at all band edges. Since this
=
filter does not have an alternation at "' 0 it only bas L + 2 alternations. From the figure we see
there are 9 alternations so L = 7. Thus, M = =
2L 2(7) 14. =
256
(b) We have
hHp(n] = -ei'"h£p(n]
HHP(ei"') = -HLP(,i(w-•))
= -A,(ei<w-•>),-;cw-•>¥
= A,(,;<w-•l)e-;.,y
= B,(ei"')•-;.,y
where
B,(ei"') = A,(,i<w-•l)
The fact that M = 14 is used to simplify the exponential term in the third line above.
(c)
II (I)
(d) The assertion is c:orrect. The original amplitude function was optimal in the Parks-McClellan
sense. The method used to create the new filter did not change the filter length, transition width,
or relative ripple sizes. All it did was slide the frequency response along the frequency axis creating
a new error function E'("') = E("' - ,r). Since translation does not change the Chebyshev error
(max IE("')I) the new filter is still optimal.
7 .39. For this filter, N = 3, so the polynomial order L is
N-1
L = --=I
2
Note that h[n] must be a type-I FIR generalized linear phase filter, since it consists of three samples,
and H (,;., ) ,;. 0 for "' = 0. h[n] can therefore be written ill the form
=
The filter must have at least L + 2 3 alternations, but no more than L + 3 = 4 alternations to satisfy
the alternation theorem, and therefore be optimal in the minimax sense. Four alternations can be
obtained if all four band edges are alternation frequencies such that the frequency response overshoots
at w = 0, undershoots at w = i, overshoots at w = f and undershoots at w = ,r.
I
Let the error in the pa.ssband and the stopband be •• and 6,. Then,
H ( ) = Y.,(s) A
' s X,(s) = s+c
(s + c)Y.,(s) = AX,(s)
dy,(t) + cy,(t) = Aze(t)
dt
(b)
dy,(t)
di,
I
t=nT
= [Az,(I) - C!/,(t}j lt=nT
= Az,(nT) - cy,(nT)
y,(nT) - y,(nT - T)
::: Az,(nT) - cy,(nT)
T
258
(c)
y[n] -y[n - 1]
T = Az[n) - cy[n)
= 1-z-
.,.----
1
+C
= H(z)
and thus the s-plane maps to the z-plane in the following manner
Therefore, the jO-ms maps to a circle of radius 1/2 that is centered at 1/2 in the z-plane. We
also see that_ the region c, < 0, i.e., the left half of the ,-plane, maps to the interior of this circle.
s-plane
z-plane
jQ
(I
If the continuous-time system is ,table, it, poles are in the left half s-plane. As shown above, these
poles map to the interior of the unit circle and so the discrete-time system will also be stable. The
stability is independent of T.
Since the jO-ms does not map to the unit circle, the ~ete-time frequency response will not be
a faithful reproduction of the continuous-time frequency response. As T gets smaller, i.e., as we
oversample more, a larger portion of the jO-ms gets mapped to the region close to the unit circle
at w = 0. Although the frequency range becomes more compressed the shape of the two responses
will look more similar. Thus, as T deaeases we improve our approximation.
(f) Substituting for the first derivative in the differential equation obtained in part (a) we get
y,(nT + T) - y,(nT) ( T)
T +cycn = Ax,(nT)
y[n + l] - y[n] []
T +eyn = Ax[n]
Y(z) A
H(z) = X( ) = 1 = H,(s) 1,-•-•
z T+c --,.-
z-1
s =
T
z = l+sT
= l + c,+ jOT
To find where the ;n axis of the s-plane maps, we let s = jO, i.e., " = O and find
z = l +jOT
Therefore, the jO-ms lies on the line 1ue{z} = 1. We also see that the region c, < 0, i.e., the left
half of the s=plane, maps to the left of this line.
z-plane
!
260
If the continuous-time system is stable, its poles are in tbe left half •-plane. As shown above, these
poles can map to a point outside tbe unit circle and ,o tbe discrete-time system will not necessarily
be stable. There are cases where ""Ying T can tum an onstable system into a stable system, but
it is not true for the general case.
Since tbe jO-axis does not map to the unit circle, the discrete-time frequency respome will not be
a faithful reproduction of the continuous-time frequency response. However, as T gets smaller our
approximation gets better for tbe same reasons outlined for the first backward difference.
7.43. (a} Just doing the integration reveal,
Using the area in the trapezoidal region to replace tbe integral above, we get
= I - z- 1 + cf (1 + z- 1 )
= H(z)
+,(jO} = He(jO)He(-jO)
+(z) = H(z)H(z- 1 )
261
(a} (i} Since H.(s) has poles at ••• H.(-•} has poles at -••·
(ii} The material in this chapter shows that under impulse invariance
A• T•A•
- - +-+ -
8 - $j; 1 -~ --.
~.1tT4z-l
Thus, going from step l to step 2 means that the autocorrelation of the discrete-time system
is a sampled version of the autocorrelation of the continuous-time system.
(iii) Since +(z) = H(z)H(z- 1 ) we can choose the poles and zeros of H(z) to be all the poles inside
the unit circle, and that choice leaves all the poles and zeros outside the unit circle for H(z- 1 ).
Consider the following example using h.(t} = ,-0 'u(t).
l 1
H.(s) =-
s+o
- and H.(-•l =- -
-s+o
+.(,} = H.(s)H.(-•i
= [,!0] [-,~0]
1/20 _ 1/20
= s+0 s-0
if 0 > 0, then
(bl Since !H.(jn)j 2 = +.(in) and ♦ (&"'}= H(&"')H(e-;,.,} = !H(&"')l2 , we see that since ¢[m] =
T.¢.(mT.), .
7.45. (a) Since the two 8ow diagrams are equivalent we have
z- 1 - a I - oz
z-•
= 1-az- 1 =-;=-;--
z-o
z = 1-oz
· = H,,(Z)lz= •-• = H,, (z-o)
H(z) -1 - -
t--a - QZ
1- oe-Jw
e-;'-oe-;•e-jw = e-;w -Q
Although a warping of the frequency scale is evident in the figure, (except when o 0, which=
corresponds to z-
1 = ,-, ), if the original system has a piecewise-constant lowpass frequency
response with cutoff frequency 8,, then the transformed system will likewise have a similar lowpass
response with cutoff frequency "'• determined by the choice of o.
xjnJ,----------0----y(n)
xjn) -----------1--+--<l,___
1
__., y[n)
H(z) -1
Htp(Z) r
- o.p -
Looking at tbe 6ow graph for H(z) we see a feedback loop with no delay. This effectively makes
the current output, y[n], a function of itself. Hence, there is no computable difference equation.
(d) y,., the 6ow graph manipulation would lead to a computable difference equation. The 6owgraph
of an FIR filter has a path without delays leading from input to output, but this does not present
any problems in terms of computation. Below is an example.
H,P(Z) bo
xjn) y(n)
xjn) ) l y(n)
r'! :: j H(z) -1
-1
b,
The transformation would destroy the linear phase of the FIR filter since the mapping between 8
=
and_ w is nonlinear. The only exception is the special case when o 0, i.e., when 8 w. =
Since there are feedback terms in the transformed filter, it must be an IIR filter. It therefore has
an infinitely long impulse response. ·
(e) Since the two 6ow diagrams are equiva.lent we have
-11 -
z-• = z
-1 z-1 - Q
1- az- 1
=z - -
z- a
QZ
z-a
z = z--
1-az
(I) •
2lt ..
-It
rc/2 1t 8
-It
......... -21t
We see from the plot of w versus 8 that a lowpass filter will not always transform into a lowpass
=
filter. Take, for example, the case when the original lowpass filter has a cutoff of 8 "/2. With
a = 0 it would transform into an allpass filter.
As shown above, if the passband of H(eiw) is the region (I -61 , 1 +6,1, then the passband of G(eiw)
is in the region [l - ~, l] which is a smaller band. However, the stop band gets bigger since it
maps to (-26, -6i, 262 - 5i].
Thus,
A = (I - 6f)
B =I
C = -26,-6~
D = U.-6i
If 61 « I and 0:, « I then,
(c) Since
and so
The new tolerance specifications can be found in a similar manner to the last section. We get,
A = l-36?-Uf
B = I
C = 0
D = ~+Ul
If 61 « I and 6, « I then,
I
266
(d) The order of the impulse response h(n] is M. Since it is linear phase it must therefore have a delay
of "f samples. To convert the two systems we must add a delay in the lower leg of each network
to match the delay that was added by the first filter.
l(nll-~t---:_l_hl_n_J_T'--hl_n_J---'~y(n]
1
2z......
The restrictions on the filter that carry over from part (a) are that it have
(i) Even symmetry
(ii) Odd Length
Hence, Type I Fm filters can be used.
The length of h[n] is 2L+ l. Since the term that is longest in the twicing system's impulse response
is the h[n] • h[n] term, the length of g[n] is 4L + 1. Since the term that is longest in the sharpening
system's impulse response is tbe h[n] • h[n] • h[n] term, the length of h5 harp[n] is 6L + 1.
7.47. We know that any system whose frequency response is of the form
L
A.(~"')= I>•(cos(w))'
l=O
can have at most L - l local maxima and minima ;ll the open interval O < w < ,r since it is in the form
of a polynomial of degree L.
If we include all endpoints of the approximation region
(b) A polynomial of degree L can have at most L - 1 local minima or maxima in an open interval.
Since A. (&"') has tiµee local extrema in the interval from O < w < .-, we lcnow L ~ 4.
Note that the optimal filter is half wave anti-symmetric if you lower its frequency response by one
half, i.e.,
A.,.(e""') = -A•,. ( .,;<•-.,>)
=
where A• .,(&"') H.,,,.(,;"') - 1/2. Another way of saying this is to say that the optimal filter is
anti-symmetric around w = .-/2 after lowering the response by 1/2. This property holds because the
optimal filter has symmetric bands with the same number of alternations. Plugging in A.,.(e-i"') =
H..,(ei"') - 1/2 into the above expression gives
h.,,,.[-n], n odd
h..,[n] ={ 0, neven,n-:/0
0.5, n=O
A sample plot of h.,,,.[n] appears below, for L = 6.
112
-1 o 1 4 s n
Note that because h.,,,.[n]= =
0 for n even, n 'I 0, a plot of h..,[n] for L 5 would have the same
=
nonzero samples, and therefore be equivalent. So the optimal filter with L 6 is really the same
= =
filter as the case of L 5, just as the optimal filter with L 4 is the same filter as the case with
L=3.
We know the filter non-optimal filter has 7 alternations. The optimal filter should be able to meet
the same specifications, but with a lower order. From part (a), we lcnow the number of alternations
must be even. Thus, the optimal filter for these specifications will have 6 alternations.
An optimal lowpass filter has either L + 2 or L + 3 alternations which means L = 4 or L = 3.
However, we showed above that these are really the same filter. Since the optimal filter has L = 4,
the filter shown in the problem cannot have L 4. =
Putting it all together we find L > 4 for the filter shown in the figure.
7.49. (a)
268
(b) The delay of tbe linear phase system is 51/2 = 25.5 samples since it is a linear phase system of
order 51. Therefore, the total delay is
Delay =
=
--
H(e1"")
25.ST + O.ST
26T
= 2.6 ms
HoUCI)
(c) H(eiOT) should cancel the effects of H 0 (jO). However, to cancel the effects of the delay introduced
by Ho(jfl) would require a noncausal filter which is not practical in this situation. Using the
relation w = nT,
oo~--"------------
0211 0.411 1t m
H(ei°')
1
0
o 0.2Jt 0.411 II {l) 0 0.2Jt "'
(d) lf H,(ifl) is also sloping across the band, lfll < ..;r, we would combine its effects with those of
H 0 (jfl) and compensate as in part (c), i.e.,
Mw. :S 1r
,r
M :5
"'•
So the maximum allowable decimation factor is
Mmax= - "
"'•
(b)
11------
Y(e"")
1/1001----------------------.
(I)
0.91! "
(c)
11------,.
,~I Wid"'J
00
.~
0.45x 50Ct>s1
"
(I)
270
Vi#>J
·~1 ~
00 0.45,,: 0.5K
"
Y(#>J
,nool
00
\
0.911
(d) After the first decimation by 50 is performed, W1 (&w) should look like the following:
II Cll
·~1 ~
oo-----'---~--.._---"'------'-----~~'--"'
0.4511 SOC!l.,
/] ,
" 211-SOC!l., 1.5511 '" w
50w,, $ 1.55,r
w,, $ 0.031,r
In general, the number of multiplies required to compute a single output sample is just N. For
a linear phase filter, however, the symmetry in the coefficients allow us to cut the number of
multiplies (roughly) in half if implementing the filter with a difference equation. The following
is a.n example of how this is accomplished using the simple Type I linear phase filter h[n] =
0.256[n] + 6(n - l] + 0.256(n - 2].
:::: 232
-lOlog,.(0.01 x 0.001) - 13 +1
2.324(0.5,r - 0.45.-)
:::: 103
H 3 (eiw) is real since A.(eiw) is real and .S. is real. It is nonnegative since A.(eiw) 2: -6,. Note
that H 3 (eiw) is an even function of w and is a zero-phase filter.
(b) H 3 (eiw) is a zero-phase filter with real coefficients. Thus, a zero at z• implies there must also be
zeros at z;, 1/z•, and 1/z,. In addition, a zero on the unit circle must be a double zero because
!
272
both its value and its derivative is zero. Note that this last property is true for H,(eiw) but not
for A.(dw). We can write H 3 (z) as
H,(z) = H2(z)H2(l/z)
where H2(z) contains all the complex conjugate zero pairs inside the unit circle and H2(l/z)
contains the corresponding complex conjugate zero pairs outside the unit circle. We factor one of
the double zeros on the unit circle and its complex conjugate zero into H2(z). The other pair on
the unit circle goes into H 2 (1/z).
Since H2(z) has its zeros on or inside the unit circle it is minimum phase ( - allow minimum
phase systems to have zeros on the unit circle in this problem). Since the zeros occur in complex
conjugate pairs, h 2 [n] is real.
(c)
where c = >'1+ 6i+d2~Y1 - 61 + 6a. Since 1 - 61 $ Ae{ei'"') :S 1 + 01 in the passband and -62 :5
A,(&w) S 62 in the stopband, we have
w E passband
w E stopband
Therefore,
M
H(eiw) = L h[n]e-;..n
.....
273
(M-1)/2 M
= L h[n]e-jwn + L h[n]e-jwn
n=0 n.=(M+l)/2
(M-1)/2 (M-1)/2
= L h[n]e-jwn + L h[M _ m],-jwM ,;wm
n=O m=O
(M-1)/2 (M-1)/2 ]
= .0-;wM/2 L h[n]eiw(M/2-n) + L h[n]e-jw(M/2-n)
[
- n=O
(M-1)/2
= ,-;wM/2
= ,-;wM/2
-
L
(JHl)/2
L
=l
2h[n]cosw(M/2- n)
2h[M;t' - n]cosw(n- ½)
Then
(M+l)/2
H(eJw) = ,-;wM/2 L b[n]cosw(n- 1/2)
¥ ¥
= 2l°"-
L.,b[n-l]cosw(n-½)+ 1°"- 1
2 L.,b[n]cosw(n- 2 )
n=l n=)
1- 1- M 1
+ b[O]cosw/2-
2 2b[+JcoswM/2
¥
= ~ L (bln] + ii{n - 1]) cosw(n - ½) + ½ii[O] cosw/2 - ½ii[ M,tl] coswM/
=l
(c) Consider
= W(w) [ii,(eiw)-tb[n]coswn]
Then
M/2
H(e'w) =,-;wM/2 L c(n] sinwn
where c(n] = 2jh(M /2 - n] for n = 1, ... , M /2.
If we follow a similar analysis as the one in part (b) we get
.Y-1 .Y-1 .Y.-1
sinw L c(n]coswn = ½ L
n=O
c(n]sinw(n + 1)-½ L
n=O
c(n]sinw(n-1)
=O
275
= 1'°
I:c[n -
2n=l l]sinwn- 1'° .
I:c!n]smwn
2n=l
+ ½c[O]sinw+ ½c[¥JsinwM/2
= ½ "t,
n=l
(c{n - 1) - c[n])sinwn + ½c[o] sinw + ½c[¥JsinwM/2
2c{O] - c[l)
n=l
2
c[n - 1) - c[n]
c(n) =
2
cf¥ - 1) n -- M
2
2
In a manner similar to that of part (c) we can find
M
L = --1
2
Hd(e'"')
sinw
W(w) = W(w) sinw
F = F
Type ry filters:
M
H(e'"') = L h[nJ,-'"'"
n=O
(M-1)/2 M
= L h[n],-i"'" + L h[n],-'"'"
A==O n=(M+l)/2
(M-1)/2 (M-1)/2
= L h[nJ,-i"'" - L h[m)e-,.,(M-m)
(M-1)/2
= 0 -J.,M/2 L h[n] ( ,-,.,(n-M/2) _ ,'"'(n-M/2))
n=O
(M-1)/2
= ,-,..M/2 L (-2j)h[n]sinw(n - M/2)
n=O
(M+l)/2
= •-iwM/2 L
..,., 2jh((M + 1)/2 - m]sinw(m - 1/2)
Then
(M+l)/2
H(e'"') L
=•-;.,M/> _, d(n) sinw(n - 1/2)
276
M-1
L :,
2
H•(<'"')
fl.(<'"') =
sinw/2
W(w) = W(w)sinw/2
F = F
(b) The filter length is 2L + l. The causal version of the fiow graph looks like
x{n]
' ..
0.5(1+z..,,)
~
_,
. ,
0.5(1+Z-.2)
z·'
•• ..__,
0.5(1+z-2) e---
~-1
.,.._
. .
(c) The fiow graph for B,(z) looks like
x{n]
The filter length is still 2L + l. The modified flow graph looks like
277
IC(n)
a,
. II o.5(1+z.,.l fr a,
'---+----l 0.5(1+z4 )
"o a,
!
278
.·!
Increasing
ao
a,
.,
8
0
0
1 cos(ro) _,
It
The picture above shows the mapping for a 0 somewhere between O and 1. The top right plot is
the mapping of
ms8 = ao + (1-ao)msw
We see that as ao increases, the transformation pushes the new passband further towards.-. The
new filter is not generally ao optimal filter since we lose ripples or alternations while keeping L
fixed. (Note that some of the original filter does not map anywhere in the new filter).
(f) In a similar manner, this choice of Qo will cause the new passband to decrease with deaeasing Oo·
7.54. (a) Let D,(z) be the z-transform of .ti.<•>{:r[nj}. Theo
(b) By taking the transform of both sides of the continuous-time differential equation one gets (assum-
ing initial rest conditions)
N M
L:O•••Y(s) = 2),s' X(s)
A:=O r=O
Solving for H,(s)
Similarly,
N M
:~:>•<• - ,-,i•Y(z) = Lb,(z -
.... r=O
z- 1 )' X(z)
Hd(z)
=> m(z) = z- z- 1
(c) First, map the continuous-time cutoff frequency into discrete-time and then make the sketch.
ejw - e-jw
0 =- -
j
- 2sin(w) =I
w=-.6
h1 [n] = h(-n]
H1(ei"') = H(e-;"')
= =
Since H(ei"') is symmetric about w 0, H(,-; .. ) H(ei"'). Thus, H 1 (ei"') H(,-;w) H(ei"'). = =
H(e 1"') is optimal in the minimax sense, so H1 (ei"') is optimal in miuirnax ..-.nse as well.
280
0$W$Wp
B,(ei"') ={ ~: w. :s :s
w ,r
h2(n] = (-lJAh(n]
=
(,-;•JAh(n]
H2(ei"') = H(ei<w+•l)
H2(ei"') is a high pass filter obtained by shifting H(ei"') by ,r along the frequency axis. H2(&"')
satisfies the alternation thereom, and is therefore optimal in the minimax sense.
0$ w :S ,r-w.
,r-wp:Sw:S1r
0:Sw $ ,r-w.
w-w,:Sw:S1r
(c) Using DTIT properties,
The filter h3(n] is the convolution of two length N sequences. Therefore, the length .of h,(n], de-
noted as N', is 2N - 1. Since N is either 11 or 13, N' must be either 21 or 25. It follows that
the polynomial order for h 3 (n], denoted as L', is either 10 or 12. For h 3 (n] to be optimal in the
minimax sense, it must have at least L' + 2 alternations. Tbns, hs(n] must abihit at least 12 alter•
nations, for the non-extraripple case, or at least 14 alternations in the extraripple case to be optimal.
A simple counting of the alternations in H3 (&"') reveals that there are 11 alternations, consisting
of the 8 alternations that were in H(ei") plus 3 where H(ei"') =
O. These are too few to satisfy
either the non-raripple case or the extraripple case. As a result, this filter is not optimal in the
minimax sense.
(d)
0$w$w,
w. :S w :S 1r
0$w$w,
w. :$ w::; 7f
(e) hs(n] is h(n] upsampled by a factor of 2. In tbe frequency domain, upsampling by a factor of 2 will
cause the frequency axis to get scaled by a factor of 1/2. Consequently, Hs(e'w) will be a bandstop
filter that satisfies the alternation theorem, with twice as many alternations as H (e1w). This filter
is optimal in the minimax sense.
0 $W $w,/2
w,/2 $ w $ .- -w,/2
.- -w,/2 $ w $ .-
0 :Sw :Sw,/2
w,/2$w$.--w,/2
.- -w,/2 $ w $ .-
7 .56. We have an odd length causal linear phase filter with values from n = 0, ... , 24. It must therefore be
either a Type I or Type ID filter.
(a) True. We know either
Type I Type ill
h(m] = h(24 - m] or h(m] = -h[24 - m]
for -oo < m < oo since the filter has linear phase. Substituting m = n + 12 we get
h[n + 12] = h[l2 - n] or h(n + 12] = -h[l2 - n]
(b) Fwe. Since the filter is linear phase it either has zeros both inside and outside the unit circle or
it has zeros only on the unit circle.
If the filter has zeros both inside a.nd outside the unit circle, its inverse has poles both. inside and
outside the unit circle. The only region of convergence that would correspond to a stable inverse
would be the ring that includes the unit circle. The inverse would therefore be two-sided and not
causal.
If the filter only has zeros on the unit circle, its inverse has poles on the unit circle and is therefore
unstable.
(c) Insufficient /nfonnatioTL If it is a Type ID filter it would have a zero at z =
-1 but if it is a Type
I filter this is not necessarily true.
(d) True. To minimize the maximum weighted approximation error is the goal of the Parks-McClellan
algorithm.
(e) True. The filter is Fm so there are no feedback paths in the signal flow graph.
(f) True. The filter has linear phase and
arg (H(e'W)] =fJ - 12w
where fJ = 0, ,r for a Type I filter or fJ = .- /2, 3r/2 for a Type ID filter. Tbe group delay is
(b)
W(w) = !· t~ (or
(or 1)
(or
f.)
f.)
~:: :~
·(or 1)
0$ lwl $ WJ
..,. :s lwl :s ""
w,:,jwj:S.-
L-1+6=L+5
(e) If M = 14, then L = M/2 = 1. The maximum number of alternations is therefore 7 + 5 = 12.
E(oo) Ii
0 .,
(f) ~ will be shown in part (g), the 3 band case can have maxima and minima in the transition
regions. It follows that we do not have to have an extremal frequency at w4 • Therefore, if we
started with an optimal maximal ripple filter and jnst slid "'• over we may move a local minimum
or maximum into the transition region, bnt there will still be enongh alternations left to satisfy
the alternation theorem. Thus, the maximum approximation error does not have to decrease.
(g) (i) If a point in the transition region has a local minimum or maximum then there is the possibility
that the surrounding points of nwcimum error do not alternate. Thus, we might lower the
number of alternations by two. Howefff, if...., started with L + 5 alternations this reduction
does not drop the number of alternations below the lower limit of L + 2 set by· the Alternation
Theorem. Therefore, local rnaxirna 'llld minima of A.(eiw) can occur in the transition regions.
Note that this is not true in the 2 band case.
(ii) If a point in the approximation bands is a local minimum or maximum, the surrounding points
of maximum error do not alternate. Thus, a local minin,um or maximum in the approximation
bands implies that the total number of alternations is reduced by two. However, if we started
with L + 5 alternations this rednction does not drop the number of alternations below the
lower limit of L + 2 set by the Alternation Theorem. Therefore, we can have a local maximum
or minimum in the approximation bands. Note that in the 2-band case we drop &om L + 3 to
L + 1 which violates the Alternation Theorem.
283
7.58. (a) In order for condition 3 to hold, G(z- 1 ) must be an allpass system, since
z- 1 = G(z- 1 )
,-,• = G(e-"")
= IG(e-jw)I e3LG(,-;")
Clearly, !G(e-Jw)I must equal unity to map the unit circle of the Z-plane onto the unit circle of
the z.plane.
