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Lecture Notes
Fall 2009
Dr. Neal Patwari
University of Utah
Department of Electrical and Computer Engineering
c _2006
ECE 5520 Fall 2009 2
Contents
1 Class Organization 8
2 Introduction 8
2.1 ”Executive Summary” . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
2.2 Why not Analog? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
2.3 Networking Stack . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
2.4 Channels and Media . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
2.5 Encoding / Decoding Block Diagram . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
2.6 Channels . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
2.7 Topic: Random Processes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
2.8 Topic: Frequency Domain Representations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
2.9 Topic: Orthogonality and Signal spaces . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
2.10 Related classes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
3 Power and Energy 13
3.1 DiscreteTime Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14
3.2 Decibel Notation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14
4 TimeDomain Concept Review 15
4.1 Periodicity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
4.2 Impulse Functions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
5 Bandwidth 16
5.1 Continuoustime Frequency Transforms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
5.1.1 Fourier Transform Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
5.2 Linear Time Invariant (LTI) Filters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
5.3 Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
6 Bandpass Signals 21
6.1 Upconversion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
6.2 Downconversion of Bandpass Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
7 Sampling 23
7.1 Aliasing Due To Sampling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24
7.2 Connection to DTFT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25
7.3 Bandpass sampling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
8 Orthogonality 28
8.1 Inner Product of Vectors . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28
8.2 Inner Product of Functions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
8.3 Deﬁnitions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
8.4 Orthogonal Sets . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
9 Orthonormal Signal Representations 32
9.1 Orthonormal Bases . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
9.2 Synthesis . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
9.3 Analysis . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
ECE 5520 Fall 2009 3
10 MultiVariate Distributions 37
10.1 Random Vectors . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38
10.2 Conditional Distributions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38
10.3 Simulation of Digital Communication Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39
10.4 Mixed Discrete and Continuous Joint Variables . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40
10.5 Expectation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42
10.6 Gaussian Random Variables . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42
10.6.1 Complementary CDF . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
10.6.2 Error Function . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
10.7 Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 44
10.8 Gaussian Random Vectors . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
10.8.1 Envelope . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
11 Random Processes 46
11.1 Autocorrelation and Power . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46
11.1.1 White Noise . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47
12 Correlation and MatchedFilter Receivers 47
12.1 Correlation Receiver . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 48
12.2 Matched Filter Receiver . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49
12.3 Amplitude . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50
12.4 Review . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50
12.5 Correlation Receiver . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
13 Optimal Detection 51
13.1 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
13.2 Bayesian Detection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 52
14 Binary Detection 52
14.1 Decision Region . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53
14.2 Formula for Probability of Error . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53
14.3 Selecting R
0
to Minimize Probability of Error . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53
14.4 LogLikelihood Ratio . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55
14.5 Case of a
0
= 0, a
1
= 1 in Gaussian noise . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55
14.6 General Case for Arbitrary Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
14.7 Equiprobable Special Case . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
14.8 Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
14.9 Review of Binary Detection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 57
15 Pulse Amplitude Modulation (PAM) 58
15.1 Baseband Signal Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58
15.2 Average Bit Energy in Mary PAM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59
16 Topics for Exam 1 60
ECE 5520 Fall 2009 4
17 Additional Problems 61
17.1 Spectrum of Communication Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61
17.2 Sampling and Aliasing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61
17.3 Orthogonality and Signal Space . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61
17.4 Random Processes, PSD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
17.5 Correlation / Matched Filter Receivers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
18 Probability of Error in Binary PAM 63
18.1 Signal Distance . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63
18.2 BER Function of Distance, Noise PSD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63
18.3 Binary PAM Error Probabilities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64
19 Detection with Multiple Symbols 66
20 Mary PAM Probability of Error 67
20.1 Symbol Error . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67
20.1.1 Symbol Error Rate and Average Bit Energy . . . . . . . . . . . . . . . . . . . . . . . . . . 67
20.2 Bit Errors and Gray Encoding . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 68
20.2.1 Bit Error Probabilities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 68
21 Intersymbol Interference 69
21.1 Multipath Radio Channel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
21.2 Transmitter and Receiver Filters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 70
22 Nyquist Filtering 70
22.1 Raised Cosine Filtering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 72
22.2 SquareRoot Raised Cosine Filtering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 72
23 Mary Detection Theory in Ndimensional signal space 72
24 Quadrature Amplitude Modulation (QAM) 74
24.1 Showing Orthogonality . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76
24.2 Constellation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77
24.3 Signal Constellations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 78
24.4 Angle and Magnitude Representation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79
24.5 Average Energy in MQAM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79
24.6 PhaseShift Keying . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79
24.7 Systems which use QAM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79
25 QAM Probability of Error 80
25.1 Overview of Future Discussions on QAM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 80
25.2 Options for Probability of Error Expressions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81
25.3 Exact Error Analysis . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81
25.4 Probability of Error in QPSK . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81
25.5 Union Bound . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 82
25.6 Application of Union Bound . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 82
25.6.1 General Formula for Union Boundbased Probability of Error . . . . . . . . . . . . . . . . 84
ECE 5520 Fall 2009 5
26 QAM Probability of Error 85
26.1 NearestNeighbor Approximate Probability of Error . . . . . . . . . . . . . . . . . . . . . . . . . 85
26.2 Summary and Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 86
27 Frequency Shift Keying 88
27.1 Orthogonal Frequencies . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
27.2 Transmission of FSK . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89
27.3 Reception of FSK . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
27.4 Coherent Reception . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
27.5 Noncoherent Reception . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
27.6 Receiver Block Diagrams . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91
27.7 Probability of Error for Coherent Binary FSK . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
27.8 Probability of Error for Noncoherent Binary FSK . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
27.9 FSK Error Probabilities, Part 2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94
27.9.1 Mary NonCoherent FSK . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94
27.9.2 Summary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 95
27.10Bandwidth of FSK . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 95
28 Frequency Multiplexing 95
28.1 Frequency Selective Fading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 95
28.2 Beneﬁts of Frequency Multiplexing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96
28.3 OFDM as an extension of FSK . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 97
29 Comparison of Modulation Methods 98
29.1 Diﬀerential Encoding for BPSK . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98
29.1.1 DPSK Transmitter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99
29.1.2 DPSK Receiver . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99
29.1.3 Probability of Bit Error for DPSK . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100
29.2 Points for Comparison . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100
29.3 Bandwidth Eﬃciency . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 101
29.3.1 PSK, PAM and QAM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 101
29.3.2 FSK . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 101
29.4 Bandwidth Eﬃciency vs.
E
b
N
0
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 102
29.5 Fidelity (P [error]) vs.
E
b
N
0
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 102
29.6 Transmitter Complexity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103
29.6.1 Linear / Nonlinear Ampliﬁers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103
29.7 Oﬀset QPSK . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 104
29.8 Receiver Complexity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
30 Link Budgets and System Design 105
30.1 Link Budgets Given C/N
0
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106
30.2 Power and Energy Limited Channels . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108
30.3 Computing Received Power . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 110
30.3.1 Free Space . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 110
30.3.2 Nonfreespace Channels . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
30.3.3 Wired Channels . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
30.4 Computing Noise Energy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112
ECE 5520 Fall 2009 6
30.5 Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112
31 Timing Synchronization 113
32 Interpolation 114
32.1 Sampling Time Notation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 115
32.2 Seeing Interpolation as Filtering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 116
32.3 Approximate Interpolation Filters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 116
32.4 Implementations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 117
32.4.1 Higher order polynomial interpolation ﬁlters . . . . . . . . . . . . . . . . . . . . . . . . . . 117
33 Final Project Overview 118
33.1 Review of Interpolation Filters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119
33.2 Timing Error Detection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119
33.3 Earlylate timing error detector (ELTED) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 120
33.4 Zerocrossing timing error detector (ZCTED) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121
33.4.1 QPSK Timing Error Detection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 122
33.5 Voltage Controlled Clock (VCC) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 122
33.6 Phase Locked Loops . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 123
33.6.1 Phase Detector . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 123
33.6.2 Loop Filter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 124
33.6.3 VCO . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 124
33.6.4 Analysis . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 124
33.6.5 DiscreteTime Implementations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 125
34 Exam 2 Topics 126
35 Source Coding 127
35.1 Entropy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 128
35.2 Joint Entropy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 130
35.3 Conditional Entropy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 130
35.4 Entropy Rate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 131
35.5 Source Coding Theorem . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
36 Review 133
37 Channel Coding 133
37.1 R. V. L. Hartley . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 133
37.2 C. E. Shannon . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 134
37.2.1 Noisy Channel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 134
37.2.2 Introduction of Latency . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 134
37.2.3 Introduction of Power Limitation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 135
37.3 Combining Two Results . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 135
37.3.1 Returning to Hartley . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 135
37.3.2 Final Results . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 135
37.4 Eﬃciency Bound . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 136
38 Review 138
ECE 5520 Fall 2009 7
39 Channel Coding 138
39.1 R. V. L. Hartley . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 138
39.2 C. E. Shannon . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 139
39.2.1 Noisy Channel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 139
39.2.2 Introduction of Latency . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 139
39.2.3 Introduction of Power Limitation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 139
39.3 Combining Two Results . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 140
39.3.1 Returning to Hartley . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 140
39.3.2 Final Results . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 141
39.4 Eﬃciency Bound . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 141
ECE 5520 Fall 2009 8
Lecture 1
Today: (1) Syllabus (2) Intro to Digital Communications
1 Class Organization
Textbook Few textbooks cover solely digital communications (without analog) in an introductory communi
cations course. But graduates today will almost always encounter / be developing solely digital communication
systems. So half of most textbooks are useless; and that the other half is sparse and needs supplemental material.
For example, the past text was the ‘standard’ text in the area for an undergraduate course, Proakis & Salehi,
J.G. Proakis and M. Salehi, Communication Systems Engineering, 2nd edition, Prentice Hall, 2001. Students
didn’t like that I had so many supplemental readings. This year’s text covers primarily digital communications,
and does it in depth. Finally, I ﬁnd it to be very wellwritten. And, there are few options in this area. I will
provide additional readings solely to provide another presentation style or ﬁt another learning style. Unless
speciﬁed, these are optional.
Lecture Notes I type my lecture notes. I have taught ECE 5520 previously and have my lecture notes from
past years. I’m constantly updating my notes, even up to the lecture time. These can be available to you at
lecture, and/or after lecture online. However, you must accept these conditions:
1. Taking notes is important: I ﬁnd most learning requires some writing on your part, not just watching.
Please take your own notes.
2. My written notes do not and cannot reﬂect everything said during lecture: I answer questions
and understand your perspective better after hearing your questions, and I try to tailor my approach
during the lecture. If I didn’t, you could just watch a recording!
2 Introduction
A digital communication system conveys discretetime, discretevalued information across a physical channel.
Information sources might include audio, video, text, or data. They might be continuoustime (analog) signals
(audio, images) and even 1D or 2D. Or, they may already be digital (discretetime, discretevalued). Our
object is to convey the signals or data to another place (or time) with as faithful representation as possible.
In this section we talk about what we’ll cover in this class, and more importantly, what we won’t cover.
2.1 ”Executive Summary”
Here is the one sentence version: We will study how to eﬃciently encode digital data on a noisy, bandwidth
limited analog medium, so that decoding the data (i.e., reception) at a receiver is simple, eﬃcient, and high
ﬁdelity.
The keys points stuﬀed into that one sentence are:
1. Digital information on an analog medium: We can send waveforms, i.e., realvalued, continuoustime
functions, on the channel (medium). These waveforms are from a discrete set of possible waveforms.
What set of waveforms should we use? Why?
ECE 5520 Fall 2009 9
2. Decoding the data: When receiving a signal (a function) in noise, none of the original waveforms will
match exactly. How do you make a decision about which waveform was sent?
3. What makes a receiver diﬃcult to realize? What choices of waveforms make a receiver simpler to imple
ment? What techniques are used in a receiver to compensate?
4. Eﬃciency, Bandwidth, and Fidelity: Fidelity is the correctness of the received data (i.e., the opposite of
error rate). What is the tradeoﬀ between energy, bandwidth, and ﬁdelity? We all want high ﬁdelity, and
low energy consumption and bandwidth usage (the costs of our communication system).
You can look at this like an impedance matching problem from circuits. You want, for power eﬃciency, to
have the source impedance match the destination impedance. In digital comm, this means that we want our
waveform choices to match the channel and receiver to maximize the eﬃciency of the communication system.
2.2 Why not Analog?
The previous text used for this course, by Proakis & Salehi, has an extensive analysis and study of analog
communication systems, such as radio and television broadcasting (Chapter 3). In the recent past, this course
would study both analog and digital communication systems. Analog systems still exist and will continue to
exist; however, development of new systems will almost certainly be of digital communication systems. Why?
• Fidelity
• Energy: transmit power, and device power consumption
• Bandwidth eﬃciency: due to coding gains
• Moore’s Law is decreasing device costs for digital hardware
• Increasing need for digital information
• More powerful information security
2.3 Networking Stack
In this course, we study digital communications from bits to bits. That is, we study how to take ones and zeros
from a transmitter, send them through a medium, and then (hopefully) correctly identify the same ones and
zeros at the receiver. There’s a lot more than this to the digital communication systems which you use on a
daily basis (e.g., iPhone, WiFi, Bluetooth, wireless keyboard, wireless car key).
To manage complexity, we (engineers) don’t try to build a system to do everything all at once. We typically
start with an application, and we build a layered network to handle the application. The 7layer OSI stack,
which you would study in a CS computer networking class, is as follows:
• Application
• Presentation (*)
• Session (*)
• Transport
• Network
ECE 5520 Fall 2009 10
• Link Layer
• Physical (PHY) Layer
(Note that there is also a 5layer model in which * layers are considered as part of the application layer.) ECE
5520 is part of the bottom layer, the physical layer. In fact, the physical layer has much more detail. It is
primarily divided into:
• Multiple Access Control (MAC)
• Encoding
• Channel / Medium
We can control the MAC and the encoding chosen for a digital communication.
2.4 Channels and Media
We can chose from a few media, but we largely can’t change the properties of the medium (although there are
exceptions). Here are some media:
• EM Spectra: (anything above 0 Hz) Radio, Microwave, mmwave bands, light
• Acoustic: ultrasound
• Transmission lines, waveguides, optical ﬁber, coaxial cable, wire pairs, ...
• Disk (data storage applications)
2.5 Encoding / Decoding Block Diagram
Information
Source
Source
Encoder
Channel
Encoder
Modulator
Channel
Information
Output
Source
Decoder
Channel
Decoder
Demodulator
Upconversion
Down
conversion
OtherComponents:
Digitalto Analog
Converter
AnalogtoDigital
Converter
Synchron
ization
Figure 1: Block diagram of a singleuser digital communication system, including (top) transmitter, (middle)
channel, and (bottom) receiver.
Notes:
• Information source comes from higher networking layers. It may be continuous or packetized.
• Source Encoding: Finding a compact digital representation for the data source. Includes sampling of
continuoustime signals, and quantization of continuousvalued signals. Also includes compression of
those sources (lossy, or lossless). What are some compression methods that you’re familiar with? We
present an introduction to source encoding at the end of this course.
ECE 5520 Fall 2009 11
• Channel encoding refers to redundancy added to the signal such that any bit errors can be corrected.
A channel decoder, because of the redundancy, can correct some bit errors. We will not study channel
encoding, but it is a topic in the (ECE 6520) Coding Theory.
• Modulation refers to the digitaltoanalog conversion which produces a continuoustime signal that can be
sent on the physical channel. It is analogous to impedance matching  proper matching of a modulation to
a channel allows optimal information transfer, like impedance matching ensured optimal power transfer.
Modulation and demodulation will be the main focus of this course.
• Channels: See above for examples. Typical models are additive noise, or linear ﬁltering channel.
Why do we do both source encoding (which compresses the signal as much as possible) and also channel
encoding (which adds redundancy to the signal)? Because of Shannon’s sourcechannel coding separation
theorem. He showed that (given enough time) we can consider them separately without additional loss. And
separation, like layering, reduces complexity to the designer.
2.6 Channels
A channel can typically be modeled as a linear ﬁlter with the addition of noise. The noise comes from a variety
of sources, but predominantly:
1. Thermal background noise: Due to the physics of living above 0 Kelvin. Well modeled as Gaussian, and
white; thus it is referred to as additive white Gaussian noise (AWGN).
2. Interference from other transmitted signals. These other transmitters whose signals we cannot completely
cancel, we lump into the ‘interference’ category. These may result in nonGaussian noise distribution, or
nonwhite noise spectral density.
The linear ﬁltering of the channel result from the physics and EM of the medium. For example, attenuation in
telephone wires varies by frequency. Narrowband wireless channels experience fading that varies quickly as a
function of frequency. Wideband wireless channels display multipath, due to multiple timedelayed reﬂections,
diﬀractions, and scattering of the signal oﬀ of the objects in the environment. All of these can be modeled as
linear ﬁlters.
The ﬁlter may be constant, or timeinvariant, if the medium, the TX and RX do not move or change.
However, for mobile radio, the channel may change very quickly over time. Even for stationary TX and RX, in
real wireless channels, movement of cars, people, trees, etc. in the environment may change the channel slowly
over time.
Transmitted
Signal
LTIFilter
h(t)
Noise
Received
Signal
Figure 2: Linear ﬁlter and additive noise channel model.
In this course, we will focus primarily on the AWGN channel, but we will mention what variations exist for
particular channels, and how they are addressed.
ECE 5520 Fall 2009 12
2.7 Topic: Random Processes
Random things in a communication system:
• Noise in the channel
• Signal (bits)
• Channel ﬁltering, attenuation, and fading
• Device frequency, phase, and timing oﬀsets
These random signals often pass through LTI ﬁlters, and are sampled. We want to build the best receiver
possible despite the impediments. Optimal receiver design is something that we study using probability theory.
We have to tolerate errors. Noise and attenuation of the channel will cause bit errors to be made by the
demodulator and even the channel decoder. This may be tolerated, or a higher layer networking protocol (eg.,
TCPIP) can determine that an error occurred and then rerequest the data.
2.8 Topic: Frequency Domain Representations
To ﬁt as many signals as possible onto a channel, we often split the signals by frequency. The concept of
sharing a channel is called multiple access (MA). Separating signals by frequency band is called frequency
division multiple access (FDMA). For the wireless channel, this is controlled by the FCC (in the US) and called
spectrum allocation. There is a tradeoﬀ between frequency requirements and time requirements, which will be
a major part of this course. The Fourier transform of our modulated, transmitted signal is used to show that
it meets the spectrum allocation limits of the FCC.
2.9 Topic: Orthogonality and Signal spaces
To show that signals sharing the same channel don’t interfere with each other, we need to show that they are
orthogonal. This means, in short, that a receiver can uniquely separate them. Signals in diﬀerent frequency
bands are orthogonal.
We can also employ multiple orthogonal signals in a single transmitter and receiver, in order to provide
multiple independent means (dimensions) on which to modulate information. We will study orthogonal signals,
and learn an algorithm to take an arbitrary set of signals and output a set of orthogonal signals with which to
represent them. We’ll use signal spaces to show graphically the results, as the example in Figure 36.
M=8 M=16
Figure 3: Example signal space diagram for Mary Phase Shift Keying, for (a) M = 8 and (b) M = 16. Each
point is a vector which can be used to send a 3 or 4 bit sequence.
ECE 5520 Fall 2009 13
2.10 Related classes
1. Prerequisites: (ECE 5510) Random Processes; (ECE 3500) Signals and Systems.
2. Signal Processing: (ECE 5530): Digital Signal Processing
3. Electromagnetics: EM Waves, (ECE 53205321) Microwave Engineering, (ECE 5324) Antenna Theory,
(ECE 5411) Fiberoptic Systems
4. Breadth: (ECE 5325) Wireless Communications
5. Devices and Circuits: (ECE 3700) Fundamentals of Digital System Design, (ECE 5720) Analog IC Design
6. Networking: (ECE 5780) Embedded System Design, (CS 5480) Computer Networks
7. Advanced Classes: (ECE 6590) Software Radio, (ECE 6520) Information Theory and Coding, (ECE 6540):
Estimation Theory
Lecture 2
Today: (1) Power, Energy, dB (2) Timedomain concepts (3) Bandwidth, Fourier Transform
Two of the biggest limitations in communications systems are (1) energy / power; and (2) bandwidth. Today’s
lecture provides some tools to deal with power and energy, and starts the review of tools to analyze frequency
content and bandwidth.
3 Power and Energy
Recall that energy is power times time. Use the units: energy is measured in Joules (J); power is measured in
Watts (W) which is the same as Joules/second (J/sec). Also, recall that our standard in signals and systems is
deﬁne our signals, such as x(t), as voltage signals (V). When we want to know the power of a signal we assume
it is being dissipated in a 1 Ohm resistor, so [x(t)[
2
is the power dissipated at time t (since power is equal to
the voltage squared divided by the resistance).
A signal x(t) has energy deﬁned as
E =
_
∞
−∞
[x(t)[
2
dt
For some signals, E will be inﬁnite because the signal is nonzero for an inﬁnite duration of time (it is always
on). These signals we call power signals and we compute their power as
P = lim
T→∞
1
2T
_
T
−T
[x(t)[
2
dt
The signal with ﬁnite energy is called an energy signal.
ECE 5520 Fall 2009 14
3.1 DiscreteTime Signals
In this book, we refer to discrete samples of the sampled signal x as x(n). You may be more familiar with the
x[n] notation. But, Matlab uses parentheses also; so we’ll follow the Rice text notation. Essentially, whenever
you see a function of n (or k, l, m), it is a discretetime function; whenever you see a function of t (or perhaps
τ) it is a continuoustime function. I’m sorry this is not more obvious in the notation.
For discretetime signals, energy and power are deﬁned as:
E =
∞
n=−∞
[x(n)[
2
(1)
P = lim
N→∞
1
2N + 1
N
n=−N
[x(n)[
2
(2)
3.2 Decibel Notation
We often use a decibel (dB) scale for power. If P
lin
is the power in Watts, then
[P]
dBW
= 10 log
10
P
lin
Decibels are more general  they can apply to other unitless quantities as well, such as a gain (loss) L(f) through
a ﬁlter H(f),
[L(f)]
dB
= 10 log
10
[H(f)[
2
(3)
Note: Why is the capital B used? Either the lowercase ‘b’ in the SI system is reserved for bits, so when the
‘bel’ was ﬁrst proposed as log
10
(), it was capitalized; or it referred to a name ‘Bell’ so it was capitalized. In
either case, we use the unit decibel 10 log
10
() which is then abbreviated as dB in the SI system.
Note that (3) could also be written as:
[L(f)]
dB
= 20 log
10
[H(f)[ (4)
Be careful with your use of 10 vs. 20 in the dB formula.
• Only use 20 as the multiplier if you are converting from voltage to power; i.e., taking the log
10
of a voltage
and expecting the result to be a dB power value.
Our standard is to consider power gains and losses, not voltage gains and losses. So if we say, for example,
the channel has a loss of 20 dB, this refers to a loss in power. In particular, the output of the channel has 100
times less power than the input to the channel.
Remember these two dB numbers:
• 3 dB: This means the number is double in linear terms.
• 10 dB: This means the number is ten times in linear terms.
And maybe this one:
• 1 dB: This means the number is a little over 25% more (multiply by 5/4) in linear terms.
With these three numbers, you can quickly convert losses or gains between linear and dB units without a
calculator. Just convert any dB number into a sum of multiples of 10, 3, and 1.
Example: Convert dB to linear values:
ECE 5520 Fall 2009 15
1. 30 dBW
2. 33 dBm
3. 20 dB
4. 4 dB
Example: Convert linear values to dB:
1. 0.2 W
2. 40 mW
Example: Convert power relationships to dB:
Convert the expression to one which involves only dB terms.
1. P
y,lin
= 100P
x,lin
2. P
o,lin
= G
connector,lin
L
−d
cable,lin
, where P
o,lin
is the received power in a ﬁberoptic link, where d is the cable
length (typically in units of km), G
connector,lin
is the gain in any connectors, and L
cable,lin
is a loss in a 1
km cable.
3. P
r,lin
= P
t,lin
G
t,lin
G
t,lin
λ
2
(4πd)
2
, where λ is the wavelength (m), d is the path length (m), and G
t,lin
and G
t,lin
are the linear gains in the antennas, P
t,lin
is the transmit power (W) and P
r,lin
is the received power (W).
This is the Friis free space path loss formula.
These last two are what we will need in Section 6.4, when we discuss link budgets. The main idea is that
we have a limited amount of power which will be available at the receiver.
4 TimeDomain Concept Review
4.1 Periodicity
Def ’n: Periodic (continuoustime)
A signal x(t) is periodic if x(t) = x(t +T
0
) for some constant T
0
,= 0 for all t ∈ R. The smallest such constant
T
0
> 0 is the period.
If a signal is not periodic it is aperiodic.
Periodic signals have Fourier series representations, as deﬁned in Rice Ch. 2.
Def ’n: Periodic (discretetime)
A DT signal x(n) is periodic if x(n) = x(n + N
0
) for some integer N
0
,= 0, for all integers n. The smallest
positive integer N
0
is the period.
ECE 5520 Fall 2009 16
4.2 Impulse Functions
Def ’n: Impulse Function
The (Dirac) impulse function δ(t) is the function which makes
_
∞
−∞
x(t)δ(t)dt = x(0) (5)
true for any function x(t) which is continuous at t = 0.
We are deﬁning a function by its most important property, the ‘sifting property’. Is there another deﬁnition
which is more familiar?
Solution:
δ(t) = lim
T→0
_
1/T, −T ≤ t ≤ T
0, o.w.
You can visualize δ(t) here as an inﬁnitely high, inﬁnitesimally wide pulse at the origin, with area one. This is
why it ‘pulls out’ the value of x(t) in the integral in (5).
Other properties of the impulse function:
• Time scaling,
• Symmetry,
• Sifting at arbitrary time t
0
,
The continuoustime unit step function is
u(t) =
_
1, t ≥ 0
0, o.w.
Example: Sifting Property
What is
_
∞
−∞
sin(πt)
πt
δ(1 −t)dt?
The discretetime impulse function (also called the Kronecker delta or δ
K
) is deﬁned as:
δ(n) =
_
1, n = 0
0, o.w.
(There is no need to get complicated with the math; this is well deﬁned.) Also,
u(n) =
_
1, n ≥ 0
0, o.w.
5 Bandwidth
Bandwidth is another critical resource for a digital communications system; we have various deﬁnitions to
quantify it. In short, it isn’t easy to describe a signal in the frequency domain with a single number. And, in
the end, a system will be designed to meet a spectral mask required by the FCC or system standard.
ECE 5520 Fall 2009 17
Periodicity
Time Periodic Aperiodic
ContinuousTime Laplace Transform
x(t) ↔ X(s)
Fourier Series Fourier Transform
x(t) ↔ a
k
x(t) ↔ X(jω)
X(jω) =
_
∞
t=−∞
x(t)e
−jωt
dt
x(t) =
1
2π
_
∞
ω=−∞
X(jω)e
jωt
dω
DiscreteTime zTransform
x(n) ↔ X(z)
Discrete Fourier Transform (DFT) Discrete Time Fourier Transform (DTFT)
x(n) ↔ X[k] x(n) ↔ X(e
jΩ
)
X[k] =
N−1
n=0
x(n)e
−j
2π
N
kn
X(e
jΩ
) =
∞
n=−∞
x(n)e
−jΩn
x(n) =
1
N
N−1
k=0
X[k]e
j
2π
N
nk
x(n) =
1
2π
_
π
n=−π
X(e
jΩ
)dΩ
Table 1: Frequency Transforms
Intuitively, bandwidth is the maximum extent of our signal’s frequency domain characterization, call it X(f).
A baseband signal absolute bandwidth is often deﬁned as the W such that X(f) = 0 for all f except for the
range −W ≤ f ≤ W. Other deﬁnitions for bandwidth are
• 3dB bandwidth: B
3dB
is the value of f such that [X(f)[
2
= [X(0)[
2
/2.
• 90% bandwidth: B
90%
is the value which captures 90% of the energy in the signal:
_
B
90%
−B
90%
[X(f)[
2
df = 0.90
_
∞
=−∞
Xd[(f)[
2
df
As a motivating example, I mention the squareroot raised cosine (SRRC) pulse, which has the following
desirable Fourier transform:
H
RRC
(f) =
_
¸
¸
_
¸
¸
_
√
T
s
, 0 ≤ [f[ ≤
1−α
2Ts _
Ts
2
_
1 + cos
_
πTs
α
_
[f[ −
1−α
2Ts
___
,
1−α
2Ts
≤ [f[ ≤
1+α
2Ts
0, o.w.
(6)
where α is a parameter called the “rolloﬀ factor”. We can actually analyze this using the properties of the
Fourier transform and many of the standard transforms you’ll ﬁnd in a Fourier transform table.
The SRRC and other pulse shapes are discussed in Appendix A, and we will go into more detail later on.
The purpose so far is to motivate practicing up on frequency transforms.
5.1 Continuoustime Frequency Transforms
Notes about continuoustime frequency transforms:
1. You are probably most familiar with the Laplace Transform. To convert it to the Fourier tranform, we
replace s with jω, where ω is the radial frequency, with units radians per second (rad/s).
ECE 5520 Fall 2009 18
2. You may prefer the radial frequency representation, but also feel free to use the rotational frequency f
(which has units of cycles per sec, or Hz. Frequency in Hz is more standard for communications; you
should use it for intuition. In this case, just substitute ω = 2πf. You could write X(j2πf) as the notation
for this, but typically you’ll see it abbreviated as X(f). Note that the deﬁnition of the Fourier tranform
in the f domain loses the
1
2π
in the inverse Fourier transform deﬁnition.s
X(j2πf) =
_
∞
t=−∞
x(t)e
−j2πft
dt
x(t) =
_
∞
f=−∞
X(j2πf)e
j2πft
df
(7)
3. The Fourier series is limited to purely periodic signals. Both Laplace and Fourier transforms are not
limited to periodic signals.
4. Note that e
jα
= cos(α) +j sin(α).
See Table 2.4.4 in the Rice book.
Example: Square Wave
Given a rectangular pulse x(t) = rect(t/T
s
),
x(t) =
_
1, −T
s
/2 < t ≤ T
s
/2
0, o.w.
What is the Fourier transform X(f)? Calculate both from the deﬁnition and from a table.
Solution: Method 1: From the deﬁnition:
X(jω) =
_
Ts/2
t=−Ts/2
e
−jωt
dt
=
1
−jω
e
−jωt
¸
¸
¸
¸
Ts/2
t=−Ts/2
=
1
−jω
_
e
−jωTs/2
−e
jωTs/2
_
= 2
sin(ωT
s
/2)
ω
= T
s
sin(ωT
s
/2)
ωT
s
/2
This uses the fact that
1
−2j
_
e
−jα
−e
jα
_
= sin(α). While it is sometimes convenient to replace sin(πx)/(πx) with
sincx, it is confusing because sinc(x) is sometimes deﬁned as sin(πx)/(πx) and sometimes deﬁned as (sin x)/x.
No standard deﬁnition for ‘sinc’ exists! Rather than make a mistake because of this, the Rice book always
writes out the expression fully. I will try to follow suit.
Method 2: From the tables and properties:
x(t) = g(2t) where g(t) =
_
1, [t[ < T
s
0, o.w.
(8)
From the table, G(jω) = 2T
s
sin(ωTs)
ωTs
. From the properties, X(jω) =
1
2
G
_
j
ω
2
_
. So
X(jω) = T
s
sin(ωT
s
/2)
ωT
s
/2
ECE 5520 Fall 2009 19
See Figure 4(a).
(a)
−4/Ts −3/Ts −2/Ts −1/Ts 0 1/Ts 2/Ts 3/Ts 4/Ts
0
Ts/2
Ts
f
T
s
s
i
n
c
(
π
f
T
s
)
(b)
−4/Ts −3/Ts −2/Ts −1/Ts 0 1/Ts 2/Ts 3/Ts 4/Ts
−50
−40
−30
−20
−10
0
f
2
0
l
o
g
1
0
(
X
(
f
)
/
T
s
)
Figure 4: (a) Fourier transform X(j2πf) of rect pulse with period T
s
, and (b) Power vs. frequency
20 log
10
(X(j2πf)/T
s
).
Question: What if Y (jω) was a rect function? What would the inverse Fourier transform y(t) be?
5.1.1 Fourier Transform Properties
See Table 2.4.3 in the Rice book.
Assume that F¦x(t)¦ = X(jω). Important properties of the Fourier transform:
1. Duality property:
x(jω) = F¦X(−t)¦
x(−jω) = F¦X(t)¦
(Confusing. It says is that you can go backwards on the Fourier transform, just remember to ﬂip the
result around the origin.)
2. Time shift property:
F¦x(t −t
0
)¦ = e
−jωt
0
X(jω)
ECE 5520 Fall 2009 20
3. Scaling property: for any real a ,= 0,
F¦x(at)¦ =
1
[a[
X
_
j
ω
a
_
4. Convolution property: if, additionally y(t) has Fourier transform X(jω),
F¦x(t) ⋆ y(t)¦ = X(jω) Y (jω)
5. Modulation property:
F¦x(t)cos(ω
0
t)¦ =
1
2
X(ω −ω
0
) +
1
2
X(ω +ω
0
)
6. Parceval’s theorem: The energy calculated in the frequency domain is equal to the energy calculated in
the time domain.
_
∞
t=−∞
[x(t)[
2
dt =
_
∞
f=−∞
[X(f)[
2
df =
1
2π
_
∞
ω=−∞
[X(jω)[
2
dω
So do whichever one is easiest! Or, check your answer by doing both.
5.2 Linear Time Invariant (LTI) Filters
If a (deterministic) signal x(t) is input to a LTI ﬁlter with impulse response h(t), the output signal is
y(t) = h(t) ⋆ x(t)
Using the above convolution property,
Y (jω) = X(jω) H(jω)
5.3 Examples
Example: Applying FT Properties
If w(t) has the Fourier transform
W(jω) =
jω
1 +jω
ﬁnd X(jω) for the following waveforms:
1. x(t) = w(2t + 2)
2. x(t) = e
−jt
w(t −1)
3. x(t) = 2
∂w(t)
∂t
4. x(t) = w(1 −t)
Solution: To be worked out in class.
Lecture 3
Today: (1) Bandpass Signals (2) Sampling
Solution: From the Examples at the end of Lecture 2:
ECE 5520 Fall 2009 21
1. Let x(t) = z(2t) where z(t) = w(t + 2). Then
Z(jω) = e
j2ω
jω
1 +jω
, and X(jω) =
1
2
Z(jω/2).
So X(jω) =
1
2
e
jω
jω/2
1+jω/2
. Alternatively, let x(t) = r(t + 1) where r(t) = w(2t). Then
R(jω) =
1
2
W(jω/2), and X(jω) = e
jω
R(jω).
Again, X(jω) =
1
2
e
jω
jω/2
1+jω/2
.
2. Let x(t) = e
−jt
z(t) where z(t) = w(t − 1). Then Z(ω) = e
−jω
W(jω), and X(jω) = Z(j(ω − 1)). So
X(jω) = e
−j(ω−1)
W(j(ω −1)) = e
−j(ω+1)
j(ω+1)
1+j(ω+1)
.
3. X(jω) = 2jω
jω
1+jω
= −
2ω
2
1+jω
.
4. Let x(t) = y(−t), where y(t) = w(1 +t). Then
X(jω) = Y (−jω), where Y (jω) = e
jω
W(jω) = e
jω
jω
1 +jω
So X(jω) = e
−jω −jω
1−jω
.
6 Bandpass Signals
Def ’n: Bandpass Signal
A bandpass signal x(t) has Fourier transform X(jω) = 0 for all ω such that [ω ±ω
c
[ > W/2, where W/2 < ω
c
.
Equivalently, a bandpass signal x(t) has Fourier transform X(f) = 0 for all f such that [f ± f
c
[ > B/2, where
B/2 < f
c
.
f
c
B
X f ( )
f
c
B
Figure 5: Bandpass signal with center frequency f
c
.
Realistically, X(jω) will not be exactly zero outside of the bandwidth, there will be some ‘sidelobes’ out of
the main band.
6.1 Upconversion
We can take a baseband signal x
C
(t) and ‘upconvert’ it to be a bandpass ﬁlter by multiplying with a cosine at
the desired center frequency, as shown in Fig. 6.
ECE 5520 Fall 2009 22
x t
C
( )
cos(2 ) pf t
c
Figure 6: Modulator block diagram.
Example: Modulated Square Wave
We dealt last time with the square wave x(t) = rect(t/T
s
). This time, we will look at:
x(t) = rect(t/T
s
) cos(ω
c
t)
where ω
c
is the center frequency in radians/sec (f
c
= ω
c
/(2π) is the center frequency in Hz). Note that I use
“rect” as follows:
rect(t) =
_
1, −1/2 < t ≤ 1/2
0, o.w.
What is X(jω)?
Solution:
X(jω) = F¦rect(t/T
s
)¦ ⋆ F¦cos(ω
c
t)¦
X(jω) =
1
2π
T
s
sin(ωT
s
/2)
ωT
s
/2
⋆ π [δ(ω −ω
c
) +δ(ω +ω
c
)]
X(jω) =
T
s
2
sin[(ω −ω
c
)T
s
/2]
(ω −ω
c
)T
s
/2
+
T
s
2
sin[(ω +ω
c
)T
s
/2]
(ω +ω
c
)T
s
/2
The plot of X(j2πf) is shown in Fig. 7.
f
c
X f ( )
f
c
Figure 7: Plot of X(j2πf) for modulated square wave example.
6.2 Downconversion of Bandpass Signals
How will we get back the desired baseband signal x
C
(t) at the receiver?
1. Multiply (again) by the carrier 2 cos(ω
c
t).
2. Input to a lowpass ﬁlter (LPF) with cutoﬀ frequency ω
c
.
This is shown in Fig. 8.
ECE 5520 Fall 2009 23
2cos(2 ) pf t
c
x t ( )
LPF
Figure 8: Demodulator block diagram.
Assume the general modulated signal was,
x(t) = x
C
(t) cos(ω
c
t)
X(jω) =
1
2
X
C
(j(ω −ω
c
)) +
1
2
X
C
(j(ω +ω
c
))
What happens after the multiplication with the carrier in the receiver?
Solution:
X
1
(jω) =
1
2π
X(jω) ⋆ 2π [δ(ω −ω
c
) +δ(ω +ω
c
)]
X
1
(jω) =
1
2
X
C
(j(ω −2ω
c
)) +
1
2
X
C
(jω) +
1
2
X
C
(jω) +
1
2
X
C
(j(ω + 2ω
c
))
There are components at 2ω
c
and −2ω
c
which will be canceled by the LPF. (Does the LPF need to be ideal?)
X
2
(jω) = X
C
(jω)
So x
2
(t) = x
C
(t).
7 Sampling
A common statement of the Nyquist sampling theorem is that a signal can be sampled at twice its bandwidth.
But the theorem really has to do with signal reconstruction from a sampled signal.
Theorem: (Nyquist Sampling Theorem.) Let x
c
(t) be a baseband, continuous signal with bandwidth B (in
Hz), i.e., X
c
(jω) = 0 for all [ω[ ≥ 2πB. Let x
c
(t) be sampled at multiples of T, where
1
T
≥ 2B to yield the
sequence ¦x
c
(nT)¦
∞
n=−∞
. Then
x
c
(t) = 2BT
∞
n=−∞
x
c
(nT)
sin(2πB(t −nT))
2πB(t −nT)
. (9)
Proof: Not covered.
Notes:
• This is an interpolation procedure.
• Given ¦x
n
¦
∞
n=−∞
, how would you ﬁnd the maximum of x(t)?
• This is only precise when X(jω) = 0 for all [ω[ ≥ 2πB.
ECE 5520 Fall 2009 24
7.1 Aliasing Due To Sampling
Essentially, sampling is the multiplication of a impulse train (at period T with the desired signal x(t):
x
sa
(t) = x(t)
∞
n=−∞
δ(t −nT)
x
sa
(t) =
∞
n=−∞
x(nT)δ(t −nT)
What is the Fourier transform of x
sa
(t)?
Solution: In the frequency domain, this is a convolution:
X
sa
(jω) = X(jω) ⋆
2π
T
∞
n=−∞
δ
_
ω −
2πn
T
_
=
2π
T
∞
n=−∞
X
_
j
_
ω −
2πn
T
__
for all ω (10)
=
1
T
X(jω) for [ω[ < 2πB
This is shown graphically in the Rice book in Figure 2.12.
The Fourier transform of the sampled signal is many copies of X(jω) strung at integer multiples of 2π/T,
as shown in Fig. 9.
Figure 9: The eﬀect of sampling on the frequency spectrum in terms of frequency f in Hz.
Example: Sinusoid sampled above and below Nyquist rate
Consider two sinusoidal signals sampled at 1/T = 25 Hz:
x
1
(nT) = sin(2π5nT)
x
2
(nT) = sin(2π20nT)
What are the two frequencies of the sinusoids, and what is the Nyquist rate? Which of them does the Nyquist
theorem apply to? Draw the spectrums of the continuous signals x
1
(t) and x
2
(t), and indicate what the spectrum
is of the sampled signals.
Figure 10 shows what happens when the Nyquist theorem is applied to the each signal (whether or not it is
valid). What observations would you make about Figure 10(b), compared to Figure 10(a)?
Example: Square vs. round pulse shape
Consider the square pulse considered before, x
1
(t) = rect(t/T
s
). Also consider a parabola pulse (this doesn’t
ECE 5520 Fall 2009 25
(a)
−1 −0.5 0 0.5 1
−1
−0.5
0
0.5
1
1.5
Time
x
(
t
)
Sampled
Interp.
(b)
−1 −0.5 0 0.5 1
−1
−0.5
0
0.5
1
1.5
Time
x
(
t
)
Sampled
Interp.
Figure 10: Sampled (a) x
1
(nT) and (b) x
2
(nT) are interpolated (—) using the Nyquist interpolation formula.
really exist in the wild – I’m making it up for an example.)
x
2
(t) =
_
1 −
_
2t
Ts
_
2
, −
1
2Ts
≤ t ≤
1
2Ts
0, o.w.
What happens to x
1
(t) and x
2
(t) when they are sampled at rate T?
In the Matlab code EgNyquistInterpolation.m we set T
s
= 1/5 and T = 1/25. Then, the sampled pulses
are interpolated using (9). Even though we’ve sampled at a pretty high rate, the reconstructed signals will not
be perfect representations of the original, in particular for x
1
(t). See Figure 11.
7.2 Connection to DTFT
Recall (10). We calculated the Fourier transform of the product by doing the convolution in the frequency
domain. Instead, we can calculate it directly from the formula.
ECE 5520 Fall 2009 26
(a)
−1 −0.5 0 0.5 1
−0.2
0
0.2
0.4
0.6
0.8
1
Time
x
(
t
)
Sampled
Interp.
(b)
−1 −0.5 0 0.5 1
−0.2
0
0.2
0.4
0.6
0.8
1
Time
x
(
t
)
Sampled
Interp.
Figure 11: Sampled (a) x
1
(nT) and (b) x
2
(nT) are interpolated (—) using the Nyquist interpolation formula.
Solution:
X
sa
(jω) =
_
∞
t=−∞
∞
n=−∞
x(nT)δ(t −nT)e
−jωt
=
∞
n=−∞
x(nT)
_
∞
t=−∞
δ(t −nT)e
−jωt
(11)
=
∞
n=−∞
x(nT)e
−jωnT
The discretetime Fourier transform of x
sa
(t), I denote DTFT¦x
sa
(t)¦, and it is:
DTFT¦x
sa
(t)¦ = X
d
(e
jΩ
) =
∞
n=−∞
x(nT)e
−jΩn
Essentially, the diﬀerence between the DTFT and the Fourier transform of the sampled signal is the relationship
Ω = ωT = 2πfT
ECE 5520 Fall 2009 27
But this deﬁnes the relationship only between the Fourier transform of the sampled signal. How can we
relate this to the FT of the continuoustime signal? A: Using (10). We have that X
d
(e
jω
) = X
sa
_
Ω
2πT
_
. Then
plugging into (10),
Solution:
X
d
(e
jΩ
) = X
sa
_
j
Ω
T
_
=
2π
T
∞
n=−∞
X
_
j
_
Ω
T
−
2πn
T
__
=
2π
T
∞
n=−∞
X
_
j
_
Ω −2πn
T
__
(12)
If x(t) is suﬃciently bandlimited, X
d
(e
jΩ
) ∝ X(jΩ/T) in the interval −π < Ω < π. This relationship holds
for the Fourier transform of the original, continuoustime signal between −π < Ω < π if and only if the
original signal x(t) is suﬃciently bandlimited so that the Nyquist sampling theorem applies.
Notes:
• Most things in the DTFT table in Table 2.8 (page 68) are analogous to continuous functions which are
not bandlimited. So  don’t expect the last form to apply to any continuous function.
• The DTFT is periodic in Ω with period 2π.
• Don’t be confused: The DTFT is a continuous function of Ω.
Lecture 4
Today: (1) Example: Bandpass Sampling (2) Orthogonality / Signal Spaces I
7.3 Bandpass sampling
Bandpass sampling is the use of sampling to perform downconversion via frequency aliasing. We assume that
the signal is truly a bandpass signal (has no DC component). Then, we sample with rate 1/T and we rely on
aliasing to produce an image the spectrum at a relatively low frequency (less than onehalf the sampling rate).
Example: Bandpass Sampling
This is Example 2.6.2 (pp. 6065) in the Rice book. In short, we have a bandpass signal in the frequency band,
450 Hz to 460 Hz. The center frequency is 455 Hz. We want to sample the signal. We have two choices:
1. Sample at more than twice the maximum frequency.
2. Sample at more than twice the bandwidth.
The ﬁrst gives a sampling frequency of more than 920 Hz. The latter gives a sampling frequency of more than
20 Hz. There is clearly a beneﬁt to the second part. However, traditionally, we would need to downconvert the
signal to baseband in order to sample it at the lower rate. (See Lecture 3 notes).
ECE 5520 Fall 2009 28
What is the sampled spectrum, and what is the bandwidth of the sampled signal in radians/sample if the
sampling rate is:
1. 1000 samples/sec?
2. 140 samples/sec?
Solution:
1. Sample at 1000 samples/sec: Ω
c
= 2π
455
500
=
455
500
π. The signal is very sparse  it occupies only
2π
10 Hz
1000 samples/sec
= π/50 radians/sample.
2. Sample at 140 samples/sec: Copies of the entire frequency spectrum will appear in the frequency
domain at multiples of 140 Hz or 2π140 radians/sec. The discretetime frequency is modulo 2π, so the
center frequency of the sampled signal is:
Ω
c
= 2π
455
140
mod 2π =
13π
2
mod 2π =
π
2
The value π/2 radians/sample is equivalent to 35 cycles/second. The bandwidth is 2π
10
140
= π/7 radi
ans/sample of sampled spectrum.
8 Orthogonality
From the ﬁrst lecture, we talked about how digital communications is largely about using (and choosing some
combination of) a discrete set of waveforms to convey information on the (continuoustime, realvalued) channel.
At the receiver, the idea is to determine which waveforms were sent and in what quantities.
This choice of waveforms is partially determined by bandwidth, which we have covered. It is also determined
by orthogonality. Loosely, orthogonality of waveforms means that some combination of the waveforms can be
uniquely separated at the receiver. This is the critical concept needed to understand digital receivers. To build
a better framework for it, we’ll start with something you’re probably more familiar with: vector multiplication.
8.1 Inner Product of Vectors
We often use an idea called inner product, which you’re familiar with from vectors (as the dot product). If we
have two vectors x and y,
x =
_
¸
¸
¸
_
x
1
x
2
.
.
.
x
n
_
¸
¸
¸
_
y =
_
¸
¸
¸
_
y
1
y
2
.
.
.
y
n
_
¸
¸
¸
_
Then we can take their inner product as:
x
T
y =
i
x
i
y
i
Note the
T
transpose ‘inbetween’ an inner product. The inner product is also denoted ¸x, y). Finally, the norm
of a vector is the square root of the inner product of a vector with itself,
x =
_
¸x, x)
ECE 5520 Fall 2009 29
A norm is always indicated with the   symbols outside of the vector.
Example: Example Inner Products
Let x = [1, 1, 1]
T
, y = [0, 1, 1]
T
, z = [1, 0, 0]
T
. What are:
• ¸x, y)?
• ¸z, y)?
• x?
• z?
Recall the inner product between two vectors can also be written as
x
T
y = xy cos θ
where θ is the angle between vectors x and y. In words, the inner product is a measure of ‘samedirectionness’:
when positive, the two vectors are pointing in a similar direction, when negative, they are pointing in somewhat
opposite directions; when zero, they’re at crossdirections.
8.2 Inner Product of Functions
The inner product is not limited to vectors. It also is applied to the ‘space’ of squareintegrable real functions,
i.e., those functions x(t) for which
_
∞
−∞
x
2
(t)dt < ∞.
The ‘space’ of squareintegrable complex functions, i.e., those functions x(t) for which
_
∞
−∞
[x(t)[
2
dt < ∞
also have an inner product. These norms are:
¸x(t), y(t)) =
_
∞
−∞
x(t)y(t)dt Real functions
¸x(t), y(t)) =
_
∞
−∞
x(t)y
∗
(t)dt Complex functions
As a result,
x(t) =
¸
_
∞
−∞
x
2
(t)dt Real functions
x(t) =
¸
_
∞
−∞
[x(t)[
2
dt Complex functions
Example: Sine and Cosine
Let
x(t) =
_
cos(2πt), 0 < t ≤ 1
0, o.w.
y(t) =
_
sin(2πt), 0 < t ≤ 1
0, o.w.
ECE 5520 Fall 2009 30
What are x(t) and y(t)? What is ¸x(t), y(t))?
Solution: Using cos
2
x =
1
2
(1 + cos 2x),
x(t) =
¸
_
1
0
cos
2
(2πt)dt =
¸
1
2
_
1 +
1
2π
sin(2πt)
_
dt
¸
¸
¸
¸
1
0
=
_
1/2
The solution for y(t) also turns out to be
_
1/2. Using sin 2x = 2 cos xsin x, the inner product is,
¸x(t), y(t)) =
_
1
0
cos(2πt) sin(2πt)dt
=
_
1
0
1
2
sin(4πt)dt
=
−1
8π
cos(4πt)
¸
¸
¸
¸
1
0
=
−1
8π
(1 −1) = 0
8.3 Deﬁnitions
Def ’n: Orthogonal
Two signals x(t) and y(t) are orthogonal if
¸x(t), y(t)) = 0
Def ’n: Orthonormal
Two signals x(t) and y(t) are orthonormal if they are orthogonal and they both have norm 1, i.e.,
x(t) = 1 y(t) = 1
(a) Are x(t) and y(t) from the sine/cosine example above orthogonal? (b) Are they orthonormal?
Solution: (a) Yes. (b) No. They could be orthonormal if they’d been scaled by
√
2.
x t
1
( )
A
A
x t
2
( )
A
A
Figure 12: Two WalshHadamard functions.
Example: WalshHadamard 2 Functions
Let
x
1
(t) =
_
_
_
A, 0 < t ≤ 0.5
−A, 0.5 < t ≤ 1
0, o.w.
x
2
(t) =
_
A, 0 < t ≤ 1
0, o.w.
ECE 5520 Fall 2009 31
These are shown in Fig. 12. (a) Are x
1
(t) and x
2
(t) orthogonal? (b) Are they orthonormal?
Solution: (a) Yes. (b) Only if A = 1.
8.4 Orthogonal Sets
Def ’n: Orthogonal Set
M signals x
1
(t), . . . , x
M
(t) are mutually orthogonal, or form an orthogonal set, if ¸x
i
(t), x
j
(t)) = 0 for all i ,= j.
Def ’n: Orthonormal Set
M mutually orthogonal signals x
1
(t), . . . , x
M
(t) are form an orthonormal set, if x
i
 = 1 for all i.
(a) (b)
Figure 13: WalshHadamard 2length and 4length functions in image form. Each function is a row of the image,
where crimson red is 1 and black is 1.
Figure 14: WalshHadamard 64length functions in image form.
Do the 64 WH functions form an orthonormal set? Why or why not?
Solution: Yes. Every pair i, j with i ,= j agrees half of the time, and disagrees half of the time. So the function
is halftime 1 and halftime 1 so they cancel and have inner product of zero. The norm of the function in row
i is 1 since the amplitude is always +1 or −1.
Example: CDMA and WalshHadamard
This example is just for your information (not for any test). The set of 64length WalshHadamard sequences
ECE 5520 Fall 2009 32
is used in CDMA (IS95) in two ways:
1. Reverse Link: Modulation: For each group of 6 bits to be send, the modulator chooses one of the 64
possible 64length functions.
2. Forward Link: Channelization or Multiple Access: The base station assigns each user a single WH function
(channel). It uses 64 functions (channels). The data on each channel is multiplied by a Walsh function so
that it is orthogonal to the data received by all other users.
Lecture 5
Today: (1) Orthogonal Signal Representations
9 Orthonormal Signal Representations
Last time we deﬁned the inner product for vectors and for functions. We talked about orthogonality and
orthonormality of functions and sets of functions. Now, we consider how to take a set of arbitrary signals and
represent them as vectors in an orthonormal basis.
What are some common orthonormal signal representations?
• Nyquist sampling: sinc functions centered at sampling times.
• Fourier series: complex sinusoids at diﬀerent frequencies
• Sine and cosine at the same frequency
• Wavelets
And, we will come up with others. Each one has a limitation – only a certain set of functions can be exactly
represented in a particular signal representation. Essentially, we must limit the set of possible functions to a
set. That is, some subset of all possible functions.
We refer to the set of arbitrary signals as:
o = ¦s
0
(t), s
1
(t), . . . , s
M−1
(t)¦
For example, a transmitter may be allowed to send any one of these M signals in order to convey information
to a receiver.
We’re going to introduce another set called an orthonormal basis:
B = ¦φ
0
(t), φ
1
(t), . . . , φ
N−1
(t)¦
Each function in B is called a basis function. You can think of B as an unambiguous and useful language to
represent the signals from our set o. Or, in analogy to color printers, a color printer can produce any of millions
of possible colors (signal set), only using black, red, blue, and green (basis set) (or black, cyan, magenta, yellow).
ECE 5520 Fall 2009 33
9.1 Orthonormal Bases
Def ’n: Span
The span of the set B is the set of all functions which are linear combinations of the functions in B. The span
is referred to as Span¦B¦ and is
Span¦B¦ =
_
N−1
k=0
a
k
φ
k
(t)
_
a
0
,...,a
N−1
∈R
Def ’n: Orthogonal Basis
For any arbitrary set of signals o = ¦s
i
(t)¦
M−1
i=0
, the orthogonal basis is an orthogonal set of the smallest size N,
B = ¦φ
i
(t)¦
N−1
i=0
, for which s
i
(t) ∈ Span¦B¦ for every i = 0, . . . , M − 1. The value of N is called the dimension
of the signal set.
Def ’n: Orthonormal Basis
The orthonormal basis is an orthogonal basis in which each basis function has norm 1.
Notes:
• The orthonormal basis is not unique. If you can ﬁnd one basis, you can use, for example, ¦−φ
k
(t)¦
N−1
k=0
as
another orthonormal basis.
• All orthogonal bases will have size N: no smaller basis can include all signals s
i
(t) in its span, and a larger
set would not be a basis by deﬁnition.
The signal set might be a set of signals we can use to transmit particular bits (or sets of bits). (As the
WalshHadamard functions are used in IS95 reverse link). An orthonormal basis for an arbitrary signal set
tells us:
• how to build the receiver,
• how to represent the signals in ‘signalspace’, and
• how to quickly analyze the error rate of the scheme.
9.2 Synthesis
Consider one of the signals in our signal set, s
i
(t). Given that it is in the span of our basis B, it can be
represented as a linear combination of the basis functions,
s
i
(t) =
N−1
k=0
a
i,k
φ
k
(t)
The particular constants a
i,k
are calculated using the inner product:
a
i,k
= ¸s
i
(t), φ
k
(t))
The projection of one signal i onto basis function k is deﬁned as a
i,k
φ
k
(t) = ¸s
i
(t), φ
k
(t))φ
k
(t). Then the signal
is equal to the sum of its projections onto the basis functions.
Why is this?
