2-Dieu Che Tuong Tu

Chapter
3
Linear Modulation . . . . . . . . . . . . . . . . . . . . 3.1.1 3.1.2 3.1.3 3.1.4 3.1.5 Double-Sideband Modulation (DSB) . . . . . . Amplitude Modulation . . . . . . . . . . . . . . 3-3 3-3 3-8 Single-Sideband Modulation . . . . . . . . . . . 3-21 Vestigial-Sideband Modulation . . . . . . . . . . 3-35 Frequency Translation and Mixing . . . . . . . . 3-38 Narrowband Angle Modulation . . . . . . . . . 3-48 Spectrum of an Angle-Modulated Signal . . . . 3-50 Power in an Angle-Modulated Signal . . . . . . 3-56 Bandwidth of Angle-Modulated Signals . . . . . 3-56 Narrowband-to-Wideband Conversion . . . . . . 3-63 Demodulation of Angle-Modulated Signals . . . 3-63 Interference in Linear Modulation . . . . . . . . 3-74 Interference in Angle Modulation . . . . . . . . 3-76
Analog Modulation
Contents
3.1
3.2
Angle Modulation . . . . . . . . . . . . . . . . . . . . 3-46 3.2.1 3.2.2 3.2.3 3.2.4 3.2.5 3.2.6
3.3
Interference . . . . . . . . . . . . . . . . . . . . . . . 3-73 3.3.1 3.3.2
3.4
Feedback Demodulators . . . . . . . . . . . . . . . . . 3-81
3-1
CHAPTER 3. ANALOG MODULATION
3.4.1 3.4.2 3.4.3 3.4.4 3.5 3.6
Phase-Locked Loops for FM Demodulation . . . 3-81 PLL Frequency Synthesizers . . . . . . . . . . . 3-102 Frequency-Compressive Feedback . . . . . . . . 3-106 Coherent Carrier Recovery for DSB Demodulation 3-108
Sampling Theory . . . . . . . . . . . . . . . . . . . . . 3-112 Analog Pulse Modulation . . . . . . . . . . . . . . . . 3-117 3.6.1 3.6.2 3.6.3 Pulse-Amplitude Modulation (PAM) . . . . . . . 3-117 Pulse-Width Modulation (PWM) . . . . . . . . . 3-119 Pulse-Position Modulation . . . . . . . . . . . . 3-119 Delta Modulation (DM) . . . . . . . . . . . . . 3-120 Pulse-Code Modulation (PCM) . . . . . . . . . 3-123 Frequency-Division Multiplexing (FDM) . . . . 3-127 Quadrature Multiplexing (QM) . . . . . . . . . . 3-130 Time-Division Multiplexing (TDM) . . . . . . . 3-131 3-135
3.7
Delta Modulation and PCM . . . . . . . . . . . . . . . 3-120 3.7.1 3.7.2
3.8
Multiplexing . . . . . . . . . . . . . . . . . . . . . . . 3-126 3.8.1 3.8.2 3.8.3
3.9
General Performance of Modulation Systems in Noise
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 efciently transmitted The carrier of the message signal is usually sinusoidal A modulated carrier can be represented as xc .t / D A.t / cos 2 fc 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.2 fc t / with A.t / placed in one-to-one correspondence with the message signal
3.1.1
Double-Sideband Modulation (DSB)
Let A.t / / m.t /, the message signal, thus xc .t / D Ac m.t / cos.2 fc t / From the modulation theorem it follows that 1 Xc .f / D Ac M.f 2
1 fc / C Ac M.f C fc / 2
3-3
m(t)
xc(t)
carrier filled envelope
DSB time domain waveforms

M(f) M(0)
f Xc(f) 1 A M(0) 2 c LSB fc
USB f
-fc
DSB spectra
Coherent Demodulation The received signal is multiplied by the signal 2 cos.2 fc t /, which is synchronous with the transmitter carrier
m(t) xc(t) xr(t) d(t) LPF yD(t)
Accos[2fct] Modulator
3-4
2cos[2fct] Channel Demodulator

For an ideal channel xr .t / D xc .t /, so d.t / D Ac m.t / cos.2 fc t / 2 cos.2 fc 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 /)
d(t) Lowpass filtering will remove the double frequency carrier term
D(f) AcM(0)
Lowpass modulation recovery filter 1 A M(0) 2 c f 2fc
1 A M(0) 2 c -2fc -W
Waveform and spectrum of d.t/
Typically the carrier frequency is much greater than the message bandwidth W , so m.t / can be recovered via lowpass ltering The scale factor Ac can be dealt with in downstream signal processing, e.g., an automatic gain control (AGC) amplier
ECE 5625 Communication Systems I 3-5
Assuming an ideal lowpass lter, the only requirement is that the cutoff frequency be greater than W and less than 2fc W The difculty 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 cos2 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 / cos2 fc 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 xed or time varying attenuation factor Even a slowly varying attenuation can be very detrimental from a distortion standpoint If say .t / D f t and m.t / D cos.2 fmt /, then yD .t / D 1 cos2 .fm 2 f /t C cos2 .fm C f /t
which is the sum of two tones Being able to generate a coherent local reference is also a practical manner
3-6 ECE 5625 Communication Systems I
One scheme is to simply square the received DSB signal

2 2 2 xr .t / D A2 c m .t / cos .2 fc t / 1 2 2 1 2 2 D Ac m .t / C Ac m .t / cos2 .2fc /t 2 2
xr(t) xr(t)
2
LPF
yD(t)
( )2
BPF very narrow (tracking) bandpass filter
divide by 2
Acos2fct
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 ltering by a narrowband bandpass lter, i.e., F fm2.t /g D k.f / C
Spectrum of m2(t) Filter this component for coherent demod k
f 2fc
Note that unless m.t / has a DC component, Xc .f / will not contain a carrier term (read .f fc ), thus DSB is also called a suppressed carrier scheme
3-7
Consider transmitting a small amount of unmodulated carrier
m(t) k Accos2fct AcM(0)/2 k << 1
xc(t)
use a narrowband filter (phase-locked loop) to extract 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.2 fc t / D Ac 1 C amn.t / cos.2 fc t / where Ac D AA0c , mn.t / is the normalized message such that min mn.t / D 1, mn.t / D and a is the modulation index aD
3-8
m.t / j min m.t /j
j min m.t /j A
A + max m(t)
A + min m(t)
xc(t) Ac(1 - a)
a<1 t Note that the envelope does not cross zero in the case of AM having a < 1 A + m(t)
m(t)
xc(t)
Bias term
Accos[2fct]
Generation of AM and a sample wavefrom
Note that if m.t / is symmetrical about zero and we dene d1 as the peak-to-peak value of xc .t / and d2 as the valley-to-valley value of xc .t /, it follows that aD proof: max m.t / D d1 d2 d1 C d2
min m.t / D j min m.t /j, so j min m.t /j/ j min m.t /j/
d1 d2 2.A C j min m.t /j/ .A D d1 C d2 2.A C j min m.t /j/ C .A j min m.t /j D Da A
3-9
The message signal can be recovered from xc .t / using a technique known as envelope detection A diode, resistor, and capacitor is all that is needed to construct and envelope detector
eo(t)
Recovered envelope with proper RC selection
xr(t)
eo(t)
0 The carrier is removed if 1/fc << RC << 1/W
Envelope detector
The circuit shown above is actually a combination of a nonlinearity and lter (system with memory) A detailed analysis of this circuit is more difcult 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 1 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
Digital signal processing based envelope detectors are also possible 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 Xc .f / D Ac .f 2 C fc / C .f C fc / fc / C Mn.f C fc /
pure carrier spectrum
aAc Mn.f 2
DSB spectrum
AM Power Efciency 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
2 2 2 hxc .t /i D hA2 c 1 C amn .t / cos .2 fc t /i A2 D c h1 C 2amn.t / C a2m2 n .t /1 C cos.2 .2fc /t i 2
If m.t / is slowly varying with respect to cos.2 fc t /, i.e., hm.t / cos !c t i ' 0;
then
2 hxc .t /i
A2 D c 1 C 2ahmn.t /i C a2hm2 n .t /i 2 A2 A2 a 2 A2 c c c 2 2 D 1 C a hm .t /i D C hm 2 n .t /i 2 2 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) Denition: AM Efciency Eff a 2 hm 2 hm2.t /i also n .t /i D D 2 .t / i 1 C a2hm2 A C hm2.t /i n
Example 3.1: Single Sinusoid AM

An AM signal of the form xc .t / D Ac 1 C a cos.2 fmt C =3/ cos.2 fc 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 nd the efciency The total power is 1000 D
2 hxc .t /i
A2 a 2 A2 c c D C h m2 n .t /i 2 2
3-13
It should be clear that in this problem mn.t / D cos.2 fmt /, so hm2 n .t /i D 1=2 and 1000 D A2 c Thus we see that A2 c D 1000 and 1515 1 D D 757:6 W Pcarrier D A2 2 c 2 and thus Psidebands D 1000 The efciency is Eff D 242:4 D 0:242 or 24.2% 1000 Pc D 242:4 W 50 D 1515:15 33 1 1 33 C 0:64 D A2 2 4 50 c
The magnitude and phase spectra can be plotted by rst expanding out xc .t / xc .t / D Ac cos.2 fc t / C aAc cos.2 fmt C =3/ cos.2 fc t / D Ac cos.2 fc t / aAc cos2 .fc C fm/t C =3 C 2 aAc C cos2 .fc fm/t =3 2
|Xc(f)| Ac/2 0.8Ac/4 -fc 0 Xc(f) 0 -/3 /3 f fc-fm fc fc+fm f
Amplitude and phase spectra for one tone AM
Example 3.2: Pulse Train with DC Offset

m(t) 2
t -1 Tm/3 Tm
Find mn.t / and the efciency E From the denition of mn.t / mn.t / D The efciency is a2hm2 n .t /i ED 1 C a 2 hm 2 n .t /i
m.t / m.t / D D m.t / j min m.t /j j 1j
To obtain hm2 n .t /i we form the time average # "Z Z Tm Tm =3 1 hm2 .t / i D .2/2 dt C . 1/2 dt n Tm 0 Tm =3 D thus 2Tm 4 2 7 1 Tm 4C 1 D C D Tm 3 3 3 3 3
.7=3/a2 7a2 ED D 1 C .7=3/a2 3 C 7a2
The best AM efciency we can achieve with this waveform is when a D 1 7 D 0:7 or 70% Eff D a D1 10 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 2 1 1 . 1/2 C .1=2/2 D 3 3 2
hm 2 n .t /i D
The efciency in this case is Eff .1=2/a2 a2 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 efciency maximum is just 50%
3-16
Example 3.3: Multiple Sinusoids

Suppose that m.t / is a sum of multiple sinusoids (multi-tone AM) M X m.t / D Ak cos.2 fk t C k /
k D1
where M is the number of sinusoids, fk values might be constrained over some band of frequencies W , e.g., fk W , and the phase values k can be any value on 0; 2 To nd mn.t / we need to nd min m.t / PM A lower bound on min m.t / is k D1 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