(b) Consider one allpass term in the product, and note that o• is real.
O$ IZI < I
Or equivalently,
l<lz-11<00
Substituting the allpass term for z- 1 gives
The inside of the unit circle of the Z-plane maps to the outside of the unit circle of the z-plane.
This is not the desired result. However, if(! - af) > 0, then
z-lz•-1
I <
I
1 <
Jzi'
l•I < 1
The inside of the unit circle of the Z-plane maps to the inside of the unit circle of the z.plane. This
is the desired result. Thus, for condition 2 to be satisfied,
1- ol > o
Ja,1 2 < 1
lo•! < 1
This condition holds for the general case as well since the general case is just a product of the
simpler allpass terms.
I
284
(c) First, it is shown that G(z- 1 ) produces the desired mapping for some value of a. Starting with
G(z- 1 },
z-1 -a
z-1 =
1- az- 1
e-Jf,r,J - a
e-;,
= 1- ae-Jw
e-;, - ae-;•e-;w e-;.., - a
=
e-;.,(l + ae-;•) = e-jl +Q
e-;161
e-;• + a
= 1 + ae-;•
e-;, +a 1 +ae;•
= 1 +ae-;• 1 + a,;•
,-;• + 2a + a 2 ei'
= 1+2acos8+a2
Using Euler's formula,
Next, an equation for o is found in terms of e, and "'•· Starting with G(z- 1 ),
z-1-o
z-, = 1-oz-l
e-;w.,. - Q
1 - ae-iw,.
e-;,,. - ae-;•,.e-;"', = e-;w,. - Q
e-;,.,. - e-;w.,. = o(e-;<,,+...,l - 1)
e-;a,. - e-;w,.
0 = e-i(l,.+w,.) - 1
e-i(l,.+w,.)/2(e-i(l,.-'"',.)/2 _ ei<B,.-w,.)/2)
= e-i(l,.+1,o1,.)f2(e-i{B,.+w,.)/2 _ ei(B,.+w,.}/2)
(i}
2
-I [1- (-0.2679) ]
=
"'• tan 2(-0.2679)
-I ( 0.9282 )
= tan -0.5358
= 21r/3
(ii)
= tan-1 [I - (0)2]
"'• 2(0)
= tan-• (oo)
= ,r/2
(iii)
2
-I [1- (0.4142) ]
=
"'• tan 2(0.4142)
= tan-• (1)
= ,r/4
(e) The first-order allpass system
G(z- 1 ) = - •-• + 0
l+oz- 1
satisfies the criteria that the unit circle in the Z-plane maps to the unit circle in the z-plane, and
that e = 0 maps tow="· Next, o is found in terms of e, and "'•·
z-• =
286
e-;w,,. + Q
e-iB,,,
= I+ ae-;w,.
-e-;,, - oe-i(w,.+I) = e-Jw,,, + o
a(l + •-i(w,+B,l) = -e-;,.,, - e-jw,,.
e-il, + e-jw.,,
a = 1 + e-i(w,.+I,.)
e-j(w,,.+l,.)/2(e-iC-w,+l,.}/2 + e-j(w,.-1.. )/2)
= - e-i(w,.+l,.)/2(ei(w-,,+l,,J/2 + e-i(w,.+l,)/2)
cos[(w, -8,)/2]
= cos[(w, + 8,)/2]
z- 1 +a
z-• = 1 + o:z- 1
e-JW + Q
= 1 +ae-,w
-e-i8 _ ae-i(w+B) = e-jw +a
.-iw(l + oe-i') = -e-i' - a
e-il +o
= I+ ae-J'
e-i'+o l+oei 6
= 1 + oe-i1 1 + oei8
.-ii + 2a + ci'ei'
= 1 + 2ocos8 + a 2
cos8 - j sin8 + 2a + a 2 ccs8 + ja 2 sin8
= 1 + 2acos8 + a 2
cos8 + 2a + a2 cos8 + j(- sin8 + a 2 sin8)
= 1 + 2ocos8 + a 2
Therefore,
-w + 7r = tan-' [2o+(a (1 +a 2
l)sin8 ]
-
)cos8 2
w = (I-a2)sin8 ]+
tan-•[ 2a+(l+o )cos8 2
"
Note that this lowpass to highpass expression is the similar to the lowpass to lowpass expression
for w found in part (c). The only difference is the additive " term, which shifts the lowpass filter
into a highpass filter. The frequency warping is plotted below.
287
(i)
w, = tan
_1 [I -2(-0.2679)
(-0.2679)
2
]
+"
-1 ( 0.9282 )
= tan -0.5358 + ,r
= 2tr/3+tr
= Sr/3
The right edge of the low pass filter gets warped to 5,r /3, which is equivalent to -,r/3. The
frequency response of this filter appears below.
... . ..
-It -it/3 0 It (I)
(ii)
w,, = tan
_1 rl - (0)
2(0)
2
]
+ .-
= tan-• (oo) + .-
= ,r/2+tr
= 3,r/2
The right edge of the low pass filter gets warped to 3,r /2, which is eqar,aleot to -ir /2. The
frequency response of this filter appears below.
288
••• • ••
-7112 0 II (I)
(iii)
1 f1- (0.4142) 2 )
"'• = tan- L 2(0.4142) +"
= tan-• (~:::) + ,r
= ,r/4+,r
= 51'/4
The right edge of the low pass filter gets warped to 5,r/4, which is equivalent to -3,r/4. Th•
frequency response of this filter appears below.
••• •••
.
-II -31114 0 31114 II (I)
289
Solutions - Chapter 8
:i:(n) = L• "•ei'f•n
1:-=-9
We note that, in accordance with the discussion of Section 8.1, the sampled signal is represented
by the summation of harmonically-related complex exponentials. The fundamental frequency of
this set of exponentials is 2.-/N, where N = 6.
Therefore, the sequence :r[n] is periodic with period 6.
(b) For any bandlimited continuous-time signal, the Nyquist Criterion ma.y be stated from Eq. (4.14b)
as:
F,?. 2FN,
where F, is the sampling rate (Hz), and FN corresponds to the highest frequency component in
the signal (also Hz).
As evident by the finite Fourier series representation of :r,(t), this continuous-time signal is, indeed,
bandlirnited with a maximum frequency of Fn = ,:-,
Hz.
Therfore, by sampling at a rate of F, = ,o"-•Hz, the Nyquist Criterion is violated, and aliasing
results.
(c) We use the analysis equation of Eq. (8.11):
N-1
X[k) =L x[nJ,-i\Hn
n=O
X(k] = t (t
n.=O m=-9
a,,.ei'fmn) ,-;'f•n
= L• L• a,,.ei(2• /l)(m-•)n
A=Om=-9
We reverse the order of the summations, and use the orthogonality relationship from Example 8.1:
• 00
Taking the infinite summation to the outside, we recognize the convolution between a,,, and shifted
impulses (Recall a,,. =
0 for Jml > 9). Thus,
00
.X(k) =6 L "•-•·
=-oo
Note that from .X(k), the aliasing is apparent. ·
292
= (1+.-,.. (tl+,-jh("fl)X(k]
= { 3X(k/3], =
k 3l
0, otherwise
(b) Using N = 2 and i[n) as in Fig PS.2-1:
N-1
X[k) = L i(nJwtn
n=G
n=G
= i(O] + i(l]e-id
= 1+2(-1)•
= {3,
-1,
k=O
k =I
Observe, from Fig. PS.2-1, that i(n] is also periodic with period 3N = 6:
3N-1
i,(k] = L i(n]w;N
n=G
= L• i(nJ,-ii""
n=G
= (1 + ,-;'f• +,-,¥•)(1 + 2(-l)l)
= (1 + ,-;'f• + ,-;'f-•)X(k/3]
= ( ~3. ::~
0, le= 1,2,4,5.
293
8.3. =
(a) The DFS coefficients will be real if i[n] is even. Only signal B can be even (i.e., is[n] is[-n];
if the origin is selected as the midpoint of either the nonzero block, or the zero block).
(b) The DFS coefficients will be imaginary if x[n] is even. None of the sequences in Fig P8.3-l can be
odd.
(c) We use the analysis equation, Eq. (8.11) and the closed form expression for a geometric series.
Assuming unit amplitudes and discarding DFS points which are zero:
= (1 - ei.. ) l - e1:•
1- eJ'ik
= 0, k = ±2, ±4,
8.4. A periodic sequence is constructed from the sequence:
as follows:
00
= La"e-;..,n
..... l
= 1- oe-P,,' lal < l
N-1 oo
= L L z(n+rNJW}n
N-1 00
= L L on+rNu(n+rN]W}n
n=Or=-oo
N-1 oo
= L Lo"+rNw¾"
n=Or=O
Rearranging the summations gives:
oo N-1
X[k] = L o•N Lon win
r=O n=O
l _ 0 N e-.j2rlt)
= LarN
r=0
oo (
1 - oe-,'t,'-
, lal < 1
= _1_(1-a11
1- aN
e-i 2d:)
1 - a.-;'t,'-
, lal <1
1
X[k] = 1- oe-;(2.t/N) ' lal <1
(c) Comparing the results of part (a) and part (b):
X[k] = X(eiw)l..,=>••JN.
8.5. {a)
z[n] = o[n]
N-1
X[k] = L o[n]W}n, 0 ::, k ::, (N - 1)
n=O
= 1
(b)
z(n] = c5[n -no], 0 $no$ (N -1)
N-1
X(kj = En=O •In - noiwin, 0 $ k $ (N - 1)
= wt""
(c)
{ 1, n even
z[n] = 0, nodd
N-1
X(k] = E:1n1win,
n=O
0 $ k $ (N -1)
(N/2)-1
= E
n=O
w],""'
1- e-i2 •"
= l - e-;(sk/N)
{ N/2, k=0,N/2
X[k] = 0, otherwise
295
(d)
{ 1, 0 5 n 5 ((N/2) - 1)
:r:[n] = 0, N/2 5 n 5 (N - 1)
N-t
X[.1:J = I;:r:fn]W;", 0 5 k 5 (N - 1)
.....
(N/2)-1
=
....I:
l - e-id
W'"
N
= 1 _ e-i(2d)/N
{ N/2, k=0
X[k] = 1 _ e !2•k/N)' k odd
0, k even, 0 5 k 5 (N -1)
(e)
e1won 0:S:n:S:(N-1)
z[n] ={ '
0, otherwise
(a) The Fourier transform of z[n]:
00
X(e'"') = L z[n]e-iwn
n.=-oc
N-t
= L ~Wone-jwn
n=O
1 _ e-i{w-wo)N
X(e'"') = 1 - e-j(w-We)
= e-i(w-.,.)(N/2 ) (sill [(w -Wo)(N/2)])
e-i(w-wo)/2 sill [(w - Wo)/2]
X(e'"') = -i(w-.,.)((N-t)/2) (sinf(w-Wo)(N/2)])
e sin ((w - Wo)/2]
(b) N-polnt DFT:
N-t
X[k] = L :r:(n]wtn, 0 5 k S (N -1)
=•
296
N-1
= L e1wonw;;
n=O
1_ •-j((2••/N)-wo)N
= 1- e-j((2rk/N)-wo)
= -;('N'--.... )(¥)sin((~ -w,,) ~)
e sin[(~ -w,,) /2)
Note that X[k] = X(&w)lw=(2d)/N
(c) Suppose Wo = (2do)/N, where ko is an integer:
1- e-i(A:-to)2'r
X[k] = 1_ e-i(•-.. )(2•)/N
8. 7. We have a six-point uniform sequence, :[n], which is nonzero for O :S n :S 5. We sample the Z-transform
of :[n] at four equally-spaced points on the unit circle.
X[k] = X(z)l,=,<>••J•>
We seek the sequence : 1 [n] which is the inverse DFT of X[k]. Recall the definition of the Z-transform: 1
00
X(z) = L :[n]z-n
n=-00
Since x[n] is zero for all n outside O :S n S 5, we may replace the infinite summation with a finite
summation. Funhermore, after substituting z = ei( 2 rk/ 4 ), we obtain
X[k]
•
= L :[n]W;", 0 :S k S 4
n=O
Note that we have taken a 4-point OFT, as specified by the sampling of the Z-transform; however, the
original sequence was of length 6. As a result, we can expect some aliasing when we return to the time
domain via the inverse DFT.
Performing the DFT,
Taking the inverse DFT by inspection, we note that there are six impulses (one for each value of n
above). However,
W4•• -- 0
"'"
"'• and w•• - w•
4. - 4'
so two points are aliased. The resulting time-domain signal is
2 2
:i[n]
.l l I I .
-1 0 1
1
2
1
3 4
•5 •6 n
297
X(e'"') = L z[n],-,..,n
n=-00
1
=
Now, sample the frequency spectra of z[n]:
Y[k] = X(e'w)lw=2d/10, 0$ k$ 9
=
•
LY[nJw,•;
n=O
Recall:
(a) We wish to obtain X(eiw)lw=h/• using the smallest DFT possible. A possible size of the DFT is
evident by the periodicity of eJWj..,=.,/•· Suppose we choose the size of the OFT to be M = S.
The data sequence is 20 points long, so we use the time-aliasing technique derived in the previous
problem. Specifically, we alias z[n] as:
00
This aliased version of z[n] is periodic with period 5 now. The 5-pt DFT is computed. The desired
value occun at a frequency corresponding to:
2.-k 4,r
N=s
For N = s, k = 2, so the desired value may be obtained as X[kJl•= 2 •
(b) Next, we wish to obtain X(e'"')lw=10./2?·
The smallest DFT is of size L = 27. Since the DFT is larger than the data block size, we pad z[n]
with 7 zeros as follows:
z[n], 0 :, n $ 19
%2 InI
{ 0,= 20:,n:,26
We take the 27-pt DFT, and the desired value corresponds to X[k] evaluated at k = 5.
298
8.10. From Fig PS.10-1, the two 8-pt sequences are related through a circular shift. Specifically,
8.11. We wish to perform the circular convolution between two 6-pt sequences. Since o:,[n] is just a shifted
impulse, the circular-convolution coincides with a circular shift of o: 1 [n] by two points.
y[n] = o:i[n]®:t,[nj
= o: 1 [n](Q),[n - 2]
= o: 1 [((n - 2)),]
4 y[n]
-1 0 I 2 3
fI
4 5
I
6
I
7
n
8.12. (a)
transforms to
3
X[k] = L cos(";'Jwf", o S k S 3
.....
The cosine term contributes only two non-zero values to the summation, giving:
X[k] = 1 - ,-i•k, 0S k S3
= 1-wr
(b)
h[n] = 2", 0S nS 3
3
H[k]. = L2"Wf", 0SkS3
.....
= 1 + 2wt + 4Wf' + BWt°
299
(c) Remember, circular convolution equals linear convolution plus aliasing. We need N ~ 3+4-1 = 6
to avoid aliasing. Since N = 4,we expect to get aliasing here. First, find 11[n] = z[n] • h[n]:
6
y[n] = z[n] • h[n]
3
2
l
n
-1 0 l 2 3 4 5 6 7
-4
-8
For this problem, aliasing means the last three points (n = 4, 5, 6) will wrap-around on top of the
first three points, giving y[n] = z[n@,[n]:
6
y[n] = z[nJ©h[n]
3
n
-1 0 l 2 3 4 5
-3
-6
(d) Using the DFT values we calculated in parts (a) and (b):
Y[k] = X[k]H[k]
= l + 2Wf + 4W;• + BW]k - w;• - 2w]• - 4Wt° - 8W;'
Since Wt' = ff1• and W;' = Wf
Y[k] = -3 - 6Wf + 3W;• + sw:•, 0 :S: k :S: 3
8.13. Using the properties of the DFT, we get y(n] = z[((n - 2)).J, that is y[n] is equal to z[n] circularly
shifted by 2. We get:
2 2
y[n]
n
012345
8.14. :r,[n] is the linear convolution of z 1(n] and :z:2(n] time-aliased to N = 8. Carrying out the 8--point
circular convolution, we get:
9 9 9 9
:r3(n]
8 8
7 7
n
0 1 2 3 4 5 6 7 8
l 24+ 1 a 2+a
y(n]
•
-1 0 l
j j j j
2 3
•4 •5 n
=
Matching the above ,equence to the one given, we get a -1, which is UDique.
8.16. Xi[k] is the 4-point DFI' of :r[n] and z,[n] is the 4-point inverse DFI' of X 1[k], therefore :r1(n] is :r[n]
time alia.ed to N = =
4. In other words, z 1[n] is one period of :i:(n] z[((n)),]. We thus have:
4 = b+ 1.
Therefore, b = 3. This is clearly UDique.
8.17. Loo~ at the ,equences, we see that z, [n] • z2(n] is non-zero for 1 $ n $ 8. The smallest N such that
.:r1[n] ~ 2[n] = :ri[n] • :r2[n] is therefore N = 9.
301
8.18. Taking the inverse DFT of X 1 [k] and using the properties of the DFT, we get:
= z[((n - m)),].
zi[n]
m = 2 works, clearly this choice is not unique, any m = 2 + 61 , where I is an integer, would work.
8.20.
X, [k] = X[k]e+j(2d2/NI.
Y,[k] = X1[k]X2[k]
From the discU5Sion of Section 8.2.5, y[n] is the result of the periodic convolution between f1[n]
and i 2 [n).
N-1
!i,[n] = L f1[m]f [n - m]
m=O
2
Since f 2 [n] is a periodic impulse, shifted by two, the resultant sequence will be a shlfted (by two).
replica of i ,In].
6 6
Using the analysis equation of Eq. (8.11), we may rigoroU5ly derive y,[n]:
X,[k] = L• f,[n]W;"
n=<>
302
L• :i:2[n)Wf"
.X2[k] =
....
= w;>
Y1[k] = .X1[k).X2[k]
= 6W;' + 5W.f1 + 4W/• + 3Wf• + 2'111• + w;•
Noting that w;• = ei'f<n) = 1 = W¥•, we use the synthesis equation of Eq. (8.12) to construct
!i,[n]. The result is identical to the sequence depicted above.
(b) The DFS of the signal illustrated in Fig. PS.21-2.is given by:
.Xs[k] = L• :i:a[n]W;"
.....
= 1+wt•
Therefore:
Y2[k) = .X,[k].X.[k]
= X1[k] + Wt' .X,[k)
Since the DFS is linear, the inverse DFS of Y2[k) is given by:
!i2[nj = :i:,[n] + :i:1[n - 4).
8.22. For a finite-length sequence z[n], with length equal to N, the periodic repetition of z[-n] is represented
by
z[((-n))N) =z[((-n+tN))N), l: integer
where the right side is justified since z[n] (and z[-n]) is periodic with period N.
The above statement holds true for any choice oft. Therefore, for l l: =
:r'[n) = { :r[n], 0 $ n $ (P - 1)
0, P:,n:,N
I
f
303
(b) Suppose N > P, consider taking a DFT which is smaller than.the data block. Of course, some
aliasing is expected. Perhaps we could introduce time aliasing to offset the effects.
Consider the N-pt inverse DFT of X[k],
l ,,_,
z[n] = N L
X[kJW,."", 0 :Sn :S (N - 1)
lo=O
Suppose X[k] was obtained as the result of an infinite summation of complex exponents:
Rearrange to get:
z[n] = L
00
m=-oo
z[m]
(
! L ,-;(2,/N)(m-n)i
,,_,
k=O
)
So, we should alias z[nj as above. Then we take the N-pt DFT to get X[k].
8.24. No. Recall that the DFT merely samples the frequency spectra. Therefore, the fact the /m{X[k]} 0 =
for 0 :S k :S (N - l) does not guarantee that the imaginary part of the continuous frequency spectra is
also zero.
For exan.ple, consider a signal which consists of an impulse centered at n = I.
z[n] ;,, o[n - l], 0 :Sn :S l
The Fourier transform is:
X(eiw) = ,-;w
Re{X(eiw)} = cos(w)
/m{X(eiw)} = -sin(w)
Note that neither is zero for all 0 :S w :S 2. Now, suppose we take the 2-pt DFT:
X[k] = W,f,
0 :S k :S l
= { 1, k=0
-1, k=l
We use the technique suggested in problem 8.28. That is, we temporarily extend the sequence such that
a periodic sequence with period 4 is formed.
i(n]
n
-4 -3 -2 -1 0 1 2 3 4 5 6 7
Now, we shift by three (to the right), and set all values outside OS n S 3 to zero.
y[n]
• •
-2 -1 0 1 2 3
•4 •5 n
8.26. (a) When multiplying the DIT of a sequence by a complex exponential, the time-domain signal
undergoes a circular shift.
For this case,
Y(k] Wt' X(k], =
OSkS5
Therefore,
y(n] = z[((n - 4))6], 0'.5n'.5S
4
3
I
2 y(n]
I
1
I• •
•
-1 0
'
l
•2 •3 4 5 6 7
n
.
(b) There are two ways to approach this problem. First, we attempt a solution by brute force.
Notice that
w~ = e-·j(2rt/N}
w-•
N = ~(2d/N) = e-;(2r/N){N-•) = w::-•
W[k] = 4 + ~2[w.•
6 + u,!-•J + [w. + w•-
6
2
6• 6
2
•] + !2[w,.
6 + w•-,.]
6'
So,
Sketching w[n):
4
II
w[n]
3/2 l 1 l 3/2
• t t t T • • n
-1 0 1 2 3 4 5 6 7
As an alternate approach, suppose we use the properties of the DIT as listed in Table 8.2.
W(k] = Xe{X[k)}
X[k] + X•[k)
= 2
w[n] = ~ IDFT{X[k)} + ~ IDFT{X.[k)}
= ~ ( :r[n] + x·[((-n))N])
For O :5 n :5 N - 1 and :r[n) real:
4
3
I• •
2 :r[N - n], for N =6
II•
1
• t •7 n
-1 0 1 2 3 4 5 6
.l f t ...
-1 0 1 2 3 4 5
q(n]
8.27. (a) The linear convolution, z 1 (n] • z 2 [n] is a sequence of length 100 + 10 - 1 = 109.
10 10
9 9
8 8
7 7 zi(n] • z,[n]
6 . 6
3
4 ,.
5 ' 5
4
3
2 2
1 1
T
-1 0 ' l 2 3 4 5 6 7 8 9
....l..---''--J..._..J...-1--'-..J........L-''--'--+- n
99 100 101 102 103 104 105 106 107 108 109
(b) The circular convolution, z 1 (n] ~ 2 [n], can be obtained by aliasing the first 9 points of the linear
convolution above:
10 10 10 10 10 10 10 10 10 10 10
llllllllll L"
01234 56789
(c) Since N? 109, the circular convolution zi[n]@,[n] will be equivalent to the linear convolution
99
of part (a).
8.28. We may approach this problem in two ways. First, the notion of modulo arithmetic may be simplified
if we utilize the implied periodic extension. That is, we redraw the original signal as if it were periodic
with period N = 4. A few periods are sufficient:
f j j f jjj i
-4 -3 -2 -1 0 l 2 3
jj i j
4 5 6 7
i(n]
n
307
To obtain z 1[n] = z[((n - 2)) 4], we shift by two (to the right) and only keep those points which lie in
the original domain of the signal (i.e. 0 :5 n :5 3):
• •
-2 -1 0
fi l
li
2
• •
3 4 5
z,[n]
To obtain :r2[n] = z[((-n)) 4 ], we fold the pseudo-periodic version of :r[n] over the origin (time-reversal),
and again we set all points outside O :5 n :5 3 equal to zero. Hence,
• •
-2 -1 0
li i i l 2 3
I
4 5
I
:r,[n]
8.29. Circular convolution equals linear convolution plus aliasing. First, we find y[n] =:r1[n] • :r2[n]:
8
6 ' 6
. 5
• 4 • 4 y[n]
• 3 3
2
'
l
T
01234567891011 ' - - n
(a) For N = 6, the last four non-zero point (6 :5 n :5 9) will alias to the first four points, giving us
Y1 [n] = z, [nJ(§):2 [n]
308
8
10 " 6
• 8
6 " 11,(n]
4
'
'
-
.
-1 0 l 2 3 4 5 6 7
n
(b) For N = 10, N ;:: 6 + 5 - l, so no aliasing occurs, and circular convolution is identical to linear
convolution.
8.30. We have a finite length sequence, whose 64-pt DFT contains only one nonzero point (for k = 32).
(a) Using the synthesis equation Eq. (8.68):
l ,,_,
:r[n] =N L X[k]Wi,"", 0 Sn S (N - l)
A:=O
Substitution yields:
32
:r[n] = !x[32]W64 "
= .!..,.;i.(32)n
64
= _!_e-71rn
64
:r[n] = !<-1)", 0SnS(N-1)
The answer is unique because we have ta.ken the 64-pt DFT of a 64-pt sequence.
(b) The sequence length is now N = 192.
:r(n]
:r[n] =
64 Sn S 191
This solution is not unique. By taking only 64 spectral samples, z(n] will be aliased in tinle.
