Tr-Rec 11

Transmission and Receiving Techniques
Prof. Michael Bank lectures

23. 05. 2011
Dear student. I w ould like to help you in this course study. For this purpose I have prepared slides which I show in auditorium. But these slides are not lectures outline. They are only illustrations. During a lectures I give explanations, proves, show connections with other courses, solve exercises. It is impossible to get ready to exam without attending lectures. I wish you successful course study. See you in lectures.
Contents
Slides Introduction, Milestones in the history of radio 6 1. Frequencies, Propagation 16 2. Amplifying 21 3. Analogue Modulations and Modulators 41 4. Digital Modulations and Modulators 80 5. UP Down Conversion, Superheterodyne and Homodyne Receiving 115 6. Receivers 134 7. Antenna in a small transceiver 176 8. IF Amplifiers 199 9. Analogue Decoding 212 10. Digital Coherent Decoding 226 11. Non Coherent Decoding 246 12. Phase-locked loop (PLL) 253 Appendix 1. Preparing to Tx-Rx-Lab 273 Appendix 2. Shortly about DWDM 281 Appendix 3 Operational Amplifier as Universal Element 287 Appendix 4 Addition to digital modulation 296 Appendix 5. Hartley modulator 307 Possible exam questions 319
Bibliography
Textbooks: 1. Schwartz, M., "Information Transmission, Modulation and Noise", 4th Ed., McGraw-Hill, 1990. 2. Tomasi,W., Electronic Communications systems Prentice Hall,1988 3. Hardy, J.K., Electronic Communication Technology, Prentice-Hall, 1986. 4. Stremler, F.G. Introduction to Communication Systems, 3rd Ed, Addison-Wesley, 1990. 5. Haykin, S.S., Communication Systems 2nd Ed., Wiley, 1983. 6. Taub, H., D.L. Schilling, Principles of Communication Systems McGraw-Hill, 1986. Additional books: 1. Edited by William Gosting. Radio Receivers. Peter Peregrines Ltd., 1986. 2. A. S. Sedra, K. E. Smith., Microelectronics Circuits. Oxford Univ. Press, 1998. 3. Rohge,U., Bucher,T., Communications Receivers McGraw-Hill, 1988. 4. M. Bank., Analog integrated circuits of the radio apparatus Radio and Communication, Moscow, 1981
Color Meaning
E = mC 2
Important formula. You have to know it.
V = I R
There is formula, which you have seen in one of previous slides.
Rin,CC > Rin,CE

To draw your attention.
5
@
Original slide
Introduction, Milestones in the history of radio
Tesla 1899
9
10 Between 1895 and 1899, Tesla claimed to have received wireless signals transmitted
over long distances, although there is no independent evidence to support this
11
12
Band select filter
Low-pass filter
LNA
DC
LO
13
14
1. Frequencies, Propagation.
3 108 m = f Hz
300 = f THz
1THz = 1012 Hz
300 m = f MHz
1. 55 m = 1550nm
15
193.55THz = 193550GHz
Micrometer (m) is 10-6m, nanometer (nm) is 10-9m, angstrom is 10-10m.
Frequency Bands
16
Relative Frequency Bandwidth
0.1 1 GHz
f/f0
2GHz
10kHz
6MHz
100MHz
106 107 108 109 1010 1011 1012 1013 1014
f( Hz)
Narrow frequency band is f/f0 << 1
17
@
Field Density Distribution Example
18
Distribution law Example ( < x < 0) 0 f ( x) = e x (0 < x < ) ( < x < 0) 0 F ( x) = 1 e x (0 < x < ) LW 1 MW 2 SW 3 x 1 x = 3.5 = 2.5 1/(mV/m) = 3.5
One-sided normal law
There is uniform law for local station signals distributions.

19
Exercises
1. Find the average of distribution (mean value) ()
+
( < x < 0) 0 f ( x) = x (0 < x < ) e ( < x < 0) 0 F ( x) = x (0 < x < ) 1 e
E =
x e x dx = 1
2. Find the probability of field density being less than average of distribution
F ( x) x= 1
1 = 1 e = 1 = 1 0.37 = 0.63 . e
1
In this case mean value is not median (!)
3. Find the probability of field density in MW frequency band being less than 1 mV/m.
F ( x) x=1 = 1 e2.51 = 1 0.08 = 0.92.

20
2. Amplifying
There are three types of amplifying
A. Amplification by control of feeding source current.
current sources
Transistors Vacuum lamp Vi

n
Vin
B. Amplification by grouping of electrons.
Tunnel diode
Resonator device (Klystron)
Long term interaction device (TWT)
C. Amplification by pump generator.
Parametrical amplifier
Amplified is used to stimulate excited Optical(atoms into emitting energy )Maser ,amplifier
D. Amplification by negative resistance using

21
A. Amplification by control of feeding source current. MOSFET Power Amplifier

There are three contradictories parameters: Output Power Frequency Band Central Frequency
Today there are amplifiers for 1 GHz and more, with power up to 1 kWt, voltage is 60 V.
Some examples
, where X band is 8 12 GHz
Two stage power amplifier structure supports

180 Mbps data rate at 8.2 GHz, 7W (38.5 dBm) output power, Gain 20.5 0.6 dB in 8.17-8.265 GHz bandwidth Overall efficiency is 26%.
The first stage: output power of 30 dBm, efficiency of 22%. The second stage: efficiency 38.5 dBm output and 35%
23
K band is 12 40 GHz OTOI - Output third order intercept (see slide 162) P1dB - Output power 1dB compressed
Tunnel diode as a amplifier

A rough approximation of the Volt-Ampere curve for a tunnel diode, showing the region of differential negative resistance
Antenna current amplifying example

Negative resistance of TD increase signal to noise ratio.
Tunnel diode
Amplifier before detector. Detector decreases L1C1 circuit quality, but TD decreases losing and increase amplifying.
TD
25 Transmitter
Amplifier limiter
VCC
Input signal
Pamela AM amplification with help limiters
Compa rator
+
Output signal
26
100%
One of possible amplifier Pamela realization
-600
vin
600
Up
Down
If Vin = Vin,max If Vin = Vin,min
Up = 60o-60o = 0o
and Doun = -60o+60o = 0o
Up = -60o-60o = -120o and Doun = 60o+60o = 120o
In case of OFDM It is possible situation when the amplitude is changing from average value to minimum or to maximum during one symbol (T = 1/F or fm = F ). The minimal frequency changing in OFDM mast be F or For Pamela system we have:
27
F 60 0 1 radian = 1 = fm
@
Time T
B. Amplification by grouping of electrons.

Vi
Particles flew out with different speed.
After some time they will be together
Time T + T
1 Star birth by Hubble
The brothers Russell and Sigurd Varian of Stanford University are generally considered to be the inventors of the klystron. Their prototype was completed in August 1937. Upon publication in 1939
P2n
U1(t)
resonator
P0
V= dz = V0 dt
Grouping of electrons
Single-cavity klystron
Speed of electrons 1, 5, 9, 13, 17 is not changing Electrons 2, 3, 4 have V < V0 Electrons 6, 7, 8 have V > V0 Before input resonator 1 2 3 4 5 6 7 8 9 After input resonator 1 2 3 4 5 6 78 9
Electrons numbers
Efficiency
Real < 20%
n is harmonic number; X is parameter of grouping
31
Klystron
In case of resonators detuning in four-six-cavity klystron one can achieve 75% and F 5%, amplifying up to 60 dB. Output power up to 0.5MWt in streaming mode and up to 50MWt in pulse mode.
Multi-cavity klystron P0
Cathode
In
Out
Collector
is some percents and F (electrical) 0.5%, F (mechanical) 25%.

Catoptrical klystron is used in receiver local oscillator and in parametrical amplifiers.
Catoptrical klystron (5mw 5W)
One resonator Reflector
Klystron applications
Klystrons produce microwave power far in excess of that developed by solid state. In modern systems, they are used from UHF (100's of MHz) up through hundreds of gigahertz (as in the Extended Interaction Klystrons in the CloudSat satellite). Klystrons can be found at work in radar, satellite and wideband high-power communication (very common in television broadcasting and EHF satellite terminals), and high-energy physics (particle accelerators and experimental reactors). At SLAC, for example, klystrons are routinely employed which have outputs in the range of 50 megawatts (pulse) and 50 kilowatts (time-averaged) at frequencies nearing 3 GHz. Popular Science's "Best of What's New 2007"[included a company using a klystron to convert the hydrocarbons in everyday materials, automotive waste, coal, oil shale, and oil sands into natural gas and diesel fuel. In St. Petersburg, FL, local cable news station Bay News 9 uses a Klystron Tube in its Klystron 9 Weather Radar to increase its range and resolution. The radar site is actually located behind the Bright House Sports Network building in Pinellas Park, FL. SLAC National Accelerator Laboratory is home to a two-mile linear acceleratorthe longest in the world. Originally a particle physics research center, SLAC is now a multipurpose laboratory for astrophysics, photon science, accelerator and particle physics research. Six scientists have been awarded the Nobel Prize for work carried out at SLAC and the future of the laboratory promises to be just as extraordinary. Menlo Park, CA Operated by Stanford University for the U.S. Dept. of Energy
Traveling-Wave Tube (TWT) - 1943

Klystron is resonators device with brief interaction of input signal and electrons energy. Therefore klystron is narrow-band amplifier with strong input signal. TWT is device with long term interaction.
E z = E 0 exp j ( t z )
= V is wave number
V
is phase velocity
Electrons 1, 3 and 5 with phase 0, and 2 don't change they velocity. Electron 2 is accelerating and electron 4 slowing down 5 9 10
1 1 Electrons gun 2, 3, 4, - anodes 7 decelerating system 5 input waveguide 9 output waveguide 8 absorbent 10 - collector 12 focusing solenoid
2-4
12
TWT type O
Electrons go through spiral and interact with spiral field.
In
Control electrode
Ez
Decelerating system
Out
Collector
TWT type M
Cathode
Description The device is an elongated vacuum tube with an electron gun (a heated cathode that emits electrons) at one end. A magnetic containment field around the tube focuses the electrons into a beam, which then passes down the middle of a wire helix that stretches from the RF input to the RF output, the electron beam finally striking a collector at the other end. A directional coupler, which can be either a waveguide or an electromagnetic coil, fed with the low-powered radio signal that is to be amplified, is positioned near the emitter, and induces a current into the helix. The helix acts as a delay line, in which the RF signal travels at near the same speed along the tube as the electron beam. The electromagnetic field due to the current in the helix interacts with the electron beam, causing bunching of the electrons (an effect called velocity modulation), and the electromagnetic field due to the beam current then induces more current back into the helix (i.e. the current builds up and thus is amplified as it passes down). A second directional coupler, positioned near the collector, receives an amplified version of the input signal from the far end of the helix. An attenuator placed on the helix, usually between the input and output helicies, prevents reflected wave from travelling back to the cathode. Higher powered TWTs often contain beryllium oxide ceramic as both a helix support rod and in some cases, as an electron collector for the TWT 35 because of its special electrical, mechanical, and thermal properties.
2. In plane x,t field moving trajectory will be straight line on the angle tg V
3. Accelerating phase 2n till (2n + 1)
Decelerating phase (2n + 1) till 2(n + 1)
1. Long axis electromagnetic wave with phase velocity V
4. Electrons with velocity v0 = V
5. There is electrons grouping round electrons with phase 2n
The gain factor of TWT is 15 35 dB, in low power TWT up to 60 dB. Bandwidth is 10% and more. In linear mode is some percents.
36 . . , 1979. 36
C. Amplification by pump generator.

Parametrical amplifier
p
q = CV
Pump Generator
mcr
C V = C V
Generator Amplifier
V = q +
Let C = C 0 + C m Sin p t = C 0 (1 + mSin p t ) m =
2s/p
Cm
C0
dq d 2q + R L dt dt 2
q = 0 C
0 =
1 R = 2L LC 0
d 2q dq 1 + 2 + 02 (1 + mSin p t ) q = 0 dt 2 dt
There are parametric amplifiers with frequency conservation and with frequency converting. Amplifiers of first type are used in satellite communication. These amplifiers have narrow frequency band.
37
fin; fP ; fin +/- fP
For achieving minimal noise temperature without cooling it is need to implement: - Varicaps with high critical frequency where is capacity modulation coefficient, C0 is middle capacity and r is varicap losses resistor . - increase pump frequency to signal frequency ratio up to 515 ( 50 .100 GHz) - Decrease losses in amplifier input part The varicaps on Schottky barrier have critical frequency more than 200GHz
f cr =
2 C 0 r
The pump generate made by diodes Gunn [Gunn diodes are similar to tunnel diodes in that they are made of materials such as GaAs or InP that exhibit a region of negative differential resistance. With appropriate biasing, dipole domains form and travel across the diode, allowing high frequency microwave oscillators to be built.]
Today No cooled parametrical amplifiers have TN up to 30K, amplifying 16 db, and relative frequency band 15% With help fluid nitrogen or helium it is possible to achieve TN = 10K The parametrical amplifier disadvantages are: The pump generator has low reliability. Large power consumption 38 Temperature changing influence 38
7 W RF Stage Example
MITEQ Driver amp. Isolator Input Isolator Matching Fujitsu FLM7785-4F Output Matching Fujitsu Input FLM7785-12F Matching Fujitsu FLM7785-12F
Att.
BPF
Output Matcing
39
BPF For harmonic suppression at the mixer output Insertion loss is not critical. Can be realized with microstrip lines Two stage narrowband power amplifier supports 180 Mbps data rate at 8.2 GHz, 7W (38.5 dBm) output power, gain of 20.5 0.6 dB in 8.17-8.265 GHz bandwidth the efficiency is 26%. The first stage of the power amplifier Class AB amplifier output power 30 dBm efficiency 22%.
39
Optical Amplifier
The pumping light is absorbed by the erbium atoms are then stimulated to emit by the longer-wavelength photons, amplifying the signal. The signal beam and pumping beam travel together down the fiber. The signal beam continually increases in strength while depleting the pump power. The second WM removes any pump photons not absorbed by the doped fiber so that they do not reach the receiver and interfere with signal detection. Doped fiber amplifiers
Doped fibre amplifiers (DFAs) are optical amplifiers which use a doped optical fibre as a gain medium to amplify an optical signal. They are related to fibre lasers. The signal to be amplified and a pump laser are multiplexed into the doped fibre, and the signal is amplified through interaction with the doping ions. The most common example is the Erbium Doped Fiber Amplifier (EDFA), where the core of a silica fiber is doped with trivalent Erbium ions (Er+3), can be efficiently pumped with a laser at 980 nm or at 1,480 nm, and exhibits gain in the 1,550 nm region. Amplification is achieved by stimulated emission of photons from dopant ions in the doped fibre. The pump laser excites ions into a higher energy from where they can decay via stimulated emission of a photon at the signal wavelength back to a lower energy level. The excited ions can also decay spontaneously (spontaneous emission) or even through non radiative processes involving interactions with phonons of the glass matrix. These last two decay mechanisms compete with stimulated emission reducing the efficiency of light amplification. The amplification window of an optical amplifier is the range of optical wavelengths for which the amplifier 40 yields a usable gain. The amplification window is determined by the spectroscopic properties of the dopant ions, the glass structure of the optical fiber, and the wavelength and power of the pump laser.
3. Analogue Modulations and Modulators Analogue Modulators

Inf. Processor Modulator Filter UpConverter Amplifier Coupler/ Filter
Inf. Modulator
Inf.
Inf. Spectrum Former
Carrier oscillator
41
Carrier oscillator
FFT-1
D/A
Basic Transmitter Parameters
1. 2. 3. 4. 5. 6. 7.
Power. Frequency Band. Out of band Transmitting. Stability. Envelope (Information) Distortions. Peak Factor Power Efficiency.
42
Measurement Scheme
Wattmeter
Inf. Source
Transmitter
Artificial Antenna
Specter Analyzer
Test Receiver
43
Requirement to Out of band Transmitting
44
Amplitude Modulation
e(t)
a (t ) = [ A0 + Ke (t ) ]Sin ( 0t + 0 ) = A(t ) Sin ( 0 t + 0 )