Current plot held >> plot(t,1 + 0.25*m/abs(min(m)),'r') >> subplot(312) >> plot(t,(1 + 0.5*m/abs(min(m))).*cos(2*pi*1000*t)) >> hold Current plot held >> plot(t,1 + 0.5*m/abs(min(m)),'r') >> subplot(313) >> plot(t,(1 + 1.0*m/abs(min(m))).*cos(2*pi*1000*t)) >> hold Current plot held >> plot(t,1 + 1.0*m/abs(min(m)),'r')
8 6 4 m(t) Amplitude 2 0 2 4 6
min m(t)
8 0 0.005 0.01 0.015 0.02 0.025 0.03 Time (s) 0.035 0.04 0.045 0.05
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
2 0 2 2 0 2
0.005
0.01
0.015
0.02
0.025
0.03
0.035
0.04
0.045
0.05
xc(t), a = 0.5
0.005
0.01
0.015
0.02
0.025
0.03
0.035
0.04
0.045
0.05
xc(t), a = 1.0
2 0 2 0 0.005 0.01 0.015 0.02 0.025 0.03 Time (s) 0.035 0.04 0.045 0.05
Modulation index comparison (fc D 1000 Hz)
To obtain the efciency of multi-tone AM we rst calculate hm2 n .t /i assuming unique frequencies hm2 n .t /i
M X k D1 2
A2 k 2j min m.t /j2
2 C 3:52 C 4:22 D D 0:3227 2 7:2462 The maximum efciency is just Eff D 0:3227 D 0:244 or 24.4% 1 C 0:3227
3-19
a D1
A remaining interest is the spectrum of xc .t / Xc .f / D Ac .f fc / C .f C fc / 2 M aAc X h j k C Ak e .f .fc C fk // 4 k D1 i j k Ce .f C .fc C fk // (USB terms)

M aAc X h j k Ak e .f C 4 k D1
.fc
fk // i
Ce
j k .f
C .fc
fk // (LSB terms)
"'( "'%( 789:;<.=,2>9,0<+?23@A03B6@6 "'% "')( "') "'&( "'& "'!( "'! "'"( " !""" #"" $"" %""
6)&&'$&/7'%8/ "(/2/9/
!"##$%&'()* !'+$,)-+./01&/ !/2/345
&"" " &"" *+,-.,/0123456
%""
$""
#"" !"""
Amplitude spectra
3-20
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 reconstructed 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 required for DSB, centered on fc is 2W Since SSB operates by transmitting just one sideband, the transmission bandwidth is reduced to just W
M(f) XDSB(f)
f W XSSB(f) LSB USB removed fc - W fc LSB removed fc XSSB(f) fc - W fc
f fc+W USB f fc+W
DSB to two forms of SSB: USSB and LSSB
The ltering required to obtain an SSB is best explained with the aid of the Hilbert transform, so we divert from text Chapter
3 back to Chapter 2 to briey study the basic properties of this transform Hilbert Transform The Hilbert transform is nothing more than a lter that shifts the phase of all frequency components by =2, i.e., H.f / D where j sgn.f /
8 f >0 <1; sgn.f / D 0; f D0 : 1; f < 0
The Hilbert transform of signal x.t / can be written in terms of the Fourier transform and inverse Fourier transform x.t O /DF 1 D h.t / where h.t / D F 1fH.f /g We can nd the impulse response h.t / using the duality theorem and the differentiation theorem d H.f / df where here H.f / D
F
j sgn.f /X.f / x.t /
! . j 2 t /h. t /
j sgn.f /, so 2j .f /
d H.f / D df
3-22
Clearly, F 1f 2j .f /g D so h.t / D and 1 t

F
2j
1 2j D j2 t t ! j sgn.f /
In the time domain the Hilbert transform is the convolution integral Z 1 Z 1 x. / x.t / x.t O /D d D d .t / 1 1 Note that since the Hilbert transform of x.t / is a shift, the Hilbert transform of x.t O / is O x.t O /D x.t / 1 =2 phase
why? observe that . j sgn.f //2 D
Example 3.4: x.t / D cos !0t

By denition O .f / D X D j sgn.f / 1 j .f 2 1 .f 2 f0/ C .f C f0/
1 fc / C j .f C f0/ 2
3-23
so from e j!0t ) .f x.t O /D
f0 /
1 1 j e j!0t C j e j!0t 2 2 j!0 t j!0 t e e D D sin !0t 2j cos !0t D sin !0t
or
It also follows that
sin !0t D cos !0t D O since x.t O /D x.t /
2 2
cos !0t
Hilbert Transform Properties 1. The energy (power) in x.t / and x.t O / are equal The proof follows from the fact that jY .f /j2 D jH.f /j2jX.f /j2 and jj sgn.f /j2 D 1 2. x.t / and x.t O / are orthogonal, that is Z 1 x.t /x.t O / dt D 0 (energy signal) Z1 T 1 lim x.t /x.t O / dt D 0 (power signal) T !1 2T T
The proof follows for the case of energy signals by generalizing Parsevals theorem Z 1 Z 1 O .f / df x.t /x.t O / dt D X.f /X 1 1 Z 1 D .j sgn.f // jX.f /j2 df D 0 1
odd even
3. Given signals m.t / and c.t / such that the corresponding spectra 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) 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 m.t / cos !0t D m.t /cos !0t D m.t / sin !0t
Analytic Signals Dene analytic signal z.t / as z.t / D x.t / C j x.t O / where x.t / is a real signal
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 characteristics
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 /
j x.t O /
Zn.f / D X.f / 1 sgn.f / ( 0; f >0 D 2X.f /; f < 0

X(f)
-W Zp(f) 2
-W Zn(f) 2
-W
The spectra of analytic signals can suppress positive or negative frequencies
Return to SSB Development

xDSB(t) m(t) Sideband Filter xSSB(t) LSB or USB
Accosct
Basic SSB signal generation
In simple terms, we create an SSB signal from a DSB signal using a sideband lter The mathematical representation of LSSB and USSB signals makes use of Hilbert transform concepts and analytic signals
DSB Signal Starting Point
-fc Formation of HL(f)
+1/2
sgn(f + fc)/2
fc
f -1/2 +1/2 -sgn(f - fc)/2
f -1/2 1 HL(f) = [sgn(f + fc) - sgn(f - fc)]/2
f -fc fc
An ideal LSSB lter
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 lter HL.f / 1 XcLSSB .f / D Ac M.f C fc / C M.f 2 1 sgn.f C fc / sgn.f 2
3-28
fc / fc /
1 XcLSSB .f / D Ac M.f C fc /sgn.f C fc / 4 C M.f fc /sgn.f fc / 1 Ac M.f C fc /sgn.f fc / 4 C M.f fc /sgn.f C fc / 1 D Ac M.f C fc / C M.f fc / 4 1 C Ac M.f C fc /sgn.f C fc / 4 M.f fc /sgn.f fc / The inverse Fourier transform of the rst term is DSB, i.e., 1 Ac m.t / cos !c t 2
F
1 ! Ac M.f C fc / C M.f 4
fc /
The second term can be inverse transformed using m.t O / so F

1 F
j sgn.f / M.f /
j!c t
M.f C fc /sgn.f C fc / D j m.t O /e

F
since m.t /e j!c t Thus
! M.f fc /
1 Ac F 1 M.f C fc /sgn.f C fc / M.f fc /sgn.f fc / 4 1 1 D Ac j m.t O /e j!c t j m.t O /e j!c t D m.t O / sin !c t 4 2
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 xcUSSB .t / D Ac m.t / cos !c t 2 1 Ac m.t O / sin !c t 2
The direct implementation of SSB is very difcult due to the requirements of the lter 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 modulation must not contain critical information at these frequencies
m(t) H(f) = -jsgn(f)
cosct
+
0
o
sinct
-90o
Carrier Osc. cosct
xc(t)
+ -
LSB USB
Phase shift modulator for SSB
3-30
Demodulation The coherent demodulator rst discussed for DSB, also works for SSB
d(t) xr(t) LPF 1/Ac scale factor included yD(t)
4cos[2fct + (t)]
Coherent demod for SSB
Carrying out the analysis to d.t /, rst we have 1 d.t / D Ac m.t / cos !c t m.t O / sin !c t 4 cos.!c t C .t // 2 D Ac m.t / cos .t / C Ac m.t / cos2!c t C .t / Ac m.t O / sin .t / Ac m.t O / sin2!c t C .t / so yD .t / D m.t / cos .t /
.t/ small
m.t O / sin .t /
'
m.t /
m.t O / .t /
The m.t O / sin .t / term represents crosstalk Another approach to demodulation is to use carrier reinsertion and envelope detection
xr(t) e(t) Envelope Detector yD(t)
Kcosct
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 nd the envelope of e.t /, which will be the nal output yD .t / Finding the envelope is a more general problem which will be useful in future problem solving, so rst consider the envelope of x.t / D a.t / cos !c t b.t / sin !c t inphase quadrature D Re a.t /e j!c t C jb.t /e j!c t D Re a.t / C jb.t / e j!c t
Q Dcomplex envelope R.t/
In a phasor diagram x.t / consists of an inphase or direct component and a quadrature component
Quadrature - Q Note: R(t) = R(t) (t) a(t)
3-32
R(t)ej(t) = a(t) + jb(t)
b(t)
In-phase - I
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 D R.t / cos !c t C .t / where .t / D tan 1b.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 1 yD .t / D Ac m.t / C K 2 sin .t / sin !c t
1 C Ac m.t O / 2
If we choose K such that .Ac m.t /=2 C K/2 then 1 yD .t / ' Ac m.t / C K 2 Note:
.Ac m.t O /=2/2,
The above analysis assumed a phase coherent reference In speech systems the frequency and phase can be adjusted to obtain intelligibility, but not so in data systems
The approximation relies on the binomial expansion 1 .1 C x/1=2 ' 1 C x for jx j 2 1
Example 3.6: Noncoherent Carrier Reinsertion

Let m.t / D cos !mt , !m K cos.!c C !/t !c and the reinserted carrier be
Following carrier reinsertion we have 1 e.t / D Ac cos !mt cos !c t 2 1 Ac sin !c t 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 o n 1 j!m t j!c t e.t / D Ac Re e e 2 o n j.!c C!/t C K Re 1 e n 1 o j.!m !/t j.!c C!/t D Re Ac e CK e 2
Q complex envelope R.t/ 3-34 ECE 5625 Communication Systems I
Finally expanding the complex envelope into the real and imaginary parts we can nd the real envelope R.t / o2 hn 1 Ac cos!m C !/t C K yD .t / D 2 n1 o2i1=2 C Ac sin.!m C !/t 2 1 ' Ac cos.!m !/t C K 2 where the last line follows for K Ac
Note that the frequency error ! causes the recovered message 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 ltering DSB such that one sideband is passed completely while only a vestige remains of the other Why VSB? 1. Simplies the lter design 2. Improves the low-frequency response and allows DC to pass undistorted 3. Has bandwidth efciency advantages over DSB or AM, similar to that of SSB
A primary application of VSB is the video portion of analog television (note HDTV has replaced this in the US) The generation of VSB starts with DSB followed by a lter that has a 2 transition band, e.g., 8 f < Fc <0; c / jH.f /j D f .f ; fc f fc C 2 :1; f >f C
c
1 |H(f)| f
fc -
fc
fc +
Ideal VSB transmitter lter amplitude response
VSB can be demodulated using a coherent demod or using carrier reinsertion and envelope detection
Transmitted Two-Tone Spectrum (only single-sided shown) A(1 - )/2 B/2
A/2 0 f f - f2 f - f1 fc f + f1 f + f2
Two-tone VSB signal