ECE 5520 Fall 2009 34
Solution: Since s
i
(t) ∈ Span¦B¦, we know there are some constants ¦a
i,k
¦
N−1
k
such that
s
i
(t) =
N−1
k=0
a
i,k
φ
k
(t)
Taking the inner product of both sides with φ
j
(t),
¸s
i
(t), φ
j
(t)) =
_
N−1
k=0
a
i,k
φ
k
(t), φ
j
(t)
_
¸s
i
(t), φ
j
(t)) =
N−1
k=0
a
i,k
¸φ
k
(t), φ
j
(t))
¸s
i
(t), φ
j
(t)) = a
i,j
So, now we can now represent a signal by a vector,
s
i
= [a
i,0
, a
i,1
, . . . , a
i,N−1
]
T
This and the basis functions completely represent each signal. Plotting ¦s
i
¦
i
in an N dimensional grid is
termed a constellation diagram. Generally, this space that we’re plotting in is called signal space.
We can also synthesize any of our M signals in the signal set by adding the proper linear combination of
the N bases. By choosing one of the M signals, we convey information, speciﬁcally, log
2
M bits of information.
(Generally, we choose M to be a power of 2 for simplicity).
See Figure 5.5 in Rice (page 254), which shows a block diagram of how a transmitter would synthesize one
of the M signals to send, based on an input bitstream.
Example: Positionshifted pulses
Plot the signal space diagram for the signals,
s
0
(t) = u(t) −u(t −1)
s
1
(t) = u(t −1) −u(t −2)
s
2
(t) = u(t) −u(t −2)
given the orthonormal basis,
φ
0
(t) = u(t) −u(t −1)
φ
1
(t) = u(t −1) −u(t −2)
What are the signal space vectors, s
0
, s
1
, and s
2
?
Solution: They are s
0
= [1, 0]
T
, s
1
= [0, 1]
T
, and s
2
= [1, 1]
T
. They are plotted in the signal space diagram in
Figure 15.
Energy:
• Energy can be calculated in signal space as
Energy¦s
i
(t)¦ =
_
−∞
∞s
2
i
(t)dt =
N−1
k=0
a
2
i,k
Proof?
ECE 5520 Fall 2009 35
1
1
f
1
f
2
Figure 15: Signal space diagram for positionshifted signals example.
• We will ﬁnd out later that it is the distances between the points in signal space which determine the bit
error rate performance of the receiver.
d
i,j
=
¸
¸
¸
_
N−1
k=0
(a
i,k
−a
j,k
)
2
for i, j in ¦0, . . . , M −1¦.
• Although diﬀerent ON bases can be used, the energy and distance between points will not change.
Example: Amplitudeshifted signals
Now consider
s
0
(t) = 1.5[u(t) −u(t −1)]
s
1
(t) = 0.5[u(t) −u(t −1)]
s
2
(t) = −0.5[u(t) −u(t −1)]
s
3
(t) = −1.5[u(t) −u(t −1)]
and the orthonormal basis,
φ
1
(t) = u(t) −u(t −1)
What are the signal space vectors for the signals ¦s
i
(t)¦? What are their energies?
Solution: s
0
= [1.5], s
1
= [0.5], s
2
= [−0.5], s
3
= [−1.5]. See Figure 16.
1
f
1
1 1 0
Figure 16: Signal space diagram for amplitudeshifted signals example.
Energies are just the squared magnitude of the vector: 2.25, 0.25, 0.25, and 2.25, respectively.
9.3 Analysis
At a receiver, our job will be to analyze the received signal (a function) and to decide which of the M possible
signals was sent. This is the task of analysis. It turns out that an orthonormal bases makes our analysis very
straightforward and elegant.
We won’t receive exactly what we sent  there will be additional functions added to the signal function we
sent.
ECE 5520 Fall 2009 36
• Thermal noise
• Interference from other users
• Selfinterference
We won’t get into these problems today. But, we might say that if we send signal m, i.e., s
m
(t) from our signal
set, then we would receive
r(t) = s
m
(t) +w(t)
where the w(t) is the sum of all of the additive signals that we did not intend to receive. But w(t) might
not (probably not) be in the span of our basis B, so r(t) would not be in Span¦B¦ either. What is the best
approximation to r(t) in the signal space? Speciﬁcally, what is ˆ r(t) ∈ Span¦B¦ such that the energy of the
diﬀerence between ˆ r(t) and r(t) is minimized, i.e.,
argmin
ˆ r(t)∈Span{B}
_
∞
−∞
[ˆ r(t) −r(t)[
2
dt (13)
Solution: Since ˆ r(t) ∈ Span¦B¦, it can be represented as a vector in signal space,
x = [x
0
, x
1
, . . . , x
N−1
]
T
.
and the synthesis equation is
ˆ r(t) =
N−1
k=0
x
k
φ
k
(t)
If you plug in the above expression for ˆ r(t) into (13), and then ﬁnd the minimum with respect to each x
k
, you’d
see that the minimum error is at
x
k
=
_
∞
−∞
r(t)φ
k
(t)dt
that is, x
k
= ¸r(t), φ
k
(t)), for k = 0, . . . , N.
Example: Analysis using a WalshHadamard 2 Basis
See Figure 17. Let s
0
(t) = φ
0
(t) and s
1
(t) = φ
1
(t). What is ˆ r(t)?
Solution:
ˆ r(t) =
_
_
_
1, 0 ≤ t < 1
−1/2, 1 ≤ t < 2
0, o.w.
Lecture 6
Today: (1) Multivariate Distributions
ECE 5520 Fall 2009 37
r t ( )
2
2
1 2
r( ) t
2
2
1 2
f
1
( ) t
1 2
f
2
( ) t
1/2
1/2
1 2
1/2
1/2
1
Figure 17: Signal and basis functions for Analysis example.
10 MultiVariate Distributions
For two random variables X
1
and X
2
,
• Joint CDF: F
X
1
,X
2
(x
1
, x
2
) = P [¦X
1
≤ x
1
¦ ∩ ¦X
2
≤ x
2
¦] It is the probability that both events happen
simultaneously.
• Joint pmf: P
X
1
,X
2
(x
1
, x
2
) = P [¦X
1
= x
1
¦ ∩ ¦X
2
= x
2
¦] It is the probability that both events happen
simultaneously.
• Joint pdf: f
X
1
,X
2
(x
1
, x
2
) =
∂
2
∂x
1
∂x
2
F
X
1
,X
2
(x
1
, x
2
)
The pdf and pmf integrate / sum to one, and are nonnegative. The CDF is nonnegative and nondecreasing,
with lim
x
i
→.−∞
F
X
1
,X
2
(x
1
, x
2
) = 0 and lim
x
1
,x
2
→.+∞
F
X
1
,X
2
(x
1
, x
2
) = 1.
Note: Two errors in Rice book dealing with the deﬁnition of the CDF. Eqn 4.9 should be:
F
X
(x) = P [X ≤ x] =
_
x
−∞
f
X
(t)dt
and Eqn 4.15 should be:
F
X
(x) = P [X ≤ x]
To ﬁnd the probability of an event, you integrate. For example, for event B ∈ S,
• Discrete case: P [B] =
(X
1
,X
2
)∈B
P
X
1
,X
2
(x
1
, x
2
)
• Continuous Case: P [B] =
_ _
(x
1
,x
2
)∈B
f
X
1
,X
2
(x
1
, x
2
)dx
1
dx
2
ECE 5520 Fall 2009 38
The marginal distributions are:
• Marginal pmf: P
X
2
(x
2
) =
x
1
∈S
X
1
P
X
1
,X
2
(x
1
, x
2
)
• Marginal pdf: f
X
2
(x
2
) =
_
x
1
∈S
X
1
f
X
1
,X
2
(x
1
, x
2
)dx
1
Two random variables X
1
and X
2
are independent iﬀ for all x
1
and x
2
,
• P
X
1
,X
2
(x
1
, x
2
) = P
X
1
(x
1
)P
X
2
(x
2
)
• f
X
1
,X
2
(x
1
, x
2
) = f
X
2
(x
2
)f
X
1
(x
1
)
10.1 Random Vectors
Def ’n: Random Vector
A random vector (R.V.) is a list of multiple random variables X
1
, X
2
, . . . , X
n
,
X = [X
1
, X
2
, . . . , X
n
]
T
Here are the Models of R.V.s:
1. The CDF of R.V. X is F
X
(x) = F
X
1
,...,Xn
(x
1
, . . . , x
n
) = P [X
1
≤ x
1
, . . . , X
n
≤ x
n
].
2. The pmf of a discrete R.V. X is P
X
(x) = P
X
1
,...,Xn
(x
1
, . . . , x
n
) = P [X
1
= x
1
, . . . , X
n
= x
n
].
3. The pdf of a continuous R.V. X is f
X
(x) = f
X
1
,...,Xn
(x
1
, . . . , x
n
) =
∂
n
∂x
1
···∂xn
F
X
(x).
10.2 Conditional Distributions
Given event B ∈ S which has P [B] > 0, the joint probability conditioned on event B is
• Discrete case:
P
X
1
,X
2
B
(x
1
, x
2
) =
_
P
X
1
,X
2
(x
1
,x
2
)
P[B]
, (X
1
, X
2
) ∈ B
0, o.w.
• Continuous Case:
f
X
1
,X
2
B
(x
1
, x
2
) =
_
f
X
1
,X
2
(x
1
,x
2
)
P[B]
, (X
1
, X
2
) ∈ B
0, o.w.
Given r.v.s X
1
and X
2
,
• Discrete case. The conditional pmf of X
1
given X
2
= x
2
, where P
X
2
(x
2
) > 0, is
P
X
1
X
2
(x
1
[x
2
) = P
X
1
,X
2
(x
1
, x
2
)/P
X
2
(x
2
)
• Continuous Case: The conditional pdf of X
1
given X
2
= x
2
, where f
X
2
(x
2
) > 0, is
f
X
1
X
2
(x
1
[x
2
) = f
X
1
,X
2
(x
1
, x
2
)/f
X
2
(x
2
)
ECE 5520 Fall 2009 39
Def ’n: Bayes’ Law
Bayes’ Law is a reformulation of he deﬁnition of the marginal pdf. It is written either as:
f
X
1
,X
2
(x
1
, x
2
) = f
X
2
X
1
(x
2
[x
1
)f
X
1
(x
1
)
or
f
X
1
X
2
(x
1
[x
2
) =
f
X
2
X
1
(x
2
[x
1
)f
X
1
(x
1
)
f
X
2
(x
2
)
10.3 Simulation of Digital Communication Systems
A simulation of a digital communication system is often used to estimate a bit error rate. Each bit can either
be demodulated without error, or with error. Thus the simulation of one bit is a Bernoulli trial. This trial E
i
is in error (E
1
= 1) with probability p
e
(the true bit error rate) and correct (E
i
= 0) with probability 1 − p
e
.
What type of random variable is E
i
?
Simulations run many bits, say N bits through a model of the communication system, and count the number
of bits that are in error. Let S =
N
i=1
E
i
, and assume that ¦E
i
¦ are independent and identically distributed
(i.i.d.).
1. What type of random variable is S?
2. What is the pmf of S?
3. What is the mean and variance of S?
Solution: E
i
is called a Bernoulli r.v. and S is called a Binomial r.v., with pmf
P
S
(s) =
_
N
s
_
p
s
e
(1 −p
e
)
N−s
The mean of S is the the same as the mean of the sum of ¦E
i
¦,
E
S
[S] = E
{E
i
}
_
N
i=1
E
i
_
=
N
i=1
E
E
i
[E
i
]
=
N
i=1
[(1 −p) 0 +p 1] = Np
We can ﬁnd the variance of S the same way:
Var
S
[S] = Var
{E
i
}
_
N
i=1
E
i
_
=
N
i=1
Var
E
i
[E
i
]
=
N
i=1
_
(1 −p) (0 −p)
2
+p (1 −p)
2
¸
=
N
i=1
_
(1 −p)p
2
+ (1 −p)(p −p
2
)
¸
= Np(1 −p)
ECE 5520 Fall 2009 40
We may also be interested knowing how many bits to run in order to get an estimate of the bit error rate.
For example, if we run a simulation and get zero bit errors, we won’t have a very good idea of the bit error rate.
Let T
1
be the time (number of bits) up to and including the ﬁrst error.
1. What type of random variable is T
1
?
2. What is the pmf of T
1
?
3. What is the mean of T
1
?
Solution: T
1
is a Geometric r.v. with pmf
P
T
1
(t) = (1 −p
e
)
t−1
p
e
The mean of T
1
is
E[T
1
] =
1
p
e
Note the variance of T
1
is Var [T
1
] = (1 − p
e
)/p
2
e
, so the standard deviation for very low p
e
is almost the same
as the expected value.
So, even if we run an experiment until the ﬁrst bit error, our estimate of p
e
will have relatively high variance.
10.4 Mixed Discrete and Continuous Joint Variables
This was not covered in ECE 5510, although it was in the Yates & Goodman textbook. We’ll often have X
1
discrete and X
2
continuous.
Example: Digital communication system in noise
Let X
1
is the transmitted signal voltage, and X
2
is the received signal voltage, which is modeled as
X
2
= aX
1
+N
where N is a continuousvalued random variable representing the additive noise of the channel, and a is a
constant which represents the attenuation of the channel. In a digital system, X
1
may take a discrete set of
values, e.g., ¦1.5, 0.5, −0.5, −1.5¦. But the noise N may be continuous, e.g., Gaussian with mean µ and variance
σ
2
. As long as σ
2
> 0, then X
2
will be continuousvalued.
Workaround: We will sometimes use a pdf for a discrete random variable. For instance, if
P
X
1
(x
1
) =
_
_
_
0.6, x
1
= 0
0.4, x
1
= 1
0, o.w.
then we would write a pdf with the probabilities as amplitudes of a dirac delta function centered at the value
that has that probability,
f
X
1
(x
1
) = 0.6δ(x
1
) + 0.4δ(x
1
−1)
This pdf has integral 1, and nonnegative values for all x
1
. Any probability integral will return the proper value.
For example, the probability that −0.5 < x
1
< 0.5 would be an integral that would return 0.6.
ECE 5520 Fall 2009 41
Example: Joint distribution of X
1
, X
2
Consider the channel model X
2
= X
1
+N, and
f
N
(n) =
1
√
2πσ
2
e
−n
2
/(2σ
2
)
where σ
2
is some known variance, and the r.v. X
1
is independent of N with
P
X
1
(x
1
) =
_
_
_
0.5, x
1
= 0
0.5, x
1
= 1
0, o.w.
.
1. What is the pdf of X
2
?
2. What is the joint p.d.f. of (X
1
, X
2
) ?
Solution:
1. When two independent r.v.s are added, the pdf of the sum is the convolution of the pdfs of the inputs.
Writing
f
X
1
(x
1
) = 0.5δ(x
1
) + 0.5δ(x
1
−1)
we convolve this with f
N
(n) above, to get
f
X
2
(x
2
) = 0.5f
N
(x
2
) + 0.5f
N
(x
2
−1)
f
X
2
(x
2
) =
1
2
√
2πσ
2
_
e
−x
2
2
/(2σ
2
)
+e
−(x
2
−1)
2
/(2σ
2
)
_
2. What is the joint p.d.f. of (X
1
, X
2
) ? Since X
1
and X
2
are NOT independent, we cannot simply multiply
the marginal pdfs together. It is necessary to use Bayes’ Law.
f
X
1
,X
2
(x
1
, x
2
) = f
X
2
X
1
(x
2
[x
1
)f
X
1
(x
1
)
Given a value of X
1
(either 0 or 1) we can write down the pdf of X
2
. So break this into two cases:
f
X
1
,X
2
(x
1
, x
2
) =
_
_
_
f
X
2
X
1
(x
2
[0)f
X
1
(0), x
1
= 0
f
X
2
X
1
(x
2
[1)f
X
1
(1), x
1
= 1
0, o.w.
f
X
1
,X
2
(x
1
, x
2
) =
_
_
_
0.5f
N
(x
2
), x
1
= 0
0.5f
N
(x
2
−1), x
1
= 1
0, o.w.
f
X
1
,X
2
(x
1
, x
2
) = 0.5f
N
(x
2
)δ(x
1
) +
0.5f
N
(x
2
−1)δ(x
1
−1)
These last two lines are completely equivalent. Use whichever seems more convenient for you. See Figure
18.
ECE 5520 Fall 2009 42
−1
0
1
2
−4
−2
0
2
4
0
X
2
X
1
Figure 18: Joint pdf of X
1
and X
2
, the input and output (respectively) of the example additive noise commu
nication system, when σ = 1.
10.5 Expectation
Def ’n: Expected Value (Joint)
The expected value of a function g(X
1
, X
2
) of random variables X
1
and X
2
is given by,
1. Discrete: E[g(X
1
, X
2
)] =
X
1
∈S
X
1
X
2
∈S
X
2
g(X
1
, X
2
)P
X
1
,X
2
(x
1
, x
2
)
2. Continuous: E[g(X
1
, X
2
)] =
_
X
1
∈S
X
1
_
X
2
∈S
X
2
g(X
1
, X
2
)f
X
1
,X
2
(x
1
, x
2
)
Typical functions g(X
1
, X
2
) are:
• Mean of X
1
or X
2
: g(X
1
, X
2
) = X
1
or g(X
1
, X
2
) = X
2
will result in the means µ
X
1
and µ
X
2
.
• Variance (or 2nd central moment) of X
1
or X
2
: g(X
1
, X
2
) = (X
1
− µ
X
1
)
2
or g(X
1
, X
2
) = (X
2
− µ
X
2
)
2
.
Often denoted σ
2
X
1
and σ
2
X
2
.
• Covariance of X
1
and X
2
: g(X
1
, X
2
) = (X
1
−µ
X
1
)(X
2
−µ
X
2
).
• Expected value of the product of X
1
and X
2
, also called the ‘correlation’ of X
1
and X
2
: g(X
1
, X
2
) = X
1
X
2
.
10.6 Gaussian Random Variables
For a single Gaussian r.v. X with mean µ
X
and variance σ
2
X
, we have the pdf,
f
X
(x) =
1
_
2πσ
2
X
e
−
(x−µ
X
)
2
2σ
2
Consider Y to be Gaussian with mean 0 and variance 1. Then, the CDF of Y is denoted as F
Y
(y) = P [Y ≤ y] =
Φ(y). So, for X, which has nonzero mean and nonunitvariance, we can write its CDF as
F
X
(x) = P [X ≤ x] = Φ
_
x −µ
X
σ
X
_
ECE 5520 Fall 2009 43
You can prove this by showing that the event X ≤ x is the same as the event
X −µ
X
σ
X
≤
x −µ
X
σ
X
Since the lefthand side is a unitvariance, zero mean Gaussian random variable, we can write the probability
of this event using the unitvariance, zero mean Gaussian CDF.
10.6.1 Complementary CDF
The probability that a unitvariance, zero mean Gaussian r.v. X exceeds some value x is one minus the CDF,
that is, 1 −Φ(x). This is so common in digital communications, it is given its own name, Q(x),
Q(x) = P [X > x] = 1 −Φ(x)
What is Q(x) in integral form?
Q(x) =
_
∞
x
1
√
2π
e
−w
2
/2
dw
For an Gaussian r.v. X with variance σ
2
X
,
P [X > x] = Q
_
x −µ
X
σ
X
_
= 1 −Φ
_
x −µ
X
σ
X
_
10.6.2 Error Function
In math, in some texts, and in Matlab, the Q(x) function is not used. Instead, there is a function called erf(x)
erf(x)
2
√
π
_
x
0
e
−t
2
dt
Example: Relationship between Q() and erf()
What is the functional relationship between Q() and erf()?
Solution: Substituting t = u/
√
2 (and thus dt = du/
√
2),
erf(x)
2
√
2π
_
√
2x
0
e
−u
2
/2
du
= 2
_
√
2x
0
1
√
2π
e
−u
2
/2
du
= 2
_
Φ(
√
2x) −
1
2
_
Equivalently, we can write Φ() in terms of the erf() function,
Φ(
√
2x) =
1
2
erf(x) +
1
2
Finally let y =
√
2x, so that
Φ(y) =
1
2
erf
_
y
√
2
_
+
1
2
ECE 5520 Fall 2009 44
Or in terms of Q(),
Q(y) = 1 −Φ(y) =
1
2
−
1
2
erf
_
y
√
2
_
(14)
You should go to Matlab and create a function Q(y) which implements:
function rval = Q(y)
rval = 1/2  1/2 .* erf(y./sqrt(2));
10.7 Examples
Example: Probability of Error in Binary Example
As in the previous example, we have a model system in which the receiver sees X
2
= X
1
+N. Here, X
1
∈ ¦0, 1¦
with equal probabilities and N is independent of X
1
and zeromean Gaussian with variance 1/4. The receiver
decides as follows:
• If X
2
≤ 1/3, then decide that the transmitter sent a ‘0’.
• If X
2
> 1/3, then decide that the transmitter sent a ‘1’.
1. Given that X
1
= 1, what is the probability that the receiver decides that a ‘0’ was sent?
2. Given that X
1
= 0, what is the probability that the receiver decides that a ‘1’ was sent?
Solution:
1. Given that X
1
= 1, since X
2
= X
1
+ N, it is clear that X
2
is also a Gaussian r.v. with mean 1 and
variance 1/4. Then the probability that the receiver decides ‘0’ is the probability that X
2
≤ 1/3,
P [error[X
1
= 1] = P [X
2
≤ 1/3]
= P
_
X
2
−1
_
1/4
≤
1/3 −1
_
1/4
_
= 1 −Q((−2/3)/(1/2))
= 1 −Q(−4/3)
2. Given that X
1
= 0, the probability that the receiver decides ‘1’ is the probability that X
2
> 1/3,
P [error[X
1
= 0] = P [X
2
> 1/3]
= P
_
X
2
_
1/4
>
1/3
_
1/4
_
= Q((1/3)/(1/2))
= Q(2/3)
ECE 5520 Fall 2009 45
10.8 Gaussian Random Vectors
Def ’n: Multivariate Gaussian R.V.
An nlength R.V. X is multivariate Gaussian with mean µ
X
, and covariance matrix C
X
if it has the pdf,
f
X
(x) =
1
_
(2π)
n
det(C
X
)
exp
_
−
1
2
(x −µ
X
)
T
C
−1
X
(x −µ
X
)
_
where det() is the determinant of the covariance matrix, and C
−1
X
is the inverse of the covariance matrix.
Any linear combination of jointly Gaussian random variables is another jointly Gaussian ran
dom variable. For example, if we have a matrix A and we let a new random vector Y = AX, then Y is also
a Gaussian random vector with mean Aµ
X
and covariance matrix AC
X
A
T
.
If the elements of X were independent random variables, the pdf would be the product of the individual
pdfs (as with any random vector) and in this case the pdf would be:
f
X
(x) =
1
_
(2π)
n
i
σ
2
i
exp
_
−
n
i=1
(x
i
−µ
X
i
)
2
2σ
2
X
i
_
Section 4.3 spends some time with 2D Gaussian random vectors, which is the dimension with which we
spend most of our time in this class.
10.8.1 Envelope
The envelope of a 2D random vector is its distance from the origin (0, 0). Speciﬁcally, if the vector is (X
1
, X
2
),
the envelope R is
R =
_
X
2
1
+X
2
2
(15)
If both X
1
and X
2
are i.i.d. Gaussian with mean 0 and variance σ
2
, the pdf of R is
f
R
(r) =
_
r
σ
2
exp
_
−
r
2
2σ
2
_
, r ≥ 0
0, o.w.
This is the Rayleigh distribution. If instead X
1
and X
2
are independent Gaussian with means µ
1
and µ
2
,
respectively, and identical variances σ
2
, the envelope R would now have a Rice (a.k.a. Rician) distribution,
f
R
(r) =
_
r
σ
2
exp
_
−
r
2
+s
2
2σ
2
_
I
0
_
rs
σ
2
_
, r ≥ 0
0, o.w.
where s
2
= µ
2
1
+µ
2
2
, and I
0
() is the zeroth order modiﬁed Bessel function of the ﬁrst kind.
Note that both the Rayleigh and Rician pdfs can be derived from the Gaussian distribution and (15) using
the transformation of random variables methods studied in ECE 5510. Both pdfs are sometimes needed to
derive probability of error formulas.
Lecture 7
Today: (1) Matlab Simulation (2) Random Processes (3) Correlation and Matched Filter Receivers
ECE 5520 Fall 2009 46
11 Random Processes
A random process X(t, s) is a function of time t and outcome (realization) s. Outcome s lies in sample space
S. Outcome or realization is included because there could be ways to record X even at the same time. For
example, multiple receivers would record diﬀerent noisy signals even of the same transmission.
A random sequence X(n, s) is a sequence of random variables indexed by time index n and realization s ∈ S.
Typically, we omit the “s” when writing the name of the random process or random sequence, and abbreviate
it to X(t) or X(n), respectively.
11.1 Autocorrelation and Power
Def ’n: Mean Function
The mean function of the random process X(t, s) is
µ
X
(t) = E[X(t, s)]
Note the mean is taken over all possible realizations s. If you record one signal over all time t, you don’t have
anything to average to get the mean function µ
X
(t).
Def ’n: Autocorrelation Function
The autocorrelation function of a random process X(t) is
R
X
(t, τ) = E[X(t)X(t −τ)]
The autocorrelation of a random sequence X(n) is
R
X
(n, k) = E[X(n)X(n −k)]
Def ’n: Widesense stationary (WSS)
A random process is widesense stationary (WSS) if
1. µ
X
= µ
X
(t) = E[X(t)] is independent of t.
2. R
X
(t, τ) depends only on the time diﬀerence τ and not on t. We then denote the autocorrelation function
as R
X
(τ).
A random process is widesense stationary (WSS) if
1. µ
X
= µ
X
(n) = E[X(n)] is independent of n.
2. R
X
(n, k) depends only on k and not on n. We then denote the autocorrelation function as R
X
(k).
The power of a signal is given by R
X
(0).
We can estimate the autocorrelation function of an ergodic, widesense stationary random process from one
realization of a powertype signal x(t),
R
X
(τ) = lim
T→∞
1
T
_
T/2
t=−T/2
x(t)x(t −τ)dt
This is not a critical topic for this class, but we now must deﬁne “ergodic”. For an ergodic random process,
its time averages are equivalent to its ensemble averages. An example that helps demonstrate the diﬀerence
between time averages and ensemble averages is the nonergodic process of opinion polling. Consider what
happens when a pollster takes either a timeaverage or a ensemble average:
ECE 5520 Fall 2009 47
• Time Average: The pollster asks the same person, every day for N days, whether or not she will vote for
person X. The time average is taken by dividing the total number of “Yes”s by N.
• Ensemble Average: The pollster asks N diﬀerent people, on the same day, whether or not they will vote
for person X. The ensemble average is taken by dividing the total number of “Yes”s by N.
Will they come up with the same answer? Perhaps; but probably not. This makes the process nonergodic.
Power Spectral Density We have seen that for a WSS random process X(t) (and for a random sequence
X(n) we compute the power spectral density as,
S
X
(f) = F¦R
X
(τ)¦
S
X
(e
jΩ
) = DTFT¦R
X
(k)¦
11.1.1 White Noise
Let X(n) be a WSS random sequence with autocorrelation function
R
X
(k) = E[X(n)X(n −k)] = σ
2
δ(k)
This says that each element of the sequence X(n) has zero covariance with every other sample of the sequence,
i.e., it is uncorrelated with X(m), m ,= n.
• Does this mean it is independent of every other sample?
• What is the PSD of the sequence?
This sequence is commonly called ‘white noise’ because it has equal parts of every frequency (analogy to
light).
For continuous time signals, white noise is a commonly used approximation for thermal noise. Let X(t) be
a WSS random process with autocorrelation function
R
X
(τ) = E[X(t)X(t −τ)] = σ
2
δ(τ)
• What is the PSD of the sequence?
• What is the power of the signal?
Realistically, thermal noise is not constant in frequency, because as frequency goes very high (e.g., 10
15
Hz), the
power spectral density goes to zero. The Rice book (4.5.2) has a good analysis of the physics of thermal noise.
12 Correlation and MatchedFilter Receivers
We’ve been talking about how our digital transmitter can transmit at each time one of a set of signals ¦s
i
(t)¦
M
i=1
.
This transmission conveys to us which of M messages we want to send, that is, log
2
M bits of information.
At the receiver, we receive signal s
i
(t) scaled and corrupted by noise:
r(t) = bs
i
(t) +n(t)
How do we decide which signal i was transmitted?
ECE 5520 Fall 2009 48
12.1 Correlation Receiver
What we’ve been talking about it the inner product. In other terms, the inner product is a correlation:
¸r(t), s
i
(t)) =
_
∞
−∞
r(t)s
i
(t)dt
But we want to build a receiver that does the minimum amount of calculation. If s
i
(t) are nonorthogonal,
then we can reduce the amount of correlation (and thus multiplication and addition) done in the receiver by
instead, correlating with the basis functions, ¦φ
k
(t)¦
N
k=1
. Correlation with the basis functions gives
x
k
= ¸r(t), φ
k
(t)) =
_
∞
−∞
r(t)φ
k
(t)dt
for k = 1 . . . N. As notation,
x = [x
1
, x
2
, . . . , x
N
]
T
Now that we’ve done these N correlations (inner products), we can compute the estimate of the received signal
as
ˆ r(t) =
K−1
k=0
x
k
φ
k
(t)
This is the ‘correlation receiver’, shown in Figure 5.1.6 in Rice (page 226).
Noisefree channel Since r(t) = bs
i
(t) +n(t), if n(t) = 0 and b = 1 then
x = a
i
where a
i
is the signal space vector for s
i
(t).
Noisy Channel Now, let b = 1 but consider n(t) to be a white Gaussian random process with zero mean and
PSD S
N
(f) = N
0
/2. Deﬁne
n
k
= ¸n(t), φ
k
) =
_
∞
−∞
n(t)φ
k
(t)dt.
What is x in terms of a
i
and the noise ¦n
k
¦? What are the mean and covariance of ¦n
k
¦?
Solution:
x
k
=
_
∞
−∞
r(t)φ
k
(t)dt
=
_
∞
−∞
[s
i
(t) +n(t)]φ
k
(t)dt
= a
i,k
+
_
∞
−∞
n(t)φ
k
(t)dt
= a
i,k
+n
k
First, n
k
is zero mean:
E[n
k
] =
_
∞
−∞
E[n(t)] φ
k
(t)dt = 0
ECE 5520 Fall 2009 49
Next we can show that n
1
, . . . , n
N
are i.i.d. by calculating the autocorrelation R
n
(m, k).
R
n
(m, k) = E[n
k
n
m
]
=
_
∞
t=−∞
_
∞
τ=−∞
E[n(t)n(τ)] φ
k
(t)φ
m
(τ)dτdt
=
_
∞
t=−∞
_
∞
τ=−∞
N
0
2
δ(t −τ)φ
k
(t)φ
m
(τ)dτdt
=
N
0
2
_
∞
t=−∞
φ
k
(t)φ
m
(t)dτdt
=
N
0
2
δ
k,m
=
_
N
0
2
, m = k
0, o.w.
Is ¦n
k
¦ WSS? Is it a Gaussian random sequence?
Since the noise components are independent, then x
k
are also independent. (Why?) What is the pdf of x
k
?
What is the pdf of x?
Solution: x
k
are independent because x
k
= a
i,k
+ n
k
and a
i,k
is a deterministic constant. Then since n
k
is
Gaussian,
f
X
k
(x
k
) =
1
_
2π(N
0
/2)
e
−
(x
k
−a
i,k
)
2
2(N
0
/2)
And, since ¦X
k
¦ are independent,
f
x
(x) =
N
k=1
f
X
k
(x
k
)
=
N
k=1
1
_
2π(N
0
/2)
e
−
(x
k
−a
i,k
)
2
2(N
0
/2)
=
1
[2π(N
0
/2)]
N/2
e
−
P
N
k=1
(x
k
−a
i,k
)
2
2(N
0
/2)
An example is in ece5510 lec07.m.
12.2 Matched Filter Receiver
Above we said that
x
k
=
_
∞
t=−∞
r(t)φ
k
(t)dt
But there really are ﬁnite limits – let’s say that the signal has a duration T, and then rewrite the integral as
x
k
=
_
T
t=0
r(t)φ
k
(t)dt
This can be written as
x
k
=
_
T
t=0
r(t)h
k
(T −t)dt
ECE 5520 Fall 2009 50
where h
k
(t) = φ
k
(T −t). (Plug in (T −t) in this formula and the Ts cancel out and only positive t is left.) This
is the output of a convolution, taken at time T,
x
k
= r(t) ⋆ h
k
(t)[
t=T
Or equivalently
x
k
= r(t) ⋆ φ
k
(T −t)[
t=T
This is Rice Section 5.1.4.
Notes:
• The x
k
can be seen as the output of a ‘matched’ ﬁlter at time T.
• This works at time T. The output for other times will be diﬀerent in the correlation and matched ﬁlter.
• These are just two diﬀerent physical implementations. We might, for example, have a physical ﬁlter with
the impulse response φ
k
(T −t) and thus it is easy to do a matched ﬁlter implementation.
• It may be easier to ‘see’ why the correlation receiver works.
Try out the Matlab code, correlation and matched filter rx.m, which is posted on WebCT.
12.3 Amplitude
Before we said b = 1. A real channel attenuates! The job of our receiver will be to amplify the signal by
approximately 1/b. This eﬀectively multiplies the noise. In general, textbooks will assume that the automatic
gain control (AGC) works properly, and then will assume the noise signal is multiplied accordingly.
Lecture 8
Today: (1) Optimal Binary Detection
12.4 Review
At a digital transmitter, we can transmit at each time one of a set of signals ¦s
i
(t)¦
M−1
i=0
. This transmission
conveys to us which of M messages we want to send, that is, log
2
M bits of information.
At the receiver, we assume we receive signal s
i
(t) corrupted by noise:
r(t) = s
i
(t) +n(t)
How do we decide which signal i ∈ ¦0, . . . , M − 1¦ was transmitted? We split the task into downconversion,
gain control, correlation or matched ﬁlter reception, and detection, as shown in Figure 19.
Down
converter
AGC Correlationor
MatchedFilter
Optimal
Detector
r t ’( )e
 2 j f t p
c
r t ’( ) r t ( ) x
b
Figure 19: A block diagram of the receiver blocks which discussed in lecture 7 & 8.
ECE 5520 Fall 2009 51
12.5 Correlation Receiver
Our signal set can be represented by the orthonormal basis functions, ¦φ
k
(t)¦
K−1
k=0
. Correlation with the basis
functions gives
x
k
= ¸r(t), φ
k
(t)) = a
i,k
+n
k
for k = 0 . . . K −1. We denote vectors:
x = [x
0
, x
1
, . . . , x
K−1
]
T
a
i
= [a
i,0
, a
i,1
, . . . , a
i,K−1
]
T
n = [n
0
, n
1
, . . . , n
K−1
]
T
Hopefully, x and a
i
should be close since i was actually sent. In an example Matlab simulation, ece5520 lec07.m,
we simulated sending a
i
= [1, 1]
T
and receiving r(t) (and thus x) in noise.
In general, the pdf of x is multivariate Gaussian with each component x
k
independent, because:
• x
k
= a
i,k
+n
k
• a
i,k
is a deterministic constant
• The ¦n
k
¦ are i.i.d. Gaussian.
The joint pdf of x is
f
X
(x) =
K−1
k=0
f
X
k
(x
k
) =
1
[2π(N
0
/2)]
N/2
e
−
P
N
k=1
(x
k
−a
i,k
)
2
2(N
0
/2)
(16)
We showed that there are two ways to implement this: the correlation receiver, and the matched ﬁlter re
ceiver. Both result in the same output x. We simulated this in the Matlab code correlation and matched filter rx.m.
13 Optimal Detection
We receive r(t) and use the matched ﬁlter or correlation receiver to calculate x. Now, how exactly do we decide
which s
i
(t) was sent?
Consider this in signal space, as shown in Figure 20.
1
f
1
1 1 0
X
Figure 20: Signal space diagram of a PAM system with an X to mark the receiver output x.
Given the signal space diagram of the transmitted signals and the received signal space vector x, what rules
should we have to decide on i? This is the topic of detection. Optimal detection uses rules designed to minimize
the probability that a symbol error will occur.
13.1 Overview
We’ll show two things:
• If each symbol i is equally probable, then the decision will be, pick the i with a
i
closest to x.
• If symbols aren’t equally probable, then we’ll need to shift the decision boundaries.
ECE 5520 Fall 2009 52
Detection theory is a major learning objective of this course. So even though it is somewhat
theoretical, it is very applicable in digital receivers. Further, detection is applicable in a wide variety of problems,
for example,
• Medical applications: Does an image show a tumor? Does a blood sample indicate a disease? Does a
breathing sound indicate an obstructed airway?
• Communications applications: Receiver design, signal detection.
• Radar applications: Obstruction detection, motion detection.
13.2 Bayesian Detection
When we say ‘optimal detection’ in the Bayesian detection framework, we mean that we want the smallest
probability of error. The probability of error is denoted
P [error]
By error, we mean that a diﬀerent signal was detected than the one that was sent. At the start of every detection
problem, we list the events that could have occurred, i.e., the symbols that could have been sent. We follow all
detection and statistics textbooks and label these classes H
i
, and write:
H
0
: r(t) = s
0
(t) +n(t)
H
1
: r(t) = s
1
(t) +n(t)
H
M−1
: r(t) = s
M−1
(t) +n(t)
This must be a complete listing of events. That is, the events H
0
∪ H
1
∪ ∪ H
M−1
= S, where the ∪ means
union, and S is the complete event space.
14 Binary Detection
Let’s just say for now that there are only two signals s
0
(t) and s
1
(t), and only one basis function φ
0
(t). Then,
instead of vectors x, a
0
and a
1
, and n, we’ll have only scalars: x, a
0
and a
1
, and n. When we talk about them
as random variables, we’ll substitute N for n and X for x. We need to decide from X whether s
0
(t) or s
1
(t)
was sent.
The event listing is,
H
0
: r(t) = s
0
(t) +n(t)
H
1
: r(t) = s
1
(t) +n(t)
Equivalently,
H
0
: X = a
0
+N
H
1
: X = a
1
+N
We use the law of total probability to say that
P [error] = P [error ∩ H
0
] +P [error ∩ H
1
] (17)
Where the cap means ‘and’. Then using Bayes’ Law,
P [error] = P [error[H
0
] P [H
0
] +P [error[H
1
] P [H
1
]
ECE 5520 Fall 2009 53
14.1 Decision Region
We’re making a decision based only on X. Over some set R
0
of values of X, we’ll decide that H
0
happened
(s
0
(t) was sent). Over a diﬀerent set R
1
of values, we’ll decide H
1
occurred (that s
1
(t) was sent). We can’t be
indecisive, so
• There is no overlap: R
0
∩ R
1
= ∅.
• There are no values of x disregarded: R
0
∪ R
1
= S.
14.2 Formula for Probability of Error
So the probability of error is
P [error] = P [X ∈ R
1
[H
0
] P [H
0
] +P [X ∈ R
0
[H
1
] P [H
1
] (18)
The probability that X is in R
1
is one minus the probability that it is in R
0
, since the two are complementary
sets.
P [error] = (1 −P [X ∈ R
0
[H
0
])P [H
0
] +P [X ∈ R
0
[H
1
] P [H
1
]
P [error] = P [H
0
] −P [X ∈ R
0
[H
0
] P [H
0
] +P [X ∈ R
0
[H
1
] P [H
1
]
Now note that probabilities that X ∈ R
0
are integrals over the event (region) R
0
.
P [error] = P [H
0
] −
_
x∈R
0
f
XH
0
(x[H
0
)P [H
0
] dx
+
_
x∈R
0
f
XH
1
(x[H
1
)P [H
1
] dx
= P [H
0
] (19)
+
_
x∈R
0
_
f
XH
1
(x[H
1
)P [H
1
] −f
XH
0
(x[H
0
)P [H
0
]
_
dx
We’ve got a lot of things in the expression in (19), but the only thing we can change is the region R
0
. Everything
else is determined by the time we get to this point. So the question is, how do you pick R
0
to minimize (19)?
14.3 Selecting R
0
to Minimize Probability of Error
We can see what the integrand looks like. Figure 21 shows the conditional probability density functions. Figure
?? shows the joint densities (the conditional pdfs multiplied by the bit probabilities P [H
0
] and P [H
1
]. Finally,
Figure ?? shows the full integrand of (19), the diﬀerence between the joint densities.
We can pick R
0
however we want  we just say what region of x, and the integral in (19) will integrate over
it. The objective is to minimize the probability of error. Which x’s should we include in the region? Should
we include x which has a positive value of the integrand? Or should we include the parts of x which have a
negative value of the integrand?
Solution: Select R
0
to be all x such that the integrand is negative.
Then R
0
is the area in which
f
XH
0
(x[H
0
)P [H
0
] > f
XH
1
(x[H
1
)P [H
1
]
If P [H
0
] = P [H
1
], then this is the region in which X is more probable given H
0
than given H
1
.
ECE 5520 Fall 2009 54
(a)
−3 −2 −1 0 1 2 3
0
0.2
0.4
0.6
0.8
1
r
P
r
o
b
a
b
i
l
i
t
y
D
e
n
s
i
t
y
f
XH
1
f
XH
2
(b)
(c)
−3 −2 −1 0 1 2 3
−0.6
−0.4
−0.2
0
0.2
0.4
r
I
n
t
e
g
r
a
n
d
f
X ∩H
2
− f
X ∩H
1
Figure 21: The (a) conditional p.d.f.s (likelihood functions) f
XH
0
(x[H
0
) and f
XH
1
(x[H
1
), (b) joint p.d.f.s
f
XH
0
(x[H
0
)P [H
0
] and f
XH
1
(x[H
1
)P [H
1
], and (c) diﬀerence between the joint p.d.f.s, f
XH
1
(x[H
1
)P [H
1
] −
f
XH
0
(x[H
0
)P [H
0
], which is the integrand in (19).
ECE 5520 Fall 2009 55
Rearranging the terms,
f
XH
1
(x[H
1
)
f
XH
0
(x[H
0
)
<
P [H
0
]
P [H
1
]
(20)
The left hand side is called the likelihood ratio. The right hand side is a threshold. Whenever x indicates that
the likelihood ratio is less than the threshold, then we’ll decide H
0
, i.e., that s
0
(t) was sent. Otherwise, we’ll
decide H
1
, i.e., that s
1
(t) was sent.
Equation (20) is a very general result, applicable no matter what conditional distributions x has.
14.4 LogLikelihood Ratio
For the Gaussian distribution, the math gets much easier if we take the log of both sides. Why can we do this?
Solution: 1. Both sides are positive, 2. The log() function is strictly increasing.
Now, the loglikelihood ratio is
log
f
XH
1
(x[H
1
)
f
XH
0
(x[H
0
)
< log
P [H
0
]
P [H
1
]
14.5 Case of a
0
= 0, a
1
= 1 in Gaussian noise
In this example, n ∼ ^(0, σ
2
N
). In addition, assume for a minute that a
0
= 0 and a
1
= 1. What is:
1. The log of a Gaussian pdf?
2. The loglikelihood ratio?
3. The decision regions for x?
Solution: What is the log of a Gaussian pdf?
log f
XH
0
(x[H
0
) = log
_
_
1
_
2πσ
2
N
e
−
x
2
2σ
2
N
_
_
= −
1
2
log(2πσ
2
N
) −
x
2
2σ
2
N
(21)
The log f
XH
1
(x[H
1
) term will be the same but with (x − 1)
2
instead of x
2
. Continuing with the loglikelihood
ratio,
log f
XH
1
(x[H
1
) −log f
XH
0
(x[H
0
) < log
P [H
0
]
P [H
1
]
x
2
2σ
2
N
−
(x −1)
2
2σ
2
N
< log
P [H
0
]
P [H
1
]
x
2
−(x −1)
2
< 2σ
2
N
log
P [H
0
]
P [H
1
]
2x −1 < 2σ
2
N
log
P [H
0
]
P [H
1
]
x <
1
2
+σ
2
N
log
P [H
0
]
P [H
1
]
ECE 5520 Fall 2009 56
In the end result, there is a simple test for x  if it is below the decision threshold, decide H
0
. If it is above
the decision threshold,
x >
1
2
+σ
2
N
log
P [H
0
]
P [H
1
]
decide H
1
. Rather than writing both inequalities each time, we use the following notation:
x
H
1
>
<
H
0
1
2
+σ
2
N
log
P [H
0
]
P [H
1
]
This completely describes the detector receiver.
For simplicity, we also write x
H
1
>
<
H
0
γ where
γ =
1
2
+σ
2
N
log
P [H
0
]
P [H
1
]
(22)
14.6 General Case for Arbitrary Signals
If, instead of a
0
= 0 and a
1
= 1, we had arbitrary values for them (the signal space representations of s
0
(t) and
s
1
(t)), we could have derived the result in the last section the same way. As long as a
0
< a
1
, we’d still have
r
H
1
>
<
H
0
γ, but now,
γ =
a
0
+a
1
2
+
σ
2
N
a
1
−a
0
log
P [H
0
]
P [H
1
]
(23)
14.7 Equiprobable Special Case
If symbols are equally likely, P [H
1
] = P [H
0
], then
P[H
1
]
P[H
0
]
= 1 and the logarithm of the fraction is zero. So then
x
H
1
>
<
H
0
a
0
+a
1
2
The decision above says that if x is closer to a
0
, decide that s
0
(t) was sent. And if x is closer to a
1
, decide that
s
1
(t) was sent. The boundary is exactly halfway in between the two signal space vectors.
This receiver is also called a maximum likelihood detector, because we only decide which likelihood function
is higher (neither is scaled by the prior probabilities P [H
0
] or P [H
1
].
14.8 Examples
Example: When H
1
becomes less likely, which direction will the optimal threshold move, towards
a
0
or towards a
1
?
Solution: Towards a
1
.
Example: Let a
0
= −1, a
1
= 1, σ
2
N
= 0.1, P [H
1
] = 0.4, and P [H
0
] = 0.6. What is the decision
threshold for x?
ECE 5520 Fall 2009 57
Solution: From (23),
γ = 0 +
0.1
2
log
0.6
0.4
= 0.05 log 1.5 ≈ 0.0203
Example: Can the decision threshold be higher than both a
0
and a
1
in this binary, onedimensional
signalling, receiver?
Given a
0
, a
1
, σ
2
N
, P [H
1
], and P [H
0
], you should be able to calculate the optimal decision threshold γ.
Example: In this example, given all of the above constants and the optimal threshold γ, calculate
the probability of error from (18).
Starting from
P [error] = P [x ∈ R
1
[H
0
] P [H
0
] +P [x ∈ R
0
[H
1
] P [H
1
]
we can use the decision regions in (22) to write
P [error] = P [x > γ[H
0
] P [H
0
] +P [x < γ[H
1
] P [H
1
]
What is the ﬁrst probability, given that r[H
0
is Gaussian with mean a
0
and variance σ
2
N
? What is the second
probability, given that x[H
1
is Gaussian with mean a
1
and variance σ
2
N
? What is then the overall probability
of error?
Solution:
P [x > γ[H
0
] = Q
_
γ −a
0
σ
N
_
P [x < γ[H
1
] = 1 −Q
_
γ −a
1
σ
N
_
= Q
_
a
1
−γ
σ
N
_
P [error] = P [H
0
] Q
_
γ −a
0
σ
N
_
+P [H
1
] Q
_
a
1
−γ
σ
N
_
14.9 Review of Binary Detection
We did three things to prove some things about the optimal detector:
• We wrote the formula for the probability of error.
• We found the decision regions which minimized the probability of error.
• We used the log operator to show that for the Gaussian error case the decision regions are separated by
a single threshold.
• We showed the formula for that threshold, both in the equiprobable symbol case, and in the general case.
Lecture 9
Today: (1) Finish Lecture 8 (2) Intro to Mary PAM
ECE 5520 Fall 2009 58
Sample exams (20078) and solutions posted on WebCT. Of course, by putting these sample exams up, you
know the actual exam won’t contain the same exact problems. There is no guarantee this exam will be the
same level of diﬃculty of either past exam.
Notes on 2008 sample exam:
• Problem 1(b) was not material covered this semester.
• Problem 3 is not on the topic of detection theory, it is on the probability topic. There is a typo: f
N
(n)
should be f
N
(t).
• Problem 6 typo: x
1
(t) =
_
t, −1 ≤ t ≤ 1
0, o.w.
and x
1
(t) =
_
1
2
(3t
2
−1), −1 ≤ t ≤ 1
0, o.w.
. That is, the “x”s
on the RHS of the given expressions should be “t”s.
Notes on 2007 sample exam:
• Problem 2 should have said “with period T
sa
” rather than “at rate T
sa
”.
• The book that year used ϕ
i
(t) instead of φ
i
(t) as the notation for a basis function, and α
i
instead of s
i
as
the signal space vector for signal s
i
(t).
• Problem 6 is on the topic of detection theory.
15 Pulse Amplitude Modulation (PAM)
Def ’n: Pulse Amplitude Modulation (PAM)
Mary Pulse Amplitude Modulation is the use of M scaled versions of a single basis function p(t) = φ
0
(t) as a
signal set,
s
i
(t) = a
i
p(t), for i = 0, . . . , M
where a
i
is the amplitude of waveform i.
Each sent waveform (a.k.a. “symbol”) conveys k = log
2
M bits of information. Note we’re calling it p(t)
instead of φ
0
(t), and a
i
instead of a
i,0
, both for simplicity, since there’s only one basis function.
15.1 Baseband Signal Examples
When you compose a signal of a number of symbols, you’ll just add each timedelayed signal to the transmitted
waveform. The resulting signal we’ll call s(t) (without any subscript) and is
s(t) =
n
a(n)p(t −nT
s
)
where a(n) ∈ ¦a
0
, . . . , a
M−1
¦ is the nth symbol transmitted, and T
s
is the symbol period (not the sample
period!). Instead of just sending one symbol, we are sending one symbol every T
s
units of time. That makes
our symbol rate as 1/T
s
symbols per second.
Note: Keep in mind that when s(t) is to be transmitted on a bandpass channel, it is modulated with a
carrier cos(2πf
c
t),
x(t) = ℜ
_
s(t)e
j2πfct
_
ECE 5520 Fall 2009 59
Rice Figures 5.2.3 and 5.2.6 show continuoustime and discretetime realizations, respectively, of PAM
transmitters and receivers.
Example: Binary Bipolar PAM
Figure 22 shows a 4ary PAM signal set using amplitudes a
0
= −A, a
1
= A. It shows a signal set using square
pulses,
φ
0
(t) = p(t) =
_
1/
√
T
s
, 0 ≤ t < T
s
0, o.w.
0 0.2 0.4 0.6 0.8 1
−1
−0.5
0
0.5
1
Time t
V
a
l
u
e
Figure 22: Example signal for binary bipolar PAM example for A =
√
T
s
.
Example: Binary Unipolar PAM
Unipolar binary PAM uses the amplitudes a
0
= 0, a
1
= A. The average energy per symbol has decreased. This
is also called OnOﬀ Keying (OOK). If the yaxis in Figure 22 was scaled such that the minimum was 0 instead
of −1, it would represent unipolar PAM.
Example: 4ary PAM
A 4ary PAM signal set using amplitudes a
0
= −3A, a
1
= −A, a
2
= A, a
3
= 3A is shown in Figure 23. It shows
a signal set using square pulses,
p(t) =
_
1/
√
T
s
, 0 ≤ t < T
s
0, o.w.
Typical Mary PAM (for all even M) uses the following amplitudes:
−(M −1)A, −(M −3)A, . . . , −A, A, . . . , +(M −3)A, +(M −1)A
Rice Figure 5.2.1 shows the signal space diagram of Mary PAM for diﬀerent values of M.
15.2 Average Bit Energy in Mary PAM
Assuming that each symbol s
i
(t) is equally likely to be sent, we want to calculate the average bit energy c
b
,
which is 1/ log
2
M times the symbol energy c
s
,
c
s
= E
__
∞
−∞
a
2
i
p
2
(t)dt
_
=
_
∞
−∞
E
_
a
2
i
¸
φ
2
0
(t)dt
= E
_
a
2
i
¸
(24)
ECE 5520 Fall 2009 60
0 0.2 0.4 0.6 0.8 1
−3
−2
−1
0
1
2
3
Time t
V
a
l
u
e
Figure 23: Example signal for 4ary PAM example.
Let ¦a
0
, . . . , a
M−1
¦ = −(M −1)A, −(M −3)A, . . . , (M −1)A, as described above to be the typical Mary PAM
case. Then
E
_
a
2
i
¸
=
A
2
M
_
(1 −M)
2
+ (3 −M)
2
+ + (M −1)
2
¸
=
A
2
M
M
m=1
(2m−1 −M)
2
=
A
2
M
M(M
2
−1)
3
= A
2
(M
2
−1)
3
So c
s
=
(M
2
−1)
3
A
2
, and
c
b
=
1
log
2
M
(M
2
−1)
3
A
2
Lecture 10
Today: Review for Exam 1
16 Topics for Exam 1
1. Fourier transform of signals, properties (12)
2. Bandpass signals
3. Sampling and aliasing
4. Orthogonality / Orthonormality
5. Correlation and Covariance
6. Signal space representation
7. Correlation / matched ﬁlter receivers
ECE 5520 Fall 2009 61
Make sure you know how to do each HW problem in HWs 13. It will be good to know the ‘tricks’ of how
each problem is done, since they will reappear.
Read over the lecture notes 17. If anything is confusing, ask me.
You may have one side of an 8.511 sheet of paper for notes. You will be provided with Table 2.4.4 and
Table 2.4.8.
17 Additional Problems
17.1 Spectrum of Communication Signals
1. Bateman Problem 1.9: An otherwise ideal mixer (multiplier) generates an internal DC oﬀset which sums
with the baseband signal prior to multiplication with the cos(2πf
c
t) carrier. How does this aﬀect the
spectrum of the output signal?
2. Haykin & Moyer Problem 2.25. A signal x(t) of ﬁnite energy is applied to a square law device whose
output y(t) is deﬁned by y(t) = x
2
(t). The spectrum of x(t) is bandlimited to −W ≤ ω ≤ W. Prove that
the spectrum of y(t) is bandlimited to −2W ≤ ω ≤ 2W.
3. Consider the pulse shape
x(t) =
_
cos
_
πt
Ts
_
, −
Ts
2
< t <
Ts
2
0, o.w.
(a) Draw a plot of the pulse shape x(t).
(b) Find the Fourier transform of x(t).
4. Rice 2.39, 2.52
17.2 Sampling and Aliasing
1. Assume that x(t) is a bandlimited signal with X(f) = 0 for [f[ > W. When x(t) is sampled at a rate
T =
1
2W
, its samples X
n
are,
X
n
=
_
_
_
1, n = ±1
2, n = 0
0, o.w.
What was the original continuoustime signal x(t)? Or, if x(t) cannot be determined, why not?
2. Rice 2.99, 2.100
17.3 Orthogonality and Signal Space
1. (from L.W. Couch 2007, Problem 249). Three functions are shown in Figure P249.
(a) Show that these functions are orthogonal over the interval (−4, 4).
(b) Find the scale factors needed to scale the functions to make them into an orthonormal set over the
interval (−4, 4).
(c) Express the waveform
w(t) =
_
1, 0 ≤ t ≤ 4
0, o.w.
in signal space using the orthonormal set found in part (b).
ECE 5520 Fall 2009 62
2. Rice Example 5.1.1 on Page 219.
3. Rice 5.2, 5.5, 5.13, 5.14, 5.16, 5.23, 5.26
17.4 Random Processes, PSD
1. (From Couch 2007 Problem 280) An RC lowpass ﬁlter is the circuit shown in Figure 24. Its transfer
function is
X(t) Y(t)
R
C
+ +
 
Figure 24: An RC lowpass ﬁlter.
H(f) =
Y (f)
X(f)
=
1
1 +j2πRCf
Given that the PSD of the input signal is ﬂat and constant at 1, i.e., S
X
(f) = 1, design an RC lowpass
ﬁlter that will attenuate this signal by 20 dB at 15 kHz. That is, ﬁnd the value of RC to satisfy the design
speciﬁcations.
2. Let a binary 1D communication system be described by two possible signals, and noise with diﬀerent
variance depending on which signal is sent, i.e.,
H
0
: r = a
0
+n
0
H
1
: r = a
1
+n
1
where α
0
= −1 and α
1
= 1, and n
0
is zeromean Gaussian with variance 0.1, and n
1
is zeromean Gaussian
with variance 0.3.
(a) What is the conditional pdf of r given H
0
?
(b) What is the conditional pdf of r given H
1
?
