CHAPTER 3
Pulse Modulation
Problem 3.1
Let 2W denote the bandwidth of a narrowband signal with carrier frequency f,. The in-phase and
quadrature components of this signal are both low-pass signals with a common bandwidth of W.
According to the sampling theorem, there is no information loss if the in-phase and quadrature
‘components are sampled at a rate higher than 2W. For the problem at hand, we have
fe = 100 kHz
2W= 10 kHz
Hence, W = 5 kHz, and the minimum rate at which it is permissible to sample the in-phase and
quadrature components is 10 kHz.
From the sampling theorem, we also know that a physical waveform can be represented over the
interval -0 << 00 by
8 = YD 4,0, (0) co)
where {@,(f)} is a set of orthogonal functions defined as
sin{nf,(¢-n/f,)}
8) = FWA
where 1 is an integer and f, is the sampling frequency. If g(#) is a low-pass signal band-limited to
W Hz, and f, > 2W, then the coefficient a, can be shown to equal g(n/f,). That is, for f, > 2W, the
orthogonal coefficients are simply the values of the waveform that are obtained when the
waveform is sampled every L/f, second,
As already mentioned, the narrowband signal is two-dimensional, consisting of in-phase and
quadrature components. In light of Eq. (1), we may represent them as follows, respectively:
gilt) = DY an/F.)0,(2)
143 solt) = DY soln/F)0,(0)
Hence, given the in-phase samples af +) and quadrature samples sf?) . We may reconstruct
oi fe
the narrowband signal g() as follows:
8(1) = g(t)cos(27f,t) — go(t)sin(2nf,0)
=> [s{F) cos(2nf,t) - go
sin(2nf0)]0,(8)
where f, = 100 kHz and f, > 10 kHz, and where the same set of orthonormal basis functions is
used for reconstructing both the in-phase and quadrature components.
144 Problem 3.2
(a) Consider a periodic train o(f) of rectangular pulses, each of duration T. The Fourier series
expansion of et) (assuming that a pulse of the train is centered on the origin) is given by
ct) = YO f, sindinf, TexpGzmnf,t)
where f, is the repetition frequency, and the amplitude of a rectangular pulse is assumed to be 1/T
(ie., each pulse has unit area). The assumption that f,T>>1 means that the spectral lines (ie.,
harmonics) of the periodic pulse train e(t) are well separated from each other.
Multiplying a message signal g(t) by c(t) yields
s(t) = o(t)g(t)
= Ef, sine(nf,T) g(t) expG2enf,0) os
‘Taking the Fourier transform of both sides of Eq.. (1) and using the frequency-shifting property of
the Fourier transform:
EG | oy eee ra @
nese
where Gif) = Flg(t)]. Thus, the spectrum S(f) consists of frequency-shifted replicas of the original
spectrum G(f), with the nthreplica being scaled in amplitude by the factor f,sine(nf,T).
145 () In accordance with the sampling theorem, let it be assumed that
« The signal g(t) is band-limited with
Gf) =0 for -W2W
Then, the different frequency-shifted replicas of G(f) involved in the construction of S(f) will not
overlap. Under the conditions described herein, the original spectrum G(f), and therefore the signal
g(t), can be recovered exactly (except for a trivial amplitude scaling) by passing s(t) through a low-
pass filter of bandwidth W.
Problem 3.3,
(a) g(t) = sine(200¢)
This sinc pulse corresponds to a bandwidth W = 100 liz. Hence, the Nyquist rate is 200 liz,
and the Nyquist interval is 1/200 seconds.
(b) g(e) = sine” (200t?
This signal may be viewed as the product of the sinc pulse sinc(200t}’ with itself.
Since multiplication in the time domain corresponds to convolution in the frequency
domain, ve find that the signal g(t) has a bandwidth equal to twice that of the sinc
pulse sin(200t}; that is, 200 Hz. The Nyquist rate of g(t) is therefore 400 Ha, and the
Nyquist interval is 1/400 seconds.
