Linear Modulation Techniques

Chapter 3
Linear Modulation Techniques
Contents
3.1 Linear Modulation . . . . . . . . . . . . . . . . . . 3-3
3.1.1 Double-Sideband Modulation (DSB) . . . . 3-3
3.1.2 Amplitude Modulation . . . . . . . . . . . . 3-8
3.1.3 Single-Sideband Modulation . . . . . . . . . 3-21
3.1.4 Vestigial-Sideband Modulation . . . . . . . . 3-35
3.1.5 Frequency Translation and Mixing . . . . . . 3-38
3.2 Interference . . . . . . . . . . . . . . . . . . . . . . 3-46
3.2.1 Interference in Linear Modulation . . . . . . 3-46
3.3 Sampling Theory . . . . . . . . . . . . . . . . . . . 3-49
3.4 Analog Pulse Modulation . . . . . . . . . . . . . . . 3-54
3.4.1 Pulse-Amplitude Modulation (PAM) . . . . . 3-54
3.4.2 Pulse-Width Modulation (PWM) . . . . . . . 3-56
3.4.3 Pulse-Position Modulation . . . . . . . . . . 3-56
3.5 Delta Modulation and PCM . . . . . . . . . . . . . . 3-57
3.5.1 Delta Modulation (DM) . . . . . . . . . . . 3-57
3.5.2 Pulse-Code Modulation (PCM) . . . . . . . 3-60
3.6 Multiplexing . . . . . . . . . . . . . . . . . . . . . 3-63
3-1
CONTENTS
3.6.1 Time-Division Multiplexing (TDM) . . . . . 3-64
3-2 ECE 5625 Communication Systems I

3.1. LINEAR MODULATION
We are typically interested in locating a message signal to some

new frequency location, where it can be efficiently transmitted
The carrier of the message signal is usually sinusoidal
A modulated carrier can be represented as

xc .t / D A.t / cos 2fc t C .t /
where A.t / is linear modulation, fc the carrier frequency, and

.t / is phase modulation
3.1 Linear Modulation

For linear modulation schemes, we may set .t / D 0 without
loss of generality
xc .t / D A.t / cos.2fc t /
with A.t / placed in one-to-one correspondence with the mes-

sage signal
3.1.1 Double-Sideband Modulation (DSB)

Let A.t / / m.t /, the message signal, thus
xc .t / D Ac m.t / cos.2fc t /
From the modulation theorem it follows that

1 1
Xc .f / D Ac M.f fc / C Ac M.f C fc /
2 2
ECE 5625 Communication Systems I 3-3
CONTENTS
xc(t) carrier filled envelope

m(t)
t t
DSB time domain waveforms

M(f) M(0)
f
Xc(f) 1
A M(0)
2 c
LSB USB
f
-fc fc
DSB spectra
Coherent Demodulation
The received signal is multiplied by the signal 2 cos.2fc t /,
which is synchronous with the transmitter carrier
m(t) xc(t) xr(t) d(t) yD(t)

LPF
Accos[2πfct] 2cos[2πfct]
Modulator Channel Demodulator

For an ideal channel xr .t / D xc .t /, so

d.t / D Ac m.t / cos.2fc t / 2 cos.2fc t /
D Ac m.t / C Ac m.t / cos.2.2fc /t /
where we have used the trig identity 2 cos2 x D 1 C cos 2x
The waveform and spectra of d.t / is shown below (assuming

m.t / has a triangular spectrum in D.f /)
Lowpass filtering will remove the
d(t) double frequency carrier term
Lowpass
D(f)
modulation
recovery filter
1 AcM(0) 1
A M(0) A M(0)
2 c 2 c
f
-2fc -W W 2fc
Waveform and spectrum of d.t/
Typically the carrier frequency is much greater than the mes-

sage bandwidth W , so m.t / can be recovered via lowpass fil-
tering
The scale factor Ac can be dealt with in downstream signal

processing, e.g., an automatic gain control (AGC) amplifier

CONTENTS
Assuming an ideal lowpass filter, the only requirement is that

the cutoff frequency be greater than W and less than 2fc W
The difficulty with this demodulator is the need for a coherent

carrier reference
To see how critical this is to demodulation of m.t / suppose that

the reference signal is of the form
c.t / D 2 cosŒ2fc t C .t /
where .t / is a time-varying phase error
With the imperfect carrier reference signal
d.t / D Ac m.t / cos .t / C Ac m.t / cosŒ2.2fc /t C .t /

yD .t / D m.t / cos .t /
Suppose that .t / is a constant or slowly varying, then the

cos .t / appears as a fixed or time varying attenuation factor
Even a slowly varying attenuation can be very detrimental from

a distortion standpoint
– If say .t / D 2f t and m.t / D cos.2fmt /, then

1
yD .t / D ŒcosŒ2.fm f /t C cosŒ2.fm C f /t
2
which is the sum of two tones
Being able to generate a coherent local reference is also a prac-

tical manner

One scheme is to simply square the received DSB signal
xr2.t / D A2cm2.t / cos2.2fc t /

1 2 2 1 2 2
D Ac m .t / C Ac m .t / cosŒ2.2fc /t
2 2
xr(t) LPF yD(t)
2
xr(t)
divide
( )2 BPF
by 2
Acos2πfct
very narrow
(tracking) band-
pass filter
Carrier recovery concept using signal squaring
Assuming that m2.t / has a nonzero DC value, then the double

frequency term will have a spectral line at 2fc which can be
divided by two following filtering by a narrowband bandpass
filter, i.e., Ffm2.t /g D kı.f / C
Filter this component

Spectrum
k
of m2(t)
for coherent demod
f
2fc
Note that unless m.t / has a DC component, Xc .f / will not

contain a carrier term (read ı.f ˙ fc ), thus DSB is also called
a suppressed carrier scheme

CONTENTS
Consider transmitting a small amount of unmodulated carrier
m(t) xc(t)
k k << 1
Accos2πfct use a narrowband filter

(phase-locked loop) to extract
AcM(0)/2
the carrier in the demod.
f
-fc fc
3.1.2 Amplitude Modulation

Amplitude modulation (AM) can be created by simply adding
a DC bias to the message signal
xc .t / D A C m.t / A0c cos.2fc t /

D Ac 1 C amn.t / cos.2fc t /
where Ac D AA0c , mn.t / is the normalized message such that

min mn.t / D 1,
m.t /
mn.t / D
j min m.t /j
and a is the modulation index
j min m.t /j
aD
A
A + max m(t) A + min m(t)

xc(t)
a<1
Ac(1 - a)
0 t
Note that the enve-

lope does not cross
zero in the case of
AM having a < 1
A + m(t)
m(t) xc(t)
Bias term A Accos[2πfct]
Generation of AM and a sample wavefrom
Note that if m.t / is symmetrical about zero and we define d1 as

the peak-to-peak value of xc .t / and d2 as the valley-to-valley
value of xc .t /, it follows that
d1 d2
aD
d1 C d2
proof: max m.t / D min m.t / D j min m.t /j, so
d1 d2 2Œ.A C j min m.t /j/ .A j min m.t /j/

D
d1 C d2 2Œ.A C j min m.t /j/ C .A j min m.t /j/
j min m.t /j
D Da
A

CONTENTS
The message signal can be recovered from xc .t / using a tech-

nique known as envelope detection
A diode, resistor, and capacitor is all that is needed to construct

and envelope detector
Recovered envelope
with proper RC
selection
eo(t)
xr(t) C R eo(t)
0 t
The carrier is removed if 1/fc << RC << 1/W
Envelope detector
The circuit shown above is actually a combination of a nonlin-

earity and filter (system with memory)
A detailed analysis of this circuit is more difficult than you

might think
A SPICE circuit simulation is relatively straight forward, but it

can be time consuming if W fc

The simple envelope detector fails if Ac Œ1 C amn.t / < 0
– In the circuit shown above, the diode is not ideal and

hence there is a turn-on voltage which further limits the
maximum value of a
The RC time constant cutoff frequency must lie between both