A0
Am
A(t ) = A0 + Am Sin ( t + )
a (t ) = A0 [1 + mSin ( t + )]Sin ( 0 t + 0 )
Am m= A0
t
2/ 2/
45
Amplitude Modulation Power

For different m:
A(t) current amplitude
m=0 m0
A02 r P0 = 2 Pmax Pmin

2 A02 (1 + m ) Amax r 2 r = P0 (1 + m ) = = 2 2 2 2
r
2
2 A02 (1 m ) Amin r 2 r = P0 (1 m ) = = 2 2
For period of
+ P
2 2 r A (t ) 2 2 r r = A0 1+ m Sin ( t + ) = A0 1+ 2 m sin( t + ) + m 2 [sin( t + ) ] = 2 2 2 2 2 r 1+ 2 m sin( t + ) + m = A0 [1 cos 2( t + ) ] 2 2 =0 =0
P (t ) =
Peak factor or Peak Average Power Ratio (PARP) =
10 log
Pmax
P( t )
Pmin = 0 P(t ) = 1,5P0
2 P(t ) = P0 1+ 0,5m
For
m = 1,
46
Pmax = 4 P0
= 4,2dB
AM Spectrum
e ( t ) = ASin ( t + ) a ( t ) = A 0 [1 + mSin ( t + =
)]Sin ( 0 t + 0 ) = A 0 [Sin ( 0 t + 0 ) + mSin ( t + )Sin ( 0 t + 0 ) ] =
mA 0 Cos [( 0 + )t + 0 + ] + = A 0 Sin ( 0 t + 0 ) 2 mA 0 Cos [( 0 )t + 0 ] = + 2 mA 0 Sin ( 0 + )t + 0 = A 0 Sin ( 0 t + 0 ) + 2 2 +

47
mA 0 Sin ( 0 )t + 0 + 2 2
Vector Diagram
A0 mA0/2
0-
0 0+
mA0/2
A0 A0
mA0/2
A0 0 0
48
Real Signal AM Modulation

Let information signal is non periodic function:
e (t ) =
V
0
( )
Cos ( t ( ) ) d
a ( t ) = [ A0 + K e ( t ) ] Sin ( 0 t + 0 ) = 1 = A0 Sin ( 0 t + 0 ) + K
V ( ) C os ( t ( ) ) d Sin ( 0 t + 0 ) 0
1 a ( t ) = A0 Sin ( 0 t + 0 ) + K V ( ) C os ( t ( ) )Sin ( 0 t + 0 ) d = 0 1 V( ) = A0 Sin ( 0 t + 0 ) + K 2 Sin ( 0 + ) t + 0 ( ) d +

0
+K
49
V( ) 2
Sin ( 0 )t + 0 + ( ) d
AM Modulator
a ( t ) = A 0 [ + mSin 1
t +
)]
Sin (
t +
DC
+
Gen
Multiplier (Balanced Modulator)
A 0 Sin ( 0 t + 0 )
50
One Transistor AM Modulator
HereVin,1 is Vc and Vin,2 is Va
V out = V c Y 21 R L Y 21 = K I e = K e V a
51
V out = K e R L V c V a
Circuit Operation
Carrier amplitude is 10 mVP and Modulation signal amplitude is 6 VP. KV = rC/re. rc = RLRC = 10 k 2k = 1667 . re = 25 mV/Ie
I e re Kv
Vb = Vcc
R2 = 10V R1 + R2
Vb Vbe Ie = ( Rb ) + Re
Vb = VCC
R1 R1 + R2
Rb = R1 R2 = 6667
For = 100 and Vbe = 0,7V I e = 0.924 mA Therefore , re = 27 KV = 61.7 For carrier Vout = KV Vin = 61.7 0,01V = 0.617VP Vbias = Vb Va Sin a t = 10 6 Sin 2 1000t
+
52
Circuit Operation (continuation)
Sin 2 1000t = 1, Vbias = Vb 6( 1) = 10 + 6 = 16V I e = 1.52mA re = 16.45 KV = K max = 101.3 Vout = 1.013V p
V = 1.013 0.617 = 0.396 V
Sin 2 1000t = +1, Vbias = Vb 6(1) = 10 6 = 4V I e = 0,328mA re = 76.3 KV = K min = 21.9 Vout = 0.219V p
V = 0.617 0.219 = 0.398 V
Vmax Vmin m= = 0.645 = 64.5% Vmax + Vmin KV = KV , 0 (1 + mSint )

53
KV , PP = KV , 0 (1 + m )
For m = 1 KV ,max = 2 KV , 0 and KV ,min = 0
Double-Sideband Modulation (DSB)

a (t ) = = A Sin Sin A 0 2
t + (
0
)
+
Sin
( +
0
t + 2
0
) = +
A 0 2 +
)t
Sin
)t
+ 2
Multiplier (Balanced Modulator) Gen
A 0 Sin ( 0 t + 0 )
54
DSB example for transmitting two color (different) signals
ES = EI cos c t + EQ sin c t = EI 2 + EQ 2 sin(c t + ) EI = arctg EQ
55
DSB waveforms in key box modulator

a b
Diode rectifier
LPF
56
Sin
Single-Sideband Modulation (SSB)

1 [Cos ( ) Cos ( + )] 2 1 [Cos ( ) + Cos ( + )] Cos Cos = 2 Sin Sin =
}
Cos(-) Cos (+)
Sin Sin
m(t)
Cos Cos
57
1 a(t ) = A0 [am (t ) Sin 0t m am (t ) Cos 0t ] 2 1 am (t ) a (t ) = d t
Hilbert Transform and SSB

H(f)
1
H(f) f
-900
H()
+900
H ( f ) = jSgn ( f )
.
If f 0 , Sgn ( f ) = 1 ; If f 0 , Sgn ( f ) = 1
a ( t ) = A[Cos ( t + ) + jSin ( t + )] = A[Q ( t ) + jI ( t ) ] A (t ) = A Q 2 (t ) + I 2 (t )
= arctg
I (t ) Q (t )
. H a ( t ) = A[ jCos ( t + ) + Sin ( t + )] =
58
( j )
= A Cos t + + jSin t + 2 2
SSB with Real Information Signal
m(t)
H
m(t)
Sin
Cos K()
()
1 a(t ) = A0 [am (t ) Sin 0t m am (t ) Cos 0t ] 2 1 am (t ) a (t ) = d H t

This is convolution integral with h(t) = 1/t, where h(t) is impulse response of H(f) system.
59
See: M. S. Roden, Digital Communication System, Prentice-Hall, 1988, P. 331.
res 0
R C
3 dB =
1 CR
= arctg
3 dB
3 dB
= 45 0
SSB Power AM
(t ) = P0 1+ 0,5m 2 P
SSB with filter method SSB with Hilbert method
PSSB(F)=P0 m2/4 PSSB=P0 m2/2
For sinusoidal information signal with m = 1, G = P(t)/PSSB = 3; For real signal if m = 0,45, G = 11
2 mmax + 2 G= 2 mmax
More precise formula:
If = 3, G = 10
60
Usually mmax = 1 then G = 2 + 1
S/N SSB and DSB Comparison
2V
N
SSB
DSB
S = ( N S = ( N
2V 2 ) N 2V )2 2 N
Noise is summed up in quadratic way Signal is summed up in linear way
In the same conditions S/N in SSB equal to S/N in DSB
61
All amplitude modulation systems have a peek factor (maximal to average ratio) more then one. These transmitter efficiency is not more then 30%.
Frequency Modulation
FM Signal Spectrum
= A0 Cos [ Sin ( t + )]Cos ( 0 t + 0 ) A0 Sin [ Sin ( t + )]Sin ( 0 t + 0 ) {A0 Cos [ Sin ( t + )]} J n ( ) = J n ( )
for odd n
a (t ) = A0 Cos [ 0 t + Sin ( t + ) + 0 ] =
and
{A0 Sin [ Sin ( t + )]}

for even n
J n ( ) = J n ( )
a (t ) = A0 Cos ( 0 t + 0 ) J 0 ( ) + 2 J 2 ( )Cos 2 ( t + ) + 2 J 4 ( )Cos 4 ( t + ) + ... +
+ A0 Cos 0 t + 0 + 2 J 1( ) Sin ( t + ) + 2 J 3 ( ) Sin 3( t + ) + ... 2
62
FM and PM Spectral Components
63
FM and PM Spectral Components

Carrier and side components amplitudes in % of non modulated carrier.
Bandwidth Phase shift
radian
64
Spectrums with the same fm(A) and the same f(B)

A B
65
Spectrums with different f
For 0,4
F = 2fm F = 2(1 + + ) 2 f
FM signal Bandwidth:
For > 1
66
Vector Diagram
67
Modulation by two sinusoidal signal
v(t ) = V0Cos( 0t + 1Sin m1t + 2 Sin m 2t ) = = ReV0e e e

j ( 0t + 1Sin m1t + 2 Sin m 2t ) j ( 1Sin m1t ) j ( 2 Sin m 2t )
= p = J p ( 1 )e
jp m1t jr 2t j ( 0 + p m1 + r m 2 )t
= r = J r ( 2 )e

v(t ) = Re V0 p ,r = J p ( 1 ) J r ( 2 )e
= V0 p ,r = J p ( 1 ) J r ( 2 )Cos( 0 + p m1 + r m 2 )t
68
@
Draw the spectrums for three cases: A. modulation Index 1 = 2 and modulation frequency m; B. modulation Index 2 = 1 and modulation frequency 2m,; C. modulation by two sinusoidal signals with frequencies m (1 = 2) and 2m (2 = 1)
Symmetrical spectrum
m2 = 2m1
Symmetrical spectrum
69
Modulation by two sinusoidal signal
J 0 ( 1 ) J 0 ( 2 ) = + 0 ,18
m p = 1, r = 0 p = 1, r = 1 p = 3, r = 1 p = 3, r = 2 J 1( 1 ) J 0 ( 2 ) = 0 , 441 J 1( 1 ) J 1( 2 ) = 0 , 254 J 3 ( 1 ) J 1( 2 ) = 0 , 057 J 3 ( 1 ) J 2 ( 2 ) = 0 , 015
+m p = 1, r = 0 p = 1, r = 1 p = 3, r = 1 p = 3, r = 2 J 1( 1 ) J 0 ( 2 ) = 0 , 441 J 1( 1 ) J 1( 2 ) = 0 , 254 J 3 ( 1 ) J 1( 2 ) = 0 , 057 J 3 ( 1 ) J 2 ( 2 ) = 0 , 015
= 0 ,738
70
= 0 ,116
Non symmetrical spectrum!
Signal and Noise in FM

Mod FM Demod FM
S/N > 1, N < S
f Const, N 1/fm
vN = Kd f
Noise
Noise spectrum in FM has a triangular form.
Preemphasis
Modulator HPF Demodulator LPF
signal
sign
al
LPF
signal
Vout 3 = 71 2 f m V in
a rctg 2 f m 2 f m
noise
noise
Signal to Noise in FM
Three factors in comparison with AM: (For FM Broadcasting system with f = 75 kHz 75 s) - Noise spectrum in FM due to triangular form - Preemphasis gives 13,2 dB, -Increasing with the same fm gives 20lg = 14 dB. and Preemphasis with = gives 4,8 dB,
S/Nout
Choosing
Vin,min
72
Frequency Modulator and oscillator Direct method
Inf.
Carrier oscillator
C1
R1
L1
C0 If C1 0.1 0.4 pF it possible to frequency change to some %.
73
Monolithic Crystal Filters

ZZ XX YY ZZ is optical axis, XX is electrical axis, YY is mechanical axis
Quartz-crystal
C1
R1
L1
0 .3 1 % s
Q 10000 L1 5 20 mHn C1 0.001 0.0001 pF
74
C0
R1 = 0 = 1
2 s
11 1 = + L1C1 L1 C1 C0
2 p
Monolithic Crystal Filters Examples
75
SSB Crystal Filters
76
Two Types IF Filter Comparison

dB
kHz
One of possible solution

77
Method for < 1

J 0 ( ) = 0 . 9975 << 1, and J 1 ( ) = J 1 ( ) If Then v ( t ) = V 0 J 0 Cos t + J 1 ( ) = 0 . 049
1 1 V 0 J 1Cos ( + ) t V 0 J 1Cos ( ) t 2 2
DSB
Inf.
AM
78
FM Osc.
900
Calculated Method
FM
Inf.
A/D
FFT
Spectrum Calculating
FFT-1
D/A
Bessel Functions
79
4. Digital Modulations and Modulators

Orthogonality
- Each signal vectors in n-dimension space are perpendicular to all another vectors or all vectors projection on one vector equals zero. - Correlation coefficient between each two signals equals zero. - Inner production of each two signals equals zero. - Impulse response of all previous signals equals zero in appearance following impulse. - Each signal it is impossible to get as a sum or difference another signals. These are definitions or main characteristics of orthogonal signals or sequences which corresponded to orthogonality requirements. We can recover desired signal from orthogonal signals group even these signal spectrums are overleaped.
80
Signal Space
Any signals
y e2
Space of signals in time 0,T by sight:

y
d
S = A sin (k0t + )
S1 S2
e2
A1 sin 1
e1 x 2 1 e1
{0 2 }, A > 0
x
A1 cos 1
0 = 2 T , k = const.
S1 = A1 sin (k0t + 1 ), S 2 = A2 sin (k0t + 2 ) 1 2 e A, E = e T 2 2 d 2 = e12 + e2 2e1e2 cos( 2 1 ) d 2 = Ex + E y 2 Ex E y

T
= 0, if x(t ) y (t ) = 0 or e1e2 cos(1 2 ) = 0

Correlation coefficient between two signals equals zero if inner production of two signals equals zero. ASint and Bcost are orthogonal signals.
=
0
x(t ) y (t ) dt Ex E y
x y x y x y
81
Correlation Coefficient
( x(t ), y (t ) )
x(t ) y (t )
T
x(t ) y (t )dt
0
1 -1 =1 when x(t)=ky(t)
(k>0)
Ex Ey
y
In vector presentation
x y = cos = xy
82
Orthogonal signals examples

Sin. and Cos.
y e2 d
A1 sin 1
2 1 e1 x
A1 cos1
10 BPSK
A1 sin 1 A1 sin 1
QPSK
d
11
e
A1 cos 1
e e e e
00
A1 cos 1
e=
83
01
d = 2 E = 2 PT
e= E
Ts = 2T
d = 2 E = 2 P 2T = 2 PT
Increasing M in MPSK
BPSK QPSK 8PSK
d =2 E RS = Rb = ES = 1 T
P = PT RS
d= 2 E R 1 RS = b = 2 2T P ES = = 2PT RS dS = 2 ES = 2 PT
d = 0.765 E Rb 1 = 3 3T P ES = = 3PT RS RS = dS =1.33 PT
dS =2 ES =2 PT
1000 0000 0001dmin Interference influence 1001 0011 dinf 1011 0010 1010 1111
84
16PSK
0110 0111 0101
1101 1100 01000100
FT = 2 FT = 1
With help the band-pass filter, which support needed pulse shaping and with help coherent demodulator, we can get F = Rb without noticeable BER increasing.
FT = 2 includes 94.8% of he whole of power. FT = 1 includes 90.2% of he whole of power.
85
Division methods
FDMA In this case information is divided corresponding to frequency. As usual, it
is impossible to give the nearest frequency for the nearest cell, because each signal has an additional radiation in neighboring frequencies (side lobes, not only main lobe). Orthogonal method (OFDM) can not help in cellular system today, because Doppler Shift can change the frequency difference between two neighboring signals.
TDMA In this case information is divided in time. In other words, several signals are
transmitting at the same frequency one after another. Transmitting velocity in this frequency is increasing, but this increase needs wider frequency band. Disadvantages for cellular system are: - bigger influence of reflection signals, -problems with multiplexing of signals from different places.
CDMA In this case group of signals is transmitting in the same frequency band and at the same
time. The signals are divided corresponding to spreading code. CDMA systems have a two advantages: better frequency efficiency and soft handover. But another communication signals in this and in neighboring cells effect decoding precision of the useful signal.
86
Multi Path Propagation Problems Solutions