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 lter has made this possible In the case of broadcast TV the carrier in included at the transmitter to insure phase coherency and easy demodulation at the TV receiver (VSB + Carrier) Very large video carrier power is required for typical TV station, i.e., greater than 100,000 W To make matters easier still, the precise VSB ltering is not performed at the transmitter due to the high power requirements, instead the TV receiver does this
Transmitter Output
Video Carrier Audio Carrier
0 -1.75 -0.75 2 interval Receiver Shaping Filter -0.75 0 1
4.0 4.5 4.75
(f - fcv) MHz
0.75
4.0
4.75
(f - fcv) MHz
Broadcast TV transmitter spectrum and receiver shaping lter
3.1.5
Frequency Translation and Mixing
Used to translate baseband or bandpass signals to some new center frequency

m(t)cos1t e(t) f f1 Local oscillator of the form BPF at f2 f2
Frequency translation system
Assuming the input signal is DSB of bandwidth 2W the mixer (multiplier) output is
local osc (LO)
e.t / D m.t / cos.!1t / 2 cos.!1 !2/t D m.t / cos.!2t / C m.t / cos.2!1 !2/t
The bandpass lter bandwidth needs to be at least 2W Hz wide Note that if an input of the form k.t / cos.!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 / cos.2!1 3!2/t ; and the bandpass lter 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 RF-Amp
IF Filt/ Amp
fIF
Env Det
Audio Amp
Local Osc. Joint tuning
Automatic gain control
For AM BT = 2W
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
AM Superheterodyne receiver
We have two choices for the local oscillator, high-side or lowside tuning
Low-side: 540 455 fLO 1145, all frequencies in kHz
1600 455 or 85
fLO
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 oscillator 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 Input 455 fLO Mixer Output Image Out of mixer BIF 560 fIF 1015 (560+455) fIF 1470 f (kHz) BRF 1470 f (kHz) Potential Image
IF BPF
1015-560 This is removed with RF BPF 455 0 1470-1015
1575 (560+1015)
f (kHz)
f (kHz) 2485 (1470+1015)
Receiver frequency plan including images
3-40
Example 3.8: A Double-Conversion Receiver

fc = 162.475 MHz (WX #4) Tunable RF-Amp 1st LO fLO1 = 173.175 MHz 10.7 MHz & 335.65 MHz 10.7 MHz IF BPF 2nd LO fLO2 = 11.155 MHz 455 kHz & 21.855 MHz 455 kHz IF BPF FM Demod
Double-conversion superheterodyne receiver
Consider a frequency modulation (FM) receiver that uses doubleconversion to receive a signal con carrier frequency 162.475 MHz (weather channel #4) Frequency modulation will be discussed in the next section The dual-conversion allows good image rejection by using a 10.7 MHz rst 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 lter The second IF lter can thus have a much narrower bandwidth by virtue of the center frequency being much lower A higher rst IF center frequency moves the image signal further away from the desired signal
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 appropriate, as an ideal multiplier is rarely available Instead active and passive circuits that approximate signal multiplication are utilized The notion of mixing comes about from passing the sum of two signals through a nonlinearity, e.g., y.t / D a1x1.t / C a2x2.t /2 C other terms 2 2 2 2 D a1 x1 .t / C 2a1a2x1.t /x2.t / C a2 x2 .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 (rst 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 Semiconductor FET (MESFET) mixer shown below
Nonlinear Device VRF VIN VLO zL VOUT
Mixer concept
+5V R2 C3 47pF L1
5 turns, 28 AWG .050 I.D.
10 C4 0.01uF L3 270nH L4 270nH C8 82pF IF
LO RF
C1 0.5pF
G1 G2
C7 D Q1 NE25139 42pF S R1 C5 47pF
C2 0.5pF L2
5 turns, 28 AWG .050 I.D.
C8 0.01uF
270
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
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 phaselocked loop applications
mixer LO source RG LO input vp( t)
D3 D4 D2 D1
RF input vi ( t)
RF source RG
IF out
vo( t) IF load RL
Passive Double-Balanced Mixer (DBM)

C9 L1 5V C8 L2 C11 R4 R3 3 2 1 T1 6
IF OUT
4:1 (200:50) TRANSFORMER 4
C10 GND IF+ GND 17 IFGND
19
20
C1 RFIN C3 C2
18
RF TAP GND RFBIAS
MAX9982
1 2 3 4 5 15 14 13 12 11
16
LO2 GND GND GND LO1 C6 C7
LO2
R1
GND
LO1
LOSEL
GND
GND
VCC
5V C4 LO SELECT
VCC
10 5V C5
825 MHz to 915 MHz SiGe High-Linearity Active DBM
3-44
Example 3.9: Single Diode Mixer
3-45
3.2
Angle Modulation
A general angle modulated signal is of the form xc .t / D Ac cos!c t C .t / Denition: Instantaneous phase of xc .t / is
i .t /
D !c t C .t /
where .t / is the phase deviation Dene: Instantaneous frequency of xc .t / is !i .t / D d .t / d i .t / D !c C dt dt
where d .t /=dt is the frequency deviation There are two basic types of angle modulation 1. Phase modulation (PM) .t / D kp m.t /
phase dev. const.
which implies xc .t / D Ac cos!c t C kp m.t / Note: the units of kp is radians per unit of m.t / If m.t / is a voltage, kp has units of radians/volt
3.2. ANGLE MODULATION
2. Frequency modulation (FM) d .t / D dt or Z .t / D kf

t0
freq. dev. const. t
kf
m.t /
m./ d C
Note: the units of kf is radians/sec per unit of m.t / If m.t / is a voltage, kf has units of radians/sec/volt An alternative expression for kf is kf D 2 fd where fd is the frequency-deviation constant in Hz/unit of m.t /
Example 3.10: Phase and Frequency Step Modulation

Consider m.t / D u.t / v We form the PM signal xPM.t / D Ac cos !c t C kp u.t / ; kp D =3 rad/v We form the FM signal Z t h i xFM.t / D Ac cos !c t C 2 fd m./ d ; fd D 3 Hz/v
/3 phase step at t = 0
fc 1 fc
3 Hz frequency step at t = 0
fc 1 fc + 3 Hz
Phase Modulation
Frequency Modulation
Phase and frequency step modulation
3.2.1
Narrowband Angle Modulation
Begin by writing an angle modulated signal in complex form xc .t / D Re Ac e j!c t e j Expand e j

.t/ .t/
in a power series
2 j!c t
xc .t / D Re Ac e
1 C j .t /
.t / 2 1, then
The narrowband approximation is j .t /j
xc .t / ' Re Ac e j!c t C jAc .t /e j!c t D Ac cos.!c t /

3-48
Ac .t / sin.!c t /
Under the narrowband approximation we see that the signal is similar to AM except it is carrier plus modulated quadrature carrier
(t) + Ac sin(ct) 90o NBFM xc(t)
NBFM modulator block diagram
Example 3.11: Single tone narrowband FM

Consider NBFM with m.t / D cos !mt Z t .t / D 2 fd cos !mt d D Now, xc .t / D cos !c t C fd sin !mt fm fd ' Ac cos !c t sin !mt sin !c t fm fd D Ac cos !c t C sin.fc C fm/t 2fm 2 fd fd sin !mt D sin !mt 2 fm fm
fd sin.fc 2fm
fm/t
This looks very much like AM

fc - fm 0 fc fc + fm
Single tone NBFM spectra
3.2.2
Spectrum of an Angle-Modulated Signal
The development in this obtains the exact spectrum of an angle modulated carrier for the case of .t / D sin !mt where is the modulation index for sinusoidal angle modulation The transmitted signal is of the form xc .t / D Ac cos !c t C sin !mt D Ac Re e j!c t e j sin !mt Note that e j sin !mt is periodic with period T D 2 =!m, thus we can obtain a Fourier series expansion of this signal, i.e., e
j sin !m t
1 X nD 1
Yne j n!mt
3-50
The coefcients are Z !m Yn D 2 Z !m D 2

=!m =!m =!m
e j sin !mt e e
j n!m t
dt dt
j.n!m t sin !m t/
=!m
Change variables in the integral by letting x D !mt , then dx D !mdt , t D =!m ! x D , and t D =!m ! x D With the above substitutions, we have Z 1 e j.nx sin x/ dx Yn D 2 Z 1 D cos.nx sin x/ dx D Jn./
0
which is a Bessel function of the rst kind order n with argument Jn./ Properties Recurrence equation: JnC1./ D n even: J n./ D Jn./ n odd: J n./ D
2n Jn./
Jn 1./
Jn./
3-51
1 0.8 0.6 0.4 0.2
J0() J1() J2()
J3()
0.2 0.4 2 4 6 8 10
Bessel function of order 03 plotted
The zeros of the Bessel functions are important in spectral analysis

First ve Bessel function zeros for order 0 5
J0() = 0
2.40483, 5.52008, 8.65373, 11.7915, 14.9309
J1() = 0
3.83171, 7.01559, 10.1735, 13.3237, 16.4706
J2() = 0
5.13562, 8.41724, 11.6198, 14.796, 17.9598
J3() = 0
6.38016, 9.76102, 13.0152, 16.2235, 19.4094
J4() = 0
7.58834, 11.0647, 14.3725, 17.616, 20.8269
J5() = 0
8.77148, 12.3386, 15.7002, 18.9801, 22.2178
3-52
Spectrum cont. We obtain the spectrum of xc .t / by inserting the series representation for e j sin !mt " # 1 X xc .t / D Ac Re e j!c t Jn./e j n!mt
nD 1
D Ac
1 X nD 1
Jn./ cos.!c C n!m/t
We see that the amplitude spectrum is symmetrical about fc due to the symmetry properties of the Bessel functions
Amplitude Spectrum (one-sided) |AcJ-1()| |AcJ-2()| |AcJ-3()| |A J ()| |AcJ-5()| c -4 fc - 5fm fc - 4fm fc - 3fm fc - 2fm fc - fm fc + fm fc + 2fm fc |AcJ1()| |AcJ2()| |AcJ3()| |AcJ4()| |AcJ5()| fc + 3fm fc + 4fm fc + 5fm f
|AcJ0()|
For PM sin !mt D kp .A sin !mt /

m.t/
) D kp A For FM Z sin !mt D kf ) D .fd =fm/A

ECE 5625 Communication Systems I 3-53 t
A cos !m d D
fd A sin !mt fm
When is small we have the narrowband case and as gets larger the spectrum spreads over wider bandwidth
1
Amplitude Spectrum
0.8 0.6 0.4 0.2 -5 1 0
= 0.2, Ac = 1 (narrowband case)
10
(f - fc)/fm
Amplitude Spectrum
0.8 0.6 0.4 0.2 -5 1 0
= 1, Ac = 1
10
(f - fc)/fm
Amplitude Spectrum
0.8 0.6 0.4 0.2 -5 1 0
= 2.4048, Ac = 1 (carrier null)
10
(f - fc)/fm
Amplitude Spectrum
0.8 0.6 0.4 0.2 -5 1 0
= 3.8317, Ac = 1 (1st sideband null)
10
(f - fc)/fm
Amplitude Spectrum
0.8 0.6 0.4 0.2 -5 0
= 8, Ac = 1 (spectrum becoming wide) (f - fc)/fm
10
The amplitude spectrum relative to fc as increases
3-54
Example 3.12: VCO FM Modulator

Consider again single-tone FM, that is m.t / D A cos.2 fmt / We assume that we know fm and the modulator deviation constant fd Find A such that the spectrum of xc .t / contains no carrier component An FM modulator can be implemented using a voltage controlled oscillator (VCO)
sensitivity fd MHz/v m(t) VCO Center Freq = fc
A VCO used as an FM modulator
The carrier term is Ac J0./ cos !c t We know that J0./ D 0 for D 2:4048; 5:5201; : : : The smallest that will make the carrier component zero is D 2:4048 D which implies that we need to set A D 2:4048
fd A fm
fm fd
3-55
Suppose that fm D 1 kHz and fd D 2:5 MHz/v, then we would need to set 1 103 D 9:6192 A D 2:4048 2:5 106 10
4
3.2.3
Power in an Angle-Modulated Signal

2 2 .t /i D A2 hxc c hcos !c t C .t / i 1 2 1 C h cos 2 !c t C .t / i A D A2 2 c 2 c
The average power in an angle modulated signal is
For large fc the second term is approximately zero (why?), thus 1 2 Pangle mod D hxc .t /i D A2 2 c which makes the power independent of the modulation m.t / (the assumptions must remain valid however)
3.2.4
Bandwidth of Angle-Modulated Signals
With sinusoidal angle modulation we know that the occupied bandwidth gets larger as increases There are an innite number of sidebands, but
n!1
lim Jn./
n lim D 0; n!1 2n n
so consider the fractional power bandwidth