(e) g(t) = sine(200e), + sinc”(200e)
The bandwidth of g(t) is determined by the highest frequency component of sinc(200t) or
sinc2(200t}, whichever one is the largest. With the bandwidth (ive., highest frequency
component of) the sinc pulse sinc(200 t) equal to 100 Hz and that of the squared sinc
pulse sinc2(200t}) equal to 200 Hz, it follows that the bandwidth of g(€) is 200 Hz.
Correspondingly, the Nyquist rate of g(t) is 400 Ilz, and its Nyquist interval is 1/400
seconds.
146 Problem 3.4
(a) The PAM wave is
s(t) = E (1 + um! (nT) Je(tent,),
where g(t) is the pulse shape, and mt(t) = m(t)/A, = cos(2mf,t). The PAM wave is
equivalent to the convolution of the instantaneously sampled [1 + um'(t)] and the pulse
shape g(t):
s(t) = { E [taum'(nt)] Stent.) } xe at)
nese
= {Ct4nm'(t)1 8 (tnt) } Ye @(t)
Tne spectrum of the PAM wave is,
SCA) = {CCC some) ye or ocr = By} acer
5
1 2 2
ett) E c6cr = By sume By]
1; m-° Ts Ts
For a rectangular pulse g(t) of duration T:0.45s, and with AT =
we have:
G(f) = AT sine(fT)
= sinc(0.45f)
For m'(t) = cos(2™fmt), and with ty = 0.25 Hz, we have:
mice) = 4 6¢#-0.25) + 8(140.25))
For T, = 18, the ideally sampled spectrum is $,(f) = [6(fom) + uMt(fom)J.
$a)
3
me
a
=1as” hd =, 35 9
The actual sampled spectrum is
147 s(t)
sine(0.U5f)[6(f-m) + pM! (f-m)]
Ss)
ors 07
0.63% i outs onttys estes outSta Ae
*s t t f 1 | ae)
2 lO 01s =O GO 0.35 07 VO Lae
(b) The ideal reconstruction filter would retain the centre 3 delta functions of S(f) or:
f z
— sas 0 bas
With no aperture effect, the two outer delta functions would have amplitude 3. Aperture
effect distorts the reconstructed signal by attenuating the high frequency portion of the
message signal.
Problem 3.5
The spectrum of the flat-top pulses is given by
H(f) = Tsinc ( fT)exp(—jnfT)
= 10™*sinc (10 f)exp(—jmf 10)
Let s(f) denote the sequence of flat-top pulses:
s(t) = SY m(nTh(t-n7,)
‘The spectrum Sf) = Fls(#)] is as follows:
Sf) =f MUP - KH)
kaw
= fH) DY MUP EF)
no
‘The magnitude spectrum |S(f)| is thus as shown in Fig. Ic.
148 4
\rol
-1/T a (a) wT f
1 (Me
fs Ww f,
(b) u
Sif)
7 0 Wt
1/T © f
VT = 10,000Hz
f, = 1,000Hz
W =400Hz
Figure 1
149 Problem 3.6
At f= 1/27, which corresponds to the highest frequency component of the message signal ae
sampling. rate equal to the Nyquist rate, we find from Eq. (6-19) that the amplitude respor
sine (O5777,)
lea cndiion
1.0) 7
re
$s 34 35 38
buy eve 17,
Figure 1
of the equalizer normalized to that at zero frequency is equal to
1 (n/2XT/T)
sine(O.ST/T,)_sin[(n/2NT/T,)]
where the ratio T/T, is equal to the duty cycle of the sampling pulses. In Fig. 1, this result is
plotted as a function of T/T,. Ideally, it should be equal to one for all values of 7/T,. For a duty
cycle of 10 percent, it is equal to 1.0041. It follows therefore that for duty cycles of less than 10
percent, the aperture effect becomes negligible, and the need for equalization may be omitted
altogether.