W and fc , hence good operation also requires that fc W

CONTENTS
Digital signal processing based envelope detectors are also pos-

sible
Historically the envelope detector has provided a very low-cost

means to recover the message signal on AM carrier
The spectrum of an AM signal is

Ac
Xc .f / D ı.f fc / C ı.f C fc /
„2 ƒ‚ …
pure carrier spectrum
aAc
C Mn.f fc / C Mn.f C fc /
„ 2 ƒ‚ …
DSB spectrum
AM Power Efficiency
Low-cost and easy to implement demodulators is a plus for
AM, but what is the downside?
Adding the bias term to m.t / means that a fraction of the total
transmitted power is dedicated to a pure carrier
The total power in xc .t / is can be written in terms of the time

average operator introduced in Chapter 2
hxc2.t /i D hA2cŒ1 C amn.t /2 cos2.2fc t /i

A2c
D hŒ1 C 2amn.t / C a2m2n.t /Œ1 C cos.2.2fc /t i
2
If m.t / is slowly varying with respect to cos.2fc t /, i.e.,
hm.t / cos !c t i ' 0;

then
A2c
hxc2.t /i 2 2

D 1 C 2ahmn.t /i C a hmn.t /i
2
A2c 2 2
A2c a2A2c 2
D 1 C a hm .t /i D C hmn.t /i
2 2
„ƒ‚… „ ƒ‚ … 2
Pcarrier Psidebands
where the last line resulted from the assumption hm.t /i D 0

(the DC or average value of m.t / is zero)
Definition: AM Efficiency
a2hm2n.t /i also hm2.t /i

Eff D 2 2
D 2
1 C a hmn.t /i A C hm2.t /i
Example 3.1: Single Sinusoid AM
An AM signal of the form
xc .t / D Ac Œ1 C a cos.2fmt C =3/ cos.2fc t /
contains a total power of 1000 W
The modulation index is 0.8
Find the power contained in the carrier and the sidebands, also
find the efficiency
The total power is
A2c a2A2c
1000 D hxc2.t /i D C hm2n.t /i
2 2
CONTENTS
It should be clear that in this problem mn.t / D cos.2fmt /, so

hm2n.t /i D 1=2 and

1 1 33
1000 D A2c C 0:64 D A2c
2 4 50
Thus we see that

50
A2c D 1000 D 1515:15
33
and
1 1515
Pcarrier D A2c D D 757:6 W
2 2
and thus
Psidebands D 1000 Pc D 242:4 W
The efficiency is
242:4
Eff D D 0:242 or 24.2%
1000
The magnitude and phase spectra can be plotted by first ex-

panding out xc .t /
xc .t / D Ac cos.2fc t / C aAc cos.2fmt C =3/ cos.2fc t /

D Ac cos.2fc t /
aAc
C cosŒ2.fc C fm/t C =3
2
aAc
C cosŒ2.fc fm/t =3
2

|Xc(f)|
Ac/2
0.8Ac/4
f
-fc 0 fc-fm fc fc+fm
Xc(f)
π/3
f
0
-π/3
Amplitude and phase spectra for one tone AM
Example 3.2: Pulse Train with DC Offset

m(t) 2
-1
Tm/3 Tm
Find mn.t / and the efficiency E
From the definition of mn.t /

m.t / m.t /
mn.t / D D D m.t /
j min m.t /j j 1j
The efficiency is
a2hm2n.t /i
ED
1 C a2hm2n.t /i
CONTENTS
To obtain hm2n.t /i we form the time average

"Z #
Tm =3 Z Tm
1
hm2n.t /i D .2/2 dt C . 1/2 dt
Tm 0 Tm =3

1 Tm 2Tm 4 2 6
D 4C 1 D C D D2
Tm 3 3 3 3 3
thus
2a2
ED
1 C 2a2
The best AM efficiency we can achieve with this waveform is
when a D 1
ˇ 2
Eff ˇ D D 0:67 or 67%
ˇ
aD1 3
Suppose that the message signal is m.t / as given here
Now min m.t / D 2 and mn.t / D m.t /=2 and

1 2 1
hm2n.t /i D . 1/2 C .1=2/2 D
3 3 2
The efficiency in this case is
.1=2/a2 a2
Eff D D
1 C .1=2/a2 2 C a2
Now when a D 1 we have Eff D 1=3 or just 33.3%
Note that for 50% duty cycle squarewave the efficiency maxi-
mum is just 50%

Example 3.3: Multiple Sinusoids
Suppose that m.t / is a sum of multiple sinusoids (multi-tone

AM)
XM
m.t / D Ak cos.2fk t C k /
kD1
where M is the number of sinusoids, fk values might be con-
strained over some band of frequencies W , e.g., fk W , and
the phase values k can be any value on Œ0; 2
To find mn.t / we need to find min m.t /

PM
A lower bound on min m.t / is kD1 Ak ; why?
The worst case value may not occur in practice depending upon
the phase and frequency values, so we may have to resort to a
numerical search or a plot of the waveform
Suppose that M D 3 with fk D f65; 100; 35g Hz, Ak D

f2; 3:5; 4:2g, and k D f0; =3; =4g rad.
>> [m,t] = M_sinusoids(1000,[65 100 35],[2 3.5 4.2],...

[0 pi/3 -pi/4], 20000);>> plot(t,m)
>> min(m)
ans = -7.2462e+00
>> -sum([2 3.5 4.2]) % worst case minimum value
ans = -9.7000e+00
>> subplot(311)
>> plot(t,(1 + 0.25*m/abs(min(m))).*cos(2*pi*1000*t))
>> hold

CONTENTS
Current plot held

>> plot(t,1 + 0.25*m/abs(min(m)),’r’)
>> subplot(312)
>> hold
Current plot held
>> plot(t,1 + 0.5*m/abs(min(m)),’r’)
>> subplot(313)
>> hold
Current plot held
>> plot(t,1 + 1.0*m/abs(min(m)),’r’)
2
m(t) Amplitude
−2
−4
−6
min m(t)
−8
0 0.005 0.01 0.015 0.02 0.025 0.03 0.035 0.04 0.045 0.05
Time (s)
Finding min m.t/ graphically
The normalization factor is approximately given by 7.246, that

is
m.t /
mn.t / D
7:246
Shown below are plots of xc .t / for a D 0:25; 0:5 and 1 using
fc D 1000 Hz

xc(t), a = 0.25
0
−2
0 0.005 0.01 0.015 0.02 0.025 0.03 0.035 0.04 0.045 0.05
2
xc(t), a = 0.5
−2
0 0.005 0.01 0.015 0.02 0.025 0.03 0.035 0.04 0.045 0.05
2
xc(t), a = 1.0
−2
0 0.005 0.01 0.015 0.02 0.025 0.03 0.035 0.04 0.045 0.05
Time (s)
Modulation index comparison (fc D 1000 Hz)
To obtain the efficiency of multi-tone AM we first calculate

hm2n.t /i assuming unique frequencies
M
X A2k
hm2n.t /i D
2j min m.t /j2
kD1
2 C 3:52 C 4:22
2
D D 0:3227
2 7:2462
The maximum efficiency is just
ˇ 0:3227
Eff ˇ D D 0:244 or 24.4%
ˇ
aD1 1 C 0:3227
CONTENTS
A remaining interest is the spectrum of xc .t /
Ac
Xc .f / D ı.f fc / C ı.f C fc /
2
M
aAc X h jk
C Ak e ı.f .fc C fk //
4
kD1
i
jk
Ce ı.f C .fc C fk // (USB terms)
M
aAc X h jk
C Ak e ı.f .fc fk //
4
kD1
i
jk
Ce ı.f C .fc fk // (LSB terms)
0.5 Carrier with