Dene the power ratio

1 2 A Pcarrier C Pk sidebands 2 c Pr D D Ptotal
Pk
nD k 1 2 A 2 c
2 Jn ./
J02./
C2
k X nD1
2 Jn ./
Given an acceptable Pr implies a fractional bandwidth of B D 2kfm (Hz) In the text values of Pr 0:7 and Pr double underlined respectively It turns out that for Pr 0:98 are single and
0:98 the value of k is IP1 C , thus
B D B98 ' 2. C 1/fm sinusoidal mod only For arbitrary modulation m.t /, dene the deviation ratio DD peak freq. deviation fd max jm.t /j D bandwidth of m.t / W
In the sinusoidal modulation bandwidth denition let ! D and fm ! W , then we obtain what is known as Carsons rule B D 2.D C 1/W Another view of Carsons rule is to consider the maximum frequency deviation f D max jm.t /jfd , then B D 2.W C f /
Two extremes in angle modulation are 1. Narrowband: D 2. Wideband: D 1 ) B D 2W 1 ) B D 2DW D 2f
Example 3.13: Single Tone FM

Consider an FM modulator for broadcasting with xc .t / D 100 cos 2 .101:1 where fd D 75 kHz/v and m.t / D cos 2 .1000/t v The value for the transmitter is fd 75 103 D AD D 75 fm 103 Note that the carrier frquency is 101.1 MHz and the peak deviation is f D 75 kHz The bandwidth of the signal is thus B ' 2.1 C 75/1000 D 152 kHz
17.5 15 12.5 10 7.5 5 2.5
106/t C .t /
B = 2( + 1)fm
Amplitude Spectrum
-76
-50
50
76
(f - 101.1 MHz) 1 kHz 100
101.1 MHz - 76 kHz

3-58
101.1 MHz
101.1 MHz + 76 kHz

Suppose that this signal is passed through an ideal bandpass lter of bandwidth 11 kHz centered on fc D 101:1 MHz, i.e., H.f / D f fc 11000 C f C fc 11000
The carrier term and ve sidebands either side of the carrier pass through this lter, resulting an output power of " # 5 2 X A 2 Pout D c J02.75/ C 2 Jn .75/ D 241:93 W 2 nD 1 Note the input power is A2 c =2 D 5000 W
Example 3.14: Two Tone FM

Finding the exact spectrum of an angle modulated carrier is not always possible The single-tone sinusoid case can be extended to multiple tone with increasing complexity Suppose that m.t / D A cos !1t C B cos !2t The phase deviation is given by 2 fd times the integral of the frequency modulation, i.e., .t / D 1 sin !1t C 2 sin !2t where 1 D Afd =f1 and 2 D Afd =f2
The transmitted signal is of the form xc .t / D Ac cos !c t C 1 sin !1t C 2 sin !2t D Ac Re e j!c t e j1 sin !1t e j2 sin 2t We have previously seen that via Fourier series expansion e j1 sin !1t D e j2 sin !1t D
1 X nD 1 1 X nD 1
Jn.1/e j n!1t Jn.2/e j n!2t
Inserting the above Fourier series expansions into xc .t /, we have ( ) 1 m X X xc .t / D Ac Re e j!c t Jn.1/e j n!1t Jm.2/e j m!2t
nD 1 mD 1
D Ac
1 X
1 X
Jn.1/Jm.2/ cos .!c C n!1 C m!2/t
nD 1 mD 1
The nonlinear nature of angle modulation is clear, since we see not only components at !c C n!1 and !c C m!2, but also at all combinations of !c C n!1 C m!2 To nd the bandwidth of this signal we can use Carsons rule (the sinusoidal formula only works for one tone) Recall that B D 2.f C W /, where f is the peak frequency deviation
The frequency deviation is fi .t / D 1 d 1 sin !1t C 2 sin !2t 2 dt D 1f1 cos.2 f1t / C 2f2 cos.2 f2t / Hz
The maximum of fi .t /, in this case, is 1f1 C 2f2 Suppose 1 D 2 D 2 and f2 D 10f1, then we see that W D f2 D 10f1 and B D 2.W Cf / D 2 10f1C2.f1C10f1/ D 2.32f1/ D 64f1
1 = 2 = 2, f2 = 10f 1 B = 2(W + f) = 2(10f1 + 2(11)f1) = 64f1
0.35 0.3 0.25 0.2 0.15 0.1 0.05
Amplitude Spectrum
(f - fc)
-40 -20 0 20 40
f1
Example 3.15: Bandlimited Noise PM and FM

This example will utilize simulation to obtain the spectrum of an angle modulated carrier The message signal in this case will be bandlimited noise having lowpass bandwidth of W Hz
In MATLAB we can generate Gaussian amplitude distributed white noise using randn() and then lter this noise using a high-order lowpass lter (implemented as a digital lter in this case) We can then use this signal to phase or frequency modulate a carrier in terms of the peak phase deviation, derived from knowledge of maxj .t /j
3-62
3.2.5
Narrowband-to-Wideband Conversion
Narrowband FM Carrier = fc1 Peak deviation = fd1 Deviation ratio = D1 Wideband FM Carrier = nfc1 Peak deviation = nfd1 Deviation ratio = nD1 xn Freq. Multiplier LO xc(t) BPF
m(t)
Narrowband FM Modulator (similar to AM)
Ac1cos[ct + (t)] Ac2cos[nct + n(t)]
Frequency translate
narrowband-to-wideband conversion
Narrowband FM can be generated using an AM-type modulator as discussed earlier (a VCO is not required, so the carrier source can be very stable) A frequency multiplier, using say a nonlinearity, can be used to make the signal wideband FM, i.e., Ac1 cos!c t C .t / ! Ac2 cosn!c t C n .t / so the modulator deviation constant of fd1 becomes nfd1
n
3.2.6
Demodulation of Angle-Modulated Signals
To demodulate FM we require a discriminator circuit, which gives an output which is proportional to the input frequency deviation For an ideal discriminator with input xr .t / D Ac cos!c t C .t /
the output is yD .t / D
xc(t)
1 d .t / KD 2 dt
yD(t)
Ideal Discriminator
Ideal FM discriminator
For FM .t / D 2 fd so
m./ d
yD .t / D KD fd m.t /
Output Signal (voltage) slope = KD
fc
Input Frequency
Ideal discriminator I/O characteristic
For PM signals we follow the discriminator with an integrator

xr(t) Ideal Discrim. yD(t)
Ideal discriminator with integrator for PM demod

For PM .t / D kp m.t / so yD .t / D KD kp m.t / We now consider approximating an ideal discriminator with:

e(t) Envelope Detector
xr(t)
yD(t)
Ideal discriminator approximation
If xr .t / D Ac cos!c t C .t / e.t / D dxr .t / D dt Ac !c C d dt sin !c t C .t /
This looks like AM provided d .t / < !c dt which is only reasonable Thus yD .t / D Ac d .t / D 2 Ac fd m.t / (for FM) dt
Relative to an ideal discriminator, the gain constant is K D D 2 Ac To eliminate any amplitude variations on Ac pass xc .t / through a bandpass limiter
Limiter xr(t) BPF Bandpass Limiter
e(t)
Envelope Detector
yD(t)
FM discriminator with bandpass limiter
We can approximate the differentiator with a delay and subtract operation e.t / D xr .t / xr .t / since lim thus e.t / ' e.t /
!0
D lim
xr .t /
xr .t
!0
dxr .t / ; dt
dxr .t / dt
In a discrete-time implementation (DSP), we can perform a similar operation, e.g. en D xn xn 1
Example 3.16: Complex Baseband Discriminator

A DSP implementation in MATLAB that works with complex baseband signals (complex envelope) is the following:
function disdata = discrim(x) % function disdata = discrimf(x) % x is the received signal in complex baseband form % % Mark Wickert
xI=real(x); % xI is the real part of the received signal xQ=imag(x); % xQ is the imaginary part of the received signal N=length(x); % N is the length of xI and xQ b=[1 -1]; % filter coefficients a=[1 0]; % for discrete derivative der_xI=filter(b,a,xI); % derivative of xI, der_xQ=filter(b,a,xQ); % derivative of xQ % normalize by the squared envelope acts as a limiter disdata=(xI.*der_xQ-xQ.*der_xI)./(xI.^2+xQ.^2);
To understand the operation of discrim() start with a general angle modulated signal and obtain the complex envelope xc .t / D Ac cos.!c t C .t // D Re Ac e j .t/e j!c t D Ac Re cos .t / C j sin .t /e j!c t The complex envelope is x Q c .t / D cos .t / C j sin .t / D xI .t / C jxQ .t / where xI and xQ are the in-phase and quadrature signals respectively A frequency discriminator obtains d .t /=dt In terms of the I and Q signals, .t / D tan
1
xQ .t / xI .t /
The derivative of .t/ is 1 d xQ .t / d .t / D dt 1 C .xQ .t /=xI .t //2 dt xI .t / 0 0 xI .t /xQ .t / xI .t /xQ .t / D 2 2 xI .t / C xQ .t /

In the DSP implementation xI n D xI .nT / and xQ n D xQ .nT /, where T is the sample period
0 0 The derivatives, xI .t / and xQ .t / are approximated by the backwards difference xI n xI n 1 and xQ n xQ n 1 respectively
To put this code into action, consider a single tone message at 1 kHz with D 2:4048 .t / D 2:4048 cos.2 .1000/t / The complex baseband (envelope) signal is x Q c .t / D e j
.t/
D e j 2:4048 cos.2
.1000/t/
A MATLAB simulation that utilizes the function Discrim() is:

>> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >>
3-68
n = 0:5000-1; m = cos(2*pi*n*1000/50000); % sampling rate = 50 kHz xc = exp(j*2.4048*m); y = Discrim(xc); % baseband spectrum plotting tool using psd() bb_spec_plot(xc,2^11,50); axis([-10 10 -30 30]) grid xlabel('Frequency (kHz)') ylabel('Spectral Density (dB)') t = n/50; plot(t(1:200),y(1:200)) axis([0 4 -.4 .4]) grid xlabel('Time (ms)') ylabel('Amplitude of y(t)')
30
20
Note: no carrier term since = 2.4048
Spectral Density (dB)
10
10
20
30 10
2 0 2 Frequency (kHz)
10
0.4 0.3 0.2 Amplitude of y(t) 0.1 0 0.1 0.2 0.3 0.4
0.5
1.5
2 Time (ms)
2.5
3.5
Baseband FM spectrum and demodulator output wavefrom
3-69
Analog Circuit Implementations A simple analog circuit implementation is an RC highpass lter followed by an envelope detector
|H(f)| 1 C 0.707 R Linear operating region converts FM to AM f
Highpass fc 1 2RC
Re Ce
Highpass
Envelope Detector
RC highpass lter + envelope detector discriminator (slope detector)
For the RC highpass lter to be practical the cutoff frequency must be reasonable Broadcast FM radio typically uses a 10.7 MHz IF frequency, which means the highpass lter must have cutoff above this frequency A more practical discriminator is the balanced discriminator, which offers a wider linear operating range
Filter Amplitude Response
|H2(f)|
|H1(f)|
f2
f1
Filter Amplitude Response
f
Linear region |H1(f)| - |H2(f)|
R L1 xc(t) L2 R
f1
C1 C2
f2
Re Re
Ce yD(t) Ce
Bandpass Envelope Detectors
Balanced discriminator operation (top) and a passive implementation (bottom)
3-71
FM Quadrature Detectors
xc(t) C1 xquad(t) Lp Cp Usually a xout(t) lowpass filter is added here Tank circuit tuned to fc
Quadrature detector schematic
In analog integrated circuits used for FM radio receivers and the like, an FM demodulator known as a quadrature detector or quadrature discriminator, is quite popular The input FM signal connects to one port of a multiplier (product device) A quadrature signal is formed by passing the input to a capacitor series connected to the other multiplier input and a parallel tank circuit resonant at the input carrier frequency The quadrature circuit receives a phase shift from the capacitor and additional phase shift from the tank circuit The phase shift produced by the tank circuit is time varying in proportion to the input frequency deviation A mathematical model for the circuit begins with the FM input signal xc .t / D Ac !c t C .t /
3.3. INTERFERENCE
The quadrature signal is xquad.t / D K1Ac sin !c t C .t / C K2 .t / dt
where the constants K1 and K2 are determined by circuit parameters The multiplier output, assuming a lowpass lter removes the sum terms, is 1 d .t / xout.t / D K1A2 sin K 2 c 2 dt By proper choice of K2 the argument of the sin function is small, and a small angle approximation yields d .t / 1 1 xout.t / ' K1K2A2 D K1K2A2 c c KD m.t / 2 dt 2
3.3
Interference
Interference is a fact of life in communication systems. A through understanding of interference requires a background in random signals 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
3.3.1
Interference in Linear Modulation