As an alternate sequence, consider
8.31. We have a 10-point sequence, :,;[n]. We want a modified sequence, :,;1 (n], such that the 10-pt. DFT of
zi[n] corresponds to
X;(k] = X(z)l,=½•'",_,,,o>+<•t>•ll
Recall the definition of the Z--transform of z(n]:
00
X(z) = L z[n]z-•
n.=-00
309
Therefore,
X(z) = L• z[n]z-•
n=O
Substituting in z = ½eil<•••/10)+(•/IO)),
We seek the signal z 1 {n], whose 10-pt. DFT is equivalent to the above expression. Recall the analysis
equation for the DFT:
X 1 [k)
•
= :[;zi{n)Wt.;', 0$ k$ 9
n=<I
= :[;:i:[n]Wf;"
n=O
7
Y{k] = L z{n]Wt°, 0 $ k $ 15
n=O
Therefore, the 16-pt. DFT of the interpolated signal contains two copies of the 8-pt. DFT of z{n]. This
is expected since Y{k] is now periodic with period 8 (see problem 8.1). Therefore, the correct choice is
C.
As a quick check, Y[0] = X{0].
t
310
Scale by -1
Shift by N
(b) To obtain X[k] from X,ik], we might try to take the inverse DFT (2N-pt) of X,ik], then take the
N-pt DFT of :r,ln] to get X[k].
However, the above approach is highly inefficient. A more reasonable approach may be achieved if
we examine the DFT analysis equations involved. First,
2N-1
X,[k] = L :ri[n]W;';, 0 :S k :S (2N - 1)
n=O
N-1
= L :r[n]W;';)
,.,_,
=O
= I: :r[nJwt12>n, 0 :S k :S (N - 1)
=O
X1 [k] = X(k/2], 0 :S k :S (N - 1)
Thus, an easier way to obtain X[k] from X 1 [k] is simply to decimate X 1 (k] by two.
~ :z:[-n].,,._ ~ :z:[n]w:•n
= L.J -2-"2N-l + L.J 2 2N-l
n=-N+l n=O
Letm= -n,
N-1 [ ] N-1 [ ]
X.[k] = ~ !!'_w,-1:n
~ 2 2N-1
+~ ~w,kn
L.J 2 2N-1
n=O n.=O
N-1
X[k] =L :z:[n]Wt", 0 5 k 5 (N -1)
n=O
and
Re{X[k]} = N-1 (2 k )
~ :z:[n]cos ';.,, n
X[k] + X'[k]
Re{X[k]} = 2
l N-1 l N-1
= 2L :z:(n]Wt" + 2L :z:(n]WN""
n=O n=O
N-1
= ½ L (:z:[n] + :z:[N - n])Wt"
n=O
So,
Re{X[k]} =DFT { ½(:z:[n] + :z:[N - n])}
8.35. From condition 1, we can determine that the sequence is of finite length (N = 5). Given:
X(~"') = 1 + A1 cosw + A2 cos2w
= 1 + A1 (~" + e-'"') + A2 (e'..., + e-i'"')
2 2
From the Fourier analysis equation, we can see by m•khing terms that:
Condition 2 yields one of the values for the amplitude constants of condition l. Since z[n] • cS[n - 3] =
:r[n - 3] = = = =
5 for n 2, we know z[-1] 5, and also that z[l] z[-1] 5. Knowing both these values =
=
tells us that A, 10.
For condition 3, we perform a circular convolution between z[((n-3)),] and w[n], a three-point sequence.
For this case, linear convolution is the same as circular convolution since N = 8 ?: 6 + 3 - l.
We know z[((n - 3)) 8 ] = z[n - 3], and convolving this with w[n] from Fig PS.35-1 gives:
22
A2 + 15
2 + 11
lli
5+A2
1> + 13
lli
2
1>
n
0 1 2 3 4 5 6 7 8
For n = 2, w[n] • z[n - 3] = 11 so A2 = 6. Thus, z[2] = z[-2] = 3, and we have fully specified z[n]:
5 5
l' lI
3 3
z[n]
•
-3
I
-2 -1
1
0 1 2
•3 n
Y[k] = X 2 [k]
= 2 + 2Wt +2wff•
+2W{ + W;"+ W:'
+ 2w;• +wt•+ wt•. o :s .1: :s s
(a) By inspection,
(b) This procedure performs the autocorrelation of a real sequence. Using tbe properties of tbe DFT,
an alternative method may be achieved with convolution:
8.37. (a)
gi[n] = z[N - 1 - n], 0 :Sn :S (N - 1)
N-1
G1 [k] = L z(N - 1- n]Wt", 0 :S k $ (N - 1)
n=O
Letm=N-1-n,
N-1
G1(k] = L z[m]W~(/'1-l-m)
m=O
N-1
= w~(N-1) L z[m]w,v•m
m=O
N-1
G1[k] = ei(2d//'I) L z[m]ei(2dm//'I)
m=O
= ei(>d/1'1 X(ei")lwz(b>/1'1)
G1(k] = H1[k]
(b)
N-1
G2[k] = L (-l)"z[n]W;', 0:, k :, (N - 1)
n=O
N-1
= L z[nJwtfl•wt·
n=O
N-1
= I: x1n1w:.,,+ti•
n=O
= X(ei")l.,=b(Hf)/N
q,[k] = He[k]
314
(c)
2N-1
G3(k] :E xfnJw.~.
= ..,.., 0 $ k $ (N - 1)
N-1 2N-l
L z[n]w;~ + L z[n -
= ..,.., N]Wfiv
=N
N-1 N-1
= L z[n]Wfiv + L z[mJw;im+N)
n=O m=O
N-1
G4 [k] = O $ k :S (N - 1)
N/2-1 N/2-1
= L
n=O
z[n]W,t12 + L
n=O
z[n + N/2]W,t12
N/2-1 N-1
= L z[n]W,t12 + L z[m]W!j';-Nl•l
A=O m=N/2
N-1
= L z[n]w}tn
n=O
= X(e''"')lw=(4rt/N)
G4 [k] = Ha[k]
(e)
2/11-1
G.[k] = I: :i:[nJwm, O :S k $ (N -1)
n=O
N-1
= L x[n]W,/'N
....0
= X(eiw)lw=(d/N)
G.(k] = H2[k]
(f)
2N-l
G6 [k] = L :r[n/2)WfN, 0 :S k $ (N - 1)
n:0
N-l
= L :r[n]W~n
n=O
= X(eiw)lw=(2d/N)
G&[k] = H,[k]
(g)
f-1
G1[k] = L x[2n]WM2, 0 :S k $ (N - 1)
n=O
The Fourier transform of ::r[n] displays generalized linear phase (see Section 5.7.2). This implies that for
::r[n] ,; 0, 0 S n S (N - 1):
::r[n] = ::r[N - 1 - n]
For N = 10,
::r[O] = :z:[9]
::r[l] = :z:[8]
:z:[2] = :z:[7]
8.39. We have two 100-pt sequences which are nonzero for the interval OS n S 99.
If x, [n] is nonzero for 10 S n S 39 only, the linear convolution
::r,[n] • :z:2[n]
is a sequence of length 40 + 100 - 1 = 139, which is nonzero for the range 10 Sn S 139.
A 100-pt circular convolution is equivalent to the linear convolution with the first 40 points aliased by
the values in the range 100 S n S 139.
Therefore, the 100-pt circular convolution will be equivalent to the linear convolution only in the range
40 Sn S 99.
8.40. (a) Since ::r[n] is 50 points long, and h[n] is 10 points long, the linear convolution y[n] = :r[n] • h[n]
=
must be 50 + 10 - 1 59 pts long.
(b) Circular convolution= linear convolutin + aliasing.
If we let y[n] = ::rjn] • h[n], a more mathematical statement of the above is given by
00
For N = 50,
::r[n]@.[n] =11[;.] + y[n + 50],
We are given: :z:[n]@).[n] =10
Hence,
11[n] + 11[n + 50] = 10, 0 S n S 49
=
Also, y[n] 5, 0 S n S 4.
Using the above information:
317
n=49 11(49) = 10
To conclude, we can determine 11[n) for 9 S n S 55 only. (Note that 11[n) for O S n S 4 is given.)
8.41. We have
A B
_ _.__ _ _.__ _ _ _ __.__ _ _~ - - n
0 9 30 39
[1
10 19
n
B•C
_ _..__ _ _ _ _....,__ _ _ _.,___ _ _ _ _....,___ n
10 28 40 58
Thus, z[n] • y[n) = w(n] is nonzero for 10 Sn S 28 and 40 Sn S 58.
318
(b) The 40-pt circluar convolution can be obtained by aliasing the linear convolution. Specifically, we
alias the points in the range 40 $ n $ 58 to the range O $ n $ 18.
Since w(n] = z(n] • y(n] is zero for O $ n $ 9, the circular convolution gin] = z(n]@\,(n] consists
of ooly the {aliased) wlues:
Also, the points of g(n] for 18 $ n $ 39 will be equiwlent to the points of w(n] in this range.
To conclude,
w(n] = g(n], 18:::,n:$39
w[n +40] - g(n], 0 $ n :$ 9
8.42. (a) The two sequences are related by the circular shift:
and
IH,[k]I = 1w,-•• H,[k]t = IH,(k]I
So, yes the magnitudes of the S-pt OFTs are equal.
(b) h,(n] is nearly like (sinx)/x.
Since H 2 (k] = .,,. H,(k]. h,(n] is a better lowpass filter.
8.43. (a) Overlap add:
If we divide the input into sections of length L, each section will have an output length:
L + 100 - 1=L + 99
Thus, the required length is
L = 256 - 99 = 157
=
If we bad 63 sections, 63 x 157 9891, there will be a remainder of 109 points. Hence, we must
pad the remaining data to 256 and use another OFT.
Therefore, we require 64 OFTs and 64 IDFTs. Since h(n] also requires a OFT, the total:
(c) Ignoring the transients at the beginning and end of the direct convolution, each output point
requires 100 multiplies and 99 adds.
overlap add:
# mult = 129(1024) = 132096
# add = 129(2048) = 264192
319
overlap save:
# mult = 131(1024) = 134144
# add = 131(2048) = 268288
direct convolution:
# mult = 100(10000) = 1000000
# add = 99(10000) = 990000
= L ri[n](-l}n
=<>
4
= 3
4 4
Q[k] = 3'5[k] + o[k -
3 3].
= !(~ + ~(-l}n)
6 3 3
= ~(l
9
+ (-1r)
l. I.I.
012345
n
320
8.45. We have:
Then:
1 •
= 7LX2[k]
lo=O
1 •
= 7 L(Re{X2[k]} + jlm{X2[k]})
lo=O
1 •
=
7 lo=O
LRe{X2[k]} , since x2[0] is real.
= g[O].
To determine the relationship between x2 [1] and g[lj, we first note that since x,[n] is real:
x(ei"'J = x·c,-'"'J.
Therefore:
X[k] = X"[N -k], k = 0, ... ,6.
\Ve thus have:
1 •
g[l] =
....
7 E&{X2[k]}W7-•
! t X2[k] + X2[klw:-•
=
=
7 >=O
7 2
. 7
l:=O
+7
2
2
[
7
!LX2klw-• !LX.[N-klw-•
7
.
k=O
1 1 •
= 2%2[1] + 14 L X2[k]W,'
•=0
•
~x2[l] + 1~ L X2[kJw,-u
=
•=0
1
= 2(x2[l] + z2[6])
1
= 2 (x2[l] + OJ
1
= 2x2[l].
8.46. (i) This corresponds to z,[n] = x,[(( ..:n))N], where N = 5. Note that this is only true for z 2[n].
(ii) X,(e'"') has linear phase corresponds to z,(n] having some internal symmetry, this is only true for
x,[n].
321
(ill) The DFT has linear phase corresponds to z,(n) (the periodic sequence obtained from z,(n]) being
symmetric, this is true for z 1 [n] and x 2 [n] only.
8.47. (a)
(b)
=
=•
L x[n]z-•J , '•~•·,
..... ;i:=2e'
=
=•
L x(n)(2eii )-•,-i¥•
.....
=
=•
L v(n],-i'i'•.
n=O
(c)
3
w(n] = ¼L W[kJw.-•·
•=<>
3
= ¼L(X(k] + X(k + 4])e+,'f••
l:=O
= ¼L
3
.t-d)
X[k]e+,'f•• + 1L X(k + 3
l.=O
4]e+,'f••
X[k]e+;'f•• + 1
3 7
= ¼L L X(k]e+;'f••
1-=0 l:=4
7
= ! L X(k]e+;'f>2n
4 >=O
= 2x[2n).
8.48. (a) No. x[n] only has N degrees of freedom and we have M ~ N constraints which can only be satisfied
if r[n] = 0. Specifically, we want
l M-1
x[n] =M
....L X[k]W;," , n = 0, ... , M - 1.
Where X[k] is the M-point DFT of :r[n], since X[k] = 0, we thus conclude that r[n] = 0, and
therefore the answer is NO.
(b) Here, we only need to make sure that when time-aliased to M samples, :r[n] is all zeros. For
example, let
:i:[n] = o[n] - o[n - 2]
then,
X(ei\'- 0 ) = I - I= 0
X(ei\'- 1
) = I - I= 0
8.49. :i: 2 [n] is r,[n] time aliased to have only N samples. Since
We get:
n= O, ... ,N-1
:i:,[n] ={ otherwise
8
Xa[k]
4 4
k
0 l 2 3 4 5 6 7
-2 -2
16
v,.[k]
8 8
k
0 l 2 3 4 5 6 7 8 9 10 11 12 13 14 15
-4 -4
(c)
8
IX,.(k]I
,;,
4 4
2 2
Q Q Q
Q Q Q Q :
•. k
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
!L
N-1
n=O
eJi;(t-r)n ={ l,
k- r
O, otherwise
= mN, m: integer
N n=0 N n=<>
= 1 , for k - r = mN
(b)
N-1 '"-IN
l ""' -~in_ 1-•'"
N L..., e' - l - eiif 1
n=O
1-ei••<•-•l
1 - e" ~<•-•>
= 0
Note that the denominator is nonzero, while the numerator will always be zero for k - r # mN.
8.52. (a) We know from Eq. (8.11) that if £ 1 [n] = i[n - m], we have:
N-1
Xi[k] = I>=[n - m)W,tn
.....
If we substitute r =n - m into this equation, we get:
r=-m
N-1-m
= w,tm I: xr,1wt·
r=-m
Using the fact that i[r] and w,t• are periodic with period N:
-1 -1
L i[rJwt· = L i[r + NJwt<•+N)
r=-m r=-m
Substituting l =r + N
N-1
L i[l]W,t'
r=-m l=N-m
X1[k] = w,tm f I:
LN-m
i[r)Wff + Nf:- i[r]Wt']
r=O
1
N-1
= w,tm L i[r)Wff
r=D
= W;;"X[k]
Hence, if :ii[n] = i[n - m), then X1[k] = Wt"' X[k).
8.53. (a) 1. The DFS of i"[n) is given by:
N-1
N
I:x"[n]Wtn = ~ i[n]W,."
l )"
(
n.=0
= X"[-k]
326
2. The DFS of :i:'[-n]:
N-1
I;x'[-n]W~"
n=O
= (=t+t :i:[l]W;/) •
= X'[k]
3. The DFS of &{:i:[n]}:
So,
X[k] = X'[-k]
X[-k] = X'[k]
X'[k] + X[-k]
&{X[k]} = 2
= &{X[-k]}
X(el"') = L f[n],-'"'"
=-00
X(elw) =};
00
(! %X(k]wN••) ,-,.,.
Rearranging the summations and combining terms:
0 $ n $ (N -1)
w[n] = { l,
0, otherwise
The window has a Fourier transform:
00
W(e-1"') = L w[n]e-;"'n
--00
N-1
= L e-;wn
n=O
1 - e-;wN
= 1 - ,-;., = .-,'f ei'f - ,-;t
= ,-;.. (¥)sin(w1)
sin(~)
328
=
3. Since z[n] :r[n]w[n], the Fourier transform of z[n] can be represented by the periodic convolution
(see Eq. (8.28)).
Integration over -.- :,; fJ :,; r reduces to the summation (note the impulse train):
Hence, the Fourier transform is obtained from the DFS via an interpolation formula.
=
(a) Suppose z[n] -z[N -1- n]. For N even, all elements of z[n] will cancel with an antisymmetric
component. For Nodd, all elements have a counterpart with opposite sign. However, z[(N -1)/2]
must also be zero.
=
Therefore, for z[n] -z[N -1 - n], X[O] 0. =
(b) Suppose z[n] = z[N - 1 - n] and N even.
N-1
X[N/2] = L z[nJWJ:'121n
n=O
N-1
= L z[n](-l)n
n=O
= z[O] - z[l] + z[2) - z[3) + · · · + z[N - 2] - z[N - l]
Therefore, X[N/2)= 0.
:r,,[n] = 1
(:r[((nllN] +:r"[((-n))N]), 0:,; n:,; (N - 1)
329
Note that:
z[((n))N] = z[n], 0 $ n $ (N - 1)
z"[((-n))N] = z"[-n + N] + z"[O]o[n] - z"[O]o[n - N]
Since,
andz,[n-N] = ~(z[n-N]+z"[N-nJ)
So,
For O :5 n :5 (N - 1):
0, otherwise
To conclude:
z,p[n], O $n $ N/2
x,,[n]
n= N/2
:i:,[n] = 2 '
x;,.[-n], -N/2<n$-l
x;,.[-n]
n = -N/2
2
8.57.
N-1 N-1
L jz[n]l = L :i:[n]z*[n]
2
n=O n.=O
Hence,
substituting:
z*(n] =
.
! L..,
N-1
X*(k]W~"
331
= ! I: l:=O
X'[k] ('f n.:=O
z[n]Wff)
= !L N-1
l=O
X'[k]X[k]
N-1 N-l
L lz[n]l 2
= ..!_
N
L IX[k]l
l=O
2
n=O
X[k] = X(e'"')L=•••/N
= B(2;k) e>!2•/N)to
A[k] = B(2;k)
2.-ac
'Y = N
(b} This statement is FALSE:
Suppose z[n] = o[n] + ½•[n - l],
X[k]
= A[kje''•,
X[k]
1 •
A[k] = 1 + -(-1)
2
and -y = 0
The Fourier transform of :r[n] is X(e'"') = 1 + ½ei"', which cannot be expressed in the form
X(e-'"') = B(w)& 0 "'.
8.59. We desire 128 samples of X(&"')Y(e'"').
Since z[n] and y[n] are 256 points long, the linear convolution, :r[n] • y[n], will be 512 points long.
We are given a 128-pt DFT only. Therefore, we must time-alias to get 128 samples. The most efficient
impiementation is:
:r[n] --+I I
m JV 1---R,[k]
y[n] --+I I
and
v[n] = v[((n + l))N] - by[((n - l))N]
Because the shift is circular, the points at n 0 and n = = (N - 1) will not be correct. Therefore, only
the points in the range l $ n $ (N - 2) are valid.
8.61. (a)
N-1
X[k] = L :r[n]e-i(>r/N)tn, 0 $ k $ (N -1)
n=O
N-1
X.,[k] = L :r[n]e-;(2.t/N+r/N)n
n=O
N-1
= E z[nje-j(,rn/N)e-J(2,r/N)l:n
n=O
n=O
N-1
= L :r[n]ei(rn/N),i(2r/N).,.
. n=O
= X_it[k]
So,
X.,[k] = X_it[N - (k + l)] , for O $ k $ (N - 1) and :r[n] real.
333
(c) (i) N-k-1 is odd when k is even. H R[k] = XM[2k], we may obtain XM[k] from R[k] as follows:
G { R[J:/2], k even
[k] = R•[(N - (k + 1))/2], k odd
where we note that
R•[(N - (k + 1))/2] = Xi,[N - (k + l)]
fork odd.
(ii)
R[k] = XM[2k]
(N/2)-1
= L z[n]e-;(4d/N+•/N)n
n=O
N-1
= L z[n]e-,<•n/N),-i('Mf)
(N/2)-1 N-1
= L z[n],-,c•n/Nle -;( '..ffl' )·+ L x[n],-i<•n/N)e -i('..ffl')
n=O n=N/2
(N/2)-1
= L (:i:[n],-i(•n/N) +z[n+N/2],-i(•n/Nl,-,1•/21),-;('Mf)
n=O
So,
r[n] = (x[n] - jx(n + N /2]),-iC•n/N), 0<- n <- (N2 -1)
(d)
X3.,(k] = X1M[k]X2M[k]
N-1
X 3.,[n] = L z 1.,[r]x2.,[((n - r))N]
r=O
From part (a):
Z1M[n] = z,[n]e-;(,n/N)
Z2M[n] = z2[nj,-i(•n/N)
X3M(n] = :t3(n],-;( .. /N)
So,
N-1
z,[n] = ,i(•n/N) L zi[r]z2[((n - r))N],-iC•/N)[((n-r))N+•l
r=O
N-1
= L :i:i[r]z,[((n - r))N],-i(•/N)[((n-r))N-(n-r)]
r=O
Since,
n-r, n>r
((n-r))N ={ -
N+n-r, n<r
((n - r))N - (n - r) = { 0, n ~r
N, n <r
334
then
e-;(,/N){((n-rJlN-(n-r)) = sgn [n _ r] = { 1, n~r
-1, n <r
and
N-1
Z3 =L z,[r]z2[((n - r))N]sgn [n - r]
=•
(e) Suppose, that for n 2: N/2:
z,.v[n] = z,[n],-;(,n/N) = 0
:z:2.v[n] = z2[n]e-;(,n/N) O =
then the modilied circular convolution is equivalent to the modified linear convolution:
Thus,
=
N-1
-
L z1.v[r]z2.v[n - r)
So,
N-1
z3[n] =L :r1[r]:z:2[n - r]e-,C•/N)(n-r) = :z:1[n] • :r2[n]
8.62. (a) We wish to compute z[nJ@,[n]:
:i:[n] • h[n] = :i:1[n] • h1[n] + :r1[n) • h2[n] • 6[n - 32] + z 2[n] • h,[n] • o[n - 32]
+ z 2 [n] • h 2 [n] • o(n - 32] • o[n - 32]
Let
!11[n] = » ,_, • •• l•l • » l•l,, l•l
112[n] = :r1[n] • h2[n] = z,(n] 2[h]
113[n] = z2[n] • h,[n] =:r2[n] ,[n]
11,[n] = :r2[n] • h2[n] =:r2[n] 64 2[n]
335
We can compute each of the above circular convolutions with two 64-pt DFTs and one 64-pt inverse
DFT.
So
= y(n) + y(n + 63),
:r[n)@).[n) 0 :Sn :S 62
The total computational cost is 12 DFTs of size N = 64.
(b) Using two 128-pt DFTs and one 128-pt inverse DFT:
(a) V = 49
(b) M = 51
(c) The points extracted correspond to the range 49 :Sn :S 99.
H(z) = L h(n)z-n
n=-oo
H(z) = 1-!,-no
2
The N-pt DFT of h(n]: (N = 4no)
•no-1
H(k] L
= n=O h[nJw;:-, 0 :S k :S (4no - I)
= 1- !w;no
2 no
1 - !e-i(•/>l•
H(k] = 2
336
(b)
H;(z) = 1
+ l
112
,_,.., l•I > 2 (1)-no for causality
h;[n] = t. G) n/no 6[n - kno]
(e)
N-1
Xn[k + NJ = L i[n]HN[n(k + N)]
n=O
= Xn[k).
337
We thus conclude that the DHS coefficients form a sequence that is also periodic with period N.
(b) We have:
= !z[n]N
= z[n].
Where we have used the fact that }::,;;;,' HN[mk)HN[nk) = N only if ((m))N = ((n))N, otherwise
it's 0.
This completes the derivation of the DHS synthesis formula.
(c) We have:
HN[a+N] = cos(2-,,,{_a;N))+sm(2,r(a;N))
21ra 2,ra
= cos( N + 2,r) + sm( N + 21r)
2.-a) . (2.-a)
= (
cosN +smN
= HN[a].