0.45 Ac = 1
0.4
Amplitude Spectra (|Xc(f)|)
0.35
0.3
0.25
Symmetrical
0.2
Sidebands for
0.15 a = 0.5
0.1
0.05
0
1000 800 600 400 200 0 200 400 600 800 1000
Frequency (Hz)
Amplitude spectra

3.1.3 Single-Sideband Modulation

In the study of DSB it was observed that the USB and LSB
spectra are related, that is the magnitude spectra about fc has
even symmetry and phase spectra about fc has odd symmetry
The information is redundant, meaning that m.t / can be recon-

structed from one or the other sidebands
Transmitting just the USB or LSB results in single-sideband

(SSB)
For m.t / having lowpass bandwidth of W the bandwidth re-

quired for DSB, centered on fc is 2W
Since SSB operates by transmitting just one sideband, the trans-

mission bandwidth is reduced to just W
M(f) XDSB(f)
f f
W fc - W fc fc+W
XSSB(f) XSSB(f)
LSB USB
USB LSB
removed removed
f
fc - W fc fc fc+W
DSB to two forms of SSB: USSB and LSSB
The filtering required to obtain an SSB signal is best explained

with the aid of the Hilbert transform, so we divert from text

CONTENTS
Chapter 3 back to Chapter 2 to briefly study the basic proper-

ties of this transform
Hilbert Transform
The Hilbert transform is nothing more than a filter that shifts
the phase of all frequency components by =2, i.e.,
H.f / D j sgn.f /
where 8
<1; f >0
ˆ
ˆ
sgn.f / D 0; f D0
ˆ
: 1; f < 0
ˆ
The Hilbert transform of signal x.t / can be written in terms of

the Fourier transform and inverse Fourier transform
1

O /DF
x.t j sgn.f /X.f /
D h.t / x.t /
where h.t / D F 1fH.f /g
We can find the impulse response h.t / using the duality theo-
rem and the differentiation theorem
d F
H.f / ! . j 2 t /h. t /
df
where here H.f / D j sgn.f /, so
d
H.f / D 2j ı.f /
df
Clearly,
F 1f 2j ı.f /g D 2j
so
2j 1
h.t / D D
j 2 t t
and
1 F
! j sgn.f /
t
In the time domain the Hilbert transform is the convolution
integral
Z 1 Z 1
x./ x.t /
O /D
x.t d D d
1 .t / 1
Note that since the Hilbert transform of x.t / is a =2 phase

O / is
shift, the Hilbert transform of x.t
OO / D
x.t x.t /
why? observe that . j sgn.f //2 D 1
Example 3.4: x.t / D cos !0t
By definition
1
XO .f / D

j sgn.f / ı.f f0/ C ı.f C f0/
2
1 1
D j ı.f f0/ C j ı.f C f0/
2 2

CONTENTS
F
so from e j!0t ) ı.f f0 /
1 1
O /D
x.t j e j!0t C j e j!0t
2 2
j!0 t j!0 t
e e
D D sin !0t
2j
or
2
cos !0t D sin !0t
It also follows that
2 2
sin !0t D cos !0t D cos !0t
OO / D
since x.t x.t /
Hilbert Transform Properties
O / are equal
1. The energy (power) in x.t / and x.t
The proof follows from the fact that jY .f /j2 D jH.f /j2jX.f /j2
and jj sgn.f /j2 D 1
O / are orthogonal, that is

2. x.t / and x.t
Z 1
O / dt D 0 (energy signal)
x.t /x.t
Z1 T
1
lim O / dt D 0 (power signal)
x.t /x.t
T !1 2T T

The proof follows for the case of energy signals by generaliz-

ing Parseval’s theorem
Z 1 Z 1
O / dt D
x.t /x.t X.f /XO .f / df
1 Z 11
D .j sgn.f // jX.f /j2 df D 0

1 „ ƒ‚ … „ ƒ‚ …
odd even
3. Given signals m.t / and c.t / such that the corresponding spec-
tra are
M.f / D 0 for jf j > W (a lowpass signal)
C.f / D 0 for jf j < W (c.t / a highpass signal)
3
then
m.t /c.t / D m.t /c.t
O /
Example 3.5: c.t / D cos !0t

Suppose that M.f / D 0 for jf j > W and f0 > W then
5 2
m.t / cos !0t D m.t /cos !0t
D m.t / sin !0t
Analytic Signals
Define analytic signal z.t / as
z.t / D x.t / C j x.t
O /
where x.t / is a real signal

CONTENTS
The envelope of z.t / is jz.t /j and is related to the envelope

discussed with DSB and AM signals
The spectrum of an analytic signal has single-sideband charac-

teristics
In particular for zp .t / D x.t / C j x.t

O /
˚
Zp .f / D X.f / C j j sgn.f /X.f /

D X.f / 1 C sgn.f /
(
2X.f /; f > 0
D
0; f <0
Note: Only positive frequencies present
Similarly for zn.t / D x.t / O /

j x.t

Zn.f / D X.f / 1 sgn.f /
(
0; f >0
D
2X.f /; f < 0
X(f) 1
f
-W W
Zp(f) 2
f
-W W
Zn(f)
2
f
-W W
The spectra of analytic signals can suppress positive or negative

frequencies
Return to SSB Development

xDSB(t)
Sideband xSSB(t)
m(t) Filter
LSB or USB
Accosωct
Basic SSB signal generation
In simple terms, we create an SSB signal from a DSB signal

using a sideband filter
The mathematical representation of LSSB and USSB signals

makes use of Hilbert transform concepts and analytic signals

CONTENTS
DSB Signal Starting Point
f
-fc fc
sgn(f + fc)/2
Formation of HL(f) +1/2
-1/2
-sgn(f - fc)/2
+1/2
-1/2
1 HL(f) = [sgn(f + fc) - sgn(f - fc)]/2
f
-fc fc
An ideal LSSB filter
From the frequency domain expression for the LSSB, we can

ultimately obtain an expression for the LSSB signal, xcLSSB .t /,
in the time domain
Start with XDSB.f / and the filter HL.f /

1
XcLSSB .f / D Ac M.f C fc / C M.f fc /
2
1
sgn.f C fc / sgn.f fc /
2

rewrite 1 1
XcLSSB .f / D Ac M.f C fc /sgn.f C fc /
4
M.f fc /sgn.f fc /
1
Ac M.f C fc /sgn.f fc /
4
M.f fc /sgn.f C fc /
rewrite 2 1
D Ac M.f C fc /sgn.f C fc /
4
M.f fc /sgn.f fc /
1
C Ac M.f C fc / C M.f fc /
4
The inverse Fourier transform of the second term is DSB, i.e.,
1 F 1
Ac m.t / cos !c t ! Ac M.f C fc / C M.f fc /
2 4
The first term can be inverse transformed using the Hilbert
transform definition
F
O /
m.t ! j sgn.f / M.f /
so
1
O /e j!c t
˚
F M.f ˙ fc /sgn.f ˙ fc / D j m.t
F
since m.t /e ˙j!c t ! M.f ˙ fc /
Thus
1 1
˚
Ac F M.f Cfc /sgn.f Cfc / M.f fc /sgn.f fc /
4
1 j!c t j!c t
1
D Ac j m.t O /e O /e
j m.t D m.t
O / sin !c t
4 2
CONTENTS
Finally,
1 1
xcLSSB .t / D Ac m.t / cos !c t C Ac m.t
O / sin !c t
2 2
Similarly for USSB it can be shown that

1 1
xcUSSB .t / D Ac m.t / cos !c t O / sin !c t
Ac m.t
2 2
The direct implementation of SSB is very difficult due to the

requirements of the filter
By moving the phase shift frequency from fc down to DC (0