(TS - symbol duration, - reflected signal delay.) TS TS >> TS <<
TDMA + Equalizing
TS TS
OFDM
CDMA
TS
87

OFDM signal
f1 =
OFDM
1 k T
f2 =
1 ( k + 1) T
fcifci+1
1 t + 1] T 1 S 2 = A 2 sin[ 2 ( k + 1 ) t + 2] T Let us A1 = A 2 = 1 1 = S 1 = A 1 sin[ 2 k

T
= 0
[sin
2 k
1 t ][sin T
2 ( k + 1)
1 t ] dt = 0 T
2/T S(f)
1/T
f1
f2
f1
88
f2
f = 1/T
MPP and OFDM Signal

E
M 1
S OFDM =
i=0
Ai Sin [ 2
1 ( k + i )t + i ] T
fci fci+1
For decreasing Multipath propagation (MPP) influence, each symbol includes Guard interval wit duration TG
TG TG
89
OFDM Signal
QPSK in TDMA QPSK in FDMA E QPSK in OFDM
FT = 1
E E
FT = 2
F In the QPSK (FT=2) in FDMA system the usage is 1bit/s/Hz also.
F In the QPSK (FT=2) in OFDM the usage is
In case of QPSK we have from 1bit/s per Hz till 2bit/s per Hz. It depends on band pass filter, pulls shaping and decoding method.
2 bit/s/Hz.
We can recover desired signal from orthogonal signals group even these signal spectrums are overleaped. .
90
Modulator and Demodulator in OFDMA is FFT processor. See appendix 4.
OFDMA Modem
Multiplexer \demultiple xer
Out. ECC
Interleaver
In. ECC
Interleaver
Spectrum creating
IFFT
DAC
Framing
Amplifier
LNA
ADC
FFT
Deinterleav.
In. ECC dec.
Deinterleav. 91
Out. ECC dec
Demultiplexer
Amplifier
Explanation about OFDMA spectrum creation.

OFDM modulation starts with constructing orthogonal spectrum, which consist of L complex components, where L is two in a power of any number . Let us L = 4 (see A on picture). Matlab commands in this stage will be: M = 2 BPSK; M = 4 QPSK; M = 16 16QAM First component in the real signal is not a DC. In order to use the Matlab IFFT command, the spectrum must satisfy the following conditions: Symmetry, In the middle and in the beginning must be zeroes, Number of component must be L power of two. In accordance with above conditions we built a spectrum with range N = 4L with following structure: 0, [L complex numbers], [2L 1 zeroes], [L conjugate numbers] (see B on picture)
f A B
f C
92
Explanation about spectrum creation.
Reminder:
In real OFDMA systems FFT range (N/2) is more than carriers number. So, there are zeroes before and after information carriers
93
Explanation about spectrum creation.
After FFT from receiving signal, we have to get N symmetrical conjugate spectral component, including our L initial components.
Reminder:
ACos ( t ) pair of two conjugate numbers 1 1 f (t ) = Ae j ( t ) + A e j ( t ) = 2 2 1 1 A[Cos ( t ) + jSin ( t )] + A[Cos ( t + ) + jSin ( t + )] = 2 2 ACos ( t )
In decoder we make FFT and receive N conjugate numbers.
After that we can get initial L spectral components. Matlab commands in this part will be: 94
FBS method
A similar idea has been used to exclude Pilot signals in OFDMA - method FBS [9]. Besides PAL phase compensation method Walsh functions is used here. For example fours order Walsh function seems like this (Walsh Adamar Matrix - WAM): 1 1 1 1 1 1 -1 1
Walsh function are orthogonal sequences. This feature is used in CDMA system for transmitting on the same frequencies some signals with opportunity to extract needed signal.
1 -1 -1
1 -1
1 -1 -1
OFDMA The main idea of FBS method it is possible to show on this easy example. Let us we have OFDMA signal with MPSK modulation. We can take for carriers with phases 1 2 3 4 f
95 95
1 2 3 4
We can transmit signal with phase 1 four times on four frequencies. It is possible to say? That it is first Walsh serial in previous example. But on the same four frequencies we can transmit second signal using second Walsh serial
1 1 1 1 f
2 2 -2 -2
f 3 -3 3 -3 f 4 -4 -4 4 f
Similarly we will use for third signal third serial and for fours signal fours serial/
So FBS method allows to transmit the same bit number, in the same frequency band and with the same power like in OFDMA system. For receiving needed signal for example third we must change signs of summing signal correspond to third signal and sum all four phases. We will take following phase constellation.
96 96
1 -1 1 -1 f
2 -2 -2 2
f 3 3 3 3 f 4 4 -4 -4 f
The main advantage of FBS method like PAL system is compensation fast phase changing in channel. It is proposed FBS 1 method for MPSK modulation and FBS 2 for MQAM modulation. In case of FBS 2 signals I and Q are transmitted separately on different Walsh series. It is possible to say, that in FBS methods are transmitted together source signals, opposite signals, conjugate signals and opposite conjugate signals.
97 97
Resulting phases sum equals 43
About FBS see part OFDMA problems and solutions
cos( ) = cos cos m sin sin

2 , m
Digital I-Q modulation

QPSK
In communication system sinusoidal signal is presented frequently like cos not like sin.
inf = r
r = 0,1,... m 1
2 u ( t ) = A cos ( k 0 t + ) + r 4 m
2 2 ) A sin( k 0 t ) sin( ) = +r +r u ( t ) = A cos( k 0 t ) cos( 4 4 m m = I cos k 0 t Q sin k 0 t I = A cos(
+r
2 ) m
Q = A sin(
+r
2 ) m
Or:
u ( t ) = Q cos k 0 t I sin k 0 t Q = A cos(
+r
2 ) m
I = A sin(
+r
2 ) m
Another explanation of sign minus in QPSK.
I-Q modulation in 8PSK
u (t ) = Re g (t ) e jk 0 t g(t) = x k + jy k
u (t ) = Re [(x k + jy k )( cos k 0 t + jsin k 0 t ] = = x k cos k 0 t y k sin k 0 t For x k = + 0 .707 , g k = + 0 .707
e jk 0 t = cos k 0 t + jsin k 0 t
u (t ) = 0 .707 A cos k 0 t 0 .707 A sin k 0 t = Acos( k 0 t + 98
Pulse shaping filter
IQ Modulator
A sin 2f 0t
I(t)
+
Q(t)
s m (t)
99
A cos 2 f 0 t
QPSK modulator
I sin Q cos
From W. Tomasi.Electronic Communications Systems, Prenttice Hall, 1988.
I-Q modulator
I cos Q sin
100 From M.K. SimonDigital Communication Techniques, Prenttice Hall, 1995.
Minimum Shift Keying (MSK)

In digital modulation, minimum-shift keying (MSK) is a type of continuous-phase frequency-shift keyng that was developed in the late 1950s and 1960s.[MSK is encoded with bits alternating between quadature components, with the Q component delayed by half the symbol period. However, instead of square pulses, MSK encodes each bit as a half sinusoid. This results in a constant-modulus signal, which reduces problems caused by non-linear distortion. MSK can also be viewed as a continuous phase frequency shift keyed (CPFSK) signal with a frequency separation of one-half the bit rate. The resulting signal is represented by the formula where aI(t) and aQ(t) encode the even and odd information respectively with a sequence of square pulses of duration 2T. This can be rewritten in a form where the phase and frequency modulation are more obvious, where bk(t) is +1 when aI(t) = aQ(t) and -1 if they are of opposite signs, and k is 0 if aI(t) is 1, and otherwise. Therefore, the signal is modulated in frequency and phase, and the phase continuously and linearly changes.
101
Power Spectral Density

2
NRZ
sin fT S P ( f ) = V2T fT
BPSK
sin[( f f c )T ] S P ( f ) = A2T ( f f c )T
2
QPSK
sin[( f f c ) 2T ] S P ( f ) = A 2T = ( f f c ) 2T
2
sin[( f f c )Ts ] = A Ts ( f f c )Ts

2
102
QAM E
16QAM
dmin dinf
16PSK
dmin 16 QAM > dmin 16 PSK
f F 2
16QAM
d = 0 .632 E R 1 RS = b = 3 4T P ES = = 4 PT RS d S = 1 .265 PT
d = 0 .39 E Rb 1 = 3 4T P ES = = 4 PT RS RS = d S = 0 .78 PT
103
Gray-Coded Mapping
104
QAM I-Q modulation

r 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 105
Ar cos( r
3 1 3 1 -1 -1 -3 -3 -3 -1 -3 -1 1 1 3 3
2 ) m
Ar sin( r
2 ) m
r
1 2 2 3 4 5 5 6 7 8 8 9 10 11 11 12 9 11 8 6 7 5
1 1 3 3 3 1 3 1 -1 -1 -3 -3 -3 -1 -3 -1
Consider the 16QAM signals: I

+3 +1 -1
10 12 4 2 3 r=1
Q
+1 +3
14 13 16 15
-3
-1 -3
Q
u (t ) = Ar cos( r
2 2 ) cos t Ar sin( r ) sin t m m
BER calculations
Base band signals

As usual we have in baseband system signal to noise ratio in form:
20 log x=
dB = V
rms
2 erfc 2
erfc( x)
Example:
p (e) =
10 log x=
Ps V = 20 log = 10 dB PN
3 .2 = 2 .26 2 erfc ( x ) = 1 .4 10 3
p ( e ) = 0 .7 10 3
There is approximate formula:
1 p (e) = exp( x 2 ) 6
In our example: 106
p (e) =
1 exp( 2 .26 2 ) = 10 3 6
The possible approach for BER calculation

20 log x= V
dB = V
rms
2 erfc 2
erfc( x)
p (e ) =
- Let we transmit BPSK or QPSK signal and we know on receiver input signal and noise powers. Doppler effect, Multipath propagation and interferences are absence. - We can calculate Eb/N0 (Eb is one bit energy, N0 is power spectral density of noise). - Let we have ideal decoder. - Now we can calculate BER: Example:
Eb E dB b N0 N0
x=
Eb N0
erfc( x) erfc( x) BER = 2
Eb = 9.6dB N0 BER = 1 10 5
x = 10
0.1
Eb dB N0
= 9.12 = 3.02
erfc(3.02) = 2 10 5
- If we implement another modulation, we can know needed power signal changing for receiving the same BER (see next slide ) - If we implement non ideal decoder we can know needed power 107 signal increasing for receiving the same BER.
Comparison with the same BER (10-5)
In previous slide we received value of BER correspond to Eb/N0. So for QPSK and Eb/N0 = 9.6dB, BER = 10-5. For another type of modulation the same BER will be more by xdB, where x = 10log (dQPSK/danother)2.
QPSK 90
-1
PT
15 0.13
2T
P*2T
(2 0.13)
= 237
24 24
0dB 9.6dB 1
33, dB 27.6
108
I-Q modulator by Alters FPGA
Asynchronous Serial Interface, or ASI, is streaming data format which often carries an MPEG Transport Stream In telecommunication, trellis modulation (also known as trellis coded modulation, or simply TCM) is a modulation scheme which allows highly efficient transmission of information over band-limited channels. Interpolation filter is filter bank for pulse shaping filter 109
110
111
112
113
114
5. UP Down Conversion, Superheterodyne and

Homodyne Receiving
115
Superheterodyne Receiver
116
Image frequency problems
If fLO > fRF , fIM = fLO + fIF = fRF + 2fIF
Image Frequency Rejection Ratio (IFRR) is where: Q is quality factor of RF circuit, k = fRF/fIF
117
IFRR = 1+ Q k
2 2
Receiver with double frequency conversion
118
Double frequency conversion example
f RF = f S = 100 MHz f IF 1 = f LO1 f S = 10 MHz f IF 2 = f IF 1 f LO 2 = 1MHz
f LO1 = 110 MHz
f LO 2 = 9 MHz
1 f IM 2 = 8 MHz
f IM 1 = f S + 2 f IF 1 = 120 MHz
1 1 f IM 2 = f LO 2 f IF 2
1 f IM 2 = f LO1 f IM 2 = 102 MHz 1 f IM 2 2 = f LO1 + f IM 2 = 118 MHz
Influences of: IF1 Filter RF Filter
fIF2
119
fIM1 fLO2 fIF1
fS fIM2 fLO1 fIM2-2 fIM1
IF2 Filter
IF1 Filter
RF Filter
@
FS
Frequency converter
Mixer - Multiplier Filter FS FLO or FLO - FS
LO
FLO
Balance Multiplier
120
Why it is indeed Multiplier ?

Z
Vout = R (I 5 + I 7 ) R (I 6 + I 8 ) = R (I 5 + I 7 I 6 I 8 ) 1. V5,9 = 0 V 3 ,11 = 0 V out = 0 00 = 0 I5 = I6 = I7 = I8
5 9
X
3
11
2.
I 5 = I 8 = I (1 + ) V out = 0 x0 = 0
V5,9 0
V 3 ,11 = 0
I 6 = I 7 = I (1 )
3.
I 5 = I (1 + )(1 + I 7 = I (1 )(1
121
V 5 ,9 0
V 3 ,11 0
) )
I 8 = I (1 + )(1
I 6 = I (1 )(1 +
) )
V out = 4 IR
x y = z
Real Multiplier Example (MC1596G)
122
Real Multiplier connection
MC1596G
Real Intermediate receiver
Out In
123
Modern Intermediate receiver
124
See Appendix 5
Direct Conversion, Homodyne receiver of Digital information

Problems: Image rejections Silicon internalization
Low-pass filter
The signal is first amplified at a low noise stage and then directly converted to the baseband or even to a direct current signal. When the frequencies of the RF and the LO signals are equal, this scheme works as a phase detector. In some literature, only when the local oscillator is synchronized in phase with the incoming carrier frequency, the receiver is called homodyne. 125
Band select filter LNA LO LO +900 Low-pass filter
DC Q
Band select filter
Low-pass Filter
To achieve maximum information, we should take both parts of signal. Its done by a method, which is called quadrature downconversion. The principle of this method is that the signal is at first divided into two channels and then downconverted by an LO signal, which has a phase shift of 90o in one channel with respect to another. The vector of the resulting signal is described as:
Signal =
I +Q
2
arg( Signal ) = = arctg
Q I
The main problem in homodyne technique is an offset caused by the LO signal leakage to the RF port of the mixer. The propagated signal reflects from the components in the front-end of the receiver and goes back to the mixer, where it is mixed down to DC. The offset can be considerable with respect to the signals to be measured. This leads to a narrower dynamic range of the electronics, because the active components get saturated easier than it would be in case of a zero offset. For example, lets take a mixer with the LO-drive equal to 1V and RF/LO isolation equal to 40 dB. In this case the offset can be as high as 10mV. Isolation of LNA can be or about 40 dB too. In case of a high sensitivity this can be a large number (this is the signal level at the output of the mixer, some amplification stages follow! See picture).
Band select filter Smin 1V Low-pass filter DC
Problems: Synphase signal influence DC offset returning
LNA 10mV 0.1mV 126 1V

LO
Nonlinearity
Another problem of the homodyne receiver, or, more concretely, of the I/Q (in-phase/quadrature) mixer, is mismatches in its branches. Assuming a mismatch of for the amplitude () and for the phase (), we can estimate the error, caused by these mismatches. In this way we get:
E IQ =
S ideal S miss 1 + 2 S ideal 2

I
For typical values of and this gives an error of .
a cos t
a (1 + ) cos( t + )
b cos( t + )
b sin( t + )
Q
In the processing of the BPSK signals it is important to know the phase with respect to the phase of the reference cavity. That means that an I/Q mixer becomes very attractive for this application. On the other side, it has too many specific hardware problems. Thats why it is desirable to apply DSP method In that case an I/Q mixer and other elements realized in forms of idealized models have no mismatches or reflections etc. Surely, the initial signal still has to be downconverted and prepared for the digitizing.
127
Direct Upconversion
1 sin sin = [cos( ) cos( + )] 2 1 cos cos = [cos( ) + cos( + )] 2
Transmitter
128
Problems Solutions
Problems: Synphase signal influence DC offset returning Nonlinearity
There are three ideas: Differential (balanced) shims Direct receiver, but fRF fLO , for example fRF = 2fLO and mixer reproduces fLO (Sub-Harmonic Mixer) Quadrature LO
129
130
131
Directly amplifying radio receiver
132
Reminding: In telecommunication, a direct-conversion receiver (DCR), also known as homodyne, synchrodyne, or zero-IF receiver, is a radio receiver design that demodulates the incoming signal by mixing it with a local oscillator signal synchronized 133 in frequency to the carrier of the wanted signal. The wanted modulation signal is obtained immediately by low-pass filtering the mixer output, without requiring further detection.
6. Receivers
Propagation Equation
Friis transmission formula:
PR 2G T G R = PT (4 d ) 2
Pt G t G r (4 d )2
In far-field region (d > ):

EV = ka Prad d
Input receiver Power is:
PR =
Received power is proportional to 1/d2 , or path loss is 20 dB/dec
Pr , dBm = 10 lg Pr , mWt
ERP = Pt G
t
Let us define the field density in receiving place.