17.5 Correlation / Matched Filter Receivers
1. A binary communication system uses the following equallylikely signals,
s
1
(t) =
_
cos
_
πt
T
_
, −
T
2
≤ t ≤
T
2
0, o.w.
s
2
(t) =
_
−cos
_
πt
T
_
, −
T
2
≤ t ≤
T
2
0, o.w.
At the receiver, a signal x(t) = s
i
(t) +n(t) is received.
(a) Is s
1
(t) an energytype or powertype signal?
(b) What is the energy or power (depending on answer to (a)) in s
1
(t)?
(c) Describe in words and a block diagram the operation of an correlation receiver.
ECE 5520 Fall 2009 63
2. Proakis & Salehi 7.8. Suppose that two signal waveforms s
1
(t) and s
2
(t) are orthogonal over the interval
(0, T). A sample function n(t) of a zeromean, white noise process is correlated with s
1
(t) and
2
(t) to
yield,
n
1
=
_
T
0
s
1
(t)n(t)dt
n
2
=
_
T
0
s
2
(t)n(t)dt
Prove that E[n
1
n
2
] = 0.
3. Proakis & Salehi 7.16. Prove that when a sinc pulse g
T
(t) = sin(πt/T)/(πt/T) is passed through its
matched ﬁlter, the output is the same sinc pulse.
Lecture 11
Today: (1) Mary PAM Intro from Lecture 9 notes (2) Probability of Error in PAM
18 Probability of Error in Binary PAM
This is in Section 6.1.2 of Rice.
How are N
0
/2 and σ
N
related? Recall from lecture 7 that the variance of n
k
came out to be N
0
/2. We had
also called the variance of noise component n
k
as σ
2
N
. So:
σ
2
N
=
N
0
2
. (25)
18.1 Signal Distance
We mentioned, when talking about signal space diagrams, a distance between vectors,
d
i,j
= a
i
−a
j
 =
_
M
k=1
(a
i,k
−a
j,k
)
2
_
1/2
For the 1D, two signal case, there is only d
1,0
,
d
1,0
= [a
1
−a
0
[ (26)
18.2 BER Function of Distance, Noise PSD
We have consistently used a
1
> a
0
. In the Gaussian noise case (with equal variance of noise in H
0
and H
1
),
and with P [H
0
] = P [H
1
], we came up with threshold γ = (a
0
+a
1
)/2, and
P [error] = P [x > γ[H
0
] P [H
0
] +P [x < γ[H
1
] P [H
1
]
ECE 5520 Fall 2009 64
Which simpliﬁed using the Q function,
P [x > γ[H
0
] = Q
_
γ −a
0
σ
N
_
P [x < γ[H
1
] = 1 −Q
_
γ −a
1
σ
N
_
= Q
_
a
1
−γ
σ
N
_
(27)
Thus
P [x > γ[H
0
] = Q
_
(a
1
−a
0
)/2
σ
N
_
P [x < γ[H
1
] = Q
_
(a
1
−a
0
)/2
σ
N
_
(28)
So both are equal, and again with P [H
0
] = P [H
1
] = 1/2,
P [error] = Q
_
(a
1
−a
0
)/2
σ
N
_
(29)
We’d assumed a
1
> a
0
, so if we had also calculated for the case a
1
< a
0
, we’d have seen that in general, the
following formula works:
P [error] = Q
_
[a
1
−a
0
[
2σ
N
_
Then, using (25) and (26), we have
P [error] = Q
_
d
0,1
2
_
N
0
/2
_
= Q
_
_
¸
d
2
0,1
2N
0
_
_
(30)
This will be important as we talk about binary PAM. This expression,
Q
_
_
¸
d
2
0,1
2N
0
_
_
Is one that we will see over and over again in this class.
18.3 Binary PAM Error Probabilities
For binary PAM (M = 2) it takes one symbol to encode one bit. Thus ‘bit’ and ‘symbol’ are interchangeable.
In signal space, our signals s
0
(t) = a
0
p(t) and s
1
(t) = a
1
p(t) are just s
0
= a
0
and s
1
= a
1
. Now we can use
(30) to compute the probability of error.
For bipolar signaling, when A
0
= −A and A
1
= A,
P [error] = Q
_
_
¸
4A
2
2N
0
_
_
= Q
_
_
¸
2A
2
N
0
_
_
For unipolar signaling, when A
0
= 0 and A
1
= A,
P [error] = Q
_
_
¸
A
2
2N
0
_
_
ECE 5520 Fall 2009 65
However, this doesn’t take into consideration that the unipolar signaling method uses only half of the energy
to transmit. Since half of the bits are exactly zero, they take no signal energy. Using (24), what is the average
energy of bipolar and unipolar signaling?
Solution: For the bipolar signalling, A
2
m
= A
2
always, so c
b
= c
s
= A
2
. For the unipolar signalling, A
2
m
equals A
2
with probability 1/2 and zero with probability 1/2. Thus c
b
= c
s
=
1
2
A
2
.
Now, we rewrite the probability of error expressions in terms of c
b
. For bipolar signaling,
P [error] = Q
_
_
¸
4A
2
2N
0
_
_
= Q
_
_
2c
b
N
0
_
For unipolar signaling,
P [error] = Q
_
_
2c
b
2N
0
_
= Q
_
_
c
b
N
0
_
Discussion These probabilities of error are shown in Figure 25.
0 2 4 6 8 10 12 14
10
−6
10
−5
10
−4
10
−3
10
−2
10
−1
10
0
E
b
/ N
0
ratio, dB
T
h
e
o
r
e
t
i
c
a
l
P
r
o
b
a
b
i
l
i
t
y
o
f
E
r
r
o
r
Unipolar
Bipolar
Figure 25: Probability of Error in binary PAM signalling.
• Even for the same bit energy c
b
, the bit error rate of bipolar PAM beats unipolar PAM.
• What is the diﬀerence in c
b
/N
0
for a constant probability of error?
• What is the diﬀerence in terms of probability of error for a given c
b
/N
0
?
Lecture 12
Today: (0) Exam 1 Return, (1) MultiHypothesis Detection Theory, (2) Probability of Error in Mary
PAM
ECE 5520 Fall 2009 66
19 Detection with Multiple Symbols
When we only had s
0
(t) and s
1
(t), we saw that our decision was based on which of the following was higher:
f
XH
0
(x[H
0
)P [H
0
] and f
XH
1
(x[H
1
)P [H
1
]
For Mary signals, we’ll see that we need to consider the highest of all M possible events H
m
,
H
0
: r(t) = s
0
(t) +n(t)
H
1
: r(t) = s
1
(t) +n(t)
H
M−1
: r(t) = s
M
(t) +n(t)
which have joint probabilities,
H
0
: f
XH
0
(x[H
0
)P [H
0
]
H
1
: f
XH
1
(x[H
1
)P [H
1
]
H
M−1
: f
XH
M−1
(x[H
M−1
)P [H
M−1
]
We will ﬁnd which of these joint probabilities is highest. For this class, we’ll only consider the case of equally
probable signals. (While equiprobable signals is sometimes not the case for M = 2 binary detection, it is very
rare in higher M communication systems.) If P [H
0
] = = P [H
M−1
] then we only need to ﬁnd the i that
makes the likelihood f
XH
i
(x[H
i
) maximum, that is, maximum likelihood detection.
Symbol Decision = arg max
i
f
XH
i
(x[H
i
)
For Gaussian (conditional) r.v.s with equal variances σ
2
N
, it is better to maximize the log of the likelihood rather
than the likelihood directly, so
log f
XH
i
(x[H
i
) = −
1
2
log(2πσ
2
N
) −
(x −a
i
)
2
2σ
2
N
This is maximized when (x − a
i
)
2
is minimized. Essentially, this is a (squared) distance between x and a
i
. So,
the decision is, ﬁnd the a
i
which is closest to x.
The decision between two neighboring signal vectors a
i
and a
i+1
will be
r
H
i+1
>
<
H
i
γ
i,i+1
=
a
i
+a
i+1
2
As an example: 4ary PAM. See Figure 26.
r
a
1
a
2
a
3
a
4
g
12
g
23
g
34
Figure 26: Signal space representation and decision thresholds for 4ary PAM.
ECE 5520 Fall 2009 67
20 Mary PAM Probability of Error
20.1 Symbol Error
The probability that we don’t get the symbol correct is the probability that x does not fall within the range
between the thresholds in which it belongs. Here, each a
i
= A
i
. Also the noise σ
2
N
= N
0
/2, as discussed above.
Assuming neighboring symbols a
i
are spaced by 2A, the decision threshold is always A away from the symbol
values. For the symbols i in the ‘middle’,
P(symbol error[H
i
) = 2Q
_
A
_
N
0
/2
_
For the symbols i on the ‘sides’,
P(symbol error[H
i
) = Q
_
A
_
N
0
/2
_
So overall,
P(symbol error) =
2(M −1)
M
Q
_
A
_
N
0
/2
_
20.1.1 Symbol Error Rate and Average Bit Energy
How does this relate to the average bit energy c
b
? From last lecture, c
b
=
1
log
2
M
(M
2
−1)
3
A
2
, which means that
A =
_
3 log
2
M
M
2
−1
c
b
So
P(symbol error) =
2(M −1)
M
Q
_
_
6 log
2
M
M
2
−1
c
b
N
0
_
(31)
Equation (31) is plotted in Figure 27.
0 2 4 6 8 10 12 14
10
−6
10
−5
10
−4
10
−3
10
−2
10
−1
10
0
E
b
/ N
0
ratio, dB
T
h
e
o
r
e
t
i
c
a
l
P
r
o
b
a
b
i
l
i
t
y
o
f
E
r
r
o
r
M=2
M=4
M=8
M=16
Figure 27: Probability of Symbol Error in Mary PAM.
ECE 5520 Fall 2009 68
20.2 Bit Errors and Gray Encoding
For binary PAM, there are only two symbols, one will be assigned binary 0 and the other binary 1. When you
make one symbol error (decide H
0
or H
1
in error) then it will cause one bit error.
For Mary PAM, bits and symbols are not synonymous. Instead, we have to carefully assign bit codes to
symbols 1 . . . M.
Example: Bit coding of M = 4 symbols
While the two options shown in Figure 28 both assign 2bits to each symbol in unique ways, one will lead to a
higher bit error rate than the other. Is one is better or worse?
1 1 1 0
“00” “01” “10” “11”
“00” “11” “10” “01” Option1:
Option2:
Figure 28: Two options for assigning bits to symbols.
The key is to recall the model for noise. It will not shift a signal uniformly across symbols. It will tend to
leave the received signal x close to the original signal a
i
. The neighbors of a
i
will be more likely than distant
signal space vectors.
Thus Gray encoding will only change one bit across boundaries, as in Option 2 in Figure 28.
Example: Bit coding of M = 8 symbols
Assign three bits to each symbol such that any two nearest neighbors are diﬀerent in only one bit (Gray
encoding).
Solution: Here is one solution.
f
1 1
“101” “100”
1 0
“001” “000”
3 2
“010” “011”
2 3
“110” “111”
Figure 29: Gray encoding for 8ary PAM.
20.2.1 Bit Error Probabilities
How many bit errors are caused by a symbol error in Mary PAM?
• One. If Gray encoding is used, the errors will tend to be just one bit ﬂipped, more than multiple bits
ﬂipped. At least at high E
b
/N
0
,
P(error) ≈
1
log
2
M
P(symbol error) (32)
• Maybe more, up to log
2
M in the worst case. Then, we need to study further the probability that x will
jump more than one decision region.
As of now  if you have to estimate a bit error rate for Mary PAM  use (32). We will show that this
approximation is very good, in almost all cases. We will also show examples in multidimensional signalling
(e.g., QAM) when this is not a good approximation.
ECE 5520 Fall 2009 69
Lecture 13
Today: (1) Pulse Shaping / ISI, (2) NDim Detection Theory
21 Intersymbol Interference
1. Consider the spectrum of the ideal 1D PAM system with a square pulse.
2. Consider the timedomain of the signal which is close to a rect function in the frequency domain.
We don’t want either: (1) occupies too much spectrum, and (2) occupies too much time.
1. If we try (1) above and use FDMA (frequency division multiple access) then the interference is outofband
interference.
2. If we try (2) above and put symbols right next to each other in time, our own symbols experience
interference called intersymbol interference.
In reality, we want to compromise between (1) and (2) and experience only a small amount of both.
Now, consider the eﬀect of ﬁltering in the path between transmitter and receiver. Filters will be seen in
1. Transmitter: Baseband ﬁlters come from the limited bandwidth, speed of digital circuitry. RF ﬁlters are
used to limit the spectral output of the signal (to meet outofband interference requirements).
2. Channel: The wideband radio channel is a sum of timedelayed impulses. Wired channels have (non
ideal) bandwidth – the attenuation of a cable or twisted pair is a function of frequency. Waveguides have
bandwidth limits (and modes).
3. Receiver: Bandpass ﬁlters are used to reduce interference, noise power entering receiver. Note that the
‘matched ﬁlter’ receiver is also a ﬁlter!
21.1 Multipath Radio Channel
(Background knowledge). The measured radio channel is a ﬁlter, call its impulse response h(τ, t). If s(t) is
transmitted, then
r(t) = s(t) ⋆ h(τ, t)
will be received. Typically, the radio channel is modeled as
h(τ, t) =
L
l=1
α
l
e
jφ
l
δ(τ −τ
l
), (33)
where α
l
and φ
l
are the amplitude and phase of the lth multipath component, and τ
l
is its time delay. Essen
tially, each reﬂection, diﬀraction, or scattering adds an additional impulse to (33) with a particular time delay
(depending on the extra path length compared to the lineofsight path) and complex amplitude (depending
on the losses and phase shifts experienced along the path). The amplitudes of the impulses tend to decay over
time, but multipath with delays of hundreds of nanoseconds often exist in indoor links and delays of several
microseconds often exist in outdoor links.
ECE 5520 Fall 2009 70
The channel in (33) has inﬁnite bandwidth (why?) so isn’t really what we’d see if we looked just at a narrow
band of the channel.
Example: 24002480 MHz Measured Radio Channel
The ‘Delay Proﬁle’ is shown in Figure 30 for an example radio channel. Actually what we observe
PDP(t) = r(t) ⋆ x(t) = (x(t) ⋆ h(t)) ⋆ x(t)
PDP(t) = r(t) ⋆ x(t) = R
X
(t) ⋆ h(t)
PDP(f) = H(f)S
X
(f) (34)
Note that 200 ns is 0.2µs is 1/ 5 MHz. At this symbol rate, each symbol would fully bleed into the next symbol.
(a)
0 100 200 300 400
0
0.05
0.1
0.15
0.2
0.25
Delay, ns
A
m
p
l
i
t
u
d
e
D
e
l
a
y
P
r
o
f
i
l
e
(b)
0 100 200 300 400
0
0.05
0.1
0.15
0.2
0.25
Delay, ns
A
m
p
l
i
t
u
d
e
D
e
l
a
y
P
r
o
f
i
l
e
Figure 30: Multiple delay proﬁles measured on two example links: (a) 13 to 43, and (b) 14 to 43.
How about for a 5 kHz symbol rate? Why or why not?
Other solutions:
• OFDM (Orthogonal Frequency Division Multiplexing)
• Direct Sequence Spread Spectrum / CDMA
• Adaptive Equalization
21.2 Transmitter and Receiver Filters
Example: Bipolar PAM Signal Input to a Filter
How does this eﬀect the correlation receiver and detector? Symbols sent over time are no longer orthogonal
– one symbol ‘leaks’ into the next (or beyond).
22 Nyquist Filtering
Key insight:
• We don’t need the leakage to be zero at all time.
ECE 5520 Fall 2009 71
• The value out of the matched ﬁlter (correlator) is taken only at times nT
s
, for integer n.
Nyquist ﬁltering has the objective of, at the matched ﬁlter output, making the sum of ‘leakage’ or ISI from
one symbol be exactly 0 at all other sampling times nT
s
.
Where have you seen this condition before? This condition, that there is an inner product of zero, is the con
dition for orthogonality of two waveforms. What we need are pulse shapes (waveforms), that when shifted in
time by T
s
, are orthogonal to each other. In more mathematical language, we need p(t) such that
¦φ
n
(t) = p(t −nT
s
)¦
n=...,−1,0,1,...
forms an orthonormal basis. The square root raised cosine, shown in Figure 32 accomplishes this. But lots of
other functions also accomplish this. The Nyquist ﬁltering theorem provides an easy means to come up with
others.
Theorem: Nyquist Filtering
Proof: A necessary and suﬃcient condition for x(t) to satisfy
x(nT
s
) =
_
1, n = 0
0, o.w.
is that its Fourier transform X(f) satisfy
∞
m=−∞
X
_
f +
m
T
s
_
= T
s
Proof: on page 833, Appendix A, of Rice book.
Basically, X(f) could be:
• X(f) = rect(fT
s
), i.e., exactly constant within −
1
2Ts
< f <
1
2Ts
band and zero outside.
• It may bleed into the next ‘channel’ but the sum of
+X
_
f −
1
T
s
_
+X(f) +X
_
f +
1
T
s
_
+
must be constant across all f.
Thus X(f) is allowed to bleed into other frequency bands – but the neighboring frequencyshifted copy of X(f)
must be lower s.t. the sum is constant. See Figure 31.
If X(f) only bleeds into one neighboring channel (that is, X(f) = 0 for all [f[ >
1
Ts
), we denote the diﬀerence
between the ideal rect function and our X(f) as ∆(f),
∆(f) = [X(f) −rect(fT
s
)[
then we can rewrite the Nyquist Filtering condition as,
∆
_
1
2T
s
−f
_
= ∆
_
1
2T
s
+f
_
for −1/T
s
≤ f < 1/T
s
. Essentially, symmetric about f = 1/(2T
s
). See Figure 31
ECE 5520 Fall 2009 72
1
2T
f
X f ( )
X f+ T ( / ) 1
X f X f+ T ( )+ ( / ) 1
1
T
Figure 31: The Nyquist ﬁltering criterion – 1/T
s
frequencyshifted copies of X(f) must add up to a constant.
Bateman presents this whole condition as “A Nyquist channel response is characterized by the transfer
function having a transition band that is symmetrical about a frequency equal to 0.5 1/T
s
.”
Activity: Doityourself Nyquist ﬁlter. Take a sheet of paper and fold it in half, and then in half the other
direction. Cut a function in the thickest side (the edge that you just folded). Leave a tiny bit so that it is not
completely disconnected into two halves. Unfold. Drawing a horizontal line for the frequency f axis, the middle
is 0.5/T
s
, and the vertical axis it X(f).
22.1 Raised Cosine Filtering
H
RC
(f) =
_
¸
_
¸
_
T
s
, 0 ≤ [f[ ≤
1−α
2Ts
Ts
2
_
1 + cos
_
πTs
α
_
[f[ −
1−α
2Ts
___
,
1−α
2Ts
≤ [f[ ≤
1+α
2Ts
0, o.w.
(35)
22.2 SquareRoot Raised Cosine Filtering
But we usually need do design a system with two identical ﬁlters, one at the transmitter and one at the receiver,
which in series, produce a zeroISI ﬁlter. In other words, we need [H(f)[
2
to meet the Nyquist ﬁltering condition.
H
RRC
(f) =
_
¸
¸
_
¸
¸
_
√
T
s
, 0 ≤ [f[ ≤
1−α
2Ts _
Ts
2
_
1 + cos
_
πTs
α
_
[f[ −
1−α
2Ts
___
,
1−α
2Ts
≤ [f[ ≤
1+α
2Ts
0, o.w.
(36)
23 Mary Detection Theory in Ndimensional signal space
We are going to start to talk about QAM, a modulation with two dimensional signal vectors, and later even
higher dimensional signal vectors. We have developed Mary detection theory for 1D signal vectors, and now
we will extend that to Ndimensional signal vectors.
Setup:
• Transmit: one of M possible symbols, a
0
, . . . , a
M−1
.
ECE 5520 Fall 2009 73
−6 −5 −4 −3 −2 −1 0 1 2 3 4 5 6 7
−0.05
0
0.05
0.1
0.15
0.2
0.25
Time, t/T
sy
B
a
s
i
s
F
u
n
c
t
i
o
n
φ
i
(
t
)
Figure 32: Two SRRC Pulses, delayed in time by nT
s
for any integer n, are orthogonal to each other.
• Receive: the symbol plus noise:
H
0
: X = a
0
+n
H
M−1
: X = a
M−1
+n
• Assume: n is multivariate Gaussian, each component n
i
is independent with zero mean and variance
σ
N
= N
0
/2.
• Assume: Symbols are equally likely.
• Question: What are the optimal decision regions?
When symbols are equally likely, the optimal decision turns out to be given by the maximum likelihood
receiver,
ˆ
i = argmax
i
log f
XH
i
(x[H
i
) (37)
Here,
f
XH
i
(x[H
i
) =
1
(2πσ
2
N
)
N/2
exp
_
−
j
(x
j
−a
i,j
)
2
2σ
2
N
_
which can also be written as
f
XH
i
(x[H
i
) =
1
(2πσ
2
N
)
N/2
exp
_
−
x −a
i

2
2σ
2
N
_
So when we want to solve (37) we can simplify quickly to:
ˆ
i = argmax
i
_
log
1
(2πσ
2
N
)
N/2
−
x −a
i

2
2σ
2
N
_
ˆ
i = argmax
i
−
x −a
i

2
2σ
2
N
ˆ
i = argmin
i
x −a
i

2
ˆ
i = argmin
i
x −a
i
 (38)
ECE 5520 Fall 2009 74
Again: Just ﬁnd the a
i
in the signal space diagram which is closest to x.
Pairwise Comparisons When is x closer to a
i
than to a
j
for some other signal space point j? Solution:
The two decision regions are separated by a straight line (Note: replace “line” with plane in 3D, or
subspace in ND). To ﬁnd this line:
1. Draw a line segment connecting a
i
and a
j
.
2. Draw a point in the middle of that line segment.
3. Draw the perpendicular bisector of the line segment through that point.
Why?
Solution: Try to ﬁnd the locus of point x which satisfy
x −a
i

2
= x −a
j

2
You can do this by using the inner product to represent the magnitude squared operator:
(x −a
i
)
T
(x −a
i
) = (x −a
j
)
T
(x −a
j
)
Then use FOIL (multiply out), cancel, and reorganize to ﬁnd a linear equation in terms of x. This is left as an
exercise.
Decision Regions Each pairwise comparison results in a linear division of space. The combined decision
region of R
i
is the space which is the intersection of all pairwise decision regions. (All conditions must be
satisﬁed.)
Example: Optimal Decision Regions
See Figure 33.
Lecture 14
Today: (1) QAM & PSK, (2) QAM Probability of Error
24 Quadrature Amplitude Modulation (QAM)
Quadrature Amplitude Modulation (QAM) is a twodimensional signalling method which uses the inphase and
quadrature (cosine and sine waves, respectively) as the two dimensions. Thus QAM uses two basis functions.
These are:
φ
0
(t) =
√
2p(t) cos(ω
0
t)
φ
1
(t) =
√
2p(t) sin(ω
0
t)
where p(t) is a pulse shape (like the ones we’ve looked at previously) with support on T
1
≤ t ≤ T
2
. That is, p(t)
is only nonzero within that window.
Previously, we’ve separated frequency upconversion and downconversion from pulse shaping. This deﬁni
tion of the orthonormal basis speciﬁcally considers a pulse shape at a frequency ω
0
. We include it here because
ECE 5520 Fall 2009 75
(a)
(b)
(c)
(d)
Figure 33: Example signal space diagrams. Draw the optimal decision regions.
ECE 5520 Fall 2009 76
it is critical to see how, with the same pulse shape p(t), we can have two orthogonal basis functions. (This is
not intuitive!)
We have two restrictions on p(t) that makes these two basis functions orthonormal:
• p(t) is unitenergy.
• p(t) is ‘low pass’; that is, it has low frequency content compared to ω
0
t.
24.1 Showing Orthogonality
Let’s show that the basis functions are orthogonal. We’ll need these cosine / sine function relationships:
sin(2A) = 2 cos Asin A
cos A−cos B = −2 sin
_
A+B
2
_
sin
_
A−B
2
_
As a ﬁrst example, let p(t) = c for a constant c , for T
1
≤ t ≤ T
2
. Are the two bases orthogonal? First, show
that
¸φ
0
(t), φ
1
(t)) = c
2
sin(ω
0
(T
2
+T
1
)) sin(ω
0
(T
2
−T
1
))
ω
0
Solution: First, see how far we can get before plugging in for p(t):
¸φ
0
(t), φ
1
(t)) =
_
T
2
T
1
√
2p(t) cos(ω
0
t)
√
2p(t) sin(ω
0
t)dt
= 2
_
T
2
T
1
p
2
(t) cos(ω
0
t) sin(ω
0
t)dt
=
_
T
2
T
1
p
2
(t) sin(2ω
0
t)dt
Next, consider the constant p(t) = c for a constant c, for T
1
≤ t ≤ T
2
.
¸φ
0
(t), φ
1
(t)) = c
2
_
T
2
T
1
sin(2ω
0
t)dt
= c
2
_
−
cos(2ω
0
t)
2ω
0
¸
¸
¸
¸
T
2
T
1
= −c
2
cos(2ω
0
T
2
) −cos(2ω
0
T
1
)
2ω
0
= −c
2
cos(2ω
0
T
2
) −cos(2ω
0
T
1
)
2ω
0
I want the answer in terms of T
2
−T
1
(for reasons explained below), so
¸φ
0
(t), φ
1
(t)) = c
2
sin(ω
0
(T
2
+T
1
)) sin(ω
0
(T
2
−T
1
))
ω
0
We can see that there are two cases:
ECE 5520 Fall 2009 77
1. The pulse duration T
2
− T
1
is an integer number of periods. That is, ω
0
(T
2
− T
1
) = πk for some integer
k. In this case, the right sin is zero, and so the correlation is exactly zero.
2. Otherwise, the numerator bounded above and below by +1 and 1, because it is a sinusoid. That is,
−
c
2
ω
0
≤ ¸φ
0
(t), φ
1
(t)) ≤
c
2
ω
0
Typically, ω
0
is a large number. For example, frequencies 2πω
0
could be in the MHz or GHz ranges.
Certainly, when we divide by numbers on the order of 10
6
or 10
9
, we’re going to get a very small inner
product. For engineering purposes, φ
0
(t) and φ
1
(t) are orthogonal.
Finally, we can attempt the proof for the case of arbitrary pulse shape p(t). In this case, we use the ‘lowpass’
assumption that the maximum frequency content of p(t) is much lower than 2πω
0
. This assumption allows us
to assume that p(t) is nearly constant over the course of one cycle of the carrier sinusoid.
This is wellillustrated in Figure 5.15 in the Rice book (page 298). In this ﬁgure, we see a pulse modulated
by a sine wave at frequency 2πω
0
. Zooming in on any few cycles, you can see that the pulse p
2
(t) is largely
constant across each cycle. Thus, when we integrate p
2
(t) sin(2ω
0
t) across one cycle, we’re going to end up with
approximately zero. Proof?
Solution: How many cycles are there? Consider the period [T
1
, T
2
]. How many times can it be divided by
π/ω
0
? Let the integer number be
L =
_
(T
2
−T
1
)ω
0
π
_
.
Then each cycle is in the period, [T
1
+ (i − 1)π/ω
0
, T
1
+ iπ/ω
0
], for i = 1, . . . , L. Within each of these cycles,
assume p
2
(t) is nearly constant. Then, in the same way that the earlier integral was zero, this part of the
integral is zero here. The only remainder is the remainder (partial cycle) period, [T
1
+Lπ/ω
0
, T
2
]. Thus
¸φ
0
(t), φ
1
(t)) = p
2
(T
1
+Lπ/ω
0
)
_
T
2
T
1
+Lπ/ω
0
sin(2ω
0
t)dt
= p
2
(T
1
+Lπ/ω
0
)
cos(2ω
0
T
2
) −cos(2ω
0
T
1
+ 2πL)
2ω
0
= p
2
(T
1
+Lπ/ω
0
)
sin(ω
0
(T
2
+T
1
)) sin(ω
0
(T
2
−T
1
))
ω
0
Again, the inner product is bounded on either side:
−
p
2
(T
1
+Lπ/ω
0
)
ω
0
≤ ¸φ
0
(t), φ
1
(t))
p
2
(T
1
+Lπ/ω
0
)
ω
0
which for very large ω
0
, is nearly zero.
24.2 Constellation
With these two basis functions, Mary QAM is deﬁned as an arbitrary signal set a
0
, . . . , a
M−1
, where each
signal space vector a
k
is twodimensional:
a
k
= [a
k,0
, a
k,1
]
T
ECE 5520 Fall 2009 78
The signal corresponding to symbol k in (Mary) QAM is thus
s(t) = a
k,0
φ
0
(t) +a
k,1
φ
1
(t)
= a
k,0
√
2p(t) cos(ω
0
t) +a
k,1
√
2p(t) sin(ω
0
t)
=
√
2p(t) [a
k,0
cos(ω
0
t) +a
k,1
sin(ω
0
t)]
Note that we could also write the signal s(t) as
s(t) =
√
2p(t)R
_
a
k,0
e
−jω
0
t
+ja
k,1
e
−jω
0
t
_
=
√
2p(t)R
_
e
−jω
0
t
(a
k,0
+ja
k,1
)
_
(39)
In many textbooks, you will see them write a QAM signal in shorthand as
s
CB
(t) = p(t)(a
k,0
+ja
k,1
)
This is called ‘Complex Baseband’. If you do the following operation you can recover the real signal s(t) as
s(t) =
√
2R
_
e
−jω
0
t
s
SB
(t)
_
You will not be responsible for Complex Baseband notation, but you should be able to read other books and
know what they’re talking about.
Then, we can see that the signal space representation a
k
is given by
a
k
= [a
k,0
, a
k,1
]
T
for k = 0, . . . , M −1
24.3 Signal Constellations
M=64QAM
Figure 34: Square signal constellation for 64QAM.
• See Figure 5.17 for examples of square QAM. These constellations use M = 2
a
for some even integer a,
and arrange the points in a grid. One such diagram for M = 64 square QAM is also given here in Figure
34.
• Figure 5.18 shows examples of constellations which use M = 2
a
for some odd integer a, and arrange the
points in a grid. These are either rectangular grids, or use squares with the corners cut out.
ECE 5520 Fall 2009 79
24.4 Angle and Magnitude Representation
You can plot a
k
in signal space and see that it has a magnitude (distance from the origin) of [a
k
[ =
_
a
2
k,0
+a
2
k,1
and angle of ∠a
k
= tan
−1
a
k,1
a
k,0
. In the continuous time signal s(t) this is
s(t) =
√
2p(t)[a
k
[ cos(ω
0
t +∠a
k
)
24.5 Average Energy in MQAM
Recall that the average energy is calculated as:
c
s
=
1
M
M−1
i=0
[a
i
[
2
c
b
=
1
M log
2
M
M−1
i=0
[a
i
[
2
where c
s
is the average energy per symbol and c
b
is the average energy per bit. We’ll work in class some
examples of ﬁnding c
b
in diﬀerent constellation diagrams.
24.6 PhaseShift Keying
Some implementations of QAM limit the constellation to include only signal space vectors with equal magnitude,
i.e.,
[a
0
[ = [a
1
[ = = [a
M−1
[
The points a
i
for i = 0, . . . , M −1 are uniformly spaced on the unit circle. Some examples are shown in Figure
35.
BPSK For example, binary phaseshift keying (BPSK) is the case of M = 2. Thus BPSK is the same as
bipolar (2ary) PAM. What is the probability of error in BPSK? The same as in bipolar PAM, i.e., for equally
probable symbols,
P [error] = Q
_
_
2c
b
N
0
_
.
QPSK M = 4 PSK is also called quadrature phase shift keying (QPSK), and is shown in Figure 35(a). Note
that the rotation of the signal space diagram doesn’t matter, so both ‘versions’ are identical in concept (although
would be a slightly diﬀerent implementation). Note how QPSK is the same as M = 4 square QAM.
24.7 Systems which use QAM
See [Couch 2007], wikipedia, and the Rice book:
• Digital Microwave Relay, various manufacturerspeciﬁc protocols. 6 GHz, and 11 GHz.
• Dialup modems: use a M = 16 or M = 8 QAM constellation.
• DSL. G.DMT uses multicarrier (up to 256 carriers) methods (OFDM), and on each narrowband (4.3kHz)
carrier, it can send up to 2
15
QAM (32,768 QAM). G.Lite uses up to 128 carriers, each with up to 2
8
= 256
QAM.
ECE 5520 Fall 2009 80
(a)
QPSK QPSK
(b)
M=8 M=16
Figure 35: Signal constellations for (a) M = 4 PSK and (b) M = 8 and M = 16 PSK.
• Cable modems. Upstream: 6 MHz bandwidth channel, with 64 QAM or 256 QAM. Downstream: QPSK
or 16 QAM.
• 802.11a and 802.11g: Adaptive modulation method, uses up to 64 QAM.
• Digital Video Broadcast (DVB): APSK used in ETSI standard.
25 QAM Probability of Error
25.1 Overview of Future Discussions on QAM
Before we get into details about the probabilities of error, you should realize that there are two main consider
ations when a constellation is designed for a particular M:
1. Points with equal distances separating them and their neighboring points tend to reduce probability of
error.
2. The highest magnitude points (furthest from the origin) strongly (negatively) impact average bit energy.
There are also some practical considerations that we will discuss
1. Some constellations have more complicated decision regions which require more computation to implement
in a receiver.
2. Some constellations are more diﬃcult (energyconsuming) to amplify at the transmitter.
ECE 5520 Fall 2009 81
25.2 Options for Probability of Error Expressions
In order of preference:
1. Exact formula. In a few cases, there is an exact expression for P [error] in an AWGN environment.
2. Union bound. This is a provable upper bound on the probability of error. It is not an approximation. It
can be used for “worst case” analysis which is often very useful for engineering design of systems.
3. Nearest Neighbor Approximation. This is a way to get a solution that is analytically easier to handle.
Typically these approximations are good at high
E
b
N
0
.
25.3 Exact Error Analysis
Exact probability of error formulas in Ndimensional modulations can be very diﬃcult to ﬁnd. This is because
our decisions regions are more complex than one threshold test. They require an integration of a ND Gaussian
pdf across an area. We needed a Q function to get a tail probability for a 1D Gaussian pdf. Now we need
not just a tail probability... but more like a section probability which is the volume under some part of the
Ndimensional Gaussian pdf.
For example, consider Mary PSK. Essentially, we must ﬁnd calculate the probability of symbol error as 1
M=8 M=16
Figure 36: Signal space diagram for Mary PSK for M = 8 and M = 16.
minus the area in the sector within ±
π
M
of the correct angle φ
i
. This is,
P(symbol error) = 1 −
_
r∈R
i
1
2πσ
2
e
−
r−α
i
2
2σ
2
(40)
This integral is a double integral, and we don’t generally have any exact expression to use to express the result
in general.
25.4 Probability of Error in QPSK
In QPSK, the probability of error is analytically tractable. Consider the QPSK constellation diagram, when
Gray encoding is used. You have already calculated the decision regions for each symbol; now consider the
decision region for the ﬁrst bit.
The decision is made using only one dimension, of the received signal vector x, speciﬁcally x
1
. Similarly,
the second bit decision is made using only x
2
. Also, the noise contribution to each element is independent. The
decisions are decoupled – x
2
has no impact on the decision about bit one, and x
1
has no impact on the decision
ECE 5520 Fall 2009 82
on bit two. Since we know the bit error probability for each bit decision (it is the same as bipolar PAM) we can
see that the bit error probability is also
P [error] = Q
_
_
2c
b
N
0
_
(41)
This is an extraordinary result – the bit rate will double in QPSK, but in theory, the bit error rate does not
increase. As we will show later, the bandwidth of QPSK is identical to that of BPSK.
25.5 Union Bound
From 5510 or an equivalent class (or a Venn diagram) you may recall the probability formula, that for two
events E and F that
P [E ∪ F] = P [E] +P [F] −P [E ∩ F]
You can prove this from the three axioms of probability. (This holds for any events E and F!) Then, using the
above formula, and the ﬁrst axiom of probability, we have that
P [E ∪ F] ≤ P [E] +P [F] . (42)
Furthermore, from (42) it is straightforward to show that for any list of sets E
1
, E
2
, . . . E
n
we have that
P
_
n
_
i=1
E
i
_
≤
n
i=1
P [E
i
] (43)
This is called the union bound, and it is very useful across communications. If you know one inequality, know
this one. It is useful when the overlaps E
i
∩ E
j
are small but diﬃcult to calculate.
25.6 Application of Union Bound
In this class, our events are typically decision events. The event E
i
may represent the event that we decide H
i
,
when some other symbol not equal to i was actually sent. In this case,
P [symbol error] ≤
n
i=1
P [E
i
] (44)
The union bound gives a conservative estimate. This can be useful for quick initial study of a modulation
type.
Example: QPSK
First, let’s study the union bound for QPSK, as shown in Figure 37(a). Assume s
1
(t) is sent. We know that
(from previous analysis):
P [error] = Q
_
_
2c
b
N
0
_
So the probability of symbol error is one minus the probability that there were no error in either bit,
P [symbol error] = 1 −
_
1 −Q
_
_
2c
b
N
0
__
2
(45)
In contrast, let’s calculate the union bound on the probability of error. We deﬁne the two error events as:
ECE 5520 Fall 2009 83
a
1
E
2
E
4
a
2
a
3
a
0
a
1
E
2
E
4
a
2
a
3
a
0
(a) (b)
Figure 37: Union bound examples. (a) is QPSK with symbols of equal energy
√
c
s
. In (b) a
1
= −a
3
=
[0,
_
3c
s
/2]
T
and a
2
= −a
0
= [
_
c
s
/2, 0]
T
.
• E
2
, the event that r falls on the wrong side of the decision boundary with a
2
.
• E
0
, the event that r falls on the wrong side of the decision boundary with a
0
.
Then we write
P [symbol error[H
1
] = P [E
2
∪ E
0
∪ E
3
]
Questions:
1. Is this equation exact?
2. Is E
2
∪ E
0
∪ E
3
= E
2
∪ E
0
? Why or why not?
Because of the answer to the question (2.),
P [symbol error[H
1
] = P [E
2
∪ E
0
]
But don’t be confused. The Rice book will not tell you to do this! His strategy is to ignore redundancies,
because we are only looking for an upper bound any way. Either way produces an upper bound, and in fact,
both produce nearly identical numerical results. However, it is perfectly acceptable (and in fact a better bound)
to remove redundancies and then apply the bound.
Then, we use the union bound.
P [symbol error[H
1
] ≤ P [E
2
] +P [E
0
]
These two probabilities are just the probability of error for a binary modulation, and both are identical, so
P [symbol error[H
1
] ≤ 2Q
_
_
2c
b
N
0
_
What is missing / What is the diﬀerence in this expression compared to (45)?
See Figure 38 to see the union bound probability of error plot, compared to the exact expression. Only at
very low E
b
/N
0
is there any noticeable diﬀerence!
Example: 4QAM with two amplitude levels
This is shown (poorly) in Figure 37(b). The amplitudes of the top and bottom symbols are
√
3 times the
ECE 5520 Fall 2009 84
0 1 2 3 4 5 6
10
−2
10
−1
E
b
/N
0
Ratio, dB
P
r
o
b
a
b
i
l
i
t
y
o
f
S
y
m
b
o
l
E
r
r
o
r
QPSK Exact
QPSK Union Bound
Figure 38: For QPSK, the exact probability of symbol error expression vs. the union bound.
amplitude of the symbols on the right and left. (They are positioned to keep the distance between points in the
signal space equal to
_
2c
s
/N
0
.) I am calling this ”2amplitude 4QAM” (I made it up).
What is the union bound on the probability of symbol error, given H
1
? Solution: Given symbol 1, the
probability is the same as above. Deﬁning E
2
and E
0
as above, these two distances between symbol a
1
and a
2
or a
0
are the same:
_
2c
s
/N
0
. Thus the formula for the union bound is the same.
What is the union bound on the probability of symbol error, given H
2
? Solution: Now, it is
P [symbol error[H
2
] ≤ 3Q
_
_
2c
b
N
0
_
So, overall, the union bound on probability of symbol error is
P [symbol error[H
2
] ≤
3 2 + 2 2
4
Q
_
_
2c
b
N
0
_
How about average energy? For QPSK, the symbol energies are all equal. Thus c
av
= c
s
. For the two
amplitude 4QAM modulation,
c
av
= c
s
2(0.5) + 2(1.5)
4
= c
s
Thus there is no advantage to the twoamplitude QAM modulation in terms of average energy.
25.6.1 General Formula for Union Boundbased Probability of Error
This is given in Rice 6.76:
P [symbol error] ≤
1
M
M−1
m=0
M−1
n=0
n=m
P [decide H
n
[H
m
]
Note the ≤ sign. This means the actual P [symbol error] will be at most this value. It is probably less than this
value. This is a general formula and is not necessarily the best upper bound. What do we mean? If I draw any
ECE 5520 Fall 2009 85
function that is always above the actual P [symbol error] formula, I have drawn an upper bound. But I could
draw lots of functions that are upper bounds, some higher than others.
In particular, for some of the error events, decide H
n
[H
m
may be redundant, and do not need to be included.
We will talk about the concept of “neighboring” symbols to symbol i. These neighbors are the ones that are
necessary to include in the union bound.
Note that the Rice book uses E
avg
where I use c
s
to denote average symbol energy. The Rice book uses E
b
where I use c
b
to denote average bit energy. I prefer c
s
to make clear we are talking about Joules per symbol.
Given this, the pairwise probability of error, P [decide H
n
[H
m
], is given by:
P [decide H
n
[H
m
] = Q
_
_
¸
d
2
m,n
2N
0
_
_
= Q
_
_
¸
d
2
m,n
2c
s
c
s
N
0
_
_
(46)
where d
m,n
is the Euclidean distance between the symbols m and n in the signal space diagram. If we use a
constant A when describing the signal space vectors (as we usually do), then, since c
s
will be proportional to
A
2
and d
2
m,n
will be proportional to A
2
, the factors A will cancel out of the expression.
Proof of (46)?
Lecture 14
Today: (1) QAM Probability of Error, (2) Union Bound and Approximations
Lecture 15
Today: (1) Mary PSK and QAM Analysis
26 QAM Probability of Error
26.1 NearestNeighbor Approximate Probability of Error
As it turns out, the probability of error is often well approximated by the terms in the Union Bound (Lecture 14,
Section 2.6.1) with the smallest d
m,n
. This is because higher d
m,n
means a higher argument in the Qfunction,
which in turn means a lower value of the Qfunction. A little extra distance means a much lower value of the
Qfunction. So approximately,
P [symbol error] ≈
N
min
M
Q
_
_
¸
d
2
min
2N
0
_
_
P [symbol error] ≈
N
min
M
Q
_
_
¸
d
2
min
2c
s
c
s
N
0
_
_
(47)
where
d
min
= min
m=n
d
m,n
, m = 0, . . . , M −1, and n = 0, . . . , M −1.
and N
min
is the number of pairs of symbols which are separated by distance d
min
.
ECE 5520 Fall 2009 86
26.2 Summary and Examples
We’ve been formulating in class a pattern or stepbystep process for ﬁnding the probability of bit or symbol
error in arbitrary Mary PSK and QAM modulations. These steps include:
1. Calculate the average energy per symbol, c
s
, as a function of any amplitude parameters (typically
we’ve been using A).
2. Calculate the average energy per bit, c
b
, using c
b
= c
s
/ log
2
M.
3. Draw decision boundaries. It is not necessary to include redundant boundary information, that is, an
error region which is completely contained in a larger error region.
4. Calculate pairwise distances between pairs of symbols which contribute to the decision boundaries.
5. Convert distances to be in terms of c
b
/N
0
, if they are not already, using the answer to step (1.).
6. Calculate the union bound on probability of symbol error using the formula derived in the previous
lecture,
P [symbol error] ≤
1
M
M−1
m=0
M−1
n=0
n=m
P [decide H
n
[H
m
]
P [decide H
n
[H
m
] = Q
_
_
¸
d
2
m,n
2N
0
_
_
7. Approximate using the nearest neighbor approximation if needed:
P [symbol error] ≈
N
min
M
Q
_
_
¸
d
2
min
2N
0
_
_
where d
min
is the shortest pairwise distance in the constellation diagram, and N
min
is the number of
ordered pairs (i, j) which are that distance apart, d
i,j
= d
min
(don’t forget to count both (i, j) and (j, i)!).
8. Convert probability of symbol error into probability of bit error if Gray encoding can be assumed,
P [error] ≈
1
log
2
M
P [symbol error]
Example: Additional QAM / PSK Constellations
Solve for the probability of symbol error in the signal space diagrams in Figure 39. You should calculate:
• An exact expression if one should happen to be available,
• The Union bound,
• The nearestneighbor approximation.
ECE 5520 Fall 2009 87
Figure 39: Signal space diagram for some example (made up) constellation diagrams.
ECE 5520 Fall 2009 88
on, ﬁrst, probability of symbol error, and then also probability of bit error. The plots are in terms of amplitude
A, but all probability of error expressions should be written in terms of c
b
/N
0
.
Lecture 16
Today: (1) QAM Examples (Lecture 15) , (2) FSK
27 Frequency Shift Keying
Frequency modulation in general changes the center frequency over time:
x(t) = cos(2πf(t)t).
Now, in frequency shift keying, symbols are selected to be sinusoids with frequency selected among a set of M
diﬀerent frequencies ¦f
0
, f
1
, . . . f
M−1
¦.
27.1 Orthogonal Frequencies
We will want our cosine wave at these M diﬀerent frequencies to be orthonormal. Consider f
k
= f
c
+k∆f, and
thus
φ
k
(t) =
√
2p(t) cos(2πf
c
t + 2πk∆ft) (48)
where p(t) is a pulse shape, which for example, could be a rectangular pulse:
p(t) =
_
1/
√
T
s
, 0 ≤ t ≤ T
s
0, o.w.
• Does this function have unit energy?
• Are these functions orthogonal?
Solution:
¸φ
k
(t), φ
m
(t)) =
_
∞
t=−∞
(2/T
s
) cos(2πf
c
t + 2πk∆ft) cos(2πf
c
t + 2πk∆ft)
= 1/T
s
_
Ts
t=0
cos(2π(k −m)∆ft)dt +
1/T
s
_
Ts
t=0
cos(4πf
c
t + 2π(k +m)∆ft)dt
= 1/T
s
_
sin(2π(k −m)∆ft)
2π(k −m)∆f
¸
¸
¸
¸
Ts
t=0
=
sin(2π(k −m)∆fT
s
)
2π(k −m)∆fT
s
Yes, they are orthogonal if 2π(k − m)∆fT
s
is a multiple of π. (They are also approximately orthogonal if
∆f is realy big, but we don’t want to waste spectrum.) For general k ,= m, this requires that ∆fT
s
= n/2, i.e.,
∆f = n
1
2T
s
= n
f
sy
2
ECE 5520 Fall 2009 89
for integer n. (Otherwise, no they’re not.)
Thus we need to plug into (48) for ∆f = n
1
2Ts
for some integer n in order to have an orthonormal basis.
What n? We will see that we either use an n of 1 or 2.
Signal space vectors a
i
are given by
a
0
= [A, 0, . . . , 0]
a
1
= [0, A, . . . , 0]
.
.
.
a
M−1
= [0, 0, . . . , A]
What is the energy per symbol? Show that this means that A =
√
c
s
.
For M = 2 and M = 3 these vectors are plotted in Figure 40.
M=2FSK M=3FSK
Figure 40: Signal space diagram for M = 2 and M = 3 FSK modulation.
27.2 Transmission of FSK
At the transmitter, FSK can be seen as a switch between M diﬀerent carrier signals, as shown in Figure 41.
FSK
Out
Signalf (t)
1
Signalf (t)
2
Switch
Figure 41: Block diagram of a binary FSK transmitter.
But usually it is generated from a single VCO, as seen in Figure 42.
Def ’n: Voltage Controlled Oscillator (VCO)
A sinusoidal generator with frequency that linearly proportional to an input voltage.
Note that we don’t need to sent square wave input into the VCO. In fact, bandwidth will be lower when we
send less steep transitions, for example, SRRC pulses.
ECE 5520 Fall 2009 90
Def ’n: Continuous Phase Frequency Shift Keying
FSK with no phase discontinuity between symbols. In other words, the phase of the output signal φ
k
(t) does
not change instantaneously at symbol boundaries iT
s
for integer i, and thus φ(t
−
+iT
s
) = φ(t
+
+iT
s
) where t
−
and t
+
are the limiting times just to the left and to the right of 0, respectively.
There are a variety of ﬂavors of CPFSK, which are beyond the scope of this course.
VCO
FSK
Out
Frequency
Signalf(t)
Figure 42: Block diagram of a binary FSK transmitter.
27.3 Reception of FSK
FSK reception is either phase coherent or phase noncoherent. Here, there are M possible carrier frequencies,
so we’d need to know and be synchronized to M diﬀerent phases θ
i
, one for each symbol frequency:
cos(2πf
c
t +θ
0
)
cos(2πf
c
t + 2π∆ft +θ
1
)
.
.
.
cos(2πf
c
t + 2π(M −1)∆ft +θ
M−1
)
27.4 Coherent Reception
FSK reception can be done via a correlation receiver, just as we’ve seen for previous modulation types.
Each phase θ
k
is estimated to be
ˆ
θ
k
by a separate phaselocked loop (PLL). In one ﬂavor of CPFSK (called
Sunde’s FSK), the carrier cos(2πf
k
t + θ
k
) is sent with the transmitted signal, to aid in demodulation (at the
expense of the additional energy). This is the only case where I’ve heard of using coherent FSK reception.
As M gets high, coherent detection becomes diﬃcult. These M PLLs must operate even though they can
only synchronize when their symbol is sent, 1/M of the time (assuming equallyprobable symbols). Also, having
M PLLs is a drawback.
27.5 Noncoherent Reception
Notice that in Figure 40, the sign or phase of the sinusoid is not very important – only one symbol exists in
each dimension. In noncoherent reception, we just measure the energy in each frequency.
This is more diﬃcult than it sounds, though – we have a fundamental problem. As we know, for every
frequency, there are two orthogonal functions, cosine and sine (see QAM and PSK). Since we will not know the
phase of the received signal, we don’t know whether or not the energy at frequency f
k
correlates highly with
the cosine wave or with the sine wave. If we only correlate it with one (the sine wave, for example), and the
phase makes the signal the other (a cosine wave) we would get a inner product of zero!
The solution is that we need to correlate the received signal with both a sine and a cosine wave at the
frequency f
k
. This will give us two inner products, lets call them x
I
k
using the capital I to denote inphase and
x
Q
k
with Q denoting quadrature.
ECE 5520 Fall 2009 91
cos(2 ) pf t
i
sin(2 ) pf t
i
x
i
Q
x
i
I
length:
() +
2
()
2
x
i
I
x
i
Q
Figure 43: The energy in a noncoherent FSK receiver at one frequency f
k
is calculated by ﬁnding its correlation
with the cosine wave (x
I
k
) and sine wave (x
Q
k
) at the frequency of interest, f
k
, and calculating the squared length
of the vector [x
I
k
, x
Q
k
]
T
.
The energy at frequency f
k
, that is,
c
f
k
= (x
I
k
)
2
+ (x
Q
k
)
2
is calculated for each frequency f
k
, k = 0, 1, . . . , M−1. You could prove this, but the optimal detector (Maximum
likelihood detector) is then to decide upon the frequency with the maximum energy c
f
k
. Is this a threshold
test?
We would need new analysis of the FSK noncoherent detector to ﬁnd its analytical probability of error.
27.6 Receiver Block Diagrams
The block diagram of the the noncoherent FSK receiver is shown in Figures 45 and 46 (copied from the Proakis
& Salehi book). Compare this to the coherent FSK receiver in Figure 44.
Figure 44: (Proakis & Salehi) Phasecoherent demodulation of Mary FSK signals.
ECE 5520 Fall 2009 92
Figure 45: (Proakis & Salehi) Demodulation of Mary FSK signals for noncoherent detection.
Figure 46: (Proakis & Salehi) Demodulation and squarelaw detection of binary FSK signals.
ECE 5520 Fall 2009 93
27.7 Probability of Error for Coherent Binary FSK
First, let’s look at coherent detection of binary FSK.
1. What is the detection threshold line separating the two decision regions?
2. What is the distance between points in the Binary FSK signal space?
What is the probability of error for coherent binary FSK? It is the same as bipolar PAM, but the symbols are
spaced diﬀerently (more closely) as a function of c
b
. We had that
P [error]
2−ary
= Q
_
_
¸
d
2
0,1
2N
0
_
_
Now, the spacing between symbols has reduced by a factor of
√
2/2 compared to bipolar PAM, to d
0,1
=
√
2c
b
.
So
P [error]
2−Co−FSK
= Q
_
_
c
b
N
0
_
For the same probability of bit error, binary FSK is about 1.5 dB better than OOK (requires 1.5 dB less energy
per bit), but 1.5 dB worse than bipolar PAM (requires 1.5 dB more energy per bit).
27.8 Probability of Error for Noncoherent Binary FSK
The energy detector shown in Figure 46 uses the energy in each frequency and selects the frequency with
maximum energy.
This energy is denoted r
2
m
in Figure 46 for frequency m and is
r
2
m
= r
2
mc
+r
2
ms
This energy measure is a statistic which measures how much energy was in the signal at frequency f
m
. The
‘envelope’ is a term used for the square root of the energy, so r
m
is termed the envelope.
Question: What will r
2
m
equal when the noise is very small?
As it turns out, given the noncoherent receiver and r
mc
and r
ms
, the envelope r
m
is an optimum (suﬃcient)
statistic to use to decide between s
1
. . . s
M
.
What do they do to prove this in Proakis & Salehi? They prove it for binary noncoherent FSK. It takes
quite a bit of 5510 to do this proof.
1. Deﬁne the received vector r as a 4 length vector of the correlation of r(t) with the sin and cos at each
frequency f
1
, f
2
.
2. They formulate the prior probabilities f
rH
i
(r[H
i
). Note that this depends on θ
m
, which is assumed to be
uniform between 0 and 2π, and independent of the noise.
f
rH
i
(r[H
i
) =
_
2π
0
f
r,θmH
i
(r, φ[H
i
)dφ
=
_
2π
0
f
rθm,H
i
(r[φ, H
i
)f
θmH
i
(φ[H
i
)dφ
(49)
Note that f
rθm,H
0
(r[φ, H
0
) is a 2D Gaussian random vector with i.i.d. components.
ECE 5520 Fall 2009 94
3. They formulate the joint probabilities f
r∩H
0
(r ∩ H
0
) and f
r∩H
1
(r ∩ H
1
).
4. Where the joint probability f
r∩H
0
(r ∩H
0
) is greater than f
rH
1
(r[H
1
), the receiver decides H
0
. Otherwise,
it decides H
1
.
5. The decisions in this last step, after manipulation of the pdfs, are shown to reduce to this decision (given
that P [H
0
] = P [H
1
]):
_
r
2
1c
+r
2
1s
H
0
>
<
H
1
_
r
2
2c
+r
2
2s
The “envelope detector” can equally well be called the “energy detector”, and it often is.
The above proof is simply FYI, and is presented since it does not appear in the Rice book.
An exact expression for the probability of error can be derived, as well. The proof is in Proakis & Salehi,
Section 7.6.9, page 430, which is posted on WebCT. The expression for probability of error in binary noncoherent
FSK is given by,
P [error]
2−NC−FSK
=
1
2
exp
_
−
c
b
2N
0
_
(50)
The expressions for probability of error in binary FSK (both coherent and noncoherent) are important, and
you should make note of them. You will use them to be able to design communication systems that use FSK.
Lecture 17
Today: (1) FSK Error Probabilities (2) OFDM
27.9 FSK Error Probabilities, Part 2
27.9.1 Mary NonCoherent FSK
For Mary noncoherent FSK, the derivation in the Proakis & Salehi book, section 7.6.9 shows that
P [symbol error] =
M−1
n=1
(−1)
n+1
_
M −1
n
_
1
n + 1
e
−log
2
M
n
n+1
E
b
N
0
and
P [error]
M−nc−FSK
=
M/2
M −1
P [symbol error]
See Figure 47.
Proof Summary: Our noncoherent receiver ﬁnds the energy in each frequency. These energy values no longer
have a Gaussian distribution (due to the squaring of the amplitudes in the energy calculation). They instead are
either Ricean (for the transmitted frequency) or Rayleigh distributed (for the “other” M −1 frequencies). The
probability that the correct frequency is selected is the probability that the Ricean random variable is larger
than all of the other random variables measured at the other frequencies.