150 Problem 3.7
Consider the full-load test tone A cos(2tf,t). Denoting the kth sample amplitude of
this signal by A), we find that the transmitted pulse is Ay g(t), where g(t) is defined by
the spectrum:
co If < Bp
Gf) =
otherwise
The mean value of the transmitted signal power is
1 its, it 2,
Eins sy Ft ray g(ty iat)
Lre UTS our,
L
2
2 ALA, eo(t) dt)
LT,
ELLA S © e@tyat
ba -LT,
where T, is the sampling period. However,
ELAAL] =
o, otherusie
Therefore,
e(tyat
Using Rayleigh's energy theorem, we may write
L gtat =f peceyi2ar
151 Therefore,
ae
The average signal power at the receiver output is A°/2. He th
to-noise ratio is given by pass ness She output signal
By choosing B,=1/2T,, we get
(SNR)g
This shows that PAM and baseband signal transmission have the same signal-to-noise ratio
for the same average transmitted power, with ‘additive white Gaussian noise, and assuming
the use of the minimum transmission bandwidth possible.
Problem 3.8
(a) The sampling interval is T, = 125 ¥s, There are 24 channels and 1 syne pulse, so the
time alloted to each channel i8 1, = T,/25 = 5 Us. The pulse duration is 1 Ms, so the
time between pulses is 4 Us.
(b) If sampled at the nyquist rate, 6.8 kHz, then T, = 147 vs, T, = 6.68 Us, and the time
between pulses is 5.68 us.
Problem 3.9
(a) The bandwidth required for each single sideband channel is 10 kHz. The total
bandwidth for 12 channels is 120 kHz.
(>) The Nyquist rate for each signal is 20 kHz. For 12 TDM signals, the total data rate
is 2N0 kliz, By using a sinc pulse whose amplitude varies in accordance with the
modulation, and with zero crossings at multiples of (1/240) ms, we need a minimum
bandwidth of 120 KHz.
152 Problem 3.10
(a) The Nyquist rate for s,(t) and s.(t) is 160 Hz. Therefore, at must be greater than
160, and the maximun R is 3:
(b) With R= 3, we may use the following signal format to multiplex the signals s,(t) and
8,(t) into a new signal, and then multiplex s,(t) and
8,(t) and 9,(t) including markers
for synchronization:
Macker
s
eA poytid Time
SS 535 535 Se SSS SSS S,
@ zero sampas
Based on this signal format, we may develop the following multiplexing system:
2400 Hz.
Clock,
FL - is
Delay Dato
baencett [serete} ae samt fe 5,ce
(e)
M [828 [e poten]
v
x
Mutbixploud
[signa
xcz
a Sampler,
153 Problem 3.11
In general, a line code can be represented as
N
s(t) = YS a,g(t—n,)
na
Let g(t) = G(f). We may then define the Fourier transform of s(t) as
N
SN = DY a,GNe
nN
-jont,
a jon’
=a) Sage
nN
where «= 2zf. The power spectral density of s(¢) is
See |
n=
SA) = lim [How 5
Ii
T+
a ae i(m=n)oT,
=IG()P lim # Y Flajagle” '
ro
n= om=N
where T is the duration of the binary data sequence, and E denotes the statistical expectation
operator. Define the autocorrelation of the binary data sequence as
R(k) = Ela,a,.,)
Indy +
By letting m =n +k and T= (2N + 1)T,, we may write
ig?
SA) = I6(Al inde ;
No keNen ar
8,7) = GDP im I
T, N-=| 2N41
154 - IeanP oly
“T Zwe @
where
1
RU) = Eldan gl = Sanne .dP; @
in
where p; is the probability of getting the product (dy dn4,); and there are J possible values for the
4, dng product. G(/) is the spectrum of the pulse-shaping signal for representing a digital symbol.