Single-Sided Spectrum Xr(f) 1 A 2 m Ac 1 A 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 receiver 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 lter yD .t / D Am cos !mt C Ai cos !i t
interference
Envelope Detection: Here we need to nd the received envelope relative to the strongest signal present Case Ac Ai
We will expand xr .t / in complex envelope form by rst noting that Ai cos.!c C !i /t D Ai cos !i t cos !c t Ai sin !i t sin !c t
3.3. INTERFERENCE
now, xr .t / D Re Ac C Am cos !mt C Ai cos !i t jAi sin !i t e j!c t Q /e j!c t D Re R.t 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 Finally, yD .t / ' Am cos !mt C Ai cos !i t
interference
Ai
Case Ai >> Ac Now the interfering term looks like the carrier and the remaining terms look like sidebands, LSSB sidebands relative to fc C fi to be specic 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!
3.3.2
Interference in Angle Modulation
Initially assume that the carrier is unmodulated xr .t / D Ac cos !c t C Ai cos.!c C !i /t In complex envelope form we have xr .t / D Re .Ac C Ai cos !i t Q / D Ac C Ai cos !i t with R.t
jAi sin !i t /e j!c t
jAi sin !i t
Q /j, The real envelope or envelope magnitude is, R.t / D jR.t p R.t / D .Ac C Ai cos !i t /2 C .Ai sin !i t /2 and the envelope phase is .t / D tan
1
Ai sin !i t Ac C Ai cos !i t
For future reference note that: tan

1
xDx
x3 x5 C 3 5
x7 C 7
jx j
' x
We can thus write that xr .t / D R.t / cos !c t C .t / If Ac Ai Ai xr .t / ' .Ac C Ai cos !i t / cos !c t C sin !i t Ac R.t/
.t/ 3-76 ECE 5625 Communication Systems I
3.3. INTERFERENCE
Case of PM Demodulator: The discriminator recovers d .t /=dt , so the output is followed by an integrator yD .t / D KD Ai sin !i t Ac
Case of FM Demodulator: The discriminator output is used directly to obtain d .t /=dt yD .t / D 1 Ai d Ai KD sin !i t D KD fi cos !i t 2 Ac dt Ac
We thus see that the interfering tone appears directly in the output for both PM and FM For the case of FM the amplitude of the tone is proportional to the offset frequency fi For fi > W , recall W is the bandwidth of the message m.t /, a lowpass lter following the discriminator will remove the interference When Ai is similar to Ac and larger, the above analysis no longer holds In complex envelope form xr .t / D Re Ac C Ai e j!i t e j!c t The phase of the complex envelope is .t / D Ac C Ai e j!i t D tan
ECE 5625 Communication Systems I 1
Ai sin !i t Ac C Ai cos !i t
3-77
We now consider Ai derivative

(t) 0.1 0.05
Ac and look at plots of .t / and the
d(t)/dt
0.6 0.4 0.2
t 1 Ai = 0.1Ac fi = 1 0.5 0.05 0.1 (t) 1 1 0.5 t 1 Ai = 0.9Ac fi = 1 0.5 0.5 1 (t) 3 2 1 t 1 Ai = 1.1Ac fi = 1 0.5 1 2 3 1 0.5 0.5 1 d(t)/dt 0.5 1 d(t)/dt 0.5 10 20 30 40 50 70 60 50 40 30 20 10 0.5 1 0.5 1 0.5 1 1 0.5 0.2 0.4 0.6 0.5 1
Phase deviation and discriminator outputs when Ai
Ac
We see that clicks (positive or negative spikes) occur in the discriminator output when the interference levels is near the signal level When Ai Ac the message signal is entirely lost and the discriminator is said to be operating below threshold
3.3. INTERFERENCE
To better see what happens when we approach threshold, apply single tone FM to the carrier .t / D Ac e jAm cos.!mt/ C Ai e j!i t Plot the discriminator output d .t /=dt with Am D 5, fm D 1, fi D 3, and various values of Ai
Ai = 0.005, 30 fi = 3 20 Am = = 5, 10 fm = 1 1 0.5 10 20 30 20 1 Ai = 0.5, fi = 3 Am = = 5, fm = 1 0.5 20 40 60 80 Ai = 0.9, fi = 3 Am = = 5, fm = 1 d(t)/dt 1 0.5 1 t 0.5 100 200 300 400 d(t)/dt Ai = 0.1, 30 fi = 3 20 Am = = 5, 10 fm = 1 t 0.5 1 1 0.5 10 20 30 d(t)/dt 0.5 1 0.5 1 d(t)/dt
Discriminator outputs as Ai approaches Ac with single tone FM D 5
The Use of Preemphasis in FM We have seen that when Ai is small compared to Ac the interference level in the case of FM demodulation is proportional to fi The generalization from a single tone interferer to background noise (text Chapter 6), shows a similar behavior, that is wide
bandwidth noise entering the receiver along with the desired FM signal creates noise in the discriminator output that has amplitude proportional with frequency (noise power proportional to the square of the frequency) In FM radio broadcasting a preemphasis boosts the high frequency content of the message signal to overcome the increased noise background level at higher frequencies, with a deemphasis lter used at the discriminator output to gain equalize/atten the end-to-end transfer function for the modulation m.t /
Discriminator Output with Interference/Noise No preemphasis
With preemphasis
Message Bandwidth 0 f1 W f r FM Mod Discrim Hd(f) |Hd(f)| C
r HP(f) |Hp(f)|
f f1 f2 f1
FM broadcast preemphasis and deemphasis ltering
The time constant for these lters is RC D 75 s (f1 D 1=.2 RC / D 2:1 kHz ), with a high end cutoff of about f2 D 30 kHz
3.4. FEEDBACK DEMODULATORS
3.4
Feedback Demodulators
The discriminator as described earlier rst converts and FM signal to and AM signal and then demodulates the AM The phase-locked loop (PLL) offer a direct way to demodulate FM and is considered a basic building block by communication system engineers
3.4.1
Phase-Locked Loops for FM Demodulation
The PLL has many uses and many different congurations, both analog and DSP based We will start with a basic conguration for demodulation of FM
Kd xr(t) Phase Detector ed(t) Loop Filter Loop Amplifier ev(t)
eo(t)
VCO Sinusoidal phase detector with inverting input
xr(t) -eo(t)
Basic PLL block diagram

Let xr .t / D Ac cos !c t C .t / eo.t / D Av sin !c t C .t / Note: Frequency error may be included in .t / .t /
Assume a sinusoidal phase detector with an inverting operation is included, then we can further write 1 ed .t / D Ac Av Kd sin 2 .t / .t /
In the above we have assumed that the double frequency term is removed (e.g., by the loop lter eventually) Note that for the voltage controlled oscillator (VCO) we have the following relationship
ev(t) VCO Kv o + d dt
but d .t / D Kv ev .t / rad/s dt Z
t
.t / D Kv
ev ./ d
In its present form the PLL is a nonlinear feedback control system

(t)
+ -
(t)
sin( ) Loop nonlinearity

(t)
ed(t)
f(t)
Loop filter impulse response
ev(t)
Nonlinear feedback control model
To shown tracking we rst consider the loop lter to have impulse response .t / (a straight through connection or unity gain amplier) The loop gain is now dened as Kt D The VCO output is Z .t / D Kt or
t
1 Ac Av Kd Kv rad/s 2
sin ./ .t /
./ d
d .t / D Kt sin .t / dt
Let .t / D .t / i.e.,
.t / and apply an input frequency step ! , d .t / D ! u.t / dt d .t / D Kt sin .t /; t dt
Now, d .t / D ! dt
0
3-83
We can now plot d =dt versus plane plot

d(t)/dt + Kt
, which is known as a phase
Stable lock point
> 0
- Kt ss
(t)
Phase plane plot (1st-order PLL)
d .t / C Kt sin .t / D !u.t / dt At t D 0 the operating point is at B Since dt is positive if d >0 !d dt d Since dt is positive if <0 !d dt is positive is negative
therefore the steady-state operating point is at A The frequency error is always zero in steady-state The steady-state phase error is
ss
Note that for locking to take place, the phase plane curve must cross the d =dt D 0 axis
The maximum steady-state value of ! the loop can handle is thus Kt The total lock range is then !c Kt ! !c C Kt ) 2Kt For a rst-order loop the lock range and the hold-range are identical For a given ! the value of ing the loop gain, i.e.,
ss ss
can be made small by increas1
D sin
! Kt
Thus for large Kt the in-lock operation of the loop can be modeled with a fully linear model since .t / .t / is small, i.e., sin .t / .t / ' .t / .t /
The s -domain linear PLL model is the following

(s) + (s) Kv/s (s) AcAvKd/2 F(s) Ev(s)
Linear PLL model
Solving for .s/ we have Kt .s/ .s/ F .s/ s Kt Kt .s/ 1 C F .s/ D .s/F .s/ s s .s/ D
3-85
or
Finally, the closed-loop transfer function is

Kt F .s/ .s/ Kt F .s/ s H.s/ D D D t .s/ s C Kt F .s/ F .s/ 1C K s
First-Order PLL Let F .s/ D 1, then we have H.s/ D Kt Kt C s
Consider the loop response to a frequency step, that is for FM, we assume m.t / D Au.t /, then Z t .t / D Akf u./ d so .s/ D Akf s2
The VCO phase output is .s/ D Akf Kt s 2.Kt C s/
The VCO control voltage should be closely related to the applied FM message To see this write Akf s Ev .s/ D .s/ D Kv Kv
3-86
Kt s.s C Kt /
Partial fraction expanding yields, Ev .s/ D thus Akf Kv 1 s e 1 s C Kt

Kt t
Akf h 1 ev .t / D Kv
A m(t)
i u.t /
1st-Order PLL frequency step response at VCO input Kv ev .t/=kf
In general, .s/ D so Ev .s/ D kf M.s/ s kf Kt Kt D M.s/ s Kv s C Kt Kv s C Kt Kt =.2 /, then kf m.t / Kv kf M.s/ s
Now if the bandwidth of m.t / is W Ev .s/
kf M.s/ ) ev .t / Kv
The rst-order PLL has limited lock range and always has a nonzero steady-state phase error when the input frequency is offset from the quiescent VCO frequency
Increasing the loop gain appears to help, but the loop bandwidth becomes large as well, which allows more noise to enter the loop Spurious time constants which are always present, but not a problem with low loop gains, are also a problem with high gain rst-order PLLs
Example 3.17: First-Order PLL Simulation Example