Example: Probability of Error for Noncoherent M = 2 case
Use the above expressions to ﬁnd the P [symbol error] and P [error] for binary noncoherent FSK.
Solution:
P [symbol error] = P [bit error] =
1
2
e
−
1
2
E
b
N
0
ECE 5520 Fall 2009 95
0 2 4 6 8 10 12
10
−6
10
−5
10
−4
10
−3
10
−2
10
−1
E
b
/N
0
Ratio, dB
P
r
o
b
a
b
i
l
i
t
y
o
f
E
r
r
o
r
M=2
M=4
M=8
M=16
M=32
Figure 47: Probability of bit error for noncoherent reception of Mary FSK.
27.9.2 Summary
The key insight is that probability of error decreases for increasing M. As M → ∞, the probability of error
goes to zero, for the case when
E
b
N
0
> −1.6dB. Remember this for later  it is related to the ShannonHartley
theorem on the limit of reliable communication.
27.10 Bandwidth of FSK
Carson’s rule is used to calculate the bandwidth of FM signals. For Mary FSK, it tells us that the approximate
bandwidth is,
B
T
= (M −1)∆f + 2B
where B is the onesided bandwidth of the digital baseband signal. For the nulltonull bandwidth of raised
cosine pulse shaping, B = (1 +α)/T
s
. Note for square wave pulses, B = 1/T
s
.
28 Frequency Multiplexing
Frequency multiplexing is the division of the total bandwidth (call it B
T
) into many smaller frequency bands,
and sending a lower bit rate on each one.
Speciﬁcally, divide your B
T
bandwidth into K subchannels, you now have B
T
/K bandwidth on each sub
channel. Note that in the multiplexed system, each subchannel has a symbol period of T
s
K, longer by a factor
of K so that the bandwidth of that subchannel has narrower bandwidth. In HW 7, you show that the total
bit rate is the same in the multiplexed system, so you don’t lose anything by this division. Furthermore, your
transmission energy is the same. The energy would be divided by K in each subchannel, so the sum of the
energy is constant.
For each band, we might send a PSK or PAM or QAM signal on that band. Our choice is arbitrary.
28.1 Frequency Selective Fading
Frequency selectivity is primarily a problem in wireless channels. But, frequency dependent gains are also
experienced in DSL, because phone lines were not designed for higher frequency signals. For example, consider
Figure 48, which shows an example frequency selective fading pattern, 10 log
10
[H(f)[
2
, where H(f) is an
ECE 5520 Fall 2009 96
B K
T
/
Severelyfaded
channel
Figure 48: An example frequency selective fading pattern. The whole bandwidth is divided into K subchannels,
and each subchannel’s channel ﬁlter is mostly constant.
example wireless channel. This H(f) might be experienced in an average outdoor channel. (The f is relative,
i.e., take the actual frequency
˜
f = f +f
c
for some center frequency f
c
.)
• You can see that some channels experience severe attenuation, while most experience little attenuation.
Some channels experience gain.
• For stationary transmitter and receiver and stationary f, the channel stays constant over time. If you put
down the transmitter and receiver and have them operate at a particular f
′
, then the loss 10 log
10
[H(f
′
)[
2
will not change throughout the communication. If the channel loss is too great, than the error rate will
be too high.
Problems:
1. Wideband: The bandwidth is used for one channel. The H(f) acts as a ﬁlter, which introduces ISI.
2. Narrowband: Part of the bandwidth is used for one narrow channel, at a lower bit rate. The power is
increased by a fade margin which makes the
E
b
N
0
high enough for low BER demodulation except in the
most extreme fading situations.
When designing for frequency selective fading, designers may use a wideband modulation method which is
robust to frequency selective fading. For example,
• Frequency multiplexing methods (including OFDM),
• Directsequence spread spectrum (DSSS), or
• Frequencyhopping spread spectrum (FHSS).
We’re going to mention frequency multiplexing.
28.2 Beneﬁts of Frequency Multiplexing
Now, bit errors are not an ‘all or nothing’ game. In frequency multiplexing, there are K parallel bitstreams,
each of rate R/K, where R is the total bit rate. As a ﬁrst order approximation, a subchannel either experiences
a high SNR and makes no errors; or is in a severe fade, has a very low SNR, and experiences a BER of 0.5 (the
ECE 5520 Fall 2009 97
worst bit error rate!). If β is the probability that a subchannel experiences a severe fade, the overall probability
of error will be 0.5β.
Compare this to a single narrowband channel, which has probability β that it will have a 0.5 probability of
error. Similarly, a wideband system with very high ISI might be completely unable to demodulate the received
signal.
Frequency multiplexing is typically combined with channel coding designed to correct a small percentage of
bit errors.
28.3 OFDM as an extension of FSK
This is section 5.5 in the Rice book.
In FSK, we use a single basis function at each of diﬀerent frequencies. In QAM, we use two basis functions
at the same frequency. OFDM is the combination:
φ
0,I
(t) =
_ _
2/T
s
cos(2πf
c
t), 0 ≤ t ≤ T
s
0, o.w.
φ
0,Q
(t) =
_ _
2/T
s
sin(2πf
c
t), 0 ≤ t ≤ T
s
0, o.w.
φ
1,I
(t) =
_ _
2/T
s
cos(2πf
c
t + 2π∆ft), 0 ≤ t ≤ T
s
0, o.w.
φ
1,Q
(t) =
_ _
2/T
s
sin(2πf
c
t + 2π∆ft), 0 ≤ t ≤ T
s
0, o.w.
.
.
.
φ
M−1,I
(t) =
_ _
2/T
s
cos(2πf
c
t + 2π(M −1)∆ft), 0 ≤ t ≤ T
s
0, o.w.
φ
M−1,Q
(t) =
_ _
2/T
s
sin(2πf
c
t + 2π(M −1)∆ft), 0 ≤ t ≤ T
s
0, o.w.
where ∆f =
1
2Ts
. These are all orthogonal functions! We can transmit much more information than possible in
Mary FSK. (Note we have 2M basis functions here!)
The signal on subchannel k might be represented as:
x
k
(t) =
_
2/T
s
[a
k,I
(t) cos(2πf
k
t) +a
k,Q
(t) sin(2πf
k
t)]
The complex baseband signal of the sum of all K signals might then be represented as
x
l
(t) =
_
2/T
s
R
_
K
k=1
(a
k,I
(t) +ja
k,Q
(t))e
j2πk∆ft
_
x
l
(t) =
_
2/T
s
R
_
K
k=1
A
k
(t)e
j2πk∆ft
_
(51)
where A
k
(t) = a
k,I
(t) +ja
k,Q
(t). Does this look like an inverse discrete fourier transform? If yes, than you can
see why it is possible to use an IFFT and FFT to generate the complex baseband signal.
1. FFT implementation: There is a particular implementation of the transmitter and receiver that use
FFT/IFFT operations. This avoids having K independent transmitter chains and receiver chains. The
ECE 5520 Fall 2009 98
FFT implementation (and the speed and ease of implementation of the FFT in hardware) is why OFDM
is popular.
Since the K carriers are orthogonal, the signal is like Kary FSK. But, rather than transmitting on one of
the K carriers at a given time (like FSK) we transmit information in parallel on all K channels simultaneously.
An example state space diagram for K = 3 and PAM on each channel is shown in Figure 49.
OFDM,3subchannels
of4aryPAM
Figure 49: Signal space diagram for K = 3 subchannel OFDM with 4PAM on each channel.
Example: 802.11a
IEEE 802.11a uses OFDM with 52 subcarriers. Four of the subcarriers are reserved for pilot tones, so eﬀectively
48 subcarriers are used for data. Each data subcarrier can be modulated in diﬀerent ways. One example is to
use 16 square QAM on each subcarrier (which is 4 bits per symbol per subcarrier). The symbol rate in 802.11a
is 250k/sec. Thus the bit rate is
250 10
3
OFDM symbols
sec
48
subcarriers
OFDM symbol
4
coded bits
subcarrier
= 48
Mbits
sec
Lecture 18
Today: Comparison of Modulation Methods
29 Comparison of Modulation Methods
This section ﬁrst presents some miscellaneous information which is important to real digital communications
systems but doesn’t ﬁt nicely into other lectures.
29.1 Diﬀerential Encoding for BPSK
Def ’n: Coherent Reception
The reception of a signal when its carrier phase is explicitly determined and used for demodulation.
ECE 5520 Fall 2009 99
For coherent reception of PSK, will always need some kind of phase synchronization in BPSK. Typically,
this means transmitting a training sequence.
For noncoherent reception of PSK, we use diﬀerential encoding (at the transmitter) and decoding (at the
receiver).
29.1.1 DPSK Transmitter
Now, consider the bit sequence ¦b
n
¦, where b
n
is the nth bit that we want to send. The sequence b
n
is a sequence
of 0’s and 1’s. How do we decide which phase to send? Prior to this, we’ve said, send a
0
if b
n
= 0, and send a
1
if b
n
= 1.
Instead of setting k for a
k
only as a function of b
n
, in diﬀerential encoding, we also include k
n−1
. Now,
k
n
=
_
k
n−1
, b
n
= 0
1 −k
n−1
, b
n
= 1
Note that 1 − k
n−1
is the complement or negation of k
n−1
– if k
n−1
= 1 then 1 − k
n−1
= 0; if k
n−1
= 0 then
1 − k
n−1
= 1. Basically, for diﬀerential BPSK, a switch in the angle of the signal space vector from 0
o
to 180
o
or vice versa indicates a bit 1; while staying at the same angle indicates a bit 0.
Note that we have to just agree on the “zero” phase. Typically k
0
= 0.
Example: Diﬀerential encoding
Let b = [1, 0, 1, 0, 1, 1, 1, 0, 0]. Assume b
0
= 0. What symbols k = [k
0
, . . . , k
8
]
T
will be sent? Solution:
k = [1, 1, 0, 0, 1, 0, 1, 1, 1]
T
These values of k
n
correspond to a symbol stream with phases:
∠a
k
= [π, π, 0, 0, π, 0, π, π, π]
T
29.1.2 DPSK Receiver
Now, at the receiver, we ﬁnd b
n
by comparing the phase of x
n
to the phase of x
n−1
. What our receiver does, is
to measure the statistic
ℜ¦x
n
x
∗
n−1
¦ = [x
n
[[x
n−1
[ cos(∠x
n
−∠x
n−1
)
as the statistic – if it less than zero, decide b
n
= 1, and if it is greater than zero, decide b
n
= 0.
Example: Diﬀerential decoding
1. Assuming no phase shift in the above encoding example, show that the receiver will decode the original
bitstream with diﬀerential decoding. Solution: Assuming φ
i
0
= 0,
ˆ
b
n
= [1, 0, 1, 0, 1, 1, 1, 0, 0]
T
.
ECE 5520 Fall 2009 100
2. Now, assume that all bits are shifted π radians and we receive
∠x
′
= [0, 0, π, π, 0, π, 0, 0, 0].
What will be decoded at the receiver? Solution:
ˆ
b
n
= [0, 0, 1, 0, 1, 1, 1, 0, 0].
Rotating all symbols by π radians only causes one bit error.
29.1.3 Probability of Bit Error for DPSK
The probability of bit error in DPSK is slightly worse than that for BPSK:
P [error] =
1
2
exp
_
−
c
b
N
0
_
For a constant probability of error, DPSK requires about 1 dB more
E
b
N
0
than BPSK, which has probability
of bit error Q
__
2E
b
N
0
_
. Both are plotted in Figure 50.
0 2 4 6 8 10 12 14
10
−6
10
−4
10
−2
10
0
E
b
/N
0
, dB
P
r
o
b
a
b
i
l
i
t
y
o
f
B
i
t
E
r
r
o
r
BPSK
DBPSK
Figure 50: Comparison of probability of bit error for BPSK and Diﬀerential BPSK.
29.2 Points for Comparison
• Linear ampliﬁers (transmitter complexity)
• Receiver complexity
• Fidelity (P [error]) vs.
E
b
N
0
• Bandwidth eﬃciency
ECE 5520 Fall 2009 101
η α = 0 α = 0.5 α = 1
BPSK 1.0 0.67 0.5
QPSK 2.0 1.33 1.5
16QAM 4.0 2.67 2.0
64QAM 6.0 4.0 3.0
Table 2: Bandwidth eﬃciency of PSK and QAM modulation methods using raised cosine ﬁltering as a function
of α.
29.3 Bandwidth Eﬃciency
We’ve talked about measuring data rate in bits per second. We’ve also talked about Hertz, i.e., the quantity of
spectrum our signal will use. Typically, we can scale a system, to increase the bit rate by decreasing the symbol
period, and correspondingly increase the bandwidth. This relationship is typically linearly proportional.
Def ’n: Bandwidth eﬃciency
The bandwidth eﬃciency, typically denoted η, is the ratio of bits per second to bandwidth:
η = R
b
/B
T
Bandwidth eﬃciency depends on the deﬁnition of “bandwidth”. Since it is usually used for comparative pur
poses, we just make sure we use the same deﬁnition of bandwidth throughout a comparison.
The key ﬁgure of merit: bits per second / Hertz, i.e., bps/Hz.
29.3.1 PSK, PAM and QAM
In these three modulation methods, the bandwidth is largely determined by the pulse shape. For root raised
cosine ﬁltering, the nullnull bandwidth is 1 +α times the bandwidth of the case when we use pure sinc pulses.
The transmission bandwidth (for a bandpass signal) is
B
T
=
1 +α
T
s
Since T
s
is seconds per symbol, we divide by log
2
M bits per symbol to get T
b
= T
s
/ log
2
M seconds per bit, or
B
T
=
(1 +α)R
b
log
2
M
where R
b
= 1/T
b
is the bit rate.
Bandwidth eﬃciency is then
η = R
b
/B
T
=
log
2
M
1 +α
See Table 2 for some numerical examples.
29.3.2 FSK
We’ve said that the bandwidth of FSK is,
B
T
= (M −1)∆f + 2B
ECE 5520 Fall 2009 102
where B is the onesided bandwidth of the digital baseband signal. For the nulltonull bandwidth of raised
cosine pulse shaping, 2B = (1 +α)/T
s
. So,
B
T
= (M −1)∆f + (1 +α)/T
s
=
R
b
log
2
M
¦(M −1)∆fT
s
+ (1 +α)¦
since R
b
= 1/T
s
for
η = R
b
/B
T
=
log
2
M
(M −1)∆fT
s
+ (1 +α)
If ∆f = 1/T
s
(required for noncoherent reception),
η =
log
2
M
M +α
29.4 Bandwidth Eﬃciency vs.
E
b
N
0
For each modulation format, we have quantities of interest:
• Bandwidth eﬃciency, and
• Energy per bit (
E
b
N
0
) requirements to achieve a given probability of error.
Example: Bandwidth eﬃciency vs.
E
b
N
0
for M = 8 PSK
What is the required
E
b
N
0
for 8PSK to achieve a probability of bit error of 10
−6
? What is the bandwidth
eﬃciency of 8PSK when using 50% excess bandwidth?
Solution: Given in Rice Figure 6.3.5 to be about 14 dB, and 2.
We can plot these (required
E
b
N
0
, bandwidth eﬃciency) pairs. See Rice Figure 6.3.6.
29.5 Fidelity (P [error]) vs.
E
b
N
0
Main modulations which we have evaluated probability of error vs.
E
b
N
0
:
1. Mary PAM, including Binary PAM or BPSK, OOK, DPSK.
2. Mary PSK, including QPSK.
3. Square QAM
4. Nonsquare QAM constellations
5. FSK, Mary FSK
In this part of the lecture we will break up into groups and derive: (1) the probability of error and (2)
probability of symbol error formulas for these types of modulations.
See also Rice pages 325331.
ECE 5520 Fall 2009 103
Name P [symbol error] P [error]
BPSK = Q
__
2E
b
N
0
_
same
OOK = Q
__
E
b
N
0
_
same
DPSK =
1
2
exp
_
−
E
b
N
0
_
same
MPAM =
2(M−1)
M
Q
_
_
6 log
2
M
M
2
−1
E
b
N
0
_
≈
1
log
2
M
P [symbol error]
QPSK = Q
__
2E
b
N
0
_
MPSK ≤ 2Q
__
2(log
2
M) sin
2
(π/M)
E
b
N
0
_
≈
1
log
2
M
P [symbol error]
Square MQAM ≈
4
log
2
M
(
√
M−1)
√
M
Q
_
_
3 log
2
M
M−1
E
b
N
0
_
2noncoFSK =
1
2
exp
_
−
E
b
2N
0
_
same
MnoncoFSK =
M−1
n=1
(
M−1
n
)
(−1)
n+1
n+1
exp
_
−
nlog
2
M
n+1
E
b
N
0
_
=
M/2
M−1
P [symbol error]
2coFSK = Q
__
E
b
N
0
_
same
McoFSK ≤ (M −1)Q
__
log
2
M
E
b
N
0
_
=
M/2
M−1
P [symbol error]
Table 3: Summary of probability of bit and symbol error formulas for several modulations.
29.6 Transmitter Complexity
What makes a transmitter more complicated?
• Need for a linear power ampliﬁer
• Higher clock speed
• Higher transmit powers
• Directional antennas
29.6.1 Linear / Nonlinear Ampliﬁers
An ampliﬁer uses DC power to take an input signal and increase its amplitude at the output. If P
DC
is the
input power (e.g., from battery) to the ampliﬁer, and P
out
is the output signal power, the power eﬃciency is
rated as η
P
= P
out
/P
DC
.
Ampliﬁers are separated into ‘classes’ which describe their conﬁguration (circuits), and as a result of their
conﬁguration, their linearity and power eﬃciency.
• Class A: linear ampliﬁers with maximum power eﬃciency of 50%. Output signal is a scaled up version of
the input signal. Power is dissipated at all times.
• Class B: linear ampliﬁers turn on for half of a cycle (conduction angle of 180
o
) with maximum power
eﬃciency of 78.5%. Two in parallel are used to amplify a sinusoidal signal, one for the positive part and
one for the negative part.
• Class AB: Like class B, but each ampliﬁer stays on slightly longer to reduce the “dead zone” at zero.
That is, the conduction angle is slightly higher than 180
o
.
ECE 5520 Fall 2009 104
• Class C: A class C ampliﬁer is closer to a switch than an ampliﬁer. This generates high distortion, but
then is bandpass ﬁltered or tuned to the center frequency to force out spurious frequencies. Class C
nonlinear ampliﬁers have power eﬃciencies around 90%. Can only amplify signals with a nearly constant
envelope.
In order to double the power eﬃciency, batterypowered transmitters are often willing to use Class C
ampliﬁers. They can do this if their output signal has constant envelope. This means that, if you look at
the constellation diagram, you’ll see a circle. The signal must never go through the origin (envelope of zero) or
even near the origin.
29.7 Oﬀset QPSK
Qchannel
Ichannel
OffsetQPSK:
Qchannel
Ichannel
QPSKwithoutoffset:
(b) (a)
Figure 51: Constellation Diagram for (a) QPSK and (b) OQPSK.
For QPSK, we wrote the modulated signal as
s(t) =
√
2p(t) cos(ω
0
t +∠a(t))
where ∠a(t) is the angle of the symbol chosen to be sent at time t. It is in a discrete set,
∠a(t) ∈ ¦
π
4
,
3π
4
,
5π
4
,
7π
4
¦
and p(t) is the pulse shape. We could have also written s(t) as
s(t) =
√
2p(t) [a
0
(t) cos(ω
0
t) +a
1
(t) sin(ω
0
t)]
The problem is that when the phase changes 180 degrees, the signal s(t) will go through zero, which precludes
use of a class C ampliﬁer. See Figure 52(b) and Figure 51 (a) to see this graphically.
For oﬀset QPSK (OQPSK), we delay the quadrature T
s
/2 with respect to the inphase. Rewriting s(t)
s(t) =
√
2p(t)a
0
(t) cos(ω
0
t) +
√
2p(t −T
s
/2)a
1
(t −T
s
/2) sin(ω
0
(t −T
s
/2))
At the receiver, we just need to delay the sampling on the quadrature half of a sample period with respect to
the inphase signal. The new transmitted signal takes the same bandwidth and average power, and has the same
E
b
/N
0
vs. probability of bit error performance. However, the envelope [s(t)[ is largely constant. See Figure 52
for a comparison of QPSK and OQPSK.
ECE 5520 Fall 2009 105
(a)
0 1 2 3 4
−0.1
0
0.1
Time t
R
e
a
l
0 1 2 3 4
−0.1
0
0.1
Time t
I
m
a
g
(b)
−0.2 −0.1 0 0.1 0.2
−0.15
−0.1
−0.05
0
0.05
0.1
0.15
0.2
I
m
a
g
Real
(c)
0 1 2 3 4
−0.1
0
0.1
Time t
R
e
a
l
0 1 2 3 4
−0.1
0
0.1
Time t
I
m
a
g
(d)
−0.2 −0.1 0 0.1 0.2
−0.15
−0.1
−0.05
0
0.05
0.1
0.15
0.2
I
m
a
g
Real
Figure 52: Matlab simulation of (ab) QPSK and (cd) OQPSK, showing the (d) largely constant envelope of
OQPSK, compared to (b) that for QPSK.
29.8 Receiver Complexity
What makes a receiver more complicated?
• Synchronization (carrier, phase, timing)
• Multiple parallel receiver chains
Lecture 19
Today: (1) Link Budgets and System Design
30 Link Budgets and System Design
As a digital communication system designer, your mission (if you choose to accept it) is to achieve:
1. High data rate
2. High ﬁdelity (low bit error rate)
3. Low transmit power
ECE 5520 Fall 2009 106
4. Low bandwidth
5. Low transmitter/receiver complexity
But this is truly a mission impossible, because you can’t have everything at the same time. So the system design
depends on what the desired system really needs and what are the acceptable tradeoﬀs. Typically some subset
of requirements are given; for example, given the bandwidth limits, the received signal power and noise power,
and bit error rate limit, what data rate can be achieved? Using which modulation?
To answer this question, it is just a matter of setting some system variables (at the given limits) and
determining what that means about the values of other system variables. See Figure 53. For example, if I was
given the bandwidth limits and C/N
0
ratio, I’d be able to determine the probability of error for several diﬀerent
modulation types.
In this lecture, we discuss this procedure, and deﬁne each of the variables we see in Figure 53, and to what
they are related. This lecture is about system design, and it cuts across ECE classes; in particular circuits,
radio propagation, antennas, optics, and the material from this class. You are expected to apply what you have
learned in other classes (or learn the functional relationships that we describe here).
Figure 53: Relationships between important system design variables (rectangular boxes). Functional relation
ships between variables are given as circles – if the relationship is a single operator (e.g., division), the operator
is drawn in, otherwise it is left as an unnamed function f(). For example, C/N
0
is shown to have a divide by
relationship with C and N
0
. The eﬀect of the choice of modulation impacts several functional relationships,
e.g., the relationship between probability of bit error and
E
b
N
0
, which is drawn as a dotted line.
30.1 Link Budgets Given C/N
0
The received power is denoted C, it has units of Watts. What is C/N
0
? It is received power divided by noise
energy. It is an odd quantity, but it summarizes what we need to know about the signal and the noise for the
purposes of system design.
• We often describe both the received power at a receiver P
R
, but in the Rice book it is typically denoted
C.
ECE 5520 Fall 2009 107
• We know the probability of bit error is typically written as a function of
E
b
N
0
. The noise energy is N
0
. The
bit energy is c
b
. We can write c
b
= CT
b
, since energy is power times time. To separate the eﬀect of T
b
,
we often denote:
c
b
N
0
=
C
N
0
T
b
=
C/N
0
R
b
where R
b
= 1/T
b
is the bit rate. In other words, C/N
0
=
E
b
N
0
R
b
What are the units of C/N
0
? Answer:
Hz, 1/s.
• Note that people often report C/N
0
in dB Hz, which is
10 log
10
C
N
0
• Be careful of Bytes per sec vs bits per sec. Commonly, CS people use Bps (kBps or MBps) when describing
data rate. For example, if it takes 5 seconds to transfer a 1MB ﬁle, then software often reports that the
data rate is 1/5 = 0.2 MBps or 200 kBps. But the bit rate is 8/5 Mbps or 1.6 10
6
bps.
Given C/N
0
, we can now relate bit error rate, modulation, bit rate, and bandwidth.
Note: We typically use Q() and Q
−1
() to relate BER and
E
b
N
0
in each direction. While you have Matlab, this
is easy to calculate. If you can program it into your calculator, great. Otherwise, it’s really not a big deal to
pull it oﬀ of a chart or table. For your convenience, the following tables/plots of Q
−1
(x) will appear on Exam
2. I am not picky about getting lots of correct decimal places.
TABLE OF THE Q
−1
() FUNCTION:
Q
−1
_
1 10
−6
_
= 4.7534 Q
−1
_
1 10
−4
_
= 3.719 Q
−1
_
1 10
−2
_
= 2.3263
Q
−1
_
1.5 10
−6
_
= 4.6708 Q
−1
_
1.5 10
−4
_
= 3.6153 Q
−1
_
1.5 10
−2
_
= 2.1701
Q
−1
_
2 10
−6
_
= 4.6114 Q
−1
_
2 10
−4
_
= 3.5401 Q
−1
_
2 10
−2
_
= 2.0537
Q
−1
_
3 10
−6
_
= 4.5264 Q
−1
_
3 10
−4
_
= 3.4316 Q
−1
_
3 10
−2
_
= 1.8808
Q
−1
_
4 10
−6
_
= 4.4652 Q
−1
_
4 10
−4
_
= 3.3528 Q
−1
_
4 10
−2
_
= 1.7507
Q
−1
_
5 10
−6
_
= 4.4172 Q
−1
_
5 10
−4
_
= 3.2905 Q
−1
_
5 10
−2
_
= 1.6449
Q
−1
_
6 10
−6
_
= 4.3776 Q
−1
_
6 10
−4
_
= 3.2389 Q
−1
_
6 10
−2
_
= 1.5548
Q
−1
_
7 10
−6
_
= 4.3439 Q
−1
_
7 10
−4
_
= 3.1947 Q
−1
_
7 10
−2
_
= 1.4758
Q
−1
_
8 10
−6
_
= 4.3145 Q
−1
_
8 10
−4
_
= 3.1559 Q
−1
_
8 10
−2
_
= 1.4051
Q
−1
_
9 10
−6
_
= 4.2884 Q
−1
_
9 10
−4
_
= 3.1214 Q
−1
_
9 10
−2
_
= 1.3408
Q
−1
_
1 10
−5
_
= 4.2649 Q
−1
_
1 10
−3
_
= 3.0902 Q
−1
_
1 10
−1
_
= 1.2816
Q
−1
_
1.5 10
−5
_
= 4.1735 Q
−1
_
1.5 10
−3
_
= 2.9677 Q
−1
_
1.5 10
−1
_
= 1.0364
Q
−1
_
2 10
−5
_
= 4.1075 Q
−1
_
2 10
−3
_
= 2.8782 Q
−1
_
2 10
−1
_
= 0.84162
Q
−1
_
3 10
−5
_
= 4.0128 Q
−1
_
3 10
−3
_
= 2.7478 Q
−1
_
3 10
−1
_
= 0.5244
Q
−1
_
4 10
−5
_
= 3.9444 Q
−1
_
4 10
−3
_
= 2.6521 Q
−1
_
4 10
−1
_
= 0.25335
Q
−1
_
5 10
−5
_
= 3.8906 Q
−1
_
5 10
−3
_
= 2.5758 Q
−1
_
5 10
−1
_
= 0
Q
−1
_
6 10
−5
_
= 3.8461 Q
−1
_
6 10
−3
_
= 2.5121
Q
−1
_
7 10
−5
_
= 3.8082 Q
−1
_
7 10
−3
_
= 2.4573
Q
−1
_
8 10
−5
_
= 3.775 Q
−1
_
8 10
−3
_
= 2.4089
Q
−1
_
9 10
−5
_
= 3.7455 Q
−1
_
9 10
−3
_
= 2.3656
ECE 5520 Fall 2009 108
10
−7
10
−6
10
−5
10
−4
10
−3
10
−2
10
−1
10
0
0
0.5
1
1.5
2
2.5
3
3.5
4
4.5
5
Value, x
I
n
v
e
r
s
e
Q
f
u
n
c
t
i
o
n
,
Q
−
1
(
x
)
30.2 Power and Energy Limited Channels
Assume the C/N
0
, the maximum bandwidth, and the maximum BER are all given. Sometimes power is the
limiting factor in determining the maximum achievable bit rate. Such links (or channels) are called power limited
channels. Sometimes bandwidth is the limiting factor in determining the maximum achievable bit rate. In this
case, the link (or channel) is called a bandwidth limited channel. You just need to try to solve the problem and
see which one limits your system.
Here is a stepbystep version of what you might need do in this case:
Method A: Start with powerlimited assumption:
1. Use the probability of error constraint to determine the
E
b
N
0
constraint, given the appropriate probability
of error formula for the modulation.
2. Given the C/N
0
constraint and the
E
b
N
0
constraint, ﬁnd the maximum bit rate. Note that R
b
= 1/T
b
=
C/N
0
E
b
N
0
,
but be sure to express both in linear units.
3. Given a maximum bit rate, calculate the maximum symbol rate R
s
=
R
b
log
2
M
and then compute the required
bandwidth using the appropriate bandwidth formula.
4. Compare the bandwidth at maximum R
s
to the bandwidth constraint: If BW at R
s
is too high, then
the system is bandwidth limited; reduce your bit rate to conform to the BW constraint. Otherwise, your
system is power limited, and your R
b
is achievable.
Method B: Start with a bandwidthlimited assumption:
ECE 5520 Fall 2009 109
1. Use the bandwidth constraint and the appropriate bandwidth formula to ﬁnd the maximum symbol rate
R
s
and then the maximum bit rate R
b
.
2. Find the
E
b
N
0
at the given bit rate by computing
E
b
N
0
=
C/N
0
R
b
. (Again, make sure that everything is in linear
units.)
3. Find the probability of error at that
E
b
N
0
, using the appropriate probability of error formula.
4. If the computed P [error] is greater than the BER constraint, then your system is power limited. Use the
previous method to ﬁnd the maximum bit rate. Otherwise, your system is bandwidthlimited, and you
have found the correct maximum bit rate.
Example: Rice 6.33
Consider a bandpass communications link with a bandwidth of 1.5 MHz and with an available C/N
0
= 82
dB Hz. The maximum bit error rate is 10
−6
.
1. If the modulation is 16PSK using the SRRC pulse shape with α = 0.5, what is the maximum achievable
bit rate on the link? Is this a power limited or bandwidth limited channel?
2. If the modulation is square 16QAM using the SRRC pulse shape with α = 0.5, what is the maximum
achievable bit rate on this link? Is this a power limited or bandwidth limited channel?
Solution:
1. Try Method A. For M = 16 PSK, we can ﬁnd
E
b
N
0
for the maximum BER:
10
−6
= P [error] =
2
log
2
M
Q
_
_
2(log
2
M) sin
2
(π/M)
c
b
N
0
_
10
−6
=
2
4
Q
_
_
2(4) sin
2
(π/16)
c
b
N
0
_
c
b
N
0
=
1
8 sin
2
(π/16)
_
Q
−1
_
2 10
−6
_¸
2
c
b
N
0
= 69.84 (52)
Converting C/N
0
to linear, C/N
0
= 10
82/10
= 1.585 10
8
. Solving for R
b
,
R
b
=
C/N
0
E
b
N
0
=
1.585 10
8
69.84
= 2.2710
6
= 2.27 Mbits/s
and thus R
s
= R
b
/ log
2
M = 2.2710
6
/4 = 5.6710
5
Msymbols/s. The required bandwidth for this
system is
B
T
=
(1 +α)R
b
log
2
M
= 1.5(2.2710
6
)/4 = 851 kHz
This is clearly lower than the maximum bandwidth of 1.5 MHz. So, the system is power limited, and can
operate with bit rate 2.27 Mbits/s. (If B
T
had come out > 1.5 MHz, we would have needed to reduce R
b
to meet the bandwidth limit.)
ECE 5520 Fall 2009 110
2. Try Method A. For M = 16 (square) QAM, we can ﬁnd
E
b
N
0
for the maximum BER:
10
−6
= P [error] =
4
log
2
M
(
√
M −1)
√
M
Q
_
_
3 log
2
M
M −1
c
b
N
0
_
10
−6
=
4
4
(4 −1)
4
Q
_
_
¸
3(4)
15
c
b
N
0
_
_
c
b
N
0
=
15
12
_
Q
−1
_
(4/3) 10
−6
_¸
2
c
b
N
0
= 27.55 (53)
Solving for R
b
,
R
b
=
C/N
0
E
b
N
0
=
1.585 10
8
27.55
= 5.7510
6
= 5.75 Mbits/s
The required bandwidth for this bit rate is:
B
T
=
(1 +α)R
b
log
2
M
= 1.5(5.7510
6
)/4 = 2.16 MHz
This is greater than the maximum bandwidth of 1.5 MHz, so we must reduce the bit rate to
R
b
=
B
T
log
2
M
1 +α
= 1.5 MHz
4
1.5
= 4 MHz
In summary, we have a bandwidthlimited system with a bit rate of 4 MHz.
30.3 Computing Received Power
We need to design our system for the power which we will receive. How can we calculate how much power will
be received? We can do this for both wireless and wired channels. Wireless propagation models are required;
this section of the class provides two examples, but there are others.
30.3.1 Free Space
‘Free space’ is the idealization in which nothing exists except for the transmitter and receiver, and can really only
be used for deep space communications. But, this formula serves as a starting point for other radio propagation
formulas. In free space, the received power is calculated from the Friis formula,
C = P
R
=
P
T
G
T
G
R
λ
2
(4πR)
2
where
• G
T
and G
R
are the antenna gains at the transmitter and receiver, respectively.
• λ is the wavelength at signal frequency. For narrowband signals, the wavelength is nearly constant across
the bandwidth, so we just use the center frequency f
c
. Note that λ = c/f
c
where c = 3 10
8
meters per
second is the speed of light.
ECE 5520 Fall 2009 111
• P
T
is the transmit power.
Here, everything is in linear terms. Typically people use decibels to express these numbers, and we will write
[P
R
]
dBm
or [G
T
]
dB
to denote that they are given by:
[P
R
]
dBm
= 10 log
10
P
R
[G
T
]
dB
= 10 log
10
G
T
(54)
In lecture 2, we showed that the Friis formula, given in dB, is
[C]
dBm
= [G
T
]
dB
+ [G
R
]
dB
+ [P
T
]
dBm
+ 20 log
10
λ
4π
−20 log
10
R
This says the received power, C, is linearly proportional to the log of the distance R. The Rice book writes the
Friis formula as
[C]
dBm
= [G
T
]
dB
+ [G
R
]
dB
+ [P
T
]
dBm
+ [L
p
]
dB
where
[L
p
]
dB
= +20 log
10
λ
4π
−20 log
10
R
where [L
p
]
dB
is called the “channel loss”. (This name that Rice chose is unfortunate because if it was positive,
it would be a “gain”, but it is typically very negative. My opinion is, it should be called “channel gain” and
have a negative gain value.)
There are also typically other losses in a transmitter / receiver system; losses in a cable, other imperfections,
etc. Rice lumps these in as the term L and writes:
[C]
dBm
= [G
T
]
dB
+ [G
R
]
dB
+ [P
T
]
dBm
+ [L
p
]
dB
+ [L]
dB
30.3.2 Nonfreespace Channels
We don’t study radio propagation path loss formulas in this course. But, a very short summary is that radio
propagation on Earth is diﬀerent than the Friis formula suggests. Lots of other formulas exist that approximate
the received power as a function of distance, antenna heights, type of environment, etc.
For example, whenever path loss is linear with the log of distance,
[L
p
]
dB
= L
0
−10nlog
10
R.
for some constants n and L
0
. Eﬀectively, because of shadowing caused by buildings, trees, etc., the average loss
may increase more quickly than 1/R
2
, instead, it may be more like 1/R
n
.
30.3.3 Wired Channels
Typically wired channels are lossy as well, but the loss is modeled as linear in the length of the cable. For
example,
[C]
dBm
= [P
T
]
dBm
−R[L
1m
]
dB
where P
T
is the transmit power and R is the cable length in meters, and L
1m
is the loss per meter.
ECE 5520 Fall 2009 112
30.4 Computing Noise Energy
The noise energy N
0
can be calculated as:
N
0
= kT
eq
where k = 1.38 10
−23
J/K is Boltzmann’s constant and T
eq
is called the equivalent noise temperature in
Kelvin. This is a topic covered in another course, and so you are not responsible for that material here. But
in short, the equivalent temperature is a function of the receiver design. T
eq
is always going to be higher than
the temperature of your receiver. Basically, all receiver circuits add noise to the received signal. With proper
design, T
eq
can be kept low.
30.5 Examples
Example: Rice 6.36
Consider a “pointtopoint” microwave link. (Such links were the key component in the telephone company’s
long distance network before ﬁber optic cables were installed.) Both antenna gains are 20 dB and the transmit
antenna power is 10 W. The modulation is 51.84 Mbits/sec 256 square QAM with a carrier frequency of 4 GHz.
Atmospheric losses are 2 dB and other incidental losses are 2 dB. A pigeon in the lineofsight path causes an
additional 2 dB loss. The receiver has an equivalent noise temperature of 400 K and an im plementation loss
of 1 dB. How far away can the two towers be if the bit error rate is not to exceed 10
−8
? Include the pigeon.
Neal’s hint: Use the dB version of Friis formula and subtract these mentioned dB losses: atmospheric losses,
incidental losses, implementation loss, and the pigeon.
Solution: Starting with the modulation, M = 256 square QAM (log
2
M = 8,
√
M = 16), to achieve
P [error] = 10
−8
,
10
−8
=
4
log
2
m
(
√
M −1)
√
M
Q
_
_
3 log
2
M
M −1
c
b
N
0
_
10
−8
=
4
8
15
16
Q
_
_
¸
3(8)
255
c
b
N
0
_
_
10
−8
=
15
32
Q
_
_
24
255
c
b
N
0
_
.
c
b
N
0
=
255
24
_
Q
−1
_
32
15
10
−8
__
2
= 319.0
The noise power N
0
= kT
eq
= 1.3810
−23
(J/K)400(K) = 5.5210
−21
J. So c
b
= 319.05.5210
−21
= 1.76
10
−18
J. Since c
b
= C/R
b
and the bit rate R
b
= 51.84 10
6
bits/sec, C = (51.84 10
6
)J(1.76 10
−18
)1/sec =
9.13 10
−11
W, or 100.4 dBW.
Switching to ﬁnding an expression for C, the wavelength is λ = 3 10
8
m/s/4 10
9
1/s = 0.075m, so:
[C]
dBW
= [G
T
]
dB
+ [G
R
]
dB
+ [P
T
]
dBW
+ 20 log
10
λ
4π
−20 log
10
R −2dB −2dB −2dB −1dB
= 20dB + 20dB + 10dBW + 20 log
10
0.075m
4π
−20 log
10
R −7dB
= −1.48dBW−20 log
10
R (55)
ECE 5520 Fall 2009 113
Plugging in [C]
dBm
= −100.4 dBW = −1.48dBW− 20 log
10
R and solving for R, we ﬁnd R = 88.3 km. Thus
microwave towers should be placed at most 88.3 km (about 55 miles) apart.
Lecture 20
Today: (1) Timing Synchronization Intro, (2) Interpolation Filters
31 Timing Synchronization
At the receiver, a transmitted signal arrives with an unknown delay τ. The received complex baseband signal
(for QAM, PAM, or QPSK) can be written (assuming carrier synchronization) as
r(t) =
k
m
a
m
(k)φ
m
(t −kT
s
−τ) (56)
where a
k
(m) is the mth transmitted symbol.
As a start, let’s compare receiver implementations for a (mostly) continuoustime and a (more) discretetime
receiver. Figure 54 has a timing synchronization loop which controls the sampler (the ADC).
Figure 54: Block diagram for a continuoustime receiver, including analog timing synchronization (from Rice
book, Figure 8.3.1).
The input signal is downconverted and then run through matched ﬁlters, which correlate the signal with
φ
n
(t −t
k
) for each n, and for some delay t
k
. For the correlation with n,
x
n
(k) = ¸r(t), φ
n
(t))
x
n
(k) =
k
m
a
m
(k)¸φ
n
(t −t
k
), φ
n
(t −kT
s
−τ)) (57)
Note that if t
k
= kT
s
+τ, then the correlation ¸φ
n
(t −t
k
), φ
n
(t −kT
s
−τ)) is highest and closest to 1. This
t
k
is the correct timing delay at each correlator for the kth symbol. But, these are generally unknown to the
receiver until timing synchronization is complete.
Figure 55 shows a receiver with an ADC immediately after the downconverter. Here, note the ADC has
nothing controlling it. Instead, after the matched ﬁlter, an interpolator corrects the sampling time problems
using discretetime processing. This interpolation is the subject of this section.
ECE 5520 Fall 2009 114
ctstime
bandpass
signal
x t ( )
2cos(2 + ) p f f t
c
2sin(2 + ) p f f t
c
PLL
p/2
ADC
Matched
Filter
Digital
Interpolator
Timing
Synch
Control
O
p
t
i
m
a
l
D
e
t
e
c
t
o
r
b
m
ADC
Matched
Filter
Digital
Interpolator
r nT ( )
sa
r nT ( )
sa
r kT ( )
sy
r kT ( )
sy
T
i
m
i
n
g
S
y
n
c
h
r
o
n
i
z
a
t
i
o
n
Figure 55: Block diagram for a digital receiver for QAM/PSK, including discretetime timing synchronization.
The industry trend is more and more towards digital implementations. A ‘software radio’ follows the idea
that as much of the radio is done in digital, after the signal has been sampled. The idea is to “bring the ADC
to the antenna” for purposes of reconﬁgurability, and reducing part counts and costs.
Another implementation is like Figure 55 but instead of the interpolator, the timing synch control block is
fed back to the ADC. But again, this requires a DAC and feedback to the analog part of the receiver, which is
not preferred. Also, because of the processing delay, this digital and analog feedback loop can be problematic.
First, we’ll talk about interpolation, and then, we’ll consider the control loop.
32 Interpolation
In this class, we will emphasize digital timing synchronization using an interpolation ﬁlter. For example, consider
Figure 56. In this ﬁgure, a BPSK receiver samples the matched ﬁlter output at a rate of twice per symbol,
unsynchronized with the symbol clock, resulting in samples r(nT).
4 6 8 10 12 14 16
−1
−0.5
0
0.5
1
Time t
V
a
l
u
e
Figure 56: Samples of the matched ﬁlter output (BPSK, RRC with α = 1) taken at twice the correct symbol
rate (vertical lines), but with a timing error. If downsampling (by 2) results in the symbol samples r
k
given by
red squares, then sampling sometimes reduces the magnitude of the desired signal.
Some example sampling clocks, compared to the actual symbol clock, are shown in Figure 57. These are
shown in degrees of severity of correction for the receiver. When we say ‘synchronized in rate’, we mean within
an integer multiple, since the sampling clock must operate at (at least) double the symbol rate.
ECE 5520 Fall 2009 115
Figure 57: (1) Sampling clock and (24) possible actual symbol clocks. Symbol clock may be (2) synchronized
in rate and phase, (3) synchronized in rate but not in phase, (4) synchronized neither in rate nor phase, with
sample clock.
In general, our receivers always deal with type (4) sampling clock error as drawn in Figure 57. That is, the
sampling clock has neither the same exact rate nor the same phase as the actual symbol clock.
Def ’n: Incommensurate
Two clocks with rates T and T
s
are incommensurate if the ratio T/T
s
is irrational. In contrast, two clocks are
commensurate if the ratio T/T
s
can be written as n/m where n, m are integers.
For example, T/T
s
= 1/2, the two clocks are commensurate and we sample exactly twice per symbol period.
As another example, if T/T
s
= 2/5, we sample exactly 2.5 times per symbol, and every 5 samples the delay
until the next correct symbol sample will repeat. Since clocks are generally incommensurate, we cannot count
on them ever repeating.
The situation shown in Figure 56 is case (3), where T/T
s
= 1/2 (the clocks are commensurate), but the
sampling clock does not have the correct phase (τ is not equal to an integer multiple of T).
32.1 Sampling Time Notation
In general, for both cases (3) and (4) in Figure 57, the correct sampling times should be kT
s
+τ, but no samples
were taken at those instants. Instead, kT
s
+ τ is always µ(k)T after the most recent sampling instant, where
µ(k) is called the fractional interval. We can write that
kT
s
+τ = [m(k) +µ(k)]T (58)
where m(k) is an integer, the highest integer such that nT < kT
s
+τ, and 0 ≤ µ(k) < 1. In other words,
m(k) =
_
kT
s
+τ
T
_
where ⌊x⌋ is the greatest integer less than function (the Matlab floor function). This means that µ(k) is given
by
µ(k) =
kT
s
+τ
T
−m(k)
Example: Calculation Example
Let T
s
/T = 3.1 and τ/T = 1.8. Calculate (m(k), µ(k)) for k = 1, 2, 3.
ECE 5520 Fall 2009 116
Solution:
m(1) = ⌊3.1 + 1.8⌋ = 4; µ(1) = 0.9
m(2) = ⌊2(3.1) + 1.8⌋ = 8; µ(1) = 0
m(3) = ⌊3(3.1) + 1.8⌋ = 11; µ(1) = 0.1
Thus your interpolation will be done: in between samples 4 and 5; at sample 8; and in between samples 11 and
12.
32.2 Seeing Interpolation as Filtering
Consider the output of the matched ﬁlter, r(t) as given in (57). The analog output of the matched ﬁlter could
be represented as a function of its samples r(nT),
r(t) =
n
r(nT)h
I
(t −nT) (59)
where
h
I
(t) =
sin(πt/T)
πt/T
.
Why is this so? What are the conditions necessary for this representation to be accurate?
If we wanted the signal at the correct sampling times, we could have it – we just need to calculate r(t) at
another set of times (not nT).
Call the correct symbol sampling times as kT
s
+τ for integer k, where T
s
is the actual symbol period used
by the transmitter. Plugging these times in for t in (59), we have that
r(kT
s
+τ) =
n
r(nT)h
I
(kT
s
+τ −nT)
Now, using the (m(k), µ(k)) notation, since kT
s
+τ = [m(k) +µ(k)]T,
r([m(k) +µ(k)]T) =
n
r(nT)h
I
([m(k) −n +µ(k)]T).
Reindexing with i = m(k) −n,
r([m(k) +µ(k)]T) =
i
r([m(k) −i]T)h
I
([i +µ(k)]T). (60)
This is a ﬁlter on samples of r(), where the ﬁlter coeﬃcients are dependent on µ(k).
Note: Good illustrations are given in M. Rice, Figure 8.4.12, and Figure 8.4.13.
32.3 Approximate Interpolation Filters
Clearly, (60) is a ﬁlter. The desired sample at [m(k) +µ(k)]T is calculated by adding the weighted contribution
from the signal at each sampling time. The problem is that in general, this requires an inﬁnite sum over i from
−∞ to ∞, because the sinc function has inﬁnite support.
Instead, we use polynomial approximations for h
I
(t):
• The easiest one we’re all familiar with is linear interpolation (a ﬁrstorder polynomial), in which we draw
a straight line between the two nearest sampled values to approximate the values of the continuous signal
between the samples. This isn’t so great of an approximation.
ECE 5520 Fall 2009 117
• A secondorder polynomial (i.e.a parabola) is actually a very good approximation. Given three points,
one can determine a parabola that ﬁts those three points exactly.
• However, the three point ﬁt does not result in a linearphase ﬁlter. (To see this, note in the time domain
that two samples are on one side of the interpolation point, and one on the other. This is temporal
asymmetry.) Instead, we can use four points to ﬁt a secondorder polynomial, and get a linearphase
ﬁlter.
• Finally, we could use a cubic interpolation ﬁlter. Four points determine a 3rd order polynomial, and result
in a linear ﬁlter.
To see results for diﬀerent order polynomial ﬁlters, see M. Rice Figure 8.24.
32.4 Implementations
Note: These polynomial ﬁlters are called Farrow ﬁlters and are named after Cecil W. Farrow, of AT&T Bell
Labs, who has the US patent (1989) for the “Continuously variable digital delay circuit”. These Farrow ﬁlters
started to be used in the 90’s and are now very common due to the dominance of digital processing in receivers.
From (60), we can see that the ﬁlter coeﬃcients are a function of µ(k), the fractional interval. Thus we
could rewrite (60) as
r([m(k) +µ(k)]T) =
i
r([m(k) −i]T)h
I
(i; µ(k)). (61)
That is, the ﬁlter is h
I
(i) but its values are a function of µ(k). The ﬁlter coeﬃcients are a polynomial function
of µ(k), that is, they are a weighted sum of µ(k)
0
, µ(k)
1
, µ(k)
2
, . . . , µ(k)
p
for a pth order polynomial ﬁlter.
Example: First order polynomial interpolation
For example, consider the linear interpolator.
r([m(k) +µ(k)]T) =
0
i=−1
r([m(k) −i]T)h
I
(i; µ(k))
What are the ﬁlter elements h
I
for a linear interpolation ﬁlter?
Solution:
r([m(k) +µ(k)]T) = µ(k)r([m(k) + 1]T) + [1 −µ(k)]r(m(k)T)
Here we have used h
I
(−1; µ(k)) = µ(k) and h
I
(0; µ(k)) = 1 −µ(k).
Essentially, given µ(k), we form a weighted average of the two nearest samples. As µ(k) → 1, we should
take the r([m(k) + 1]T) sample exactly. As µ(k) → 0, we should take the r(m(k)T) sample exactly.
32.4.1 Higher order polynomial interpolation ﬁlters
In general,
h
I
(i; µ(k)) =
p
l=0
b
l
(i)µ(k)
l
A full table of b
l
(i) is given in Table 8.1 of the M. Rice handout.
Note that the i indices seem backwards.
ECE 5520 Fall 2009 118
For the 2nd order Farrow ﬁlter, there is an extra degree of freedom – you can select parameter α to be in
the range 0 < α < 1. It has been shown by simulation that α = 0.43 is best, but people tend to use α = 0.5
because it is only slightly worse, and division by two is extremely easy in digital ﬁlters.
Example: 2nd order Farrow ﬁlter
What is the Farrow ﬁlter for α = 0.5 which interpolates exactly halfway between sample points?
Solution: From the problem statement, µ = 0.5. Since µ
2
= 0.25, µ = 0.5, µ
0
= 1, we can calculate that
h
I
(−2; 0.5) = αµ
2
−αµ = 0.125 −0.25 = −0.125
h
I
(−1; 0.5) = −αµ
2
+ (1 +α)µ = −0.125 + 0.75 = 0.625
h
I
(0; 0.5) = −αµ
2
+ (α −1)µ + 1 = −0.125 −0.25 + 1 = 0.625
h
I
(1; 0.5) = αµ
2
−αµ = 0.125 −0.25 = −0.125
(62)
Does this make sense? Do the weights add up to 1? Does it make sense to subtract a fraction of the two
more distant samples?
Example: Matlab implementation of Farrow Filter
My implementation is called ece5520 lec20.m and is posted on WebCT. Be careful, as my implementation
uses a loop, rather than vector processing.
Lecture 21
Today: (1) Timing Synchronization (2) PLLs
33 Final Project Overview
s n [ ]
MF N/2
h n p nT [ ]= ( )
cos( ) W
0
n 2
MF N/2
h n p nT [ ]= ( )
sin( ) W
0
n 2
Timing
Error
Detector
Interp.
Filter
Loop
Filter
VCC
v n ( )
m
e n ( )
x
I
( ) n
x
Q
( ) n
symbol
strobe
Interp.
Filter
r
I
( ) k
r
Q
( ) k
Preamble
Detector
Symbol
Decision
DATA
Bits
All
Bits
Figure 58: Flow chart of ﬁnal project; timing synchronization for QPSK.
ECE 5520 Fall 2009 119
For your ﬁnal project, you will implement the above receiver. It adds these blocks to your QPSK implemen
tation:
1. Interpolation Filter
2. Timing Error Detector (TED)
3. Loop Filter
4. Voltage Controlled Clock (VCC)
5. Preamble Detector
These notes address the motivation and overall design of each block.
Compared to last year’s class, you have the advantage of having access to the the Rice book, Section 8.4.4,
which contains much of the code for a BPSK implementation. When you put these together and implement
them for QPSK, you will need to understand how and why they work, in order to ﬁx any bugs.
33.1 Review of Interpolation Filters
Timing synchronization is necessary to know when to sample the matched ﬁlter output. We want to sample at
times [n+µ]T
s
, where n is the integer part and µ is the fractional oﬀset. Often, we leave out the T
s
and simply
talk about the index n or fractional delay µ.
Implementations may be continuous time, discrete time, or a mix. We focus on the discrete time solutions.
• Problem: After the matched ﬁlter, the samples may be at incorrect times, and in modern discretetime
implementations, there may be no analog feedback to the ADC.
• Solution: From samples taken at or above the Nyquist rate, you can interpolate between samples to ﬁnd
the desired sample.
However this solution leads to new problems:
• Problem: True interpolation requires signiﬁcant calculation – the sinc ﬁlter has inﬁnite impulse response.
• Solution: Approximate the sinc function with a 2nd or 3rd order polynomial interpolation ﬁlter, it works
nearly as well.
• Problem: How do you know when the correct symbol sampling time should be?
• Solution: Use a timing locked loop, analogous to a phased locked loop.
33.2 Timing Error Detection
We consider timing error detection blocks that operate after the matched ﬁlter and interpolator in the receiver,
which has output denoted x
I
(n). The job of the timing error detector is to produce a voltage error signal e(n)
which is proportional to the correct fractional oﬀset between the current sampling oﬀset (ˆ µ) and the correct
sampling oﬀset.
There are several possible timing error detection (TED) methods, as related in Rice 8.4.1. This is a good
source for detail on many possible methods. We will talk through two common discretetime implementations:
1. Earlylate timing error detector (ELTED)
ECE 5520 Fall 2009 120
2. Zerocrossing timing error detector (ZCTED)
We’ll use a continuoustime realization of a BPSK received matched ﬁlter output, x(t), to illustrate the
operation of timing error detectors. You can imagine that the interpolator output x
I
(n) is a sampled version of
x(t), hopefully with some samples taken at exactly the correct symbol sampling times.
Figure 59(a) shows an example eye diagram of the signal x(t) and Figure 59(b) shows its derivative ˙ x(t). In
both, time 0 corresponds to a correct symbol sampling time.
(a)
−1.5 −1 −0.5 0 0.5 1 1.5
−1
−0.5
0
0.5
1
Time from Symbol Sample Time
x
(
t
)
V
a
l
u
e
(b)
−1.5 −1 −0.5 0 0.5 1 1.5
−5
0
5
x 10
−4
Time from Symbol Sample Time
S
l
o
p
e
o
f
x
(
t
)
Figure 59: A sample eyediagram of a RRCshaped BPSK received signal (post matched ﬁlter), x(t). Here (a)
shows the signal x(t) and (b) shows its derivative ˙ x(t).
33.3 Earlylate timing error detector (ELTED)
The earlylate TED is a discretetime implementation of the continuoustime “earlylate gate” timing error
detector. In general, the error e(n) is determined by the slope and the sign of the sample x(n), at time n. In
particular, consider for BPSK the value of
˙ x(t)sgn ¦x(t)¦
Since the derivative ˙ x(t) is close to zero at the correct sampling time, and increases away from the correct
sampling time, we can use it to indicate how far from the sampling time we might be.
Figure 59 shows that the derivative is a good indication of timing error when the bit is changing from 1,
to 1, to 1. When the bit is constant (e.g., 1, to 1, to 1) the slope is close to zero. When the bit sequence is
ECE 5520 Fall 2009 121
from 1, to 1, to 1, the slope will be somewhat negative even when the sample is taken at the right time, and
when the bit is changing from 1, to 1, to 1, the slope will be somewhat positive even when the sample is taken
at the right time. These are imperfections in the ELTED which (hopefully) average out over many bits. In
particular, we can send alternating bits during the synchronization period in order to help the ELTED more
quickly converge.
The signum function sgn ¦x(t)¦ just corrects for the sign:
• A positive slope would mean we’re behind for a +1 symbol, but would mean that we’re ahead for a 1
symbol.
• In the opposite manner, a negative slope would mean we’re ahead for a +1 symbol, but would mean that
we’re behind for a 1 symbol.