Eqs. (1) and (2) provide the basis for evaluating the spectra of the specified line codes.
(a) Unipolar NRZ signaling
For rectangular NRZ pulse shapes, the Fourier-transform pair is
a= Aree x) FG(f) = AT,sinc(fT,)
For unipolar NRZ signaling, the possible levels for a’s are +4 and 0. For equiprobable
symbols, we have the following autocorrelation values:
20) = tare to =A/2
4
RU) = SnAg 2):Pi
ia
2 2
Aan OMOMOEAg
= Fafa de Ga 4 for dl>o
Thus
Rk) =f A fork = 0 GB)
74 for k#0 Therefore, the power spectral density for unipolar NRZ signals, using formulas (1) and (3), is
3e
i
ae som t ral
TA 4
is
But,
where 8(f) is a delta function in the frequency domain. Hence,
AT, 2
SCA) = sino
Ty)
ia]
We also note that sinc(f7,) = 0 at f = +,n#0; we thus get
AT,
sue teine(sTy)[ +5
(b) Polat Non-return-to-zero Signaling
For polar NRZ signaling, the possible values for a’s are +A and -A, Assuming equiprobable
symbols, we have
RO) = Say4,),P;
ist
For k #0, we have
156 ‘
RE) = YA ntg 0):Pi
i=l
5 CA) 5
=0
Thus,
R(k) = | A for k= 0 4)
0 for k#0
‘The power spectral density for this case, using formulas (1) and (4), is
S(f) = A’T,sine*(fT,)
(c) Return-to-zero Signaling
The pulse shape used for return-to-zero signaling is given by (x ) We therefore have
T,/2
GY) spPsine(f1,/2)
The autocorrelation for this case is the same as that for unipolar NRZ signaling. Therefore, the
power spectral density of RZ signals is
2
aT,
Sf) = sine’ gancTall+
(d) Bipolar Signals
The permitted values of level a for bipolar signals are +A, -A, and 0, where binary symbol 1 is
represented alternately by +4 and -A, and binary 0 is represented by level zero. We thus have
the following autocorrelation function values: Fork>1,
RW) = Day.
Thus,
as for k =
ae AS for (a | °
0 for |k| > 1
‘The pulse duration for this case is equal to T,/2. Hence,
Gin = Bai ine 22) ©
Using Equations (1), (5) and (6), the power spectral density of bipolar signals is
jot, -joT,,
be |
sine ZA 1 -cos(2n/T,)]
158 (ec) Manchester Code,
The permitted values of a’s in the Manchester code are +A and -A. Hence,
latte ata lta leg?
R(O) = GAP HAY + GCAY? + 34?)
=a
For k#0,
4
RE) = Daya, dis =
=0
Thus,
2
mua 4 for k = 0
0
for k#0
U
The pulse shape of Manchester signaling is given by
tT y/4 (t-T,/4y
Hn = 0S) (FH)
The pulse spectrum is therefore
Gf)
fT) ~ioty/4
sine)
3
srind p22)
159 Therefore, the power spectral density of Manchester NRZ has the form
Problem 3.12
Power spectral density of a binary data stream will not be affected by the use of differential
encoding. The reason for this statement is that differential encoding uses the same pulse shaping
functions as ordinary encoding methods. If the number of bits is high, then the probability of a
symbol one and symbol zero are the same for both cases,
Problem 3.13
(a)
| (mt) Ty oT
(b) an = ):
0, otherwise
Equivalently, we may write
a(t) = cos( FF Aree)
where rect(®) is @ rectangular function of unit amplitude and unit duration. The Fourier
transform of g(t) is given by
Gf) = FS r-A rez] sinc(fT,)
where A denotes the pulse amplitude and * denotes convolution in the frequency domain.