Hz) the implementation is much more reasonable (this applies
to a DSP implementation as well)
The phase shift is not perfect at low frequencies, so the modu-

lation must not contain critical information at these frequencies
m(t) cosωct +
0 o Carrier Osc.
cosωct xc(t)
H(f) = -90o
-jsgn(f) sinωct + LSB
- USB
Phase shift modulator for SSB

Demodulation
The coherent demodulator first discussed for DSB, also works
for SSB
d(t) yD(t)
xr(t) LPF
1/Ac scale factor

4cos[2πfct + θ(t)] included
Coherent demod for SSB
Carrying out the analysis to d.t /, first we have

1
d.t / D Ac m.t / cos !c t ˙ m.tO / sin !c t 4 cos.!c t C .t //
2
D Ac m.t / cos .t / C Ac m.t / cosŒ2!c t C .t /
Ac m.t
O / sin .t / ˙ Ac m.t
O / sinŒ2!c t C .t /
so
yD .t / D m.t / cos .t / m.t
O / sin .t /
.t/ small
' m.t / m.t
O /.t /
O / sin .t / term represents crosstalk
– The m.t
Another approach to demodulation is to use carrier reinsertion
and envelope detection
e(t) Envelope
xr(t) yD(t)
Detector
Kcosωct

CONTENTS
e.t / D xr .t / C K cos !c t

1 1
D Ac m.t / C K cos !c t ˙ Ac m.t
O / sin !c t
2 2
To proceed with the analysis we must find the envelope of e.t /,

which will be the final output yD .t /
Finding the envelope is a more general problem which will be

useful in future problem solving, so first consider the envelope
of
x.t / D a.t / cos !c t b.t / sin !c t

„ƒ‚… „ƒ‚…
inphase quadrature
D Re a.t /e j!c t C jb.t /e j!c t
˚
j!c t
˚
D Re Œa.t / C jb.t / e
„ ƒ‚ …
Q
R.t/Dcomplex envelope
In a phasor diagram x.t / consists of an inphase or direct com-

ponent and a quadrature component
Quadrature - Q
~
Note: R(t) =
R(t)ejθ(t) = a(t) + jb(t)

R(t)
b(t)
θ(t)
In-phase - I
a(t)

where the resultant R.t / is such that
a.t / D R.t / cos .t /

b.t / D R.t / sin .t /
which implies that

x.t / D R.t / cos .t / cos !c t sin .t / sin !c t

D R.t / cos !c t C .t /
where .t / D tan 1Œb.t /=a.t /
The signal envelope is thus given by

p
R.t / D a2.t / C b 2.t /
The output of an envelope detector will be R.t / if a.t / and

b.t / are slowly varying with respect to cos !c t
In the SSB demodulator

s
2 2
1 1
yD .t / D Ac m.t / C K C Ac m.t
O /
2 2
If we choose K such that .Ac m.t /=2 C K/2 .Ac m.t

O /=2/2,
then
1
yD .t / ' Ac m.t / C K
2
Note:
– The above analysis assumed a phase coherent reference

– In speech systems the frequency and phase can be ad-
justed to obtain intelligibility, but not so in data systems

CONTENTS
– The approximation relies on the binomial expansion
1=2 1
.1 C x/ ' 1 C x for jxj 1
2
Example 3.6: Noncoherent Carrier Reinsertion
Let m.t / D cos !mt, !m !c and the reinserted carrier be

K cosŒ.!c C !/t
Following carrier reinsertion we have
1
e.t / D Ac cos !mt cos !c t
2
1
Ac sin !mt sin !c t C K cos .!c C !/t
2
1
D Ac cos .!c ˙ !m/t C K cos .!c C !/t
2
We can write e.t / as the real part of a complex envelope times

a carrier at either !c or !c C !
In this case, since K will be large compared to Ac =2, we write
1 n
˙j!m t j!c t
o
e.t / D Ac Re e e
2 n o
j.!c C!/t
C KRe 1 e
n 1 o
j.˙!m !/t j.!c C!/t
D Re Ac e CK e
„2 ƒ‚ …
Q
complex envelope R.t/

Finally expanding the complex envelope into the real and imag-
inary parts we can find the real envelope R.t /
hn 1 o2
yD .t / D Ac cosŒ˙!m C !/t C K
2
n1 o2i1=2
C Ac sinŒ.˙!m C !/t
2
1
' Ac cosŒ.!m !/t C K
2
where the last line follows for K Ac
Note that the frequency error ! causes the recovered mes-

sage signal to shift up or down in frequency by !, but not
both at the same time as in DSB, thus the recovered speech
signal is more intelligible
3.1.4 Vestigial-Sideband Modulation

Vestigial sideband (VSB) is derived by filtering DSB such that
one sideband is passed completely while only a vestige remains
of the other
Why VSB?
1. Simplifies the filter design

2. Improves the low-frequency response and allows DC to
pass undistorted
3. Has bandwidth efficiency advantages over DSB or AM,
similar to that of SSB

CONTENTS
A primary application of VSB is the video portion of analog

television (note HDTV replaces this in the US with 8VSB1)
The generation of VSB starts with DSB followed by a filter

that has a 2ˇ transition band, e.g.,
8
<0; f < Fc ˇ
ˆ
ˆ
jH.f /j D f .f2ˇc ˇ/ ; fc ˇ f fc C ˇ
ˆ
:1; f >f Cˇ
ˆ
c
|H(f)|
f
fc - β fc fc + β
Ideal VSB transmitter filter amplitude response
VSB can be demodulated using a coherent demod or using car-

rier reinsertion and envelope detection
Transmitted Two-Tone Spectrum
(only single-sided shown)
B/2
A(1 - ε)/2
Aε/2
f
0 f - f2 f - f1 fc f + f1 f + f2
Two-tone VSB signal

1
http://www.tek.com/document/primer/fundamentals-8vsb

Suppose the message signal consists of two tones
m.t / D A cos !1t C B cos !2t
Following the DSB modulation and VSB shaping,

1
xc .t / D A cos.!c !1/t
2
1 1
C A.1 / cos.!c C !1/t C B cos.!c C !2/t
2 2
A coherent demod multiplies the received signal by 4 cos !c t

to produce
e.t / D A cos !1t C A.1 / cos !1t C B cos !2t

D A cos !1t C B cos !2t
which is the original message signal
The symmetry of the VSB shaping filter has made this possible
In the case of broadcast TV the carrier in included at the trans-

mitter to insure phase coherency and easy demodulation at the
TV receiver (VSB + Carrier)
– Very large video carrier power was required for typical

TV station, i.e., greater than 100,000 W
– To make matters easier still, the precise VSB filtering is
not performed at the transmitter due to the high power
requirements, instead the TV receiver did this
– Further study is needed on today’s 8VSB

CONTENTS
Transmitter Video Carrier

Output Audio Carrier
(f - fcv) MHz
-1.75 -0.75 0 4.0 4.5 4.75
2β interval
Receiver 1
Shaping
Filter
(f - fcv) MHz
-0.75 0 0.75 4.0 4.75
Broadcast TV transmitter spectrum and receiver shaping filter
3.1.5 Frequency Translation and Mixing

Used to translate baseband or bandpass signals to some new
center frequency
m(t)cosω1t
e(t) BPF
at
f2 f
f
f1 f2
Local oscillator of the form

2cos[(w1 ± w2 )t] = 2 cos(wLO t )
Frequency translation system
Assuming the input signal is DSB of bandwidth 2W the mixer

(multiplier) output is
local osc (LO)
‚ …„ ƒ
e.t / D m.t / cos.!1t / 2 cos.!1 ˙ !2/t
D m.t / cos.!2t / C m.t / cosŒ.2!1 ˙ !2/t