For a discrete time system, you get only an approximation of the slope unless the sampling rate SPS (or
N, as it is called in the Rice book) is high. We’d estimate ˙ x(t) from the samples out of the interpolator and see
that
e(n −1) = ¦x
I
(n) −x
I
(n −2)¦ sgn ¦x
I
(n −1)¦
Here, we don’t divide by the time duration 2T
s
because it is just a scale factor, and we just need something
proportional to the timing error. This factor contributes to the gain K
p
of this part in the loop. This timing
error detector has a theoretical gain K
p
which shown in Figure 8.4.7 of the Rice book.
33.4 Zerocrossing timing error detector (ZCTED)
The zerocrossing timing error detector (ZCTED) is also described in detail in Section 8.4.1 of the Rice book.
In particular see Figure 8.4.8 of the Rice book. This error detector assumes that the sampling rate is SPS = 2.
It is described here for BPSK systems.
The error is,
e(k) = x((k −1/2)T
s
+ ˆ τ)[ˆ a(k −1) − ˆ a(k)] (63)
where the ˆ a(k) term is the estimate of the kth symbol (either +1 or −1),
ˆ a(k −1) = sgn ¦x((k −1)T
s
+ ˆ τ)¦ , (64)
ˆ a(k) = sgn ¦x(kT
s
+ ˆ τ)¦ . (65)
The error detector is nonzero at symbol sampling times. That is, if the symbol strobe is not activated at sample
n, then the error e(n) = 0.
In terms of n, because every second sample is a symbol sampling time, (63) can be rewritten as
e(n) = x
I
(n −1)[sgn¦x
I
(n −2)¦ −sgn¦x
I
(n)¦] (66)
This operation is drawn in Figure 60.
Basically, if the sign changes between n −2 and n, that indicates a symbol transition. If there was a symbol
transmission, the intermediate sample n−1 should be approximately zero, if it was taken at the correct sampling
time.
The theoretical gain in the ZCTED is twice that of the ELTED. The factor of two comes from the diﬀerence
of signs in (66), which will be 2 when the sign changes. In the project, you will need to look up K
p
in Figure
8.17 of Rice, which is a function of the excess bandwidth of the RRC ﬁlter used, and then multiply it by 2 to
ﬁnd the gain K
p
of your ZCTED.
ECE 5520 Fall 2009 122
x n
I
( )
TimingErrorDetector
z
1
z
1
x n
I
( 1) x n
I
( 2)
symbol
strobe
sgn sgn
e n ( )
Figure 60: Block diagram of the zerocrossing timing error detector (ZCTED), where sgn indicates the signum
function, 1 when positive and 1 when negative. The input strobe signal activates the sampler when it is the
correct symbol sampling time. When this happens, and the symbol has switched, the output is 2x
I
(n − 1),
which would be approximately zero if the sample clock is synchronized.
33.4.1 QPSK Timing Error Detection
When using QPSK, both the inphase and quadrature signals are used to calculate the error term. The error is
simply the sum of the two errors calculated for each signal. See (8.100) in the Rice book for the exact expression.
33.5 Voltage Controlled Clock (VCC)
We’ll use a decrementing VCC in the project. This is shown in Figure 61. (The choice of incrementing or
decrementing is arbitrary.) An example trace of the numerically controlled oscillator (NCO) which is the heart
of the VCC is shown in Figure 62.
VoltageControlledClock(VCC)
v n ( )
1/2
W n ( )
z
1
modulo1
NCO n ( 1)
underflow?
Strobehigh:change to
= ( 1)/ ( )
m
m NCO n W n
symbol
strobe
symbol
strobe
m
Figure 61: Block diagram of decrementing VCC with control voltage v(n).
NCO n ( 2)
...
NCO n ( )
NCO n ( 1)
m
W n ( )
Figure 62: Numericallycontrolled oscillator signal NCO(n).
ECE 5520 Fall 2009 123
The NCO starts at 1. At each sample n, the NCO decrements by 1/SPS + v(n). Once per symbol, (on
average) the NCO voltage drops below zero. When this happens, the strobe is set to high, meaning that a
sample must be taken.
When the symbol strobe is activated, the fractional interval is also recalculated. It is,
µ(n) =
NCO(n −1)
W(n)
(67)
When the strobe is not activated, µ(n) is kept the same, i.e., µ(n) = µ(n − 1). You will prove that (67) is the
correct form for µ(n) in the HW 9 assignment.
33.6 Phase Locked Loops
Carrier synchronization is done using a phaselocked loop (PLL). This section is included for reference. For
those of you familiar with PLLs, the operation of the timing synchronization loop may be best understood by
analogy to the continuous time carrier synchronization loop.
Phase
Detector
Loop
Filter
VCO
A f t t cos(2 + ( )) p q
c g( ( ) ) q t q( ) t
v( ) t
cos(2 + ( )) p q f t t
c
q ò ( )= ( ) t k v x dx
0
¥
t
Figure 63: General block diagram of a phaselocked loop for carrier synchronization [Rice].
Consider a generic PLL block diagram in Figure 63. The incoming signal is a carrier wave,
Acos(2πf
c
t +θ(t))
where θ(t) is the phase oﬀset. It is a function of time, t, because it may not be constant. For example, it may
include a frequency oﬀset, and then θ(t) = ∆ft +α. In this case, the phase oﬀset is a ramp.
The job of the the PLL is to estimate φ(t). We call this estimate
ˆ
θ(t). The error in the phase estimate is
θ
e
(t) = θ(t) −
ˆ
θ(t)
If the PLL is perfect, θ
e
(t) = 0.
33.6.1 Phase Detector
If the estimate of the phase is not perfect, the phase detector produces a voltage to correct the phase estimate.
The function g(θ
e
) may look something like the one shown in Figure 64.
Initially, let’s ignore the loop ﬁlter.
1. If the phase estimate is too low, that is, behind the carrier, then g(θ
e
) > 0 and the VCO increases the
phase of the VCO output.
2. If the phase estimate is too high, that is, ahead of the carrier, then g(θ
e
) < 0 and the VCO decreases the
phase of the VCO output.
3. If the phase estimate is exactly correct, then g(θ
e
) = 0 and VCO output phase remains constant.
This is the eﬀect of the ‘S’ curve function g(θ
e
) in Figure 64.
ECE 5520 Fall 2009 124
g( ) q
e
q
e
Figure 64: Typical phase detector inputoutput relationship. This curve is called an ‘S’ Curve. because the
shape of the curve looks like the letter ‘S’ [Rice].
33.6.2 Loop Filter
The loop ﬁlter is just a lowpass ﬁlter to reduce the noise. Note that in addition to the Acos(2πf
c
t + θ(t))
signal term we also have channel noise coming into the system. We represent the loop ﬁlter in the Laplace or
frequency domain as F(s) or F(f), respectively, for continuous time. We will be interested in discrete time
later, so we could also write the Ztransform response as F(z).
33.6.3 VCO
A voltagecontrolled oscillator (VCO) simply integrates the input and uses that integral as the phase of a cosine
wave. The input is essentially the gas pedal for a race car driving around a track  increase the gas (voltage),
and the engine (VCO) will increase the frequency of rotation around the track.
Note that if you just raise the voltage for a short period and then reduce it back (a constant plus rect
function input), you change the phase of the output, but leave the frequency the same.
33.6.4 Analysis
To analyze the loop in Figure 63, we end up making simpliﬁcations. In particular, we model the ‘S’ curve
(Figure 64) as a line. This linear approximation is generally ﬁne when the phase error is reasonably small, i.e.,
θ
e
= 0. If we do this, we can redraw Figure 63 as it is drawn in Figure 65.
F s ( )
Q( ) s
V s ( )
Q( ) s
k
0
s
k
p
phase
equivalent
perspective
Figure 65: Phaseequivalent and linear model for the continuoustime PLL, in the Laplace domain [Rice].
In the left half of Figure 65, we write just Θ(s) and
ˆ
Θ(s) even though the physical process, as we know, is
the cosine of 2πf
c
plus that phase. This phaseequivalent model of the PLL allows us to do analysis.
ECE 5520 Fall 2009 125
We can write that
Θ
e
(s) = Θ(s) −
ˆ
Θ(s)
= Θ(s) −Θ
e
(s)k
p
F(s)
k
0
s
To get the transfer function of the ﬁlter (when we consider the phase estimate to be the output), then some
manipulation shows you that
H
a
(s) =
ˆ
Θ(s)
Θ(s)
=
k
p
F(s)
k
0
s
1 +k
p
F(s)
k
0
s
=
k
0
k
p
F(s)
s +k
0
k
p
F(s)
You can use H
a
(s) to design the loop ﬁlter F(s) and gains k
p
and k
0
to achieve the desired PLL goals. Here
those goals include:
1. Desired bandwidth.
2. Desired response to particular phase error models.
The latter deserves more explanation. For example, phase error models might be: step input; or ramp input.
Designing a ﬁlter to respond to these, but not to AWGN noise (as much as possible), is a challenge. The Rice
book recommends a 2nd order proportionalplusintegrator loop ﬁlter,
F(s) = k
1
+
k
2
s
This ﬁlter has a proportional part (k
1
) which is just proportional to the input, which responds to the input
during phase oﬀset. The ﬁlter also has an integrator part (k
2
/s) which integrates the phase input, and in case
of a frequency oﬀset, will ramp up the phase at the proper rate.
Note: A accelerating frequency change wouldn’t be handled properly by the given ﬁlter. For example, if the
temperature of the transmitter or receiver varied quickly, the frequency of the crystal will also change quickly.
However, this is not typically fast enough to be a problem in most radios.
In this case there are four parameters to set, k
0
, k
1
, k
2
, and k
p
. The Rice Section C.2 details how to set these
parameters to achieve a desired bandwidth and a desired damping factor (to avoid ringing, but to converge
quickly). You will, in Homework 9, design the particular loop ﬁlter to use in your ﬁnal project.
33.6.5 DiscreteTime Implementations
You can alter F(s) to be a discrete time ﬁlter using Tustin’s approximation,
1
s
→
T
2
1 +z
−1
1 −z
−1
Because it is only a secondorder ﬁlter, its implementation is quite eﬃcient. See Figure C.2.1 (p 733, Rice book)
for diagrams of this discretetime PLL. Equations C.56 and C.57 (p 736) show how to calculate the constants
K
1
and K
2
, given the desired natural frequency θ
n
, the normalized bandwidth B
n
T, damping coeﬃcient ζ, and
other gain constants K
0
and K
p
.
ECE 5520 Fall 2009 126
Note that the bandwidth B
n
T is the most critical factor. It is typically much less than one, (e.g., 0.005).
This means that the loop ﬁlter eﬀectively averages over many samples when setting the input to the VCO.
Lecture 22
Today: (1) Exam 2 Review
• Exam 2 is Tue Apr 14.
• I will be out of town MonWed. The exam will be proctored. Please come to me with questions today or
tomorrow.
• Oﬃce hours: Today: 23:30. Friday 1112, 34.
34 Exam 2 Topics
Where to study:
• Lectures covered: 8 (detection theory), 9, 1119. Lectures 20,21 are not covered. No material from 17
will be directly tested, but of course you need to know some things from the start of the semester to be
able to perform well on the second part of this course.
• Homeworks covered: 48. Please review these homeworks and know the correct solutions.
• Rice book: Chapters 5 and 6.
What topics:
1. Detection theory
2. Signal space
3. Digital modulation methods:
• Binary, bipolar vs. unipolar
• Mary: PAM, QAM, PSK, FSK
4. Intersymbol interference, Nyquist ﬁltering, SRRC pulse shaping
5. Diﬀerential encoding and decoding for BPSK
6. Energy detection for FSK
7. Gray encoding
8. Probability of Error vs.
E
b
N
0
.
• Standard modulation method. Both know the formula and know how it can be derived.
• A nonstandard modulation method, given signal space diagram.
• Exact expression, union bound, nearestneighbor approximation.
ECE 5520 Fall 2009 127
9. Bandwidth assuming SRRC pulse shaping for PAM/QAM/PSK, FSK, concept of frequency multiplexing
10. Comparison of digital modulation methods:
• Need for linear ampliﬁers (transmitter complexity)
• Receiver complexity
• Bandwidth eﬃciency
Types of questions (format):
1. Short answer, with a 12 sentence limit. There may be multiple correct answers, but there may be many
wrong answers (or right answer / wrong reason combinations).
2. True / false or multiple choice.
3. Work out solution. Example: What is the probability of bit error vs.
E
b
N
0
for this signal space diagram?
4. System design: Here are some engineering requirements for this communication system. From this list of
possible modulation, which one would you recommend? What bandwidth and bit rate would it achieve?
You will be provided the table of the Qinv function, and the plot of Qinv. You can use two sides of an 8.5 x 11
sheet of paper.
Lecture 23
Today: (1) Source Coding
• HW 9 is due today at 5pm. OH today 23pm, Monday 45pm.
• Final homework: HW 10, which will be due April 28 at 5pm.
35 Source Coding
This topic is a semesterlong course at the graduate level in itself; but the basic ideas can be presented pretty
quickly.
One good reference:
• C. E. Shannon, “A mathematical theory of communication,” Bell System Technical Journal, vol. 27, pp.
379423 and 623656, July and October, 1948. http://www.cs.belllabs.com/cm/ms/what/shannonday/
paper.html
We have done a lot of counting of bits as our primary measure of communication systems. Our information
source is measured in bits, or in bits per second. Modulation schemes’ bandwidth eﬃciency is measured in bits
per Hertz, and energy eﬃciency is energy per bit over noise PSD. Everything is measured in bits!
But how do we measure the bits of a source (e.g., audio, video, email, ...)? Information can often be
represented in many diﬀerent ways. Images and sound can be encoded in diﬀerent ways. Text ﬁles can be
presented in diﬀerent ways.
Here are two misconceptions:
ECE 5520 Fall 2009 128
1. Using the ﬁle size tells you how much information is contained within the ﬁle.
2. Take the log
2
of the number of diﬀerent messages you could send.
For example, consider a digital black & white image (not grayscale, but truly black or white).
1. You could store it as a list of pixels. Each pixel has two possibilities (possible messages), thus we could
encode it in log
2
2 or one bit per pixel.
2. You could simply send the coordinates of the pixels of one of the colors (e.g.all black pixels).
How many bits would be used in these two representations? What would make you decide which one is more
eﬃcient?
You can see that equivalent representations can use diﬀerent number of bits. This is the idea behind source
compression. For example, .zip or .tar ﬁles represent the exact same information that was contained in the
original ﬁles, but with fewer bits.
What if we had a ﬁxed number of bits to send any image, and we used the sparse B&W image coding scheme
(2.) above? Sometimes, the number of bits in the compressed image would exceed what we had allocated. This
would introduce errors into the image.
Two types of compression algorithms:
• Lossless: e.g., Zip or compress.
• Lossy: e.g., JPEG, MP3
Note: Both “zip” and the unix “compress” commands use the LempelZiv algorithm for source compression.
So what is the intrinsic measure of bits of text, an image, audio, or video?
35.1 Entropy
Entropy is a measure of the randomness of a random variable. Randomness and information, in nontechnical
language, are just two perspectives on the same thing:
• If you are told the value of a r.v. that doesn’t vary that much, that telling conveys very little information
to you.
• If you are told the value of a very “random” r.v., that telling conveys quite a bit of information to you.
Our technical deﬁnition of entropy of a random variable is as follows.
Def ’n: Entropy
Let X be a discrete random variable with pmf p
X
(x
i
) = P [X = x]. Here, there is a ﬁnite or countably inﬁnite
set S
X
, and x ∈ S
X
. We will shorten the notation by using p
i
as follows:
p
i
= p
X
(x
i
) = P [X = x
i
]
where ¦x
1
, x
2
, . . .¦ is an ordering of the possible values in S
X
. Then the entropy of X, in units of bits, is deﬁned
as,
H [X] = −
i
p
i
log
2
p
i
(68)
Notes:
ECE 5520 Fall 2009 129
• H [X] is an operator on a random variable, not a function of a random variable. It returns a (deterministic)
number, not another random variable. This it is like E[X], another operator on a random variable.
• Entropy of a discrete random variable X is calculated using the probability values of the pmf of X, p
i
.
Nothing else is needed.
• The sum will be from i = 1 . . . N when [S
X
[ = N < ∞.
• Use that 0 log 0 = 0. This is true in the limit of xlog x as x → 0
+
.
• All “log” functions are logbase2 in information theory unless otherwise noted. Keep this in mind when
reading a book on information theory. The “reason” the units are bits is because of the base2 of the
log. Actually, when theorists use log
e
or the natural log, they express information in “nats”, short for
“natural” digits.
Example: Binary r.v.
A binary (Bernoulli) r.v. has pmf,
p
X
(x) =
_
_
_
s, x = 1
1 −s, x = 0
0, o.w.
What is the entropy H [X] as a function of s?
Solution: Entropy is given by (68) and is:
H[X] = −s log
2
s −(1 −s) log
2
(1 −s)
The solution is plotted in Figure 66.
0 0.2 0.4 0.6 0.8 1
0
0.2
0.4
0.6
0.8
1
P[X=1]
E
n
t
r
o
p
y
H
[
X
]
Figure 66: Entropy of a binary r.v.
Example: Nonuniform source with ﬁve messages
Some signals are more often close to zero (e.g.audio). Model the r.v. X to have pmf
p
X
(x) =
_
¸
¸
¸
¸
¸
¸
_
¸
¸
¸
¸
¸
¸
_
1/16, x = 2
1/4, x = 1
1/2, x = 0
1/8, x = −1
1/16, x = −2
0, o.w.
ECE 5520 Fall 2009 130
What is its entropy H [X]?
Solution:
H [X] =
1
2
log
2
2 +
1
4
log
2
4 +
1
8
log
2
8 + 2
1
16
log
2
16
=
15
8
bits (69)
Other questions:
1. Do you need to know what the symbol set S
X
is?
2. Would multiplying X by 2 change its entropy?
3. Would an arbitrary onetoone function change the entropy of X?
35.2 Joint Entropy
Def ’n: Joint Entropy
The joint entropy of two random variables X
1
, X
2
with event sets S
X
1
and S
X
2
is deﬁned as
H[X
1
, X
2
] = −
x
1
∈S
X
1
x
2
∈S
X
2
p
X
1
,X
2
(x
1
, x
2
) log
2
p
X
1
,X
2
(x
1
, x
2
) (70)
For N joint random variables, X
1
, . . . , X
N
, entropy is
H[X
1
, . . . , X
N
] = −
x
1
∈S
X
1
x
N
∈S
X
N
p
X
1
,...,X
N
(x
1
, . . . , x
N
) log
2
p
X
1
,...,X
N
(x
1
, . . . , x
N
)
What is the entropy for N i.i.d. random variables? You can show that
H[X
1
, . . . , X
N
] = −N
x
1
∈S
X
1
p
X
1
(x
1
) log
2
p
X
1
(x
1
) = NH(X
1
)
The entropy of N i.i.d. random variables has N times the entropy of any one of them. In addition, the entropy
of any N independent (but possibly with diﬀerent distributions) r.v.s is just the sum of the entropy of each
individual r.v.
When r.v.s are not independent, the joint entropy of N r.v.s is less than N times the entropy of one of them.
Intuitively, if you know some of them, because of the dependence or correlation, the rest that you don’t know
become less informative. For example, the B&W image, since pixels are correlated in space, the joint r.v. of
several neighboring pixels will have less entropy than the sum of the individual pixel entropies.
35.3 Conditional Entropy
How much additional entropy is in the joint random variables X
1
, X
2
compared just to one of them? This is
often an important question because it answers the question, “How much additional information do I get from
both, compared to just one of them?”. We call this diﬀerence the conditional entropy, H[X
2
[X
1
]:
H[X
2
[X
1
] = H[X
2
, X
1
] −H[X
1
]. (71)
ECE 5520 Fall 2009 131
What is an equation for H[X
2
[X
1
] as a function of the joint probabilities p
X
1
,X
2
(x
1
, x
2
) and the conditional
probabilities p
X
2
X
1
(x
2
[x
1
).
Solution: Plugging in (68) for H[X
2
, X
1
] and H[X
1
],
H[X
2
[X
1
] = −
x
1
∈S
X
1
x
2
∈S
X
2
p
X
1
,X
2
(x
1
, x
2
) log
2
p
X
1
,X
2
(x
1
, x
2
) +
x
1
∈S
X
1
p
X
1
(x
1
) log
2
p
X
1
(x
1
)
= −
x
1
∈S
X
1
x
2
∈S
X
2
p
X
1
,X
2
(x
1
, x
2
) log
2
p
X
1
,X
2
(x
1
, x
2
) +
x
1
∈S
X
1
_
_
x
2
∈S
X
2
p
X
1
,X
2
(x
1
, x
2
)
_
_
log
2
p
X
1
(x
1
)
= −
x
1
∈S
X
1
x
2
∈S
X
2
p
X
1
,X
2
(x
1
, x
2
) (log
2
p
X
1
,X
2
(x
1
, x
2
) −log
2
p
X
1
(x
1
))
= −
x
1
∈S
X
1
x
2
∈S
X
2
p
X
1
,X
2
(x
1
, x
2
) log
2
p
X
1
,X
2
(x
1
, x
2
)
p
X
1
(x
1
)
= −
x
1
∈S
X
1
x
2
∈S
X
2
p
X
1
,X
2
(x
1
, x
2
) log
2
p
X
2
X
1
(x
2
[x
1
) (72)
Note the asymmetry – there is the joint probability multiplied by the log of the conditional probability. This
is not like either the joint or the marginal entropy.
We could also have multivariate conditional entropy,
H[X
N
[X
N−1
, . . . , X
1
] = −
x
N−1
∈S
X
N−1
x
1
∈S
X
1
p
X
1
,...,X
N
(x
1
, x
N
) log
2
p
X
N
X
N−1
,...,X
1
(x
N
[x
N−1
, . . . , x
1
)
which is the additional entropy (or information) contained in the Nth random variable, given the values of the
N −1 previous random variables.
35.4 Entropy Rate
Typically, we’re interested in discretetime random processes, in which we have random variables X
1
, X
2
, . . ..
Since there are inﬁnitely many of them, the joint entropy of all of them may go to inﬁnity as N → ∞. For this
case, we are more interested in the rate. How many additional bits, in the limit, are needed for the average r.v.
as N → ∞?
Def ’n: Entropy Rate
The entropy rate of a stationary discretetime random process, in units of bits per random variable (a.k.a.
source output), is deﬁned as
H = lim
N→∞
H[X
N
[X
N−1
, . . . , X
1
].
It can be shown that entropy rate can equivalently be written as
H = lim
N→∞
1
N
H[X
1
, X
2
, . . . , X
N
].
Example: Entropy of English text
Let X
i
be the ith letter or space in a common English sentence. What is the sample space S
X
i
? Is X
i
uniform
on that space?
ECE 5520 Fall 2009 132
What is H[X
i
]? Solution: I had Matlab read in the text of Shakespeare’s Romeo and Juliet. See Figure
67(a). For this pmf, I calculated an entropy of H = 4.1199. The Proakis & Salehi book mentions that this value
for general English text is about 4.3.
What is H[X
i
, X
i+1
]? Solution: Again, using Matlab on Shakespeare’s Romeo and Juliet, I calculated the
entropy of the joint pmf of each twoletter combination. This gives me the twodimensional pmf shown in Figure
??(b). I calculate an entropy of 7.46, which is 2 3.73. For the threeletter combinations, the joint entropy was
10.04 = 3 3.35. For fourletter combinations, the joint entropy was 11.98 = 4 2.99.
You can see that the average entropy in bits per letter is decreasing quickly.
(a)
sp f k p u z
0
0.05
0.1
0.15
0.2
Space or Letter
P
o
c
c
u
r
r
e
n
c
e
(b) Second Space or Letter
F
i
r
s
t
S
p
a
c
e
o
r
L
e
t
t
e
r
sp f k p u z
sp
f
k
p
u
z
0.005
0.01
0.015
0.02
0.025
0.03
Figure 67: PMF of (a) single letters and (b) twoletter combinations (including spaces) in Shakespeare’s Romeo
and Juliet.
What is the entropy rate, H? Solution: For N = 10, we have H = 1.3 bits/letter [taken from Proakis &
Salehi Section 6.2].
35.5 Source Coding Theorem
The key connection between this mathematical deﬁnition of entropy and the bit rate that we’ve been talking
about all semester is given by the source coding theorem. It is one of the two fundamental theorems of
information theory, and was introduced by Claude Shannon in 1948.
Theorem: A source with entropy rate H can be encoded with arbitrarily small error probability, at any rate
R (bits / source output) as long as R > H. Conversely, if R < H, the error probability will be bounded away
from zero, independent of the complexity of the encoder and the decoder employed.
Proof: Proof: Using typical sequences. See Shannon’s original 1948 paper.
Notes:
• Here, an ‘error’ occurs when your compressed version of the data is not exactly the same as the original.
Example: B&W images.
• R is our 1/T
b
.
• Theorem fact: Information measure (entropy) gives us a minimum bit rate.
• What is the minimum possible rate to encode English text (if you remove all punctuation)?
• The theorem does not tell us how to do it – just that it can be done.
ECE 5520 Fall 2009 133
• The theorem does not tell us how well it can be done if N is not inﬁnite. That is, for a ﬁnite source, the
rate may need to be higher.
Lecture 24
Today: (1) Channel Capacity
36 Review
Last time, we deﬁned entropy,
H[X] = −
i
p
i
log
2
p
i
and entropy rate,
H = lim
N→∞
1
N
H[X
1
, X
2
, . . . , X
N
].
We showed that entropy can be used to quantify information. Given our information source X or ¦X
i
¦, the
value of H[X] or H gives us a measure of how many bits we actually need to use to encode, without loss, the
source data.
The major result was the Shannon’s source coding theorem, which says that a source with entropy rate H
can be encoded with arbitrarily small error probability, at any rate R (bits / source output) as long as R > H.
Any lower rate than H would guarantee loss of information.
37 Channel Coding
Now, we turn to the noisy channel. This discussion of entropy allows us to consider the maximum data rate
which can be carried without error on a bandlimited channel, which is aﬀected by additive White Gaussian
noise (AWGN).
37.1 R. V. L. Hartley
Ralph V. L. Hartley (born Nov. 30, 1888) received the A.B. degree from the University of Utah in 1909. He
worked as a researcher for the Western Electric Company, involved in radio telephony. Here he developed the
“Hartley oscillator”. Afterwards, at Bell Laboratories, he developed relationships useful for determining the
capacity of bandlimited communication channels. In July 1928, he published in the Bell System Technical
Journal a paper on “Transmission of Information”.
Hartley was particularly inﬂuenced by Nyquist’s result. When transmitting a sequence of pulses, each of
duration T
sy
, Nyquist determined that the pulse rate was limited to two times the available channel bandwidth
B,
1
T
sy
≤ 2B.
In Hartley’s 1928 paper, he considered digital transmission in pulseamplitude modulated systems. The
pulse rate was limited to 2B, as described by Nyquist. But, depending on how pulse amplitudes were chosen,
each pulse could represent more or less information.
In particular, Hartley assumed that the maximum amplitude available to the transmitter was A. Then,
Hartley made the assumption that the communication system could discern between pulse amplitudes, if they
ECE 5520 Fall 2009 134
were at separated by at least a voltage spacing of A
δ
. Given that a PAM system operates from 0 to A in
increments of A
δ
, the number of diﬀerent pulse amplitudes (symbols) is
M = 1 +
A
A
δ
Note that early receivers were modiﬁed AM envelope detectors, and did not deal well with negative amplitudes.
Next, Hartley used the ‘bit’ measure to quantify the data which could be encoded using M amplitude levels,
log
2
M = log
2
_
1 +
A
A
δ
_
Finally, Hartley quantiﬁed the data rate using Nyquist’s relationship to determine the maximum rate C, in
bits per second, possible from the digital communication system,
C = 2Blog
2
_
1 +
A
A
δ
_
(Note that the Hartley transform came later in his career in 1942).
37.2 C. E. Shannon
What was left unanswered by Hartley’s capacity formula was the relationship between noise and the minimum
amplitude separation between symbols. Engineers would have to be conservative when setting A
δ
to ensure a
low probability of error.
Furthermore, the capacity formula was for a particular type of PAM system, and did not say anything
fundamental about the relationship between capacity and bandwidth for arbitrary modulation.
37.2.1 Noisy Channel
Shannon did take into account an AWGN channel, and used statistics to develop a universal bound for capacity,
regardless of modulation type. In this AWGN channel, the ith symbol sample at the receiver (after the matched
ﬁlter, assuming perfect syncronization) is y
i
,
y
i
= x
i
+z
i
where X is the transmitted signal and Z is the noise in the channel. The noise term z
i
is assumed to be i.i.d.
Gaussian with variance P
N
.
37.2.2 Introduction of Latency
Shannon’s key insight was to exchange latency (time delay) for reduced probability of error. In fact, his capacity
bound considers simultaneously demodulating sequences of received symbols, y = [y
1
, . . . , y
n
], of length n. All
n symbols are received before making a decision. This late decision will decide all values of y = [x
1
, . . . , x
n
]
simultaneously. Further, Shannon’s proof considers the limiting case as n → ∞.
This asymptotic limit as n → ∞ allows for a proof using the statistical convergence of a sequence of random
variables. In particular, we need a law called the law of large numbers (ECE 6962 topic). This law says that
the following event,
1
n
n
i=1
(y
i
−x
i
)
2
≤ P
N
happens with probability one, as n → ∞. In other words, as n → ∞, the measured value y will be located
within an ndimensional sphere (hypersphere) of radius
√
nP
N
with center x.
ECE 5520 Fall 2009 135
37.2.3 Introduction of Power Limitation
Shannon also formulated the problem as a powerlimited case, in which the average power in the desired signal
x
i
was limited to P. That is,
1
n
n
i=1
x
2
i
≤ P
This combination of signal power limitation and noise power results in the fact that,
1
n
n
i=1
y
2
i
≤
1
n
n
i=1
x
2
i
+
1
n
n
i=1
(y
i
−x
i
)
2
≤ P +P
N
As a result
y
2
≤ n(P +P
N
)
This result says that the vector y, with probability one as n → ∞, is contained within a hypersphere of radius
_
n(P +P
N
) centered at the origin.
37.3 Combining Two Results
The two results, together, show how we many diﬀerent symbols we could have uniquely distinguished, within
a period of n sample times. Hartley asked how many symbol amplitudes could be ﬁt into [0, A] such that they
are all separated by A
δ
. Shannon’s formulation asks us how many multidimensional amplitudes x
i
can be ﬁt
into a hypersphere of radius
_
n(P +P
N
) centered at the origin, such that hyperspheres of radius
√
nP
N
do
not overlap. This is shown in P&S in Figure 9.9 (pg. 584).
This number M is the number of diﬀerent messages that could have been sent in n pulses. The result of
this geometrical problem is that
M =
_
1 +
P
P
N
_
(
n/2) (73)
37.3.1 Returning to Hartley
Adjusting Hartley’s formula, if we could send M messages now in n pulses (rather than 1 pulse) we would adjust
capacity to be:
C =
2B
n
log
2
M
Using the M from (76) above,
C =
2B
n
n
2
log
2
_
1 +
P
P
N
_
= Blog
2
_
1 +
P
P
N
_
37.3.2 Final Results
Finally, we can replace the noise variance P
N
with the relationship between noise twosided PSD N
0
/2, by
P
N
= N
0
B
ECE 5520 Fall 2009 136
So ﬁnally we have the ShannonHartley Theorem,
C = Blog
2
_
1 +
P
N
0
B
_
(74)
Typically, people also use W = B so you’ll see also
C = W log
2
_
1 +
P
N
0
W
_
(75)
This result says that a communication system can operate at bit rate C (in a bandlimited channel
with width W given power limit P and noise value N
0
), with arbitrarily low probability of error.
Shannon also proved that any system which operates at a bit rate higher than the capacity C will certainly
incur a positive bit error rate. Any practical communication system must operate at R
b
< C, where R
b
is the
operating bit rate.
Note that the ratio
P
N
0
W
is the signal power divided by the noise power, or signal to noise ratio (SNR). Thus
the capacity bound is also written C = W log
2
(1 +SNR).
37.4 Eﬃciency Bound
Recall that c
b
= PT
b
. That is, the energy per bit is the power multiplied by the bit duration. Thus from (74),
C = W log
2
_
1 +
c
b
/T
b
N
0
W
_
or since R
b
= 1/T
b
,
C = W log
2
_
1 +
R
b
W
c
b
N
0
_
Here, C is just a capacity limit. We know that our bit rate R
b
≤ C, so
R
b
W
≤ log
2
_
1 +
R
b
W
c
b
N
0
_
Deﬁning η =
R
b
W
,
η ≤ log
2
_
1 +η
c
b
N
0
_
This expression can’t analytically be solved for η. However, you can look at it as a bound on the bandwidth
eﬃciency as a function of the
E
b
N
0
ratio. This relationship is shown in Figure 70.
Lecture 25
Today: (1) Channel Capacity
• Lecture Tue April 28, is canceled. Instead, please the Robert Graves lecture 3:05 PM in Room 105 WEB,
“Digital Terrestrial Television Broadcasting”.
• The ﬁnal project is due Tue, May 5, 5pm. No late projects are accepted, because of the late due date.
If you think you might be late, set your deadline to May 4.
• Everyone gets a 100% on the “Discussion Item” which I never implemented.
ECE 5520 Fall 2009 137
(a)
0 5 10 15 20 25 30
0
2
4
6
8
10
12
14
E
b
/N
0
ratio (dB)
B
i
t
s
/
s
e
c
o
n
d
p
e
r
H
e
r
t
z
(b)
0 5 10 15 20 25 30
10
−1
10
0
10
1
E
b
/N
0
ratio (dB)
B
i
t
s
/
s
e
c
o
n
d
p
e
r
H
e
r
t
z
Figure 68: From the ShannonHartley theorem, bound on bandwidth eﬃciency, η, on a (a) linear plot and (b)
log plot.
ECE 5520 Fall 2009 138
38 Review
Last time, we deﬁned entropy,
H[X] = −
i
p
i
log
2
p
i
and entropy rate,
H = lim
N→∞
1
N
H[X
1
, X
2
, . . . , X
N
].
We showed that entropy can be used to quantify information. Given our information source X or ¦X
i
¦, the
value of H[X] or H gives us a measure of how many bits we actually need to use to encode, without loss, the
source data.
The major result was the Shannon’s source coding theorem, which says that a source with entropy rate H
can be encoded with arbitrarily small error probability, at any rate R (bits / source output) as long as R > H.
Any lower rate than H would guarantee loss of information.
39 Channel Coding
Now, we turn to the noisy channel. This discussion of entropy allows us to consider the maximum data rate
which can be carried without error on a bandlimited channel, which is aﬀected by additive White Gaussian
noise (AWGN).
39.1 R. V. L. Hartley
Ralph V. L. Hartley (born Nov. 30, 1888) received the A.B. degree from the University of Utah in 1909. He
worked as a researcher for the Western Electric Company, involved in radio telephony. Afterwards, at Bell Labo
ratories, he developed relationships useful for determining the capacity of bandlimited communication channels.
In July 1928, he published in the Bell System Technical Journal a paper on “Transmission of Information”.
Hartley was particularly inﬂuenced by Nyquist’s sampling theorem. When transmitting a sequence of pulses,
each of duration T
s
, Nyquist determined that the pulse rate was limited to two times the available channel
bandwidth B,
1
T
s
≤ 2B.
In Hartley’s 1928 paper, he considered digital transmission in pulseamplitude modulated systems. The
pulse rate was limited to 2B, as described by Nyquist. But, depending on how pulse amplitudes were chosen,
each pulse could represent more or less information.
In particular, Hartley assumed that the maximum amplitude available to the transmitter was A. Then,
Hartley made the assumption that the communication system could discern between pulse amplitudes, if they
were at separated by at least a voltage spacing of A
δ
. Given that a PAM system operates from 0 to A in
increments of A
δ
, the number of diﬀerent pulse amplitudes (symbols) is
M = 1 +
A
A
δ
Note that early receivers were modiﬁed AM envelope detectors, and did not deal well with negative amplitudes.
Next, Hartley used the ‘bit’ measure to quantify the data which could be encoded using M amplitude levels,
log
2
M = log
2
_
1 +
A
A
δ
_
ECE 5520 Fall 2009 139
Finally, Hartley quantiﬁed the data rate using Nyquist’s relationship to determine the maximum rate C, in
bits per second, possible from the digital communication system,
C = 2Blog
2
_
1 +
A
A
δ
_
39.2 C. E. Shannon
What was left unanswered by Hartley’s capacity formula was the relationship between noise and the minimum
amplitude separation between symbols. Engineers would have to be conservative when setting A
δ
to ensure a
low probability of error.
Furthermore, the capacity formula was for a particular type of PAM system, and did not say anything
fundamental about the relationship between capacity and bandwidth for arbitrary modulation.
39.2.1 Noisy Channel
Shannon did take into account an AWGN channel, and used statistics to develop a universal bound for capacity,
regardless of modulation type. In this AWGN channel, the ith symbol sample at the receiver (after the matched
ﬁlter, assuming perfect synchronization) is y
i
,
y
i
= x
i
+z
i
where X is the transmitted signal and Z is the noise in the channel. The noise term z
i
is assumed to be
i.i.d. Gaussian with variance E
N
= N
0
/2.
39.2.2 Introduction of Latency
Shannon’s key insight was to exchange latency (time delay) for reduced probability of error. In fact, his capacity
bound considers ndimensional signaling. So the received vector is y = [y
1
, . . . , y
n
], of length n. These might be
truly an ndimensional signal (e.g., FSK), or they might use multiple symbols over time (recall that symbols at
diﬀerent multiples of T
s
are orthogonal). In either case, Shannon uses all n dimensions in the constellation – the
detector must use all n samples of y to make a decision. In the multiple symbols over time, this late decision
will decide all values of x = [x
1
, . . . , x
n
] simultaneously. Further, Shannon’s proof considers the limiting case as
n → ∞.
This asymptotic limit as n → ∞ allows for a proof using the statistical convergence of a sequence of random
variables. In particular, we need a law called the law of large numbers. This law says that the following event,
1
n
n
i=1
(y
i
−x
i
)
2
≤ E
N
happens with probability one, as n → ∞. In other words, as n → ∞, the measured value y will be located
within an ndimensional sphere (hypersphere) of radius
√
nE
N
with center x.
39.2.3 Introduction of Power Limitation
Shannon also formulated the problem as a energylimited case, in which the maximum symbol energy in the
desired signal x
i
was limited to E. That is,
1
n
n
i=1
x
2
i
≤ E
ECE 5520 Fall 2009 140
This combination of signal energy limitation and noise energy results in the fact that,
1
n
n
i=1
y
2
i
≤
1
n
n
i=1
x
2
i
+
1
n
n
i=1
(y
i
−x
i
)
2
≤ E +E
N
As a result
y
2
≤ n(E +E
N
)
This result says that the vector y, with probability one as n → ∞, is contained within a hypersphere of radius
_
n(E +E
N
) centered at the origin.
39.3 Combining Two Results
The two results, together, show how we many diﬀerent symbols we could have uniquely distinguished, within
a period of n sample times. Hartley asked how many symbol amplitudes could be ﬁt into [0, A] such that they
are all separated by A
δ
. Shannon’s formulation asks us how many multidimensional amplitudes x
i
can be ﬁt
into a hypersphere of radius
_
n(E +E
N
) centered at the origin, such that hyperspheres of radius
√
nE
N
do
not overlap. This is shown in Figure 69.
n
E
E
(
+
)
N
radius
nE
N
Figure 69: Shannon’s capacity formulation simpliﬁes to the geometrical question of: how many hyperspheres of
a smaller radius
√
nE
N
ﬁt into a hypersphere of radius
_
n(E +E
N
)?
This number M is the number of diﬀerent messages that could have been sent in n pulses. The result of
this geometrical problem is that
M =
_
1 +
E
E
N
_
n/2
(76)
39.3.1 Returning to Hartley
Adjusting Hartley’s formula, if we could send M messages now in n pulses (rather than 1 pulse) we would adjust
capacity to be:
C =
2B
n
log
2
M
Using the M from (76) above,
C =
2B
n
n
2
log
2
_
1 +
E
E
N
_
= Blog
2
_
1 +
E
E
N
_
ECE 5520 Fall 2009 141
39.3.2 Final Results
Since energy is power multiplied by time, E = PT
s
=
P
2B
where P is the maximum signal power and B is the
bandwidth, and E
N
= N
0
/2, we have the ShannonHartley Theorem,
C = Blog
2
_
1 +
P
N
0
B
_
. (77)
This result says that a communication system can operate at bit rate C (in a bandlimited channel
with width B given power limit E and noise value N
0
), with arbitrarily low probability of error.
Shannon also proved that any system which operates at a bit rate higher than the capacity C will certainly
incur a positive bit error rate. Any practical communication system must operate at R
b
< C, where R
b
is the
operating bit rate.
Note that the ratio
E
N
0
B
is the signal power divided by the noise power, or signal to noise ratio (SNR). Thus
the capacity bound is also written C = Blog
2
(1 +SNR).
39.4 Eﬃciency Bound
Another way to write the maximum signal power P is to multiply it by the bit period and use it as the maximum
energy per bit, i.e., c
b
= PT
b
. That is, the energy per bit is the maximum power multiplied by the bit duration.
Thus from (77),
C = Blog
2
_
1 +
c
b
/T
b
N
0
B
_
or since R
b
= 1/T
b
,
C = Blog
2
_
1 +
R
b
B
c
b
N
0
_
Here, C is just a capacity limit. Be know that our bit rate R
b
≤ C, so
R
b
B
≤ log
2
_
1 +
R
b
B
c
b
N
0
_
Deﬁning η =
R
b
B
(the spectral eﬃciency),
η ≤ log
2
_
1 +η
c
b
N
0
_
This expression can’t analytically be solved for η. However, you can look at it as a bound on the bandwidth
eﬃciency as a function of the
E
b
N
0
ratio. This relationship is shown in Figure 70. Figure 71 is the plot on a logy
axis with some of the modulation types discussed this semester.
ECE 5520 Fall 2009 142
0 5 10 15 20 25 30
0
2
4
6
8
10
12
14
E
b
/N
0
ratio (dB)
B
i
t
s
/
s
e
c
o
n
d
p
e
r
H
e
r
t
z
Figure 70: From the ShannonHartley theorem, bound on bandwidth eﬃciency, η.
−10 0 10 20 30
10
−2
10
−1
10
0
10
1
E
b
/N
0
(dB)
B
a
n
d
w
i
d
t
h
E
f
f
i
c
i
e
n
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4−FSK
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16−QAM
64−QAM
256−QAM
1024−QAM
8−PSK
16−PSK
32−PSK
64−PSK
Figure 71: From the ShannonHartley theorem bound with achieved bandwidth eﬃciencies of MQAM, MPSK,
and MFSK.
ECE 5520 Fall 2009
2
Contents
1 Class Organization 2 Introduction 2.1 ”Executive Summary” . . . . . . . . . . . 2.2 Why not Analog? . . . . . . . . . . . . . . 2.3 Networking Stack . . . . . . . . . . . . . . 2.4 Channels and Media . . . . . . . . . . . . 2.5 Encoding / Decoding Block Diagram . . . 2.6 Channels . . . . . . . . . . . . . . . . . . 2.7 Topic: Random Processes . . . . . . . . . 2.8 Topic: Frequency Domain Representations 2.9 Topic: Orthogonality and Signal spaces . 2.10 Related classes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8 8 8 9 9 10 10 11 12 12 12 13
3 Power and Energy 13 3.1 DiscreteTime Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14 3.2 Decibel Notation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14 4 TimeDomain Concept Review 15 4.1 Periodicity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15 4.2 Impulse Functions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16 5 Bandwidth 5.1 Continuoustime Frequency Transforms 5.1.1 Fourier Transform Properties . . 5.2 Linear Time Invariant (LTI) Filters . . . 5.3 Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16 17 19 20 20
6 Bandpass Signals 21 6.1 Upconversion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21 6.2 Downconversion of Bandpass Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22 7 Sampling 23 7.1 Aliasing Due To Sampling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24 7.2 Connection to DTFT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25 7.3 Bandpass sampling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27 8 Orthogonality 8.1 Inner Product of Vectors . . 8.2 Inner Product of Functions 8.3 Deﬁnitions . . . . . . . . . . 8.4 Orthogonal Sets . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28 28 29 30 31
32 9 Orthonormal Signal Representations 9.1 Orthonormal Bases . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33 9.2 Synthesis . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33 9.3 Analysis . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
ECE 5520 Fall 2009
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10 MultiVariate Distributions 10.1 Random Vectors . . . . . . . . . . . . . . . . . 10.2 Conditional Distributions . . . . . . . . . . . . 10.3 Simulation of Digital Communication Systems . 10.4 Mixed Discrete and Continuous Joint Variables 10.5 Expectation . . . . . . . . . . . . . . . . . . . . 10.6 Gaussian Random Variables . . . . . . . . . . . 10.6.1 Complementary CDF . . . . . . . . . . 10.6.2 Error Function . . . . . . . . . . . . . . 10.7 Examples . . . . . . . . . . . . . . . . . . . . . 10.8 Gaussian Random Vectors . . . . . . . . . . . . 10.8.1 Envelope . . . . . . . . . . . . . . . . .
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11 Random Processes 46 11.1 Autocorrelation and Power . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46 11.1.1 White Noise . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47 12 Correlation and MatchedFilter Receivers 12.1 Correlation Receiver . . . . . . . . . . . . . 12.2 Matched Filter Receiver . . . . . . . . . . . 12.3 Amplitude . . . . . . . . . . . . . . . . . . . 12.4 Review . . . . . . . . . . . . . . . . . . . . . 12.5 Correlation Receiver . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47 48 49 50 50 51
13 Optimal Detection 51 13.1 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51 13.2 Bayesian Detection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 52 14 Binary Detection 14.1 Decision Region . . . . . . . . . . . . . . . . . 14.2 Formula for Probability of Error . . . . . . . 14.3 Selecting R0 to Minimize Probability of Error 14.4 LogLikelihood Ratio . . . . . . . . . . . . . . 14.5 Case of a0 = 0, a1 = 1 in Gaussian noise . . . 14.6 General Case for Arbitrary Signals . . . . . . 14.7 Equiprobable Special Case . . . . . . . . . . 14.8 Examples . . . . . . . . . . . . . . . . . . . . 14.9 Review of Binary Detection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 52 53 53 53 55 55 56 56 56 57
15 Pulse Amplitude Modulation (PAM) 58 15.1 Baseband Signal Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58 15.2 Average Bit Energy in M ary PAM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59 16 Topics for Exam 1 60
.3 Binary PAM Error Probabilities . . . . . . . . 25. . . . . . . . . . 24. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18. . 17. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5 Correlation / Matched Filter Receivers . . . . . . . . . . . . . .6. . 17. . . . . . . . . . .ECE 5520 Fall 2009 4 61 61 61 61 62 62 63 63 63 64 66 . . .3 Signal Constellations . 25. . . . . . . . . . 24. . . . . . . . . 24. . . . . . . . . . 18. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17. . .3 Orthogonality and Signal Space . . . . . . . . . . . . . . . . .1 General Formula for Union Boundbased . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . PSD . . . . . . . . . . . . . . . . . . . . 69 21. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1 Bit Error Probabilities . . . . . . . . . . . . .1 Overview of Future Discussions on QAM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 72 23 M ary Detection Theory in N dimensional signal space 24 Quadrature Amplitude Modulation (QAM) 24. . . . . . . . . 24. . . . . . . . . . . . . . . 25. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2 Sampling and Aliasing . . . . . . . . . . .2 SquareRoot Raised Cosine Filtering . . . . . . . . . . . . . . . . . . . . . 24. . . . . . 24. . . . . . . . . . . . . . . . . . . . . .2 BER Function of Distance. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2 Options for Probability of Error Expressions . . . . . . . . . . 21 Intersymbol Interference 69 21. 19 Detection with Multiple Symbols 20 M ary PAM Probability of Error 20. . . . . . . . . . . . . . . . . .1 Multipath Radio Channel . . . . . . . . . . . . . . . . .1 Showing Orthogonality . . . . Noise PSD . . . . . . . . .4 Probability of Error in QPSK . . . .2 Bit Errors and Gray Encoding . . . . . . . . . . . 20.7 Systems which use QAM . . . . 25.2 Transmitter and Receiver Filters . . . . . .1 Signal Distance . . .2. . . . . . . . .4 Random Processes. . . . . . . . . . . . . . . . . . 18 Probability of Error in Binary PAM 18. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 72 74 76 77 78 79 79 79 79 80 80 81 81 81 82 82 84 25 QAM Probability of Error 25. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . Probability . . . . . . . . . . . . . . . . . . . . . . .6 Application of Union Bound . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . of Error . . . . . . . . . . . . . . . . . . .5 Union Bound . . . . . . . . 67 67 67 68 68 17 Additional Problems 17. . . . . . . . . . . . . . . . . .1 Symbol Error . . . . . . . . .3 Exact Error Analysis . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 70 22 Nyquist Filtering 70 22. . . . . . . . . . . . . . . . . . .5 Average Energy in MQAM . . . . . . . . . . . . . . .2 Constellation . . .1 Spectrum of Communication Signals . . . . . . . .1. . . . . . . . . . . . 72 22. . . . . . . . . . . . . .6 PhaseShift Keying .1 Symbol Error Rate and Average Bit Energy 20. . . . .1 Raised Cosine Filtering . . . . . . 17. 20. . . . . . . . . . . . . . . . . . . 25. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4 Angle and Magnitude Representation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88 88 89 90 90 90 91 93 93 94 94 95 95 28 Frequency Multiplexing 95 28. . . . 29. . . . . .3. . . . . .2 FSK . . .4 Bandwidth Eﬃciency vs. . . . . . . 29. . . . . . . . . . . .1 M ary NonCoherent FSK . . . . . . . . . . . . . . . . .9. .2 Transmission of FSK . . .ECE 5520 Fall 2009 5 26 QAM Probability of Error 85 26. . . . . . . . . . . . . . . . . . . .3. . . . . . .10Bandwidth of FSK . . . . . . .1.2 Power and Energy Limited Channels 30. . . . . . . .1 Linear / Nonlinear Ampliﬁers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 97 29 Comparison of Modulation Methods 29. . . . . . . . . .3 Reception of FSK . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98 98 99 99 100 100 101 101 101 102 102 103 103 104 105 105 106 108 110 110 111 111 112 . . . . . . . . . . . . . . . . . . . . . . . . . 0 29. . . . . . . . . . . . . . . . .2 Summary and Examples . . . . . . . . . . . . . . . . . . . 27. . . . . . . .2 DPSK Receiver . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . Part 2 . . . . . . . . . . . . . . . . . . . 0 E 29. . . . . . . . . . . . . . . . . . . . . .1 NearestNeighbor Approximate Probability of Error . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29. . . . . . . . . . . . . . . . . . . . . . . 27. 30. . . . 30. . . . . . . . . . . . . . .6 Receiver Block Diagrams . . . 29. . . . . . . . . . . . . . . . . . . .7 Oﬀset QPSK . . . . . . . . . . . 27. . . . . . . . . . . . . . . . . . . . 30. . . . . . . . . . . . . . . . . . . . . . . . .6. . . . .2 Summary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3 OFDM as an extension of FSK . . . . . . .3 Wired Channels .1 Frequency Selective Fading . . .8 Receiver Complexity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .8 Probability of Error for Noncoherent Binary FSK 27. . . 29. . . . 27. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30. . . . . . . . . . . . . . . . . . . 27. .3. . . 29. . . . . . . . . . . . . . . . . . . . .2 Points for Comparison . . . . . . . . . . . . . . . . . . . 29. . . . . . . . . . . .1. . . . . . . . . . . . . . . .5 Noncoherent Reception . . . . . . . . . . . . . . . . . . . . . . . . . .5 Fidelity (P [error]) vs. . . . . .2 Nonfreespace Channels . .1 Diﬀerential Encoding for BPSK . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .7 Probability of Error for Coherent Binary FSK . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27. . . . . . . . . . . . . .3 Bandwidth Eﬃciency . 29. . . Nb .1. . . . . . . . . . . . .4 Computing Noise Energy . . . .1 Orthogonal Frequencies . 29. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . PAM and QAM . . . . . . . . . . . . . . . 86 27 Frequency Shift Keying 27. .3. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27. . . . . . . . . .4 Coherent Reception . . . . . 27. . . . . . . 27. . . . . . . . . . . . . . . . . .1 DPSK Transmitter . .3. . 30. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30 Link Budgets and System Design 30. . . . . 96 28. Nb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3 Probability of Bit Error for DPSK 29. . . . . . . . . . . . . . . . . . . . . . . . .3 Computing Received Power . . .1 Free Space . . . . . . . . . . . . . . . . .9. . . . . . . . . . . . . . . . .6 Transmitter Complexity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .9 FSK Error Probabilities. . . . . . . . . . . 95 28. . . . . . . . . . . . . . . . . . . . . . . .1 PSK. . 85 26. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1 Link Budgets Given C/N0 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . E 29. 27. . .2 Beneﬁts of Frequency Multiplexing . . . . . . . . .
. . . . . . . . . . .2. . . .2 Final Results . . . . . . . . . . . . . . . . 37. . . . 37. . . 33. . . . 124 . .2 Seeing Interpolation as Filtering . . . . . . . . . . . . . . . . 33. . 33. . . . . . . . . . . . . . .2 Loop Filter . . . . . . . 130 . . . 120 . . . . . . . . . 131 . . . . . . . . . . . . . . . . . . 33. . . . . . . . . . . . . . . . . . . . . . 135 . . . . . . . . . . . . . .5 Examples . .5 DiscreteTime Implementations . . . . . . 123 . . . . . . . . . . .1 Noisy Channel . . . . . . . . . 35. . . . . . . . . . . . . . . . . . . . . . .3 VCO . . . . .1 R. . . . . . Shannon . . . . 124 . . . . . . . . . . . . . 37.6 Phase Locked Loops . . . . . . . . . . . . . . . . . . .6. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37. . . . . 119 . . . . . . . . . . . . . . . 33. . . . . . . . . . . . . . . . . . . . . . . 37. . . . . . . . . . . . . . . .1 QPSK Timing Error Detection . . . . . . 33. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 122 . . . . . .3. . E. . . . . . . . . .2 Timing Error Detection . . . . . . . . . . . . . . . . . . . . . . 35. 32. . . . . .ECE 5520 Fall 2009 6 30. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33. . . . . . . . . 115 . . . . . . . . . . . .6. . . . . . . . . . . . . . . . . . . . . . 117 118 . . . . . 124 . . . . . . . . . . . . 123 . . . . . . 37. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4 Analysis . . 135 . . . . . . .3 Approximate Interpolation Filters . . . . . . . . . . . . . . . . . . 128 .4 Entropy Rate . .5 Voltage Controlled Clock (VCC) . . . . . . . . . . . . . . V. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33. . . . . . . . . . . . . .3 Introduction of Power Limitation 37. . . .1 Phase Detector . . . 135 . . . . . . . .4 Eﬃciency Bound . . .3 Combining Two Results . . . . . . . . . . 122 . . . . . . . 121 . . . . . 134 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1 Review of Interpolation Filters . . . . . . 33. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 133 . . . 119 . . . . . . . 136 138 . . . . . 32. .1 Sampling Time Notation . . . . . . . . . . . 37. . . . . . . . . . . .2. . . . . 134 . . . . . . . . .3. . . . . . . . . . . . . . . . .4. . . . . . . 135 . . . . . . . . . . . . . . . . . . . 35. . . . . . . . . . . . . . . . . . . . . . . . . . . . 32. . . . . . . . . . . . .2 Joint Entropy . . 32. .5 Source Coding Theorem 36 Review 37 Channel Coding 37. . . . 130 . . . . . . . . 34 Exam 2 Topics 35 Source Coding 35. . . . . . . . . . . . . . . . . . . . . 38 Review . . . . . . . . . . . . 116 . . . . . . . . 112 31 Timing Synchronization 32 Interpolation 32. .6. . . . . . 132 133 133 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33. . 134 . . . . . .2 C. . . . . . . . . . . . L. . . . . . . . .6. . . . . 113 114 . . . . . . . . . . .4 Zerocrossing timing error detector (ZCTED) 33. . . 117 . . . . . . . .3 Conditional Entropy . . . .1 Entropy . . . . . . . . . . . . . . . . . . . 125 126 127 . . . . . . . . . . . . 116 . . . . . . . . . . . . .6. . .3 Earlylate timing error detector (ELTED) . . . . . . . . . . . . . . .4. . . . . . . . . . . . . . 33 Final Project Overview 33. . . .2 Introduction of Latency . . . . . . . . . . .1 Returning to Hartley . . . . . . . . . . . .2. . . . . . . . . . . . . . . . . . . . Hartley . . . .1 Higher order polynomial interpolation ﬁlters . . . . . . . . . . . . . . . . . .4 Implementations . . . . . . . . .