Tool such as MATLAB, MATLAB with Simulink, VisSim/Comm, ADS, and others provide an ideal environment for simulating PLLs at the system level Circuit level simulation of PLLs is very challenging due to the need to simulate every cycle of the VCO The most realistic simulation method is to use the actual bandpass signals, but since the carrier frequency must be kept low to minimize the simulation time, we have difculties removing the double frequency term from the phase detector output By simulating at baseband, using the nonlinear loop model, many PLL aspects can be modeled without worrying about how to remove the double frequency term A complex baseband simulation allows further capability, but will not be discussed at his time The most challenging aspect of the simulation is dealing with the integrator found in the VCO block (Kv =s )
We consider a discrete-time simulation where all continuoustime waveforms are replaced by their discrete-time counterparts, i.e., xn D x.nT / D x.n=f s/, where fs is the sample frequency and T D 1=fs is the sampling period The input/output relationship of an integration block can be approximated via the trapezoidal rule yn D yn 1 C T xn C xn 2 1
function [theta,ev,phi_error] = PLL1(phi,fs,loop_type,Kv,fn,zeta) % [theta, ev, error, t] = PLL1(phi,fs,loop_type,Kv,fn,zeta) % % % Mark Wickert, April 2007 T = 1/fs; Kv = 2*pi*Kv; % convert Kv in Hz/v to rad/s/v if loop_type == 1 % First-order loop parameters Kt = 2*pi*fn; % loop natural frequency in rad/s elseif loop_type == 2 % Second-order loop parameters Kt = 4*pi*zeta*fn; % loop natural frequency in rad/s a = pi*fn/zeta; else error('Loop type must be 1 or 2'); end % Initialize integration approximation filters filt_in_last = 0; filt_out_last = 0; vco_in_last = 0; vco_out = 0; vco_out_last = 0; % Initialize working and final output vectors n = 0:length(phi)-1; theta = zeros(size(phi)); ev = zeros(size(phi)); phi_error = zeros(size(phi)); % Begin the simulation loop
for k = 1:length(n) phi_error(k) = phi(k) - vco_out; % sinusoidal phase detector pd_out = sin(phi_error(k)); % Loop gain gain_out = Kt/Kv*pd_out; % apply VCO gain at VCO % Loop filter if loop_type == 2 filt_in = a*gain_out; filt_out = filt_out_last + T/2*(filt_in + filt_in_last); filt_in_last = filt_in; filt_out_last = filt_out; filt_out = filt_out + gain_out; else filt_out = gain_out; end % VCO vco_in = filt_out; vco_out = vco_out_last + T/2*(vco_in + vco_in_last); vco_in_last = vco_in; vco_out_last = vco_out; vco_out = Kv*vco_out; % apply Kv % Measured loop signals ev(k) = vco_in; theta(k) = vco_out; end
To simulate a frequency step we input a phase ramp Consider an 8 Hz frequency step turning on at 0.5 s and a -12 Hz frequency step turning on at 1.5 s .t / D 2
>> >> >> >> >> >> >> >>
3-90
8.t
0:5/u.t
0:5/
12.t
1:5/u.t
1:5/
t = 0:1/1000:2.5; idx1 = find(t>= 0.5); idx2 = find(t>= 1.5); phi1(idx1) =2*pi* 8*(t(idx1)-0.5).*ones(size(idx1)); phi2(idx2) = 2*pi*12*(t(idx2)-1.5).*ones(size(idx2)); phi = phi1 - phi2; [theta, ev, phi_error] = PLL1(phi,1000,1,1,10,0.707); plot(t,phi_error); % phase error in radians
0.927
Phase Error, (t) (t), (rad)
0.5
With Kt = 2(10) and Kv = 2(1) rad/s/v, we know that with the 8 Hz step ev(t) = 8, so working backwards, sin( - ) = 8/10 = 0.8 and - = 0.927 rad.
-0.412
0.5 0 0.5 1 Time (s) 1.5 2 2.5
Phase error for input within lock range
In the above plot we see the nite rise-time due to the loop gain being 2 .10/ This is a rst-order lowpass step response The loop stays in lock since the frequency swing either side of zero is within the 10 Hz lock range Suppose now that a single positive frequency step of 12 Hz is applied, the loop unlocks and cycle slips indenitely; why?
>> >> >> >> >> >> phi = 12/8*phi1; % scale frequency step from 8 Hz to 12 Hz [theta, ev, phi_error] = PLL1(phi,1000,1,1,10,0.707); subplot(211) plot(t,phi_error) subplot(212) plot(t,sin(phi_error))
3-91
Phase Error ((t) (t))
100
50
le Cyc
slips
0.5
1 Time (s)
1.5
2.5
Phase Error sin((t) (t))
1 0.5 0 0.5 1 0 0.5 1 Time (s) 1.5 2 2.5
Phase error for input exceeding lock range by 2 Hz
By plotting the true phase detector output, sin .t / .t /, we see that the error voltage is simply not large enough to pull the VCO frequency to match the input which is offset by 12 Hz In the phase plane plot shown earlier, this scenario corresponds to the trajectory never crossing zero
3-92
Second-Order Type II PLL To mitigate some of the problems of the rst-order PLL, we can include a second integrator in the open-loop transfer function A common loop lter for building a second-order PLL is an integrator with lead compensation sCa F .s/ D s The resulting PLL is sometimes called a perfect second-order PLL since two integrators are now in the transfer function In text Problem 3.52 you analyze the lead-lag loop lter sCa F .s/ D sC a which creates an imperfect, or nite gain integrator, secondorder PLL Returning to the integrator with phase lead loop lter, the closedloop transfer function is H.s/ D Kt F .s/ Kt .s C a/ D 2 s C Kt F .s/ s C Kt s C Kt a
The transfer function from the input phase to the phase error .t / is G.s/ D or G.s/ D 1
.s/ .s/ .s/ D .s/ .s/ s2 H.s/ D 2 s C Kt s C Kt a

3-93
In standard second-order system notation we can write the denominator of G.s/ D 1 H.s/ (and also H.s/) as
2 s 2 C Kt s C Kt a D s 2 C 2 !ns C !n
where !n D Kt a D natural frequency in rad/s r 1 Kt D D damping factor 2 a p
For an input frequency step the steady-state phase error is zero Note the hold-in range is innite, in theory, since the integrator contained in the loop lter has innite DC gain To verify this we can use the nal value theorem
ss
! s2 D lim s s !0 s 2 s 2 C Kt s C Kt a !s D lim 2 D0 s !0 s C Kt s C Kt a .t / by inverse Laplace transform! 2 s 2 C 2 !ns C !n
In exact terms we can nd ing .s/ D The result for < 1 is

2
! .t / D p !n 1
3-94
!n t
sin !n 1
2t
u.t /
Example 3.18: Second-Order PLL Simulation Example

As a simulation example consider a loop designed with fn D 10 Hz and D 0:707 Kt D 2 !n D 2 0:707 2 10 D 88:84 2 10 !n D D 44:43 aD 2 2 0:707 The simulation code of Example 3.17 includes the needed loop lter via a software switch The integrator that is part of the loop lter is implemented using the same trapezoidal formula as used in the VCO We input a 40 Hz frequency step and observe the VCO control voltage (ev .t /) as the loop rst slips cycles, gradually pulls in, then tracks the input signal that is offset by 40 Hz The VCO gain Kv D 1 v/Hz or 2 rad/s, so ev .t / effectively corresponds to the VCO frequency deviation in Hz
>> >> >> >> >> >>
t = 0:1/1000:2.5; idx1 = find(t>= 0.5); phi(idx1) = 2*pi*40*(t(idx1)-0.5).*ones(size(idx1)); [theta, ev, phi_error] = PLL1(phi,1000,2,1,10,0.707); plot(t,ev) axis([0.4 0.8 -10 50])
3-95
50 VCO Control Voltage ev(t) (Kv = 1 Hz/v)
40
Cycle slipping, but pulling in to match 40 Hz frequency step
30
Cycle slipping stops, and the loop settles
20
10
10 0.4
0.45
0.5
0.55
0.6 Time (s)
0.65
0.7
0.75
0.8
VCO control voltage for a 40 Hz frequency step
Example 3.19: Bandpass Simulation of FM Demodulation

Baseband PLL simulations are very useful and easy to implement, but sometimes a full bandpass level simulation is required The MATLAB simulation le PLL1.m is modied to allow passband simulation via the function le PLL2.m The phase detector is a multiplier followed by a lowpass lter to remove the double frequency term
function [theta, ev, phi_error] = PLL2(xr,fs,loop_type,Kv,fn,zeta) % [theta, ev, error, t] = PLL2(xr,fs,loop_type,Kv,fn,zeta) %
% % Mark Wickert, April 2007 T = 1/fs; % Set the VCO quiescent frequency in Hz fc = fs/4; % Design a lowpass filter to remove the double freq term [b,a] = butter(5,2*1/8); fstate = zeros(1,5); % LPF state vector Kv = 2*pi*Kv; % convert Kv in Hz/v to rad/s/v if loop_type == 1 % First-order loop parameters Kt = 2*pi*fn; % loop natural frequency in rad/s elseif loop_type == 2 % Second-order loop parameters Kt = 4*pi*zeta*fn; % loop natural frequency in rad/s a = pi*fn/zeta; else error('Loop type musy be 1 or 2'); end % Initialize integration approximation filters filt_in_last = 0; filt_out_last = 0; vco_in_last = 0; vco_out = 0; vco_out_last = 0; % Initialize working and final output vectors n = 0:length(xr)-1; theta = zeros(size(xr)); ev = zeros(size(xr)); phi_error = zeros(size(xr)); % Begin the simulation loop for k = 1:length(n) % Sinusoidal phase detector (simple multiplier) phi_error(k) = 2*xr(k)*vco_out; % LPF to remove double frequency term [phi_error(k),fstate] = filter(b,a,phi_error(k),fstate); pd_out = phi_error(k); % Loop gain gain_out = Kt/Kv*pd_out; % apply VCO gain at VCO % Loop filter if loop_type == 2 filt_in = a*gain_out; filt_out = filt_out_last + T/2*(filt_in + filt_in_last);
end
filt_in_last = filt_in; filt_out_last = filt_out; filt_out = filt_out + gain_out; else filt_out = gain_out; end % VCO vco_in = filt_out + fc/(Kv/(2*pi)); % bias to quiescent freq. vco_out = vco_out_last + T/2*(vco_in + vco_in_last); vco_in_last = vco_in; vco_out_last = vco_out; vco_out = Kv*vco_out; % apply Kv; vco_out = sin(vco_out); % sin() for bandpass signal % Measured loop signals ev(k) = filt_out; theta(k) = vco_out;
Note that the carrier frequency is xed at fs =4 and the lowpass lter cutoff frequency is xed at fs =8 The double frequency components out of the phase detector are removed with a fth-order Butterworth lowpass lter The VCO is modied to include a bias that shifts the quiescent frequency to fc D fs =4 The VCO output is not simply a phase deviation, but rather a sinusoid with argument the VCO output phase We will test the PLL using a single tone FM signal
>> >> >> >> >> >> >> >>
3-98
t = 0:1/4000:5; xr = cos(2*pi*1000*t+2*sin(2*pi*10*t)); psd(xr,2^14,4000) axis([900 1100 -40 30]) % Process signal through PLL [theta, ev, phi_error] = PLL2(xr,4000,1,1,50,0.707); plot(t,ev) axis([0 1 -25 25])
30 20 Power Spectrum of xr(t) (dB) 10 0 10 20 30 40 900
920
940
960
980 1000 1020 Frequency (Hz)
1040
1060
1080
1100
Single tone FM input spectrum having fm D 10 Hz and f D 20 Hz