And:
HN[a+b) = cos(2.-('.;.,+b))+sm(2.-('.;.,+b))
(d) We have:
N-1
DH S(z[n - no]) = I: z[n - no]HN[nk]
-=O
N-1-no
= }: z[n]HN[(n +no)k]
n.=-ftcl
338
N-1-no
= L :i:(n](HN[nk]CN[nok] + HN[-nk]SN(nok])
N-1-no N-1-no
= CN[nok] L :i:(n]HN(nk] + SN(nok] L :i:(n]HN[-nk]
n=-no n=-no
N-1 N-1
= CN(nok] L :i:(n]HN[nk] + SN(nok] L :i:(n]HN(-nk]
n=O n=O
= CN(nok]XH[k] + SN(nok]XH[-k]
(e) We have:
N-1
DHT{i,[n]} = DHT{L z1[m]x2(((n - m))N]}
m=O
N-1 N-1
XHl[k] = L (L z1[m]x2[((n - m))N])HN[nk]
n=O m=O
N-1 N-1
= L z,[m] L z2(((n - m))N]HN[nk]
m=O n.=O
N-1
= L z,[m]DHT{z2[((n - m))N]}
m=O
N-1
= L z,[m](XH2[k]CN[mk] + XH2[((-k))N]SN[mk]) (using PS.65-7)
N-1 N-1
= L z,[m]XH2[k]CN(mk] + L z,[m]XH2[((-k))N]SN[mk]
m=O m=O
= 'i:'
=O
z,[m]XH>[k](HN[mk] + HN[-mk])
2
+ 'i:'
m=O
z,[m]XH2[((-k))N](HN(mk] -2HN[-mk])
l l
=
2XH2[k](XH1[k] +XH1[((-k))N]) + 2XH2[((-k))N](XH1(k]-XH1[((-k))N])
= 2l XH1(k](XH2[k] + XH2[((-k))N]) + 2l XH,[((-k))N](XH2[k]- XH2[((-k))N])
= ~• 2rkn . 2,rkn
L., :[n](cos( JV) - j sm( JV))
....,
N-1
-· L :[n](CN[kn) - jSN[kn))
=O
then:
N-1
L o:[n]CN[kn]
=D
=
N-1
L :[n]SN[kn)
...., =
We thus get:
N-1
XH[k) L o:[n](CN[kn] + SN[kn))
= ....,
(g) We have:
N-1
Xg[k] = L :[n)(CN[kn] + SN(kn))
=O
Therefore:
N-1
L o:[n]CN[kn] =
=D
N-1
L o:[n]SN[kn] =
=D
We thus get:
N-1
X[k] = L :[n)e-i",F-
=D
N-1
= L o:[n](CN[kn] - jSN[kn))
=D
340
.X(k] = .i(.,;"ll,.=W
= X(,i2•12N)(l + (-I)')
Therefore:
k even
k odd
We thus want:
k even
k odd
341
Thus:
y(n] = ! w[n-N]
w(n)
0
, 0-S.n-S.N-1
N-S.n-S.2N-1
otherwise
= { 1 , lwl -s. {
0 , otherwise
Since h(n) is FIR, we assume it is non-zero over O -s_ n -S. N. The phase of H(e 1 ~) should be set
such that h(n) is symmetric about the center of its range, i.e. ~- Therefore, the phase of H(e'~)
should be e' "f- . So one possible H(k) may be:
that is:
!
ei¼• 0 < k < !f.
H(k) = 0 : o~he~
,it• '
4N - !f. < k < 4N
2 - -
!
X(k)H'(k] 0-S. k -S. ~
Y2(k] = 0 otherwise
X(k - 3N]H'[k - 3N] 4N-!f.<k<4N
2 - -
2N-3
=
z 1 (n) _1_ "
2N-2L..,
N-1
.... X 1 (k]ei••tn/(2N-2)
2N-3
Note that:
2N-3 N-2
L X''[2N - 2 - kjei2dn/(2N-2) = L X"[r]_,i2r(2N-2-r)n/(2N-2)
l=N
N-2
= L X"(k]e-j2rb/(2N-2)
l=l
therefore:
l N-1 N-2
:r,(n] = --=--("
2N-2 ~
X"[k]_,i2dn/(2N-2) + "X"[k]e-;2dn/(2N-2lj
~
h=O k=l
N-2
= 2
1
N _ (X"(O] + X''[N -1]_,i2rn +
2
L
X"[k]{.,i2r•n/(2N-2) + 0 -;2dn/(2N-2)))
•=1
1 N-2
= -2N-2 -(X''[O] + X"[N - l]ei'" + "X"[k]2cos( dn ))
~ N-1
k=l
and:
= N ~ 1 (L o[k]X'1[k]cos(;:'1 ))
•=0
o[k] ={ ! k=0andN-1
I$ k $N-2.
This completes the derivation.
8.69.
v[n] = z2[2n]
therefore, for k =0, I, ... , N - I:
Furthermore, we have:
N-1
2Re{e-i¼Y-V[k]} = 2Re{,-i¼Y- L v[n]e-i"P}
n=O
N-1
= 2&{I; v[n],-,<~(n+¼n}
n=O
N-1
= 2 L Re{v[n],-il'w'(n+¼))}
n=O
N-l 2,rk l
= 2 L v[n]cos(N(n + ))
=O
4
= 2 I:[J
n=O
vncos ("k{4n+l))
2
N.
2&{,-,Wv[k]} = 2Re{X[k],-'*}
2N-l
= 2 L Re{:r[n],-i*( 2n+l)}
n=O
N-1
[ ] cos ("k(2nN+ 1)) .
2 L :rn
= 2
n=O
8.70. Substituting the expression for X2[k] from equation (8.174) into equation (8.175), we get:
344
l 2N-l .
:r2[n] = -
2N •=0
L X2[k]e'....../(2N)
1 N-1 2N-1
= 2N(X•2[0] + L
X"(k]ei""f(2N)e1'2•ln/(2N) _ L xe2(2N _ k]ei""f(2N)_,j2dn/(2N))
Cl k=N+l
1 N-1 · 2N-l
= w<X"[0] + L
X'.[k]ei••(2n+1)/(2N) _ L X"[2N _ j;J_,id(2n+l)/(2N))
k=l i=N+l
l N-1 N-1
=
2
N(X"[0] + L
X'2[kj_,id(2n+l)/(2N) _ L
X"[kJ.,i•(2N--<e)(2■+1)/(2N))
l=-1 l=l
l N-1 N-1
= 2
N(X"[0] + L
X"(k]_,id(2n+l)/(2N) +L X"[k]e-j•l(h+l)/(2N))
i=l i=l
l N-1
= 2N(X''[0] + L x·'[k](ei••<••+l)/(2N) + .-jd(2n+l)/(2N)))
•=1
N-1
(X'2[0] + ~ X' 2[k] cos("k( 2n + l)))
1
= -2N L.., 2N .
•=1
Furthermore:
= ! •=0L
N-1
P[k]X''[k] cos{"k(~',:/ l))
P[k] = { ! k=O
l:Sk:SN-1.
This completes the deri,...tion.
8. 71. First we derive Parsev..l's theorem for the DFT.
Let :r[n] be an N point sequence and define y[n] as follows:
y[n] = :r:[nJ®:r*[((-n))N]-
Using the properties of the DFT, we have:
2N-3 N-1
L !z,[n]l 2 =2 L lz[n]I' - !z[O]l 2 - )z/N -1]1
2
•
n=O
We thus conclude:
N-1 N-1
21
,/_ (2
2
L JX"fk]I' - !X''[O]I' - IX"[N -1]1 2 ) =2 L jz[n]j 2 - lr[O]l
2
- lrfN -1]!
2
.
n=O n=O
We thus conclude:
-
N-1 N-1
~(2 L IX[k]J 2 IX[O]l 2) = 2 L lz[n]j 2.
....
-
2
347
Solutions - Chapter 9
9.1. There are several possible approaches to this problem. Two are presented below.
Solution #1: Use the program to compute the DFT of X(k], yielding the sequence g[n].
N-1
=L
g[n]
.... X[k],-;2dn/N
Then, compute
1
x[n] = Ng[((N - n))N]
for n = 0, ... , N - 1. We demonstrate that this solution produces the inverse DFT below.
1
x[n] = Ng[((N - n))N]
N-l
= .!_ L X[k],-;2wl(N-n)/N
N •=0
N-1
= .!_ L X[k],i2dn/N
N •=0
Solution #2: Take the complex conjugate of X[k], and then compute its DFT using the program,
yielding the sequence /[n].
N-1
/[n] =L X"[k],-;••>n/N
i:=0
Then, compute
1
x[n] = Nf"[n]
We demonstrate that this solution produces the inverse DFT below.
:r[n] = !r[n]
= !L N-1
1=0
X[kj,i2•ln/N
X[2] = L x[nJwr
n=O
2
= x[O] + x[I]W; + x[2Jw: + x(3Jw: + x[4Jw: + z[SJW/ 0 + x(6]W;
+ x[7]W; 4
= x(O] + x(l]W; + x[2](-l) + x(3](-W/) + x(4)(I) + z[S]W;
+ x(6](-1) + z(7)(-W;)
350
Each input sample contributes the proper amount to the output DFI' sample.
9.3. (a) The input should be placed into A[r] in bit-reversed order.
A[O] = z[O]
A[l] = z[4]
A[2] = z[2]
A[3) = z[6]
A[4) = z[l]
A[S) = z[S)
A[6] = z[3]
A[7] = z[7]
X[k] '
= L(-W,)"W,.•
n=()
'
= I:<-1rw,;w,;•
n=O
'
= I;cw.-')"w,w,;•
n=()
'
= I;w;<•-,)
n=O
1- w•-•
= 1- w;-•
= So[k-3]
A sketch of D[r] wis provided below.
0 1 2 3 4 5 6 7 r
1 '
'
0 1 2 3 4 5 6 7 r
9.4. (a) In any stage, N/2 butterflies must be computed. In the mth stage, there are 2m-l different
coefficients.
(b) Looking at figure 9.10, we notice that the coefficients are
We therefore have
for Wn = wo +nD.w or
for Wn = wo + (n - 19)1:,..,.
9.9. In this problem, we are using butter:fly flow graphs to compute a DFT. These computations are done in
place, in an array of registers. An example flow graph for a N = = =
8, (or v log2 8 3), decimation-in-
time DFT is provided below.
353
(a) The difference between l 1 and lo can be foood by using the figure above. For example, in the first
stage, the array elements A[4] and A[S] comprise a butterfly. Thus, l1 - lo = =
5 - 4 1. This
difference of 1 holds for all the other butterflies in the first stage. Looking at the other stages, we
find
stage m = 1: l1 - lo 1 =
stage m = 2: l1 - lo 2 =
stage m = 3: l1 - lo = 4
From this we find that the difference, in general, is
form=l, ... ,v
(b) Again looking at the figure, we notice that for stage 1, there are 4 butterflies with the same twiddle
factor. The lo for these butterflies are 0, 2, 4, and 6, which we see differ by 2. For stage 2, there are
two butterflies with the same twiddle factor. Consider the butterflies with the IV,\' twiddle factor.
The lo for these two butterflies are O and 4, which differ by 4. Note that in the last stage, there
are no butterflies with the same twiddle factor, as the four twiddle factors are unique. Thus, we
found
stage m = 1: ti.lo 2 =
stage m = 2: ti.lo =4
stage m = 3: n/a
9.10. This is an application of the causal version of the chirp transform with
0 = 0000 -+ 0000 = 0
l = 0001 -+ 1000 = 8
2 = 0010 -+ 0100 = 4
3 = OOll -+ 1100 = 12
4 = 0100 -+ 0010 = 2
5 = 0101 -+ 1010 = 10
6 = 0ll0 -+ 0110 = 6
7 = 0111 -+ 1110 = 14
8 = 1000 -+ 0001 = l
9 = 1001 -+ 1001 = 9
10 = 1010 -+ 0101 = 5
ll = 1011 -+ 1101 = 13
12 = 1100 -+ 0011 = 3
13 = ll0l -+ 1011 = 11
14 = 1110 -+ 0111 = 7
15 = 1111 -+ 1111 = 15
The new sample order is 0, 8, 4, 12, 2, 10, 6, 14, 1, 9, 5, 13, 3, 11, 7, 15.
9.12. False. lt is possible by rearranging the order in which the nodes appear in the signal !low graph.
However, the computation cannot be carried out in-place.
9.13. Only them = 1 stage will have this form. No other stage of a N = 16 radix-2 decimation-in-frequency
FFT will have a W1 & term raised to an odd power. ·
9.14. The possible values of r for each of the four stages are
~= 1, r=O
m=2, r=0,4
m=3, T =0,2,4,6
m=4, r = 0, 1,2,3,4,5,6, 7
355
N Program A Program B
2 4 20
4 16 80
8 64 240
16 256 640
32 1024 1600
64 4096 3840
Thus, we see that a sequence with length N = 64 is the shortest sequence for which Program B runs
faster than Program A.
9.16. The possible values for r for each of the four stages are
m= 1, r=0
m=2, r =0,4
m=3, r =0,2,4,6
m=4, r =0, 1,2,3,4, 5,6, 7
where WN is the twiddle factor for each stage. Since the particular butterfly shown bas r = 2, the stages
which have this butterfly are
m=3,4
9.17. The FFT is a decimation-in-time algorithm, since the decimation-in-frequency algorithm bas only 10,'2
terms in the last stage.
9.18. lf the N 1 = 1021 point DFT was calculated using the convolution sum directly it would take Nf
multiplications. lf the N 2 = 1024 point DFT was calculated using the FFT it would take N2 log2 N2
multiplications. Assuming that the number of multiplications is proportional to the calculation time
the ratio of the two times is
N'f 1021 2
N2 log2 N2 = 1024 log2 1024 = lOl.S s:: lOO
which would explain the results.
9.19. X(e'••I•) corresponds to the k = 3 index of a length N = 8 DFT. Using the flow graph of the
second-order recursive system for Goertzel's algorithm,
a = 2cos(2;k)
= 2cos (2"i3))
= -v2
b = -wt
-e-ib/8
=
1 +i
=
v2
356
9.20. First, we derive a relationship between the X 1 (ei") and X(ei") using the shift and time reversal
properties of the DTFT.
r,[n] = z[32 - n]
X1 (ei") = X(,-;"),-,-.2w
Looking at the figure we see that calculating 11[32] is just an application of the Goertzel algorithm with
=
k 7 and N = 32. Therefore,
11[32] = x,[1]
= X,(ei")l.,=W
= X(,-;"),-;.,321
w=fi
= X(,-;tt ),-;(if )32
= X(,-;tt)
Note that if we put x[n] through the system directly, we would be evaluating X(z) at the conjugate
location on the unit circle, i.e., at w = +71r/l6.
9.21. (a) Assume x[n] = 0, for n < 0 and n > N - 1. From the figure, we see that
Y•[n] = x[n] + Wty.[n - l]
Starting with n =0, and iterating this recursive equation, we find
Y>[O] = z[O]
11•[1] = r[l] + wtx[0]
Y•[2] = i[2] + Wtx[l] + Wff:i:[0]
X(z)
1- w-•,-
N
1
1- 1
2z- cos(~) + z-2
= X(z)
1- w-•z-
N
1
Therefore, Y•[n] = x[n] + Wty,[n - l]. This is the same difference equation as in part (a).
357
9.22. The flow graph for 16 point radix-2 decimation-in-time FFT algorithm is shown below•
x[8)
w;. X[1)
x[4) X[2)
x[12)
w;. X[3]
x[2) X(4]
x[10]
w;. X(5]
x[6) X[6]
-
x[14]
vi;, X[7J
x[1) , X(8]
vi;,
x[9) _,
x[5)
x[13]
vi;,
x{3}
x[11)
w;.
x[7)
x[15]
vi;,
To determine the number of real multiplications and additions required to implement the flow graph,
358
consider the number of real multiplications and additions introduced by each of the coefficients Wt:
W/'6 :0 real multiplications + 0 real additions
Wt, : 0 real multiplications+ 0 real additions (W140 (a + jb) =b - aj)
W{, : 2 real multiplications+ 2 real additions (Wr6 (a + jb) = ,1/-(a + b) + i,1/-(b- a))
Wf6 : 2 real multiplications+ 2 real additions similarly
Wf, : 4 real multiplications+ 2 real additions
Wf, : 4 real multiplications+ 2 real additions
Wf, : 4 real mnltiplications + 2 real additions
U'i, : 4 real mnltiplications + 2 real additions
The contribution of all the Wt 's on the flow graph is 28 real multiplications and 20 real additions. The
butterflies contribute O real multiplications and 32 real additions per stage. Since there are four stages,
the butterfiles contribute O real multiplications and 128 real additions. In total, 28 real multiplications
and 148 real additions are required to implement the flow graph.
9.23. (a) Setting up the butterfly's system of equations in matrix form gives
Solving for
gives
1/2
X{O] x(O]
w:,2
X{4J x(1]
1/2
Xl2J x(2]
w:,2
X{6] x(3]
1/2
Xl1J x(4]
w'12
8
XIS] x(S]
1/2
X{3] x(6]
w:,2
Xf7l
(c) The modification is made by removing all factors of 1/2, changing all Wj;' to W,v, and ~labeling
the input and the output, as shown in the flow graph below.
w'
x(4) ' X{1]
YI'
x(6] ' X13I
w'
x(S] • XIS]
(d) Ye.,. In general, for each decimation-ill-time FFT algorithm there ezists a decimation-in-frequency
FFT algorithm that corresponds to interchanging the input and output and reversing the direction
of all the arrows in the flow graph.
360
9.24. (a) Using the figure, it is observed that each output Y[k] is a scaled version of X[k]. The scaling
factor is W[k], which is found to be
k =01234567
W[k] = 1GGG2GG2 G2 G3
Y(k] = W(k]X[k]
= W(k]X[kjW'(k]
= X(k]
9.25. Let z, be the z-plane locations of the 25 points uniformly spaced on an arc of a circle of radius 0.5
from -'If /6 to 2,r /3. Then
where
71'
WO =
6
&, = (5;) (;4)
5.-
=
144
This is similar to the expression for X(e'~) using the chirp transform algorithm. The oiJ.ly difference is
the (0.5)-• term. Setting
we get
N-,
X(z.) = w•· /2 L g\n\w-(•-•l' /2
...0
using the result of the chirp transform algorithm. A procedure for computing X (z) at the points •• is
then
361
• Multiply the sequence x(n] by the sequence (O.sJ-••-;"'••w•'/2 to form g[n].
• Convolve g(n] with the sequence w-•' /2 •
• Multiply this result by the sequence w•'l 2 to form X(z•J-
A block diagram of this system appears below.
g{n]
x[n] ---l(:X}------1( l----i(_X}--➔
9.26.
2N-l
Y(k] = L y(n],-;(~l••
n=O
N-1 2N-l
= L 2
e-j(r/N)n e-j(2r/N)(k/2)n + L e-j(r/N)n 2 e-j(2 .. /N)(li:/2)n
n.=O n=N
N-1 N-1
= L 2
e-i(•/N)n e-j(211/N)(k/2)n + L e-i(•/N)(l+N)2 e-j(2w/N)(k/2)(l+N)
n=O l=O
N-1 N-1
= L 2
e-i(•/N)n e-j(2r/N)(A:/2)n + e-i•lc L e-j(•/N)(l 2+2Nl+N 2 )e-i<2•/N}(i/2)l
n=O l:sO
N-1 N-1
= L 2
e-j(r/N)n e-j{2r/N).(li:/2)n + (-l)k L e-j(r/N)l 2 e-j(2w/N)(1/2)J
n=O l=O
N-1
= (1 + (-1)•) L ,-;(r/NJ•',-;(2r/N)(•/2)n
n=O
= { 2X(k/2], k even
0, k odd
Thus,
k even
Y(k] = { 0,
k odd
9.27. Let
y[n] = .-j2rn/627:r(n]
Then
Y(e'"') = X(e'l"'+tM)
Let y'(n] =:E:=-oo y(n + 256m], 0 :5 n :S 255, and let Y'(k] be the 256 point DFT of y'[n]. Then
9.28. (a) The problem states that the effective frequency spacing, !!./, should be 50 Hz or less. This
constrains N such that
1
!!./ = NT :S 50
1
N '2:. SOT
~ 200
Since the sequence length L is 500, and N must be a power of 2, we might conclude that the
minimum -value for N is 512 for computing the desired samples of the z-transform.
However, we can compute the samples with N equal to 256 by using time aliasing. In this technique,
we would zero pad z[n] to a length of 512, then form the 256 point sequence
We could then compute 256 samples of the z-transform of y[n]. The effective frequency spacing of
these samples would be 1/(NT)"' 39 Hz whlch is lower than the 50 Hz specification.
Note that these samples also correspond to the ..,.,n-indexed samples of a length 512 sampled
z-transform of z[nj. Problem 9.30 discusses this technique of time aliasing in more detail.
(b) u,t
y(n] = (1.25rz[n]
Then, using the modulation property of the z-transform, Y(z) = X(0.8z) and so Y[k] = X(0.8e;,r1/N).
9.29. (a) We offet two solutions to this problem.
Solution #1: Looking at the DFT of the sequence, we find
N-1
X[k] = L z(nJe-j2dn/N
=O
(N/2)-1 N-1
= L z[n],-j2dn/N + L z[n]e-j2dn/N
n=O n.=N/2
(N/2)-1 (N/2)-1
= L z(n]e-;2,•n/N + L z(r + (N/2)Je-j2d[,+(N/2)1/N
n=O r=O
(N/2)-1
= L z(nl[l - (-l}"]e-;,,>n/N
=O
= 0, k even
Solution #2: Alternatively, we can use the circular shift property of the DFT to find
X[k] = -X(k],-;<~>•<f>
= -(-I)'X[k]
= (-1)>+ 1 X[k]
When k is even, we have X(k] = -X[k] whlch can only be true if X[k] = O.
363
=
for k 0, ... , N /2 - 1. Thus, we can compute the odd-indexed DIT values using one N /2 point
DIT plus a small amount of extra computation.
9.30. (a) Note that we can write the even-indexed values of X[k] as X[2k] for k = 0, ... , (N/2) - 1. From
the definition of the DIT, we find
N-1
X[2k] = L :t[n]e-;2,(2t)n/N
.....
N/2-1
= L :t[n],-;,p;,,•n
.....
N/2-1
+ L :t[n + (N/2)],-_;.,~;,,1n,-;,J;,,<NJ2>•
.....
N/2-1
= L (:t(n] + :t(n + (N/2)])e-;TP7l!tn
.....
= Y[k]
M-1
L L
00
= :t[n +rM],-;2,>(n+rM)/M.,i2r(rJl)t/M
r=-oo n=O
Y(k] = L :t[l],-;..t1/Jt
l=-oo
= X(ei•••fM)
Thus, the result from Part (a) is a special case of this result if we let M =
N /2. In Part (a), there
are only two r terms for which y[n] is nonzero in the range n = 0, ... , (N /2) - 1.
364
(c) We can write the odd-indexed values of X[k] as X(2k + l] fork = 0, --- , (N/2) - l. From the
definition of the DFT, we find
N-1
X[2k + l] = L z(nJe-i..(2Hl)n/N
.....,
N-1
= L z(n]e-i2rn/N•-;2r(2k)n/N
.....,
(N/2)-1 (N/2)-1
= L z[n]e-;2rn/N•-i 1Jj,, •n + L z[n + (N/2)J•-;2r{n+(N/2)1/N•-;;lr,•ln+(N/2))
n=O n=O
(N/2)-1
= L [(z(n] - z(n + (N/2)])•-;~n] •-;rM,y>n
n=O
Let
(z(n] - z[n + (N/2)])e-i< 2•/N)n, 0 $ n $ (N/2) - l
11(n] = {
0, otherwise
Then Y[k] = X[2k + l]. Thus, The algorithm for computing the odd-indexed DIT values is as
follows.
step 1: Form the sequence
step 2: Compute the N/2 point DFT of y(n], yielding the sequence Y(k].
step 3: 'l'he odd-indexed values of X[k] are then X(k] = Y[(k - 1)/2], k = 1, 3, ... , N - l.
9.31. (a) Since x(n] is real, z[n] = x•[n], and X(k] is conjugate symmetric.
N-1
X(k] = L x"[n]e-;~•n
n=O
N-1 )"
= (
~ z(n]ei~•n•-j~Nn
= X"(N-k]
In these expressions, the subscripts "E" and "0" denote even and odd symmetry, respectively, and
the subscripts "R" and "I" denote real and imaginary parts, respectively.
365
GER[k) = X1ER(k]
the odd and real part of G[k] is
GoR[k] = -X201[k]
the even and imaginuy part of G[k] is
GEr[k] = X2ER(k]
and the odd and imaginary part of G[k] is
Gor[k] = X,or[k]
Having established these relationships, it is easy to come up with expressions for Xi[k] and X 2(k].
(c) An N = 2" point FFT requires (N/2) log2 N complex multiplications and N log2 N complex addi-
tions. This is equivalent to 2N log2 N real multiplications and 3N log2 N real additions.
{i) The two N-point FFTs, X 1[k] and X 2[k], require a total of 4Nlog2 N real multiplications and
6N log2 N real additions.
{ii) Computing the N-point FFT, G[k], requires 2N log2 N real multiplications and 3Nlog2 N real
additions. Then, the computation of GER(k), GEr[k], Gor[k], and GoR(k) from G[k] requires
approximately 4N real multiplications and 4N real additions. Then, the formation of X 1 (k] and
X2(k] from GER(k), GEr[k], Gor(k), and GoR[k] requires no real additions or multiplications.
So this technique requires a total of approximately 2N log2 N + 4N real multiplications and
3N log2 N + 4N real additions.