. . . . . . . . . . . . . . .2 C. . . . . . . . . . 39. . . . . . . . . . L. . . . . . . . . . . . . . . . . . . .2 Final Results . . . . . . . . . . . . . . . . . Shannon . . . . . . . . . . .1 Returning to Hartley . 39. . . . . Hartley . . . . .2 Introduction of Latency . . . . . . . . . . . . . . . . . . .3. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39. . . . .1 R. . . . . .2. . . . . . . . . . . . . . . . . . . . . 39. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . V. . . 39.2. . . . . . . . . . . . . . . . . . . . . . . . . . 39. . . . . . . . . .1 Noisy Channel . . . . E. . . .3. .2. . . . . . . . . . . .ECE 5520 Fall 2009 7 138 138 139 139 139 139 140 140 141 141 39 Channel Coding 39. . . . . . . . . . . . . . . . . . . . . . . . . . .4 Eﬃciency Bound . . . . . . . . . .3 Combining Two Results . . . . . . . . . . . . . . . .3 Introduction of Power Limitation 39. . . . . . . . . 39. . . . . . . . . . . . . . . . . . . . . . . . . . . .
and/or after lecture online.1 ”Executive Summary” Here is the one sentence version: We will study how to eﬃciently encode digital data on a noisy. For example. I will provide additional readings solely to provide another presentation style or ﬁt another learning style. They might be continuoustime (analog) signals (audio. there are few options in this area. and that the other half is sparse and needs supplemental material. If I didn’t. and I try to tailor my approach during the lecture. In this section we talk about what we’ll cover in this class. Prentice Hall. Salehi. These can be available to you at lecture. you must accept these conditions: 1. or data. Proakis and M. This year’s text covers primarily digital communications. 2nd edition. I ﬁnd it to be very wellwritten. discretevalued). the past text was the ‘standard’ text in the area for an undergraduate course. bandwidthlimited analog medium. The keys points stuﬀed into that one sentence are: 1. and more importantly. not just watching. Finally.e. Information sources might include audio. Proakis & Salehi. eﬃcient.e. I’m constantly updating my notes. Please take your own notes.G. you could just watch a recording! 2 Introduction A digital communication system conveys discretetime. text. J. These waveforms are from a discrete set of possible waveforms. realvalued. 2001. i. 2. on the channel (medium). continuoustime functions. I have taught ECE 5520 previously and have my lecture notes from past years. discretevalued information across a physical channel. Communication Systems Engineering. and does it in depth. images) and even 1D or 2D. video. My written notes do not and cannot reﬂect everything said during lecture: I answer questions and understand your perspective better after hearing your questions. 2. and highﬁdelity...ECE 5520 Fall 2009 8 Lecture 1 Today: (1) Syllabus (2) Intro to Digital Communications 1 Class Organization Textbook Few textbooks cover solely digital communications (without analog) in an introductory communications course. they may already be digital (discretetime. However. What set of waveforms should we use? Why? . so that decoding the data (i. So half of most textbooks are useless. Or. Digital information on an analog medium: We can send waveforms. And. Unless speciﬁed. But graduates today will almost always encounter / be developing solely digital communication systems. these are optional. Lecture Notes I type my lecture notes. what we won’t cover. Taking notes is important: I ﬁnd most learning requires some writing on your part. Students didn’t like that I had so many supplemental readings. Our object is to convey the signals or data to another place (or time) with as faithful representation as possible. even up to the lecture time. reception) at a receiver is simple.
. In digital comm. The 7layer OSI stack. Decoding the data: When receiving a signal (a function) in noise. and device power consumption • Bandwidth eﬃciency: due to coding gains • Moore’s Law is decreasing device costs for digital hardware • Increasing need for digital information • More powerful information security 2. we study digital communications from bits to bits. none of the original waveforms will match exactly. has an extensive analysis and study of analog communication systems. 2. Bandwidth. to have the source impedance match the destination impedance. That is.e. wireless car key).ECE 5520 Fall 2009 9 2. for power eﬃciency. and ﬁdelity? We all want high ﬁdelity. To manage complexity. WiFi. and we build a layered network to handle the application. In the recent past. How do you make a decision about which waveform was sent? 3. development of new systems will almost certainly be of digital communication systems. this course would study both analog and digital communication systems. Eﬃciency. and Fidelity: Fidelity is the correctness of the received data (i. There’s a lot more than this to the digital communication systems which you use on a daily basis (e. which you would study in a CS computer networking class. What is the tradeoﬀ between energy. this means that we want our waveform choices to match the channel and receiver to maximize the eﬃciency of the communication system. and low energy consumption and bandwidth usage (the costs of our communication system). we study how to take ones and zeros from a transmitter.3 Networking Stack In this course. iPhone.2 Why not Analog? The previous text used for this course. Why? • Fidelity • Energy: transmit power. wireless keyboard. by Proakis & Salehi. we (engineers) don’t try to build a system to do everything all at once. however.g. is as follows: • Application • Presentation (*) • Session (*) • Transport • Network . send them through a medium. You want. You can look at this like an impedance matching problem from circuits. such as radio and television broadcasting (Chapter 3).. Analog systems still exist and will continue to exist. bandwidth. We typically start with an application. Bluetooth. and then (hopefully) correctly identify the same ones and zeros at the receiver. What makes a receiver diﬃcult to realize? What choices of waveforms make a receiver simpler to implement? What techniques are used in a receiver to compensate? 4. the opposite of error rate).
optical ﬁber. It is primarily divided into: • Multiple Access Control (MAC) • Encoding • Channel / Medium We can control the MAC and the encoding chosen for a digital communication. mmwave bands.4 Channels and Media We can chose from a few media. coaxial cable. Notes: • Information source comes from higher networking layers. the physical layer has much more detail.) ECE 5520 is part of the bottom layer. including (top) transmitter.ECE 5520 Fall 2009 • Link Layer • Physical (PHY) Layer 10 (Note that there is also a 5layer model in which * layers are considered as part of the application layer. Microwave. waveguides. Includes sampling of continuoustime signals. and (bottom) receiver.5 Encoding / Decoding Block Diagram Source Encoder Channel Encoder Modulator Upconversion Information Source Other Components: Digital to Analog Converter Analog to Digital Converter Synchronization Channel Information Output Source Decoder Channel Decoder Demodulator Downconversion Figure 1: Block diagram of a singleuser digital communication system.. the physical layer. Also includes compression of those sources (lossy.. . . Here are some media: • EM Spectra: (anything above 0 Hz) Radio. 2. wire pairs. What are some compression methods that you’re familiar with? We present an introduction to source encoding at the end of this course. (middle) channel. It may be continuous or packetized. In fact. but we largely can’t change the properties of the medium (although there are exceptions). • Source Encoding: Finding a compact digital representation for the data source. or lossless). • Disk (data storage applications) 2. light • Acoustic: ultrasound • Transmission lines. and quantization of continuousvalued signals.
• Channels: See above for examples. The noise comes from a variety of sources. However. etc. And separation. attenuation in telephone wires varies by frequency. and how they are addressed. in the environment may change the channel slowly over time. These other transmitters whose signals we cannot completely cancel. Typical models are additive noise. In this course. and scattering of the signal oﬀ of the objects in the environment. Wideband wireless channels display multipath. we lump into the ‘interference’ category. A channel decoder. Narrowband wireless channels experience fading that varies quickly as a function of frequency. diﬀractions. or timeinvariant. These may result in nonGaussian noise distribution.proper matching of a modulation to a channel allows optimal information transfer. due to multiple timedelayed reﬂections. the channel may change very quickly over time.ECE 5520 Fall 2009 11 • Channel encoding refers to redundancy added to the signal such that any bit errors can be corrected. the TX and RX do not move or change. but predominantly: 1. • Modulation refers to the digitaltoanalog conversion which produces a continuoustime signal that can be sent on the physical channel. but we will mention what variations exist for particular channels. It is analogous to impedance matching . We will not study channel encoding. The linear ﬁltering of the channel result from the physics and EM of the medium. reduces complexity to the designer. The ﬁlter may be constant. or nonwhite noise spectral density. movement of cars. Well modeled as Gaussian. can correct some bit errors. For example. or linear ﬁltering channel. Interference from other transmitted signals. if the medium. 2. for mobile radio. Noise Transmitted Signal LTI Filter h(t) Received Signal Figure 2: Linear ﬁlter and additive noise channel model. like layering. but it is a topic in the (ECE 6520) Coding Theory. Modulation and demodulation will be the main focus of this course. trees. because of the redundancy. people. like impedance matching ensured optimal power transfer. thus it is referred to as additive white Gaussian noise (AWGN). we will focus primarily on the AWGN channel. . Why do we do both source encoding (which compresses the signal as much as possible) and also channel encoding (which adds redundancy to the signal)? Because of Shannon’s sourcechannel coding separation theorem. He showed that (given enough time) we can consider them separately without additional loss.6 Channels A channel can typically be modeled as a linear ﬁlter with the addition of noise. All of these can be modeled as linear ﬁlters. Even for stationary TX and RX. Thermal background noise: Due to the physics of living above 0 Kelvin. in real wireless channels. and white. 2.
2. and timing oﬀsets These random signals often pass through LTI ﬁlters. For the wireless channel. or a higher layer networking protocol (eg. The concept of sharing a channel is called multiple access (MA). in order to provide multiple independent means (dimensions) on which to modulate information. We’ll use signal spaces to show graphically the results. The Fourier transform of our modulated. and are sampled. M=8 M=16 Figure 3: Example signal space diagram for M ary Phase Shift Keying.7 Topic: Random Processes Random things in a communication system: • Noise in the channel • Signal (bits) • Channel ﬁltering. This means. . We can also employ multiple orthogonal signals in a single transmitter and receiver. in short. We want to build the best receiver possible despite the impediments. phase. transmitted signal is used to show that it meets the spectrum allocation limits of the FCC. There is a tradeoﬀ between frequency requirements and time requirements. we need to show that they are orthogonal. for (a) M = 8 and (b) M = 16. This may be tolerated. Optimal receiver design is something that we study using probability theory. and learn an algorithm to take an arbitrary set of signals and output a set of orthogonal signals with which to represent them. this is controlled by the FCC (in the US) and called spectrum allocation. attenuation.9 Topic: Orthogonality and Signal spaces To show that signals sharing the same channel don’t interfere with each other. Each point is a vector which can be used to send a 3 or 4 bit sequence. Separating signals by frequency band is called frequencydivision multiple access (FDMA).8 Topic: Frequency Domain Representations To ﬁt as many signals as possible onto a channel. that a receiver can uniquely separate them. TCPIP) can determine that an error occurred and then rerequest the data. as the example in Figure 36.ECE 5520 Fall 2009 12 2. which will be a major part of this course. 2. We will study orthogonal signals. we often split the signals by frequency. We have to tolerate errors. Noise and attenuation of the channel will cause bit errors to be made by the demodulator and even the channel decoder. and fading • Device frequency.. Signals in diﬀerent frequency bands are orthogonal.
3 Power and Energy Recall that energy is power times time. (ECE 5324) Antenna Theory. power is measured in Watts (W) which is the same as Joules/second (J/sec). recall that our standard in signals and systems is deﬁne our signals. Today’s lecture provides some tools to deal with power and energy. Signal Processing: (ECE 5530): Digital Signal Processing 3. Also.10 Related classes 1. 2. Advanced Classes: (ECE 6590) Software Radio. such as x(t). .ECE 5520 Fall 2009 13 2. Fourier Transform Two of the biggest limitations in communications systems are (1) energy / power . Networking: (ECE 5780) Embedded System Design. (CS 5480) Computer Networks 7. and starts the review of tools to analyze frequency content and bandwidth. Energy. (ECE 6540): Estimation Theory Lecture 2 Today: (1) Power. so x(t)2 is the power dissipated at time t (since power is equal to the voltage squared divided by the resistance). (ECE 3500) Signals and Systems. dB (2) Timedomain concepts (3) Bandwidth. Prerequisites: (ECE 5510) Random Processes. Breadth: (ECE 5325) Wireless Communications 5. (ECE 6520) Information Theory and Coding. (ECE 53205321) Microwave Engineering. Use the units: energy is measured in Joules (J). E will be inﬁnite because the signal is nonzero for an inﬁnite duration of time (it is always on). These signals we call power signals and we compute their power as P = lim 1 T →∞ 2T T −T x(t)2 dt The signal with ﬁnite energy is called an energy signal. as voltage signals (V). (ECE 5411) Fiberoptic Systems 4. When we want to know the power of a signal we assume it is being dissipated in a 1 Ohm resistor. (ECE 5720) Analog IC Design 6. A signal x(t) has energy deﬁned as E= ∞ −∞ x(t)2 dt For some signals. Devices and Circuits: (ECE 3700) Fundamentals of Digital System Design. and (2) bandwidth. Electromagnetics: EM Waves.
energy and power are deﬁned as: E = ∞ n=−∞ x(n)2 N n=−N (1) x(n)2 P 1 = lim N →∞ 2N + 1 (2) 3. I’m sorry this is not more obvious in the notation. Just convert any dB number into a sum of multiples of 10. whenever you see a function of n (or k. If Plin is the power in Watts. Our standard is to consider power gains and losses. not voltage gains and losses. taking the log10 of a voltage and expecting the result to be a dB power value. You may be more familiar with the x[n] notation. l. Example: Convert dB to linear values: (4) . it is a discretetime function.e. i. such as a gain (loss) L(f ) through a ﬁlter H(f ). Remember these two dB numbers: • 3 dB: This means the number is double in linear terms.ECE 5520 Fall 2009 14 3. and 1. 3. or it referred to a name ‘Bell’ so it was capitalized. Note that (3) could also be written as: [L(f )]dB = 20 log10 H(f ) Be careful with your use of 10 vs. With these three numbers. 20 in the dB formula. whenever you see a function of t (or perhaps τ ) it is a continuoustime function. you can quickly convert losses or gains between linear and dB units without a calculator. the output of the channel has 100 times less power than the input to the channel. • Only use 20 as the multiplier if you are converting from voltage to power. But. so when the ‘bel’ was ﬁrst proposed as log10 (·). [L(f )]dB = 10 log 10 H(f )2 (3) Note: Why is the capital B used? Either the lowercase ‘b’ in the SI system is reserved for bits. the channel has a loss of 20 dB.1 DiscreteTime Signals In this book..2 Decibel Notation We often use a decibel (dB) scale for power. it was capitalized. Matlab uses parentheses also. so we’ll follow the Rice text notation.they can apply to other unitless quantities as well. m). In either case. we refer to discrete samples of the sampled signal x as x(n). And maybe this one: • 1 dB: This means the number is a little over 25% more (multiply by 5/4) in linear terms. this refers to a loss in power. Essentially. In particular. • 10 dB: This means the number is ten times in linear terms. for example. So if we say. we use the unit decibel 10 log 10 (·) which is then abbreviated as dB in the SI system. then [P ]dBW = 10 log 10 Plin Decibels are more general . For discretetime signals.
30 dBW 2. where d is the cable length (typically in units of km). This is the Friis free space path loss formula.lin = 100Px.lin 2. where Po. 40 mW Example: Convert power relationships to dB: Convert the expression to one which involves only dB terms.lin and Gt.lin is the transmit power (W) and Pr.lin .ECE 5520 Fall 2009 15 1. Pr.2 W 2. Pt.lin is the received power (W).4. d is the path length (m).lin . and Gt.lin is the gain in any connectors. for all integers n.lin t. 33 dBm 3.lin = Pt.lin (4πd)2 are the linear gains in the antennas. where λ is the wavelength (m). If a signal is not periodic it is aperiodic. G G λ2 4 4.lin t. as deﬁned in Rice Ch.lin L−d cable.lin is the received power in a ﬁberoptic link. Def ’n: Periodic (discretetime) A DT signal x(n) is periodic if x(n) = x(n + N0 ) for some integer N0 = 0. .1 TimeDomain Concept Review Periodicity Def ’n: Periodic (continuoustime) A signal x(t) is periodic if x(t) = x(t + T0 ) for some constant T0 = 0 for all t ∈ R. 3. The main idea is that we have a limited amount of power which will be available at the receiver. 0.lin = Gconnector. The smallest positive integer N0 is the period. 1. The smallest such constant T0 > 0 is the period. when we discuss link budgets. These last two are what we will need in Section 6. Po. 4 dB Example: Convert linear values to dB: 1. Gconnector.lin is a loss in a 1 km cable. Periodic signals have Fourier series representations. Py. and Lcable. 2. 20 dB 4.
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4.2
Impulse Functions
Def ’n: Impulse Function The (Dirac) impulse function δ(t) is the function which makes
∞ −∞
x(t)δ(t)dt = x(0)
(5)
true for any function x(t) which is continuous at t = 0. We are deﬁning a function by its most important property, the ‘sifting property’. Is there another deﬁnition which is more familiar? Solution: δ(t) = lim
T →0
1/T , −T ≤ t ≤ T 0, o.w.
You can visualize δ(t) here as an inﬁnitely high, inﬁnitesimally wide pulse at the origin, with area one. This is why it ‘pulls out’ the value of x(t) in the integral in (5). Other properties of the impulse function: • Time scaling, • Symmetry, • Sifting at arbitrary time t0 , The continuoustime unit step function is u(t) = 1, t ≥ 0 0, o.w.
Example: Sifting Property ∞ What is −∞ sin(πt) δ(1 − t)dt? πt
The discretetime impulse function (also called the Kronecker delta or δK ) is deﬁned as: δ(n) = 1, n = 0 0, o.w.
(There is no need to get complicated with the math; this is well deﬁned.) Also, u(n) = 1, n ≥ 0 0, o.w.
5
Bandwidth
Bandwidth is another critical resource for a digital communications system; we have various deﬁnitions to quantify it. In short, it isn’t easy to describe a signal in the frequency domain with a single number. And, in the end, a system will be designed to meet a spectral mask required by the FCC or system standard.
ECE 5520 Fall 2009 Periodicity Time ContinuousTime Periodic Aperiodic Laplace Transform x(t) ↔ X(s) Fourier Transform x(t) ↔ X(jω) ∞ X(jω) = t=−∞ x(t)e−jωt dt ∞ 1 x(t) = 2π ω=−∞ X(jω)ejωt dω zTransform x(n) ↔ X(z) Discrete Time Fourier Transform (DTFT) x(n) ↔ X(ejΩ ) −jΩn X(ejΩ ) = ∞ n=−∞ x(n)e π 1 x(n) = 2π n=−π X(ejΩ )dΩ
17
Fourier Series x(t) ↔ ak DiscreteTime Discrete Fourier Transform (DFT) x(n) ↔ X[k] 2π N −1 X[k] = n=0 x(n)e−j N kn 2π N −1 1 x(n) = N k=0 X[k]ej N nk
Table 1: Frequency Transforms Intuitively, bandwidth is the maximum extent of our signal’s frequency domain characterization, call it X(f ). A baseband signal absolute bandwidth is often deﬁned as the W such that X(f ) = 0 for all f except for the range −W ≤ f ≤ W . Other deﬁnitions for bandwidth are • 3dB bandwidth: B3dB is the value of f such that X(f )2 = X(0)2 /2. • 90% bandwidth: B90% is the value which captures 90% of the energy in the signal:
B90% −B90% ∞ =−∞
X(f )2 df = 0.90
Xd(f )2 df
As a motivating example, I mention the squareroot raised cosine (SRRC) pulse, which has the following desirable Fourier transform: √ 0 ≤ f  ≤ 1−α Ts , 2Ts πTs Ts 1−α 1−α (6) HRRC (f ) = 1 + cos α f  − 2Ts , 2Ts ≤ f  ≤ 1+α 2 2Ts 0, o.w.
where α is a parameter called the “rolloﬀ factor”. We can actually analyze this using the properties of the Fourier transform and many of the standard transforms you’ll ﬁnd in a Fourier transform table. The SRRC and other pulse shapes are discussed in Appendix A, and we will go into more detail later on. The purpose so far is to motivate practicing up on frequency transforms.
5.1
Continuoustime Frequency Transforms
Notes about continuoustime frequency transforms: 1. You are probably most familiar with the Laplace Transform. To convert it to the Fourier tranform, we replace s with jω, where ω is the radial frequency, with units radians per second (rad/s).
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2. You may prefer the radial frequency representation, but also feel free to use the rotational frequency f (which has units of cycles per sec, or Hz. Frequency in Hz is more standard for communications; you should use it for intuition. In this case, just substitute ω = 2πf . You could write X(j2πf ) as the notation for this, but typically you’ll see it abbreviated as X(f ). Note that the deﬁnition of the Fourier tranform 1 in the f domain loses the 2π in the inverse Fourier transform deﬁnition.s X(j2πf ) = x(t) =
∞ t=−∞ ∞ f =−∞
x(t)e−j2πf t dt X(j2πf )ej2πf t df (7)
3. The Fourier series is limited to purely periodic signals. Both Laplace and Fourier transforms are not limited to periodic signals. 4. Note that ejα = cos(α) + j sin(α). See Table 2.4.4 in the Rice book. Example: Square Wave Given a rectangular pulse x(t) = rect(t/Ts ), x(t) = 1, −Ts /2 < t ≤ Ts /2 0, o.w.
What is the Fourier transform X(f )? Calculate both from the deﬁnition and from a table. Solution: Method 1: From the deﬁnition:
Ts /2
X(jω) =
t=−Ts /2
e−jωt dt
Ts /2 t=−Ts /2
=
1 e−jωTs /2 − ejωTs /2 = −jω sin(ωTs /2) sin(ωTs /2) = 2 = Ts ω ωTs /2
1 This uses the fact that −2j e−jα − ejα = sin(α). While it is sometimes convenient to replace sin(πx)/(πx) with sincx, it is confusing because sinc(x) is sometimes deﬁned as sin(πx)/(πx) and sometimes deﬁned as (sin x)/x. No standard deﬁnition for ‘sinc’ exists! Rather than make a mistake because of this, the Rice book always writes out the expression fully. I will try to follow suit. Method 2: From the tables and properties:
1 −jωt e −jω
x(t) = g(2t) where g(t) =
1, t < Ts 0, o.w.
(8)
1 From the table, G(jω) = 2Ts sin(ωTs ) . From the properties, X(jω) = 2 G j ω . So ωTs 2
X(jω) = Ts
sin(ωTs /2) ωTs /2
It says is that you can go backwards on the Fourier transform. Duality property: x(jω) = F {X(−t)} x(−jω) = F {X(t)} (Confusing.4.1. Question: What if Y (jω) was a rect function? What would the inverse Fourier transform y(t) be? 5. 20 log10 (X(j2πf )/Ts ).ECE 5520 Fall 2009 19 See Figure 4(a). Important properties of the Fourier transform: 1. just remember to ﬂip the result around the origin. and (b) Power vs.) 2.3 in the Rice book. Ts Ts sinc(π f Ts) Ts/2 0 (a) −4/Ts −3/Ts −2/Ts −1/Ts 0 f 1/Ts 2/Ts 3/Ts 4/Ts 0 −10 (X(f)/T ) 20 log s −20 10 −30 −40 (b) −50 −4/Ts −3/Ts −2/Ts −1/Ts 0 f 1/Ts 2/Ts 3/Ts 4/Ts Figure 4: (a) Fourier transform X(j2πf ) of rect pulse with period Ts . Time shift property: F {x(t − t0 )} = e−jωt0 X(jω) . Assume that F {x(t)} = X(jω).1 Fourier Transform Properties frequency See Table 2.
5. Convolution property: if. jω 1 + jω Lecture 3 Today: (1) Bandpass Signals (2) Sampling Solution: From the Examples at the end of Lecture 2: . additionally y(t) has Fourier transform X(jω). ∞ ∞ ∞ 1 X(jω)2 dω X(f )2 df = x(t)2 dt = 2π ω=−∞ f =−∞ t=−∞ So do whichever one is easiest! Or. check your answer by doing both.3 Examples Example: Applying FT Properties If w(t) has the Fourier transform W (jω) = ﬁnd X(jω) for the following waveforms: 1. Y (jω) = X(jω) · H(jω) 5. F {x(at)} = ω 1 X j a a 4. Scaling property: for any real a = 0. Modulation property: 1 1 F {x(t)cos(ω0 t)} = X(ω − ω0 ) + X(ω + ω0 ) 2 2 6. the output signal is y(t) = h(t) ⋆ x(t) Using the above convolution property.2 Linear Time Invariant (LTI) Filters If a (deterministic) signal x(t) is input to a LTI ﬁlter with impulse response h(t). Parceval’s theorem: The energy calculated in the frequency domain is equal to the energy calculated in the time domain. F {x(t) ⋆ y(t)} = X(jω) · Y (jω) 5.ECE 5520 Fall 2009 20 3. x(t) = e−jt w(t − 1) 3. x(t) = w(1 − t) Solution: To be worked out in class. x(t) = w(2t + 2) 2. x(t) = 2 ∂w(t) ∂t 4.
ECE 5520 Fall 2009 21 1. Let x(t) = z(2t) where z(t) = w(t + 2). and X(jω) = ejω R(jω). X(jω) = 2jω 1+jω = − 1+jω . X(jω) will not be exactly zero outside of the bandwidth. Equivalently. X(jω) = 1 ejω 1+jω/2 . jω 2ω 3. 1 + jω 2 jω/2 So X(jω) = 1 ejω 1+jω/2 . 2 2. Alternatively. Let x(t) = e−jt z(t) where z(t) = w(t − 1). where y(t) = w(1 + t). let x(t) = r(t + 1) where r(t) = w(2t). jω 1 + jω 6 Bandpass Signals Def ’n: Bandpass Signal A bandpass signal x(t) has Fourier transform X(jω) = 0 for all ω such that ω ± ωc  > W/2. and X(jω) = Z(jω/2). 2 jω/2 Again. 2 4. a bandpass signal x(t) has Fourier transform X(f ) = 0 for all f such that f ± fc  > B/2. . Then 2 1 R(jω) = W (jω/2). Then Z(ω) = e−jω W (jω). So j(ω+1) X(jω) = e−j(ω−1) W (j(ω − 1)) = e−j(ω+1) 1+j(ω+1) . where B/2 < fc . there will be some ‘sidelobes’ out of the main band. 6. as shown in Fig. Then Z(jω) = ej2ω 1 jω . B X(f) B fc fc Figure 5: Bandpass signal with center frequency fc .1 Upconversion We can take a baseband signal xC (t) and ‘upconvert’ it to be a bandpass ﬁlter by multiplying with a cosine at the desired center frequency. and X(jω) = Z(j(ω − 1)). Then X(jω) = Y (−jω). Let x(t) = y(−t). Realistically. where Y (jω) = ejω W (jω) = ejω −jω So X(jω) = e−jω 1−jω . where W/2 < ωc . 6.
ECE 5520 Fall 2009 22 xC(t) cos(2pfct) Figure 6: Modulator block diagram. Example: Modulated Square Wave We dealt last time with the square wave x(t) = rect(t/Ts ). o. 6.2 Downconversion of Bandpass Signals How will we get back the desired baseband signal xC (t) at the receiver? 1. Multiply (again) by the carrier 2 cos(ωc t). X(f) fc fc Figure 7: Plot of X(j2πf ) for modulated square wave example. we will look at: x(t) = rect(t/Ts ) cos(ωc t) where ωc is the center frequency in radians/sec (fc = ωc /(2π) is the center frequency in Hz). This is shown in Fig. . 8. −1/2 < t ≤ 1/2 rect(t) = 0. 7. Input to a lowpass ﬁlter (LPF) with cutoﬀ frequency ωc .w. Note that I use “rect” as follows: 1. What is X(jω)? Solution: X(jω) = F {rect(t/Ts )} ⋆ F {cos(ωc t)} sin(ωTs /2) 1 Ts ⋆ π [δ(ω − ωc ) + δ(ω + ωc )] X(jω) = 2π ωTs /2 Ts sin[(ω − ωc )Ts /2] Ts sin[(ω + ωc )Ts /2] X(jω) = + 2 (ω − ωc )Ts /2 2 (ω + ωc )Ts /2 The plot of X(j2πf ) is shown in Fig. 2. This time.
. 7 Sampling A common statement of the Nyquist sampling theorem is that a signal can be sampled at twice its bandwidth. 2πB(t − nT ) (9) Proof: Not covered. (Does the LPF need to be ideal?) X2 (jω) = XC (jω) So x2 (t) = xC (t). . Then xc (t) = 2BT ∞ n=−∞ xc (nT ) sin(2πB(t − nT )) . • Given {xn }∞ n=−∞ . where T ≥ 2B to yield the ∞ sequence {xc (nT )}n=−∞ . Let xc (t) be sampled at multiples of T . continuous signal with bandwidth B (in 1 Hz).e.ECE 5520 Fall 2009 23 x(t) 2cos(2pfct) LPF Figure 8: Demodulator block diagram. Theorem: (Nyquist Sampling Theorem. Assume the general modulated signal was.) Let xc (t) be a baseband. i. But the theorem really has to do with signal reconstruction from a sampled signal. how would you ﬁnd the maximum of x(t)? • This is only precise when X(jω) = 0 for all ω ≥ 2πB. x(t) = xC (t) cos(ωc t) 1 1 X(jω) = XC (j(ω − ωc )) + XC (j(ω + ωc )) 2 2 What happens after the multiplication with the carrier in the receiver? Solution: X1 (jω) = X1 (jω) = 1 X(jω) ⋆ 2π [δ(ω − ωc ) + δ(ω + ωc )] 2π 1 1 XC (j(ω − 2ωc )) + XC (jω) + 2 2 1 1 XC (jω) + XC (j(ω + 2ωc )) 2 2 There are components at 2ωc and −2ωc which will be canceled by the LPF. Notes: • This is an interpolation procedure. Xc (jω) = 0 for all ω ≥ 2πB.
The Fourier transform of the sampled signal is many copies of X(jω) strung at integer multiples of 2π/T . 9. Also consider a parabola pulse (this doesn’t . Example: Sinusoid sampled above and below Nyquist rate Consider two sinusoidal signals sampled at 1/T = 25 Hz: x1 (nT ) = sin(2π5nT ) x2 (nT ) = sin(2π20nT ) What are the two frequencies of the sinusoids. and indicate what the spectrum is of the sampled signals. round pulse shape Consider the square pulse considered before.12. compared to Figure 10(a)? Example: Square vs. sampling is the multiplication of a impulse train (at period T with the desired signal x(t): xsa (t) = x(t) xsa (t) = What is the Fourier transform of xsa (t)? Solution: In the frequency domain.1 Aliasing Due To Sampling ∞ n=−∞ Essentially. as shown in Fig. this is a convolution: Xsa (jω) = X(jω) ⋆ = = 2π T ∞ n=−∞ ∞ n=−∞ δ(t − nT ) x(nT )δ(t − nT ) 2π T ∞ n=−∞ δ ω− 2πn T 2πn T for all ω (10) X j ω− 1 X(jω) T for ω < 2πB This is shown graphically in the Rice book in Figure 2. x1 (t) = rect(t/Ts ). Figure 10 shows what happens when the Nyquist theorem is applied to the each signal (whether or not it is valid). What observations would you make about Figure 10(b). and what is the Nyquist rate? Which of them does the Nyquist theorem apply to? Draw the spectrums of the continuous signals x1 (t) and x2 (t). Figure 9: The eﬀect of sampling on the frequency spectrum in terms of frequency f in Hz.ECE 5520 Fall 2009 24 7.
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1.5 1 0.5
Sampled Interp.
x(t)
0 −0.5 −1 −1 −0.5
(a)
1.5 1 0.5
0 Time
0.5
1
Sampled Interp.
x(t)
0 −0.5 −1 −1 −0.5
(b)
0 Time
0.5
1
Figure 10: Sampled (a) x1 (nT ) and (b) x2 (nT ) are interpolated (—) using the Nyquist interpolation formula. really exist in the wild – I’m making it up for an example.) x2 (t) = 1− 0,
2t Ts 2 1 , − 2Ts ≤ t ≤ 1 2Ts
o.w.
What happens to x1 (t) and x2 (t) when they are sampled at rate T ? In the Matlab code EgNyquistInterpolation.m we set Ts = 1/5 and T = 1/25. Then, the sampled pulses are interpolated using (9). Even though we’ve sampled at a pretty high rate, the reconstructed signals will not be perfect representations of the original, in particular for x1 (t). See Figure 11.
7.2
Connection to DTFT
Recall (10). We calculated the Fourier transform of the product by doing the convolution in the frequency domain. Instead, we can calculate it directly from the formula.
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1 0.8
Sampled Interp.
x(t)
0.6 0.4 0.2 0 −0.2 −1 −0.5 0 Time 0.5 1
(a)
1 0.8
Sampled Interp.
x(t)
0.6 0.4 0.2 0 −0.2 −1 −0.5 0 Time 0.5 1
(b)
Figure 11: Sampled (a) x1 (nT ) and (b) x2 (nT ) are interpolated (—) using the Nyquist interpolation formula. Solution: Xsa (jω) = = =
∞ ∞
t=−∞ n=−∞ ∞ n=−∞ ∞ n=−∞
x(nT )δ(t − nT )e−jωt
∞ t=−∞
x(nT )
δ(t − nT )e−jωt
(11)
x(nT )e−jωnT
The discretetime Fourier transform of xsa (t), I denote DTFT {xsa (t)}, and it is: DTFT {xsa (t)} = Xd (ejΩ ) =
∞ n=−∞
x(nT )e−jΩn
Essentially, the diﬀerence between the DTFT and the Fourier transform of the sampled signal is the relationship Ω = ωT = 2πf T
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But this deﬁnes the relationship only between the Fourier transform of the sampled signal. How can we Ω relate this to the FT of the continuoustime signal? A: Using (10). We have that Xd (ejω ) = Xsa 2πT . Then plugging into (10), Solution: Xd (ejΩ ) = Xsa j = = 2π T 2π T
∞ n=−∞ ∞ n=−∞
Ω T X j X j Ω 2πn − T T Ω − 2πn T (12)
If x(t) is suﬃciently bandlimited, Xd (ejΩ ) ∝ X(jΩ/T ) in the interval −π < Ω < π. This relationship holds for the Fourier transform of the original, continuoustime signal between −π < Ω < π if and only if the original signal x(t) is suﬃciently bandlimited so that the Nyquist sampling theorem applies. Notes: • Most things in the DTFT table in Table 2.8 (page 68) are analogous to continuous functions which are not bandlimited. So  don’t expect the last form to apply to any continuous function. • The DTFT is periodic in Ω with period 2π. • Don’t be confused: The DTFT is a continuous function of Ω.
Lecture 4 Today: (1) Example: Bandpass Sampling (2) Orthogonality / Signal Spaces I
7.3
Bandpass sampling
Bandpass sampling is the use of sampling to perform downconversion via frequency aliasing. We assume that the signal is truly a bandpass signal (has no DC component). Then, we sample with rate 1/T and we rely on aliasing to produce an image the spectrum at a relatively low frequency (less than onehalf the sampling rate). Example: Bandpass Sampling This is Example 2.6.2 (pp. 6065) in the Rice book. In short, we have a bandpass signal in the frequency band, 450 Hz to 460 Hz. The center frequency is 455 Hz. We want to sample the signal. We have two choices: 1. Sample at more than twice the maximum frequency. 2. Sample at more than twice the bandwidth. The ﬁrst gives a sampling frequency of more than 920 Hz. The latter gives a sampling frequency of more than 20 Hz. There is clearly a beneﬁt to the second part. However, traditionally, we would need to downconvert the signal to baseband in order to sample it at the lower rate. (See Lecture 3 notes).
It is also determined by orthogonality. Sample at 1000 samples/sec: Ωc = 2π × 10 Hz 2π = π/50 radians/sample. x . 1000 samples/sec? 2. 1000 samples/sec 455 500 = 455 500 π. This is the critical concept needed to understand digital receivers. This choice of waveforms is partially determined by bandwidth. Sample at 140 samples/sec: Copies of the entire frequency spectrum will appear in the frequency domain at multiples of 140 Hz or 2π140 radians/sec. and what is the bandwidth of the sampled signal in radians/sample if the sampling rate is: 1. x = x. the idea is to determine which waveforms were sent and in what quantities. we talked about how digital communications is largely about using (and choosing some combination of) a discrete set of waveforms to convey information on the (continuoustime. y= .1 Inner Product of Vectors Then we can take their inner product as: We often use an idea called inner product. Finally. The bandwidth is 2π 140 = π/7 radians/sample of sampled spectrum. so the center frequency of the sampled signal is: Ωc = 2π × 455 140 mod 2π = 13π 2 mod 2π = π 2 10 The value π/2 radians/sample is equivalent to 35 cycles/second. which we have covered. 8. the norm of a vector is the square root of the inner product of a vector with itself. . . 8 Orthogonality From the ﬁrst lecture. If we have two vectors x and y. y . orthogonality of waveforms means that some combination of the waveforms can be uniquely separated at the receiver. x1 y1 x2 y2 x= . The inner product is also denoted x. . 140 samples/sec? Solution: 1. xn yn xT y = i xi y i Note the T transpose ‘inbetween’ an inner product.ECE 5520 Fall 2009 28 What is the sampled spectrum. . To build a better framework for it. The signal is very sparse . we’ll start with something you’re probably more familiar with: vector multiplication. which you’re familiar with from vectors (as the dot product).it occupies only 2. realvalued) channel. The discretetime frequency is modulo 2π. Loosely. At the receiver.
. those functions x(t) for which ∞ −∞ ∞ −∞ x2 (t)dt < ∞. when zero. y = [0. 0. These norms are: x(t). 1]T .w. they are pointing in somewhat opposite directions. o. x(t) x(t) = = ∞ −∞ ∞ −∞ = = ∞ −∞ ∞ −∞ x(t)y(t)dt x(t)y ∗ (t)dt Real functions Complex functions x2 (t)dt x(t)2 dt Real functions Complex functions Example: Sine and Cosine Let x(t) = y(t) = cos(2πt). .e. o. 0]T .e. z = [1.. The ‘space’ of squareintegrable complex functions. In words. i. they’re at crossdirections. It also is applied to the ‘space’ of squareintegrable real functions.ECE 5520 Fall 2009 · 29 symbols outside of the vector. the two vectors are pointing in a similar direction. y ? • x ? • z ? Recall the inner product between two vectors can also be written as xT y = x y cos θ where θ is the angle between vectors x and y. when negative. A norm is always indicated with the Example: Example Inner Products Let x = [1. What are: • x. 1. y ? • z. 0 < t ≤ 1 0. i. those functions x(t) for which x(t)2 dt < ∞ also have an inner product. 1. y(t) x(t). y(t) As a result.2 Inner Product of Functions The inner product is not limited to vectors. sin(2πt). 8. 0 < t ≤ 1 0.w. the inner product is a measure of ‘samedirectionness’: when positive. 1]T .
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What are x(t) and y(t) ? What is x(t), y(t) ?
1 Solution: Using cos2 x = 2 (1 + cos 2x), 1
x(t)
=
0
cos2 (2πt)dt =
1 2
1+
1 sin(2πt) dt 2π
1
=
0
1/2
The solution for y(t) also turns out to be x(t), y(t) =
1/2. Using sin 2x = 2 cos x sin x, the inner product is,
1
cos(2πt) sin(2πt)dt
0 1
=
0
1 sin(4πt)dt 2
1
=
−1 cos(4πt) 8π
=
0
−1 (1 − 1) = 0 8π
8.3
Deﬁnitions
Def ’n: Orthogonal Two signals x(t) and y(t) are orthogonal if x(t), y(t) = 0 Def ’n: Orthonormal Two signals x(t) and y(t) are orthonormal if they are orthogonal and they both have norm 1, i.e., x(t) = 1 y(t) = 1
(a) Are x(t) and y(t) from the sine/cosine example above orthogonal? (b) Are they orthonormal? √ Solution: (a) Yes. (b) No. They could be orthonormal if they’d been scaled by 2.
x1(t) A
x2(t) A
A
A
Figure 12: Two WalshHadamard functions. Example: WalshHadamard 2 Functions Let 0 < t ≤ 0.5 A, −A, 0.5 < t ≤ 1 x1 (t) = 0, o.w. A, 0 < t ≤ 1 0, o.w.
x2 (t) =
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These are shown in Fig. 12. (a) Are x1 (t) and x2 (t) orthogonal? (b) Are they orthonormal? Solution: (a) Yes. (b) Only if A = 1.
8.4
Orthogonal Sets
Def ’n: Orthogonal Set M signals x1 (t), . . . , xM (t) are mutually orthogonal, or form an orthogonal set, if xi (t), xj (t) = 0 for all i = j. Def ’n: Orthonormal Set M mutually orthogonal signals x1 (t), . . . , xM (t) are form an orthonormal set, if xi = 1 for all i.
(a)
(b)
Figure 13: WalshHadamard 2length and 4length functions in image form. Each function is a row of the image, where crimson red is 1 and black is 1.
Figure 14: WalshHadamard 64length functions in image form. Do the 64 WH functions form an orthonormal set? Why or why not? Solution: Yes. Every pair i, j with i = j agrees half of the time, and disagrees half of the time. So the function is halftime 1 and halftime 1 so they cancel and have inner product of zero. The norm of the function in row i is 1 since the amplitude is always +1 or −1. Example: CDMA and WalshHadamard This example is just for your information (not for any test). The set of 64length WalshHadamard sequences
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is used in CDMA (IS95) in two ways: 1. Reverse Link: Modulation: For each group of 6 bits to be send, the modulator chooses one of the 64 possible 64length functions. 2. Forward Link: Channelization or Multiple Access: The base station assigns each user a single WH function (channel). It uses 64 functions (channels). The data on each channel is multiplied by a Walsh function so that it is orthogonal to the data received by all other users.
Lecture 5 Today: (1) Orthogonal Signal Representations
9
Orthonormal Signal Representations
Last time we deﬁned the inner product for vectors and for functions. We talked about orthogonality and orthonormality of functions and sets of functions. Now, we consider how to take a set of arbitrary signals and represent them as vectors in an orthonormal basis. What are some common orthonormal signal representations? • Nyquist sampling: sinc functions centered at sampling times. • Fourier series: complex sinusoids at diﬀerent frequencies • Sine and cosine at the same frequency • Wavelets And, we will come up with others. Each one has a limitation – only a certain set of functions can be exactly represented in a particular signal representation. Essentially, we must limit the set of possible functions to a set. That is, some subset of all possible functions. We refer to the set of arbitrary signals as: S = {s0 (t), s1 (t), . . . , sM −1 (t)} For example, a transmitter may be allowed to send any one of these M signals in order to convey information to a receiver. We’re going to introduce another set called an orthonormal basis: B = {φ0 (t), φ1 (t), . . . , φN −1 (t)} Each function in B is called a basis function. You can think of B as an unambiguous and useful language to represent the signals from our set S. Or, in analogy to color printers, a color printer can produce any of millions of possible colors (signal set), only using black, red, blue, and green (basis set) (or black, cyan, magenta, yellow).
it can be represented as a linear combination of the basis functions. 9. An orthonormal basis for an arbitrary signal set tells us: • how to build the receiver. (As the WalshHadamard functions are used in IS95 reverse link).. The signal set might be a set of signals we can use to transmit particular bits (or sets of bits). The value of N is called the dimension of the signal set. .1 Orthonormal Bases Def ’n: Span The span of the set B is the set of all functions which are linear combinations of the functions in B.. . si (t). Then the signal is equal to the sum of its projections onto the basis functions. φk (t) φk (t). .ECE 5520 Fall 2009 33 9. . you can use.aN−1 ∈R Def ’n: Orthogonal Basis For any arbitrary set of signals S = {si (t)}M −1 . for which si (t) ∈ Span{B} for every i = 0.k = si (t). The span is referred to as Span{B} and is Span{B} = N −1 k=0 ak φk (t) a0 . If you can ﬁnd one basis. Notes: N −1 • The orthonormal basis is not unique. Given that it is in the span of our basis B. si (t) = N −1 k=0 ai. and a larger set would not be a basis by deﬁnition. φk (t) The projection of one signal i onto basis function k is deﬁned as ai.2 Synthesis Consider one of the signals in our signal set.k are calculated using the inner product: ai. • All orthogonal bases will have size N : no smaller basis can include all signals si (t) in its span. i=0 N −1 B = {φi (t)}i=0 . Def ’n: Orthonormal Basis The orthonormal basis is an orthogonal basis in which each basis function has norm 1. {−φk (t)}k=0 as another orthonormal basis. for example.. M − 1. • how to represent the signals in ‘signalspace’. Why is this? .k φk (t) The particular constants ai. and • how to quickly analyze the error rate of the scheme..k φk (t) = si (t). the orthogonal basis is an orthogonal set of the smallest size N .
1 . Example: Positionshifted pulses Plot the signal space diagram for the signals. By choosing one of the M signals. speciﬁcally. . and s2 = [1.5 in Rice (page 254). this space that we’re plotting in is called signal space.ECE 5520 Fall 2009 N Solution: Since si (t) ∈ Span{B}.k φk (t) N −1 k=0 N −1 k=0 ai. .k φk (t). s0 . 0]T . and s2 ? Solution: They are s0 = [1. φj (t) = ai. See Figure 5. we know there are some constants {ai. Energy: • Energy can be calculated in signal space as Energy{si (t)} = −∞ φ1 (t) = u(t − 1) − u(t − 2) ∞s2 (t)dt = i N −1 k=0 a2 i. now we can now represent a signal by a vector.N −1 ]T This and the basis functions completely represent each signal. φj (t) = ai.j So. log2 M bits of information.k }k −1 such that 34 si (t) = Taking the inner product of both sides with φj (t).k Proof? . which shows a block diagram of how a transmitter would synthesize one of the M signals to send. ai. φj (t) si (t). 1]T . We can also synthesize any of our M signals in the signal set by adding the proper linear combination of the N bases. si (t). s1 = [0. s0 (t) = u(t) − u(t − 1) s1 (t) = u(t − 1) − u(t − 2) s2 (t) = u(t) − u(t − 2) given the orthonormal basis. φj (t) = N −1 k=0 ai. φj (t) si (t). ai. s1 . we choose M to be a power of 2 for simplicity).0 . Plotting {si }i in an N dimensional grid is termed a constellation diagram. . (Generally. φ0 (t) = u(t) − u(t − 1) What are the signal space vectors. Generally. They are plotted in the signal space diagram in Figure 15. we convey information. based on an input bitstream. 1]T .k φk (t). . si = [ai.
our job will be to analyze the received signal (a function) and to decide which of the M possible signals was sent. • We will ﬁnd out later that it is the distances between the points in signal space which determine the bit error rate performance of the receiver. M − 1}. s2 = [−0. respectively.k − aj. s1 = [0. • Although diﬀerent ON bases can be used.25.3 Analysis At a receiver. . Example: Amplitudeshifted signals Now consider s0 (t) = 1. and 2. .25. . It turns out that an orthonormal bases makes our analysis very straightforward and elegant. j in {0. Energies are just the squared magnitude of the vector: 2. 0.25.there will be additional functions added to the signal function we sent. .5[u(t) − u(t − 1)] N −1 k=0 (ai. What are the signal space vectors for the signals {si (t)}? What are their energies? Solution: s0 = [1.5]. This is the task of analysis. See Figure 16.5[u(t) − u(t − 1)] s3 (t) = −1.5]. 9.5].j = for i. 1 0 1 f1 Figure 16: Signal space diagram for amplitudeshifted signals example. We won’t receive exactly what we sent .5]. the energy and distance between points will not change.ECE 5520 Fall 2009 35 f2 1 f1 1 Figure 15: Signal space diagram for positionshifted signals example.k )2 s1 (t) = 0. .25.5[u(t) − u(t − 1)] s2 (t) = −0.5[u(t) − u(t − 1)] φ1 (t) = u(t) − u(t − 1) and the orthonormal basis. s3 = [−1. di. 0.
. and the synthesis equation is r (t) = ˆ N −1 k=0 xk φk (t) If you plug in the above expression for r (t) into (13). what is r (t) ∈ Span{B} such that the energy of the ˆ diﬀerence between r(t) and r(t) is minimized. ˆ x = [x0 . so r(t) would not be in Span{B} either. But w(t) might not (probably not) be in the span of our basis B. . .e. ˆ argmin r (t)∈Span{B} ˆ ∞ −∞ r ˆ(t) − r(t)2 dt (13) Solution: Since r(t) ∈ Span{B}. . i. x1 . . you’d ˆ see that the minimum error is at ∞ r(t)φk (t)dt xk = −∞ that is.. N . But. Example: Analysis using a WalshHadamard 2 Basis See Figure 17. we might say that if we send signal m. xk = r(t). . it can be represented as a vector in signal space. What is r(t)? ˆ Solution: 0≤t<1 1.w. . Let s0 (t) = φ0 (t) and s1 (t) = φ1 (t).ECE 5520 Fall 2009 • Thermal noise • Interference from other users • Selfinterference 36 We won’t get into these problems today. 1 ≤ t < 2 o. Lecture 6 Today: (1) Multivariate Distributions . r (t) = ˆ −1/2. φk (t) . i. xN −1 ]T ..e. 0. What is the best approximation to r(t) in the signal space? Speciﬁcally. and then ﬁnd the minimum with respect to each xk . then we would receive r(t) = sm (t) + w(t) where the w(t) is the sum of all of the additive signals that we did not intend to receive. for k = 0. . sm (t) from our signal set.
x2 ) = 0 and limx1 .X2 (x1 .X2 (x1 . For example.X2 )∈B PX1 . x2 ) = 1. x2 ) The pdf and pmf integrate / sum to one. Eqn 4. and are nonnegative. 10 MultiVariate Distributions For two random variables X1 and X2 .1/2 r(t) 2 1 2 2 f2(t) 1/2 1 .x2 )∈B . • Joint pdf: fX1 .X2 (x1 . you integrate.X2 (x1 . Note: Two errors in Rice book dealing with the deﬁnition of the CDF.X2 (x1 .−∞ FX1 .9 should be: x FX (x) = P [X ≤ x] = and Eqn 4.15 should be: fX (t)dt −∞ FX (x) = P [X ≤ x] To ﬁnd the probability of an event.ECE 5520 Fall 2009 37 f1(t) 1/2 1 . The CDF is nonnegative and nondecreasing. • Discrete case: P [B] = • Continuous Case: P [B] = (X1 . x2 ) = ∂2 ∂x1 ∂x2 FX1 .1/2 r(t) 2 1 2 2 2 1 2 Figure 17: Signal and basis functions for Analysis example. x2 ) = P [{X1 = x1 } ∩ {X2 = x2 }] It is the probability that both events happen simultaneously. x2 )dx1 dx2 (x1 . x2 ) = P [{X1 ≤ x1 } ∩ {X2 ≤ x2 }] It is the probability that both events happen simultaneously.X2 (x1 . • Joint pmf: PX1 . x2 ) fX1 . with limxi →.+∞ FX1 .x2 →. • Joint CDF: FX1 .X2 (x1 .X2 (x1 . for event B ∈ S.
2. X2 ) ∈ B o. where fX2 (x2 ) > 0. x2 ) = • Continuous Case: fX1 .X2 (x1 . X is FX (x) = FX1 . x2 )dx1 Two random variables X1 and X2 are independent iﬀ for all x1 and x2 . P [B] (X1 . the joint probability conditioned on event B is • Discrete case: PX1 . . . Xn .. . .V. . . .X2 (x1 . . X is fX (x) = fX1 . ..2 Conditional Distributions Given event B ∈ S which has P [B] > 0. .) is a list of multiple random variables X1 .Xn (x1 .X2 (x1 .X2 (x1 . . X2 . . where PX2 (x2 ) > 0. . xn ) = P [X1 = x1 . x2 ) = Given r.. 0. The pdf of a continuous R. The CDF of R.. . X = [X1 . xn ) = ∂n ∂x1 ···∂xn FX (x). . . .. (X1 .1 Random Vectors Def ’n: Random Vector A random vector (R..w.Xn (x1 . x2 ) = fX2 (x2 )fX1 (x1 ) 10. . . The pmf of a discrete R.V.. • PX1 .V. P [B] 0. . Xn = xn ].s: 1.X2 B (x1 . 10. . X2 ) ∈ B o.V. . is fX1 X2 (x1 x2 ) = fX1 . . x2 ) fX1 ... .V. Xn ]T Here are the Models of R. The conditional pmf of X1 given X2 = x2 .X2 (x1 .s X1 and X2 . x2 ) = PX1 (x1 )PX2 (x2 ) • fX1 .v.X2 (x1 .w. .Xn (x1 . .x2 ) . .X2 (x1 . . xn ) = P [X1 ≤ x1 . . x2 )/fX2 (x2 ) PX1 . 3.X2 B (x1 .. X is PX (x) = PX1 . X2 .X2 (x1 . Xn ≤ xn ]. x2 )/PX2 (x2 ) • Continuous Case: The conditional pdf of X1 given X2 = x2 ..x2 ) .ECE 5520 Fall 2009 38 The marginal distributions are: • Marginal pmf: PX2 (x2 ) = • Marginal pdf: fX2 (x2 ) = x1 ∈SX1 x1 ∈SX1 PX1 . • Discrete case. is PX1 X2 (x1 x2 ) = PX1 .. fX1 .
1.ECE 5520 Fall 2009 Def ’n: Bayes’ Law Bayes’ Law is a reformulation of he deﬁnition of the marginal pdf..). What type of random variable is Ei ? Simulations run many bits. with pmf PS (s) = N s p (1 − pe )N −s s e The mean of S is the the same as the mean of the sum of {Ei }. and count the number of bits that are in error.v. Thus the simulation of one bit is a Bernoulli trial. or with error. This trial Ei is in error (E1 = 1) with probability pe (the true bit error rate) and correct (Ei = 0) with probability 1 − pe .i. Each bit can either be demodulated without error.d. x2 ) = fX2 X1 (x2 x1 )fX1 (x1 ) or fX1 X2 (x1 x2 ) = fX2 X1 (x2 x1 )fX1 (x1 ) fX2 (x2 ) 39 10. It is written either as: fX1 . N N ES [S] = E{Ei } N Ei = i=1 i=1 EEi [Ei ] = i=1 [(1 − p) · 0 + p · 1] = N p We can ﬁnd the variance of S the same way: N N VarS [S] = Var{Ei } N Ei = i=1 i=1 VarEi [Ei ] = i=1 N (1 − p) · (0 − p)2 + p · (1 − p)2 (1 − p)p2 + (1 − p)(p − p2 ) = i=1 = N p(1 − p) . What type of random variable is S? 2. and S is called a Binomial r. Let S = N Ei . What is the mean and variance of S? Solution: Ei is called a Bernoulli r. say N bits through a model of the communication system. and assume that {Ei } are independent and identically distributed i=1 (i. What is the pmf of S? 3.X2 (x1 .v.3 Simulation of Digital Communication Systems A simulation of a digital communication system is often used to estimate a bit error rate.