The bandpass filter bandwidth needs to be at least 2W Hz wide
Note that if an input of the form k.t / cosŒ.!1 ˙2!2/t is present

it will be converted to !2 also, i.e.,
e.t / D k.t / cos.!2t / C k.t / cosŒ.2!1 ˙ 3!2/t ;
and the bandpass filter output is k.t / cos.!2t /
The frequencies !1 ˙ 2!2 are the image frequencies of !1 with

respect to !LO D !1 ˙ !2
Example 3.7: AM Broadcast Superheterodyne Receiver
Tunable IF Filt/ Env Audio

RF-Amp Amp Det Amp
fIF
Automatic gain
Local control For AM BT = 2W
Osc.
Joint tuning
AM Broadcast Specs: fc = 540 to 1600 kHz on 10 kHz spacings

carrier stability
Modulated audio flat 100 Hz to 5 kHz
Typical fIF = 455 kHz
Classical AM superheterodyne receiver
We have two choices for the local oscillator, high-side or low-

side tuning

CONTENTS
– Low-side: 540 455 fLO 1600 455 or 85 fLO

1145, all frequencies in kHz
– High-side: 540 C 455 fLO 1600 C 455 or 995
fLO 2055, all frequencies in kHz
The high-side option is advantageous since the tunable oscil-

lator or frequency synthesizer has the smallest frequency ratio
fLO,max=fLO,min D 2055=995 D 2:15
Suppose the desired station is at 560 kHz, then with high-side

tuning we have fLO D 560 C 455 D 1015 kHz
The image frequency is at fimage D fc C 2fIF D 560 C 2

455 D 1470 kHz (note this is another AM radio station center
frequency
Desired Potential Image

BRF
Input
f (kHz)
455 560 1470
fLO fIF fIF
f (kHz)
BIF 1015 1470
Mixer IF BPF (560+455)
Output
f (kHz)
Image 1015-560 1575
Out of This is removed (560+1015)
mixer with RF BPF
f (kHz)
455 2485
0 1470-1015
(1470+1015)
Receiver frequency plan including images

Example 3.8: A Double-Conversion Receiver

fc = 162.475 MHz
(WX #4) 10.7 MHz & 455 kHz &
335.65 MHz 21.855 MHz
Tunable 10.7 MHz 455 kHz FM
RF-Amp IF BPF IF BPF Demod
1st 2nd
LO LO
fLO1 = 173.175 MHz fLO2 = 11.155 MHz
Double-conversion superheterodyne receiver (Lab 4)
Consider a frequency modulation (FM) receiver that uses double-

conversion to receive a signal on carrier frequency 162.475
MHz (weather channel #4 here in Colorado Springs)
– Frequency modulation will be discussed in the next sec-

tion
The dual-conversion allows good image rejection by using a

10.7 MHz first IF and then can provide good selectivity by
using a second IF at 455 kHz; why?
– The ratio of bandwidth to center frequency can only be so

small in a low loss RF filter
– The second IF filter can thus have a much narrower band-
width by virtue of the center frequency being much lower
A higher first IF center frequency moves the image signal fur-

ther away from the desired signal

CONTENTS
– For high-side tuning we have fimage D fc C 2fIF D fc C

21:4 MHz
Double-conversion receivers are more complex to implement
Mixers
The multiplier that is used to implement frequency translation

is often referred to as a mixer
In the world of RF circuit design the term mixer is more ap-

propriate, as an ideal multiplier is rarely available
Instead active and passive circuits that approximate signal mul-

tiplication are utilized
The notion of mixing comes about from passing the sum of two
signals through a nonlinearity, e.g.,
y.t / D Œa1x1.t / C a2x2.t /2 C other terms

D a12x12.t / C 2a1a2x1.t /x2.t / C a22x22.t /
In this mixing application we are most interested in the center

term

ydesired.t / D 2a1a2 x1.t / x2.t /
Clearly this mixer produces unwanted terms (first and third),

and in general many other terms, since the nonlinearity will
have more than just a square-law input/output characteristic

A diode or active device can be used to form mixing products

as described above, consider the dual-gate MEtal Semiconduc-
tor FET (MESFET) mixer shown below
Nonlinear Device
VRF
VIN zL VOUT
VLO
Mixer concept
+5V
R2
10Ω
C3
47pF C4
0.01uF
L1 L3
5 turns, 28 AWG L4
.050 I.D. 270nH IF
C1 270nH
0.5pF G1 C8
LO D Q1 C7
G2 NE25139 42pF 82pF
RF S
C2
0.5pF
R1 C5 C8
L2 0.01uF
5 turns, 28 AWG 47pF
.050 I.D. 270Ω
Dual-Gate MESFET Active Mixer
The double-balanced mixer (DBM), which can be constructed

using a diode ring, provides better isolation between the RF,
LO, and IF ports
When properly balanced the DBM also allows even harmonics

to be suppressed in the mixing operation

CONTENTS
A basic transformer coupled DBM, employing a diode ring, is

shown below, followed by an active version
The DBM is suitable for use as a phase detector in phase-

locked loop applications
mixer
LO source LO RF RF source
input D2 D1
input
RG vp(t) vi (t) RG
D3 D4
IF vo(t)
out
IF load
RL
Passive Double-Balanced Mixer (DBM)
C9 T1
3 6
L1 R3 IF OUT
5V
2 4:1 (200:50)
TRANSFORMER
C8
C11 1 4
L2 R4
C10
GND
GND
GND
IF+
IF-
20
19
18
17
16
C1
RF MAX9982 LO2
RFIN 1 15 LO2
TAP GND C7
2 14
C3 C2 GND GND
3 13
RFBIAS GND
4 12
R1 GND LO1
5 11 LO1
C6
10
6
9
GND
GND
LOSEL
VCC
VCC
5V 5V
C4 C5
LO SELECT
825 MHz to 915 MHz SiGe High-Linearity Active DBM

Example 3.9: Single Diode Mixer
VI Probe for PSD

AM Signal at 560 kHz,
10 kHz Tone Message Signal Sum Mixing Diode
V_out
R
R R I_Probe
ap_dio_1N4148_19930601 R2
R3 R1 I_Probe1
D1 R=50 Ohm
VtSine R=50 Ohm R R R=50 Ohm
AM_ModTuned Splitter_Resistive_6dB
SRC4 MOD1 R6 X3 R8
Vdc=0 V ModIndex=0.5 R=50 Ohm R=50 Ohm
Amplitude=0.5 V Fnom=560 kHz
Freq=560 kHz Rout=50 Ohm
VtSine
SRC5 VtSine
Vdc=0 V SRC2
Vdc=0 V
Amplitude=1.0 V
Freq=10 kHz Amplitude=2 V
Freq=1.015 MHz TRANSIENT
f
PspecTran Tran
Tran1
PspecTran
PspecTran1 StopTime=400.0 usec
PspecTran1=pspec_tran(V_out, 0,I_Probe1.i,5kHz,200) MaxTimeStep=10.0 nsec
-10
Input Signal (RF)

-20 455 kHz IF Output
LO Mix
Power (dBm)
@ 1015 kHz
-30
10 kHz
sidebands
-40
-50
400 420 440 460 480 500 520 540 560 580 600
freq, KHz
ADS single diode mixer simulation: 560 kHz ! 455 kHz

CONTENTS
3.2 Interference
Interference is a fact of life in communication systems. A through
understanding of interference requires a background in random sig-
nals analysis (Chapter 6 of the text), but some basic concepts can
be obtained by considering a single interference at fc C fi that lies
close to the carrier fc
3.2.1 Interference in Linear Modulation