(d) Starting with
N-1
X[k] = L z(n],-j2dn/N
....0
and separating z(n] into its even and odd numbered parts, we get
step 1: Form the sequence g[n] = z[2n] + jz[2n + l], which has length N /2.
step 2: Compute G[k], the N/2 point DFT of g[n].
step 3: Separate G[k] into the four parts, fork= l, ... , (N/2) - 1
1
GoR[k] =
2(GR[k] - GR[(N/2) - kl)
l
GsR[k] =
2(GR[k] + GR[(N/2)- kl)
l
Gor[k] =
2(Gr[k] - Gr[(N/2) - kl)
(d) For these signals, N is large enough so that circular convolution of :i:[n] and h[n] and the linear con-
volution of :z;[n] and h[n] produce the same result. Counting the number of complex multiplications
for the procedure in pa.rt (b) we get
Since there a.re 4 real multiplications for every complex multiplication we see that the procedure
takes 6Nlog,, N + 4N real multiplications. Using the answer from part (a), we see that the direct
=
method requires (N /2)(N /2) N2 /4 real multiplications.
The following table shows that the sma.llest N =
2• for which the FFT method requires fewer
multiplications than the direct method is 256.
9.33. (a) For each L point section, P - 1 samples a.re discarded, leaving L - P +l output samples. The
complex multiplications a.re:
L point FF!' of input: (L/2) log2 L = v2• /2
Multiplication of filter and section DFI': L 2•=
L point inverse FFT: (L/2) log2 L = v2• /2
Total per section: 2•(v + 1)
Therefore,
Complex Multiplications 2• ( v + l)
Output Sample
= 2• - P + l
Note we assume here that H[k] ba.s been precalculated.
(b) The figure below plots the number of complex multiplications per sample \'etS1IS v. For v 12, =
the number of multiplies per sample reaches a minimum of 14.8. In comparison, direct evaluation
of the convolution sum would require 500 complex multiplications per output sample.
368
,S! 25
I
., 20
'
'S
CL
~ 15
' "
"
8.
!!
0
10
I
'§
5 •
::! 0
10 11 12 13 14 15 16 17 18 19 20
V
Although ,., : 9 is the first valid choice for overlap-save method, it is not plotted since the value is
so large (in the hundreds) it would obscure the graph.
(c)
! IOverlap/Save IDirect j
I ,., I 2"(•+1)
2"' -
2"' 1+1
I 2•-l I
1 2 1
2 4 2
3 6.4 4
4 8.9 8
5 11.3 16
9.34. This problem asks that"we find eight equally spaced inverse DFT coefficients using the chirp tranSform
. algorithm. The book derives the algorithm for the forward DFT. However, with some minor tweaking,
it is easy to formulate an inverse DFT. First, we start with the in?er&e DFT relation
N-1
:r[n] = .!_ L X[k]ei'ru/N
N ,._.,
N-1
:r[n,] : .!, L X[kjd2wn,•/N
N •=0
369
Next, we define
.O.n = 1
n, = no+ l.O.n
where l = 0, ... , 7. Substituting this into the equation above gives
:i:[n,] =
Defining
we find
N-1
z[n,] =~ L X(kJei2•no•tNw-tt
4=0
Using the relation
lk = ~[l' + k 2 - (k - l) 2 l
we get
Let
Then,
11[0] = z[1020)
11[1) = z[1021)
11[2) = z{1022)
y(3) = z(1023)
y[4) = :i:[O)
11(5] = :i:(1)
11[6) = :i:[2]
11(7) == :i:[3]
370
= { ,:[n], iL :S n :S iL + 127,
0, otherwise
Using the above we can implement the system with the following block diagram.
x[n]
Shift
b -iL FFT-1
Multiply f---1.i
w[n] = u[n]-u[n-128) ---1~
256- I IFFf-2 y,[n]
256- t
FFT-1
h[n)---1~
256- t
The FFT size was chosen as the next power of 2 higher than the length of the linear convolution. This
insures the circular convolution implied by multiplying DFTs corresponds to linear convolution as well.
Neoov = N., + Nb - 1
= 128+ 64- I
= 191
NpPT =256
9.36. (a) The flow graph of a decimation-in-frequency radix-2 FFT algorithm for N = 16 is shown below.
371
lC(OJ X(OJ
lC(1J
w';,
X(8J
lC(2J X(4J
w';,
lC(3J X(12J
lC(4J X(2J
w';,
lC(SJ X(10J
lC(6J X(6J
w';,
x[7J _, _, X(14]
lC(8J X{1J
w';,
lC(9) X{9J
xf10) X{SJ
w';, X(13)
x[11J
x{12) X{3J
lC(13)
w';. X(11]
lC{14) X{7J
w';, X[15]
lC(15]
,tO] X(OJ
>11)
w:. X[BJ
w:. X[4]
w:. X[2]
w:. X[6)
w:. X[1)
w:. Xl5J
w:. X(3]
w:. X[7]
(c) The pruned butterflies can be used in ( 11 - I') stages. For simplicity, assume that N /2 complex
multiplies are required in each unpruned stage. Counting all W~ terms gives
we note that :r[2n + l] = h[n], and :r[2n - l] = h[n - l] for n = 0, l, ... , ~ - 1. We then get
So
H[k] - F[k]
- wicw;,• - wi >
Therefore,
X(k] = G[k] + WtH(k]
G[k]+ F(k]
= w-•
N - w•
N
j F(k]
= G[k] - 2sin(2,rk/N)
Clearly, we need to compute X(0] and X[N/2] with a separate formula since the sin(2,rk/N) = 0
=
for /c 0 and k N/2.=
(c) For each stage of the FFT, the equations
X[0] = G[0] + F(0]
X[N/2] = G[0] - F(0]
require 2 real additions each, since the values G[0] and F[0] may be complex. We therefore require
a total of 4 real additions to implement these two equations per stage.
For a single stage, the equation
l . F[/c]
X[k] = G(ic] - 2; sin(2,r/c/N) k #- 0, N /2
requires (N - 2)/2 multiplications of the purely imaginary "twiddle factor" terms by the complex
coefficents of F[k] fork#- 0, N/2. The number of multiplications were halved using the symmetry
sin(2,r(/c + N/2)/N) =- sin(2,rk/N) and the fact that F[k] is periodic with period N/2. Since
multiplying a complex number by a purely imaginary number takes 2 real multiplies, we see that
the equation requires a total of (N - 2) real multiplies per stage.
We also need (N - 2) complex additions to add the G(k] and modified F[k] terms for k f- 0, N /2.
Since a complex addition requires two real additions, we see that the equation takes a total of
2(N - 2) real additions per stage.
Putting this all together with the fact that there are log2 N stages gives us the totals
Real Multiplications = (N - 2) log2 N
Real Additions = 2N log2 N
Note that this is approximately half the computation of that of the standard FFT.
(d) The division by sin(2,rk/N) fork near 0 and N/2 can cause X[k] to get quite large at these values
of k. Imagine a signal z 1 [n], and signal z 2 [n] formed from z 1 [n] by adding a small amount of white
noise. Using this FFT algorithm, the two FFTs X 1 [k] and X2[k] can vary greatly at such values
of k.
9.38. (a)
N-1
X(2k] = L z(n]W~
n=O
(N/2)-1
= L (z[n]W~+z(n+(N/2)JW!>(n+(N/•ll)
-=O
(N/2)-1
= L (:i:[n] + :i:[n + (N/2)]) wi•n
n=O
375
In the derivation above, we used the fact that W},N = 1. Since W;l" = W},12 , X[2k] has been
expressed as an N/2 point DFT of the sequence z[n] + z[n + (N/2)], n = 0, 1, ... ,(N/2) - 1.
(b)
N-1
X[4k + l] = L z[nJwt•.Ht)n
n=O
(N/4)-1 .
= L (z[n]w;w:0n + :,;(n + (N/4)Jw;+(Nf•>w:,•<•+(N/•))
n=O
In the derivation above, we used the fact that Wf:14 = -j, W;/"14 = j, Wf: 12 = -1, and W},N = 1.
Since wt,••= WJ:,14 , X[4k + I] has been expressed as a N/4 point DFT. But we need to multiply
the sequence (z[n] - z[n + (N/2)]) - j(z[n + (N/4)) - z[n + (3N/4)]) by the twiddle factor w;,
0 $ n $ (N/4) - I before we compute the N /4 point DFT.
The other odd-indexed terms can be shown in the sa.me way to be
(N/4)-1
X[4k + 3] = L {(z[n] - z[n + (N/2)])
n=O
Parts (a) and (b) show that we can replace the computation of an N point DFT with the compu-
tation of one N/2 point DFT, two N/4 point DFTs, and some extra complex arithmetic.
(c) Assume N = 16 and define
g(n] = z[n] + z[n + (N /2)], n = 0, 1, ... , (N /2) - 1
/,(n] = z[n] - z[n + (N /2)], n = 0, 1, ... , (N/4) - I
J,[n] = z[n+(N/4)]-z(n+(3N/4)], n=0,l, ... ,(N/4)-1
)
376
g(O)
x(OJ X[OJ
x(1J X[SJ
x(2) X[4J
x(3) X[12J
II-point OFT
x(4)
X[2J
x(SJ X[10J
x(6J X[6J
x(7J X[14J
x(BJ
~. X[1J
,
w,.
x(9J X[9J
x(10J
w.. 4-pooitOFT
X[S]
><!12]
~. X[3]
x(13]
w:. X[11J
x(14J
w,. 4-pooit OFT
X[7J
x(15]
w,. X[15]
-1
-j
X[OJ
~. X[8J
lC[4)
~. )([12)
X[2J
~. )([10)
lC[6J
-1
~. )([14)
lC[1)
~. X[9J
~. X[5J
~. )([13)
X[3J
~. X(11J
~. lC[7J
~. )([15)
(d) The flow diagram for the regular radix-2 decim&tion-in-frequency algorithm is shown in the next
figure for N = 16. Not cowmng trivial multiplications by wi, we find that there are 17 complex
multiplications total. Of these 17 complex multiplications, 7 are multiplications by wt. = -j.
Since a multiplication by - j can be done with ,zero real multiplications, and a complex multipli-
cation requires 4 real multiplications, we find that the total number of real multiplications for the
=
decimation-in-frequency algorithm to be (10)(4) 40.
Taking a look at the split-radix algorithm, we find again that there are 17 complex multiplications.
378
In this case, however, 9 of these are by Wt. = -j. Thus, it takes a total of (8)(4) = 32 real
multiplications to implement this ftow graph.
lqO]
w:. lq8]
X[4]
w:. lq12]
)(12]
w:. lq10]
X{6]
w:. lq14]
x(B] lq1]
x(10] lCl5]
x(12] X(3]
x(14] X(7)
we notice that, to first order, the 8, and 8,+1 differ by a factor of 2. ·(Note that these formulae are
in radians). This approximate factor of 2 for sucessive 8, is confirmed by looking at some values
of 8,: 80 = 45°, 81 = 26.6°, 92 = 14.0°, 93 = 7.1°. So we have a set of angles whose values are
decreasing by about a factor of 2.
You can add and subtract these 8, angles to form any angle O < 8 < ,r /2. The error is bound by
8M = arctan(2-M), the angle that would be included next in the sum. H the error were greater
than 8.11, then one of the a, terms must have been incorrect. The inclusion of the Mth term must
bring the sum closer to 8.
ao = +l
-8 = 0080
fori=ltoM-1
if (8 > 8)
else
a,= +l
end for
(c) Note that (X + jY)(l + ja,r') = (X - a,y2-•) + j(Y + a,x2-•). Hence, the recursion is simply
multiplying by M complex numbers of the form (1 + ja,2-'). These can be represented in polar
form:
(I+ ja,2-') = ✓1 + 2-2<_,io, .......,(,--1
= G;.tJa;I;
N-1
X[3k] = L z[n]W1,..
n=O
N/3-1 2N/3-l N-1
= L z[n]W1,.. + L zjn]W1,.. + L z[n]W1,...
A=O n=N/3 ,,,_2N/3
Substituting m = n- N /3 into the second summation, a.nd m = n-2N/3 into the third summation
gives
= L
n=O
(z[n] + z[n + N/3] + z{n + 2N/3])wm.
Substituting m = n-N/3 into the second summation, a.nd m = n~2N/3 into the third summation
gives
N/3-1 N/3-1
X[3k + l] = L z[nJw;(3t+i) + L z[m + N/3Jwtm+N/3H3t+l)
n=O m=O
N/3-1
+ L
m=O
z[m + 2N/3JW1"'+2N/3)(3t+t)
.
N/3-1 N/3-1
= L z[n]w;<St+I) + L z[m + N 1a1w;<n+l)w~•w;1•
n=O m=O
N/3-1
+ L z[m + 2N/3Jw;<St+i>wlf•w:,ivi•
m=O
N/3-1
= L (z[n] + :z:[n + N1a1w;1• + :z:[n + 2N/3JWJ,"/3)w;:<St+I)
n=O
381
=
Define the sequence
13
z 2 [n] = (z[n] + z[n + N/3Jw::1• + z[n + 2N/3JW;,"' )W.v
The 3-point DFT of z 2 [n] is X2 [k] = X[3k + 1]. Next, z 3 [n] is found in a similar manner. Starting
with the definition of the OFT,
N-1
X[k] = L z[n]~•
n=<>
N-1
X[3k + 2] = L :r[nJw;;<•>+•l
n=<>
N/3-1 2N/3-1 N-1
= L :r[n]w,:;(3>+2) + L z[nJw;;<>H2) + L :r[n]w,:;(3t+2)
n=O n=N/3 n=2N/3
Substituting m = n- N/3 into the second summation, and m = n-2N/3 into the third summation
gives
N/3-1 N/3-1
X[3k + 2] = L :r[nJw;;(3H2) + L :r[m + N/3JW1m+N/3)(3t+2)
n=O m=1f·
N/3-1
+ L :r[m + 2N/3JW1m+2N/3)(3H2)
N/3-1 N/3-1
= L :r[n]w,:;(3H2) + L :r[m + N/3Jw;<at+2lwff•w~N/3
n=O m=O
N/3-1
+ L :r[m + 2N/3Jw;<•>+•>w,v•w<;13
N/3-1
= L (:r[n] + z[n + N/3]W~N/ 3
+ z[n + 2N/3Jw';/ 3Jw,:;<at+ 2>
n=O
N/3-l
= L (:r[n] + :r[n + N/3]W;/'13 + :r[n + 2N/3]w';13 )Wl,ftWM3
X[k] = L :r[n]W;'
n=<>
X[O] = z[O] + :r[l] + z[2]
X[l] = :r[O] + :r[l]WJ + :r[2]Wf
X[2] = :r[O] + :r[l]Wf + :r[2JW: =:r[O] + z[l]Wf + z[2]WJ
The butterfly for the 3 point DIT is drawn below.
382
xm - ,(q)
x,., - ,IPJ
(d) Using the results from parts (a) and (b), the flow graph is drawn below.
383
x,(OJ
X(OJ
x,(11
X(3J
x,(2I
X(6]
~ xJOJ
X(1]
w'
• •,[11 N•3DFT X(4]
vi, x,r2I
X[7J
~ x;oI
X(2]
w: •Pl X(8)
•
384
x 1(0J
,c(O) X)OJ
,c(2J
~ OJ
X)6)
x,[11 X)1)
,c(3)
w'
,c(4) N=3
OFT
• 91 I N•3
OFT
X)4)
,c(SJ
w: X[7J
,c(6J X)2J
X)8J
,c(8J - - - - - - - - - - -
(fl A direct implementation of the 9 point DFT equation requires !l2 = 81 complex multiplications.
The system in part (e), in constrast, requires 4 complex multiplications for each 3 point DFT, and
an additional 4 from the twiddle factors, if we do not count the trival ~ multiplications. In total,
the system in part (e) requires 28 complex multiplications. In general, a radix-3 FFT of a sequence
of length N = 3" requires appn,:r;imauly
Note that this formula for a radix-3 FFT is of the form N log, N. The constant multiplier, ½,·is
significantly larger than that of a radix-2 FFT. This is because a radix-2 butterfly bas no complex
multiplications, while in part (c) we found that a radix-3 butterfly has 4 complex multiplications.
Also note that this formula is an upper bound, since some of the N twiddle factors in the v - 1
stages will be trivial. However, the formula is a good estimate.
9.41. (a)
N-1
X(k] = h"[k] l):r[n]h"(n])h[k - n]
n=O
N-1
= e-id 2 /N } : z{n]e-i•n2/N e3"•(t2-2&n+n2)/N
n=O
N-1
= L :r[n],-;2rn.t/N
"'"°
(b)
N-1
X[k +NJ= h"[k + NJ L :r[n]h"[n]h[k + N - n]
=O
h"[k + NJ = 2
,-;r(t+N) /N
= 2 2
e-j,r(l: +U.N+N }/N
= e-id /N e-j,rN
2
= h"[k],-,,N
So
N-1
X[k + NJ = h"[kJ,-i•N L :r[n]h"[n]h[k - n]ei•N
= X[k]
x{k]---ilXr---ll y{k]
I\•
h [k]
:t2[k] = L :z:,[t]h[k - l]
l=-oo
N-1
= L :z:[tJ•-;•t'tN.,;•<•-t)'/N
t=O
k E (0, ... , 2N - l]
Therefore,
{d)
2N-1
H(z) = L .,id'/N,-•
•=0
M-12M-1
= L L 2
~1t(r+lM) /Nz-(r+tM)
r=O l=O
M-l 2M-1
== L L ei'" 2 2
/N ei2•rl/M ei•l z-r z-LM
r=O l::=O
= 3:' r=O
.,;rr'/N ,-, rt\.,;2r,/.V ,-.v)'(-1)'2]
l ,=0
= M-1
~
. ,
e1•r /N z-r
[J - ,i2r<(2M)/.V z-2.V']
~
...... I+ ei2r,/.V z-M
= ~· ei"' /N -, [ l - .-2.w• ]
~ z 1 + ei(b/Al)rz-M
is drawn below.
387
z-M
z-1 .
-z-'Ar -1
~
.
z-M
z-1
~
ei',r/M
z-M
z-1
-ei2"(M-1 )IM
=
(f) Complex multiplications: Since we are only interested in y[k] for k N, N + 1, ... , 2N - 1 we
do not need to calculate the complex multiplications on the output side of each parallel branch
until k ? N. Thus,
Complex additions: The complex additions on the output side of each parallel branch do not
need to be computed until k ,:: N. Thus,
z-1 ,s,-..
ft'" •2
2 cos"\
lm{X[k]}
sin u,k
z-1
e1
e3
-1 "\ = 2ld<IN
(b)
Re{X(k]} = Re{y,[N]}
Im{X(k]} = lm{yo[N]}
Since the output of interest is the Nth sample, we need only consider the ,ariance at time N.
The noise ei(n] is input to both hR(n] and hr[n]. Using the techniques from chapter 6, we find the
variance of the noise is
N
cr~(N] = ,r., + ..~, I>~(n]
n=O
389
N
cry[N] = a;, + u!, L hJ[n].
n=O
Let 8=21fk/N.
N
:Ecos•en
n=O n=O
N
= ¼L(e''n + e-ifn)2
n=O
N
= !. L(e'"n + 2 + ,-,nn)
4n=O
1 ( 1 _ ei21(N+I) 1_ ,-i2'(N+l))
= 4 1 - ei28 + 2(N + 1) + 1 - e-;21
= ¼(1 + 2(N + l) + 1)
= N +l
2
Similarly, L:=0 h1{n] = N /2.
Therefore,
2-2B
uh(n] = 12 (1 + (N/2) + l)
2-2B
= 12 (N +4)/2
2 8
unn] = ~; (l + (N/2))
2-2B
= 12 (N +2)/2
9.43.
N-1 N-1
X(k] =L :r[n]cos(2.-kn/N)-j L :r(n]sin(2.-kn/N).
n=O n-o
For k # 0, there are N - 1 multiplies in the computation of the real part and the imaginary part:
9.44. (a)
(b) The conditions a.re not suflicient to guarantee that overflow cannot occur.
9.45. (a) First, note that each stage has N /2 butterfiles. In the first stage, all the multiplications a.re
=
~ +l. In the second stage, half a.re +land the other half a.re wJt•
= -j. Successive stages
have half the number of the previous stage. In general,
and
m= I
Number of - j multiplications in stage m = { O,
N/2"',
(b) If we assume that all the + 1 and -j multiplications a.re done noiselessly, then the noise variance
will be different at each output node. This is easily seen by looking at Figure 9.10, where we see
for example that X[O] will be noise-free, while X[l] will not be noise-free. Thus, a noise analysis
would be required for each output node separately. A somewhat simpler approach would be to
assume that since the first two stages consist of only +land -j multiplications, these two stages
can be performed noiselessly. Each output node is connected to all N/2 butterfiles in the first stage
and to N /4 butterflies in the second stage. Thus, if the first two stages a.re performed noiselessly,
a better estimate of the number of independent noise sources contributing to the output is
N N N
N - 1- 2 - 4 =4 - 1.
Note that all the odd indexed outputs will have exactly (N/4) - 1 of these noise sources, while
the even indexed outputs will have less. In fact, X[O], X[N/4], X[N/2], X[3N/4] will be noiseless.
X[N/8), X[3N/8], X[SN/8], and X[7N/8] will have one noise source. It is possible to continue this
analysis for all X[k], but clearly, a complicated formula would be required to desaibe the number
of noise sources for all even k. We have shown that
Thus, \h.<e uum.ber of noise sources is upper bounded (N/4) - 1. Using this bound, we can get a
"<I>."'"-~~"'-~~~~-
£[1F[k]I'] $ ( ~ - l)uJi
When the scaling is done at the input, an upper bound on the noise-to-signal ratio is found to be
Another approach to this noise analysis is to compute the average noise at the output by using the
average number of noisy butterfties connected to an output node. This style of analysis is used in
Weinstein.
(c) Now assume as before that the first two stages are noiseless. Thus, equation 9.67 would not include
the first two stages.
•-1
l'(IF(k]l 2] < a1 L 2<•-ml(~)2v-2m-2
m=2
•-1
= a1 L (~ i•-m-2
m=2
•-3
= 2at I:<~)·
= 20-2
B
c-
i::0
(!J--2)
2
1- !
2
= 4'71(1-(!)"-2)
2
2 4)
= 4as(l- -
N
Thus,
l'[IF[k]l 2] 12Na1N- 4
l'[IX[k]l 2) :::. N
< 12(N -4)"1
X.,(q]
where
£(1F[k]l2] = (N - l)u1
since each noise source propagates along a unity gain path to the output nodes.
Thus, the results for the decimation-in-frequency FFT are identical to those for the decimation-in-time
FFT. This is true for both cases oC scaling either at the input of the FFT by 1/N, or at the input of
each stage of the FFT by 1/2.
(a)
(b)
Thus we have
Re{Y,[k]} = Ev{Re{Y.[k]}}
Re{Y2[k]} = Ev{Im{Y3[k]}}
Im{Y1[k]} = Od{Im{Y,[k]}}
Im{Y2[k]} = -Od{Re{Y,[k]}}
and so
l
X,[k) =
2[Re{Y.[k]} + Re{Y,[N - kl}]
X 1(k]=Re{Y1(k]}
U,(k] =;Im{Y,(k]}
and so
X [k] = Im{Y,(k]} k _, O N
s 2sm,;. r , 2
Note that X 3 (0] cannot be recovered using this technique, and if N is even, neither can X3[N/2].
394
(f) In part (b) replace X 3 and X, with U3 and U, and use the re3Ult of (d) to give
9.48. First, we find an expression for samples of the system function H(z ).
H(z) = '"-~
<C'" b,z -•
l - Lt=• a,z-l
'<C'M b -j2•kT/N
H(ei' ..I") = l -..:.,,=0
<C'" ••a,,-;,..,,,,
'L..lt=l
Now assume N, M ~ 5ll. Let b[n] = bn and
1
a[n) ={ '
a,.,
Let B[k], A[k] be the 512 pt DFTs of b[n), and a[n]. Then
H(ei'dfm) = B[k]
A[k]
9.49. (a) It is interesting to note that (linear) convolution and polynomial multiplication are the same
operation. Many m&thematical software tools, like Matlab, perform polynomial multiplication
using convolution. Here, we replace
L-1
.,_,
p(z) = L a,z', q(z) =L b,z"
;.:() ;.:()
with
L-1
.,_,
p[n] = L a;c5[n - i], q[n] = L b,6[n - i)
;.:()
...,
Then,
r(n) = p[n] • q[n).
The coefficients in r[n) will be identically equal to those of r(z). We can compute r(n] with circular
convolution, inste.ad of linear convolution, by zero padding p[n) and q[n] to a length N = L+ M -1.
This zero padding ensures that linear convolution and circular convolntion will give the same resnlt.