From Poynting`s theorem we can find power density
Pd =
ERP 4 d
P d = EH
E 2 = = 120 H 120
P R = Pd A e f , i s G R A e f ,is =
Another case:
E2 = 120 H 2 Pd = EH = 120
Aef ,m =
See Balanis
Aef,is is effective isotropic aperture Aef,m is maximum effective aperture D is antenna directivity, for isotropic antenna D = 1
So : or E
E mV
=
m )
30 ERP d
m
wt
= 173
dB ( kwt )
ERP d km
kwt
134
dB ( V
= 104 . 8 + ERP
20 log d
ERP is Effective Radiated Power

km
Antenna in mobile device

Some of antenna parameters
-Radiation Resistance
R rad
P rad = 2 I rms
-Antenna Impedance
Vin Za = = Ra + jX a = Rrad + Rloss + jX a I in
- Gain (G) is a ratio of the maximal radiation intensity from the specified antenna to the radiation intensity from a loss-free isotropic antenna with the same input power (Pin). - Directivity (D) is a ratio of the maximal radiation intensity from the specified antenna to the radiation intensity from a isotropic antenna with the same radiation power (Prad). -Radiation Efficiency
G a = D
0 < a < 1
135
Some of antenna parameters (continuation)
-Effective height (he(m)) is a ratio of source voltage of antenna equivalent (VV) to field density (EV/m). For example:
Za
V = E* he
SI = SI
h
he
136
@
Dipole and Monopole Parameters in Open Space Dipole: h h Monopole: h
Dipole h = /4: Rrad = Ra =73.2, G = 2.15 dB, he = /. Dipole h << /4: Rrad 20(mh)2, G = 1.76 dB, he = h. Monopole h = /4 in open space: Rrad 52, he = /2. Monopole h << /4 in open space: Rrad 14(mh)2, he = h/2.
137
m(= ) =
137
Monopole h = /4 above ground in transmitter: Rrad 36, G = 5.15 dB, he = /2. Monopole h << /4 above ground in transmitter: Rrad 10(mh)2, G = 4.76 dB, he = h/2. Monopole h = /4 above ground in receiver: Rrad 36, G = -1.15 dB, he = /2. Monopole h << /4 above ground in receiver: Rrad 10(mh)2, G = -2.24 dB, he = h/2. In case A: Dipole GT * GR ? Dipole: GT * GR Monopole: (GT+3dB) * (GT-3dB) = GT * GR
In case B: Monopole -
138
Conclusion: pass from dipole in free space to vertical monopole above ground simultaneously in transmitter and in receiver does not change power balance. 138
*
he = S N
1. 2. 3. 4.
Magnetic (frame) antenna Effective height (he(m)) of frame antenna is:

2 SN
where
If antenna dimensions are less than , he of frame antenna is less than he of monopole. But magnetic antenna has some advantages:
is frame square is coil number
Small dimensions, It has non isotropic pattern, It is possible to increase he, if antenna is part of resonance circuit, then: It is possible to increase he, if antenna has a ferrite core, then: where is relative magnetic penetration
he = he =
2 SN 2 SN
Q Q
139
139
Perpetual resonance
Field Source
L sh = RH = C where L sh is shield
inductivit y
Shield Frame for Field Generator calibration
E =
RH = 150
Antenna
V 2 QS
= 150
RH = 150
Generator
Eg
140
Field Generator
Sensitivity
Z = R + jX
Noise Parameters
Z
EN,R
2 N ,R
= 4 KTR
f
E
2 N ,R
Y = G + jB IN,R
2 N ,R
= 4 KTG f =
V in , N
G = 1/R
K = 1,37*10-23 J/K
Noise Spectral Density
If T = 293o, KT = 400*10-23 J
141
SN =
@
NF,dB TNK
Low Noise Amplifiers
Mixer
On bipolar transistor
On no cooled transistor Schottky (FET) No cooled parametrical Cooled parametrical (20K) Maser Relic noise (3,4K) F,GHz
re he p os m at
142
142
K band amplifier nonlinear parameters
In telecommunications, standing wave ratio (SWR) is the ratio of the amplitude of a partial standing wave at an antinode (maximum) to the amplitude at an adjacent node (minimum), in an electrical transmission line. The SWR is usually defined as a voltage ratio called the VSWR, for voltage standing wave ratio. For example, the VSWR value 1.2:1 denotes a maximum standing wave amplitude that is 1.2 times greater than the minimum standing wave value. It is also possible to define the SWR in terms of current, resulting in the ISWR, which has the same 143 numerical value. The power standing wave ratio (PSWR) is defined as the square of the VSWR.
0.5W X/Ku-Band Power Amplifier MAAPGM0034
Third-order intercept point
In telecommunications, a third-order intercept point (IP3 or TOI) is a measure for weakly nonlinear systems and devices, for example receivers, linear amplifiers and mixers. It is based on the idea that the device nonlinearity can be modeled using a low-order polynomial, derived by means of Taylor series expansion. The thirdorder intercept point relates nonlinear products caused by the thirdorder nonlinear term to the linearly amplified signal, in contrast to the second-order intercept point that uses second order terms. The intercept point is a purely mathematical concept, and does not correspond to a practically occurring physical power level. In many cases, it lies beyond the damage threshold of the device. The intercept point is obtained graphically by plotting the output power versus the input power both on logarithmic scales (e.g., decibels). Two curves are drawn; one for the linearly amplified signal at an input tone frequency, one for a nonlinear product. On a logarithmic scale, the function xn translates into a straight line with slope of n. Therefore, the linearly amplified signal will exhibit a slope of 1. A third-order nonlinear product will increase by 3 dB in power when the input power is raised by 1 dB.
145
http://en.wikipedia.org/wiki/Third-order_intercept_point
*
Both curves are extended with straight lines of slope 1 and n (3 for a third-order intercept point). The point where the curves intersect is the intercept point. It can be read off from the input or output power axis, leading to input or output intercept point, respectively (IIP3/OIP3). Input and output intercept point differ by the small-signal gain of the device.
The third-order intercept point (TOI) is a property of the device transfer function O (see diagram). This transfer function relates the output signal voltage level to the input signal voltage level. We assume a linear device having a transfer function whose small signal form may be expressed in terms of a power series containing only odd terms, making the transfer function an odd function of input signal voltage, i.e., O[s(t)] = O[s(t)]. Where the signals passing through the actual device are modulated sinusoidal voltage waveforms (e.g., RF amplifier), device nonlinearities can be expressed in terms of how they affect individual sinusoidal signal components.
146
*
For example, say the input voltage signal is the sine wave and the device transfer function produces an output of the form where G is the amplifier gain and D3 is cubic distortion. We may substitute the first equation into the second and, using the trigonometric identity we obtain the device output voltage waveform as
The output waveform contains the original waveform, cos(t), plus a new harmonic term, cos(3t), the third-order. The coefficient of the cos(t) harmonic has two terms, one that varies linearly with V and one that varies with the cube of V. In fact, the coefficient of cos(t) has nearly the same form as the transfer function, except for the factor on the cubic term. In other words, as signal level V is increased, the level of the cos(t) term in the output eventually levels off, similar to how the transfer function levels off. Of course, the coefficients of the higher-order harmonics will increase (with increasing V) as the coefficient of the cos(t) term levels off (the power has to go somewhere).
147
@
Transformer in RF stage
1:n R1 R2 = R1*n2
n:1 RL RL
KV y21RL
148
y21RL n KV = y21RLn n
2
Noise Figure
Noise figure (NF) is ratio between output noise power in real stage and output noise power in ideal stage, if noise source in ideal stage is noise of resistor, which sees stage input. Ideal stage in this definition is the same stage, but all elements including transistor do not give a noise.
N F ,Rec
NF ,2 1 q2 N F ,3 1 q3 = N F ,1 + + + K P,1 q1 KP,1KP,2 q2
- noise figures of stages 1,2 - power amplification of stages 1,2 - mismatch coefficient between transistor and resonance circuit
NF ,1,2,3... KP,1,2,3... q1, q2 ...
If KP,1>>10, then NF = NF,1

149
TN = T(NF-1)
Receiver Antenna system sensitivity
E min( V / m )
Where K is Bolzman constant, T is Kelvin temperature, D is system parameters influence B is receiver stages till detector bandwidth, Ra is antenna impedance or impedance which sees receiver first stage, NF,V is noise Figure of receiver first stage with respect to voltage.
2 VV
1 = D 4 KTBR a N F ,V he
PAV
4 Ra
= D2
KTBNF , P
Ra does not effect maximal available power.
For receiver with FET transistor the condition for better sensitivity is maximal value of he/Ra.
150
@
AM Receiver
EV =
m
4 KT fRa N F PS out PN mhe
PS 23 If out = 20dB; m = 0,3; KT = 400 10 , then PN E V

m
4,2 = he ( m )
m
f ( kHz ) Ra ( k ) N F
he is effective (radiation,
virtual) height
Vin ( V ) = EV he ( m )
f 151
is a minimum of frequency band after detector and half frequency band before decoder
@
FM Receiver
PS out f ( m , ) PN h e ( m ) m max 3
E V
m
4 KT f IF ( kHz ) R a ( k ) N F
f ( m , ) If then
1 = m
arctg m 1 3 m
PS out = 26 dB ; m = 1; PN E V
m m
f = 75 kHz ; = 75 s ; Fm = 15 kHz
0,2 = he ( m ) he ( m )
f IF ( kHz ) R a ( k ) N F
V V = E V
152
Exponential Modulation
Noise in FM
Mod FM Demod FM
S/N > 1,
N < S
f Const, N 1/fm
vN = Kd f
Mod FM
Demod FM
153
Noise spectrum in FM has a triangular form.
@ *
Signal to Noise in FM Three factors in comparison with AM:

(For FM Broadcasting system with f = 75 kHz Preemphasis with = 75 s) and
- Noise spectrum in FM due to triangular form gives 4,8 dB, - Pre emphasis gives 13,2 dB, -Increasing with the same fm gives 20lg = 14 dB.
154
Choosing
S/Nout F Vin,min
155
@
AM Receiver with Frame antenna
E V
m
f is average frequency,
1 3he ( mm )
f f kHz La ,mHn N F Qa
La and Qa are inductivity and quality of laden antenna (after injecting in receiver), f is a minimum of frequency band after detector and half frequency band before decoder,
he is effective (radiation, virtual) height
156
* In various books and articles it is possible to read the following at first sight contradictory statements.
Receiver input noise voltage is proportional to square root of antenna resistance (Ra), therefore receiver sensitivity is better if Ra is smaller. Noise available power does not depend on antenna resistance.
Where is the truth?

157
@
Possible Dependence between NF and Ra. For receiver with FET transistor the condition for better sensitivity is maximal value of he/Ra.
4 .2 = he n R g fN F 4 .2 = he n
With the help of transformer we can change resisting, but we can not change sensitivity. For example:
R a n fN F
2
E mV
4 .2 = he
R a fN F
But possible NF depends on resistor, which it sees. In this case we do can change the sensitivity the with help of transformer.
158
NF
Rg,opt
Rg
Design Receiver RF part example

You have to design the receiver RF part and to define sensitivity. Frequency range f = 1.8 2.2 MHz, frequency band after detector f = 1kHz, first transistor noise figure NF = 2, load antenna quality Qa = 20, (relative magnetic penetration) of ferrite = 400, 0 (air relative
penetration) = 410-7.
1.
Choosing varicap 10 40 pF. We have f2 /f1 = 1.2, or we need C1/C2 = 1.22 = 1.5, then we implement the following scheme.
C ad + 40 = 1.5, C ad = 50, C ev = C ad + 10
Cad
(10 + 50 ) (40 + 50 ) = 74 pF
Cad2 Cad1
Cascade connection is also possible . This case advantage is varicap voltage decreasing. 2. Calculating L for fev = 2MHz.
f =
159
1 2 L LC
L =
1 = 0 . 09 mHn 2 2 4 f C
L0 =
= 4 . 5 Hn
Design Receiver RF part example (continuation)

3. Calculating coil number N.
2S rP Let S = 1 cm 2 and r = 0 . 2 mm . S = R 2 , R = 0 . 56 cm and P = 2 R = 3 . 5 cm . N = 3 .9 4 2 P ln

4. Calculating effective antenna height without C .
L = 2 PN
ln
2S , rP
L N
2
S is coil square, r is wire radius, P is coil perimeter
so
he =
2 N S mm 2
mm
for
f ev = 2 MHz
= 150 10 3 mm
and
h e = 6 . 7 mm
he = 2 SN
5. Calculating sensitivity .
160
E V
m
1 3 h e ( mm )
f f kHz L a , mHn N Qa
= 54
N = N
V
m
Think about this questions

RL
h=
, N F = N F ,opt = N F , min
+
- increase of amplifying from antenna till detector, - implementation of first transistor with bigger , - implementation of deep feed back in RF amplifier, - implementation of deep feed back in IF amplifier, - increase of antenna height, - putting first transistor in refrigerator, - implementation of several antennas with one transistor, - implementation of some antennas with several transistors.
161
Is it possible to improve receiver sensitivity without making worse remaining receiver parameters by implementing the following techniques?
Feed back does not improve signal to noise ratio

V out R1 R 0 + R1
R0
R1
Vout
Output voltage (with distortions) returns to input with opposite phase, but in input there is signal adding without distortion from Ein. Negative feed back decreases distortion level in amplifier, so it can increase the signal to noise ratio.
Ein
KV =
R0 if Rin > R1 R1
It is not correct in the case if distortions and noise source EN is first transistor input. In this case distortion source short-circuits the R1 . R0 R1
162
EN
Bonus Question
Increasing h of antenna more than /4 does not improve sensitivity.
Why in car window glass one implements antenna with h > /4 ?
163
Selectivity
Spurious Channels Causes:
1. Local oscillator and signal harmonics
f IF = n1 f RF n2 f 2 ... mfLO
2. First stages non linearity f(x) x f(x) x
164
First stages non linearity