160 Using the replication property of the delta function 8(/), we get
+ sine (rls
+ asine(n(/-F)) sine (7 or
J
7)
a
2
Note that the two spectral components sine (Ti f 2) and sine (rf fe 7) overlap in
0
the frequency interval -(1/T,) < f < (I/T;), hence the presence of cross-product terms in Eq.
().
Figure 1 plots the normalized power spectral density S()/(A77)/4) versus the normalized
frequency fT), The interesting point to note in this figure is the significant reduction in the
power spectrum of the puls
shaped data stream x(t) in the interval -L/T, << Ty
(©) The power spectral density of the standard form of polar NRZ signaling is
S(f) = AT ysine*(fT,)
Comparing this expression with that of Eq. (1), we observe the following differences:
SS
Polar NRZ ing | Polar NRZ signals using
cosine pulses rectangular pulses
feo 0 “7,
f= 22/T, ATy4 0
161
@) rk
162
14
12h
= = ©
wt ws”
oat Problem 3.14
(4
Ch
a
TELE = FR
Ppl ps
© HILAL
163 roblem 3.15(a)
Pr Problem 3.15(b)
cay"
dn tt boo tot
165 Problem 3.16
‘The minimum number of bits per sample is 7 for a signal-to-quantization noise ratio of 40 dB.
Hence,
(™* number of sampl
in a duration of 10s
) = 8000 10
8x10" samples
The minimum storage is therefore
=7x8x 10*
= 5.6.x 105
= 560 kbits
166 Problem 3.17
Suppose that baseband signal m(t) is modeled as the sample function of a Gaussian random
process of zero mean, and that the amplitude range of m(t) at the quantizer input extends from
4Armg t0 4Arm,: We find that samples of the signal m(t) will fall outside the amplitude range 8Aymg
with a probability of overload that is less than 1 in 10* If we further assume the use of a binary
code with each code word having a length n, so that the number of quantizing levels is 2", we find
that the resulting quantizer step size is
3 = SAnms @
aR
Substituting Eq. (1) to the formula for the output signal-to-quantization noise ratio, we get
(SNR), = son @
Expressing the signal-to-noise ratio in decibels:
10logg(SNR) = GR - 7.2 (3)
This formula states that each bit in the code word of a PCM system contributes 6dB to the signal-
to-noise ratio. It gives a good description of the noise performance of a PCM system, provided that
the following conditions are satisfied:
1, The system operates with an average signal power above the error threshold, so that the
effect of transmission noise is made negligible, and performance is thereby limited
essentially by quantizing noise alone.
2. ‘The quantizing error is uniformly distributed.
3. The quantization is fine enough (say R > 6) to prevent signal-correlated patterns in the
quantizing error waveform.
4. The quantizer is aligned with the amplitude range from -4Armg 0 4Armg-
In general, conditions (1) through (3) are true of toll quality voice signals. However, when demands
on voice quality are not severe, we may use a coarse quantizer corresponding to R < 6. In such a
case, degradation in system performance is reflected not only by a lower signal-to-noise ratio, but
also by an undesirable presence of signal-dependent patterns in the waveform of quantizing error.
167 Problem 3.18
(a) Let the message bandwidth be W. Then, sampling the message signal at its Nyquist
rate, and using an R-bit code to represent each sample of the message signal, we find that
the bit duration is
le
* aR
The bit rate is
1
wt BR
%
‘The maximum value of message bandwidth is therefore
10°
7
7)
Ynax = “2
x
x
= 3.57 x 10° te
(b) The output signal-to-quantizing noise ratio is given by (see Example 2):
10 10819 (SNR) = 1.8 + OR
1.8467
43,8 dB
Problem 3.19
Let a signal amplitude lying in the range
1 1
HO sr iy tg hy
be represented by the quantized amplitude x,. The instantaneous square value of the error
is (exy)*, Let the probability density function of the input signal be f(x). If the
step size 6; is small in relation to the input signal excursion, then fy(x) varies little
within the quantum step and may be approximated by fy(x.