(b) We can implement the circular convolution of p[n] and q[n] using the following procedure.
step 1: Take the DFTs of p[n) and q[n) using the FFT program. This gives PIA:] and Q[k).
step 2: Multiply to get R(A:) = P[k)Q[A:).
step 3: Take the inverse DFT of R(k) using the FFT program. This gives r[n).
395
Here, we assumed that the FFT program also computes inverse DFTs. If not, it is a relatively
simple matter to modify the input to the program so that its output is an inverse DFT. (See
problem 9.1).
While it may seem that this procedure is more work, for Jong sequences, it is actually more efficient.
The direct computation of r[n] requires approximately (L + M) 2 real multiplications, since a; and
b, are real. Assoming that a length L + M FFT computation takes ((L + M)/2) Jog2 (L + M)
complex multiplications, we count the complex multiplications required il> the procedure described
above to be
Note the resemblance to p(x) and q(x) of part (a). We form the signals
L-1
u(n] = L u,o(n - i]
i=O
Jt-1
111n1 = L .,,.rn - iJ
i=O
and use the procedure described in part (b). This computes the product u • v in binary. For
=
L 8000 and M =
1000, this procedure requires approximately
(d) For the (forward and inverse) FFI's, the mean-square value of the output noise is (L + M)o-~.
While ~ will be small, as there are 16 bits, the noise can be significant, since L + M is a large
number.
9.50. (a) Using the definition of the discrete Hartley transform we get
HN[a+b) = [cos(21rb/N)+sin(2,rb/N)]cos(2.-a/N)
+ [cos(-2.-b/N) + sin(-21rb/N)] sin(2ira/N)
= HN[b)CN[a) + HN[-b)SN(a)
(b) To obtain a fast algorithm for computation of the discrete Hartley transform, we can proceed as
in the decimation-in-time FFI' algorithm; i.e.,
(N/2)-1 (N/2)-1
XH[k) = L :t[2r]HN[2rk] + L :t[2r + l]HN((2r + l)k]
r=O r=O
(N/2)-1 (N/2)-1
= L :t[2r]HN[2rk] + L :t[2r + l]HN[2rk]CN[k]
r=O r=O
(N/2)-1
+ L :t(2r + l]HN[((-2rk))N]SN[k]
r=O
(N/2)-1
+ L z[2r + l]HN12[((-rklJN12)SN(k]
397
where
(N/2)-1
F[A:] = I: z[2r]HN12[rk]
r=O
is the N/2-pomt DHT of the e\'ell-mdexed pomts and
(N/2)-1
G[A:] = I:
- z[2r + l]HN12[rk]
is the N /2-pomt DHT of the odd-mdexed pomts. As m the derivation of the decimation-in-time
FFT algorithm, we can continue to divide the sequences m half if N is a power of 2. Thus the
indexmg will be exactly the same except that we have to access G[((-A:))N12] as well as G[k] and
F[k]; i.e., the "butterfly" is slightly more complicated. The fast Hartley transform will require
N log2 N operations as in the case of the DFT, but the multiplies and adds will be real instead of
complex.
9.51. (a)
l 10 0 0 0 0 0 l 0 0 0 0 0 0 0
l -1 0 0 0 0 0 0 0 1 0 0 0 0 0 0
0 0 1 l 0 0 0 0 0 0 ~ 0 0 0 0 0
F, = 0 0 1 -1 0 0 0 0
T,=
0 0 0 0 w; 0 0 0
0 0 0 0 1 1 0 0 0 0 0 0 1 0 0 0
0 0 0 0 1 -1 0 0 0 0 0 0 0 l 0 0
0 0 0 0 0 0 l l 0 0 0 0 0 0 wo8 0
0 0 0 0 0 0 1 -1 0 0 0 0 0 0 0 W;
l 0 1 0 0 0 0 0 1 0 0 0 0 0 0 0
0 1 0 1 0 0 0 0 0 1 0 0 0 0 0 0
1 0 -1 0 0 0 0 0 0 0 l 0 0 0 0 0
0 1 0 -1 0 0 0 0 0 0 0 1 0 0 0 0
F2 = 0 0 0 0 1 0 1 0
T2 = 0 0 0 0 ~ 0 0 0
0 0 0 0 0 1 0 1 0 0 0 0 0 II-'.'
8 0 0
0 0 0 0 1 0 -1 0 0 0 0 0 0 0 w•• 0
0 0 0 0 0 1 0 -1 0 0 0 0 0 0 0 W'8
1 0 0 0 1 0 0 0
0 1 0 0 0 1 0 0
0 0 l 0 0 0 1 0
0 0 0 1 0 0 0 1
F, = 1 0 0 0 -1 0 0 0
0 1 0 0 0 -1 0 0
0 0 1 0 0 0 -1 0
0 0 0 1 0 0 0 -1
(b)
QH = FfTfFfzj'Ff
= F1TiF2T2F,
398
xfOJ
xf1J y(1]
-1
vf.
xf2J y(2]
xf3)
xf4] y(4)
xfSJ
xf6] y(6]
x(7J -1 y{7J
-1 -1
This structure is the decimation in frequency FIT with the twiddle factors conjugated and therefore
calculates
N • IDFI'{z[n]}
(c) Knowing that Q calculates the DIT and 11Q
8 calculates the IDFT, we should realize that cas-
cading the two should just return the original signal. More formally we have
= H[k]X[k]
Next, the orthogonality of the basis vectors is shown to be a necessary requirement for the circular
convolution property. We start again with the circular convolution of :r[n] and h[n]
N-1
y[n] = L :r[m]h[((n - m))N]
m=O
Substituting the inverse DFT for :r[m] and h[((n - m))N] gives
y[n] = }:
m=O
(! }: X[k Jw,,••"') (~ ~ H[k,Jw;:,;••«n-m))N)
l:1-0
1
l:2=0
= }:
m=O
(!}: X[k,Jw,,••"') (! ~ H[k,Jw;:.;••<•-m))
.k1 =O i:2:0
Therefore, the circular convolution property holds as long as the basis vectors are orthogonal.
(b)
400
X[k] = ( (t z[n]4n>))
17
= ((1 · 1 + 2 · 4• + 3 · 16• + 0 · 64•))17
= ((1 +2·4• +3-16•))17
H[k] = ( (t h[n)4n•)) 17
= ((3· 1 + l ·4• +O· 16• +0·64•))17
= ((3+4•Jh,
Y[O] = ((24)),, = 7
Y(l] = ((42)),, = 8
Y[2] = ((4)),, = 4
Y(3] = ((112))17 = 10
N- 1 = 13
w;;' = 13
1
((N- N)h, =((W;i 1Wiv )),, =((13 · 4))11 =((52))17 = 1
401
(e)
9.53. (a) The tables below list the values for n and k obtained with the index maps.
n2 k,
IO I l I2 0 l 2
n, 0 I(, ' i2 k, 0 0 2 4
l I3 j 4 I5 l l 3 5
As shown, the index maps only produce n = 0, ... , 5 and k = 0, ... , 5.
(b) Malring the substitution we get
(i) Let G(k,, n2] be the N = 2 point DFTs of the inner parenthesis; i.e.,
1
G(k., n 2] = L :i:(3n 1 + n2]Wt• "',
n,=O
This calculates 3 DFTs, one for each column of the index map associated with n. Since the
DFT size is 2, we can perform these with simple butterflies and use no multiplications.
(ii) Let G(k1, n2] be the set of 3 column DFTs multiplied by the twiddle factors.
0 $ k1 $ 1,
{ 0$n2 $2.
(iii) The outer sum calculates two N = 3 point DFTs, one for each of the two -values of k1.
2
X(k, +2J:2] =L G(k,,n2]Wf'"',
n:a=O
The only complex multiplies are due to the twiddle factors. Therefore, there are 10 complex
multiplies. The direct implementation requires N 2 = 62 = 36 complex multiplies (a little less if
you do not count multiplies by l or -1).
403
(f) The alternate index map can be found be reversing the roles of n and k; i.e.,
n = n1 + 2n2 for n 1 = 0, 1; n2 = 0, 1, 2
k=3.l:1 +k2 for .l:1 = 0, l; k2 = 0, 1,2
405
Solutions - Chapter 10
L = (16,ooosa:1 es)(20x10-•sec)
= 320 samples
(b) The frame rate is the number of frames of data processed per second, or equivalently, the num-
ber of OFT computations done per second. Since the window is a,lvanced 40 samples between
computations of the OFT, the frame rate is
(c) The most straightforward solution to this problem is to say that since the window length Lis 320,
we need N 2: Lin order to do the DFT. Therefore, a value of N = 512 meets the criteria of N 2: L,
=
N 2". However, since the windows overlap, we can find a smaller N.
Since the window advances 40 samples between computations, we really only need 40 valid samples
for each DFT in order to reconstruct the original input signal. If we time alias the windowed data,
=
we can use a smaller DFT length than the window length. With N 256, 64 samples will be time
=
aliased, and remaining 192 samples will be valid. However, with N 128, all the samples will be
aliased. Therefore, the minimum me of N is 256.
(d) Using the relation
Starting with
2.-k
!l• = NT'
we find
2,r(-100)
!l-100 = (1000)(1/10,000)
= -2.-(1000) rad/s
2.-(100)
!l,oo = (1000)(1/10, 000)
= 2.-(1000) rad/s
2r(-420)
n_,20 = (1000)(1/10,000)
= -2r(4200) rad/s
2,r(420)
a... = {1000)(1/10,000)
= 2r(4200) rad/s
410
Consequently,
1
Xe(j 0) lns-2r(1000) = 10,000
1
X.(j 0) ln=2r(1000) = 10,000
1
X.(j 0) ln=-2r(4200) = 2,000
1
x.u o> ln=••<<2001 = 2,000
Note that all expressions for Xe(j 0) have been multiplied by the sampling period T = 1/10, 000 because
sampling the continuous-time signal ze(t) involves multiplication by 1/T.
10. 7. The Hamming window's mainlobe is l>.wml = 1~, radians wide. We want
l,,wml < "
8,r
s
.
100
L-1 100
L ~ 801
Because the window length is constrained to be a power of 2, we see that
Lmin = 1024
10.8. All window• expect th, Blackman 5ati5/y the criteria. Using the table, and noting that the window
length N =
M + 1, we find
Rectangular:
4,r
~Wml = M+l
4,r
= 256
= ~<~rad
64 - 25
The resolution of the rectangular window satisfies the criteria.
Bartlett, Hanning, Hamming:
= 255
. .
= 31.875 S 25 rad
The resolution of the Bartlett, Hanning, and Jfarnrning windows satisfies the aiteria.
Blackman:
12,r
=M
12..
= 255
= -"-t~rad
21.25 25
The Blackman window does not have a frequency resolution of at least ,r /25 radians. Therefore,
this window does not satisfy the criteria.
411
Clearly, the cosines in z 1 [n] are too closely spaced in frequency to produce distinct peaks.
In z,[n], we have a small amplitude cosine which will be obscnrred by the large sidelobes from the
rectangular window. The peak will therefore not be visible.
The only signal from which we would expect to see two distinct peaks is z2[n].
LI./= NT
1
Thus, to satisfy the criterion that the frequency spacing between consecutive DFT samples is 1 Hz or
less we must have
LI./
1
s 1
NT s 1
1
T ?:
N
1
T ?: 1024 sec
However, we must also satisfy the Sampling Theorem to avoid aliasing. We therefore have the addition
restriction that,
1
f ?: 200 li2
1
T S sec
200
Putting the two constraints together we find
_l_<T<-1-
1024 - - 200
1
Tmin = 1024 sec
10.11. The equivalent frequency spacing is
.11.n = =
2.-
NT = 2.-
(Sl92)(50µs) 15.34 rad/s
or
LI./ =-.11.n = 2,r 2.44 Hz
412
t.J $ 5
1
$ 5
NT
1
N ?: ST
8000
?: 5
?: 1600 samples
10.13. Since w[n] is the rectangular window and we are using N = 36 we have
JS
X,[k] = L :r[rR + m]e-i(2r/ •>•m 3
m=O
= DFI'{:r[rR + n]}
Because :r[n] is zero outside the range O $ n $ 71, X,[k] will be zero except when r = 0 or r = l.
'When r = 0, the 36 points in the sum of the DFI' only include the section
,i(i,)3n + ,-;(i,)3n
cos(,m/6) = 2
of :r[n]. Therefore, we can use the properties of the DFI' to find
When r = 1, the 36 points in the sum of the DFI' only include the section
,;ci,)On + ,-;(ft)9n
cos(,m/2) =
2
of :r[n]. Therefore, we can use the properties of the DFI' to find
10.14. The signals :r2 [n], :r3 [n], and :r,[n] could be :r[nj, as described below.
413
Looking at the figure, it is clear that there are two nonzero OFT coefficients at k = 8, and k = 16.
These correspond to frequencies
(2w)(8)
w, =
. 128
= -8 rad
(2w)(l6)
"'2 = 128
= !:4 rad
Also notice that the magnitude of the DFT coefficient at k = 16 is about 3 times that of the OFT
coeflicient at k = 8.
• :,;,(n): The second cosine term has a frequency of .26.- rad, which is neither 1r/8 rad or 1r/4 rad.
Consequently, :,; 1 [n] is not consistent with the information shown in the figure.
• :,;2[n]: This signal is consistent with the information shown in the figure. The peaks occur at the
correct locations, and are scaled properly.
• :,;,[n]: This signal is consistent with the information shown in the figure. The peaks occur at the
correct locations, and are scaled properly.
• z,[n]: This signal has a cosine term with frequency .-/16 rad, which is neither 7'/8 rad or .-/4 rad.
Consequently, :,; 4 [n] is not consistent with the information shown in the figure.
• 3:s[n]: This signal bas sinusoids with the correct frequencies, but the scale factors on the two terms
are not consistent witb the information shown in the figure.
• [n]: This signal is consistent with the information shown in the figure. Note that phase infor-
:,; 6
mation is not represented in the DIT magnitude plot.
w;[n] = w0 + >.n
This describes a line with slope ,\ and intercept Wo- Thus,
or
t.fl = 21ft.f = 61.4 rad/s
10.l 7. We should choose Method£.
414
Method 1: This doubles the number of samples we take of the frequeocy wria.ble, but does not change
the frequeocy resolution. The size of the main lobe from the window remains the same.
Method 2: Thu improves the frel/uency resolution ,;,..,., the main lobe from the window get, mu,ller.
Method 3: This increases the time resolution (the ability to distinguish events in time), but does not
affect the frequency resolution.
Method 4: This will decrease the frequeocy resolution since the main lobe from the window increases.
This is a strange thing to do since there are samples of x[n] that do not get used in the transform.
Method 5: This will only improve the resolution if we can ignore any problems due to sidelobe leakage.
For example, changing to a recta.ngular window will improve our ability to resolve two equal
amplitude sinusoids. In most cases, however, "!' need to worry about sidelobe levels. A large
sidelobe might mask the presence of a low amplitude signal. Since we do not know ahead of time
the nature of the signal we are trying to analyze. changing to a rectangnlar window may actually
make things worse. Thus, in general, changing to a rectangular window will not necessarily increase
the frequency resolution.
10.18. No, the peaks will not have the same height. The peaks in V2 (&") will be larger than those in l'i (,;").
First, note that the Fourier transform of the rectangular window bas a higher peak than that of the
Hamming window. H this is not obvious, consider Figure 7.21, and recall that the Fourier transform of
an L-point window w[n], evaluated at DC (w = 0), is
L-1
W(&O) = L w[n]
n=O
Let the rectangular window be WR(n], and the Hamming window be WH(n]. It is clear from the figure
=
(where M L+l) that
L-1 L-1
L "'R[n] > L WH(n]
n=O n=O
Therefore,
WR(&O) > WH(.,.i0 )
Thus, the Fourier transform of the rectangular window has a higher peak than that of the Hamming
window.
Now recall that the multiplication of two signals in the time domain corresponds to a periodic convolution
in the frequency domain. So in the frequency domain, l'i (&") is the convolution of two scaled impulses
from the sinusoid, with the Fourier transform of the L-point Hamming window, WH(&"). This results
in two scaled copies of WH(&"), centered at the frequencies of the sinusoid. Similarly, V.(&") consists
of two scaled copies of WR (ei"), also centered at the frequencies of the sinusoid. The scale factor is the
same in both cases, resulting from the Fourier transform of the sinusoid.
Since the peaks of the Fourier transform of the rectangular window are higher than those of the Hamming
window, the peaks in V2(&") will be larger than those in V,(ei").
10.19. Using the approximation given in the chapter
L-
~ 24.-(A., + 12)
155Am1 +1
we find for A., = 30 dB and A,., = ~ rad,
L = 24.-(30+ 12)
155(,r/40) +1
z 261.1 ➔ 262
415
10.20. (a) The best sidelobe attenuation expected nnder these constraints is
24.-(A,1 + 12)
L ::
155.0..., +1
24r(A., + 12)
512 ::
155(.-/100) +1
A,1 :: 21 dB
(b) The two sinusoidal components are separated by at least .- /50 radians. Since the largest allowable
mainlobe width is 1t /100 radians, we know that the peak of the DFT magnitude of the weaker
sinusoidal component will not be located in tbe mainlobe of tbe DFT magnitude of the stronger
sinusoidal component. Thus, we only need to consider the sidelobe height of the stronger compo-
nent.
Converting 21 dB attenuation back &om dB gives
-21 dB = 20log10 m
m = 0.0891
Since the amplitude of tbe stronger sinusoidal component is 1, the amplitude of the weaker sinu-
soidal component must be greater than 0.0891 in order for the weaker sinusoidal component to be
seen over the sidelobe of the stronger sinusoidal component.
10.21. We have
v[n] = cos(2,rn/5)w[n]
ei2•n/• + ,-;2,n/5]
= [ w[n]
2
In order to label V(eiw) correctly, we must find the mainlobe height, strongest sidelobe height, and the
first nulls of W(e'w).
Mainlobe Height of W(eiw), The peak height is at w = 0 for which we can use l'hopital's rule to find
W(ei 0 ) = 32 cos(l6w)
cos(w/2)
I
w=O
= 32
Strongest Sidelobe height of W(eiw): The strongest sidelobe height for the rectangular window is
=
13 dB below the main peak height. Therefore, since 13 dB 0.2239 we have
First Nulls of W(eiw): The first nulls can be found be noting that W(eiw) = O when sin(l6w) = 0
Thus, the first nulls occur at
2..
w=±-
32
Therefore, JV(&w)l looks like
416
16 I V(J">, I
-2KIS ♦ 2Kl'32
2lr/5 - 2"/32
_,, lt Ill
Note that the numbers used above for the heights are not exact because we are adding two copies of
W(e-'w) to get V(eiw) and the exact values for the heights will depend on relative phase and location of
the two copies. However, they are a very good approximation and the error is small.
10.22. The 'instantaneous frequency' of :r[n], denoted as >.[n], can be determined by taking the derivative
with respect to n of the argument of the cosine term. This gives
0.5
0.45
0.4
0.35
0.3
"
~ 0.25
..,
0.2
0.15
0.1
..
0.05 ---
0
2000 4000 6000 8000 10000 12000 14000 16000
Sample number (n)
417
Here, we see a cosine plot shifted up the frequency (>./2,r) axis by a constant. A$ is customary in a
spectrogram, only the frequencies 0 $ >./2,r $ 0.5 are plotted.
10.23. In this problem, we relate the DFT X(k] of a discrete-time signal z[n] to the continuous-time Fourier
transform X.(jO) of tbe continuous-time signal z.(t). Since z(n] is obtained by sampling z.(t),
z[n] = z.(nT)
which is equivalent to
X(ei"') = { ½Xe (i'f), for OS w < .-
fXc (j~), for.- Sw < 2,r
Since the DFT is a sampled version of X(e;"'),
for0Sk$N-l
we find
'Xc: (·2d)
T" 1NT , for0$k< If
X[k] = { 1X ( •2•<•-Nl) forf:S:k$N-l
~ C: J NT I
Breaking up the DFT into two terms J.iJo, this is nnececessaary to relate the negative frequencies of X.(jO)
'f
to the proper indicies S k S N -1 in X(k].
Method 1: Using the above equation for X[k], and plugging in values of N = 4000, and T = 25/JS, we
find
X,[k] = { 40,000Xc (j2,r -10 · k), for 0 $ k S 1999
40, OOOX. (j2,r · 10 · (k - 4000)), for 2000 $ k S 3999
Therefore, we see this does not provide the desired samples. A sketch is provided below, for a
triangular-shaped X.(jO).
-I ..... .t.t=10Hz
418
Method 2: This time we plug in values of N = 4000, and T = SOµs to find
X,[l:] = { 20, OOOX, U2.- · 5 - l:), for O$ l: $ 1999
20, ooox, u2.- ·5 · (.1: - 4000)) , for 2000 $ l: $ 3999
Therefore, we see this does provide the desired samples. A sketch is provided below.
-1 f-- Af=5Hz
Method 3: Noting that z3[n] = z 2[n] + z2 (n - ~], we get
X3[k] = X2[k] + (-1)• X2[k]
-1 f-- Af=5Hz
419
10.24. (a) In this problem, we relate the DFT X[k] of a discrete-time signal :t[n] to the continuous-time
Fourier tranform X,(j!l) of the continuous-time signal :z:,(t). Since :z:[n] is obtained by sampling
:r,(t),
:r[n] = :r,(nT)
for -1r :5 w :5 1r
which is equivalent to
we find
1X ( ]W'f
,.
•2••)
C 1 for0:,k<f
X[k] = { 'X ( -2•<;,TN))
'f' C J I forf:,k:,N-1
Breaking up the DFT into two terms like this is necessary to relate the negative frequencies of
f
X,(jn) to the proper indicies S k S N - I in X[k].
Y[k] = oX,(j2,r · 10 · k)
is correct. To understand the effect of each step in the procedure, it helps to draw some frequency
domain plots. Assume the spectrum of the original signal :t,(t) looks like
420
Q
-~~(1~0000-==~)-----1-----~211(:-:-:1~0000)
Sampling this continuous-time signal will produce the discrete-time signal z[n], with a spectrum
X(ei°') .
o X{kJ = Samples of X( el "')
Next, we form
X[k], 0$k$250
W(k] = 0, 251 S k S 749
{
X[k], 750SkS999
and find w(n] as the inverse DFT of W[k].
421
W[k]
1/T
Before going on, we should plot the Fourier transform, W(e'w), of w[n]. It will look like
1/T
1---___..!ID,.C:,.l.~,il,r::,.L.~,£;;,.(.__ _ _ .,
2 1t
"
W(e'w) goes through the DFT points and therefore is equal to samples of X,(iO) at these points
for O ::; k :=; 250 and 750 ::; k :=; 999, but it is not equal to X,(;11) between those frequencies.
Furthermore, W(e'w) = 0 at the DFT frequencies for 251 ::, k :=; 749, but it is not zero between
those frequencies; i.e. we can not do ideal lowpass filtering using the DFT.
Now we define
¥[n]={ w[2n], 0 ::, n ::; 499
0, 500 ::; n ::; 999
and let Y[k] be the DFT of y[n]. First note that Y(~) is
Y(~) = !w(~I•)
2
+ !w(ei<w-••l/2)
2
which looks like
1
422
~c:.L;:_1..Q.Q..C:,,J.J.£:.,CI..Q.Q..C>.LJ.£:..C.L..l..£:::..Q.LJ,,L~ (I)
It 2 it 4it
00 JI
Y[n, .\) = L L h[k]z[n + m - k]w[m]•-;•m
JI 00
we find
M
= L h[kJ•-;u X[n - k, >.)
>=O
H the window is long compared to M, then a small time shift in X[n, >.) won't radically alter the
spectrum, and
X[n - k, >.) ::: X[n, >.)
Consequently,
M
Y[n, >.) ::: L h[k],-;u X[n, >.)
::: H(ei')X[n, >.)
10.26. Plugging in the relation for c,,.[m] into tbe equation for /(w) gives
/(w) =
l L-1 [L-1
LU m=~-l) ~ v[n]v[n + m] ,-,wm
]
L-1 L-1
= L~ L v[n] L v[n + m]e-;wm
n=O m.=-(L-1)
Note that for all values of O :o; n :o; L - I, the second summation will be over all non-zero values of v[l]
in the range O S l S L - I. M. a result,
l L-1 L-1
/(w) = - L v[n]eiwn L v[l],-;wt
LUn=0 l=<>
1
= -LU V"(eiw)V(ei"')
1
= - LU
IV(ei"')l 2
Note that in this analysis, we have assumed that v[n] is a real sequence.