X1 may take a discrete set of values. For example. What type of random variable is T1 ? 2. What is the pmf of T1 ? 3. Workaround: We will sometimes use a pdf for a discrete 0.5. our estimate of pe will have relatively high variance. Example: Digital communication system in noise Let X1 is the transmitted signal voltage.v. the probability that −0.ECE 5520 Fall 2009 40 We may also be interested knowing how many bits to run in order to get an estimate of the bit error rate. if x1 = 0 x1 = 1 o. and a is a constant which represents the attenuation of the channel. Gaussian with mean µ and variance σ 2 . 0. ..5. then X2 will be continuousvalued. then we would write a pdf with the probabilities as amplitudes of a dirac delta function centered at the value that has that probability. But the noise N may be continuous.5 < x1 < 0. which is modeled as X2 = aX1 + N where N is a continuousvalued random variable representing the additive noise of the channel. For instance. 0. 1. we won’t have a very good idea of the bit error rate.4 Mixed Discrete and Continuous Joint Variables This was not covered in ECE 5510.4. For example. −0. e.g. although it was in the Yates & Goodman textbook. −1. In a digital system.g. with pmf PT1 (t) = (1 − pe )t−1 pe The mean of T1 is E [T1 ] = 1 pe Note the variance of T1 is Var [T1 ] = (1 − pe )/p2 .5. Let T1 be the time (number of bits) up to and including the ﬁrst error. PX1 (x1 ) = 0. e.6δ(x1 ) + 0. We’ll often have X1 discrete and X2 continuous..w. random variable. As long as σ 2 > 0.6. so the standard deviation for very low pe is almost the same e as the expected value. if we run a simulation and get zero bit errors. even if we run an experiment until the ﬁrst bit error. Any probability integral will return the proper value.6. 10. What is the mean of T1 ? Solution: T1 is a Geometric r.5 would be an integral that would return 0.5}. {1. and nonnegative values for all x1 . and X2 is the received signal voltage. fX1 (x1 ) = 0. So.4δ(x1 − 1) This pdf has integral 1.
the pdf of the sum is the convolution of the pdfs of the inputs.5fN (x2 ) + 0. o. 0.5δ(x1 − 1) we convolve this with fN (n) above. . to get fX2 (x2 ) = 0.w. x2 ) = 0.v. What is the joint p.s are added.5fN (x2 ).d.X2 (x1 . When two independent r. 0. o. fX1 . and fN (n) = √ 1 2πσ 2 e−n 2 /(2σ 2 ) where σ 2 is some known variance. X2 ) ? Solution: 1. X2 ) ? Since X1 and X2 are NOT independent.f. of (X1 . x1 = 1 .5fN (x2 )δ(x1 ) + 0.X2 (x1 . of (X1 .v. x1 = 0 0. What is the pdf of X2 ? 2.X2 (x1 . x2 ) = down the pdf of X2 .w. x2 ) = fX1 .5fN (x2 − 1)δ(x1 − 1) These last two lines are completely equivalent.w. x1 = 1 0.ECE 5520 Fall 2009 41 Example: Joint distribution of X1 . o. and the r. Use whichever seems more convenient for you.f.5. 1. fX1 . Writing fX1 (x1 ) = 0.5fN (x2 − 1).5δ(x1 ) + 0. x1 = 1 0. x2 ) = fX2 X1 (x2 x1 )fX1 (x1 ) Given a value of X1 (either 0 or 1) we can write fX1 . x1 = 0 PX1 (x1 ) = 0. It is necessary to use Bayes’ Law. x1 = 0 fX2 X1 (x2 1)fX1 (1).5fN (x2 − 1) 1 2 2 2 2 √ e−x2 /(2σ ) + e−(x2 −1) /(2σ ) fX2 (x2 ) = 2 2 2πσ 2.d.X2 (x1 . See Figure 18.5. we cannot simply multiply the marginal pdfs together. What is the joint p. X2 Consider the channel model X2 = X1 + N . X1 is independent of N with 0. So break this into two cases: fX2 X1 (x2 0)fX1 (0).
X with mean µX and variance σX . 10. X2 )fX1 . • Covariance of X1 and X2 : g(X1 .X2 (x1 . for X. X2 )] = X1 ∈SX1 X1 ∈SX1 X2 ∈SX2 X2 ∈SX2 g(X1 . X2 ) = X1 or g(X1 . fX (x) = 1 2 2πσX e− (x−µX )2 2σ 2 Consider Y to be Gaussian with mean 0 and variance 1. we can write its CDF as FX (x) = P [X ≤ x] = Φ x − µX σX . also called the ‘correlation’ of X1 and X2 : g(X1 .ECE 5520 Fall 2009 42 4 2 0 −1 0 0 X 1 1 −2 2 −4 X 2 Figure 18: Joint pdf of X1 and X2 . X2 )] = Typical functions g(X1 . the input and output (respectively) of the example additive noise communication system. 10. we have the pdf.5 Expectation Def ’n: Expected Value (Joint) The expected value of a function g(X1 .X2 (x1 . X2 ) = (X2 − µX2 )2 . x2 ) g(X1 . which has nonzero mean and nonunitvariance. Continuous: E [g(X1 . • Variance (or 2nd central moment) of X1 or X2 : g(X1 . 2 2 Often denoted σX1 and σX2 . the CDF of Y is denoted as FY (y) = P [Y ≤ y] = Φ(y). X2 ) = X2 will result in the means µX1 and µX2 . x2 ) 2. X2 ) = (X1 − µX1 )2 or g(X1 . X2 ) are: • Mean of X1 or X2 : g(X1 . X2 ) = X1 X2 . 1. • Expected value of the product of X1 and X2 . Then.v. So. X2 ) of random variables X1 and X2 is given by.6 Gaussian Random Variables 2 For a single Gaussian r. Discrete: E [g(X1 . X2 )PX1 . when σ = 1. X2 ) = (X1 − µX1 )(X2 − µX2 ).
so that 1 Φ(y) = erf 2 y √ 2 + 1 2 0 √ 2x . Instead.v.1 Complementary CDF The probability that a unitvariance. 1 − Φ(x). we can write Φ(·) in terms of the erf(·) function. that is.6. we can write the probability of this event using the unitvariance. erf(x) 2 √ 2π = 2 √ 2x e−u 2 /2 du 1 2 √ e−u /2 du 2π 0 √ 1 = 2 Φ( 2x) − 2 Equivalently. there is a function called erf(x) erf(x) 2 √ π x 0 e−t dt 2 Example: Relationship between Q(·) and erf(·) What is the functional relationship between Q(·) and erf(·)? √ √ Solution: Substituting t = u/ 2 (and thus dt = du/ 2). √ 1 1 Φ( 2x) = erf(x) + 2 2 Finally let y = √ 2x. the Q(x) function is not used.6. ∞ 1 2 √ e−w /2 dw 2π P [X > x] = Q 10. Q(x).2 Error Function x − µX σX =1−Φ x − µX σX In math. in some texts. zero mean Gaussian random variable. Q(x) = P [X > x] = 1 − Φ (x) What is Q(x) in integral form? Q(x) = x 2 For an Gaussian r. zero mean Gaussian CDF.v. X exceeds some value x is one minus the CDF. 10. This is so common in digital communications. X with variance σX .ECE 5520 Fall 2009 You can prove this by showing that the event X ≤ x is the same as the event x − µX X − µX ≤ σX σX 43 Since the lefthand side is a unitvariance. and in Matlab. it is given its own name. zero mean Gaussian r.
P [errorX1 = 1] = P [X2 ≤ 1/3] = P 1/3 − 1 X2 − 1 ≤ 1/4 1/4 = 1 − Q(−4/3) = 1 − Q ((−2/3)/(1/2)) 2./sqrt(2)). The receiver decides as follows: • If X2 ≤ 1/3.* erf(y. since X2 = X1 + N .ECE 5520 Fall 2009 44 Or in terms of Q(·). 10.v. Given that X1 = 1.1/2 . Given that X1 = 0. then decide that the transmitter sent a ‘0’.7 Examples Example: Probability of Error in Binary Example As in the previous example. Given that X1 = 0. Then the probability that the receiver decides ‘0’ is the probability that X2 ≤ 1/3. the probability that the receiver decides ‘1’ is the probability that X2 > 1/3. 1. what is the probability that the receiver decides that a ‘1’ was sent? Solution: 1. • If X2 > 1/3. P [errorX1 = 0] = P [X2 > 1/3] = P X2 1/4 > 1/3 1/4 = Q ((1/3)/(1/2)) = Q(2/3) . then decide that the transmitter sent a ‘1’. 1} with equal probabilities and N is independent of X1 and zeromean Gaussian with variance 1/4. Given that X1 = 1. what is the probability that the receiver decides that a ‘0’ was sent? 2. with mean 1 and variance 1/4. it is clear that X2 is also a Gaussian r. we have a model system in which the receiver sees X2 = X1 + N . Here. Q(y) = 1 − Φ(y) = 1 1 − erf 2 2 y √ 2 (14) You should go to Matlab and create a function Q(y) which implements: function rval = Q(y) rval = 1/2 . X1 ∈ {0.
fR (r) = r σ2 +s exp − r 2σ2 2 2 I0 rs σ2 0. X is multivariate Gaussian with mean µX .V.8. which is the dimension with which we spend most of our time in this class. Rician) distribution. the pdf would be the product of the individual pdfs (as with any random vector) and in this case the pdf would be: fX (x) = 1 (2π)n i 2 σi n exp − i=1 (xi − µXi )2 2 2σXi Section 4. r≥0 o. then Y is also a Gaussian random vector with mean AµX and covariance matrix ACX AT .a. This is the Rayleigh distribution.i. and identical variances σ 2 .3 spends some time with 2D Gaussian random vectors. and covariance matrix CX if it has the pdf. 0). X2 ). Speciﬁcally.w. Lecture 7 Today: (1) Matlab Simulation (2) Random Processes (3) Correlation and Matched Filter Receivers . and I0 (·) is the zeroth order modiﬁed Bessel function of the ﬁrst kind. if the vector is (X1 . if we have a matrix A and we let a new random vector Y = AX. o. Gaussian with mean 0 and variance σ 2 . If instead X1 and X2 are independent Gaussian with means µ1 and µ2 . . the pdf of R is fR (r) = r σ2 r exp − 2σ2 . An nlength R. fX (x) = 1 (2π)n det(CX ) 1 −1 exp − (x − µX )T CX (x − µX ) 2 −1 where det() is the determinant of the covariance matrix.V.d.8 Gaussian Random Vectors Def ’n: Multivariate Gaussian R. the envelope R is R= 2 2 X1 + X2 (15) If both X1 and X2 are i. and CX is the inverse of the covariance matrix.w. r ≥ 0 2 0. For example. 2 1 Note that both the Rayleigh and Rician pdfs can be derived from the Gaussian distribution and (15) using the transformation of random variables methods studied in ECE 5510. where s2 = µ2 + µ2 .ECE 5520 Fall 2009 45 10. the envelope R would now have a Rice (a. If the elements of X were independent random variables.k. Any linear combination of jointly Gaussian random variables is another jointly Gaussian random variable. Both pdfs are sometimes needed to derive probability of error formulas. 10.1 Envelope The envelope of a 2D random vector is its distance from the origin (0. respectively.
µX = µX (n) = E [X(n)] is independent of n. and abbreviate it to X(t) or X(n). Consider what happens when a pollster takes either a timeaverage or a ensemble average: . widesense stationary random process from one realization of a powertype signal x(t). respectively.1 Autocorrelation and Power Def ’n: Mean Function The mean function of the random process X(t. A random process is widesense stationary (WSS) if 1. The power of a signal is given by RX (0). its time averages are equivalent to its ensemble averages. you don’t have anything to average to get the mean function µX (t). 2. µX = µX (t) = E [X(t)] is independent of t. k) = E [X(n)X(n − k)] Def ’n: Widesense stationary (WSS) A random process is widesense stationary (WSS) if 1. s) is a function of time t and outcome (realization) s. We then denote the autocorrelation function as RX (k). we omit the “s” when writing the name of the random process or random sequence. s) is µX (t) = E [X(t. Outcome or realization is included because there could be ways to record X even at the same time. τ ) depends only on the time diﬀerence τ and not on t. but we now must deﬁne “ergodic”. Typically. 11. If you record one signal over all time t. We can estimate the autocorrelation function of an ergodic. k) depends only on k and not on n. s)] Note the mean is taken over all possible realizations s. RX (τ ) = lim 1 T →∞ T T /2 t=−T /2 x(t)x(t − τ )dt This is not a critical topic for this class. For example. An example that helps demonstrate the diﬀerence between time averages and ensemble averages is the nonergodic process of opinion polling. 2. A random sequence X(n. multiple receivers would record diﬀerent noisy signals even of the same transmission. s) is a sequence of random variables indexed by time index n and realization s ∈ S. For an ergodic random process. RX (t. We then denote the autocorrelation function as RX (τ ). RX (n.ECE 5520 Fall 2009 46 11 Random Processes A random process X(t. Def ’n: Autocorrelation Function The autocorrelation function of a random process X(t) is RX (t. Outcome s lies in sample space S. τ ) = E [X(t)X(t − τ )] The autocorrelation of a random sequence X(n) is RX (n.
every day for N days. on the same day. but probably not. it is uncorrelated with X(m). i=1 This transmission conveys to us which of M messages we want to send. whether or not she will vote for person X.2) has a good analysis of the physics of thermal noise. SX (f ) = F {RX (τ )} SX (ejΩ ) = DTFT {RX (k)} 11. 12 Correlation and MatchedFilter Receivers We’ve been talking about how our digital transmitter can transmit at each time one of a set of signals {si (t)}M . The Rice book (4. • Ensemble Average: The pollster asks N diﬀerent people. The time average is taken by dividing the total number of “Yes”s by N . At the receiver. 1015 Hz). • Does this mean it is independent of every other sample? • What is the PSD of the sequence? This sequence is commonly called ‘white noise’ because it has equal parts of every frequency (analogy to light).g. whether or not they will vote for person X.1. we receive signal si (t) scaled and corrupted by noise: r(t) = bsi (t) + n(t) How do we decide which signal i was transmitted? .e. Will they come up with the same answer? Perhaps. m = n. The ensemble average is taken by dividing the total number of “Yes”s by N .1 White Noise Let X(n) be a WSS random sequence with autocorrelation function RX (k) = E [X(n)X(n − k)] = σ 2 δ(k) This says that each element of the sequence X(n) has zero covariance with every other sample of the sequence. This makes the process nonergodic. Power Spectral Density We have seen that for a WSS random process X(t) (and for a random sequence X(n) we compute the power spectral density as.. thermal noise is not constant in frequency. log2 M bits of information. For continuous time signals. i. Let X(t) be a WSS random process with autocorrelation function RX (τ ) = E [X(t)X(t − τ )] = σ 2 δ(τ ) • What is the PSD of the sequence? • What is the power of the signal? Realistically. the power spectral density goes to zero.. because as frequency goes very high (e. that is. white noise is a commonly used approximation for thermal noise.ECE 5520 Fall 2009 47 • Time Average: The pollster asks the same person.5.
{φk (t)}N . correlating with the basis functions. φk (t) = for k = 1 . the inner product is a correlation: r(t). . .k + nk First. x = [x1 . Deﬁne nk = n(t). N . Noisy Channel Now.1 Correlation Receiver ∞ −∞ What we’ve been talking about it the inner product. then we can reduce the amount of correlation (and thus multiplication and addition) done in the receiver by instead. if n(t) = 0 and b = 1 then x = ai where ai is the signal space vector for si (t). . .k + n(t)φk (t)dt = ai. shown in Figure 5.1. xN ]T Now that we’ve done these N correlations (inner products).ECE 5520 Fall 2009 48 12. . φk = ∞ n(t)φk (t)dt. If si (t) are nonorthogonal. nk is zero mean: E [nk ] = ∞ −∞ E [n(t)] φk (t)dt = 0 . let b = 1 but consider n(t) to be a white Gaussian random process with zero mean and PSD SN (f ) = N0 /2. si (t) = r(t)si (t)dt But we want to build a receiver that does the minimum amount of calculation. . x2 . we can compute the estimate of the received signal as K−1 k=0 ∞ −∞ r(t)φk (t)dt r(t) = ˆ xk φk (t) This is the ‘correlation receiver’. −∞ What is x in terms of ai and the noise {nk }? What are the mean and covariance of {nk }? Solution: xk = = ∞ −∞ ∞ −∞ r(t)φk (t)dt [si (t) + n(t)]φk (t)dt ∞ −∞ = ai. Noisefree channel Since r(t) = bsi (t) + n(t). In other terms. Correlation with the basis functions gives k=1 xk = r(t).6 in Rice (page 226). As notation.
Then since nk is fXk (xk ) = And.m. (Why?) What is the pdf of xk ? What is the pdf of x? Solution: Gaussian. N 1 2π(N0 /2) e − (xk −ai. and then rewrite the integral as T xk = t=0 r(t)φk (t)dt T This can be written as xk = t=0 r(t)hk (T − t)dt . . xk are independent because xk = ai.i. then xk are also independent.d. . k). since {Xk } are independent.ECE 5520 Fall 2009 49 Next we can show that n1 .k ) 2(N0 /2) 12.2 Matched Filter Receiver xk = ∞ t=−∞ Above we said that r(t)φk (t)dt But there really are ﬁnite limits – let’s say that the signal has a duration T . 2 Is {nk } WSS? Is it a Gaussian random sequence? Since the noise components are independent. m=k δk. o.k e 2(N0 /2) 2π(N0 /2) 2 = k=1 = An example is in ece5510 lec07.k )2 2(N0 /2) fx (x) = k=1 N fXk (xk ) (x −a ) 1 − k i. nN are i.k + nk and ai.k is a deterministic constant. by calculating the autocorrelation Rn (m. .w. k) = E [nk nm ] = = = = ∞ t=−∞ ∞ t=−∞ ∞ τ =−∞ ∞ τ =−∞ E [n(t)n(τ )] φk (t)φm (τ )dτ dt N0 δ(t − τ )φk (t)φm (τ )dτ dt 2 N0 ∞ φk (t)φm (t)dτ dt 2 t=−∞ N0 N0 2 . 1 − e N/2 [2π(N0 /2)] PN 2 k=1 (xk −ai.m = 0. . Rn (m.
log2 M bits of information. correlation or matched ﬁlter reception. The output for other times will be diﬀerent in the correlation and matched ﬁlter. In general. This transmission i=0 conveys to us which of M messages we want to send. . . . Notes: • The xk can be seen as the output of a ‘matched’ ﬁlter at time T . 12. we assume we receive signal si (t) corrupted by noise: r(t) = si (t) + n(t) How do we decide which signal i ∈ {0. M − 1} was transmitted? We split the task into downconversion. (Plug in (T − t) in this formula and the T s cancel out and only positive t is left. This eﬀectively multiplies the noise. textbooks will assume that the automatic gain control (AGC) works properly. r’(t)e j2pfct r’(t) Downconverter AGC r(t) Correlation or Matched Filter x Optimal Detector b Figure 19: A block diagram of the receiver blocks which discussed in lecture 7 & 8.3 Amplitude Before we said b = 1.4 Review At a digital transmitter. for example. At the receiver.m. and detection.) This is the output of a convolution. which is posted on WebCT. have a physical ﬁlter with the impulse response φk (T − t) and thus it is easy to do a matched ﬁlter implementation. xk = r(t) ⋆ hk (t)t=T Or equivalently xk = r(t) ⋆ φk (T − t)t=T This is Rice Section 5. . that is. as shown in Figure 19. We might. • These are just two diﬀerent physical implementations. Lecture 8 Today: (1) Optimal Binary Detection 12.ECE 5520 Fall 2009 50 where hk (t) = φk (T − t). • It may be easier to ‘see’ why the correlation receiver works. . we can transmit at each time one of a set of signals {si (t)}M −1 . A real channel attenuates! The job of our receiver will be to amplify the signal by approximately 1/b. correlation and matched filter rx. • This works at time T . Try out the Matlab code. gain control.4. and then will assume the noise signal is multiplied accordingly.1. taken at time T .
what rules should we have to decide on i? This is the topic of detection. We simulated this in the Matlab code correlation and matched filter rx.i.0 .k + nk for k = 0 . Now. nK−1 ]T Hopefully. Both result in the same output x. .1 . In general.5 Correlation Receiver Our signal set can be represented by the orthonormal basis functions. • If symbols aren’t equally probable. Correlation with the basis k=0 functions gives xk = r(t).1 Overview We’ll show two things: • If each symbol i is equally probable. Optimal detection uses rules designed to minimize the probability that a symbol error will occur.ECE 5520 Fall 2009 51 12. 13 Optimal Detection We receive r(t) and use the matched ﬁlter or correlation receiver to calculate x. The joint pdf of x is K−1 fX (x) = k=0 1 − e fXk (xk ) = N/2 [2π(N0 /2)] PN 2 k=1 (xk −ai. x and ai should be close since i was actually sent. 1]T and receiving r(t) (and thus x) in noise.k ) 2(N0 /2) (16) We showed that there are two ways to implement this: the correlation receiver. . {φk (t)}K−1 . then we’ll need to shift the decision boundaries. . ai. We denote vectors: x = [x0 . pick the i with ai closest to x.k is a deterministic constant • The {nk } are i. 13. then the decision will be. 1 0 X 1 f1 Figure 20: Signal space diagram of a PAM system with an X to mark the receiver output x. . . n1 . .m. φk (t) = ai. Gaussian. K − 1.m. Given the signal space diagram of the transmitted signals and the received signal space vector x. xK−1 ]T ai = [ai. the pdf of x is multivariate Gaussian with each component xk independent. as shown in Figure 20. .k + nk • ai.d. . . In an example Matlab simulation. . . ece5520 lec07. x1 . ai. because: • xk = ai. . . and the matched ﬁlter receiver. we simulated sending ai = [1. .K−1 ]T n = [n0 . . how exactly do we decide which si (t) was sent? Consider this in signal space.
Then using Bayes’ Law. 13. H0 : H1 : We use the law of total probability to say that P [error] = P [error ∩ H0 ] + P [error ∩ H1 ] Where the cap means ‘and’.e. it is very applicable in digital receivers. we list the events that could have occurred. At the start of every detection problem. i. That is. we’ll have only scalars: x.2 Bayesian Detection When we say ‘optimal detection’ in the Bayesian detection framework. • Radar applications: Obstruction detection. When we talk about them as random variables. and n. we’ll substitute N for n and X for x. 14 Binary Detection Let’s just say for now that there are only two signals s0 (t) and s1 (t). motion detection. we mean that we want the smallest probability of error. we mean that a diﬀerent signal was detected than the one that was sent.ECE 5520 Fall 2009 52 Detection theory is a major learning objective of this course. P [error] = P [errorH0 ] P [H0 ] + P [errorH1 ] P [H1 ] (17) X = a0 + N X = a1 + N r(t) = s0 (t) + n(t) r(t) = s1 (t) + n(t) . and only one basis function φ0 (t). a0 and a1 . instead of vectors x. So even though it is somewhat theoretical. signal detection. Further. and S is the complete event space. We need to decide from X whether s0 (t) or s1 (t) was sent. the symbols that could have been sent. the events H0 ∪ H1 ∪ · · · ∪ HM −1 = S. where the ∪ means union. H0 : H1 : Equivalently. • Medical applications: Does an image show a tumor? Does a blood sample indicate a disease? Does a breathing sound indicate an obstructed airway? • Communications applications: Receiver design. The event listing is. We follow all detection and statistics textbooks and label these classes Hi . detection is applicable in a wide variety of problems. and n. Then. a0 and a1 . The probability of error is denoted P [error] By error.. and write: H0 : H1 : HM −1 : ··· r(t) = s0 (t) + n(t) r(t) = s1 (t) + n(t) r(t) = sM −1 (t) + n(t) ··· This must be a complete listing of events. for example.
We can’t be indecisive. So the question is.ECE 5520 Fall 2009 53 14. Figure ?? shows the full integrand of (19). Over a diﬀerent set R1 of values.2 Formula for Probability of Error So the probability of error is P [error] = P [X ∈ R1 H0 ] P [H0 ] + P [X ∈ R0 H1 ] P [H1 ] (18) The probability that X is in R1 is one minus the probability that it is in R0 . Over some set R0 of values of X. so • There is no overlap: R0 ∩ R1 = ∅. then this is the region in which X is more probable given H0 than given H1 . .we just say what region of x.1 Decision Region We’re making a decision based only on X.3 Selecting R0 to Minimize Probability of Error We can see what the integrand looks like. Everything else is determined by the time we get to this point. Which x’s should we include in the region? Should we include x which has a positive value of the integrand? Or should we include the parts of x which have a negative value of the integrand? Solution: Select R0 to be all x such that the integrand is negative. since the two are complementary sets. The objective is to minimize the probability of error. we’ll decide that H0 happened (s0 (t) was sent). P [error] = P [H0 ] − + x∈R0 x∈R0 fXH0 (xH0 )P [H0 ] dx fXH1 (xH1 )P [H1 ] dx (19) fXH1 (xH1 )P [H1 ] − fXH0 (xH0 )P [H0 ] dx = P [H0 ] + x∈R0 We’ve got a lot of things in the expression in (19). Then R0 is the area in which fXH0 (xH0 )P [H0 ] > fXH1 (xH1 )P [H1 ] If P [H0 ] = P [H1 ]. we’ll decide H1 occurred (that s1 (t) was sent). 14. • There are no values of x disregarded: R0 ∪ R1 = S. and the integral in (19) will integrate over it. Figure ?? shows the joint densities (the conditional pdfs multiplied by the bit probabilities P [H0 ] and P [H1 ]. P [error] = (1 − P [X ∈ R0 H0 ])P [H0 ] + P [X ∈ R0 H1 ] P [H1 ] P [error] = P [H0 ] − P [X ∈ R0 H0 ] P [H0 ] + P [X ∈ R0 H1 ] P [H1 ] Now note that probabilities that X ∈ R0 are integrals over the event (region) R0 . Figure 21 shows the conditional probability density functions. but the only thing we can change is the region R0 . Finally. We can pick R0 however we want . the diﬀerence between the joint densities. how do you pick R0 to minimize (19)? 14.
which is the integrand in (19).4 −0.8 f XH XH 1 2 Probability Density 0.2 X∩H −f 2 X ∩H 1 Integrand 0 −0.f.d.s.ECE 5520 Fall 2009 54 1 f 0.f.d.2 0 −3 −2 −1 (a) 0 r 1 2 3 (b) 0.s fXH0 (xH0 )P [H0 ] and fXH1 (xH1 )P [H1 ].s (likelihood functions) fXH0 (xH0 ) and fXH1 (xH1 ).4 f 0. .4 0.6 0.f.d.2 −0.6 −3 −2 −1 (c) 0 r 1 2 3 Figure 21: The (a) conditional p. and (c) diﬀerence between the joint p. (b) joint p. fXH1 (xH1 )P [H1 ] − fXH0 (xH0 )P [H0 ].
fXH1 (xH1 ) P [H0 ] < fXH0 (xH0 ) P [H1 ] (20) The left hand side is called the likelihood ratio. The loglikelihood ratio? 3.e. a1 = 1 in Gaussian noise 2 In this example. the math gets much easier if we take the log of both sides. The right hand side is a threshold. the loglikelihood ratio is fXH1 (xH1 ) P [H0 ] < log log fXH0 (xH0 ) P [H1 ] 14. Now.. σN ). Continuing with the loglikelihood ratio. 14.4 LogLikelihood Ratio For the Gaussian distribution. Whenever x indicates that the likelihood ratio is less than the threshold. The decision regions for x? Solution: What is the log of a Gaussian pdf? log fXH0 (xH0 ) = log 1 2 2πσN e 2 − x2 2σ N (21) 1 x2 2 = − log(2πσN ) − 2 2 2σN The log fXH1 (xH1 ) term will be the same but with (x − 1)2 instead of x2 . i. The log() function is strictly increasing. that s0 (t) was sent. Why can we do this? Solution: 1. we’ll decide H1 . applicable no matter what conditional distributions x has. What is: 1. Otherwise. 2. Both sides are positive. assume for a minute that a0 = 0 and a1 = 1. then we’ll decide H0 . log fXH1 (xH1 ) − log fXH0 (xH0 ) < log [H0 ] [H1 ] 2 2 (x − 1) [H0 ] x 2 − 2 [H1 ] 2σN 2σN P [H0 ] 2 x2 − (x − 1)2 < 2σN log P [H1 ] P [H0 ] 2 2x − 1 < 2σN log P [H1 ] P [H0 ] 1 2 + σN log x < 2 P [H1 ] P P P < log P .. that s1 (t) was sent.5 Case of a0 = 0.ECE 5520 Fall 2009 55 Rearranging the terms. n ∼ N (0. The log of a Gaussian pdf? 2. In addition. i. Equation (20) is a very general result.e.
a1 = 1. and P [H0 ] = 0. there is a simple test for x . P [H1 ] = P [H0 ]. What is the decision threshold for x? . decide that s0 (t) was sent. we could have derived the result in the last section the same way. As long as a0 < a1 . which direction will the optimal threshold move. σN = 0.ECE 5520 Fall 2009 56 In the end result. then x = 1 and the logarithm of the fraction is zero.6. because we only decide which likelihood function is higher (neither is scaled by the prior probabilities P [H0 ] or P [H1 ].4. we use the following notation: x H1 > < H0 1 P [H0 ] 2 + σN log 2 P [H1 ] This completely describes the detector receiver. instead of a0 = 0 and a1 = 1.1.7 Equiprobable Special Case P [H1 ] P [H0 ] H1 > < H0 If symbols are equally likely. P [H1 ] = 0. So then a0 + a1 2 The decision above says that if x is closer to a0 . but now. And if x is closer to a1 . P [H0 ] 1 2 x > + σN log 2 P [H1 ] decide H1 . 2 Example: Let a0 = −1. we’d still have r H1 > < H0 γ. Rather than writing both inequalities each time. γ= 2 σN P [H0 ] a0 + a1 + log 2 a1 − a0 P [H1 ] (23) 14. If it is above the decision threshold. The boundary is exactly halfway in between the two signal space vectors.6 General Case for Arbitrary Signals If. decide H0 .8 Examples Example: When H1 becomes less likely. 14. This receiver is also called a maximum likelihood detector. For simplicity.if it is below the decision threshold. we had arbitrary values for them (the signal space representations of s0 (t) and s1 (t)). decide that s1 (t) was sent. towards a0 or towards a1 ? Solution: Towards a1 . we also write x H1 > < H0 γ where γ= 1 P [H0 ] 2 + σN log 2 P [H1 ] (22) 14.
05 log 1. onedimensional signalling.0203 2 0. • We showed the formula for that threshold. γ =0+ 0. P [H1 ].9 Review of Binary Detection We did three things to prove some things about the optimal detector: • We wrote the formula for the probability of error. both in the equiprobable symbol case.5 ≈ 0.ECE 5520 Fall 2009 57 Solution: From (23).4 Example: Can the decision threshold be higher than both a0 and a1 in this binary. Example: In this example. given that xH1 is Gaussian with mean a1 and variance σN of error? Solution: P [x > γH0 ] = Q γ − a0 σN γ − a1 a1 − γ P [x < γH1 ] = 1 − Q =Q σN σN γ − a0 P [error] = P [H0 ] Q + P [H1 ] Q σN a1 − γ σN 14. and in the general case. Lecture 9 Today: (1) Finish Lecture 8 (2) Intro to Mary PAM . you should be able to calculate the optimal decision threshold γ. given all of the above constants and the optimal threshold γ. • We found the decision regions which minimized the probability of error.6 0. • We used the log operator to show that for the Gaussian error case the decision regions are separated by a single threshold. receiver? 2 Given a0 . Starting from P [error] = P [x ∈ R1 H0 ] P [H0 ] + P [x ∈ R0 H1 ] P [H1 ] we can use the decision regions in (22) to write P [error] = P [x > γH0 ] P [H0 ] + P [x < γH1 ] P [H1 ] 2 What is the ﬁrst probability. given that rH0 is Gaussian with mean a0 and variance σN ? What is the second 2 ? What is then the overall probability probability.1 log = 0. and P [H0 ]. a1 . calculate the probability of error from (18). σN .
There is no guarantee this exam will be the same level of diﬃculty of either past exam. by putting these sample exams up. it is modulated with a carrier cos(2πfc t). . 15 Pulse Amplitude Modulation (PAM) Def ’n: Pulse Amplitude Modulation (PAM) M ary Pulse Amplitude Modulation is the use of M scaled versions of a single basis function p(t) = φ0 (t) as a signal set.1 Baseband Signal Examples When you compose a signal of a number of symbols. it is on the probability topic. That makes our symbol rate as 1/Ts symbols per second. 1 2 2 (3t − 1). x(t) = ℜ s(t)ej2πfc t .0 . o. There is a typo: fN (n) should be fN (t). . . −1 ≤ t ≤ 1 and x1 (t) = 0.w. M where ai is the amplitude of waveform i. you know the actual exam won’t contain the same exact problems. t. Each sent waveform (a. • Problem 6 is on the topic of detection theory. 0. That is. . and Ts is the symbol period (not the sample period!).ECE 5520 Fall 2009 58 Sample exams (20078) and solutions posted on WebCT.k. • Problem 6 typo: x1 (t) = Notes on 2007 sample exam: • Problem 2 should have said “with period Tsa ” rather than “at rate Tsa ”. since there’s only one basis function. Of course. Instead of just sending one symbol. for i = 0. si (t) = ai p(t). “symbol”) conveys k = log2 M bits of information. The resulting signal we’ll call s(t) (without any subscript) and is s(t) = n a(n)p(t − nTs ) where a(n) ∈ {a0 . Note we’re calling it p(t) instead of φ0 (t). on the RHS of the given expressions should be “t”s. Notes on 2008 sample exam: • Problem 1(b) was not material covered this semester. . 15. • The book that year used ϕi (t) instead of φi (t) as the notation for a basis function. both for simplicity. and αi instead of si as the signal space vector for signal si (t). aM −1 } is the nth symbol transmitted. . Note: Keep in mind that when s(t) is to be transmitted on a bandpass channel. we are sending one symbol every Ts units of time.a. . • Problem 3 is not on the topic of detection theory. −1 ≤ t ≤ 1 .w. you’ll just add each timedelayed signal to the transmitted waveform. the “x”s o. . and ai instead of ai.
.w. 0 ≤ t < Ts φ0 (t) = p(t) = 0. +(M − 1)A Rice Figure 5.2. Example: 4ary PAM A 4ary PAM signal set using amplitudes a0 = −3A. which is 1/ log 2 M times the symbol energy Es .6 show continuoustime and discretetime realizations. . The average energy per symbol has decreased. a1 = A.ECE 5520 Fall 2009 59 Rice Figures 5. This is also called OnOﬀ Keying (OOK). . √ 1/ Ts . 0 ≤ t < Ts p(t) = 0. it would represent unipolar PAM. 15.5 Value 0 −0. Es = E = E ∞ −∞ 2 ai a2 p2 (t)dt = i ∞ −∞ E a2 φ2 (t)dt i 0 (24) . Example: Binary Unipolar PAM Unipolar binary PAM uses the amplitudes a0 = 0. It shows a signal set using square pulses. respectively.4 Time t 0. Example: Binary Bipolar PAM Figure 22 shows a 4ary PAM signal set using amplitudes a0 = −A.5 −1 0 0. +(M − 3)A. of PAM transmitters and receivers. . o. .2 Average Bit Energy in Mary PAM Assuming that each symbol si (t) is equally likely to be sent. −(M − 3)A. we want to calculate the average bit energy Eb . Typical M ary PAM (for all even M ) uses the following amplitudes: −(M − 1)A. a2 = A. .1 shows the signal space diagram of Mary PAM for diﬀerent values of M .6 0. If the yaxis in Figure 22 was scaled such that the minimum was 0 instead of −1.3 and 5. o.2.w. A. −A.2 0. .8 1 Figure 22: Example signal for binary bipolar PAM example for A = √ Ts .2. 1 0. a3 = 3A is shown in Figure 23. a1 = −A. . a1 = A. It shows a signal set using square pulses. √ 1/ Ts .
Signal space representation 7. Sampling and aliasing 4. . as described above to be the typical Mary PAM case. properties (12) 2.6 0. Orthogonality / Orthonormality 5. .4 Time t 0.8 1 Figure 23: Example signal for 4ary PAM example. (M − 1)A. Bandpass signals 3. . −(M − 3)A. Correlation / matched ﬁlter receivers . . .2 0. Fourier transform of signals. aM −1 } = −(M − 1)A.ECE 5520 Fall 2009 60 3 2 1 Value 0 −1 −2 −3 0 0. . Correlation and Covariance 6. 3 and Eb = 1 (M 2 − 1) 2 A log2 M 3 Lecture 10 Today: Review for Exam 1 16 Topics for Exam 1 1. . Let {a0 . . Then E a2 i = = A2 (1 − M )2 + (3 − M )2 + · · · + (M − 1)2 M A2 M M m=1 (2m − 1 − M )2 = A2 M (M 2 − 1) M 3 (M 2 − 1) = A2 3 So Es = (M 2 −1) 2 A .
25.9: An otherwise ideal mixer (multiplier) generates an internal DC oﬀset which sums with the baseband signal prior to multiplication with the cos(2πfc t) carrier. When x(t) is sampled at a rate n = ±1 n=0 o. Bateman Problem 1. Ts 2 17. πt Ts . if x(t) cannot be determined. 17 17. If anything is confusing. − Ts < t < 2 o. 3. A signal x(t) of ﬁnite energy is applied to a square law device whose output y(t) is deﬁned by y(t) = x2 (t). 4). Prove that the spectrum of y(t) is bandlimited to −2W ≤ ω ≤ 2W . Three functions are shown in Figure P249. 4).1 Additional Problems Spectrum of Communication Signals 1. Rice 2.4 and Table 2. (b) Find the Fourier transform of x(t). why not? 17. in signal space using the orthonormal set found in part (b). . It will be good to know the ‘tricks’ of how each problem is done. Consider the pulse shape x(t) = (a) Draw a plot of the pulse shape x(t). 4. You may have one side of an 8. You will be provided with Table 2. since they will reappear. (from L. Xn = 2. 2. (b) Find the scale factors needed to scale the functions to make them into an orthonormal set over the interval (−4. 2. ask me.W.w.100 What was the original continuoustime signal x(t)? Or. Rice 2.99.5×11 sheet of paper for notes.4.4. The spectrum of x(t) is bandlimited to −W ≤ ω ≤ W . its samples Xn are. Problem 249). Haykin & Moyer Problem 2. Assume that x(t) is a bandlimited signal with X(f ) 1 T = 2W .2 Sampling and Aliasing = 0 for f  > W . 1. Couch 2007. (a) Show that these functions are orthogonal over the interval (−4. 1.39.w. 0. Read over the lecture notes 17. o.3 Orthogonality and Signal Space 1.52 cos 0. (c) Express the waveform w(t) = 1. 2.8.ECE 5520 Fall 2009 61 Make sure you know how to do each HW problem in HWs 13. 0 ≤ t ≤ 4 0.w. How does this aﬀect the spectrum of the output signal? 2.
w. . H0 : H1 : r = a0 + n0 r = a1 + n1 where α0 = −1 and α1 = 1.13. i.5 Correlation / Matched Filter Receivers 1. 3. −T ≤ t ≤ 2 o. H(f ) = Y (f ) 1 = X(f ) 1 + j2πRCf Given that the PSD of the input signal is ﬂat and constant at 1. ﬁnd the value of RC to satisfy the design speciﬁcations. SX (f ) = 1.e. πt T T 2 T 2 − cos 0.w. Let a binary 1D communication system be described by two possible signals. −T ≤ t ≤ 2 o. and n1 is zeromean Gaussian with variance 0.2.1 on Page 219. At the receiver. and noise with diﬀerent variance depending on which signal is sent. πt T . (From Couch 2007 Problem 280) An RC lowpass ﬁlter is the circuit shown in Figure 24. 5.3.14.23. design an RC lowpass ﬁlter that will attenuate this signal by 20 dB at 15 kHz. a signal x(t) = si (t) + n(t) is received. (a) Is s1 (t) an energytype or powertype signal? (b) What is the energy or power (depending on answer to (a)) in s1 (t)? (c) Describe in words and a block diagram the operation of an correlation receiver.4 Random Processes. i...16. 5. Rice Example 5. That is. 2. PSD 1. A binary communication system uses the following equallylikely signals. 5.5.e. s1 (t) = s2 (t) = cos 0.1. and n0 is zeromean Gaussian with variance 0. Its transfer function is + R C + X(t)  Y(t)  Figure 24: An RC lowpass ﬁlter.ECE 5520 Fall 2009 62 2. 5. .1. 5.26 17. Rice 5. 5. (a) What is the conditional pdf of r given H0 ? (b) What is the conditional pdf of r given H1 ? 17.
the output is the same sinc pulse. we came up with threshold γ = (a0 + a1 )/2. In the Gaussian noise case (with equal variance of noise in H0 and H1 ). A sample function n(t) of a zeromean.k ) 2 d1.1 Signal Distance We mentioned. How are N0 /2 and σN related? Recall from lecture 7 that the variance of nk came out to be N0 /2. Suppose that two signal waveforms s1 (t) and s2 (t) are orthogonal over the interval (0. Prove that when a sinc pulse gT (t) = sin(πt/T )/(πt/T ) is passed through its matched ﬁlter. Lecture 11 Today: (1) Mary PAM Intro from Lecture 9 notes (2) Probability of Error in PAM 18 Probability of Error in Binary PAM This is in Section 6.2 BER Function of Distance. T ). 2 (25) 18. Proakis & Salehi 7. there is only d1. Proakis & Salehi 7.2 of Rice. T n1 = 0 T s1 (t)n(t)dt s2 (t)n(t)dt 0 n2 = Prove that E [n1 n2 ] = 0. We had 2 also called the variance of noise component nk as σN .ECE 5520 Fall 2009 63 2.1. two signal case. Noise PSD We have consistently used a1 > a0 .0 = a1 − a0  (26) 18.j = ai − aj = For the 1D. So: 2 σN = N0 .0 . when talking about signal space diagrams.k − aj. a distance between vectors. k=1 (ai. M 1/2 di. and P [error] = P [x > γH0 ] P [H0 ] + P [x < γH1 ] P [H1 ] . and with P [H0 ] = P [H1 ]. white noise process is correlated with s1 (t) and 2 (t) to yield.16.8. 3.
using (25) and (26).ECE 5520 Fall 2009 64 Which simpliﬁed using the Q function. our signals s0 (t) = a0 p(t) and s1 (t) = a1 p(t) are just s0 = a0 and s1 = a1 . so if we had also calculated for the case a1 < a0 . P [x > γH0 ] = Q γ − a0 σN γ − a1 P [x < γH1 ] = 1 − Q σN =Q a1 − γ σN (27) Thus P [x > γH0 ] = Q P [x < γH1 ] = Q So both are equal. we have P [error] = Q d0. and again with P [H0 ] = P [H1 ] = 1/2. we’d have seen that in general. This will be important as we talk about binary PAM. when A0 = 0 and A1 = A. For bipolar signaling.1 2 N0 /2 = Q d2 0. Thus ‘bit’ and ‘symbol’ are interchangeable. when A0 = −A and A1 = A. In signal space. the following formula works: a1 − a0  P [error] = Q 2σN Then. Now we can use (30) to compute the probability of error.3 Binary PAM Error Probabilities For binary PAM (M = 2) it takes one symbol to encode one bit. This expression.1 2N0 (30) Is one that we will see over and over again in this class. P [error] = Q (a1 − a0 )/2 σN (29) (a1 − a0 )/2 σN (a1 − a0 )/2 σN (28) We’d assumed a1 > a0 . 2 d0. A2 2N0 P [error] = Q .1 Q 2N0 18. 2 2 4A 2A = Q P [error] = Q 2N0 N0 For unipolar signaling.
Thus Eb = Es = 1 A2 . so Eb = Es = A2 . 10 0 Theoretical Probability of Error 10 10 10 10 10 10 −1 Unipolar Bipolar −2 −3 −4 −5 −6 0 2 4 6 8 10 E / N ratio. A2 = A2 always. For the unipolar signalling.ECE 5520 Fall 2009 65 However. what is the average energy of bipolar and unipolar signaling? Solution: For the bipolar signalling. dB b 0 12 14 Figure 25: Probability of Error in binary PAM signalling. Since half of the bits are exactly zero. (2) Probability of Error in Mary PAM . the bit error rate of bipolar PAM beats unipolar PAM. Using (24). • What is the diﬀerence in Eb /N0 for a constant probability of error? • What is the diﬀerence in terms of probability of error for a given Eb /N0 ? Lecture 12 Today: (0) Exam 1 Return. • Even for the same bit energy Eb . we rewrite the probability of error expressions in terms of Eb . this doesn’t take into consideration that the unipolar signaling method uses only half of the energy to transmit. 2 4A 2Eb =Q P [error] = Q 2N0 N0 For unipolar signaling. (1) MultiHypothesis Detection Theory. For bipolar signaling. they take no signal energy. 2 Now. P [error] = Q Discussion 2Eb 2N0 =Q Eb N0 These probabilities of error are shown in Figure 25. A2 m m equals A2 with probability 1/2 and zero with probability 1/2.
So. that is. For this class. . (While equiprobable signals is sometimes not the case for M = 2 binary detection. Essentially. a1 g12 a2 g23 a3 g34 a4 r Figure 26: Signal space representation and decision thresholds for 4ary PAM. H0 : H1 : HM −1 : which have joint probabilities.v. See Figure 26. we’ll only consider the case of equally probable signals.ECE 5520 Fall 2009 66 19 Detection with Multiple Symbols When we only had s0 (t) and s1 (t). maximum likelihood detection. this is a (squared) distance between x and ai . the decision is. it is better to maximize the log of the likelihood rather than the likelihood directly.i+1 = ai + ai+1 2 As an example: 4ary PAM. it is very rare in higher M communication systems. we saw that our decision was based on which of the following was higher: fXH0 (xH0 )P [H0 ] and fXH1 (xH1 )P [H1 ] For M ary signals. Symbol Decision = arg max fXHi (xHi ) i 2 For Gaussian (conditional) r. ﬁnd the ai which is closest to x. so (x − ai )2 1 2 log fXHi (xHi ) = − log(2πσN ) − 2 2 2σN This is maximized when (x − ai )2 is minimized. we’ll see that we need to consider the highest of all M possible events Hm . The decision between two neighboring signal vectors ai and ai+1 will be r Hi+1 > < Hi γi. H0 : H1 : HM −1 : ··· fXH0 (xH0 )P [H0 ] fXH1 (xH1 )P [H1 ] fXHM −1 (xHM −1 )P [HM −1 ] ··· ··· r(t) = s0 (t) + n(t) r(t) = s1 (t) + n(t) r(t) = sM (t) + n(t) ··· We will ﬁnd which of these joint probabilities is highest.) If P [H0 ] = · · · = P [HM −1 ] then we only need to ﬁnd the i that makes the likelihood fXHi (xHi ) maximum.s with equal variances σN .
ECE 5520 Fall 2009 67 20 20. 10 0 which means that 3 log2 M Eb M2 − 1 6 log2 M Eb M 2 − 1 N0 (31) 2(M − 1) Q M Theoretical Probability of Error 10 10 10 10 10 10 −1 −2 −3 −4 −5 M=2 M=4 M=8 M=16 −6 0 2 4 6 8 10 E / N ratio. Also the noise σN = N0 /2. For the symbols i in the ‘middle’. Here. P (symbol errorHi ) = 2Q For the symbols i on the ‘sides’. P (symbol errorHi ) = Q So overall. Assuming neighboring symbols ai are spaced by 2A.1 2(M − 1) Q M A N0 /2 A N0 /2 A N0 /2 Symbol Error Rate and Average Bit Energy (M 2 −1) 2 1 A . P (symbol error) = 20. as discussed above. the decision threshold is always A away from the symbol values. log2 M 3 How does this relate to the average bit energy Eb ? From last lecture.1 Mary PAM Probability of Error Symbol Error The probability that we don’t get the symbol correct is the probability that x does not fall within the range 2 between the thresholds in which it belongs. dB b 0 12 14 Figure 27: Probability of Symbol Error in M ary PAM. Eb = A= So P (symbol error) = Equation (31) is plotted in Figure 27. .1. each ai = Ai .
When you make one symbol error (decide H0 or H1 in error) then it will cause one bit error. up to log2 M in the worst case.2 Bit Errors and Gray Encoding For binary PAM. Instead. It will tend to leave the received signal x close to the original signal ai . M . “110” “111” “101” “100” “000” “001” “011” “010” f1 1 0 2 1 3 3 2 Figure 29: Gray encoding for 8ary PAM. As of now .. in almost all cases. The neighbors of ai will be more likely than distant signal space vectors. QAM) when this is not a good approximation. Then. At least at high Eb /N0 . we have to carefully assign bit codes to symbols 1 . Example: Bit coding of M = 8 symbols Assign three bits to each symbol such that any two nearest neighbors are diﬀerent in only one bit (Gray encoding). Solution: Here is one solution. Example: Bit coding of M = 4 symbols While the two options shown in Figure 28 both assign 2bits to each symbol in unique ways.g. the errors will tend to be just one bit ﬂipped. as in Option 2 in Figure 28.2.use (32). We will show that this approximation is very good. one will be assigned binary 0 and the other binary 1.ECE 5520 Fall 2009 68 20. For M ary PAM. . Is one is better or worse? Option 1: “00” “11” “01” “10” Option 2: “00” “01” “11” “10” 1 0 1 Figure 28: Two options for assigning bits to symbols. . there are only two symbols. we need to study further the probability that x will jump more than one decision region.if you have to estimate a bit error rate for M ary PAM . We will also show examples in multidimensional signalling (e. P (error) ≈ 1 P (symbol error) log2 M (32) • Maybe more. It will not shift a signal uniformly across symbols. more than multiple bits ﬂipped. . If Gray encoding is used. 20. bits and symbols are not synonymous. The key is to recall the model for noise.1 Bit Error Probabilities How many bit errors are caused by a symbol error in M ary PAM? • One. one will lead to a higher bit error rate than the other. Thus Gray encoding will only change one bit across boundaries.
Channel: The wideband radio channel is a sum of timedelayed impulses. The amplitudes of the impulses tend to decay over time. RF ﬁlters are used to limit the spectral output of the signal (to meet outofband interference requirements). our own symbols experience interference called intersymbol interference. Filters will be seen in 1. each reﬂection. t) will be received. (33) where αl and φl are the amplitude and phase of the lth multipath component. Consider the spectrum of the ideal 1D PAM system with a square pulse. Wired channels have (nonideal) bandwidth – the attenuation of a cable or twisted pair is a function of frequency. 2. The measured radio channel is a ﬁlter. 2. Typically. consider the eﬀect of ﬁltering in the path between transmitter and receiver. noise power entering receiver. 1. Waveguides have bandwidth limits (and modes). 3. we want to compromise between (1) and (2) and experience only a small amount of both. the radio channel is modeled as L h(τ. then r(t) = s(t) ⋆ h(τ. Essentially. t). and (2) occupies too much time. and τl is its time delay. speed of digital circuitry. Receiver: Bandpass ﬁlters are used to reduce interference. t) = l=1 αl ejφl δ(τ − τl ).1 Multipath Radio Channel (Background knowledge). Transmitter: Baseband ﬁlters come from the limited bandwidth. Note that the ‘matched ﬁlter’ receiver is also a ﬁlter! 21. If we try (2) above and put symbols right next to each other in time. (2) NDim Detection Theory 21 Intersymbol Interference 1. Consider the timedomain of the signal which is close to a rect function in the frequency domain. diﬀraction. 2. but multipath with delays of hundreds of nanoseconds often exist in indoor links and delays of several microseconds often exist in outdoor links. or scattering adds an additional impulse to (33) with a particular time delay (depending on the extra path length compared to the lineofsight path) and complex amplitude (depending on the losses and phase shifts experienced along the path). . In reality. call its impulse response h(τ.ECE 5520 Fall 2009 69 Lecture 13 Today: (1) Pulse Shaping / ISI. If we try (1) above and use FDMA (frequency division multiple access) then the interference is outofband interference. If s(t) is transmitted. We don’t want either: (1) occupies too much spectrum. Now.
0.25 0.2µs is 1/ 5 MHz.2 0.1 0.2 Transmitter and Receiver Filters Example: Bipolar PAM Signal Input to a Filter How does this eﬀect the correlation receiver and detector? Symbols sent over time are no longer orthogonal – one symbol ‘leaks’ into the next (or beyond). At this symbol rate.05 0 0 100 (a) (b) 200 Delay. How about for a 5 kHz symbol rate? Why or why not? Other solutions: • OFDM (Orthogonal Frequency Division Multiplexing) • Direct Sequence Spread Spectrum / CDMA • Adaptive Equalization 21. and (b) 14 to 43. Example: 24002480 MHz Measured Radio Channel The ‘Delay Proﬁle’ is shown in Figure 30 for an example radio channel.1 0.15 0. Key insight: .05 0 0 Amplitude Delay Profile 100 200 Delay. each symbol would fully bleed into the next symbol.25 Amplitude Delay Profile 0.2 0. Actually what we observe PDP(t) = r(t) ⋆ x(t) = (x(t) ⋆ h(t)) ⋆ x(t) PDP(t) = r(t) ⋆ x(t) = RX (t) ⋆ h(t) PDP(f ) = H(f )SX (f ) (34) Note that 200 ns is 0. ns 300 400 0.15 0. 22 Nyquist Filtering • We don’t need the leakage to be zero at all time.ECE 5520 Fall 2009 70 The channel in (33) has inﬁnite bandwidth (why?) so isn’t really what we’d see if we looked just at a narrow band of the channel. ns 300 400 Figure 30: Multiple delay proﬁles measured on two example links: (a) 13 to 43.
1 If X(f ) only bleeds into one neighboring channel (that is. forms an orthonormal basis.. n = 0 0. In more mathematical language.1.0.−1. ∆(f ) = X(f ) − rect(f Ts ) then we can rewrite the Nyquist Filtering condition as. at the matched ﬁlter output. is the condition for orthogonality of two waveforms.. Where have you seen this condition before? This condition.t. i. o. But lots of other functions also accomplish this. The square root raised cosine. ∆ 1 −f 2Ts =∆ 1 +f 2Ts 1 Ts + X(f ) + X f + 1 Ts + ··· for −1/Ts ≤ f < 1/Ts . we denote the diﬀerence between the ideal rect function and our X(f ) as ∆(f ).. Theorem: Nyquist Filtering Proof: A necessary and suﬃcient condition for x(t) to satisfy x(nTs ) = is that its Fourier transform X(f ) satisfy ∞ m=−∞ 1..e. X(f ) = 0 for all f  > Ts ). What we need are pulse shapes (waveforms). See Figure 31. Essentially. for integer n. Thus X(f ) is allowed to bleed into other frequency bands – but the neighboring frequencyshifted copy of X(f ) must be lower s. making the sum of ‘leakage’ or ISI from one symbol be exactly 0 at all other sampling times nTs .w. Appendix A. shown in Figure 32 accomplishes this. of Rice book. X f+ m Ts = Ts Proof: on page 833.. Basically. X(f ) could be: 1 • X(f ) = rect(f Ts ). The Nyquist ﬁltering theorem provides an easy means to come up with others. symmetric about f = 1/(2Ts ). • It may bleed into the next ‘channel’ but the sum of ··· + X f − must be constant across all f . are orthogonal to each other. that there is an inner product of zero..ECE 5520 Fall 2009 • The value out of the matched ﬁlter (correlator) is taken only at times nTs . the sum is constant.. See Figure 31 . that when shifted in time by Ts . we need p(t) such that {φn (t) = p(t − nTs )}n=. 71 Nyquist ﬁltering has the objective of. exactly constant within − 2Ts < f < 1 2Ts band and zero outside.
. which in series. Take a sheet of paper and fold it in half. Unfold. and now we will extend that to N dimensional signal vectors. Cut a function in the thickest side (the edge that you just folded). 2Ts Ts (36) HRRC (f ) = 1 + cos πTs f  − 1−α .w. 1−α ≤ f  ≤ 1+α 2 α 2Ts 2Ts 2Ts 0. ≤ f  1−α 2Ts ≤ 1+α 2Ts (35) But we usually need do design a system with two identical ﬁlters. a0 .” Activity: Doityourself Nyquist ﬁlter.5/Ts . 0 ≤ f  ≤ 1−α 2Ts o. aM −1 . We have developed M ary detection theory for 1D signal vectors. Ts 2 1 + cos πTs α 22. . 22. In other words. Drawing a horizontal line for the frequency f axis. the middle is 0.ECE 5520 Fall 2009 72 X(f) + X(f+1/T) X(f) X(f+1/T) f 1 1 2T T Figure 31: The Nyquist ﬁltering criterion – 1/Ts frequencyshifted copies of X(f ) must add up to a constant. . √ 0 ≤ f  ≤ 1−α Ts . a modulation with two dimensional signal vectors. 23 Mary Detection Theory in N dimensional signal space We are going to start to talk about QAM. Bateman presents this whole condition as “A Nyquist channel response is characterized by the transfer function having a transition band that is symmetrical about a frequency equal to 0. f  − 1−α 2Ts .5 × 1/Ts . o. produce a zeroISI ﬁlter. one at the transmitter and one at the receiver. we need H(f )2 to meet the Nyquist ﬁltering condition. .2 SquareRoot Raised Cosine Filtering 0. Setup: • Transmit: one of M possible symbols.w. and the vertical axis it X(f ).1 Raised Cosine Filtering HRC (f ) = Ts . . and then in half the other direction. Leave a tiny bit so that it is not completely disconnected into two halves. and later even higher dimensional signal vectors.
j )2 2 1 2 (2πσN )N/2 x − ai 2 2σN x − ai 2 2σN So when we want to solve (37) we can simplify quickly to: ˆ = argmax log i i 1 2 (2πσN )N/2 2 2 − ˆ = argmax − x − ai i 2 2σN i ˆ = argmin x − ai 2 i i ˆ = argmin x − ai i i (38) . • Assume: Symbols are equally likely.25 Basis Function φ (t) 0. each component ni is independent with zero mean and variance σN = N0 /2. fXHi (xHi ) = which can also be written as fXHi (xHi ) = 1 2 (2πσN )N/2 exp − exp − j (xj 2 2σN − ai. are orthogonal to each other. ˆ = argmax log fXH (xHi ) i (37) i i Here.1 0. • Question: What are the optimal decision regions? When symbols are equally likely.ECE 5520 Fall 2009 73 0. • Receive: the symbol plus noise: H0 : HM −1 : ··· X = a0 + n X = aM −1 + n ··· • Assume: n is multivariate Gaussian. t/Tsy 3 4 5 6 7 Figure 32: Two SRRC Pulses. delayed in time by nTs for any integer n.15 0.05 0 −0.2 0.05 −6 −5 −4 −3 −2 −1 i 0 1 2 Time. the optimal decision turns out to be given by the maximum likelihood receiver.