Single-Sided Spectrum
Xr(f) Ac
1 1
A A
2 m 2 m
Ai
f
fc - fm fc fc + fm fc + fi
AM carrier with single tone interference
If a single tone carrier falls within the IF passband of the re-

ceiver what problems does it cause?
Coherent Demodulator

xr .t / D Ac cos !c t C Am cos !mt cos !c t
C Ai cos.!c C !i /t
– We multiply xr .t / by 2 cos !c t and lowpass filter
yD .t / D Am cos !mt C Ai cos !i t
„ ƒ‚ …
interference
Envelope Detection: Here we need to find the received enve-

lope relative to the strongest signal present

3.2. INTERFERENCE
– Case Ac Ai
– We will expand xr .t / in complex envelope form by first
noting that
Ai cos.!c C!i /t D Ai cos !i t cos !c t Ai sin !i t sin !c t
now,
˚
xr .t / D Re Ac C Am cos !mt C Ai cos !i t
jAi sin !i t e j!c t

Q j!c t
˚
D Re R.t /e
so
Q /j
R.t / D jR.t
h
D .Ac C Am cos !mt C Ai cos !i t /2
i1=2
2
C .Ai sin !i t /
' Ac C Am cos !mt C Ai cos !i t
assuming that Ac Ai
– Finally,
yD .t / ' Am cos !mt C Ai cos !i t

„ ƒ‚ …
interference
– Case Ai >> Ac
– Now the interfering term looks like the carrier and the re-
maining terms look like sidebands, LSSB sidebands rela-
tive to fc C fi to be specific

CONTENTS
– From SSB envelope detector analysis we expect

1
yD .t / ' Am cos.!i C !m/t C Ac cos !i t
2
1
C Am cos.!i !m/t
2
and we conclude that the message signal is lost!

3.3. SAMPLING THEORY
3.3 Sampling Theory

We now return to text Chapter 2, Section 8, for an introduc-
tion/review of sampling theory
Consider the representation of continuous-time signal x.t / by

the sampled waveform
" 1 # 1
X X
xı .t / D x.t / ı.t nTs / D x.nTs /ı.t nTs /
nD 1 nD 1
x(t) xδ(t)
Sampling
t t
0 -Ts 0 Ts 2Ts 3Ts 4Ts 5Ts
How is Ts selected so that x.t / can be recovered from xı .t /?
Uniform Sampling Theorem for Lowpass Signals

Given
Ffx.t /g D X.f / D 0; for f > W
then choose
1
Ts < or fs > 2W .fs D 1=Ts /
2W
to reconstruct x.t / from xı .t / and pass xı .t / through an ideal
LPF with cutoff frequency W < B < fs W
2W D Nyquist frequency
fs =2 D folding frequency
CONTENTS
proof:
1
" #
X
Xı .f / D X.f / fs ı.f nfs /
nD 1
but X.f / ı.f nfs / D X.f nfs /, so

1
X
Xı .f / D fs X.f nfs /
nD 1
X0 X(f)
f
-W W
Lowpass
reconstruction Xδ(f) Guard band
filter = fs - 2W
X0 fs
... ...
fs > 2W
f
-fs -W 0 W fs-W fs
Aliasing
fs < 2W X0 fs
... ...
f
-2fs -fs 0 fs 2fs
Spectra before and after sampling at rate fs
As long as fs W > W or fs > 2W there is no aliasing

(spectral overlap)

To recover x.t / from xı .t / all we need to do is lowpass filter

the sampled signal with an ideal lowpass filter having cutoff
frequency W < fcutoff < fs W
In simple terms we set the lowpass bandwidth to the folding

frequency, fs =2
Suppose the reconstruction filter is of the form

f
H.f / D H0… e j 2f t0
2B
we then choose W < B < fs W
For input Xı .f /, the output spectrum is

j 2f t0
Y .f / D fs H0X.f /e
and in the time domain
y.t / D fs H0x.t t0 /
If the reconstruction filter is not ideal we then have to design

the filter in such a way that minimal desired signal energy is re-
moved, yet also minimizing the contributions from the spectral
translates either side of the n D 0 translate
The reconstruction operation can also be viewed as interpolat-

ing signal values between the available sample values
Suppose that the reconstruction filter has impulse response h.t /,

CONTENTS
then
1
X
y.t / D x.nTs /h.t nTs /
nD 1
1
X
D 2BH0 x.nTs /sincŒ2B.t t0 nTs /
nD 1
where in the last lines we invoked the ideal filter described

earlier
Uniform Sampling Theorem for Bandpass Signals

If x.t / has a single-sided bandwidth of W Hz and
Ffx.t /g D 0 for f > fu
then we may choose

2fu
fs D
m
where
fu
mD ;
W
which is the greatest integer less than or equal to fu=W
Example 3.10: Bandpass signal sampling
1 X(f)
0.8
0.6
0.4
0.2
f
4 2 2 4
Input signal spectrum

In the above signal spectrum we see that
W D 2; fu D 4 fu=W D 2 ) m D 2
so
2.4/
fs D D4
2
will work
The sampled signal spectrum is

1
X
Xı .f / D 4 X.f nfs /
nD 1
Recover with
bandpass filter Xδ(f)
4
3
2
1
f
15 10 5 5 10 15
-3fs -2fs -fs fs 2fs 3fs
Spectrum after sampling

CONTENTS
3.4 Analog Pulse Modulation

The message signal m.t / is sampled at rate fs D 1=Ts
A characteristic of the transmitted pulse is made to vary in a

one-to-one correspondence with samples of the message signal
A digital variation is to allow the pulse attribute to take on

values from a finite set of allowable values
3.4.1 Pulse-Amplitude Modulation (PAM)

PAM produces a sequence of flat-topped pulses whose ampli-
tude varies in proportion to samples of the message signal
Start with a message signal, m.t /, that has been uniformly

sampled
1
X
mı .t / D m.nTs /ı.t nTs /
nD 1
The PAM signal is

1
X t .nTs C =2/
mc .t / D m.nTs /…
nD 1

m(t)
mc(t)
τ
t
0 τ Ts 2Ts 3Ts 4Ts
PAM waveform

3.4. ANALOG PULSE MODULATION
It is possible to create mc .t / directly from mı .t / using a zero-

order hold filter, which has impulse response

t =2
h.t / D …

and frequency response
jf
H.f / D sinc.f /e
mδ(t) h(t) mc(t)
How does h.t / change the recovery operation from the case of
ideal sampling?
– If Ts we can get by with just a lowpass reconstruc-

tion filter having cutoff frequency at fs =2 D 2=Ts
– In general, there may be a need for equalization if is on
the order of Ts =4 to Ts =2
Lowpass
sinc() function reconstruction
envelope filter
f
-fs -W W fs
mc(t) Lowpass m(t)
Recovery of m.t/ from mc .t/

CONTENTS
3.4.2 Pulse-Width Modulation (PWM)

A PWM waveform consists of pulses with width proportional
to the sampled analog waveform
For bipolar m.t / signals we may choose a pulse width of Ts =2

to correspond to m.t / D 0
The biggest application for PWM is in motor control
It is also used in class D audio power amplifiers
A lowpass filter applied to a PWM waveform recovers the

modulation m.t /
PWM Signal
1
0.5
t
20 10 10 20
0.5
Analog input m(t)
1
Example PWM signal
3.4.3 Pulse-Position Modulation

With PPM the displacement in time of each pulse, with re-
spect to a reference time, is proportional to the sampled analog
waveform
The time axis may be slotted into a discrete number of pulse

positions, then m.t / would be quantized
Digital modulation that employs M slots, using nonoverlap-

ping pulses, is a form of M -ary orthogonal communications

3.5. DELTA MODULATION AND PCM
– PPM of this type is finding application in ultra-wideband

communications
PPMSignal
1
0.5
t
20 10 10 20
0.5
Analog input m(t) 1
Example PPM signal
3.5 Delta Modulation and PCM

This section considers two pure digital pulse modulation schemes
Pure digital means that the output of the modulator is a binary

waveform taking on only discrete values
3.5.1 Delta Modulation (DM)