)+ Then, the mean-square value of
the error due to signals falling within this quantum is
? tyra
168 a
(2)
Therefore, eliminating fy(x,) between Eqs. (1) and (2), we get
2,22
EIQf] = +
The total mean-square value of the quantizing error is the sum of that contributed by each
of the several quanta. Hence,
169 Problem 3.20
(a)
=
=
Voltage
Quaticn
output
Quadkzer
eutpok
Voltane
Tyla.
a Tone
es ey
it 83
8% *
. Quoted
one anpor
(>)
10 Problem 3
The quantizer has the following input-output curve:
Ouspur
At the sampling instants we havet
t m(t) code
3/8 2 0011
-1/8 v2 0011
1/8 3v2 1100
43/8 3v2 1100
And the coded waveform is (assuming on-off signaling):
a pouty Time (Seconds)
+ 3
8 7
roblem 3.2:
The transmitted code words are:
t/t, code
1 001
010
3 on
4 100
5 101
6 110
mt The sampled analog signal is
Problem
(a) The probability p, of any binary symbol being inverted by transmission through the
system is usually quite small, so that the probability of error after n regenerations in
the system is very nearly equal ton p,. For very large n, the probability of more than
one inversion must be taken into account. Let p, denote the probability that a binary
symbol is in error after transmission through the Complete system. ‘Then, p, is also the
probability of an odd mumber of errors, since an even number of errors"restores the
original value. Counting zero as an even number, the probability of an even number of
errors is 1-p,. Hence
Pp (1-P4)+ (1-1
Poet Py
C2 PLHP,
This is a linear difference equation of the first order. Its solution is
1 a
Py =e (1-1-2,
(bv) If p, is very small and n is not too large, then
(1-2) 9" = 1-2) yn
and
1
Py = gli-(-2p,n)]
= pyr
172 Problem 3.24 - Regenerative repeater for PCM
Three basic functions are performed by regenerative repeaters: equalization, timing and decision-
making.
Equalization: The equalizer shapes the incoming pulses so as to compensate for the effects of
amplitude and phase distortion produced by the imperfect transmission characteristics of the
channel.
‘Timing: The timing circuitry provides a periodic pulse train, derived from the received pulses, for
sampling the equalized pulses at the instants of time where the signal-to-noise ratio is maximum.
Decision-making: The extracted samples are compared to a predetermined threshold to make
decisions. In each bit interval, a decision is made whether the received symbol is 1 or 0 on the
basis of whether the threshold is exceeded or not.
Problem 3.25
m(1) = Atanh(Br)
To avoid slope overload, we require
|dm(1)|
ml | w
am) = 4 Bsech?(Br) Q)
dt
Hence, using Eq. (2) in (1):
A> max(ABsech*(Br)) xT, @)
Since sech(Br) = —1.
cosh (Bi)
Bi
it follows that the maximum value of sech(fr) is 1, which occurs at time 1 = 0, Hence, from Eq. (3)
we find that A > ABT,
173 Problem 3.26
The modulating wave is
mt) = A, cos(2nf,t)
The slope of m(t) is
an(t)
at
W2nf hy sin(2nft)
The maximum slope of m(t) is equal to 2nf,A,
The maximum average slope of the approximating signal m,(t) produced by the delta
modulator is 6/T,, where 6 is the step size and T, is the sampling period. The limiting
value of 4, is therefore given by
é
anf, > S
mn ? TS
or
6
A> sq
in? Bef, Ty
Assuming a load of 1 ohm, the transmitted power is AC/2, Therefore, the maximum
power that may be transmitted without slopeoverload distortion is equal to 6°/8x"r272,
174 Problem 3.27
Is= Wxyquist
Frxyquist = 6-8 KHz
f= 10x 6.8 x 10° = 6.8 x 10* Hz.