To analyze the nonlinear properties of commonly used Taylor series.
f ( x ) = y 0 + k 1 x + k 2 x 2 + k 3 x 3 ... df ( x 0 ) 1 d 2 f ( x0 ) ( x x0 ) + ( x x0 ) 2 + y = y0 + 2 dx dx 2 1 d 3 f ( x0 ) 1 d n f ( x0 ) ( x x 0 ) 3 ... + ( x x0 ) n 6 dx 3 n! dx n 1 d n f ( x0 ) kn = n! dx n
165
First stages non linearity (continuation) x = V1Sin 1t + V2 Sin 2t

Term k1x k2x2 Freq. 1 2 0 2 1 2 2 1 2 1+1/2IF k3x3 1 2 21 2 1 22 31 32 Amplitude k1V1 k1V2 0,5k2(V12+V22) 0,5k2V12 0,5k2V22 k2V1V2 0,75k3V1 (V12+2V22) 0,75k3V2 (V22+2V12) 0,75k3 V12 V2 0,75k3 V1 V22 0,25k3V13 0,25k3V23 Cross Modulation Blocking Effect IP2 2(1+1/2IF)-21 =1F Commentary
IP3 IP3
166
Let us FS = 100 MHz, F1 = 102 MHz F2 = 103 MHz Than 2F1 - F2 = 101MHz near to FS
Whistles
f IF = n1 f RF n 2 f 2 ... mf LO
fRF fLO + fRF fLO +/- 2fRF fLO +/- 3fRF fLO +/- 4fRF fLO +/- 5fRF fLO +/- 6fRF 2fRF 5fRF fLO 5fRF 2fLO 3fRF fLO 3fLO 5fRF 4fRF 2fLO
There are whistles if:
f IF = n 1 f RF mf or f IF = mf
LO
LO RF
nf
2fLO 2fRF 4fRF 3fLO 3fLO 3fRF 4fLO 4fRF
5fLO 6fRF 4fLO 6fRF 5fLO 5fRF 2fLO 4fRF 2fLO 3fRF 3fRF fLO 5fRF 3fLO 6fLO 6fRF 3fRF 6fRF 3fLO 3fLO 6fRF 2fRF fLO 5fRF 4fLO 2fLO 5fRF 5fRF 6fLO 3fLO 4fRF 4fRF 6fRF 2fLO 6fRF 2fLO 6fRF 5fLO 4fRF 5fLO 2fLO-6fRF 6fRF 4fLO 3fRF 2fLO 4fLO 5fRF 2fRF 3fLO
6fRF 5fRF
0.1 167
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1.1
fS/fLO=fRF/fRF+fIF
If we now restrict our attention to that portion of the cos(t) coefficient which varies linearly with V, and then ask ourselves, at what input voltage level, V, will the coefficients of the first and third order terms have equal magnitudes (i.e., where the magnitudes intersect), we find that this happens when which is the Output Third-Order Intercept Point (OTOI). So, we see that the TOI input power level is simply 4/3 times the ratio of the gain and the cubic distortion term in the device transfer function. The smaller the cubic term is in relation to the gain, the more linear the device is and the higher the TOI is, which clearly makes sense. The TOI, being related to the magnitude squared of the input voltage waveform, is a power quantity, typically measured in milliwatts (mW). The TOI is usually well beyond operational power levels, as the input waveform would be severely distorted at this point. The TOI is closely related to the amplifier's "1 dB compression point," which is defined as that point or at which the total coefficient of the cos(t) term is 1 dB below the linear portion of that coefficient. We can relate the 1 dB compression point to the TOI as follows. Since 1 dB = 20 log10 1.122, we may or say, in a voltage sense, that the 1dB compression point occurs when
or or In a power sense (V2 is a power quantity), a factor of 0.10875 corresponds to 9.636 dB, so by this approximate analysis, the 1 dB compression point occurs roughly 9.6 dB below the TOI. Recall: decibel figure = 10 dB log10(power ratio) = 20 dB log10(voltage ratio).
168
Whistles (continuation)
Example: We have a Receiver with fRF = 280MHz and fIF = 70MHz . Are there a whistles? Yes! f RF 280 = = 0.8 4 f RF 3 f LO and 5 f LO 6 f RF f RF + f IF 280 + 70
4 280 3 350 = 70 and 5 350 6 280 = 70

280MHz is not too good !
Notes:
0.7
169
0.8
0.9
1.1
f Rf
f RF + f IF
f RF F RF f IF
Out of band channels
f RF , ad = mf LO f IF
f fRF fRF,ad
f RF , ad >> f RF
f RF
RL
+
170
@ *
Frequency Planning Algorithm

Defining of receiver
selectivity situation Next frequency
Selectivity Parameters measurement
Frequency bands for all selected channels
First frequency choosing
Intermodulation channel (IP3) adding
Frequency bands for first channel
Next frequency choosing is possible?
Yes
Not
The End
171
* Frequency Planning Algorithm (exercise)

1. Defining of Receiver Selectivity situation.
1.1 In frequency band 300 400 MHz you have to choose frequencies for maximum number of communications. 1.2 Receiver has IF1 = 40 MHz, IF2 =1 MHz, bandwidth F = 0.1 MHz. 1.3 RF bandwidth 60 MHz. 1.4 Maximal ration transmitter signal to another receiver input signal is 60 dB.
2. Selectivity Parameters measurement.

Measurements results: 2.1 Adjust Frequency Rejection (AF) 40 dB, 2.2 First Image Rejection (IR) 30 dB, 2.3 Second Image Rejection 70 dB, 2.4 Cross modulation and Blocking Effect 80 dB,
2.5 Spurious Channels IP3 50 dB.

172
With regard to 1.4 we can not use parameters 2.3 and 2.4.
* Exercise (continuation)
Useful signal Adjusted channel Image channel IP3 channel
3. First frequency (in MHz) choosing (300).

300 400
4. Frequency Bands for first channel.

300 AF 315
IP3 (315*2-330=300, see 1.3)
380
IR
400
5. Second frequency choosing (316).

300
173
317 We need two additional signals for IP3
400
300
315
330
* Exercise (continuation)
6. Frequency Bands for first and second channels.
300 397
7. Total result
?
300 317 334 351 382 397
Conclusion: It is possible to get 5 communications.
174
Conclusion
AGC
AM suppression
Image Rejection
IP3
Blocking Effect, Cross modulation
Whistles
Out of band channels
Adjust Frequency Rejection
175
7. Antenna in a small transceiver
Current distribution along monopole is Monopole /4 In top of monopole current equal zero. As usual we take /4 monopole. In case of /2 radiation compensates. One can use /2 monopole but he has to implement open non radiate /4 wire /4
/4
10.7 mm
Lcoil2 =190.42mm
177
Lfull =238.26mm
12.8 mm
Lcoil2 =46.34mm Lfull =91.56mm
Lcoil1 =120.42mm Lfull =164.81mm
178
12.8 mm
Monopole and plat problems
Tx
Rx
Tx
Rx
179
One of well know ideas is Many-stored cophased antennas
Antipodal wave is absorbed by coils
Antipodal wave is absorbed by /2 line loops
180
With Dipole
With Monopole
With PIFA
Without antenna ?
1960
2000
2012
years
PIFA
Inverse L Antenna (ILA) Inverse F Antenna (IFA)
Ground
Feeding
Feeding
F 1.5%
= 4(H + L)
S influences on matching and does not on resonance frequency.
Double Inverse F Antenna (DIFA)

Passive L radiator
Maybe F and some L antennas ?
F 4%
Active F radiator
It is desirable 10% !
182
Instead of some L antennas it is possible use planar technology.

See http://mobiblues.ru/post_1234816635.html
Planar Inverted-F Antenna (PIFA)
F = 10%, Prad/Pfeed 65%

Ground
Connection to ground
Feeding
Flat element
There is recommendation in www.aligent.com to choose h = 0.04
183
Dielectric intercalation
Dielectric intercalation and rabbet widen frequency band. Feeding point influences on antenna resonance behaviors.
Rabbet
The slot line in flat element allows decreasing SAR (L. W. Li, M. S. Leong, P. S.
Slot line Kooi, and T. S. Yeo. Specific Absorbtion Rates in Human Head Due to Handset Antennas: A Comporative Study Using FDTD Method. Journal of Electromagnetic Waves and Applications, Vol. 14, 2000, pp. 987-1000).
Feeding
184
Antenna pifa
185
MB Main Idea
Version 1
Dipole
+ _
/2
+ _
Version 2
+ _
+ _
186
+ _
+ _
Some Explanations
Dipole
MB
d
Lets consider 2 narrow lines, separated by distance d, as shown in the Figure If distance d is smaller than /4, the same currents will flow in the lines and there will be no coupling effects between them. If d is close to /4 , but not equal to /4, the current induced from neighboring line will have a different phase, because of delay. if d = /4, there will be no coupling because of orthogonality. But , in this case antenna pattern is not circular.
/2 loop
+ _
/4
Thats why the width of the radiation part of the PCB must be less than 1/8 (more explanations in next slide).
187
d >/8
/4
Let us come back to basis MB antenna. If we use PCB like antenna, so we haven't ground. So we have to use differential amplifier
/2 loop
+
MB
1/2
188
First stage MB antenna delaine line is printed line

If the PCB is wider than 1/8 we propose the following version in which the feed points are far from the PCB edges and separated by 1/8 If one of the current feeds is from the top edge as shown, the current distribution in the PCB will be uniform. This reduced the coupling effects
d >/8
/4
189
/2 loop
MB and PIFA simulations
PIFA from C S T
190
190
MB
191
191
PIFA- simulations of gain
192
192
MB antenna- simulations at far field
193
193
PIFA- simulations at far field
194
194
Comparison between PIFA and MB antenna
Antenna type
Freq. [GHz]
S11 [dB]
Rad. efficiency [dB]
Total
efficiency
[dB]
Directivity Far field [dBi]
Vin at excitation port for E = 30mV/m at distance 5m, [mV]
Gain [dB]
MB antenna
1.305
-24
-0.002
-0.0139
4.382
16.565
4.384
PIFA
0.9
-15.9
-0.224
-2.094
2.148
27.3
1.92
195
195
Experimental results
196
196
PIFA
197 197
Second stage MB antenna delaine line is separate device

There are three MB antennas using two delay lines. With help electronic keys (for example by PIN diodes) one can made one antenna for three frequencies
1 5 2 3 30 mm 6 4
1,2 = 0.2810-9s 1,5 = 0.2710-9s
3,4 = 0.8310-9s 3,6 = 0.7910-9s
70 mm
0.9 GHz
1.8 GHz
1.9 GHz
~
198
1234
12
34
15
36
8. IF Amplifiers
199
IF Amplifier
Resistive Amplifier
In
+
IC Resistive Amplifier
Out In
200
Single and Double Resonant Circuit

y= V = 0 V = i
2 i 2 0
I = 0 I = i
i 0 = 1 2 0 =
2i 0 0 ,7
0 Q= 0 ,7
0
0,7 2 (I 0) 1 i i
= Q
for n stages
for one stage
tg
n
y=
201
1 1+ 2
1 y= 2 1+
Double Resonant Circuit

CCkC k
C L L C C L L C
= kQ > 1
y = y = y =
+ 2
(1 ) + (1 + )
2 2
1+
2 2
< 1 = 1
202
+ 2
2 n
(1 ) + (1 + )
2 2
2 4 +
4
Double Resonant Circuit (continuation)
Frequency Characteristics
Phase Characteristics
203
Transformed Double Resonant Circuit

C(1) R R 2L2 C2/2 R C(2) n2 R
L1/2
2C1
L(1)
L(2)
(1 )
n2 L
L1 = L2 = C1 = R ( f 2 f1 ) f 2 f1 R 2 4 f 0 f 2 f1 4 f 02 R
(2)
L (1 ) = L (2) C C
(1 )
L1 + 2 L2 2 = 2n 2L2 C2 = 2n2
= 2C1
204
1 C2 = ( f 2 f 1 )R
(2)
Transformed Double Resonant Circuit

C(1) R R 2L2 C2/2 R C(2) n2 R
L1/2
2C1
L(1)
L(2)
(1 )
n2 L
L1 = L2 = C1 = R ( f 2 f1 ) f 2 f1 R 2 4 f 0 f 2 f1 4 f 02 R
(2)
L (1 ) = L (2) C C
(1 )
L1 + 2 L2 2 = 2n 2L2 C2 = 2n2
= 2C1
205
1 C2 = ( f 2 f 1 )R
(2)
It is given:
Filter Number Calculation Example f IF = 10 .7 MHz , Q = 80 , f ad .ch = 500 kHz
Find the filter number in cases of single and double resonant circuits with = 1 for adjusted channel attenuation more then 60 dB.
Solution: 10700 f 0 .7 = = 133 .75 kHz 80

Single resonant circuits
ad .ch =
2 500 = 7 .48 133 .75
60 dB 1 1000
Double resonant circuits
1 = 1 1000 2 1 + 7 .48 3 n= = 3 .37 log 0 .13

In this case we need four circuits and four stages.
206
2 = 1 1000 4 4 + 7 .48 3 n= 2. log 0 .036

In this case we need four circuits and two stages.
Monolithic Crystal Filters

ZZ XX YY ZZ is optical axis, XX is electrical axis, YY is mechanical axis
Quartz-crystal
C1
R1
L1
0 .3 1 % s
Q 10000 L1 5 20 mHn C1 0.001 0.0001 pF
207
C0
R1 = 0 = 1
2 s
11 1 = + L1C1 L1 C1 C0
2 p
208
209
SSB Crystal Filters
210
Two Types IF Filter Comparison

dB
kHz
One of possible solution

211
9. Analogue Decoding
Envelope (non coherent) detector
AM detector e(t) R v(out)
FM signal
FM/AM
AM detector
Weak signal mode i(d)
Strong signal mode i(d) V= e(d)
0.7V
e(d)
212
Weak signal mode
id = id0 + k1id + k2 (id ) + ...