The message signal m.t / is encoded into a binary sequence
which corresponds to changes in m.t / relative to reference
waveform ms .t /
DM gets its name from the fact that only the difference from
sample-to-sample is encoded
The sampling rate in combination with the step size are the two
primary controlling modulator design parameters

CONTENTS
m(t) + 1 ∆(t) xc(t)

d(t) -1
−
Pulse Modulator
ms(t) =
Control the
δ0 step size
Delta modulator with step size parameter ı0
Start-up transient
1
m(t) (blue)
m(t) and ms(t)
0.5
ms(t) (red)
0
Slope δ0 = 0.15
−0.5
overload
−1
0 0.005 0.01 0.015 0.02 0.025 0.03 0.035 0.04 0.045 0.05
Time (s)
0.5
xc(t)
−0.5
−1
0 0.005 0.01 0.015 0.02 0.025 0.03 0.035 0.04 0.045 0.05
Time (s)
Delta modulator waveforms
The maximum slope that can be followed is ı0=Ts

A MATLAB DM simulation function is given below
function [t_o,x,ms] = DeltaMod(m,fs,delta_0,L)

% [t,x,ms] = DeltaMod(m,fs,delta_0,L)
%
% Mark Wickert, April 2006
n = 0:(L*length(m))-1;
t_o = n/(L*fs);
ms = zeros(size(m));
x = zeros(size(m));
ms_old = 0; % zero initial condition
for k=1:length(m)
x(k) = sign(m(k) - ms_old);
ms(k) = ms_old + x(k)*delta_0;
ms_old = ms(k);
end
x = [x; zeros(L-1,length(m))];
x = reshape(x,1,L*length(m));
The message m.t / can be recovered from xc .t / by integrating

and then lowpass filtering to remove the stair step edges (low-
pass filtering directly is a simplification)
Slope overload can be dealt with through an adaptive scheme
– If m.t / is nearly constant keep the step size ı0 small

– If m.t / has large variations, a larger step size is needed
With adaptive DM the step size is controlled via a variable gain

amplifier, where the gain is controlled by square-law detecting
the output of a lowpass filter acting on xc .t /

CONTENTS
m(t) + 1 ∆(t) xc(t)
d(t) -1
−
Pulse Modulator
ms(t)
VGA
( )2 LPF
Means to obtain a variable step size DM
3.5.2 Pulse-Code Modulation (PCM)

Each sample of m.t / is mapped to a binary word by
1. Sampling
2. Quantizing
3. Encoding
m(t) PCM
Sampler Quantizer Encoder Output
Equivalent
Views
m(t) Sample Analog to Parallel Serial
& Digital to Serial Data
Hold Converter n Converter

Quant. Encoded
Quantizer Bits: n = 3, q = 2n = 8
Level Output
7 111
6 110 m(t)
5 101
4 100
3 011
2 010
1 001
0 000 t
0 Ts 2Ts 3Ts 4Ts 5Ts 6Ts 7Ts
Encoded Serial PCM Data: 001 100 110 111 110 100 010 010 ...
3-Bit PCM encoding
Assume that m.t / has bandwidth W Hz, then
– Choose fs > 2W
– Choose n bits per sample (q D 2n quantization levels)
– ) 2nW binary digits per second must be transmitted
Each pulse has width no more than

1
. /max D;
2nW
so using the fact that the lowpass bandwidth of a single pulse
is about 1=.2 / Hz, we have that the lowpass transmission
bandwidth for PCM is approximately
B ' kW n;
where k is a proportionality constant
When located on a carrier the required bandwidth is doubled

CONTENTS
Binary phase-shift keying (BPSK), mentioned earlier, is a pop-

ular scheme for transmitting PCM using an RF carrier
Many other digital modulation schemes are possible
The number of quantization levels, q D log2 n, controls the

quantization error, assuming m.t / lies within the full-scale range
of the quantizer
Increasing q reduces the quantization error, but also increases

the transmission bandwidth
The error between m.kTs / and the quantized value QŒm.kTs /,
denoted e.n/, is the quantization error
If n D 16, for example, the ratio of signal power in the samples

of m.t /, to noise power in e.n/, is about 95 dB (assuming m.t /
stays within the quantizer dynamic range)
Example 3.11: Compact Disk Digital Audio
CD audio quality audio is obtained by sampling a stereo source

at 44.1 kHz
PCM digitizing produces 16 bits per sample per L/R audio

channel

3.6. MULTIPLEXING
0.163 mm
One Frame of 12 Audio Samples
Synch Sub Parity Parity

(27 bits) Code Data (96 bits) (32 bits) Data (96 bits) (32 bits)
(8 bits)
CD recoding frame format
The source bit rate is thus 2 16 44:1ksps D 1:4112 Msps
Data framing and error protection bits are added to bring the
total bit count per frame to 588 bits and a serial bit rate of
4.3218 Mbps
3.6 Multiplexing
It is quite common to have multiple information sources lo-
cated at the same point within a communication system
To simultaneously transmit these signals we need to use some

form of multiplexing

CONTENTS
There is more than one form of multiplexing available to the

communications engineer
In this chapter we consider time-division multiplexing, while

in Chapter 4 frequency division multiplexing is described
3.6.1 Time-Division Multiplexing (TDM)
Time division multiplexing can be applied to sampled analog

signals directly or accomplished at the bit level
We assume that all sources are sample at or above the Nyquist

rate
Both schemes are similar in that the bandwidth or data rate of

the sources being combined needs to be taken into account to
properly maintain real-time information flow from the source
to user
For message sources with harmonically related bandwidths we

can interleave samples such that the wideband sources are sam-
pled more often
To begin with consider equal bandwidth sources

3.6. MULTIPLEXING
Info. Info.
Source 1 User 1
Synchronization
Required
Info. Info.
Channel
Source 2 User 2
...
...
Commutators
Info. Info.
Source N User N
For equal bandwidth: s1s2s3 s1s2s3 s1s2s3 s1s2s3 s1s2s3 s1s2s3 s1s2s3 ....
Analog TDM (equal bandwidth sources)
Suppose that m1.t / has bandwidth 3W and sources m2.t /, m3.t /,

and m4.t / each have bandwidth W , we could send the samples
as
s1s2s1s3s1s4s1s2s1 : : :
with the commutator rate being fs > 2W Hz
The equivalent transmission bandwidth for multiplexed signals

can be obtained as follows
– Each channel requires greater than 2Wi samples/s

– The total number of samples, ns , over N channels in T s
is thus
N
X
ns D 2Wi T
iD1
– An equivalent signal channel of bandwidth B would pro-

duce 2BT D ns samples in T s, thus the equivalent base-

CONTENTS
band signal bandwidth is

N
X
BD Wi Hz
i D1
which is the same minimum bandwidth required for FDM

using SSB
– Pure digital multiplexing behaves similarly to analog mul-
tiplexing, except now the number of bits per sample, which
takes into account the sample precision, must be included
– In the earlier PCM example for CD audio this was taken
into account when we said that left and right audio chan-
nels each sampled at 44.1 ksps with 16-bit quantizers,
multiplex up to
2 16 44; 100 D 1:4112 Msps

3.6. MULTIPLEXING
Example 3.12: Digital Telephone System

The North American digital TDM hierarchy is based on a sin-
gle voice signal sampled at 8000 samples per second using a
7-bit quantizer plus one signaling bit
The serial bit-rate per voice channel is 64 kbps
North American Digital TDM Hierarchy)