A
2.>max|
idm(1))
dt
For the sinusoidal signal m(t) = A,,Sin(2rtfyf), we have
POs raf pA yC08(2%F yf)
dt
Hence,
|dm(t)| 2: ri
em = Pha
or, equivalently,
4
T,
Ti Arb max
Therefore,
Andean = oto
mlmax ~ TX 20x Fy
~ AL:
2th m
01x68 x 10°
2nx 10°
u
1osv
175 Problem 3.28
(a) From the solution to Problem 3.27, we have
a)
‘The average signal power = £
=e)
With slope overload avoided, the only source of quantization of noise is granular noise
Replacing A/2 for PCM with A for delta modulation, we find that the average quantization
noise power is A7/3; for more details, see the solution to part (b) of Problem 3.30. The
waveform of the reconstruction error (i.e., granular quantization noise) is a pattern of bipolar
binary pulses characterized by (1) duration = T, = I/f,, and (2) average power = A/3. Hence,
the autocorrelation function of the quantization noise is triangular in shape with a peak value
of A?/3 and base 27,, as shown in Fig. 1:
Rol) >
v3
Fig. 1
From random process theory, we recall that
SoAjao = [Rola
which, for the problem at hand, yields
‘Typically, in delta modulation the sampling rate f, is very large compared to the highest frequency
component of the original message signal. We may therefore approximate the power spectral
density of the granular quantization noise as
176 2
Soif=) A3F, “WS SSW
0, otherwise
where W is the bandwidth of the reconstruction filter at the demodulator output. Hence, the
average quantization noise power is
20°W
3f5
w
N= J So(fdf = Q)
-w
Substituting Eq. (2) into (1), we get
of 2 LmAY? W
wa (Say
_ 8x fw
af
(b) Correspondingly, output signal-to-noise ratio is
(3)
SNR = —— >,
(81° f,A°W)/3f?
ah
lor Ww
Problem 3.29
@as dhs
2 hn
17 2xmx 10x
af
0) (SNR) = SS
ax’ fw
3
fe e3e (50x13)
16m? 10°x5x 10°
= 475
In decibels,
(SNR) oq, = LOlog 49475
= 268 4B
Problem 3.30
(@) For linear delta modulation, the maximum amplitude of a sinusoidal test signal that can be
used without slope-overload distortion is
Af,
A= aap,
0.1.x 60x 10° 3
—veo ff, = 2x3x10°
2nx 1x10
= 0.95V
(b) i)
Under the pre-filtered condition, it is reasonable to assume that the granular quantization
noise is uniformly distributed between -A and +A. Hence, the variance of the quantization
noise is
178 (SNR),
prefiltered =
_ 3x 0.95?
Sieean ik
2x0.
= 135
= 21.3 dB
(ii)The signal-to-noise ratio under the post-filtered condition is
3
(3) =F
(Weestneea ” Ten? fa
(60
(1y’x3
= 1367
= 313 dB
‘The filtering gain in signal-to-noise ratio due to the use of a reconstruction filter at the
demodulator output is therefore 31.3 - 21,3 = 10 dB.
179 Problem 3.31
Let the sinusoidal signal m(t) = Asinaog, where @p = 2xfo
The autocorrelation of the signal is
2
R(t) =
mn c0s(Wot)
A
2a
_ ( 1
R,(L) = 5 °°5( %0%* ia)
2
a
= F.e0s(0.1)
For this problem, we thus have
Ry, = (Ry, (O))
(R,(1)]
(a) The optimum solution is given by
Wo = Rie,
2
Fos (0.1)
= 2 = 00s(0.1)
ae
2
= 0.995
(©) Jig = Ry(0) = TRG Ey
2 2 2
5 4-4 .08(0.1) x 4-cos(0.1)/(A?/2)
180 (1 cos?(0.1))
np
= 0.00547
Problem 3.32
1 08 06]
= |08 1 08)
0.608 1
= [os, 06, 04)”
(a) Wo = RY
-1
1 08 06] [os
= |08 1 0.8} 0.6)
0.608 1} [o4|
0.875
=l 0
-0.125
(©) Jin = R,(0)— 4, Ri 'r,
r,
= R,(O0)—1 Wo
0.875
08,06, 0.4]| 0
—0.125]
(0.8 x0.875- 04x 0.125)
0.7 + 0.05
0.35
181 Problem 3.33
RK =|! 08
08 1
(@) wo = Rr
= | 0.8889
-O.1LIL
() Imin = R,(O)— ERY
10.6444
= 0.3556
which is slightly worse than the result obtained with a linear predictor using three unit delays
(ic., three coefficients). This result is intuitively satisfying.