2 2 t
If et = ed = Et Cos0t
id = id0 + k1Et Cos0t + k2 Et2 (Cos0t )2 + ... E E id = id0 + k1Et Cos0t + k2 Cos20t + k2 2 2 2 Et id = k2 2
2 t
213
Weak signal mode (continuation)
Et = E (1 + mSin t ) k2 2 k2 2 m2 m2 id = E (1 + mSin t ) 2 = E (1 + + 2 mSin t Cos 2 t ) 2 2 2 2 k2 2 k2 2 m 2 id , = E 2 mSin t id , 2 = E Cos 2 t 2 2 2 m2 2 =m THD = If m = 100 %, THD = 25 % 2m 4
214
Strong signal mode

i(d) V= e(d)
V out = V in K d m K d 0 ,8 0 , 9 2
For R >> Ri,
<< RC << 2
1
EI 2
V out I out
215
V out I out I1 2E E E 1 1 1 2 R in R
R in = R
@
Input resistance for envelope decreasing
C2 R C2 > C1 AM signal R
C1
R in , =
R R 2
If C2 and
R = 0,
m0
C2
m=0
216
@
Coherent AM Detector
mA0 { ... + Sin ( 0 + / )t + 0 / + ( ) + ... } Arec Sin( 0t + rec ) 2 2
Carrier recovery
K mix
Disadvantage
m A0 Arec Sin ( t + ) 2
Advantages: Small non linear distortions No problem with non symmetrical signal All type analogue modulations
217
Carrier recovery necessity
This is selective decoder
Envelope FM detector
Discriminator
Vout R= R= Vin
FM/AM
AM detector
218
FM /AM
f = 0 V2/2 Vm1 V2/2 V3 Vm2
+ f V2/2 V3 Vm1
219
- f V2/2 Vm2 V2/2 V2/2 Vm1 V3 Vm2
Ratio Detector
Vout R C R= R= Vin
L1 L3 = 4 k12 2
220
FM/AM
AM detector
@
m and Vout calculation
U U = 1 + j + j c oe (1 + j )2 + 1 1 + j j I c R oe = 2 (1 + j )2 + 2 f = f0 1 I 2 R
Detectors
m 1
m 2
Where f f
0
is deviation
= k
1 2
is loaded circuit quality
m =
m1
U
m
m 2
2U
Um = Um1 = Um2 ( =0)
m =
221
1 + ( + 2
)2
1 + (
2
)2
(1 +
Vout 2U m mKd
2
1+
+ 4
coefficients,depended on Um, Kd 0.6 0.8
Kd is detection
@
Attendant AM suppression
R1 R2
R1 R2
2 is Vout
AM causes in-phase voltage changing Um1 and Um2. In absent frequency modulation (f = 0 or = 0) the same Um1 and Um2 changing das not effect Vout. In present frequency modulation (f 0 or 0) Um1 Um2 , detection coefficients Kd1 Kd2 and AM causes Vout changing (Vout,AM). In ratio detector Vout,AM more less then in discriminator, because Co stabilizes voltage between 6 and 7 and voltage ratio in 2 changing little. Moreover Co decrease input resistance for envelope. This effect result in am decreasing. Vout,AM
m0
222
m=0
C2 C2
Discriminator Ratio Detector

f
@
Coherent FM Detector
11
v x = V x Sin t
vx
/2
vy
vout
v y = V y Sin t + 2 After LPF V xV y V xV y v out = Cos = Sin 2 2 2 = arctg = arctg 2 0 .7

2 << 0 . 7 If v out = V xV y 0 .7
2i 0 0 ,7
223
=2 0 , 7
= m Cos t , m Cos t 0 .7
v out = V xV y
@
Additional tunings in FM Receiver
Additional circuit
Suppression Method
Useful tuning Additional tuning Additional tuning IF Ampl. Limiter FM Detector Ampl.
Additional circuit 224
- When we do have a non symmetrical AM signal? - Which type of analogue modulation it is possible to detect with the help of envelope detector? - What the input resistance for envelope is? - Why do we implement two LC circuits not one in envelope FM detector? - Is additional tunings in FM receiver suppression is selectivity parameter?
225
10. Digital Coherent Decoding

Carrier Recovery Methods
(Reference Oscillation Selectors ROS)
Receiving signal Without Pilot Signal Receiving signal With Pilot Signal
Passive (disconnected, without feed back)
Active (closed, with feed back)
226
Carrier filtration
Modulation removing
PLL
Filtration Method for AM
Narrow band pass filter
Amplifier with AGC
Amplifier
Limiter
Band pass filter
227
11
Reference Oscillation Selectors for MPSK Modulation
This is frequency multiplication method for example. To increase m times the argument of a trigonometric function, decreasing the PM signal with 2/m phase hops, the phase modulation is removed, and the phase of the received signal does not depend on information phase hops.
At two phase modulation m = 2 (BPSK or DBPSK) multiplier can be a quadrator.
228
Carrier in DSB Receiver

is 1800
AM signal is 1800
Carrier reduced AM signal
229
Carrier component in DSB spectrum is absent, because half FFT period the phase is and another half is + .
Carrier Recovery in DSB Receiver
/2
Phase Look Loop (PLL)
Phase Detector (Mixer) 1 Input Loop Filter F(s) Discriminator output 3 i - 0 = /2
PLL can not make carrier recovery. Its Tracking Filter output goal is carrier 2 stabilization.
Voltage Control Oscillator (VCO) fVCO Capture range
v 2 (t )
230
fi Look range
Receiving in Continuous Channel Discrete Communication

It is known: -Source and coder behavior, -Modulator structure, -Channel mathematical model. What is the demodulator, which guarantees the best receiving quality. For example
( i = 1, 2 ,3 ,... m )
After modulator we have m signals u i ( t ) in time ( 0 t T ) The receiver gets signal z ( t ) You can find one of m hypothesizes.
Let we have after encoder m possible symbols. b i
for m = 4, bi are 00, 01, 11, 10. Error
z ( t ) bi
z (t ) = s j (t ) + n (t ) s j (t )
231
Non overlapping zones of space signal.
- Useful signal in receiving place - Additive interference Not error
n (t )
Receiving criterion of quality

Z(t) bi It is correct with probability P(bi/z) -
Demodulator
posterior probability (after experiment). Maximum posterior probability rule: Byes formula
P (b j z ) = P (b j ) P (b j ) w ( z b j ) w( z )
P(bi z ) > P(b j z ), j = 1,..., m. ji or max P(bi z )

i
prior probability of bj
Modulator and channel behavior.
P(bi ) w( z bi ) > P(b j ) w( z b j ) j = 1,..., m. ji or max[ P(bi ) w( z bi ))]

i
For choosing bi all J I (m 1) inequalities are:
i, j =
w( z bi ) w( z b j )
>
P(b j ) P(bi ) 1 m
If P(bi ) = P(b j ) = i, j > 1
232
Sometimes there is additional clean or zero hypothesis: no signal only interfere:
w ( z bi ) w( z 0) i > j
or
= i ,0 = i
for all
ji
i
max[ i ]
This is correct for 1 P(bi ) = P (b j ) = m
This is likelihood ratio relatively zero hypothesis.
In many cases (for finite signals) one can implement following rule:
w ( z bi ) > w ( z b j )
233
Receiving Methods
Coherent methods Transmitted
signal variants are completely known The channel does not cause random amplitude and phase shifts. The receiver has signal examples with the same amplitude, frequency and phase.
11
Quasi-coherent methods
Signal variants have a random and uniformly distributed initial phase. The channel does not cause random amplitude and phase shifts. The receiver can get signal examples with help carrier recovery from receiving signal.
Non coherent methods

for unknown shape signals The channel causes random amplitude and phase shifts. This method is impossible to implement for modulations including opposite signals (with phases and + ). So this method cannot implement for MPSK or MQAM. But it is possible to implement for phase-difference modulation.
234
Optimum Incoherent methods
Auto correlated methods
Optimal receiving, Coherent method

Let will be only noise is probably value with spectral density N0
z ( t ) = s i ( t ) + n ( t ), s i ( t ) = ku i ( t )
( 0 t T ), ( i = 1 , 2 ,..., m )
where all are known.
All signals si(t) finite with duration T and it is not multi path propagation, Doppler effect and so on. There is full synchronization. We don't know n(t) and i. We have to find likelihood ratio relatively s(t) = 0, and z(t) = n(t).
The finite signal have a in finite frequency band. In this case we have Space signal with n = . Therefore we suppose that our noise is quasi white noise in band F=n/2T , 1 where n >> 1. T Zero hypothesis [ Z(t) is noise]. Let take n sections in one symbol time.
t =
2F
The samples in these sections Z1, ,Zn uninvited for quasi white noise, therefore we can get n-dimension probability density foe these samples:
w( z1 , z 2 ,..., z n ; t1 , t 2 ,..., t n where 2 = N 0 F

235
1 0= exp 2 ( 2 ) n 2 1
z 2 (t k ), k =1
n
Transmitting signal bi hypothesis. In this case:
n(t ) = z (t ) si (t ) w( z1 , z 2 ,..., z n ; t1 , t 2 ,..., t n 1 bi = exp 2 ( 2 ) n 2 1

1 = exp 2 0 2
n
[ z (t k ) si (t k )]2 , k =1
n
The likelihood ratio relatively zero hypothesis for n sections will be:
=
[n] i
w( z1 , z 2 ,..., z n ; t1 , t 2 ,..., t n bi w( z1 , z 2 ,..., z n ; t1 , t 2 ,..., t n
[ z (tk ) si (tk )] +
2 k =1
z 2 (t k ) 2 2 k =1 1
n
2 = N0 F =
[n] i
N0 2t 1 [ z (t k ) si (t k )] t + N0 k =1
2 n
1 = exp N0
z 2 (t k )t k =1
n
We can choose i with maximum
[in ] or ln [in ]
1 [ z (t k ) si (t k )] t + N0 k =1
2 n
236
1 ln = N0
[n] i
z 2 (t k )t
k =1
2 [n] ln i = N0
Let come back to white noise.
1 z (tk )si (tk )t N k =1 0

T
si (tk )t
2 k =1
F , n , t 0
T
2 2 1 [n] ln i = lim ln i = z (t ) si (t )dt si2 (t )dt n N N0 N0 0 0 0
The solution algorithm will be:T

i
z (t )s (t )dt 0.5E z (t )s (t )dt 0.5E ,

i j j 0 0
j = 0,..., m 1,
E j = s 2 (t )dt j
0
The correlator calculates this value

237
( z , si ) = z (t ) si (t )dt
0
Therefore this method named correlation
Coherent demodulator by using correlator (active filter)

T
11
z ( t ) s i ( t ) dt 0 . 5 E
z ( t ) s j ( t ) dt 0 . 5 E j ,
j = 0 ,..., m 1 ,
It is possible to realize correlator algorithm be following circuit: :
G1
E1/2
Gi Z(t) Ei/2
Gm Em/2
max
bi ?
Falloff
238
If Ei = const. we can find. This algorithm is not depend on Ei
T max z (t ) si (t )dt i 0
11
Systems with Ei = const (systems with active pause)are most popular specially for channels with selective fading. In these systems the difficult problem is synchronization (T borders and falloff time knowledge).
Signals in systems with active pause examples:
Signal in system with passive pause examples:

239
In simple binary system there is only one unequally:

T
z (t ) s
0
T 1
( t ) dt 0 . 5 E 1 >
z (t ) s
0
T 2
( t ) dt 0 . 5 E 2
or
z (t ) s
0
( t ) dt >
s ( t ) dt = S 1 ( t ) S 2 ( t )
- differential signal - threshold level R

Vc
Solution
= 0 .5 ( E 1 E 2 )
For binary system with passive pause we have following algorithm. In this case:
z(t)
C
threshol d level ()
s1 (t ) = a, s2 (t ) = 0, s (t ) = s1 (t ) = a, a, a 2T E1 (t ) = a T , E2 (t ) = 0 = 2
2
In this decoder one must choose RC>>T, then in T moment:
Now:
T
a 2T aT z (t )adt > z (t )dt > 2 2 0 0
1 VC = RC
z (t ) dt
0
240
aT 2 RC
Optimal receiving by matching filter using
11
( z , si ) It is possible to calculate scalar product with help passive linear filter with constant parameters.
= z (t ) si (t ) dt
0
T
Let we have filter with time response g() y (t ) = g (t ) z (t )d [sometimes h(t)] . For input signal z(t) we will get 0 output signal y(t) Let we choose g() so as the receiving value in time t = T will be equal to scalar product. So:
g ( ) = si (T )
T i
y (T ) =
s (T ) z (T )d = s ( t ) z ( t )dt
i 0 0
= ( z , si )
Matching with signal si(t) filter is filter with time response
g ( ) = as (t0 )
a and are const.

241
Finite function g(t) is mirror image of finite signal s(t)
11
We can find matching filter transfer function with help Furrier transform
k ( j ) MF =
g (t ) exp( j t ) dt = a s (t

t ) exp( j t ) dt =
S ( j )
= a s ( ) exp[ j (t 0 t )]d = aS ( j ) exp( j t 0 )
S ( j ) is conjugated term to s(t) spectral density

S(t) spectral components with big energy have a most response in matching filter. All components are adding in phase due to in s(t) equals in matching filter. Therefore all correspond components are adding in phase.
S ( j )
242
For signal z(t) = s(t) + n(t) we can get matching filter output signal in time T:
11
y (T ) = a z (T ) s (T )d = a z ( x) s ( x)dx a ( z , s )
0 0
Now we can get Coherent demodulator by using matching filters MF1
E1/2
It is possible to show, that matching filter allows to get maximal value of Eb/N0 in comparison with any linear circuit.
Ei/2 Z(t)
MFi
max
bi ?
Falloff
Em/2
243
MFm
Matching filter changes a signal form, but its task is another getting solution yes or no.
Matching filter realization

Delay line for T
11
LPF
With help this circuit we can get any signal s(t), because:
sin 2F (t k t ) s (t ) = a k 2F (t k t ) k =0 1 t = a k = s ( k t ), 2F F - s(t) bandwidth

n
This is transversal filter. If input is a it is s(t) generator, if input is b this is matching filter for s(t).
244
11
Realization difficulty in active filter (correlator) and in matching filter approximately the same. Correlator demands precision phase synchronization of coherent signal. Matching filter demands precision time for getting output sample.
Vin
VMF,out
Vcor,out
245
11. Non Coherent Decoding

Difference methods
Difference encoding
Let us 1 does not change polarity, -1 change.
11
Phase-difference modulation (PDM)
Information After encoder After decoder After encoder After decoder Information After encoder After decoder One error after encoder After decoder 246
1 1 -1 1 1 -1 1 -1 -1 1 1 -1 -1 -1 1 1 -1 1 1 1 1 -1 1 1 -1 1 -1 -1 1 1 -1 -1 -1 -1 -1 1 -1 -1 1 1 -1 1 1 1 1 -1 -1 1 1 1 -1 1 1 -1 1 -1 -1 1 1 -1 -1 -1 1 1 -1 1 1 1 1 -1 1 1 -1 1 -1 -1 1 1 -1 -1 1 1 1 -1 1 1 1 1 -1 1 -1 1 1 -1 -1 1
Polarity changing 1 1 -1 1 1 1 -1 1 1 - 1
Let one of m equiprobable and equally powerful signals with duration and a random, uniformly distributed initial phase and the white Gaussian noise n(t) come to the demodulator input
S j (t ) = S j (t ) cos + S j (t ) sin x ( t ) = S j ( t ) + n ( t )
for
j = 1, 2,..., m
S j is the Gilbert transformation of signal Sj.