Digital No. of 64 kbps
Signal Bit Rate PCM VF Transmission
Sys. Number R (Mb/s) Channels Media Used
DS-0 0.064 1 Wire pairs
T1 DS-1 1.544 24 Wire pairs
T1C DS-1C 3.152 48 Wire pairs
T2 DS-2 6.312 96 Wire pairs
T3 DS-3 44.736 672 Coax, radio, fiber
DS-3C 90.254 1344 Radio, fiber
DS-4E 139.264 2016 Radio, fiber, coax
T4 DS-4 274.176 4032 Coax, fiber
DS-432 432.00 6048 Fiber
T5 DS-5 560.160 8064 Coax, fiber
Consider the T1 channel which contains 24 voice signals
Eight total bits are sent per voice channel at a sampling rate of
8000 Hz
The 24 channels are multiplexed into a T1 frame with an extra

bit for frame synchronization, thus there are 24 8 C 1 D 193
bits per frame

CONTENTS
Frame period is 1=8000 D 0:125 ms, so the serial bit rate is

193 8000 D 1:544 Mbps
Four T1 channels are multiplexed into a T2 channel (96 voice

channels)
Seven T2 channels are multiplexed into a T3 channel (672

voice channels)
Six T3 channels are multiplexed into a T4 channel (4032 voice

channels)

Linear Modulation Techniques

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Linear Modulation Techniques

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Chapter 3

Linear Modulation Techniques

3.6.1 Time-Division Multiplexing (TDM) . . . . . 3-64

3-2 ECE 5625 Communication Systems I

 We are typically interested in locating a message signal to some

 The carrier of the message signal is usually sinusoidal

 A modulated carrier can be represented as

where A.t / is linear modulation, fc the carrier frequency, and

3.1 Linear Modulation

with A.t / placed in one-to-one correspondence with the mes-

3.1.1 Double-Sideband Modulation (DSB)

 From the modulation theorem it follows that

xc(t) carrier filled envelope

DSB time domain waveforms

m(t) xc(t) xr(t) d(t) yD(t)

Modulator Channel Demodulator

3-4 ECE 5625 Communication Systems I

 For an ideal channel xr .t / D xc .t /, so

 The waveform and spectra of d.t / is shown below (assuming

Waveform and spectrum of d.t/

 Typically the carrier frequency is much greater than the mes-

 The scale factor Ac can be dealt with in downstream signal

ECE 5625 Communication Systems I 3-5

 Assuming an ideal lowpass filter, the only requirement is that

 The difficulty with this demodulator is the need for a coherent

 To see how critical this is to demodulation of m.t / suppose that

c.t / D 2 cosŒ2fc t C .t /

where .t / is a time-varying phase error

 With the imperfect carrier reference signal

d.t / D Ac m.t / cos .t / C Ac m.t / cosŒ2.2fc /t C .t /

 Suppose that .t / is a constant or slowly varying, then the

 Even a slowly varying attenuation can be very detrimental from

– If say .t / D 2f t and m.t / D cos.2fmt /, then

 Being able to generate a coherent local reference is also a prac-

3-6 ECE 5625 Communication Systems I

 One scheme is to simply square the received DSB signal

xr2.t / D A2cm2.t / cos2.2fc t /

xr(t) LPF yD(t)

 Assuming that m2.t / has a nonzero DC value, then the double

Filter this component

for coherent demod

 Note that unless m.t / has a DC component, Xc .f / will not

ECE 5625 Communication Systems I 3-7

 Consider transmitting a small amount of unmodulated carrier

Accos2πfct use a narrowband filter

3.1.2 Amplitude Modulation

xc .t / D A C m.t / A0c cos.2fc t /

where Ac D AA0c , mn.t / is the normalized message such that

and a is the modulation index

A + max m(t) A + min m(t)

Note that the enve-

Bias term A Accos[2πfct]

Generation of AM and a sample wavefrom

 Note that if m.t / is symmetrical about zero and we define d1 as

d1 d2 2Œ.A C j min m.t /j/ .A j min m.t /j/

ECE 5625 Communication Systems I 3-9

 The message signal can be recovered from xc .t / using a tech-

 A diode, resistor, and capacitor is all that is needed to construct

 The circuit shown above is actually a combination of a nonlin-

 A detailed analysis of this circuit is more difficult than you

 A SPICE circuit simulation is relatively straight forward, but it

3-10 ECE 5625 Communication Systems I

 The simple envelope detector fails if Ac Œ1 C amn.t / < 0

We are typically interested in locating a message signal to some

The carrier of the message signal is usually sinusoidal

A modulated carrier can be represented as

From the modulation theorem it follows that

For an ideal channel xr .t / D xc .t /, so

The waveform and spectra of d.t / is shown below (assuming

Typically the carrier frequency is much greater than the mes-

The scale factor Ac can be dealt with in downstream signal

Assuming an ideal lowpass filter, the only requirement is that

The difficulty with this demodulator is the need for a coherent

To see how critical this is to demodulation of m.t / suppose that

c.t / D 2 cosŒ2fc t C .t /

where .t / is a time-varying phase error

With the imperfect carrier reference signal

d.t / D Ac m.t / cos .t / C Ac m.t / cosŒ2.2fc /t C .t /

Suppose that .t / is a constant or slowly varying, then the

Even a slowly varying attenuation can be very detrimental from

– If say .t / D 2f t and m.t / D cos.2fmt /, then

Being able to generate a coherent local reference is also a prac-

One scheme is to simply square the received DSB signal

xr2.t / D A2cm2.t / cos2.2fc t /

Assuming that m2.t / has a nonzero DC value, then the double

Note that unless m.t / has a DC component, Xc .f / will not

Consider transmitting a small amount of unmodulated carrier

xc .t / D A C m.t / A0c cos.2fc t /

Note that if m.t / is symmetrical about zero and we define d1 as

d1 d2 2Œ.A C j min m.t /j/ .A j min m.t /j/

The message signal can be recovered from xc .t / using a tech-

A diode, resistor, and capacitor is all that is needed to construct

The circuit shown above is actually a combination of a nonlin-

A detailed analysis of this circuit is more difficult than you

A SPICE circuit simulation is relatively straight forward, but it

The simple envelope detector fails if Ac Œ1 C amn.t / < 0

The RC time constant cutoff frequency must lie between both

Digital signal processing based envelope detectors are also pos-

Historically the envelope detector has provided a very low-cost

The spectrum of an AM signal is

The total power in xc .t / is can be written in terms of the time

hxc2.t /i D hA2cŒ1 C amn.t /2 cos2.2fc t /i

If m.t / is slowly varying with respect to cos.2fc t /, i.e.,

a2hm2n.t /i also hm2.t /i

An AM signal of the form

xc .t / D Ac Œ1 C a cos.2fmt C =3/ cos.2fc t /

The modulation index is 0.8

The total power is

It should be clear that in this problem mn.t / D cos.2fmt /, so

Thus we see that

The magnitude and phase spectra can be plotted by first ex-

xc .t / D Ac cos.2fc t / C aAc cos.2fmt C =3/ cos.2fc t /

Find mn.t / and the efficiency E

From the definition of mn.t /

To obtain hm2n.t /i we form the time average

Suppose that the message signal is m.t / as given here

Now min m.t / D 2 and mn.t / D m.t /=2 and

Now when a D 1 we have Eff D 1=3 or just 33.3%

Suppose that m.t / is a sum of multiple sinusoids (multi-tone

To find mn.t / we need to find min m.t /

Suppose that M D 3 with fk D f65; 100; 35g Hz, Ak D

The normalization factor is approximately given by 7.246, that

To obtain the efficiency of multi-tone AM we first calculate

The maximum efficiency is just

A remaining interest is the spectrum of xc .t /

The information is redundant, meaning that m.t / can be recon-

Transmitting just the USB or LSB results in single-sideband

For m.t / having lowpass bandwidth of W the bandwidth re-

Since SSB operates by transmitting just one sideband, the trans-

The filtering required to obtain an SSB signal is best explained

The Hilbert transform of signal x.t / can be written in terms of

Note that since the Hilbert transform of x.t / is a =2 phase

It also follows that