25 20 VCO Control Voltage ev(t) (Kv = 1 Hz/v) 15 10 5 0 5 10 15 20 25 0 0.1 0.2 0.3 0.4 0.5 0.6 Time (s) 0.7 0.8 0.9 1
Recovered modulation at VCO input, ev .t/
3-99
General Loop Transfer Function and Steady-State Errors We have see that for arbitrary loop lter F .s/ the closed-loop transfer function H.s/ is H.s/ D Kt F .s/ s C Kt F .s/
and the loop error function G.s/ D 1 H.s/ is s G.s/ D s C Kt F .s/ In tracking receiver applications of the PLL we need to consider platform dynamics which give rise to a phase deviation of the received signal of the form .t / D Rt 2 C 2 ft C
0
u.t /
which is a superposition of a phase step, frequency step, and a frequency ramp In the s -domain we have .s/ D 2 R 2 f 0 C C s3 s2 s
From the nal value theorem the loop steady-state phase error is 2 R 2 f 0 G.s/ C C ss D lim s s !0 s3 s2 s If we generalize the loop lters we have been considering to the form F .s/ D
3-100
1 2 a b s C as C b D 1 C C s2 s s2
we have for G.s/ s3 G.s/ D 3 s C Kt s 2 C Kt as C Kt b Depending upon the values chosen for a and b , we can create a 1st, 2nd, or 3rd-order PLL using this F .s/ The steady-state phase error when using this loop lter is
ss
s 0s 2 C 2 fs C 2 R D lim 3 s !0 s C Kt s 2 C Kt as C Kt b
ss ?
What are some possible outcomes for
3-101
3.4.2
PLL Frequency Synthesizers
A frequency synthesizer is used to generate a stable, yet programmable frequency source A frequency synthesizer is often used to allow digital tuning of the local oscillator in a communications receiver One common frequency synthesis type is known as indirect synthesis With indirect synthesis a PLL is used to create a stable frequency source The basic block diagram of an indirect frequency synthesizer is the following
fref 1 M Phase Detector Loop Filter VCO fout
fref Freq Div M
fout N
1 N Freq Div
Indirect frequency synthesis using a PLL
When locked the frequency error is zero, thus fout D N M fref
3-102
Example 3.20: A PLL Synthesizer for Broadcast FM

In this example the synthesizer will provide the local oscillator signal for frequency converting the FM broadcast band from 88.1 to 107.9 MHz down to an IF of 10.7 MHz The minimum channel spacing should be 200 kHz We will choose high-side tuning for the LO, thus 88:1 C 10:7 98:8 fLO fLO 107:9 C 10:7 MHz 118:6 MHz
The step size must be 200 kHz so the frequency must be no larger than 200 kHz To reduce the maximum frequency into the divide by counter a frequency offset scheme will be employed The synthesizer with offset oscillator is the following
Freq Div fref 1 M Phase Detector Loop Filter VCO fout
fref M = 200 kHz
fmix N
1 N
fmix
Difference Freq Div Frequency foffset
FM broadcast band synthesizer producing fLO for fIF D 10:7 MHz

Choose foffset < fout then fmix D fout
foffset, and for locking
fref fmix Nfref D ) fout D C foffset M N M Note that Fmix D Nfref=M and fout D fmix C foffset, by virtue of the low side tuning assumption for the offset oscillator Let fref=M D 200 kHz and foffset D 98:0 MHz, then Nmax D and Nmin D 118:6 98:0 D 103 0:2 98:8 98:0 D4 0:2
To program the LO such that the receiver tunes all FM stations step N from 4; 5; 6; : : : ; 102; 103
3-104
Example 3.21: Simple PLL Frequency Multiplication

A scheme for multiplication by three is shown below:
Input at fc Hard limit sinusoidal input if needed Phase Detector Loop Filt. & Ampl.
Input Spectrum
VCO Centered at 3fc xVCO = Acos[2(3fc)t]
f 0 fc 3fc
PLL as a Frequency Multiplier
3-105
Example 3.22: Simple PLL Frequency Division

A scheme for divide by two is shown below:
VCO Output Input at fc
Phase Detector
Loop Filt. & Ampl.
2T0 VCO Output Spectrum
t VCO Centered at fc/2 Lowpass Filter Keep the Fundamental xLPF = Acos[2(fc/2)t]
0 fc/2
PLL as a Frequency Divider
3.4.3
Frequency-Compressive Feedback
ed(t) BPF eo(t) x(t) Discrim ev(t) Demod. Output VCO
xr(t)
Frequency compressive feedback PLL

If we place a discriminator inside the PLL loop a compressing action occurs Assume that xr .t / D Ac cos!c t C .t / and Z ev .t / D Av sin .!c Then,
blocked by BPF t
!o/t C Kv
ev ./ d
1 ed .t / D Ac Av 2
!o/t C other terms Z t sin!ot C .t / Kv ev ./ d

passed by BPF
sin .2!c
so x.t / D h 1 Ac Av sin !ot C .t / 2 Z Kv

t
ev ./ d
Assuming an ideal discriminator ev .t / D or ev .t / 1 C

1 d .t / KD 2 dt
Kv ev .t /
Kv KD KD d .t / D 2 2 dt
3-107
For FM d .t /=dt D 2 fd m.t /, so ev .t / D KD fd m.t / 1 C Kv KD =.2 /
which is the original modulation scaled by a constant The discriminator input must be x.t / D 1 1 Ac Ad sin !ot C .t / 2 1 C Kv KD =.2 /
Assuming that Kv KD =.2 / 1 we conclude that the discriminator input has been converted to a narrowband FM signal, which is justies the name frequency compressive feedback
3.4.4
Coherent Carrier Recovery for DSB Demodulation
Recall that a DSB signal is of the form xr .t / D m.t / cos !c t A PLL can be used to obtain a coherent carrier reference directly from xr .t / Here we will consider the squaring loop and the Costas loop
xr(t)
m(t)cos(ct + ) Am2(t)cos(2ct + 2) ( )2 Loop Filter x2 VCO -90o 0o
LPF cos(ct + )
m(t)cos()
Bsin(2ct + 2)
static phase error
Squaring Loop
LPF 0o -90o LPF m(t)cos() 1 2 m (t)sin2 2 xr(t) cos(ct + ) sin(ct + ) VCO Loop Filter k sin(2)
m(t)sin()
Costas Loop
Note: For both of the above loops m2.t / must contain a DC component The Costas loop or a variation of it, is often used for carrier recovery in digital modulation Binary phase-shift keying (BPSK), for example, can be viewed as DSB where 1 X m.t / D dnp.t nT /
nD 1 ECE 5625 Communication Systems I 3-109
where dn D 1 represents random data bits and p.t / is a pulse shaping function, say ( 1; 0 t T p.t / D 0; otherwise Note that in this case m2.t / D 1, so there is a strong DC value present
m(t)
1 0.5 0.5 1 1 0.5 2 4 6 8 10
t/T
m(t)cos(ct)
t/T
2 0.5 1 4 6 8 10
BPSK modulation
Digital signal processing techniques are particularly useful for building PLLs In the discrete-time domain, digital communication waveforms are usually processed at complex baseband following some form of I-Q demodulation
LPF 0
o
rI(t)
A/D
xIF(t)
-90o LPF
Sampling fs clock cos[2fcLt + L] fs rQ(t) A/D
DiscreteTime
r[n] = rI[n] + jrQ[n]
IF to discrete-time complex baseband conversion

y[n]
To Symbol Synch
x[n]
From Matched Filter
j [ n ]
( )M 2M 2 1
[n]
v[n] Error Generation
Im( )
e[n]
LUT NCO
z 1
kp
z 1
Loop Filter
ka
M th-power digital PLL (DPLL) carrier phase tracking loop

From Matched Filter
x[n]
To Symbol Synch L-Tap Delay [n] L-Tap MA FIR

e j( )
y[n]
[n] F( ) Rect. to Polar [n] M
1 arg() M
e j( )
Non-Data Aided (NDA) feedforward carrier phase tracking

3.5
Sampling Theory
We now return to text Chapter 2, Section 8, for an introduction/review of sampling theory Consider the representation of continuous-time signal x.t / by the sampled waveform " 1 # 1 X X x .t / D x.t / .t nTs / D x.nTs /.t nTs /
nD 1 nD 1
x(t) Sampling
x(t)
-Ts
Ts 2Ts 3Ts 4Ts 5Ts
How is Ts selected so that x.t / can be recovered from x .t /? Uniform Sampling Theorem for Lowpass Signals Given F fx.t /g D X.f / D 0; then choose Ts < 1 2W or fs > 2W .fs D 1=Ts / for f > W
to reconstruct x.t / from x .t / and pass x .t / through an ideal LPF with cutoff frequency W < B < fs W 2W D Nyquist frequency fs =2 D folding frequency
3.5. SAMPLING THEORY
proof: " X .f / D X.f / but X.f / .f fs

1 X nD 1
# .f nfs /
nfs / D X.f X .f / D fs
1 X nD 1
nfs /, so X.f
X(f)
nfs /
X0
-W Lowpass reconstruction filter ... fs > 2W -fs fs < 2W ... -W 0 X(f) X0 fs
f Guard band = fs - 2W ... f
fs-W Aliasing
fs
X0 fs
... f
-2fs
-fs
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 x .t / all we need to do is lowpass lter the sampled signal with an ideal lowpass lter having cutoff frequency W < fcutoff < fs W In simple terms we set the lowpass bandwidth to the folding frequency, fs =2 Suppose the reconstruction lter is of the form H.f / D H0 we then choose W < B < fs f e 2B W
j 2 f t0
For input X .f /, the output spectrum is Y .f / D fs H0X.f /e and in the time domain y.t / D fs H0x.t t0 /
j 2 f t0
If the reconstruction lter is not ideal we then have to design the lter in such a way that minimal desired signal energy is removed, 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 interpolating signal values between the available sample values Suppose that the reconstruction lter has impulse response h.t /,
3.5. SAMPLING THEORY
then y.t / D
1 X nD 1
x.nTs /h.t
1 X nD 1
nTs / t0 nTs /
D 2BH0
x.nTs /sinc2B.t
where in the last lines we invoked the ideal lter described earlier Uniform Sampling Theorem for Bandpass Signals If x.t / has a single-sided bandwidth of W Hz and F fx.t /g D 0 for then we may choose fs D where mD 2fu m f > fu
fu ; W which is the greatest integer less than or equal to fu=W
Example 3.23: Bandpass signal sampling

1 0.8 0.6 0.4 0.2 4 2
X(f)
Input signal spectrum

In the above signal spectrum we see that W D 2; so fs D will work The sampled signal spectrum is X .f / D 4
Recover with bandpass filter
4 3 2 1 15 10 5
1 X nD 1
fu D 4
fu=W D 2 ) m D 2 2.4/ D4 2
X.f
nfs /
X(f)
f
5 10 15
-3fs
-2fs
-fs
fs
2fs
3fs
Spectrum after sampling
3-116
3.6. ANALOG PULSE MODULATION
3.6
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 nite set of allowable values
3.6.1
Pulse-Amplitude Modulation (PAM)
PAM produces a sequence of at-topped pulses whose amplitude varies in proportion to samples of the message signal Start with a message signal, m.t /, that has been uniformly sampled 1 X m .t / D m.nTs /.t nTs /
nD 1
The PAM signal is mc .t / D

1 X nD 1
m.nTs /
.nTs C =2/
m(t) mc(t) 0 Ts 2Ts 3Ts 4Ts t
PAM waveform
It is possible to create mc .t / directly from m .t / using a zeroorder hold lter, which has impulse response h.t / D and frequency response H.f / D sinc.f /e
m(t) h(t)
j f
=2
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 reconstruction lter having cutoff frequency at fs =2 D 2=Ts In general, there may be a need for equalization if t au is on the order of Ts =4 to Ts =2
sinc() function envelope Lowpass reconstruction filter
-fs
-W
fs
mc(t)
Lowpass
m(t)
Recovery of m.t/ from mc .t/

3.6. ANALOG PULSE MODULATION
3.6.2
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 ampliers A lowpass lter applied to a PWM waveform recovers the modulation m.t /
PWM Signal
1 0.5
t
20 10 Analog input m(t) 0.5 1 10 20
Example PWM signal
3.6.3
Pulse-Position Modulation
With PPM the displacement in time of each pulse, with respect 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 nonoverlapping pulses, is a form of M -ary orthogonal communications
PPM of this type is nding application in ultra-wideband communications