The error probability will be minimum, if the demodulator makes a decision in favor of such a signal Si, for which the following inequality is correct for any j i.
x(t ) Si (t )dt + x(t ) S i (t )dt > x(t ) S j (t )dt + x(t ) S j (t )dt 0 0 0 0
For PDM = 2T. Therefore the algorithm of the optimum incoherent demodulator is a follows: Within the interval of the nth symbol, a decision should be made in favor of the signal Si with the phase difference i.
for 0 < t T a sin t Si (t ) = a sin(t + i ) for T < t 2T

247
This method analyze see in Y. Okunev, Phase and Phase-Difference Modulation in Digital communications, Artech House, 1997.
The reference oscillator (O) in incoherent demodulator is incoherented, and it has random initial phase. At the same time the frequency of the reference oscillator should coincide with the frequency of the PM signal carrier; slight frequency deviation are admissible, but they result in a noise immunity decrease.
248
Auto correlated Signal Processing

Let us consider two variants of the transmitted signal S1(t) and S2(t) in time 2B interval 0, 2 B
S 1 ( t ) = i i ( t ) S 2 ( t ) = i i ( t )
i =1 i =1
i i - orthonormalized basic functions

2B
For the case of signals with identical 2B 2B energy 2 i = i2

i =1 i =1
- dimension of expected signal
i = S1 (t ) i (t ) dt
0
likelihood algorithm will be:

2B x (t ) i (t ) dt > x (t )i (t ) dt i =1 0 i =1 0 2B 2 2
i = S 2 (t ) i (t ) dt
0
- unknown factors
If > S1 is transmitted, if < S2 is transmitted. f(t) is an arbitrary function. We can choose the following basic functions ( 0= 2/T):
For two symbols of PDM we can write:
249
For i = 1 S1, S2 and basic functions are shown:
The basic of S1 contains the even harmonics of the f = /T and the subspace of S2 contains odd ones:
S 1 (t ) = S 2 (t ) =
i =1
a 2 i sin 2 i
t + b 2 i cos 2 i
i =1
a 2 i 1 sin( 2 i 1 )
t + b 2 i 1 cos( 2 i 1 )
250
The getting integrals within the limits (0, T) are proprtional to the decomposition coefficients of the signal received within the interval ) to T, and the integrals within the limits (T, 2T) are proportional to the decomposition coefficients of the signal received within the interval T to 2T. Hence, the sum of products of these coefficients is proportional to the scalar product of two adjacent signal chips, that is, can be written
11
J = sgn x(t ) x(t T )dt

T
2T
Thus, the use of the optimum algorithm of unknown shape signal reception, realizing the generalized maximum likelihood rule, results in the autocorrelated scheme of PDM signal processing as shown below.
251
Comparison with the Shannon limit
Bandwidth-per-bit-rate comparison.
252
12. Phase-locked loop (PLL)

Phase Detector (Mixer)
11
Input
Loop Filter F(s)
Tracking Filter output
2
Voltage Control Oscillator (VCO)
Discriminator output
v 2 (t )
Implementation Examples:
- PLL like a synthesizer - For carrier recovery - FM Detector
253
Stable frequency source 1 Modulate signal 1 Modulate signal 1
Local frequency source 2 Carrier 2 Information 3
11
PLL like a Frequency Gain Control (FGC)
Information Discriminator
Very Low Pass Filter
254
11
PLL like a synthesizer
255
Phase-locked loop another block diagram
11
256
Phase-locked loop Loop operation

e (t ) = vi (t ) vo (t ) = VsV0 cos( i t + i ) sin( i t + o ) = 1 = VsVo [sin( i o ) sin( 2 i + i + o ) ] K d sin( i o ) K d ( i o ) 2
Phase Detector (PD) Input Loop Filter F(s)
Vd = K ( i 0 )
11
vi (t ) = V s cos( i t + i )
Tracking Filter output
v 0 ( t ) = V 0 Sin ( i t + 0 )
Discriminator output
Voltage Control Oscillator (VCO)
v 2 (t )
257
In case of tracing i = 0
Phase Discriminators
Very Low Pass Filter
258
Phase tuning transformer

f = 0 V2/2 Vm1 V2/2 V3 Vm2
+ f V2/2 V3 Vm1
259
- f V2/2 Vm2 V2/2 V2/2 Vm1 V3 Vm2
PLL
11
Phase Detector
(PD)
Vd = K
d
( i 0 )
vi (t ) = V s cos( i t + i )
v 0 ( t ) = V 0 Sin ( i t + 0 )
F(s)
(VCO)
v 2 (t )
v d = K m vi (t ) v 0 (t ), where If sec ond harmony
K m is const . after LPF we have
( 2 i ) is absent
K mV sV0 vd = Sin ( i 0 ) or v d = K d Sin ( i 0 ) 2 i 0 = e e is error. In receiving time (hold-in

range or after capture) 260
vd = K d e
VCO
(PD)
vi (t ) = Vs cos(it + i )
v 0 ( t ) = V 0 sin( i t + 0 )
0 (s) = K 0
V2 (s) s
V d = K d ( i 0 )
11
F(s)
(rad/sec)
(VCO)
v2 (t )
= K 0 v 2 (t ) K 0 =
d 0 = K 0 v2 (t ), after Laplace transform dt t V2 ( s ) = K 0 v2 (t ) dt s 0 ( s ) = K 0V2 ( s ) and 0 ( s ) = K 0 s 0 dx (t ) = sx ( s ) dt
rad in sec Volt
or
Hz
if
F Ko = V
arctgK0 V2
Phase change in VCO output is integral from input VCO voltage.
t F (s) f (t ) dt = , s 0
261
f i = f 0 + K 0 v 2 (t ) ( K 0 in Hz ) V v0 (t ) = V0 Sin 2 f 0 + K 0 v 2 (t ) dt
Loop filter
(PD)
vi (t ) = Vs cos( i t + i )
v 0 ( t ) = V 0 Sin ( i t + 0 )
0 (s) = K 0
V2 (s) s
11
V d = K d ( i 0 )
F(s)
(VCO)
v2 (t )
Second order Low Pass Filter Passive Active
262
PLL
Active Second order Low Pass Filter
F ( s ) = L{x (t )}= x (t ) e st dt
0
A ( sCR 2 + 1) F2 ( s ) = sCR 2 + 1 + (1 A )( sCR 1 ) If s 2 + 1 A , F2 ( s ) = s 1
263
Transform function
Usual feed back unit i
Kd F(t)
11
0
K0
L
v i (t ) = V s cos( i t + i )
F (t ) K d L = , but i 1 + K 0 F (t ) K d 0 LK0 F (t ) K d K 0 = = i i 1 + F (t ) K d K 0
(PD)
Vd = K
d
0)
F(s)
v 0 ( t ) = V 0 Sin ( i t + 0 )
0 (s) = K
0
V2 (s) s
(VCO)
v 2 (t )
K0 (s) K 0K d F (s) s = = H (s) = 0 K i (s) s + K 0K d F (s) 1 + K d F (s) 0 s K d F (s)

264
KV = K0KdK(s) = K0KdKf is open loop-gain. If after filter there is amplifier with Ka:
KV = K0KdKf Ka
Transform function of PLL with Active Second order Low Pass Filter
H (s) =
K 0 K d ( s 2 + 1) 1 s 2 + K 0 K d ( s 2 + 1) 1
2
2 n s + n H (s) = 2 2 s + 2 n s + n where
=
2 n
K0K d
and
1 = 2 n = 1 Q 2
H(j) dB = 0.3 = 0.7 = 3.0
Amplifying
+2 0
10 265 0.7 1 2 3 /n
Another LPF circuits
2 n s + n H (s) = H 0 2 2 s + 2 n s + n
2
Below you can see elements calculation if parameters and n are known. In calculation result one can get relative values r and c. The real values you can get by: Where
Ri = ri R
Ci =
is normalizing resistor, is real cutoff or resonance frequency.
ci n R
As usually if segment Q < 15 it is possible to made each segment by one OA. If segment Q > 15 it is better to made each segment by three OA.
266
2 n
n2
H0
in
out
1 c2
1 1 1 + + r r r 3 4 1
1 r3 r4 c 2 c5
1 r1r3 c 2 c5
Uin
out
K1
1 1 1 k + + r1c1 r2 c1 r2 c 2
1 r1 r2 c1c 2
k r1r2 c1c 2
Uin
Uout
1 r5 c 2
r3 r1r2 r4 c1c 2
r3 r1r2 r4 c1c 2
267
Phase error in PLL with Active Filter

(PD)
V d = K d ( i 0 )
11
vi (t ) = Vs cos(i t + i )
v 0 ( t ) = V 0 Sin ( i t + 0 )
0 (s) = K 0
V2 (s) s
F(s)
(VCO)
v2 (t )
If i i +
0 = i +e
e = ( i 0 ) = K0Kd
If
268
< e <
Ampl . , e 0
Tracking and Acquisition
11
fVCO
i - 0 = /2
fi L H or Fmax
Capture range or two Pull-in range L = 2n(LPF) Look range or two Hold-in Range H
Fmax = ( / 2 rad ) K d K f K a K o Fmax = ( / 2 rad ) K V depend on Ampl., on n(LPF) and on fmax of VCO
269
S/Nphase
Timepull-in
@
PLL with different Lock-in range zone
(PD)
F(s)1 A1 F(s)2 A2
In tuning time In receiving time
(VCO)
v2 (t )
VREF
If we change only n , e does not change. If we change n and A , e is changing.
270
Costas Loop
u i (t ) = Ai m (t ) sin [ i t + i (t ) ] u o1 (t ) = Ao sin( i t + o ) u o 2 ( t ) = Ao cos( i t + o )
11 Here the reference oscillation is generated by the VCO, which with two correlators, a multiplier and Loop Filter form an PLL unit. The VCO is controlled through the filter by a signal equal to the product of the received signal projections onto the orthogonal reference oscillations. This control signal is proportional to a sine of a double phase difference between the received signal and the reference oscillation. If the i is m*1800 ( for example in BPSK or DSB ), VCO output signal does not depend on i. Actually this is Carrier recover for BPSK, DBPSK or DSB signals.
Ai A o AA m ( t ) cos( i o ) i o m ( t ) cos( 2 i t + i + o ) 2 2 AA AA u d 2 ( t ) = i o m ( t ) sin( i o ) + i o m ( t ) sin( 2 i t + i + o ) 2 2 u d 1 (t ) =

ud
Ai Ao m (t ) cos( i o ) 2 A A m (t ) u Q (t ) = i o sin( i o ) 2 u I (t ) =
o )]
[Ai Aom(t )]2 sin( (t ) =

8 8
i
o i + o
[Ai Ao m(t )]2 sin[2( )+

8
[Ai Aom(t )]2 sin(0) + [Ai Aom(t )]2 sin[2( )]

o
[Ai Ao ]2 sin[2(
271
o )]
In receiving time (hold-in range or after capture), in case of small frequency shift.
u f (t ) =
[Ai Ao ]2 sin [2(

8
o )]
u f (t )
Kd [2( 0 i ) ] = K d ( 0 i ) 2
- What is purpose of PLL in synthesizer? - Is the phase discriminator a multiplier? - What is the influence of quality and order of active filter on selectivity? - Way hold-in range is higher then lock-in range usually? - In what way frequency multiplying make remodulation of mPSK signal? - Does a DSB signal have a carrier? - What is the difference between PLL and Costas loop?
272
Appendix 1.
Preparing to Tx-Rx-Lab
Superheterodyne Receiver
Homodyne Receiver
Band select filter Low-pass filter
LNA
DC
273
LO
Modulations and Modulators
ES
f << f 0
274
@
Basic Receiver Parameters, Antennas
Sensitivity Frequency range Impedance Noise figure Signal to noise ratio Amplification Dynamic range Selectivity Bandwidth Adjust Frequency Rejection
275
Spurious Channels
@
Basic Receiver Parameters (continuation)
Image Rejection Cross modulation Blocking Effect AM suppression ( ), AM
Distortion
Total Harmonic Distortion (THD) Intermodulation Subjective sound quality
Heterodyne radiation
276
Measurement scheme for Receiver Sensitivity

m0
Filter for Modulation Frequency
S/N
Voltmeter
Field source
Antenna
Receiver Generator Antenna equivalent Filter for all frequency band
m=0
Field source
Antenna
m0
Receiver
Filter for Modulation Frequency Filter for all frequency band and Rejection Filter for Modulation Frequency
SINAD
Voltmeter
Generator
277
Antenna equivalent
m0
Two signal selectivity measurements
Ant G G1 Ant equiv G2
Wide band filter Rec
F measur.
SA
278
Sound Signal Distortion

Total Harmonic Distortion
THD =
V + V + ...
2 2 2 3
V12 + V22 + V32 ...
100 0
0
V + V + ...
2 2 2 3
V12
1000 0
f
Intermodulation
Vf1in Vf2in
Vf1-f2 Vf1 Vf2 Vf1+f2
Kinter mod =
279
Vf 1 f 2 Vf 1
1000 0
CE stage
Real Multiplier Example (MC1596G)
Vin
280
Appendix 2.
Shortly about DWDM
1. 55 m = 1550nm
193.55THz = 193550GHz
Wavelength-division multiplexing (WDM)
281
Erbium-Doped Fiber Amplifier (EDFA)
Pout
A preamplification B follower (basic amplifying) C saturation condition (buster)
Amplifying
In one step amplifier amplifying in C is 16dBm. With two pump generator it is possible to get 26 dBm Noise figure in A is 5 6 dB
Filter
282
EDFA amplification and wave length
Spectral density, dBm in 0.1 nm
nm
283
For example CATV
284
One of several problems in DWDM systems is Dispersion
285
Decibels
V1/V2 P1/P2 R
Zero Levels in lines

P0 = 1 mW 0dBm
If R = 600 , V0 = 0,775 V and I0 = 1,29 mA LP = 10 log(P/P0)
P = V 2 / R P1 / P2 = V12 / V22
NV = 20 log(V/V0) NI = 20 log(I/I0)
20lg( V1/V2 ) = 10 lg (P1/P2)

dB 0 3 6 10 20 V1/V2 1 2 2 3 10 P1/P2 1 2 4 10 100
Zero Levels in RF
0dBm is 1 mW If R = 50 , OdBm correspond to 22.4 mV
V1 / V2 = 300 = 10 10 3 (20 + 20 + 10)dB = 50dB

286
Appendix 3
Operational Amplifier as Universal Element

About Slew Rate: Let us Then
v in = V in sin 2 ft dV = 2 fV out cos 2 ft dt Maximum SR corresponds to cos 2 f max t = 1 SR = 2 f max V out , max
Ideal OA has:
KV,0 = , Zin = , Zout = 0 , f1 = , Slew Rate (SR) =

Let us KV,0 , but KV,0 > Z0/Z1
Since [ K V ,0 or Z in = , V = out ] V0 U in Z 0 + I 1 = I 0 , but I 1 =
Then
Vin V0 and Z1
I0 =
V0 Vout Z0
(V V0 ) Z 0 (V0 Vout ) Z 1 =0 I 1 I 0 = in Z1Z 0 Vout Z 0 Vout Z 1 + + Vout Z 1 = 0 K V ,0 K V ,0 Z KV = 0 Z1 1 + KU = 1 1 Z (1 + 0 ) K V ,0 Z1
I0 I1 Vin Z1 U0
Z0
+
If
V [ K V = out ] Vin [ = Z1 ] Z 0 + Z1
KV,0
Uout
K V , 0 >> 1 Z0 Z1
Z0 1 Z1 1 + 1 K V ,0
KV = Z in = Z 1
287
Different configurations with OA

I0 I1 Z1 U0 Vin Z0
Difference amplifier
KV,0
Uout I1 V1 V2 V011 Z11 Z01 Z1 V01 I0 Z0
KU,0
Vout
K V = (1 +
Z0 ) Z1
1 1 1+ K V ,0
If
K V , 0 >> 1 Z0 1 = Z1
KV = 1 + Z in =
If
K V , 0 >> 1 and Z0 (V 2 V1 ) Z1
1 1 Z 0 = Z 0 , Z1 = Z1
Vout =
288
Z out 0
I0 I1 Vin Z1 U0
Z0 Zout Vout Z0 Vout Zout
KV,0
Vout Vout ZL
ZL Z1
V out = K V , 0 (V in V out )
V out = V out
1 Z out Z out 1+ + ZL Z 0 + Z out
Zout does not influence, if
Z out < Z L and
Z out < Z o
289
Z in ( Z d Z com )
Z0 Z1 Vin1
+
Zcom
1+ V out = V in 2
Vin2
Zd
Vout
Zcom
Z0 Z0 Z0 + Z 1 Zcom Z1 V in1 1 1 1+ 1+ K U ,0 K V ,0
1 Z Z Z0 1+ 0 + 0 + Z 1 Zd Zcom
Zin does not influence, if
Z 1 < Zd and
Z 1 < Zcom
290
Operational Amplifier implementation

Weighted summer
Vin,i
V out = V in ,i
1
R0 Ri
Voltage follower
Rin = , Rout = 0 k v = 1
Integrator
Vin
Vout =
1 C o R1
V
0
in
dt
291
Partial integrator

f =
6dB/oct
1 2 C 0 R 0 1 2 C 0 R 2
f =
R0
Differentiator
Vin
C1
KV,0
V out = R 0 C 1
dV in dt
292

Stabilitrons with R = Rlim I
Limiter
Vout
Vin V
Big resistance region
Vin
Vout
V out ,lim = V in
1 R0 2 R lim
Precision limiter
V out , lim = V in
RL
R 0 R L + R 0 R + RR L R1 R L
293
See amplitude modulator also.
Comparator
Zero comparator Vin Vout
Vout Vin
One side comparator Vin
Vout
Vout
Vout
Vin
Comparator with hysteresis Vin Vout
Vout
Vin
294
OA and Multiplier Implementation Examples