424
10.27. (a) Since z[n] has length L, the aperiodic function, c,,[m], will he 2£ - 1 points long. Therefore,
in order for the aperiodic correlation function to equal the periodic correlation fuction, c.,[m], for
0 $ m $ L - 1, we require that the inverse DFT is not time aliased. So, the minimum inverse
DFT length N min is
Nmin 2£-1 =
(b) H we require M points to he unaliased, we can have L - M aliased points. Therefore, for c.,(m] =
c,,[m] for O $ m $ M - 1, the minimum inverse DFT length Nmin is
Nmin = 2L - l - (L - M)
= L+M-1
ws[m] = L
n.=-oo
WR[n]wa(n + m]
00
= I: WR[k - m]wa[k]
00
= L w11[k]w11(-(m - k)]
.t=-oo
= WR(m]. WR[-m]
The convolution above is the triangular signal desaibed by the symmetric Bartlett window formula.
This is shown graphically below for a few critical cases of m.
Consider m = -(M - 1). This is first value of m for which the two signals overlap.
m = -(M - 1) case
•. wR[k+(M-1)]
k
T
k=-(M-1)
m=Ocase
Consider m = (M - 1). This is last value of m for which the two signals overlap.
m = (M-1) case
• WR[k-(M-1)]
T k
k=M-1
T m
m=-(M-1)
From part (a), we know that the Bartlett window can be found by convolving WR[m] with WR[-m].
In the frequency domain, we therefore have,
Ws(_,;,.,) = WR(eiw)WR(•-;w)
= [ l sm(wM/2) -jw(M-1)/2] [ l sin(-wM/2leiw(M-l)/2]
,Ill sm(w/2) ,Ill sin(-w/2)
2
= .!_ [sm(wM/2)]
M sm(w/2)
(c) The power spectrum, defined as the Fourier transform of the aperiodic autocorrelation sequence,
is always no1111egative. Thus, any window that can be represented as an aperiodic autocorrelation
sequence will have a no1111egative Fourier transform. So to generate other finite-length window se-
quences, w[n], that have no1111egative Fourier transforms, simply take the aperiodic autocorrelation
of an input sequence, :i:[n].
we find
. l _ ,-iw[2(M-l)+l]
= ,'w(M-1)
1- e-,w
.
. 1- e-jw(2M'-l)
= ,'w(M-1) _
1- e-JW
e,iw(Jt-1) _ e-;wM
= 1- e-;w
.-;w/2[,iw(M-1/2) _ ,-;..(M-1/2)]
= e-jw/2[eiw/2 _ 0 -;../2]
,;w(M-1/2) _ ,-iw(Jl-1/2)
= eiw/2 _ e-;w/2
= 2jsin{w(M - ½)]
2jsm(w/2)
= sm[w(M - ½ll
sm(w/2)
427
or
. smw-
· [ 2At-lj
2
-
WR(e'"') = sin(w/2)
where 2M - I is the window length. A sketch of WR (&"') appears below.
-2"/(2M-1) 2"/(2M-1)
Bartlett (triangular): W8 (&"') is the Fourier transform of a triangular signal,
-(M-1) 0
II
which is the convolution of a rectangular signal,
x[m)
~
' ' '~ b ,, ~ ,o ~ ~ ...- 112
-
-(M-1)/2 0 (M-1)/2
=
with itself. Tha.t is, wa[m] :[m] • :[m].
Above, we found the Fourier tra.nform of a rectangular window, as
. sin[w2¥J
WR(e'"') = sin(w/2)
where 2M - I was the length of the window. We can use this result to find the Fourier
transform of :[m]. The signal :i:[m] is similar to the rectaz,gular window, the difference being
428
Ws(.,;"') = [x(.,;"')]'
2
= ..!_ [sin(wM/2)]
.M sin(w/2)
A sketch of Ws(&"') appears below.
0 It
"'
-21</M
JI anning/Hamming: Starting with
wH[m] = (o + ,B cos[nn/(M - 1)]) wR[m)
WH(m] = (o + i.,;•m/(M'-l) + ~,-,..,,./(M'-l)) WR(m]
= 0
sin[w (M - ½)l ~ [sin((w- -.f.:r)(M - ½ll]
sin(w/2) +2 sin((w· ,,".. 1 )/2)
-
(I)
-It It
429
(b) Rectangular: The approximate mainlobe width, and the approximate variance ratio, F, for the
rectangular window are found below for large M.
In part (a), we found the Fourier transform of the rectangular window as
2.-n
w = 2M-1
Plugging in n = 1 gives us half the mainlobe bandwidth.
2,r
½Mainlobe bandwidth = 2M-1
4,r
Mainlobe bandwidth =
2M-1
Mainlobe bandwidth
l (M-1)
F = Q L w
2
[m]
m=-(M'-1)
1
= -(2M -1)
Q
2M
Q
Bartlett (triangular): The approximate mainlobe width, and the approximate variance ratio,
F, for the Bartlett window are found below for large M.
M(M - 1)(2M - 1)
}:m' = 6
m=O
430
F
__ Q1 L 1-M(M-l) lml)
(
2
--(M-1)
= .!.(2'f:'(1-!!!.)'-1]
Q . .. =0 M
l [ M-1 ( 2M-1 M-1 ]
L 1 - -M__,
= Q- 2 ,..... L m + -M2.,...
L m2 - 1
= .!_ [2M _ 4(M - l}M 2(M - l}M(2M - 1) _ l]
Q 2M + 6M2
:: h [2M-2M+ 2~]
2M
::
3Q
Hanning/JI arnrning-. We c..an approximate the mainlobe bandwidth by analyzing the Fourier
transform derived in Part (a). Looking at one of the terms from this expression,
n,r "
"' = M - (1/2) + M - 1
n,r .-
:: - +-
M M
"(n + 1)
::
M
So the mainlobe bandwidth for this term is
½Mainlobe bandwidth :: ;,
2
Mainlobe bandwidth :: "
ii
Not< that the peak value for this term occnrs at a frequency w:: 1r/M.
A similar analysis c..an be applied to the other terms in Fourier transform derived in Part {a).
The mainlobe bandwidth for the term
sin[w (M - ½)]
a .
sm(w/2)
431
is also 21t/M. Note that the peak value for this term occun at a frequency w =O.
A sample plot of these three terms, for {J = 2a and large M is shown below.
Noting that
Jl-1 ( )
L cos M":1 = -1
--(M-1)
•
2nn )
M-1
L
m=-{Al-1}
(
COSM-1 =1
432
we conclude
~ 2:(o•+~•)
10.30. (a) Using the definition of the time-dependent Fourier transform we find
13
X(O, k] = L :[m}e-j(2•/7)•m
m=O
< 13
= L :(m],-j(2•/7)•m + L :(1Je-j(2•/7)"'
m=O J=7
By plotting :[m)
'
•••
-1 0 1 2 3 4 5 6 7 8 9 10 n
we see that :[m] + :r[m + 7] = I for O :S m :S 6. Thus,
X[O,k]
•
= L(l),-;c2,11l•m
m=O
= 1:>J'T{l}
= 7o(k]
(b) If we follow tbe same procedure we used in part (a) we find
13
X[n, k] = L :(n + mJe-;C •/7)>m 2
m=O
< 13
= L :[n + mJ,-;(2•/Tl>m + l:=[n + ij,-;<••/7).,
m=O l=?
OTFT: IX (ei°')I
0 OFT:IX}JI
(I)
0 2!t
434
The maximum possible error, Omax error, of the frequency estimate is one half of the frequency
resolution of the DFI'.
1 2.-
Om.ax =
error
=
.
2NT
NT
For the system parameters of N = 32, and T = 10-4, this is
Omax error =982 rad/s
(d) To develop a procedure to get an exact estimate of Oo, it helps to derive X,.[k]. First, let's find
the Fourier transform of :t,.[n] = :t[n]w[n], where w[n] is an N-point rectangular window.
N-l
Xw(~'"') = L e''-'one-;,.,n
n=O
N-1
= L e-j(w-wo)n
=<I
(21rk/N - "'°)(N - I)
l X ,. [kl = +m,r
2
where the m,r term comes from the fact that the term
lin{{2d:/N -.,.)N/2]
lin{(21rk/N - "'°)/2]
can change sign (i.e. become negative or positive), and thereby offset the phase by .- radians. In
addition, this term accounts for wrapping the phase, so that the phase stays in the range [-,r, .-J.
435
In these equations, wo is the estimate found in Part (c). So we would look for values of m in the
range (Lm,,,.nJ, fm,.... l]- Similar expressions bold for p.
Once wo is known, we can find !lo using the relation !lo =wo/T.
10.32. For each part, we use the definition of the time-dependent Fourier transform,
00
:t1 (n + m]w[m],-;>m + b L
-=-00
= aX1 [n, >.) + bX2[n, >.)
(b) Shifting: using y[n] = :r[n - no],
Y[n, >.) = L
.. y[n + m]w[m],-i-'m
m=-oo
436
00
= X[n - fl(),).)
= L00
.,;wo(n+m):[n + m)w[m],-;>m
m=-oo
= L00
.,;w,n:[n + m)w[ml,-;(>-wo)m
m=-oo
= [X[n, ->.)]'
= X'[n, ->.)
10.33. (a) We are given that ¢ 0 (T) = C {zc(t)zc(t + T)}. Since z[n] = Zc(nT),
¢[m] = C {:[n]z[n + ml}
= C {zc(nT)z0 (nT + mT)}
= <Pc(mT)
(b) P(w) a.nd P 0 (fl) are the transforms of ¢[ml a.nd ¢c(T) respectively. Since ¢[ml is a sampled version
of <Pc(T), P(w) a.nd Pc(fl) are related by
(c) The condition is that no aliasing occurs when sampling. Thus, we require that P0 (fl) = 0 for
lfll ;;:: f so that
lwl <"
10.34. In this problem, we are given
• z[n] = A cos(WQn + 8) + e[n]
• 8 is a nniform random variable on O to 2.-
• e[n] is a.n independent, zero mean random variable
437
2
A £ {½cos (2won + wom + 211) + ½ cos(wom)}
Next, note that
£ {e(n]} =0
As a result, the two middle terms drop out. Finally, note that since e(n] is a sequence of zero-mean
variables that are uncorrelated with each other,
-.a.r [½ JV(.1:]1 2
] ::: .P;,(w)
This equation can be used to find the approximate variance of JX(k]i2. We substitute the signal
X(k] for V(k], the DFT length N for L, and use the power spectrum
(b) To achieve a 10 Hz or less spacing between samples of the power spectrum, we require
1
:,; 10 Hz
NT
1
N ~
lOT
20,000
~
10
~ 2, 000 samples
K = Q
L
200,000
= 2048
= 97.66 segments
Il we zero-pad the last segment so that it amtains 2048 samples, we will have K = 98 segments.
(d) The key to reducing the variance is to use more segments. Two methods are discussed beltrw.
=
Note that in both methods, we want the segments to be length L 2048 so that we maintain the
frequency spacing.
439
(i) Decreasing the length of the segments to /oth their length, and then zero-padding them to
=
L 2048 samples will increase K by a factor of 10. Accordingly, the variance will decrease by
a factor of 10. However, the frequency resolution will be reduced.
(ii) U we increase the data record to 2,000,000 samples, we can keep the window length the same
and increase K by a factor of 10. ·
10.37. (a) Taking the expected value of
~[m] = 21
1(
1•
-•
I(w),,;"'md,.,
gives
we find
= _1_1• [.!..1"
2,rLU -•
P,.(6)
2r -•
C...,(,,;1.,-•>j_,j"'mciu] dB
Note we can change the limits of integration of the inner integral to be [-", ,rj because we are
integrating over the whole period. Doing this gives
E {¢[ml} = _1_ 1•
2,r LU -•
P,,(e),,;•m
2,r -•
[.!..1'
c• .,(e'"'')_,j"''md,.,'] dB
= 2
,r~U 1_: P.,(6),,;•m {c.,.,[m]}dB
= L~c,..,[m] ( 2~ 1_: P•• (e),,;•mdB]
l
= wc....lmJ;•• [ml
(b)
¢p(ml = L
..
By applying the sampling theorem to Fourier transforms, we see that
¢,,(m+rNJ
E ~p[ml} =
..
r=-oo
L E ~•• [m + rNJ}
r=-oo
1 ..
= LU L
c,..,[m+rN]¢,,[m+rNJ
--oo
which is a time aliased version of E {¢,,[ml}.
(c) N should be chosen so that no time aliasing occurs. Since ¢,,[ml is 2L-1 points long, we should
choose N ~ 2L.
= Ql [M-1
L :r[n]:r[n + ml
.....
M-1 M-1 ]
+ L :r[n + M]:r[n + M +ml+ ... + L :r[n + (K - l}M]:r[n + (K - l}M + m)
..... .....
= bL LK-1 JJ-1
i=O n=O
:r:[n+iM]:r[n+iM +ml
K-1
= .!. :E c.[mJ
Q.,..
where
M-1
e;(m] = L :r:[n + iM]:r[n + iM + m] forO:Sm:SM-1
.....
(b) We can rewrite the expression for e;[m] from part (a) as
M-1
e;(ml = L :r[n + iM]:r[n + iM + ml
.....
JI-J /11-1
= L :r(n+iM]z[n+iM +m)+ L 0-z[n+iM +ml
,,,_,
=<> =M
= L :z:,(n)11,[n +m)
.....
where
.r,[n) = { .r[n + iM], 0::,n::,M-1
0, M::,n:SN-1
!
441
and
11,[n] = :r[n + iM]
Thus, the correlations e;[m) can be obtained by computing N-point linear correlations. Next, we
show that for N ~ 2M - 1, circular correlation is equivalent to linear correlation.
can be expressed as
ey.[m) = Cs,[-m]
N-1
= L :r,[((n - m))N]!l,[n)
n=O
N-l
= L :r:[((m - n))N)!f,[n]
n=O
where :r'[n] = :r[-n]. Note that this is a circular convolution of :r,[-n] with y,[n). Thus, we have
expressed the circular correlation of :r,[n) with y,[n] as a circular convolution of :r,[-n] with 11,[n].
Now recall from chapter 8 that the circular convolution of two M point signals is equivalent to
their linear convolution when N ~ 2M - l. Since we can express the circular correlation in terms
of a circular convolution, this result applies to circular correlation as well. Therefore, we see that
if N ~ 2M -1,
e;[m] c,[m) = for0SmSM-1
Thus, the minimum value of N is 2M - l.
(c} A procedure for computing ¢,.[m] is described below.
step 1: Compute X,[k] and Y;(k], which are the N ~ 2M - l point DFTs of :r,[n] and 11,[n].
=
step 2: Multiply X,[k] and Y;"[k) point by point, yielding C,[k) C,[k) X,[k]Y,•[k). =
step 3: Repeat the above two steps for all data (K times}, then compute
• J K-1
+•• [k) = Q L C,[k) for0SkSN-1
i=O
If
2 · log2 N · K = KNlog2 N, for step I
KN, for step 2
N, for divide by Q operation in step 3
/f log2 N for step 4
c[n,m]
00
we find
c(n,m] = 2.1•
2• -•
IX[n, .>.)i'ei'md.>.
=
00
L L
00
l=-oo r=-oo
:i:[n + l]w[l]:i:[n + r]w[r] ( 1,r
2
1· ._
"'
)
.-,>(-l+r)_,i>md>.
The 6[m - I + r] term is zero everwhere except when m - I + r = 0. Therefore, we can replace the
two sums of land r with one sum over r, by substituting l == m + r.
00
c(n, m] = 21
. 1•-• IX[n, >.)I' _,;>m d>.
00
where
h,,.{r] = w[-r)w[-(m + r)J
(c) To compute c[n,m] by causal operations, we see that
h,,.[r] =w[-r]w[-(m + r)]
requires that w[r] must be zero for
-r < 0
r > 0
and w[r] must be zero for
-(m+r) < 0
m+r > 0
r > -m
Thus, w[r] must be zero for r > min(0, -m). If m is positive, then w[r] must be zero for r > 0.
This is equivalent to the requirement that w[-r] must be zero for r < 0.
(d) Plugging in
2: 0
w[-r} = { 0,a' ' r
r<O
into hm(r] = w[-r]w(-(m + r)], we find
r ~O,r ~-m
h,,.(r] = { 0, otherwise
Ta.king the z-transform of this expression gives
00
Hm(z) = L h,,.(r}z-•
445
Again we have assumed that mis positive. H lzl > a2, then
a'"
Hm(z) =
1- a 2 z- 1
h,,,[r] = a'"6[r] + a 2 h,,,(r - 1]
Using this in the equation for c[n, m] gives
00
= c'":r[n]:r(n - m] + c2 c(n - l, m]
a2
[ ] -{
w-r - ra', r;::O
0, r <0
To get the z-transform H,.(z), recall the z-transform property: r:r(r] +-> -z•~•!_ Using this
property, we find
ra2'u[r] ~
(1- a•z-1)2
a 2 z- 1 (1 +a2 z- 1 )
(1- a•z-•)3
we get
00
y(n] ,
, .
x[n]
.
X
. '"\
-+, ,. -+ . c[n,m)
z-1 ·,I,; z-1
I,
am+2(m+ 1)z-m
-~
" 3a2
.
.. -~
I, ·v -1
-3a' 2
z-1
I ,
. X '
I,
z-1 a6
'
z-'
am+4(1-m)z-m '" " .
,'
= L :r[n - m]ho(m].,;;.m
m.-==-oc
L <1Z1[n + m]ho[-m]e-;>m + L b:i:2(n + m]ho[-m)e-;>m = aX1 [n, .>.) + bX2 [n, .>.)
m=-co --00
Note that most typical window sequences are lowpass in nature, and are centered around a fre-
= =
quency of w 0. Since Ho(e'") W(,-i") is the Fourier transform of a window which is Jowpass
in nature, the signal S (e'") is also lowpass.
The signal s(n] = X[n, .>.) is mnltiplied by a oomplex exponential ei•n. This modulation shifts the
frequency response of S(e'") so that it is centered at w = >..
h{n] = s{n]e''"
H(e"") = S (,;cw->))
Since S(e'"') is lowpass filter centered at w = 0, the overall system is a bandpass filter centered at
w=>..
448
(c) First, it is shown th.at the individual outputs Y•[n] are samples (in the,\ dimension) of the time-
dependent Fourier transform.
00
= L :i:[n + m]w[mje-;h>m/N
m=-oo
= X[n, ,\)J.=2d/N
Next, it is shown that the overall output is y[n] = Nw[O]:i:[n].
N-1
y[n] = L Y•[n]
N-1 oc
= L L :i:[n + m]w[m],-;2,>m/N
i:=O m.=-00
oo N-1
= L L x[n + m]w[m],-;2dm/N
oo N-1
= L :i:[n + m]w[m] L ,-;2,>m/N
m=-oo i=O
N6lmJ
= Nw[OJ:i:[n]
(d) Consider a single channel,
decimator expander
X ( ,i(w+>,l) Ho(,;,-,)
so the output of the decimator is
½R-1
~ X (,;«w-2,1)/R+>,>) Ho (,;<w-2,1)/R)
R-1
....L Ho (ei<w->,)) Go (ei<w->,>)
N-1
+ L X (ei(w-2sl/R)) ½L Go (ei(w->,>) Ho (ei<w->,-2•1/R))
l=l k=O
Aliasing Component
(e) Yes, it is possible. G0 (_,iw) = NH0 (&w) will yield exact reconstruction.
(f) See chapter 7 in "Multirate Digital Signal Processing" by Crochiere and Rabiner, 1983.
(g) Once again, we consider a single channel,
decirnator expander
From Part (a), we know that the output of the filter ho(n] is
00
Therefore,
N-1 cc oo
= L L 90(n - lR] L :i:(m]ho[RI - m]e-;2<•(m-n)/N
•=o l=-oo m=-oo
N-1
= NL L 9o[n-lR]ho[lR-n+rN]:i:(n-rN]
co 00
y(n] = L h(k]:r(n - k]
k=-oo
= L L h(k]h(/]9'>.. [I + m - k]
~-ool=-oo
9'>.. (1 + m - k] = u;o[I + m - k]
Substituting this into the expression for 9'>,.[m] gives
00 00
4>,,[m] = ~ L L h[k]h[l]o[I + m - k]
00
= u; L h(l + m]h[I]
l=-oo
Note that 00
(b) Taking the DTFT of 1/>n[m] will give the power density spectrum t,.(w).
+.. (w) = m=f;oc { ~ ,f:.. h[I + m]h[ij} •-;.,m
oc oc
= ~ L h[ij L h[l + m],-;.,m
l=-oo m=-oo
The relation for +.. (z) above is found by multiplying two polynomials in z. The highest power
of z in +.. (z) is z>L which arises from tbe multiplication of the k = 0 and I= M coefficients.
The smallest power of z in +n(z) is z-M which arises from the multiplication of the k = M
and I= 0 coeflicents. Thus, ,.,(m] is nonzero only in the interval 1ml SM.
(d) For an AR process,
H(z) = bo
_.N a.11:z -•
1 - L....t;:=1
bo
Since
+.. (z) = a;H(z)H•(z)
X X
Im
X X
Re
Nth order zero X X
Nth order zero at z = ~
X X
By performing a partial fraction expansion on +,,(z) we find that each pole pair contributes a
sequence of the form A•or' 1
~<{ml
and therefore
~·-
~ M
N
'-•(m] = L A•ol"'
k=l
1
454
9'>.,[mJ = q,.,[-mJ
= £ {11[n - mJy[n]}
= £ { 11[n - mJ (t.aw[n - k) + :z:[nJ)}
N
= L "•£ {y[n - m]y[n - kl} + £ {y[n - m):z:[n]}
t=t
N
= L at9'>.,[m - k] + 9'>,.[-m)
t=l
N
= L 4t¢..,,[m - kJ + ¢,,[mJ
l:=l
Form= 0,
N
9'>.,,[oJ = L ... ¢.,[-kJ + ¢.,[oJ
bl
¢,,[OJ = £ {:z:(nJy[n]}
Note that :r:(nJ is uncorrelaled with the y(n - k], for k = 1, ... , N. Therefore,
¢,,(OJ= a;
Thus,
N
q,.,,[OJ = L ...q,.,,[-k] + ~
1st
N
= L citq,.,,[kJ + a!
t=l
455
= L a.¢.,(m - k]
lr-1
¢.. (m - k) = ¢yy(k - m]
= ¢.,(Im - kl]
Thus,
,, ,,
L a•¢.. [lm - kl] = L a.¢.. (m - k]
bl ,l;:l
x(n) = x,(nT)
= :6 t GY.'ej(2•/16)kn
t:-4
1/2
••• •••
1/4
16. k
-16 -4 0 4
z(n] =
•
..!_ ""' X(k]ei<•• 1••>0n
16 L,
-=-•
7
= .!. I: X(k]ei<
16 ..,_,
2•/ll)b
= IDFS{ X(k]}
456
However, since the period we use in the sum of the IDFS is unimportant we ·can also write
,.
.:i:(n] = .!_ L X[k]~(2r/16)•n
16 ....
= IDFS{X(k]}
= IDIT{Xo(k]}
where Xo[k] is the period of X[k] starting at zero, i.e.,
G[kj
112 112
114 1/4
118 118
1/16 1/16
k
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
We can apply the same idea as we did in part (a), except now the DFS and DFT siu should be
32 instead of 16. Going through the same steps will lead us to the sequence Q(k] that looks like:
Q{kj
1/2 112
1/4
20 31 k
0 4 8 12 16 24 28
457
(Here we have assumed o = 1). We see that we can interpolate in the time domain by zero padding
in the middle of the D FT samples.
10.43. (a) Using the relation,
0$_k$.N/2
N/2$.k$.N
where N is the DFT length and T is the sampling period, the continuous-time frequencies corre-
sponding to the DFT indices k = 32 and k = 231 are
32
(2:xi){l/20, 000)
= 2500 Hz
231-256
(256)(1/20, 000)
= -1953 Hz
(b) Since
i[n] = :i:(n]wR(n]
the DTFT of :r[n] is simply the periodic convolution of X(e'"') with WR(<'"').
(c) Multiplication in the time domain corresponds to periodic convolution in the frequency domain, as
shown in pa.rt (b). To evaluate this periodic convolution at the frequency w32 = 2,r(32)/L, {where
L = N = 256) corresponding to the k = 32 DFT coeflicient, we first shift the window W0 ,,(e'"')
to "'32· Then, we multiply the shifted window with X(e'"'), and integrate the result. In order for
!
l, w = 0
w.,,(e'"')= o, w=±21r/L
0, 21fk/L, fork= 2,3, ... ,L-2
Note that we are only specifying W0 ,,(e'"') at the DFT frequencies w = 2,rk/ L, fork = 0, ... , L-1.
(d) Note that the L point DFT of a rectangular window of length Lis
L-1
WR(k] = ~)l)e-/2d/L
=<I
1- e-;z.i
= 1- e-j2•11L
= U(k]
W0 ,,(e'"') is only specified at DFT frequenciesw = 2rk/L, and it can take on other values between
these frequencies. Therefore, the DTFT of w.,,(e'"') can be written in terms of WR(e'"') and two
shifted versions of WR(<'"').
w ••,[n]
2/256
1/256
10.44. (a) After the Jowpass filter, the highest frequency in the signal is 6,,;. To avoid aliasing in the
downsampler we must have
twM $ ,r
,r
M $ 6w
N
$
2kA
N
Mma:,;=-
2k,.