Problem 3.34
Input signal variance = R,(0)
The normalized autocorrelation of the input signal for a lag of one sample interval is,
= 0.75
Error variance = R,(0)-R,(1)Rz!(0)R,(1)
= R,(0)(1-p2(1))
182 RO)
Processing gain. = ———*—__
RO) ~P-(1))
2.2857
Expressing the processing gain in dB, we have
3.59 dB
10log 19(2.2857)
Problem 3.35
Processing gain
(a) Three-tap predictor:
Processing gain = 2.8571
4.56 dB
(b) Two-tap predictor:
Processing gain = 2.8715
4.49 dB
Therefore, the use of a three-tap predictor in the DPCM system results an improvement of
4.56 - 4.49 = 0.07 dB over the corresponding system using a two-tap predictor.
Problem 3.
(a) For DPCM, we have 10log9(SNR)o = 01 + 6n dB
For PCM, we have 10logyg(SNR)g = 4.77 + 6n - 20log;o(log(1 + 11))
where n is the number of quantization levels
SNR of DPCM SNR = ce + 6n, where -3<0< 15
For n=8, the SNR is in the range of 45 to 63 dBs.
SNR of Pt
SNR = 4.77 + 6n - 20logyo(log(2.56))
4.77 +48 - 14.8783
=38dB
Therefore, the SNR improvement resulting from the use of DPCM is in the range of 7 to 25,
4B.
(b) Let us assume that m) bits/sample are used for DPCM and » bits/sample for PCM
If = 15 dB, then we have
15 +6m, = 6n- 10.0
10+15,
6
Rearranging: (n ~ nj)
= 4.18
which, in effect, represents a saving of about 4 bits/sample due to the use of DPCM.
If, on the other hand, we choose
-3 dB, we have
-3 +6n, = 6n - 10
Rearranging: (n-n,) = 12-3
= 101
which represents a saving of about 1 bit/sample due to the use of DPCM.
184, Problem 3.37
The transmitting prediction filter operates on exact samples of the signal, whereas the receiving
prediction filter operates on quantized samples.
Problem 3.38
Matlab codes
4% Problem 3.38, CS: Haykin
Ylat-topped PAM signal
Yand magnitude spectrum
o Mathini Sellathurai
Yaata
£3=8000; % sample frequency
ts=1.260-4; Yit/ts
pulse_duration=5e-S; Ypulse duration
% sinusoidal sgnai;
260-6; Ysampling frequency of signal
£4=80000;
t= (0:td: 100¥td);
m=10000;
sesin(fmet);
% PAM signal generation
pan_s=PAM(s,td, ts, pulse
figure(1);hold on
185 plot(t,s,’—-");
plot(t(1:length(pam.s)),pam_s);
xlabel('tine’)
ylabel (‘magnitude’)
Legend ‘signal’, *PAN-signal’);
% Computing magnitude spectrum S(#) of the signal
a=((abs(t2t(pam.s)).-2));
a=a/nax(a);
f=t50(ts/24:ts%(t5/ta) : CLength(a))+£a¢ (£2/24);
figure (2)
plot(t,a);
xlabel(’frequency’);
ylabel (‘magnitude’)
4% finding the zeros
index=tind(a
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