Voltage divisor
Root extraction circuit
~ 1 R0 Vy Vout
295
Vx Vout Vy Vout Vx
Vx Vout
Appendix 4
Addition to digital modulation

Ideal Filter
Ideal normalized low-pass filter
H(f ) = H(f ) e
j ( f )
1 H(f ) = 0 H ( f ) e j 2 ft df =
f < fu f fu
h (t ) = 1 { H ( f )} =
fu
fu
e j 2 f ( t t0 ) df =
sin 2 f u (t t0 ) = 2 fu = 2 f u sin c 2 f u (t t 0 ) 2 f u (t t0 )
Sinc = sin /
296
Practical formulas for fu = Fm and t0 = 0 K() A -Fm 0 Fm

2 Fm
(nT)
LPF
h(nT)
1 h (t ) = 2
j t
K ( ) e jt d
Sin 2Fm t = 2 Fm A 2Fm t
A h (t ) = 2
2 Fm
A 1 j t d = e 2 jt
h (t )
2 Fm 2 Fm
2FmA t
297
2Fmt 4Fmt
Short impulse spectrum

x(t) h -T/2 0 T/2
0 x (t ) = h 0
j t T /2
t < T / 2 T / 2 < t < T / 2 t >T /2
S ( ) =
x (t ) e
dt =h
T / 2
j t
h dt = e j t j
T /2 T / 2
e j e j Sin = 2j
S ( ) = hT Sin T 2
T 2
Frequency and time responses duality.
299
Time and Spectrum
Fm = f1 =
1 2T
T
Periodic signal with period 2T
2T =
x (t ) = C k e jk1t
1 Ck = T
1 x (t ) = 2
x (t )e jk1 dt
S ( )e jt dt
No periodic signal
S ( ) =
x (t )e jt dt
300
K ( ) =
U out
U out = K ( ) U in
U in

Sin. and Cos.
y e2 d
A1 sin 1
2 1 e1 x
A1 cos1
10 BPSK
A1 sin 1 A1 sin 1
QPSK
d
11
e
A1 cos 1
e e e e
00
A1 cos 1
e=
301
01
d = 2 E = 2 PT
e= E
Ts = 2T
d = 2 E = 2 P 2T = 2 PT
B2. Orthogonal FSK (Orthogonal Frequency Shift Keying)

x 0 (t ) = x1 ( t ) = E cos 2 f 0 t = E cos 2 k ft E cos 2 f 1 t = E cos 2 ( k + 1) ft
y
0 tT
There are orthogonal signals if f = f1 f0 = 1/T.
For orthogonal signals we have:

S1
e2 e1 d e= E
S2
(
0
E cos 2 k ft
d =
E cos 2 ( k + 1 ) ft dt = 0
2 E 2
Eb/N0 difference with BPSK is 3dB
d BPSK = d FSK
If f1 f 0 = f = 1 T
1 = 2 f t = 2 T = 2 T 302
There are frequency shift only
Frequency efficiency is 1/3 bit/s/Hz
Orthogonal MFSK
x1 (t ) = x 2 (t ) = x3 (t ) =
E cos 2f1t E cos 2f 2 t E cos 2f 3t

Rbit = (log 2 M ) 1 bits T sec
All signals are orthogonal to each other

S 2 (t )
E
d = 2 E
S1 (t )
S 3 (t )
Ts = T log 2 M f = f i f i 1 = 1 T log 2 M
F
303
F =
1 M T log 2 M
Binary Modulations
BPSK
A1 sin 1
FSK
y S1 d e1 S2 x
y S1
ASK
e= E e
A1 cos 1
e= E
e2
e= E
e2 = 0
e1 x
d = 2 E = 2 PT
d= 2 E
d =
304
System Parameter (q = 2FT)

FT = 2 includes 94.8% of he whole of power. FT = 1 includes 90.2% of he whole of power.
Power spectral density of random NRZ base-band signal
q =1
Minimal System Parameter S/N ISI
With help the band-pass filter, which support needed pulse shaping and with help coherent demodulator, we can get F = Rb without noticeable BER increasing.
q 2 1
q = 2 FT = 2
BPSK 0.5bit/s/Hz 1bit/s/Hz QPSK 1bit/s/Hz 2bit/s/Hz
305 In FSK frequency efficiency is 1/3 bit/s/Hz
@ f and t
f t 1
Short pulse spectrum x(t) q
S ( ) = x (t ) e jt dt
0
e j t 1
S ( ) = x (t ) dt = q
0
e j 1, << 1
306
if
S()
1/
Appendix 5
Hartley modulator
Up and down conversions make by mixer two sinusoidal signals. As a result we receive two signals with sum and difference frequencies. One of these two signals needless. There are many cases where this signal deleting is difficult problem. In some cases it is possible to solve this problems by wall known Hartley modulator
Ralph Vinton Lyon Hartley

(November 30, 1888 May 1, 1970) was an electronics researcher. He invented the Hartley oscillator and the Hartley transform, and contributed to the foundations of information theory.
The main idea of Hartley Modulator is following trigonometrically equitation using.
sin sin =
1 [cos( ) cos( + )] 2 1 cos cos = [cos( ) + cos( + )] 2
Sin
Single-Sideband Modulation (SSB)

1 [Cos ( ) Cos ( + )] 2 1 [Cos ( ) + Cos ( + )] Cos Cos = 2 Sin Sin =
}
Cos(-) Cos (+)
Sin Sin
m(t)
Cos Cos
308
1 a(t ) = A0 [am (t ) Sin 0t m am (t ) Cos 0t ] 2 1 am (t ) a (t ) = d t
Direct Upconversion
1 sin sin = [cos( ) cos( + )] 2 1 cos cos = [cos( ) + cos( + )] 2
Transmitter
309
Direct Conversion, Homodyne receiver of Digital information
Problems: Image rejections Silicon internalization
Low-pass filter
The signal is first amplified at a low noise stage and then directly converted to the baseband or even to a direct current signal. When the frequencies of the RF and the LO signals are equal, this scheme works as a phase detector. In some literature, only when the local oscillator is synchronized in phase with the incoming carrier frequency, the receiver is called homodyne. 310
Band select filter LNA LO LO +900 Low-pass filter
DC Q
Band select filter
Low-pass Filter
To achieve maximum information, we should take both parts of signal. Its done by a method, which is called quadrature downconversion. The principle of this method is that the signal is at first divided into two channels and then downconverted by an LO signal, which has a phase shift of 90o in one channel with respect to another. The vector of the resulting signal is described as:
Signal =
I +Q
2
arg( Signal ) = = arctg
Q I
The main problem in homodyne technique is an offset caused by the LO signal leakage to the RF port of the mixer. The propagated signal reflects from the components in the front-end of the receiver and goes back to the mixer, where it is mixed down to DC. The offset can be considerable with respect to the signals to be measured. This leads to a narrower dynamic range of the electronics, because the active components get saturated easier than it would be in case of a zero offset. For example, lets take a mixer with the LO-drive equal to 1V and RF/LO isolation equal to 40 dB. In this case the offset can be as high as 10mV. Isolation of LNA can be or about 40 dB too. In case of a high sensitivity this can be a large number (this is the signal level at the output of the mixer, some amplification stages follow! See picture).
Band select filter Smin 1V Low-pass filter DC
Problems: Synphase signal influence DC offset returning
LNA 10mV 0.1mV 311 1V

LO
Nonlinearity
Another problem of the homodyne receiver, or, more concretely, of the I/Q (in-phase/quadrature) mixer, is mismatches in its branches. Assuming a mismatch of for the amplitude () and for the phase (), we can estimate the error, caused by these mismatches. In this way we get:
E IQ =
S ideal S miss 1 + 2 S ideal 2

I
For typical values of and this gives an error of .
a cos t
a (1 + ) cos( t + )
b cos( t + )
b sin( t + )
Q
In the processing of the BPSK signals it is important to know the phase with respect to the phase of the reference cavity. That means that an I/Q mixer becomes very attractive for this application. On the other side, it has too many specific hardware problems. Thats why it is desirable to apply DSP method In that case an I/Q mixer and other elements realized in forms of idealized models have no mismatches or reflections etc. Surely, the initial signal still has to be downconverted and prepared for the digitizing.
312
Problems Solutions
There are three ideas: Differential (balanced) shims Direct receiver, but fRF fLO , for example fLO reproduces fLO (Sub-Harmonic Mixer) Quadrature LO Problems: Synphase signal influence DC offset returning and mixer Nonlinearity
= 2 fRF
313
Using LO with frequency fRF /2 does not solve problem always. There are another solutions today: wideband and low IF methods. Image rejection in these cases it is possible to solve by Hartley method
1 [cos( ) cos( + )] 2 1 cos cos = [cos( ) + cos( + )] 2 1 sin cos = [sin ( ) + sin ( + )] 2 sin sin =
314
Wideband-IF Arch.
315
316
Laboratory
317
FLO = FS /2 Method
TR module elements
Some Abbreviations. TP Test Point. DCA Digital Control Attenuator. Amp Amplifier. SPDT Switch Single Pole, Double Throw. SPST - Switch Single Pole, Single Throw.
Some clarifications. Driver transforms Single ended signal to differential. DCA attenuates signal power from -1 to -31dB with step 1dB. Splitter splits signal for observation at the Test Point. Option optional attenuators. Number in the frame Signal power in this point (dBm). Number without frame Loss/Gain of this component (dB). There is component name under (or above) each component. Processor (CPU): Sends Control word to synthesizer through SPI. Produces digital modulation signal for transmitting. Transforms received analog modulated signal to digital. Measures BER. Transforms analog audio signal from microphone to digital. Transforms received audio digital signal to analog for speaker.
Possible exam questions Part 1.

1. Exercise, slide 20 2. Amplifiers, principal, advantages and disadvantages 2.1 Tunnel diode using, slide 25 2.2 Pamela, slides 27, 27 2.3 Klystron, slides 30 33 2.4 TWT, slides 34 36 2.5 Parametrical amplifier, slides 37, 38, 40 2.6 Why and when we prefer limiter amplifier, slide 26 3. Give definition for one of transmitter parameters and its measurement method , slides 42 44 4. Exercise, slide 46 5. Draw and explain AM modulator block diagram, slide 50 6. Draw and explain AM modulator by one transistor, slide 51 7. Draw and explain DSB modulator block diagram, slide 54 8. Draw and explain SSB modulator block diagram and its problems, slides 57 59 9. Exercise on FM by two sinusoidal signal, slides 69 70 10. S/N in FM, slides 71, 72 11. Direct method of FM, slides 73, 74.
Part 2.
1. Signal space, orthogonal signals, distance d, slides 81, 83 2. BPSK, QPSK, 16PSK comparison, slide 83, 84 3. Divisions methods and MPP problems, slides 86, 87 4. OFDMA system, slides 89, 90 5. OFDMA modem, slide 91 6. Universal IQ modulator, slide 98 7. Exercise, slide 105 8. Exercise, slide 107
Part 3.
1. 2. 3. 4. 5. 6. 7. 8. 9. Give definition for one of receiver parameters and its measurement method , slides 275 278 Sound Signal Distortion, slide 279, Ideal Operational Amplifier, slide 287, OA implementation example, slides 288 293. Superheterodyne and double Receiver, image rejection problems, slides 116 118, Exercise, slide 119 Balance multiplier, implementation problems, slides 120 123, Direct Up conversion using Hartley modulator 307, 309 Homodyne receiver and its problems, slides 310 - 313
10. First stages non linearity slides 165 166 11. Envelope detector, Weak signal mode, slides 212 214 12. Envelope detector, Strong signal mode, slides 215, 216 13. Coherent FM Detector, slide 223 14. Reference Oscillation Selectors for MPSK Modulation, slides 226, 228 15. Coherent digital receiving method, slides 234, 238 16. Optimal receiving by matching filter using slides 241 245 17. Difference encoding and decoding, slides 246, 251 18. PLL principal, slides 253,255, 257, 269 19. SSB detector problems 20. Costas loop, slide 271
*
The end.
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Transmission and Receiving Techniques

Prof. Michael Bank lectures

Rin,CC > Rin,CE

Introduction, Milestones in the history of radio

over long distances, although there is no independent evidence to support this

Band select filter

Micrometer (m) is 10-6m, nanometer (nm) is 10-9m, angstrom is 10-10m.

Relative Frequency Bandwidth

106 107 108 109 1010 1011 1012 1013 1014

Narrow frequency band is f/f0 << 1

One-sided normal law

There is uniform law for local station signals distributions.

In this case mean value is not median (!)

F ( x) x=1 = 1 e2.51 = 1 0.08 = 0.92.

Transistors Vacuum lamp Vi

B. Amplification by grouping of electrons.

Resonator device (Klystron)

Long term interaction device (TWT)

C. Amplification by pump generator.

D. Amplification by negative resistance using

A. Amplification by control of feeding source current. MOSFET Power Amplifier

Two stage power amplifier structure supports

Tunnel diode as a amplifier

Antenna current amplifying example

Pamela AM amplification with help limiters

One of possible amplifier Pamela realization

If Vin = Vin,max If Vin = Vin,min

and Doun = -60o+60o = 0o

Up = -60o-60o = -120o and Doun = 60o+60o = 120o

B. Amplification by grouping of electrons.

Particles flew out with different speed.

After some time they will be together

1 Star birth by Hubble

Real < 20%

n is harmonic number; X is parameter of grouping

is some percents and F (electrical) 0.5%, F (mechanical) 25%.

Catoptrical klystron (5mw 5W)

One resonator Reflector

Traveling-Wave Tube (TWT) - 1943

Electrons go through spiral and interact with spiral field.

3. Accelerating phase 2n till (2n + 1)

Decelerating phase (2n + 1) till 2(n + 1)

1. Long axis electromagnetic wave with phase velocity V

4. Electrons with velocity v0 = V

5. There is electrons grouping round electrons with phase 2n

C. Amplification by pump generator.

Let C = C 0 + C m Sin p t = C 0 (1 + mSin p t ) m =

fin; fP ; fin +/- fP

3. Analogue Modulations and Modulators Analogue Modulators

Inf. Spectrum Former

Basic Transmitter Parameters

Requirement to Out of band Transmitting

a (t ) = [ A0 + Ke (t ) ]Sin ( 0t + 0 ) = A(t ) Sin ( 0 t + 0 )

Amplitude Modulation Power

A02 r P0 = 2 Pmax Pmin

2 2 r A (t ) 2 2 r r = A0 1+ m Sin ( t + ) = A0 1+ 2 m sin( t + ) + m 2 [sin( t + ) ] = 2 2 2 2 2 r 1+ 2 m sin( t + ) + m = A0 [1 cos 2( t + ) ] 2 2 =0 =0

Peak factor or Peak Average Power Ratio (PARP) =

)]Sin ( 0 t + 0 ) = A 0 [Sin ( 0 t + 0 ) + mSin ( t + )Sin ( 0 t + 0 ) ] =

mA 0 Cos [( 0 + )t + 0 + ] + = A 0 Sin ( 0 t + 0 ) 2 mA 0 Cos [( 0 )t + 0 ] = + 2 mA 0 Sin ( 0 + )t + 0 = A 0 Sin ( 0 t + 0 ) + 2 2 +

Real Signal AM Modulation

1 a ( t ) = A0 Sin ( 0 t + 0 ) + K V ( ) C os ( t ( ) )Sin ( 0 t + 0 ) d = 0 1 V( ) = A0 Sin ( 0 t + 0 ) + K 2 Sin ( 0 + ) t + 0 ( ) d +

Multiplier (Balanced Modulator)

One Transistor AM Modulator