PPMSignal
1 0.5
t
20 10
Analog input m(t)
0.5 1
10
20
Example PPM signal
3.7
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
3.7.1
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
3.7. DELTA MODULATION AND PCM
m(t)
+ d(t)
(t)
xc(t)
-1 Pulse Modulator
ms(t) =
Control the step size
Delta modulator with step size parameter 0

Start-up transient
1 m(t) and ms(t) 0.5 0 0.5 1 0 0.005 0.01 0.015 0.02 0.025 0.03 Time (s) 0.035 0.04 0.045 0.05
Slope overload
m(t) (blue) ms(t) (red) 0 = 0.15
1 0.5 xc(t) 0 0.5 1 0 0.005 0.01 0.015 0.02 0.025 0.03 Time (s) 0.035 0.04 0.045 0.05
Delta modulator waveforms
The maximum slope that can be followed is 0=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 ltering to remove the stair step edges (lowpass ltering directly is a simplication) Slope overload can be dealt with through an adaptive scheme If m.t / is nearly constant keep the step size 0 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 amplier, where the gain is controlled by square-law detecting the output of a lowpass lter acting on xc .t /
m(t)
+ ms(t) d(t)
(t)
xc(t)
-1 Pulse Modulator VGA
( )2
LPF
Means to obtain a variable step size DM
3.7.2
Pulse-Code Modulation (PCM)
Each sample of m.t / is mapped to a binary word by 1. Sampling 2. Quantizing 3. Encoding

m(t) Equivalent Views m(t) Sample & Hold Analog to Digital Converter n Parallel to Serial Converter Serial Data PCM Output
Sampler
Quantizer
Encoder
3-123
Quant. Encoded Level Output 7 111 6 110 5 101 4 100 3 011 2 010 1 001 0 000 0
Quantizer Bits: n = 3, q = 2n = 8 m(t)
t 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 ; 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 . /max D B ' kW n; where k is a proportionality constant When located on a carrier the required bandwidth is doubled
Binary phase-shift keying (BPSK), mentioned earlier, is a popular 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 Qm.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 3.24: 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
0.163 mm
One Frame of 12 Audio Samples

Synch Sub (27 bits) Code (8 bits) Data (96 bits) Parity (32 bits) Data (96 bits) Parity (32 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
3.8
Multiplexing
It is quite common to have multiple information sources located at the same point within a communication system To simultaneously transmit these signals we need to use some form of multiplexing
3-126
3.8. MULTIPLEXING
There is more than one form of multiplexing available to the communications engineer
3.8.1
Frequency-Division Multiplexing (FDM)
With FDM the idea is to locate a group of messages on different subcarriers and then sum then together to form a new baseband signal which can then be modulated onto the carrier
m1(t) Mod #1 fsc1 m2(t) Mod #2 fsc2 Composite baseband ... Mod #N fscN RF Mod fc xc(t) Lower bound on the composite signal bandwidth
mN(t)
FDM transmitter
At the receiver we rst demodulate the composite signal, then separate into subcarrier channels using bandpass lters, then demodulate the messages from each subcarrier
BPF fsc1
Sub Car. Demod #1
yD1(t)
RF Demod
BPF fsc2 ... BPF fscN
Sub Car. Demod #2 ... Sub Car. Demod #N
yD2(t) ... yDN(t)
FDM receiver/demodulator
The best spectral efciency is obtained with SSB subcarrier modulation and no guard bands At one time this was the dominant means of routing calls in the public switched telephone network (PSTN) In some applications the subcarrier modulation may be combinations both analog and digital schemes The analog schemes may be combinations of amplitude modulation (AM/DSM/SSB) and angle modulation (FM/PM)
3-128
3.8. MULTIPLEXING
Example 3.25: FM Stereo

l(t) + + + r(t) 19 kHz Pilot x2 Freq. Mult Pilot Carrier l(t) + r(t)
l(t) - r(t)
+ + 38 kHz
xb(t)
FM Mod
xc(t)
19 kHz pilot
fc
Xb(f)
Other subcarrier services can occupy this region
15
19
23
38
53
f (kHz)
FM stereo transmitter
Mono output l(t) + r(t) l(t)
xr(t)
FM Discrim
xb(t)
LPF W = 15 kHz LPF W = 15 kHz Coherent demod of DSB on 38 kHz subcarrier
BPF fc = 19 kHz x 2 Freq Mult
l(t) - r(t)
r(t)
FM stereo receiver
3.8.2
Quadrature Multiplexing (QM)

Accosct 2cosct LPF xc(t) Channel xr(t) d1(t) yD1(t)
m1(t)
m2(t) d2(t) Acsinct 2sinct
LPF
yD2(t)
QM modulation and demodulation
With QM quadrature (sin/cos) carrier are used to send independent message sources The transmitted signal is xc .t / D Ac m1.t / cos !c t C m2.t / sin !c t If we assume an imperfect reference at the receiver, i.e., 2 cos.!c t C /, we have d1.t / D Ac m1.t / cos m2.t / sin C m1.t / cos.2!c t C / C m2.t / sin.2!c t C /
LPF removes these terms
yD1.t / D Ac m1.t / cos C m2.t / sin The second term in yD1.t / is termed crosstalk, and is due to the static phase error
3.8. MULTIPLEXING
Similarly yD2.t / D Ac m2.t / cos m1.t / sin
Note that QM acheives a bandwidth efciency similar to that of SSB using adjacent two subcarriers or USSB and LSSB together on the same subcarrier
3.8.3
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 ow from the source to user For message sources with harmonically related bandwidths we can interleave samples such that the wideband sources are sampled more often To begin with consider equal bandwidth sources
Info. Source 1 Info. Source 2 ...
Synchronization Required Channel
Info. User 1 Info. User 2 ...
Commutators Info. Source N Info. 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
i D1
An equivalent signal channel of bandwidth B would produce 2BT D ns samples in T s, thus the equivalent base3-132 ECE 5625 Communication Systems I
3.8. MULTIPLEXING
band signal bandwidth is BD

N X i D1
Wi Hz
which is the same minimum bandwidth required for FDM using SSB Pure digital multiplexing behaves similarly to analog multiplexing, 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 channels each sampled at 44.1 ksps with 16-bit quantizers, multiplex up to 2 16 44; 100 D 1:4112 Msps
3-133
Example 3.26: Digital Telephone System

The North American digital TDM hierarchy is based on a single 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 Sys. Number R (Mb/s) Channels DS-0 0.064 1 T1 DS-1 1.544 24 T1C DS-1C 3.152 48 T2 DS-2 6.312 96 T3 DS-3 44.736 672 DS-3C 90.254 1344 DS-4E 139.264 2016 T4 DS-4 274.176 4032 DS-432 432.00 6048 T5 DS-5 560.160 8064
Transmission Media Used Wire pairs Wire pairs Wire pairs Wire pairs Coax, radio, ber Radio, ber Radio, ber, coax Coax, ber Fiber Coax, ber
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
3.9. GENERAL PERFORMANCE OF MODULATION SYSTEMS IN NOISE
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)
3.9
General Performance of Modulation Systems in Noise

Regardless of the modulation scheme, the received signal xr .t /, is generally perturbed by additive noise of some sort, i.e., xr .t / D xc .t / C n.t / where n.t / is a noise process The pre- and post-detection signal-to-noise ratio (SNR) is used as a gure or merit
3-135
xr(t)
Pre-Det. Filter P (SNR)T = T <n2(t)> Common to all systems
Demod/ Detector (SNR)D =
yD(t) <m2(t)> <n2(t)>
FM
PCM q = 256 (SNR)D PCM q = 64
1 D=
FM =2
0
5
Nonlinear modulation systems have a distinct threshold in noise
= ,D
FM
,D
SB D Q B, mod S S e B, nt D S e D er h Co
(SNR)T
General modulation performance in noise
3-136

2-Dieu Che Tuong Tu

Uploaded by

Document Information

Original Description:

Copyright

Available Formats

Share this document

Share or Embed Document

Sharing Options

Did you find this document useful?

Is this content inappropriate?

Copyright:

Available Formats

2-Dieu Che Tuong Tu

Uploaded by

Copyright:

Available Formats

Chapter

Angle Modulation . . . . . . . . . . . . . . . . . . . . 3-46 3.2.1 3.2.2 3.2.3 3.2.4 3.2.5 3.2.6

Interference . . . . . . . . . . . . . . . . . . . . . . . 3-73 3.3.1 3.3.2

Feedback Demodulators . . . . . . . . . . . . . . . . . 3-81

CHAPTER 3. ANALOG MODULATION

3.4.1 3.4.2 3.4.3 3.4.4 3.5 3.6

Delta Modulation and PCM . . . . . . . . . . . . . . . 3-120 3.7.1 3.7.2

Multiplexing . . . . . . . . . . . . . . . . . . . . . . . 3-126 3.8.1 3.8.2 3.8.3

General Performance of Modulation Systems in Noise

ECE 5625 Communication Systems I

3.1. LINEAR MODULATION

Double-Sideband Modulation (DSB)

CHAPTER 3. ANALOG MODULATION

carrier filled envelope

DSB time domain waveforms

f Xc(f) 1 A M(0) 2 c LSB fc

2cos[2fct] Channel Demodulator

3.1. LINEAR MODULATION

Lowpass modulation recovery filter 1 A M(0) 2 c f 2fc

Waveform and spectrum of d.t/

CHAPTER 3. ANALOG MODULATION

3.1. LINEAR MODULATION

One scheme is to simply square the received DSB signal

BPF very narrow (tracking) bandpass filter

Carrier recovery concept using signal squaring

ECE 5625 Communication Systems I

CHAPTER 3. ANALOG MODULATION

Consider transmitting a small amount of unmodulated carrier

m(t) k Accos2fct AcM(0)/2 k << 1

m.t / j min m.t /j

3.1. LINEAR MODULATION

Generation of AM and a sample wavefrom

ECE 5625 Communication Systems I

CHAPTER 3. ANALOG MODULATION

Recovered envelope with proper RC selection

0 The carrier is removed if 1/fc << RC << 1/W

3.1. LINEAR MODULATION

CHAPTER 3. ANALOG MODULATION

pure carrier spectrum

3.1. LINEAR MODULATION

Example 3.1: Single Sinusoid AM

ECE 5625 Communication Systems I

CHAPTER 3. ANALOG MODULATION

3.1. LINEAR MODULATION

|Xc(f)| Ac/2 0.8Ac/4 -fc 0 Xc(f) 0 -/3 /3 f fc-fm fc fc+fm f

Amplitude and phase spectra for one tone AM

Example 3.2: Pulse Train with DC Offset

m.t / m.t / D D m.t / j min m.t /j j 1j

CHAPTER 3. ANALOG MODULATION

.7=3/a2 7a2 ED D 1 C .7=3/a2 3 C 7a2

The efciency in this case is Eff .1=2/a2 a2 D D 1 C .1=2/a2 2 C a2

ECE 5625 Communication Systems I

3.1. LINEAR MODULATION

Example 3.3: Multiple Sinusoids

>> subplot(311) >> plot(t,(1 + 0.25*m/abs(min(m))).*cos(2*pi*1000*t)) >> hold

CHAPTER 3. ANALOG MODULATION

Finding min m.t/ graphically

3.1. LINEAR MODULATION

Modulation index comparison (fc D 1000 Hz)

A2 k 2j min m.t /j2

ECE 5625 Communication Systems I

CHAPTER 3. ANALOG MODULATION

A remaining interest is the spectrum of xc .t / Xc .f / D Ac .f fc / C .f C fc / 2 M aAc X h j k C Ak e .f .fc C fk // 4 k D1 i j k Ce .f C .fc C fk // (USB terms)

>> subplot(311) >> plot(t,(1 + 0.25m/abs(min(m))).cos(2pi1000*t)) >> hold