(b) The Courier transform of x,[n] looks like
so M = 6 is the largest M we can use that avoid• aliasing With this choioe of M the fourier
transform of x,[n] looks like
459
-It 0 It (I)
Taking the OFT of z,[n] gives us N samples of X,(eiw) spaced 27f/N apart in frequency. By
examining the figures above we see that these samples correspond to the desired samples of X(e;w)
which will he spaced 2.0.,,,/N apart inside the region -.0.,,, < w < t:.w.
Note that after downsampling the endpoints of the region alias. Therefore, we cannot trust the
values our new OFT provides at those points. However, the way the problem is set up we already
know the values at the endpoints from the original OFT.
(c) The system p{n] periodically replicates XN[n] to create XN[n]. Then, the upsampler inserts M -1
zeros in betweeen each sample of .XN[n]. Thus, the samples k, - I;,._ and k, + I;,._ which border
the zoom region in the original OFT map to M(k, - k,._) and M(k, + k,._). The system h[n] then
interpolates between the nonzero points filling in the "missing" samples. Since the linear phase
filter is length 513 it adds a delay of M/2 = 512/2 = 256 samples so the desired samples of XNM[n]
now lie in the region
k; = Mk,+ 256
k~ = Mk,._
X(ei"'J
It (I)
460
0 N-1 k
After periodically replicating and upsampling by M we have a signal that looks like
M-1 zeros
...
l
•••
0 M(N-1) n
Filtering by h[n] then interpolates between the samples. XNM[n] is shown below if we assume that
h[n] is the ideal zero phase filter. The points with an x correspond to the interpolated points.
~[n]
interpolated points
.. . i •••
0 M(N-1) n
where
k' = Mkc
C
k'c,. = Mkc,.
Solutions - Chapter 11
11.1. Using the fact that :t,[n] is the inverse transform of?u we get
?u{X(,;w)} = 2 _ ,.,;w - ,u-;w
Im{X(,;w)} = 2asinw
11.2. Taking the inverse transform of ?u{X(,;"')} = 5/4- wsw, we get
5 1 1
:z:,(n] = o[n] - iln + l] -
4 2o(n - l]
Since x(n] is causal, we can rewver it from :z:,[n]
5
x(n] = 2:t,[n]u[n] - :z:,(O]o(n] = 4"[n] - o[n - l],
1
:z:[n] = o(n] - iln - l]
but this does not satisfy the conditions on z[n] given in the problem statement.
1
:.[n] =o[n - l] - -o[n - 2]
2
wbicb satisfies all the constraints. The idea behind this choice is that cascading 4 signal with an allpass
system does not change ~be magnitude squared response.
Another choice that works is X(,;w) = ½(1- :ze-;.,)•-;w for which we get
1
:z:(n) = 26(n - 1) - o(n - 2]
The idea behind this choice was to flip the r.ero to its zeciprocal location outside the unit circle. This
has the same magnitude squared response up to a scaliDg factor; hence, the ½ierm.
466
1 1 1
:z:,[n] = 2o(n] - ;.6fn + 2] - iln - 2]
. 1 1
X,(e"") = 2 - 2cos2w.
where X,(dw) = !(X(~) +X•(~)] is the conjugau ,vmmetric part of X(~). Since X(dw) = 0 for
-.- :S w < 0 we have
= { 2X,(&"'), 0 :S w <"'
0, otherwise
= { l-cos2w, 0:Sw<O'
0, otherwise
Thus,
and
Im{X(eiw)} = 0.
About Notation: XR(eiw) with a capital R is the real part of X(eiw). X,(e'w) with a small r is the
conjugate symmetric part of X(&"') which is complex-valued in general.
11.S. The Hilbert transform can be viewed as a filter with frequency response
X,(ei"') = H(ei"')X,(ei"')
= -j.-o(w - wo) + j,ro(w + wo)
z,(n] = sinwon
(b) Similarly, z,(n] = - cos won.
(c) ;r,[n] is the ideal low pass filter
0:Sw:Sw.
-we:Sw:$0
"'• :S lwl :S,,.
1- cosw.n
I 1 2,r1
:tin=-
-we
1e"'" uw---
1
2,r O
Jc- ur.&1=
,m
467
3 3
z 0 [n] = -i[n + 4] + a[n + lj - a[n - 1] + i[n - 4]
Because z[n] is real and causal we can recover most of x[n], i.e.,
z[n] = 2z0 [n]u[n] + z[O]o[n]
= z[O]o[n] - U[n - lj + 35[n - 4]
The extra information given to us allows us to find z[O],
6 = X(e'"')I., ...
"" z[n]e'"(OJ
= I:
n.=-oc
= z[O]- 2 + 3
Plugging this into our equation for z[n] we find
11.7. (a) Given the imaginary part of X(&"'), we can take the inverse DTFT to find the odd part of z[n],
denoted z 0 [n].
Any z[O] will result in a correct solution to this problem. Setting z[O] = 0 gives the result
z[n] = -<l[n - lj - U[n - 2]
468
(b) No, the answer to part (a) is not unique, since any choice for :r(O] will result in a correct solution.
11.8. Using Euler's identity and the fact that :r0 (n] is the inverse transform of jX1(ei"') we find
jX1(~) = 3jsin2w
= 3( ,;2w -;e-;2w)
3
:r.(n] = 2(6[n + 2] - o(n - 2])
Because :r(n] is real and causal we can recover all of :r(n] except at n =0,
:r(n] = 2:r0 (n]u(n] + :r(O]o[n]
= -36(n - 2] + :i:(O]o[n]
Therefore,
:r[n] + :r(-n]
:r,(n] =
2
(-36(n - 2] + :i:(O]o[n]) + (-36[n + 2] + :r[O]o(n])
= 2
3 3
= - iln + 2] + :i:(O]o(n] - iln - 2]
XR(ei"') = -~ei..,
2
+ :r(O] - ~,-;2w
2
= :r[O] - 3 cos 2w
Thus, XR2(ei"') and XR3(ei"') are possible if :r(O] = -1 and :r(O] = 0 respectively.
11.9. (a) Given the imaginary part of X(ei"'), we can take the inverse DTIT to find the odd part of :i:[n],
denoted :i:0 (n].
3+l+z(O] = 3
z(O] = -1
Therefore,
z(n] = -3o(n - l] - o[n - 3] - o(n]
(h) Yes, the answer to part (a) is unique. The specification of X(eiw) at w = ,r allowed us to find a
unique z(n].
11.10. Factoring the magnitude squared response we get
IH(,)1
2 = {I - ½z-1
)(1 - ½z)
(I+ 2z- 1 )(1 + 2z)
= H(z)H"(I/z")
Since h(n] is stable and causal and has a stable aod causal inverse, it must he a minimum phase system.
It therefore has all its poles and zeros inside the unit circle which allows us to uniquely identify H(z)
from JH(z)! 2 • ·
H(z) = 1+2z
izl > ½
Thus,
0 < w < ,.
< 0
Therefore, the real part of X (&"') is
_;"'ll2
IH( e- = -10 - -2 cosw = 1 - -2 cosw + -1 = ( 1 - 1___ ,.,
. ) ( 1 - -e'
1 ·.,)
9 3 3 9 3 3
= H(ei"')H"(ei"')
H(ei"') = 1- !,-;.,
3
1
h[n] = o(n] - 1]
3o[n -
(b) No. We can fiud a new system by taking the zero from the original system and ftipping it to its
reciprocal location. This only changes the magnitude squared response by a scaling factor. If we
compensate for the scaling factor the two magnitude squared responses will be the same. Thus, we
find
= !c1
3
- 3,-;"'l
1
h(n) = 3"[n] - U[n - 1]
Taking the inverse DTFT of XR(e 1"') gives the conjugate-symmetric part of :i:[n], denoted as :i:,(n].
1 1 1 1 1 1
:i:,[n] = - ,a(n + 2) + 2"1[n + 1) + ,a[n + 1) + 6[n) + 2o[n - 1) - ,a(n - 1) + ,a[n - 2]
2 2 2 2
Using the relation x(n] = 2x,[n]u[n] - x,[0]o[n],
XR(&w) = f: (D • cos(kw)
= 1+ ~
k=O
= o[n] +
z[n]-z[-n]
Ler
l=l
00
2 6[n + k]
:z:0 [n] =
= 2I ~ 2 00 er
2
(o[n + k] - o[n - kl)
472
j ~ (1)' sin(kw)
00
= 2
j ~ (1)' sin(kw)
00
= 2
Thus,
= L (1)'
00
X1(e'"')
_
....
2 sin(kw)
11.15. Given X,(eJ"'), we can take the inverse DTFT of jX,(,;"') to find the odd part of z(n], denoted z 0 (n].
Im {X(.-J"')} = sinw
= _!_e3"' _ 1-e-iw
2j 2j
L
n=-oo
z[n]=3
-1 +z(O] = 3
z(O] = 4
Therefore,
z(n) = 4o(n] - o[n - l]
11.16. Using Euler's identity and the fact that z,[n] is the inverse transform of XR(eJ"') we have
XR(ei"') = 2 - 4cos(3w)
= 2 - 2(eJ"' + ,-;.. )
= 2+4
"F 7
Thus, there is no real, causal sequence that satisfies both conditions.
11.17. There is more than one way to solve this problem. Two solutions are presented below -
Solution 1: Yes, it is possible to determine :r[n] uniquely. Note that X(k], the 2 point DFT of a real
signal :r(n], is also real, as demonstrated below.
1
X(k] = L :r(n]e-;2...•12
n=O
1
X(k] = L:r[n](-clr•
n=O
Thus,
X[O] = :r(O] + :z:[l]
X(l] = :z:[O] - :z:(l]
Clearly, if :z:[n] is real, then X[k] is real. Therefore, we can conclude that the imaginary part XI[k]
is zero.
Therefore, the inverse DFT of XR[k] is :z:[n], computed below.
1
:z:(n] = ½L XR[k],;••n>f•
k=O
1
:z:(n] = ½LXR(k](-l)n>
k=O
1
:z:[O] = 2{XR(O] + XR(l])
= -1
1
:z:(l] = 2
{XR(O] - XR(l])
= 3
Thus,
:z:(n] = -5(n] + 36(n - l]
Solution 2: Start by making the assumption that X(k] is complex, i.e., X1[k] is nonzero and XR[k] =
U(k] - 4J(k - l]. Then, because z.,.(n] is the inverse DFT of XR[k] we find
z.,.(n] = ! t
2.,..
X R[kj,;••,../2
1
= ! LXR[k]{-l)n>
2.,..
474
and
= 21 (XR[0] + XR[l])
= -1
= 21 (XR[0] - XR[l])
= 3
z,,[n] = -o[n] + 36[n - l]
Because z[n] is real and causal, we can determine it from z.,[n]
z.,[n], n=0
2:t.,[n], 0 < n<N/2
z[n] =
z.,[N/2], n=N/2
0, otherwise
With N = 2 we bave
z[n] = -o[n] + 36[n - 1)
If we began by making the assumption that X[k) was real, i.e., Xr[k) = 0 and X[k) = XR[k] =
26[k) - 46[k - 1) than by taking the inverse transform we find that
z[n] = z.,[n] = -o[k] + 36[k - 1)
This is the same answer we got before. Since there was no ambiguities in our determination of
z[n], we conclude that z[n) can be uniquely determined.
The next problem shows that when N > 2, we cannot necessarily uniquely determine z[n) from
XR[k] unless we make additional assumptions about z[n] such as periodic causality. When N > 2
the two assumptions we used above leads to two different sequences with the same XR[k].
11.18. Sequence 1: Fork= 0, 1,2 we have
XR[k] = 9o(k] + 66[k - 1) + 66[((k + l)),]
and XR[k] = 0 for any other k. Using the DFT properties and taking the inverse DFT we find for
n =•l,2,3
z,,[n) = 3 + 2 ( ,'(2•/l)n + •-j(2•/3)n)
= 3 + 4 cos(2irn/3)
= 76[n] + o[n - 1) + o[n - 2)
If we let x[n) = z,,[n) we have the desired sequence.
Sequence 2: If we assume z[n) is periodically causal, we can nse the following property to solve for
x[n] from z,,[n):
z.,[0], n = 0
z[n) = 2:t.,[n), 0 < n <
{ 0,
1
otherwise
Note that this is only true for odd N. For even N, we wowd also need to handle the n = N /2
point as shown in the chapter. We have
z.,[0J, n = 0
z[n) = 2:t.,[n), n = 1
{ 0, otherwise
= 76[n) + 26[n - 1)
475
11.19. Given the real part of X[k], we can take the inverse DIT to find the even periodic part of :i:[n],
denoted 2'cp[n].
Using the inverse DIT relation,
we find
1
:i:cp[0] = 4(4+1+2+1)=2
:i:cp[l] = !.(4+j-2-j)=!.
4 . 2
1
%cp[2] = -(4-1+2-1) = 1
4
2'cp[3] = -I (4 - ,. - 2 +1.) =-
1
4 2
Thus,
1 1
:i:cp[n] = 2o[n] + i5[n - 1] + o[n - 2] + i5[n - 3]
Next, we can relate the odd periodic and even periodic parts of :i:[n] using
:i:cp[n], 0 < n < N/2
z.,.[n] = -zcp[n], N/2 < n:, N - 1
{ 0, otherwise
Performing this operation gives
1 1
z.,.[n] = -o[n - l] - -o(n - 3]
2 2
Taking the DIT of z.,.{n] yields jX1[k]. Using the DIT relation,
N-1
jX1(k] = L :i:.,.[n]W"'
=<>
we find
iX1(0J = (o+½+o-½) =0
iX1(3J = (o+½+o+~) =i
Thus,
jX,[k] = -;o[k - l] + jo[k - 3]
11.20. As the following shows, the second condition implies :i:[0] = 1.
z[0] · = !_
6.,_.
t X[k]ei<••l•l""I
-
•
= ~ LX[k]
=
....
1
. 476
:t2(n]- :t 2[((-nll•l
= 2
1
= -3 (o(n - 4] - o[((n + 4)).]) -
.
!3 (o[n - 5] - 6[((n + 5)).])
:t3(n] - :t3(((-n)) 6 ]
:t.,.(n] = 2
1 1
= 3
(o(n - l] - 6(((n + 1))6 ]) -
3 (6(n - 2] - o[((n + 2)) 6])
For n < 0 or n > 5, these sequences are zero. Since the transform of :t.,(n] is jX,[k] we find for
k = 0, ... ,5
= -~jsin(rk/3) + ~jsin(2rk/3)
= j_!_(-6[k-2]+6[k-4])
../3
Thus, both x 2 [n] and :t 3 (n] are consistent with the information given.
XR(pe.iw) = U(p,w)
= l+p- 1 acosw
Since au
7ii
= ! av we have
p&.1 '
Since ~ = -¼ ~ we have,
¥.' -¼~
ap-• sinw + K'(p) = ap-• sinw
Thus,
K'(p) = 0
K(p) = C
477
Since :i:(n] is real V(p,w) is an odd function of w. Hence, V(p,O) = 0, implying that C = 0.
Therefore,
XR(&"') = 1 + ocosw
= Q . Q
1 + -t:'W + -e-JW
2 2
.
0
:i:,(n] = o(n] +
Q
2o(n + l] + 2o(n - l]
Because :i:(n] is real and causal, we can recover ,: 0 (n] from x,[n] as follows
{ ,:,(n], n>O
:i: 0 [n] = 0, n=O
-:i:,(n], n<O
0 Q
= --o(n + 1] + -o[n - l]
2 2
Thus,
Note that we could have obtained :i:(n] directly from ,:,(n] as follows
2 2Z -N/2 1 -N/2
1-z-l l-z- 1 - +•
1- z-N/2 + z-1 _ z-1-N/2
= l-z- 1 l•I -Io
Sampling this we find
V'N(k] = UN(<2d/N)
1- (-1)• + ,-;2d/N _ ,-j2d/N(-J)t
= I - ,-;2d/N
478
When k = 0 we get 0/0 which, if the function was continuous, you would use l'Hopital's rule. In this
case the function is disaete so that is not available to us. One route to the answer is to use the definition
oftbe DFS
17N[0] = f, iiN[n]e-i~ln1
l:=O k=O
N
= LiiN(n]
l=O
= N
N, =
k 0,
UN[k) = -2j cot(rk/N), k odd,
{
0, keven,kj0
11.23. (a) Because :r,,[n] is the inverse DFT of X11(k) we ha.ve for n = 0, ... ,N - 1 and k = 0, ... ,N - 1
X[k)+ X"[k]
X11[k] = 2
:r[n] +x•[((-n))N]
:r.,[n] = 2
or equivalently, if we periodica.lly extend these sequences with period N
_[ l =
%en
z[n] + z(-n]
2
Note that since the signal is real f"(-n] = f[-n].
The first period of f[n] is zero from n = M ton= N - 1. JIN = 2(M - 1) there is no overlap of
f(n] and f[-n] except at n = 0 and n = N/2. We can therefore recover f[n] from f,[n] with the
following:
_ { ~,[n], n=l, ... ,N/2-1
:i:[n] = :r,[n], n = 0, N/2
0, n=M, ... ,N-1
If we tried to make N any smaller, the overlap of f[n] and z[-n] would prevent the recovery of :r(n].
Consequently, the smallest value of N we can use to recover X(.i:] from X11(.i:] is N = 2(M - 1).
(b) If N = 2(M - 1),
where
2, n = 1,2, ... ,N/2-1
UN[n] = 1, n = 0, N/2
{ 0, otherwise
= 2u[n] - 2u[n - N/2] - o[n] + o[n - N/2]
Taking the DFT of :r[n] we find
where
= DFT{2u[n]-2u[n - N/2] - o[n] + 6[n -N/2]}
1-(-l)k +e-j2d/N -e-j2d/N(-l)k
= k=0, ... ,N-1
1_ e-j2d/N
N, k=0,
= -2j cot(,rk/N),
{ 0,
0 < k < N - 1, k odd
otherwise
H,(&"') = 2j
= [HER(•;"')+ HoR(ei"')) + j[HE1(ei"') + Ho,(ei"')]
2j
[HER(ei"') - HoR(ei"')) - ;[HE1(e'"') - Ho,(e'"'))
2j
Thus,
H,._(ei"') = HER(e'"') Hc(ei"') = BE1(e"")
Hs(e'"') = Ho1(e'"') HD(e"") = -HoR(e'"')
480
. I - jH(,,;<.,-tl)
H1p(e"") = 2
The simplification in the last step used the fact that hip[n]
=
1/2 for n O.
...
= m(•n(•I is zero for even n and equals
.
Find h1p[n] :
Taking the inverse DTFT of H1p(,,;") yields
Using the fact that h[n] is zero for n = 0 and n even we can reduce this to
(c) The linear phase causes a delay of n• = M/2 in the responses. H nd is not an integer, then we
interpret hJp[n] and h(n] as
sin(.-(n-n•)/2)
= .-(n - nd)
2 sin2 (.-(n - nd)/2)
= .- (n - nd)
Then,
where i.[n] and h1pln] are the causal FIR approrimations to h(n] and h1p[n]. Similarly,
•
h1pfn] ={ sin(,r(n - n.)/2)h.[ I
2 • n +2 n
!o[ - n,lw [nI' Meven
sin(,r(n - nd)/2)h{n}, Modd
(d) The lowpass filter com,sponding to the first filter in the example looks like
481
1.2
1 t------------
0.8 M = 18, ll = 2.629
0.6
0.4
0.2
The lowpass filter corresponding to the second filter in the example looks like
1.2
11--------
0.8 M=17,jl=2.44
0.6
0.4
0.2
11.26. (a) The example shown here samples at the Nyquist rate of T = ,r /(fl, + <lfl) as in the chapter's
example, but the bandpass signal is such that <lfl/(fl, +an)= 3/5. Then, 2,r/(<lf!T) = 10/3.
482
1 sC(j!l)
nC , ,nC+.!IOC n
1/T
2/T
=
(b) U 2,r/(A!lT) M + e, where Mis an integer and e some fraction, then using the Nyquist rate of
2,r /T = 2(0, + AO) will force ~ecirnalion by M. M. jnst shown, this choice for T causes 54(eP")
to have intervals of zero. Instead, choose T such that 2,r/(AOT) is the next highest integer
2,r
A!lT =M+l.
Y(eiw) contains roughly half the frequency spectrum as X(eiw), we can reconstruct X(eiw) from Y(eiw).
We can accomplish this by recognizing that since z[n] is real, X (eiw) must be conjugate symmetric.
The output of the system, y[n], has a Fourier transform Y(eiw) that is the product of X(eiw) and
H(dw). Therefore, Y(eiw) will correspond to
At first glance, it may seem like X(e/W) = Y(eiw) + Y-(,-iw). This is close to the right answer, but it
=
doesn't take into consideration the fact that Y(eiw) is non-zero at w 0 and w ,r. Thus, the solution=
X(eiw) = Y(eiW) + rc,-;w), will be incorrect at w = O and w = .,
since Y(eiw) and Y-(eiw) will
overlap at these frequencies. It is necessary to pay special attention to these frequencies to get the right
answer. Let
Z(dw) ={
0, _ w O,w = = .-
Y(e'w), otherwise
Alternatively, we can express Z(eiw) with the constants a and b defined as
00
a = Y(eiwlL=0 = L y[n]
n=-00
00
b = Y(eiw)L=• = L y[n](-1)"
n=-00
H(z) = F(l/z)
= ....!.., J HR(v- 1 ) (z-• + ") dv, lzl ~ 1
2.-J Tc z-• - v v
where HR(v) = 1u{H(ei1 )}.
484
:E
n=-oc
1l{z[n]}z[n] = 2~ 1-• •
H(eiw)X(ei"')X(e-;w) dw
where
H(eiw) = { -:i
3
O < w <,,.
_,,. <w < 0
but the integral = 0 since the integrand is an odd function over the symmetric interval.
(c) Since 1l{z[n]} = :t[n] • h[n]
1l{z[n] • y[n]} = (z[n] • y[n]) • h[n]
= (:t[n] • h[n]) • y[n]
= z[n] • (y[n] • h[n])
h[n]
z,[n] z,[n]
h[n]
(b) The cross-correlation between input and output is just the convolution of 4',.,.(m] and h(m],
00
4',.,.(m] = L h(k]4',.,.(m- k]
l=-oo
= L h(-l]¢,.,.(m- l]
l=-oo
00
= - L h(l]¢•• ,.(m - l]
l=-co
= -¢,.,,(m]
since h(n] = -h(-n] and ¢,.,.(m] = ¢,••• (-m).
{c) Starting from the definition of the autocorrelation and using the linearity of the expectation oper-
ator we get
we get
0, O<w< .-
P.,(w) = { 4t,.-.(e'"'), -.- < w < 0
11.31. (a) As shown in the figure below, the system recoutructs the original bandpass signal. As in the
example, T =.-/(rt,+ Art) and M = 5.
486
2/(ST)
V.<e"'J
(I)
21T
1/T Y,(e"'J
1 y C(jO) = s C(j!l)
-nC
h,.;(n]
(e) Using the information from part (b) we find
11(n] = 11,(n] • h;(n]
= (11,.(n] + ;11,.[n]) • (h,.;(n] + jh;,[n])
= (11,.(n] • h,.;(n] -11,.(n] • h;;[n]) +j(y,.[n] • h,.;[n] + 11,.[n] • h;;[n))
Jlr[n]
We can now redraw the figure using only real operations:
yJnJ - tM
Y,.(n]
hJnJ
y)n]
Ideal
DIC
Convener -
yJnJ
- tM
Y.,(n]
l>Jn]
-
• -1
(d) From comparing the top and bottom figures in the answer to part (a), it is evident that the desired
complex system response is given by:
H(eiw)={ 1, -1r<w<O
0, 0:, w $1f
X(z>
488
JI; J,I~
log{l -ctz- 1) = - f: nz
ctn -n
n=I
1 1•1 > lal
00 /3"
log(l - .Bz) = - L nz",
n=l
l•I > I.B- 1 1
-1
.B-n
= I: -n z-n' l•I > I.B-'I
n=-00
From the equations above we can identify the following z-transform pain;
an
--u[n -1] +-+ log(l - a,- 1 ), l•I > lal
n
.B-n
-u[-n - l] +-+ log(l -
n
.Bz), l•I > I.B- 1 1
log{A), n=O
n>O
i[n] =
n<O
(e) From the results of part (d), we see if i(n] is causal, all the b1 and d1 terms must be zero. But the
expression for X (z) shows these terms correspond to the zeros and poles outside the unit circle.
We conclude that all the zeros and poles af X(z) are inside the unit circle, i.e., :z:[n] is a minimum
phase sequence.