Lecture 14 Today: (1) QAM & PSK. The combined decision region of Ri is the space which is the intersection of all pairwise decision regions. We include it here because . Draw a point in the middle of that line segment. This deﬁnition of the orthonormal basis speciﬁcally considers a pulse shape at a frequency ω0 . These are: √ 2p(t) cos(ω0 t) φ0 (t) = √ φ1 (t) = 2p(t) sin(ω0 t) where p(t) is a pulse shape (like the ones we’ve looked at previously) with support on T1 ≤ t ≤ T2 . and reorganize to ﬁnd a linear equation in terms of x. Thus QAM uses two basis functions. (All conditions must be satisﬁed. cancel. respectively) as the two dimensions. p(t) is only nonzero within that window. 3. Draw the perpendicular bisector of the line segment through that point. or subspace in N D).ECE 5520 Fall 2009 74 Again: Just ﬁnd the ai in the signal space diagram which is closest to x. Previously. To ﬁnd this line: 1. 2. (2) QAM Probability of Error 24 Quadrature Amplitude Modulation (QAM) Quadrature Amplitude Modulation (QAM) is a twodimensional signalling method which uses the inphase and quadrature (cosine and sine waves. Why? Solution: Try to ﬁnd the locus of point x which satisfy x − ai 2 = x − aj 2 You can do this by using the inner product to represent the magnitude squared operator: (x − ai )T (x − ai ) = (x − aj )T (x − aj ) Then use FOIL (multiply out). we’ve separated frequency upconversion and downconversion from pulse shaping. Decision Regions Each pairwise comparison results in a linear division of space. Draw a line segment connecting ai and aj .) Example: Optimal Decision Regions See Figure 33. That is. This is left as an exercise. Pairwise Comparisons When is x closer to ai than to aj for some other signal space point j? Solution: The two decision regions are separated by a straight line (Note: replace “line” with plane in 3D.
. Draw the optimal decision regions.ECE 5520 Fall 2009 75 (a) (b) (c) (d) Figure 33: Example signal space diagrams.
Are the two bases orthogonal? First.1 Showing Orthogonality Let’s show that the basis functions are orthogonal. see how far we can get before plugging in for p(t): T2 φ0 (t). φ1 (t) = c2 sin(ω0 (T2 + T1 )) sin(ω0 (T2 − T1 )) ω0 Solution: First. (This is not intuitive!) We have two restrictions on p(t) that makes these two basis functions orthonormal: • p(t) is unitenergy. with the same pulse shape p(t). φ1 (t) = c2 T2 sin(2ω0 t)dt T1 = c2 − = −c2 cos(2ω0 t) 2ω0 T2 T1 cos(2ω0 T2 ) − cos(2ω0 T1 ) 2ω0 cos(2ω0 T2 ) − cos(2ω0 T1 ) = −c2 2ω0 I want the answer in terms of T2 − T1 (for reasons explained below). φ1 (t) We can see that there are two cases: = c2 sin(ω0 (T2 + T1 )) sin(ω0 (T2 − T1 )) ω0 . φ1 (t) = T1 √ √ 2p(t) cos(ω0 t) 2p(t) sin(ω0 t)dt T2 = 2 T1 T2 p2 (t) cos(ω0 t) sin(ω0 t)dt = T1 p2 (t) sin(2ω0 t)dt Next. φ0 (t). consider the constant p(t) = c for a constant c. so φ0 (t). for T1 ≤ t ≤ T2 . for T1 ≤ t ≤ T2 . we can have two orthogonal basis functions. it has low frequency content compared to ω0 t. that is. let p(t) = c for a constant c . show that φ0 (t). 24.ECE 5520 Fall 2009 76 it is critical to see how. • p(t) is ‘low pass’. We’ll need these cosine / sine function relationships: sin(2A) = 2 cos A sin A A+B cos A − cos B = −2 sin 2 sin A−B 2 As a ﬁrst example.
ak. T2 ]. . Thus φ0 (t).ECE 5520 Fall 2009 77 1.0 . Within each of these cycles. assume p2 (t) is nearly constant. when we integrate p2 (t) sin(2ω0 t) across one cycle. This is wellillustrated in Figure 5. Certainly. . That is.1 ]T . because it is a sinusoid. is nearly zero. Thus. . we’re going to end up with approximately zero.15 in the Rice book (page 298). . The only remainder is the remainder (partial cycle) period. In this ﬁgure. we can attempt the proof for the case of arbitrary pulse shape p(t). . For engineering purposes. Proof? Solution: How many cycles are there? Consider the period [T1 . [T1 + (i − 1)π/ω0 . In this case. . − c2 c2 ≤ φ0 (t). the right sin is zero. in the same way that the earlier integral was zero. L= π Then each cycle is in the period. we see a pulse modulated by a sine wave at frequency 2πω0 . L. [T1 + Lπ/ω0 . φ1 (t) = p2 (T1 + Lπ/ω0 ) = p2 (T1 + Lπ/ω0 ) T2 sin(2ω0 t)dt T1 +Lπ/ω0 cos(2ω0 T2 ) − cos(2ω0 T1 + 2πL) 2ω0 sin(ω0 (T2 + T1 )) sin(ω0 (T2 − T1 )) = p2 (T1 + Lπ/ω0 ) ω0 Again. for i = 1. ω0 (T2 − T1 ) = πk for some integer k. T1 + iπ/ω0 ]. . T2 ]. when we divide by numbers on the order of 106 or 109 .2 Constellation With these two basis functions. In this case. where each signal space vector ak is twodimensional: ak = [ak. Finally. For example. φ0 (t) and φ1 (t) are orthogonal. φ1 (t) ω0 ω0 which for very large ω0 . the inner product is bounded on either side: − p2 (T1 + Lπ/ω0 ) p2 (T1 + Lπ/ω0 ) ≤ φ0 (t). frequencies 2πω0 could be in the MHz or GHz ranges. Zooming in on any few cycles. Otherwise. and so the correlation is exactly zero. φ1 (t) ≤ ω0 ω0 Typically. aM −1 . Then. M ary QAM is deﬁned as an arbitrary signal set a0 . you can see that the pulse p2 (t) is largely constant across each cycle. ω0 is a large number. the numerator bounded above and below by +1 and 1. How many times can it be divided by π/ω0 ? Let the integer number be (T2 − T1 )ω0 . this part of the integral is zero here. This assumption allows us to assume that p(t) is nearly constant over the course of one cycle of the carrier sinusoid. 2. we’re going to get a very small inner product. . 24. That is. The pulse duration T2 − T1 is an integer number of periods. we use the ‘lowpass’ assumption that the maximum frequency content of p(t) is much lower than 2πω0 .
• See Figure 5.0 cos(ω0 t) + ak. One such diagram for M = 64 square QAM is also given here in Figure 34. These are either rectangular grids. .17 for examples of square QAM. If you do the following operation you can recover the real signal s(t) as √ s(t) = 2R e−jω0 t sSB (t) You will not be responsible for Complex Baseband notation.0 + jak. • Figure 5. .0 2p(t) cos(ω0 t) + ak.1 sin(ω0 t)] = Note that we could also write the signal s(t) as √ 2p(t)R ak.ECE 5520 Fall 2009 78 The signal corresponding to symbol k in (M ary) QAM is thus s(t) = ak.1 2p(t) sin(ω0 t) √ 2p(t) [ak. M − 1 24. Then. .1 ) This is called ‘Complex Baseband’.1 e−jω0 t s(t) = √ = 2p(t)R e−jω0 t (ak. . . we can see that the signal space representation ak is given by ak = [ak.1 ]T for k = 0. ak.0 e−jω0 t + jak. you will see them write a QAM signal in shorthand as sCB (t) = p(t)(ak. or use squares with the corners cut out.1 ) (39) In many textbooks.18 shows examples of constellations which use M = 2a for some odd integer a.0 φ0 (t) + ak.3 Signal Constellations M=64 QAM Figure 34: Square signal constellation for 64QAM.0 .0 + jak. These constellations use M = 2a for some even integer a. but you should be able to read other books and know what they’re talking about. and arrange the points in a grid.1 φ1 (t) √ √ = ak. and arrange the points in a grid.
G. i. . each with up to 28 = 256 QAM. You can plot ak in signal space and see that it has a magnitude (distance from the origin) of ak  = and angle of ∠ak = tan−1 In the continuous time signal s(t) this is s(t) = √ 2p(t)ak  cos(ω0 t + ∠ak ) 24.DMT uses multicarrier (up to 256 carriers) methods (OFDM).1 ak. .. and the Rice book: • Digital Microwave Relay. binary phaseshift keying (BPSK) is the case of M = 2. it can send up to 215 QAM (32.6 PhaseShift Keying Some implementations of QAM limit the constellation to include only signal space vectors with equal magnitude.0 . and on each narrowband (4. a0  = a1  = · · · = aM −1  The points ai for i = 0.3kHz) carrier. Some examples are shown in Figure 35. and 11 GHz. M − 1 are uniformly spaced on the unit circle. and is shown in Figure 35(a). so both ‘versions’ are identical in concept (although would be a slightly diﬀerent implementation).ECE 5520 Fall 2009 79 24. 24. 24.. We’ll work in class some examples of ﬁnding Eb in diﬀerent constellation diagrams. i. P [error] = Q N0 QPSK M = 4 PSK is also called quadrature phase shift keying (QPSK). various manufacturerspeciﬁc protocols. .7 Systems which use QAM See [Couch 2007]. Thus BPSK is the same as bipolar (2ary) PAM.4 Angle and Magnitude Representation a2 + a2 k. . Note that the rotation of the signal space diagram doesn’t matter.768 QAM).Lite uses up to 128 carriers. .5 Average Energy in MQAM Recall that the average energy is calculated as: Es = Eb = 1 M M −1 i=0 ai 2 M −1 i=0 1 M log2 M ai 2 where Es is the average energy per symbol and Eb is the average energy per bit. for equally probable symbols. BPSK For example. What is the probability of error in BPSK? The same as in bipolar PAM. G.e. wikipedia. 6 GHz. 2Eb .0 k.e. • Dialup modems: use a M = 16 or M = 8 QAM constellation. Note how QPSK is the same as M = 4 square QAM. • DSL.1 ak.
• Cable modems. uses up to 64 QAM. The highest magnitude points (furthest from the origin) strongly (negatively) impact average bit energy. • 802. Downstream: QPSK or 16 QAM.ECE 5520 Fall 2009 80 QPSK QPSK (a) M=8 M=16 (b) Figure 35: Signal constellations for (a) M = 4 PSK and (b) M = 8 and M = 16 PSK.11g: Adaptive modulation method. with 64 QAM or 256 QAM. Some constellations have more complicated decision regions which require more computation to implement in a receiver. Upstream: 6 MHz bandwidth channel. Some constellations are more diﬃcult (energyconsuming) to amplify at the transmitter.11a and 802. There are also some practical considerations that we will discuss 1. 2. • Digital Video Broadcast (DVB): APSK used in ETSI standard. you should realize that there are two main considerations when a constellation is designed for a particular M : 1. 2. .1 QAM Probability of Error Overview of Future Discussions on QAM Before we get into details about the probabilities of error. 25 25. Points with equal distances separating them and their neighboring points tend to reduce probability of error.
Consider the QPSK constellation diagram. and x1 has no impact on the decision . 0 25. π minus the area in the sector within ± M of the correct angle φi . consider M ary PSK. P (symbol error) = 1 − r∈Ri 1 − e 2πσ 2 r−α i 2 2σ 2 (40) This integral is a double integral. but more like a section probability which is the volume under some part of the N dimensional Gaussian pdf. the noise contribution to each element is independent.3 Exact Error Analysis Exact probability of error formulas in N dimensional modulations can be very diﬃcult to ﬁnd. Similarly. there is an exact expression for P [error] in an AWGN environment. the second bit decision is made using only x2 . now consider the decision region for the ﬁrst bit. 3. we must ﬁnd calculate the probability of symbol error as 1 M=8 M=16 Figure 36: Signal space diagram for M ary PSK for M = 8 and M = 16. This is.4 Probability of Error in QPSK In QPSK. This is a way to get a solution that is analytically easier to handle. Nearest Neighbor Approximation. Exact formula. Now we need not just a tail probability. Essentially. You have already calculated the decision regions for each symbol. For example. This is because our decisions regions are more complex than one threshold test. This is a provable upper bound on the probability of error.2 Options for Probability of Error Expressions In order of preference: 1. of the received signal vector x. We needed a Q function to get a tail probability for a 1D Gaussian pdf. Union bound. The decision is made using only one dimension. and we don’t generally have any exact expression to use to express the result in general.. the probability of error is analytically tractable. In a few cases. 2. They require an integration of a N D Gaussian pdf across an area. speciﬁcally x1 . It can be used for “worst case” analysis which is often very useful for engineering design of systems. It is not an approximation. Also. when Gray encoding is used.. E Typically these approximations are good at high Nb . The decisions are decoupled – x2 has no impact on the decision about bit one. 25.ECE 5520 Fall 2009 81 25.
we have that P [E ∪ F ] ≤ P [E] + P [F ] . . our events are typically decision events. using the above formula. E2 . Assume s1 (t) is sent. 25. 25. and it is very useful across communications. The event Ei may represent the event that we decide Hi . If you know one inequality. En we have that n n (42) P i=1 Ei ≤ P [Ei ] i=1 (43) This is called the union bound. n P [symbol error] ≤ P [Ei ] i=1 (44) The union bound gives a conservative estimate. . Example: QPSK First. as shown in Figure 37(a). Since we know the bit error probability for each bit decision (it is the same as bipolar PAM) we can see that the bit error probability is also P [error] = Q 2Eb N0 (41) This is an extraordinary result – the bit rate will double in QPSK. It is useful when the overlaps Ei ∩ Ej are small but diﬃcult to calculate. P [symbol error] = 1 − 1−Q 2Eb N0 2 (45) In contrast. the bandwidth of QPSK is identical to that of BPSK. This can be useful for quick initial study of a modulation type. We know that (from previous analysis): 2Eb P [error] = Q N0 So the probability of symbol error is one minus the probability that there were no error in either bit. As we will show later. In this case.5 Union Bound From 5510 or an equivalent class (or a Venn diagram) you may recall the probability formula. Furthermore. that for two events E and F that P [E ∪ F ] = P [E] + P [F ] − P [E ∩ F ] You can prove this from the three axioms of probability. (This holds for any events E and F !) Then. the bit error rate does not increase.ECE 5520 Fall 2009 82 on bit two. and the ﬁrst axiom of probability. but in theory. let’s study the union bound for QPSK.6 Application of Union Bound In this class. from (42) it is straightforward to show that for any list of sets E1 . when some other symbol not equal to i was actually sent. know this one. We deﬁne the two error events as: . let’s calculate the union bound on the probability of error. .
both produce nearly identical numerical results.ECE 5520 Fall 2009 83 a1 a1 E2 a2 (a) a3 E4 a0 (b) a3 √ Es . and both are identical. 0]T .). Then we write P [symbol errorH1 ] = P [E2 ∪ E0 ∪ E3 ] Questions: 1. In (b) a1 = −a3 = E2 a2 E4 a0 Figure 37: Union bound examples. we use the union bound. 3Es /2]T and a2 = −a0 = [ Es /2. However. (a) is QPSK with symbols of equal energy [0. • E0 . P [symbol errorH1 ] = P [E2 ∪ E0 ] But don’t be confused. The Rice book will not tell you to do this! His strategy is to ignore redundancies. Is this equation exact? 2. • E2 . so P [symbol errorH1 ] ≤ 2Q 2Eb N0 What is missing / What is the diﬀerence in this expression compared to (45)? See Figure 38 to see the union bound probability of error plot. Then. the event that r falls on the wrong side of the decision boundary with a2 . and in fact. P [symbol errorH1 ] ≤ P [E2 ] + P [E0 ] These two probabilities are just the probability of error for a binary modulation. Only at very low Eb /N0 is there any noticeable diﬀerence! Example: 4QAM with two amplitude levels √ This is shown (poorly) in Figure 37(b). it is perfectly acceptable (and in fact a better bound) to remove redundancies and then apply the bound. The amplitudes of the top and bottom symbols are 3 times the . Either way produces an upper bound. Is E2 ∪ E0 ∪ E3 = E2 ∪ E0 ? Why or why not? Because of the answer to the question (2. the event that r falls on the wrong side of the decision boundary with a0 . because we are only looking for an upper bound any way. compared to the exact expression.
What is the union bound on the probability of symbol error. given H1 ? Solution: Given symbol 1. It is probably less than this value. For the twoamplitude 4QAM modulation. amplitude of the symbols on the right and left. 2(0. the symbol energies are all equal. What is the union bound on the probability of symbol error. the probability is the same as above. the union bound on probability of symbol error is P [symbol errorH2 ] ≤ 3·2+2·2 Q 4 2Eb N0 How about average energy? For QPSK.ECE 5520 Fall 2009 84 Probability of Symbol Error 10 −1 10 −2 QPSK Exact QPSK Union Bound 0 1 2 3 4 Eb/N0 Ratio. Deﬁning E2 and E0 as above. (They are positioned to keep the distance between points in the signal space equal to 2Es /N0 .1 General Formula for Union Boundbased Probability of Error This is given in Rice 6. Thus the formula for the union bound is the same. dB 5 6 Figure 38: For QPSK.76: P [symbol error] ≤ 1 M M −1 M −1 m=0 n=0 n=m P [decide Hn Hm ] Note the ≤ sign. This means the actual P [symbol error] will be at most this value. these two distances between symbol a1 and a2 or a0 are the same: 2Es /N0 . it is 2Eb N0 P [symbol errorH2 ] ≤ 3Q So.) I am calling this ”2amplitude 4QAM” (I made it up). given H2 ? Solution: Now. This is a general formula and is not necessarily the best upper bound.5) + 2(1. What do we mean? If I draw any .6. Thus Eav = Es . overall. the union bound. the exact probability of symbol error expression vs. 25.5) = Es Eav = Es 4 Thus there is no advantage to the twoamplitude QAM modulation in terms of average energy.
M − 1. The Rice book uses Eb where I use Eb to denote average bit energy.n Es = Q (46) P [decide Hn Hm ] = Q 2N0 2Es N0 where dm. the pairwise probability of error. I prefer Es to make clear we are talking about Joules per symbol. .n means a higher argument in the Qfunction. Nmin d2 min Q P [symbol error] ≈ M 2N0 Nmin d2 Es min P [symbol error] ≈ Q (47) M 2Es N0 where dmin = min dm. (2) Union Bound and Approximations Lecture 15 Today: (1) Mary PSK and QAM Analysis 26 26.n . and do not need to be included.n is the Euclidean distance between the symbols m and n in the signal space diagram. for some of the error events. and Nmin is the number of pairs of symbols which are separated by distance dmin . Section 2. I have drawn an upper bound. .n will be proportional to A2 .n .1) with the smallest dm. P [decide Hn Hm ]. some higher than others. . since Es will be proportional to 2 A2 and dm. M − 1. A little extra distance means a much lower value of the Qfunction. and n = 0. . . . This is because higher dm. But I could draw lots of functions that are upper bounds. decide Hn Hm may be redundant.ECE 5520 Fall 2009 85 function that is always above the actual P [symbol error] formula. Given this. the factors A will cancel out of the expression.1 QAM Probability of Error NearestNeighbor Approximate Probability of Error As it turns out. So approximately. Note that the Rice book uses Eavg where I use Es to denote average symbol energy. Proof of (46)? Lecture 14 Today: (1) QAM Probability of Error. which in turn means a lower value of the Qfunction. . the probability of error is often well approximated by the terms in the Union Bound (Lecture 14.n dm. m=n m = 0. . is given by: 2 2 dm.6. These neighbors are the ones that are necessary to include in the union bound. then. If we use a constant A when describing the signal space vectors (as we usually do). In particular. We will talk about the concept of “neighboring” symbols to symbol i. .
P [symbol error] ≤ 1 M M −1 M −1 m=0 n=0 n=m P [decide Hn Hm ] P [decide Hn Hm ] = Q d2 m. Calculate the average energy per symbol. P [error] ≈ 1 P [symbol error] log2 M Example: Additional QAM / PSK Constellations Solve for the probability of symbol error in the signal space diagrams in Figure 39. 4. These steps include: 1. 2. Calculate the union bound on probability of symbol error using the formula derived in the previous lecture.2 Summary and Examples We’ve been formulating in class a pattern or stepbystep process for ﬁnding the probability of bit or symbol error in arbitrary Mary PSK and QAM modulations. an error region which is completely contained in a larger error region. 3. j) which are that distance apart. as a function of any amplitude parameters (typically we’ve been using A). Draw decision boundaries. Approximate using the nearest neighbor approximation if needed: Nmin d2 min Q P [symbol error] ≈ M 2N0 where dmin is the shortest pairwise distance in the constellation diagram. using Eb = Es / log2 M . i)!). Convert distances to be in terms of Eb /N0 . • The nearestneighbor approximation. Calculate the average energy per bit. 6. and Nmin is the number of ordered pairs (i. di. if they are not already. Es . 8.n 2N0 7. You should calculate: • An exact expression if one should happen to be available. . Eb . that is. j) and (j.).ECE 5520 Fall 2009 86 26. using the answer to step (1.j = dmin (don’t forget to count both (i. • The Union bound. It is not necessary to include redundant boundary information. Convert probability of symbol error into probability of bit error if Gray encoding can be assumed. Calculate pairwise distances between pairs of symbols which contribute to the decision boundaries. 5.
ECE 5520 Fall 2009 87 Figure 39: Signal space diagram for some example (made up) constellation diagrams. .
but all probability of error expressions should be written in terms of Eb /N0 . . o.w. ∆f = n fsy 1 =n 2Ts 2 . and thus √ (48) φk (t) = 2p(t) cos(2πfc t + 2πk∆f t) where p(t) is a pulse shape. symbols are selected to be sinusoids with frequency selected among a set of M diﬀerent frequencies {f0 . Lecture 16 Today: (1) QAM Examples (Lecture 15) . fM −1 }. f1 . which for example. Frequency modulation in general changes the center frequency over time: Now.ECE 5520 Fall 2009 88 on. (2) FSK 27 Frequency Shift Keying x(t) = cos(2πf (t)t). .) For general k = m. could be a rectangular pulse: √ 1/ Ts . i. this requires that ∆f Ts = n/2. ﬁrst. . in frequency shift keying. 27.1 Orthogonal Frequencies We will want our cosine wave at these M diﬀerent frequencies to be orthonormal.. The plots are in terms of amplitude A. (They are also approximately orthogonal if ∆f is realy big. Consider fk = fc + k∆f . φm (t) = ∞ t=−∞ Ts (2/Ts ) cos(2πfc t + 2πk∆f t) cos(2πfc t + 2πk∆f t) cos(2π(k − m)∆f t)dt + cos(4πfc t + 2π(k + m)∆f t)dt t=0 Ts t=0 = 1/Ts t=0 Ts 1/Ts = 1/Ts = sin(2π(k − m)∆f t) 2π(k − m)∆f sin(2π(k − m)∆f Ts ) 2π(k − m)∆f Ts Yes.e. and then also probability of bit error. • Does this function have unit energy? • Are these functions orthogonal? Solution: φk (t). probability of symbol error. but we don’t want to waste spectrum. 0 ≤ t ≤ Ts p(t) = 0. they are orthogonal if 2π(k − m)∆f Ts is a multiple of π.
FSK can be seen as a switch between M diﬀerent carrier signals. as shown in Figure 41. bandwidth will be lower when we send less steep transitions. What n? We will see that we either use an n of 1 or 2. (Otherwise. In fact. . . A] √ What is the energy per symbol? Show that this means that A = Es . no they’re not. as seen in Figure 42. aM −1 = [0. But usually it is generated from a single VCO. SRRC pulses.2 Transmission of FSK At the transmitter. .) 1 Thus we need to plug into (48) for ∆f = n 2Ts for some integer n in order to have an orthonormal basis. M=2 FSK M=3 FSK Figure 40: Signal space diagram for M = 2 and M = 3 FSK modulation. . . 27. . . Note that we don’t need to sent square wave input into the VCO.ECE 5520 Fall 2009 89 for integer n. 0] a1 = [0. . for example. Def ’n: Voltage Controlled Oscillator (VCO) A sinusoidal generator with frequency that linearly proportional to an input voltage. 0. . A. For M = 2 and M = 3 these vectors are plotted in Figure 40. . . 0. Signal f1(t) Signal f2(t) Switch FSK Out Figure 41: Block diagram of a binary FSK transmitter. . . Signal space vectors ai are given by a0 = [A. 0] . . .
If we only correlate it with one (the sine wave. just as we’ve seen for previous modulation types. we don’t know whether or not the energy at frequency fk correlates highly with the cosine wave or with the sine wave. These M PLLs must operate even though they can only synchronize when their symbol is sent. which are beyond the scope of this course. the carrier cos(2πfk t + θk ) is sent with the transmitted signal. respectively. the phase of the output signal φk (t) does not change instantaneously at symbol boundaries iTs for integer i. and thus φ(t− + iTs ) = φ(t+ + iTs ) where t− and t+ are the limiting times just to the left and to the right of 0. for every frequency. Also. cosine and sine (see QAM and PSK). cos(2πfc t + 2π(M − 1)∆f t + θM −1 ) 27. though – we have a fundamental problem. the sign or phase of the sinusoid is not very important – only one symbol exists in each dimension.ECE 5520 Fall 2009 90 Def ’n: Continuous Phase Frequency Shift Keying FSK with no phase discontinuity between symbols. VCO Frequency Signal f(t) FSK Out Figure 42: Block diagram of a binary FSK transmitter. . This will give us two inner products. having M PLLs is a drawback. and the phase makes the signal the other (a cosine wave) we would get a inner product of zero! The solution is that we need to correlate the received signal with both a sine and a cosine wave at the frequency fk . . lets call them xI using the capital I to denote inphase and k xQ with Q denoting quadrature. Since we will not know the phase of the received signal. so we’d need to know and be synchronized to M diﬀerent phases θi . ˆ Each phase θk is estimated to be θk by a separate phaselocked loop (PLL). In one ﬂavor of CPFSK (called Sunde’s FSK).3 Reception of FSK FSK reception is either phase coherent or phase noncoherent. This is the only case where I’ve heard of using coherent FSK reception.5 Noncoherent Reception Notice that in Figure 40.4 Coherent Reception FSK reception can be done via a correlation receiver. for example). In noncoherent reception. 27. one for each symbol frequency: cos(2πfc t + θ0 ) cos(2πfc t + 2π∆f t + θ1 ) . As we know. There are a variety of ﬂavors of CPFSK. there are two orthogonal functions. coherent detection becomes diﬃcult. In other words. to aid in demodulation (at the expense of the additional energy). we just measure the energy in each frequency. 1/M of the time (assuming equallyprobable symbols). Here. This is more diﬃcult than it sounds. k . 27. there are M possible carrier frequencies. As M gets high.
1. k k The energy at frequency fk .ECE 5520 Fall 2009 sin(2pfit) 91 xQ i length: (xI )2 + (xQ)2 i i xI i cos(2pfit) Figure 43: The energy in a noncoherent FSK receiver at one frequency fk is calculated by ﬁnding its correlation with the cosine wave (xI ) and sine wave (xQ ) at the frequency of interest. xQ ]T . 27. M −1. You could prove this. Figure 44: (Proakis & Salehi) Phasecoherent demodulation of M ary FSK signals. . fk . but the optimal detector (Maximum likelihood detector) is then to decide upon the frequency with the maximum energy Efk . . Is this a threshold test? We would need new analysis of the FSK noncoherent detector to ﬁnd its analytical probability of error. and calculating the squared length k k of the vector [xI . . Q Efk = (xI )2 + (xk )2 k is calculated for each frequency fk .6 Receiver Block Diagrams The block diagram of the the noncoherent FSK receiver is shown in Figures 45 and 46 (copied from the Proakis & Salehi book). . . that is. Compare this to the coherent FSK receiver in Figure 44. k = 0.
Figure 46: (Proakis & Salehi) Demodulation and squarelaw detection of binary FSK signals.ECE 5520 Fall 2009 92 Figure 45: (Proakis & Salehi) Demodulation of M ary FSK signals for noncoherent detection. .
5 dB worse than bipolar PAM (requires 1. Now. to d0.5 dB less energy per bit). given the noncoherent receiver and rmc and rms . but the symbols are spaced diﬀerently (more closely) as a function of Eb .ECE 5520 Fall 2009 93 27. . What is the detection threshold line separating the two decision regions? 2.8 Probability of Error for Noncoherent Binary FSK The energy detector shown in Figure 46 uses the energy in each frequency and selects the frequency with maximum energy. the envelope rm is an optimum (suﬃcient) statistic to use to decide between s1 . let’s look at coherent detection of binary FSK. What is the distance between points in the Binary FSK signal space? What is the probability of error for coherent binary FSK? It is the same as bipolar PAM. 2. 2 Question: What will rm equal when the noise is very small? As it turns out. 2π frHi (rHi ) = = 0 2π 0 fr. Note that this depends on θm .θmHi (r.5 dB better than OOK (requires 1. the spacing between symbols has reduced by a factor of So P [error]2−Co−F SK = Q For the same probability of bit error. We had that d2 0. They formulate the prior probabilities frHi (rHi ). . H0 ) is a 2D Gaussian random vector with i. What do they do to prove this in Proakis & Salehi? They prove it for binary noncoherent FSK. so rm is termed the envelope. It takes quite a bit of 5510 to do this proof. 1.5 dB more energy per bit). 2 This energy is denoted rm in Figure 46 for frequency m and is 2 2 2 rm = rmc + rms This energy measure is a statistic which measures how much energy was in the signal at frequency fm . The ‘envelope’ is a term used for the square root of the energy.i. which is assumed to be uniform between 0 and 2π. Hi )fθm Hi (φHi )dφ (49) Note that frθm. .d. binary FSK is about 1. 27.1 = Eb N0 √ 2Eb .1 P [error]2−ary = Q 2N0 √ 2/2 compared to bipolar PAM. f2 . and independent of the noise.H0 (rφ. 1. but 1.7 Probability of Error for Coherent Binary FSK First.Hi (rφ. sM . φHi )dφ frθm. Deﬁne the received vector r as a 4 length vector of the correlation of r(t) with the sin and cos at each frequency f1 . components.
Where the joint probability fr∩H0 (r ∩ H0 ) is greater than frH1 (rH1 ). the derivation in the Proakis & Salehi book. Eb 1 (50) P [error]2−N C−F SK = exp − 2 2N0 The expressions for probability of error in binary FSK (both coherent and noncoherent) are important.9. The expression for probability of error in binary noncoherent FSK is given by. and is presented since it does not appear in the Rice book. The decisions in this last step. Example: Probability of Error for Noncoherent M = 2 case Use the above expressions to ﬁnd the P [symbol error] and P [error] for binary noncoherent FSK.9. 5. which is posted on WebCT. are shown to reduce to this decision (given that P [H0 ] = P [H1 ]): 2 2 r1c + r1s H0 > < H1 2 2 r2c + r2s The “envelope detector” can equally well be called the “energy detector”. section 7. Otherwise. They formulate the joint probabilities fr∩H0 (r ∩ H0 ) and fr∩H1 (r ∩ H1 ).ECE 5520 Fall 2009 3. The proof is in Proakis & Salehi. The above proof is simply FYI. Proof Summary: Our noncoherent receiver ﬁnds the energy in each frequency. Lecture 17 Today: (1) FSK Error Probabilities (2) OFDM 27.9 27. it decides H1 . after manipulation of the pdfs. Section 7. as well.6. You will use them to be able to design communication systems that use FSK. The probability that the correct frequency is selected is the probability that the Ricean random variable is larger than all of the other random variables measured at the other frequencies.6. Solution: 1 − 1 Eb P [symbol error] = P [bit error] = e 2 N0 2 M/2 P [symbol error] M −1 . They instead are either Ricean (for the transmitted frequency) or Rayleigh distributed (for the “other” M − 1 frequencies).9 shows that P [symbol error] = and P [error]M −nc−F SK = M −1 n=1 (−1)n+1 n E M −1 1 − log2 M n+1 Nb 0 e n+1 n See Figure 47. the receiver decides H0 . and it often is.1 FSK Error Probabilities. These energy values no longer have a Gaussian distribution (due to the squaring of the amplitudes in the energy calculation). page 430. Part 2 M ary NonCoherent FSK For M ary noncoherent FSK. An exact expression for the probability of error can be derived. and you should make note of them. 94 4.
Speciﬁcally. consider Figure 48. Furthermore. the probability of error E goes to zero. each subchannel has a symbol period of Ts K. As M → ∞. which shows an example frequency selective fading pattern. For each band. In HW 7. Note for square wave pulses. frequency dependent gains are also experienced in DSL. But. divide your BT bandwidth into K subchannels. 27. because phone lines were not designed for higher frequency signals. you show that the total bit rate is the same in the multiplexed system. 10 log 10 H(f )2 .6dB. Our choice is arbitrary. so the sum of the energy is constant. The energy would be divided by K in each subchannel. BT = (M − 1)∆f + 2B where B is the onesided bandwidth of the digital baseband signal. 27. Note that in the multiplexed system.10 Bandwidth of FSK Carson’s rule is used to calculate the bandwidth of FM signals. B = (1 + α)/Ts . For M ary FSK. longer by a factor of K so that the bandwidth of that subchannel has narrower bandwidth. For example. it tells us that the approximate bandwidth is.it is related to the ShannonHartley 0 theorem on the limit of reliable communication. 28. Remember this for later .2 Summary The key insight is that probability of error decreases for increasing M . your transmission energy is the same. you now have BT /K bandwidth on each subchannel. 28 Frequency Multiplexing Frequency multiplexing is the division of the total bandwidth (call it BT ) into many smaller frequency bands. we might send a PSK or PAM or QAM signal on that band. so you don’t lose anything by this division. dB 10 12 0 Figure 47: Probability of bit error for noncoherent reception of Mary FSK.1 Frequency Selective Fading Frequency selectivity is primarily a problem in wireless channels.ECE 5520 Fall 2009 −1 95 10 10 −2 Probability of Error 10 −3 10 −4 10 −5 10 −6 M=2 M=4 M=8 M=16 M=32 2 4 6 8 Eb/N0 Ratio. for the case when Nb > −1. and sending a lower bit rate on each one. For the nulltonull bandwidth of raisedcosine pulse shaping. B = 1/Ts .9. where H(f ) is an .
We’re going to mention frequency multiplexing.2 Beneﬁts of Frequency Multiplexing Now. • For stationary transmitter and receiver and stationary f . or is in a severe fade. Wideband: The bandwidth is used for one channel. has a very low SNR. example wireless channel.ECE 5520 Fall 2009 BT/K 96 Severely faded channel Figure 48: An example frequency selective fading pattern. 2. designers may use a wideband modulation method which is robust to frequency selective fading. there are K parallel bitstreams. Some channels experience gain. For example. • Directsequence spread spectrum (DSSS). This H(f ) might be experienced in an average outdoor channel. If you put down the transmitter and receiver and have them operate at a particular f ′ . while most experience little attenuation.5 (the .) • You can see that some channels experience severe attenuation. (The f is relative.. Problems: 1. where R is the total bit rate.e. which introduces ISI. a subchannel either experiences a high SNR and makes no errors. the channel stays constant over time. and experiences a BER of 0. • Frequency multiplexing methods (including OFDM). take the actual frequency f = f + fc for some center frequency fc . 28. each of rate R/K. When designing for frequency selective fading. As a ﬁrst order approximation. then the loss 10 log10 H(f ′ )2 will not change throughout the communication. bit errors are not an ‘all or nothing’ game. than the error rate will be too high. The power is E increased by a fade margin which makes the Nb high enough for low BER demodulation except in the 0 most extreme fading situations. The whole bandwidth is divided into K subchannels. If the channel loss is too great. at a lower bit rate. or • Frequencyhopping spread spectrum (FHSS). ˜ i. The H(f ) acts as a ﬁlter. and each subchannel’s channel ﬁlter is mostly constant. In frequency multiplexing. Narrowband: Part of the bandwidth is used for one narrow channel.
w. Compare this to a single narrowband channel. 0 ≤ t ≤ Ts o.Q (t))ej2πk∆f t Ak (t)ej2πk∆f t k=1 2/Ts R (51) where Ak (t) = ak.w. 0 ≤ t ≤ Ts o. o. OFDM is the combination: φ0. o. 2/Ts cos(2πfc t + 2π∆f t). than you can see why it is possible to use an IFFT and FFT to generate the complex baseband signal.w. a wideband system with very high ISI might be completely unable to demodulate the received signal. Similarly.Q (t). 0 ≤ t ≤ Ts 0.I (t) = φ1. 0. 2/Ts sin(2πfc t + 2π∆f t). In FSK. we use two basis functions at the same frequency.5 probability of error.I (t) + jak. 0.I (t) + jak. 2/Ts sin(2πfc t). Frequency multiplexing is typically combined with channel coding designed to correct a small percentage of bit errors. φM −1. FFT implementation: There is a particular implementation of the transmitter and receiver that use FFT/IFFT operations. 0 ≤ t ≤ Ts 0.Q (t) sin(2πfk t)] The complex baseband signal of the sum of all K signals might then be represented as K xl (t) = xl (t) = 2/Ts R k=1 K (ak. which has probability β that it will have a 0. 1.Q (t) = 0. These are all orthogonal functions! We can transmit much more information than possible in M ary FSK.w.3 OFDM as an extension of FSK This is section 5. This avoids having K independent transmitter chains and receiver chains.5 in the Rice book. .Q (t) = φ1. 2/Ts cos(2πfc t). (Note we have 2M basis functions here!) The signal on subchannel k might be represented as: xk (t) = 2/Ts [ak. The . o.I (t) = φ0. 28. . 2/Ts cos(2πfc t + 2π(M − 1)∆f t). 0 ≤ t ≤ Ts o. Does this look like an inverse discrete fourier transform? If yes.w. 2/Ts sin(2πfc t + 2π(M − 1)∆f t). If β is the probability that a subchannel experiences a severe fade.I (t) cos(2πfk t) + ak. 1 where ∆f = 2Ts . the overall probability of error will be 0.w.ECE 5520 Fall 2009 97 worst bit error rate!).I (t) = φM −1. 0 ≤ t ≤ Ts 0. In QAM. we use a single basis function at each of diﬀerent frequencies.Q (t) = .5β.
Thus the bit rate is 250 × 103 OFDM symbols subcarriers coded bits Mbits 48 4 = 48 sec OFDM symbol subcarrier sec Lecture 18 Today: Comparison of Modulation Methods 29 Comparison of Modulation Methods This section ﬁrst presents some miscellaneous information which is important to real digital communications systems but doesn’t ﬁt nicely into other lectures. 29.11a IEEE 802.11a is 250k/sec. so eﬀectively 48 subcarriers are used for data. Each data subcarrier can be modulated in diﬀerent ways.1 Diﬀerential Encoding for BPSK Def ’n: Coherent Reception The reception of a signal when its carrier phase is explicitly determined and used for demodulation.11a uses OFDM with 52 subcarriers. One example is to use 16 square QAM on each subcarrier (which is 4 bits per symbol per subcarrier). An example state space diagram for K = 3 and PAM on each channel is shown in Figure 49. . Since the K carriers are orthogonal. 3 subchannels of 4ary PAM Figure 49: Signal space diagram for K = 3 subchannel OFDM with 4PAM on each channel. But. OFDM. rather than transmitting on one of the K carriers at a given time (like FSK) we transmit information in parallel on all K channels simultaneously. the signal is like Kary FSK.ECE 5520 Fall 2009 98 FFT implementation (and the speed and ease of implementation of the FFT in hardware) is why OFDM is popular. Four of the subcarriers are reserved for pilot tones. The symbol rate in 802. Example: 802.
1. The sequence bn is a sequence of 0’s and 1’s. 1. π]T 29. 1. for diﬀerential BPSK. . a switch in the angle of the signal space vector from 0o to 180o or vice versa indicates a bit 1. 0. we’ve said. and if it is greater than zero. π. send a0 if bn = 0. we ﬁnd bn by comparing the phase of xn to the phase of xn−1 . 0. ˆn = [1.1. 0]T .2 DPSK Receiver Now. 0]. 1. in diﬀerential encoding. . b . 0. show that the receiver will decode the original bitstream with diﬀerential decoding. bn = 0 1 − kn−1 . Basically. 0. 0. if kn−1 = 0 then 1 − kn−1 = 1. 0. π. bn = 1 Note that 1 − kn−1 is the complement or negation of kn−1 – if kn−1 = 1 then 1 − kn−1 = 0. 1. consider the bit sequence {bn }. π. 1. is to measure the statistic ∗ ℜ{xn xn−1 } = xn xn−1  cos(∠xn − ∠xn−1 ) as the statistic – if it less than zero. . What our receiver does. 0. 0. Example: Diﬀerential decoding 1. at the receiver. and send a1 if bn = 1. 1. How do we decide which phase to send? Prior to this.1. 1. 0. 0. 1. 1. while staying at the same angle indicates a bit 0. Solution: Assuming φi0 = 0. k8 ]T will be sent? Solution: k = [1. Assume b0 = 0. kn = kn−1 . we use diﬀerential encoding (at the transmitter) and decoding (at the receiver).ECE 5520 Fall 2009 99 For coherent reception of PSK. . 29. 0. What symbols k = [k0 . we also include kn−1 . For noncoherent reception of PSK. Note that we have to just agree on the “zero” phase. π. Assuming no phase shift in the above encoding example. Instead of setting k for ak only as a function of bn . Typically k0 = 0. Now. Example: Diﬀerential encoding Let b = [1. 0. decide bn = 0. 1. this means transmitting a training sequence. decide bn = 1. Typically. 1. 1]T These values of kn correspond to a symbol stream with phases: ∠ak = [π.1 DPSK Transmitter Now. where bn is the nth bit that we want to send. will always need some kind of phase synchronization in BPSK.
0. 0.3 Probability of Bit Error for DPSK The probability of bit error in DPSK is slightly worse than that for BPSK: P [error] = Eb 1 exp − 2 N0 Eb N0 For a constant probability of error. assume that all bits are shifted π radians and we receive ∠x′ = [0. 10 0 Probability of Bit Error 10 −2 10 −4 10 −6 BPSK DBPSK 2 4 6 8 Eb/N0.1. What will be decoded at the receiver? Solution: ˆn = [0. which has probability . DPSK requires about 1 dB more of bit error Q 2Eb N0 than BPSK. dB 10 12 14 0 Figure 50: Comparison of probability of bit error for BPSK and Diﬀerential BPSK. b Rotating all symbols by π radians only causes one bit error. Now. 29. 0. π. 0]. Both are plotted in Figure 50. 0. 1. 1. π. • Bandwidth eﬃciency Eb N0 . 0. π. 29. 1. 0].2 Points for Comparison • Linear ampliﬁers (transmitter complexity) • Receiver complexity • Fidelity (P [error]) vs. 1. 0.ECE 5520 Fall 2009 100 2. 0.
Typically. and correspondingly increase the bandwidth. 29..e.1 PSK. we just make sure we use the same deﬁnition of bandwidth throughout a comparison. bps/Hz.e.67 1.0 3. to increase the bit rate by decreasing the symbol period. is the ratio of bits per second to bandwidth: η = Rb /BT Bandwidth eﬃciency depends on the deﬁnition of “bandwidth”.0 4. 29. 29. This relationship is typically linearly proportional. the quantity of spectrum our signal will use.5 0. i. Def ’n: Bandwidth eﬃciency The bandwidth eﬃciency.0 α=1 0. PAM and QAM In these three modulation methods. the nullnull bandwidth is 1 + α times the bandwidth of the case when we use pure sinc pulses. we divide by log2 M bits per symbol to get Tb = Ts / log2 M seconds per bit. typically denoted η. The transmission bandwidth (for a bandpass signal) is BT = 1+α Ts Since Ts is seconds per symbol. The key ﬁgure of merit: bits per second / Hertz.5 1. i.0 α = 0.0 6.0 101 Table 2: Bandwidth eﬃciency of PSK and QAM modulation methods using raised cosine ﬁltering as a function of α.5 2.67 4. For root raised cosine ﬁltering.3.ECE 5520 Fall 2009 η BPSK QPSK 16QAM 64QAM α=0 1. Since it is usually used for comparative purposes. or BT = where Rb = 1/Tb is the bit rate. BT = (M − 1)∆f + 2B .0 2.3 Bandwidth Eﬃciency We’ve talked about measuring data rate in bits per second. Bandwidth eﬃciency is then η = Rb /BT = See Table 2 for some numerical examples.3. We’ve also talked about Hertz.. the bandwidth is largely determined by the pulse shape.2 FSK (1 + α)Rb log2 M log2 M 1+α We’ve said that the bandwidth of FSK is. we can scale a system.33 2.
See also Rice pages 325331. DPSK. E We can plot these (required Nb . FSK. and 2. So. Eb N0 Eb N0 : Main modulations which we have evaluated probability of error vs.5 Fidelity (P [error]) vs. OOK.5 to be about 14 dB. For the nulltonull bandwidth of raisedcosine pulse shaping.3. bandwidth eﬃciency) pairs. . Nonsquare QAM constellations 5.3. 2B = (1 + α)/Ts . and E • Energy per bit ( Nb ) requirements to achieve a given probability of error.4 Bandwidth Eﬃciency vs. BT = (M − 1)∆f + (1 + α)/Ts = since Rb = 1/Ts for η = Rb /BT = Rb {(M − 1)∆f Ts + (1 + α)} log2 M If ∆f = 1/Ts (required for noncoherent reception). Mary PAM. Mary PSK. 3. 2. 0 29. 0 E Example: Bandwidth eﬃciency vs. we have quantities of interest: • Bandwidth eﬃciency. log2 M (M − 1)∆f Ts + (1 + α) log2 M M +α η= 29. including Binary PAM or BPSK. See Rice Figure 6. Square QAM 4. Nb for M = 8 PSK 0 E What is the required Nb for 8PSK to achieve a probability of bit error of 10−6 ? What is the bandwidth 0 eﬃciency of 8PSK when using 50% excess bandwidth? Solution: Given in Rice Figure 6. Mary FSK In this part of the lecture we will break up into groups and derive: (1) the probability of error and (2) probability of symbol error formulas for these types of modulations.6. Eb N0 For each modulation format. 1. including QPSK.ECE 5520 Fall 2009 102 where B is the onesided bandwidth of the digital baseband signal.
but each ampliﬁer stays on slightly longer to reduce the “dead zone” at zero.1 Linear / Nonlinear Ampliﬁers An ampliﬁer uses DC power to take an input signal and increase its amplitude at the output. Ampliﬁers are separated into ‘classes’ which describe their conﬁguration (circuits). the conduction angle is slightly higher than 180o . That is. • Class B: linear ampliﬁers turn on for half of a cycle (conduction angle of 180o ) with maximum power eﬃciency of 78. • Class AB: Like class B. .6 Transmitter Complexity What makes a transmitter more complicated? • Need for a linear power ampliﬁer • Higher clock speed • Higher transmit powers • Directional antennas 29. Power is dissipated at all times.6. If PDC is the input power (e.5%. and as a result of their conﬁguration. Output signal is a scaled up version of the input signal. Two in parallel are used to amplify a sinusoidal signal. 29. from battery) to the ampliﬁer. • Class A: linear ampliﬁers with maximum power eﬃciency of 50%. one for the positive part and one for the negative part. the power eﬃciency is rated as ηP = Pout /PDC .ECE 5520 Fall 2009 Name BPSK OOK DPSK MPAM QPSK MPSK Square MQAM 2noncoFSK MnoncoFSK 2coFSK McoFSK = Eb 1 2 exp − 2N0 M −1 M −1 (−1)n+1 n=1 ( n ) n+1 exp 103 P [symbol error] =Q =Q = = 1 2 2Eb N0 Eb N0 E − Nb 0 6 log2 M Eb M 2 −1 N0 P [error] same same same ≈ ≈ 1 log2 M P exp 2(M −1) Q M [symbol error] 2Eb N0 =Q ≤ 2Q E 2(log2 M ) sin2 (π/M ) Nb 0 ≈ = E − n log2 M Nb n+1 0 1 log2 M P [symbol error] √ 3 log2 M Eb ( M −1) 4 √ Q log2 M M −1 N0 M same = M/2 M −1 P M/2 M −1 P [symbol error] same =Q ≤ (M − 1)Q Eb N0 E log2 M Nb 0 = [symbol error] Table 3: Summary of probability of bit and symbol error formulas for several modulations..g. and Pout is the output signal power. their linearity and power eﬃciency.
In order to double the power eﬃciency. They can do this if their output signal has constant envelope. we just need to delay the sampling on the quadrature half of a sample period with respect to the inphase signal. . if you look at the constellation diagram. Rewriting s(t) √ √ s(t) = 2p(t)a0 (t) cos(ω0 t) + 2p(t − Ts /2)a1 (t − Ts /2) sin(ω0 (t − Ts /2)) At the receiver. and has the same Eb /N0 vs. . Can only amplify signals with a nearly constant envelope. we delay the quadrature Ts /2 with respect to the inphase. This generates high distortion. batterypowered transmitters are often willing to use Class C ampliﬁers. which precludes use of a class C ampliﬁer. However. The signal must never go through the origin (envelope of zero) or even near the origin.ECE 5520 Fall 2009 104 • Class C : A class C ampliﬁer is closer to a switch than an ampliﬁer. π 3π 5π 7π ∠a(t) ∈ { . See Figure 52(b) and Figure 51 (a) to see this graphically. . See Figure 52 for a comparison of QPSK and OQPSK. We could have also written s(t) as √ s(t) = 2p(t) [a0 (t) cos(ω0 t) + a1 (t) sin(ω0 t)] The problem is that when the phase changes 180 degrees. It is in a discrete set. you’ll see a circle. we wrote the modulated signal as √ s(t) = 2p(t) cos(ω0 t + ∠a(t)) where ∠a(t) is the angle of the symbol chosen to be sent at time t. but then is bandpass ﬁltered or tuned to the center frequency to force out spurious frequencies. The new transmitted signal takes the same bandwidth and average power. For QPSK. } 4 4 4 4 and p(t) is the pulse shape. 29. the envelope s(t) is largely constant. This means that. the signal s(t) will go through zero. For oﬀset QPSK (OQPSK). Class C nonlinear ampliﬁers have power eﬃciencies around 90%. probability of bit error performance.7 Oﬀset QPSK QPSK without offset: Qchannel Offset QPSK: Qchannel Ichannel Ichannel (a) (b) Figure 51: Constellation Diagram for (a) QPSK and (b) OQPSK.
High ﬁdelity (low bit error rate) 3.1 Real 0 −0.1 0 1 2 Time t 3 4 Imag 0.1 0 Real 0. compared to (b) that for QPSK.05 0 −0.1 (c) 0 1 2 Time t 3 4 (d) −0.1 0.2 0.2 0.2 Real 0 −0.1 0.1 −0.05 −0.8 Receiver Complexity What makes a receiver more complicated? • Synchronization (carrier. High data rate 2.15 −0.1 0.1 (a) 0. Low transmit power .1 Imag 0 −0. showing the (d) largely constant envelope of OQPSK. 29.15 0.2 −0.2 Figure 52: Matlab simulation of (ab) QPSK and (cd) OQPSK.1 0 1 2 Time t 3 4 (b) −0.15 −0. timing) • Multiple parallel receiver chains Lecture 19 Today: (1) Link Budgets and System Design 30 Link Budgets and System Design As a digital communication system designer.1 0 Real 0.2 0.1 105 0.1 0. your mission (if you choose to accept it) is to achieve: 1.1 Imag 0 −0. phase.1 0 1 2 Time t 3 4 Imag 0.15 0.ECE 5520 Fall 2009 0.05 −0.05 0 −0.
Low bandwidth 5. given the bandwidth limits. I’d be able to determine the probability of error for several diﬀerent modulation types. C/N0 is shown to have a divide by relationship with C and N0 . See Figure 53. This lecture is about system design.1 Link Budgets Given C/N0 The received power is denoted C. It is an odd quantity. in particular circuits.ECE 5520 Fall 2009 106 4. radio propagation. it is just a matter of setting some system variables (at the given limits) and determining what that means about the values of other system variables. E e. You are expected to apply what you have learned in other classes (or learn the functional relationships that we describe here). Low transmitter/receiver complexity But this is truly a mission impossible. if I was given the bandwidth limits and C/N0 ratio. and bit error rate limit. • We often describe both the received power at a receiver PR . we discuss this procedure. and the material from this class. the operator is drawn in. Typically some subset of requirements are given. the relationship between probability of bit error and Nb . for example.. Figure 53: Relationships between important system design variables (rectangular boxes). which is drawn as a dotted line. and to what they are related. the received signal power and noise power. and it cuts across ECE classes. because you can’t have everything at the same time. Functional relationships between variables are given as circles – if the relationship is a single operator (e. it has units of Watts. The eﬀect of the choice of modulation impacts several functional relationships. For example. otherwise it is left as an unnamed function f (). For example. 0 30. What is C/N0 ? It is received power divided by noise energy.g..g. division). what data rate can be achieved? Using which modulation? To answer this question. So the system design depends on what the desired system really needs and what are the acceptable tradeoﬀs. and deﬁne each of the variables we see in Figure 53. . but it summarizes what we need to know about the signal and the noise for the purposes of system design. optics. In this lecture. but in the Rice book it is typically denoted C. antennas.
1947 Q−1 8 × 10−4 = 3.7507 Q−1 5 × 10−2 = 1.4316 Q−1 4 × 10−4 = 3.8782 Q−1 3 × 10−3 = 2. modulation.6708 2 × 10−6 = 4.4172 6 × 10−6 = 4.6114 3 × 10−6 = 4. In other words.3408 Q−1 1 × 10−1 = 1.4089 Q−1 9 × 10−3 = 2. Commonly.1559 Q−1 9 × 10−4 = 3.5121 Q−1 7 × 10−3 = 2. we can now relate bit error rate.ECE 5520 Fall 2009 107 E • We know the probability of bit error is typically written as a function of Nb . • Note that people often report C/N0 in dB Hz.8808 Q−1 4 × 10−2 = 1.5 × 10−1 = 1. Q Q−1 Q−1 Q−1 Q−1 Q−1 Q−1 Q−1 Q−1 Q−1 Q−1 Q−1 Q−1 Q−1 Q−1 Q−1 Q−1 Q−1 Q−1 Q−1 −1 1× = 4. Given C/N0 . then software often reports that the data rate is 1/5 = 0.2905 Q−1 6 × 10−4 = 3.6153 −1 Q 2 × 10−4 = 3.2816 Q−1 1. C/N0 = Hz.6521 Q−1 5 × 10−3 = 2.9444 5 × 10−5 = 3.7478 Q−1 4 × 10−3 = 2.4051 Q−1 9 × 10−2 = 1.6 × 106 bps.8461 7 × 10−5 = 3.3528 Q−1 5 × 10−4 = 3. since energy is power times time.2649 1. E Note: We typically use Q (·) and Q−1 (·) to relate BER and Nb in each direction.6449 Q−1 6 × 10−2 = 1.4652 5 × 10−6 = 4. CS people use Bps (kBps or MBps) when describing data rate.1735 2 × 10−5 = 4. The 0 bit energy is Eb . if it takes 5 seconds to transfer a 1MB ﬁle.0364 Q−1 2 × 10−1 = 0. Otherwise. We can write Eb = CTb . For example. While you have Matlab. I am not picky about getting lots of correct decimal places. and bandwidth.2389 Q−1 7 × 10−4 = 3.84162 Q−1 3 × 10−1 = 0. it’s really not a big deal to pull it oﬀ of a chart or table. bit rate.25335 Q−1 5 × 10−1 = 0 .0128 4 × 10−5 = 3.8906 6 × 10−5 = 3. The noise energy is N0 .7534 1.4758 Q−1 8 × 10−2 = 1. which is 10 log10 C N0 Eb N 0 Rb What are the units of C/N0 ? Answer: • Be careful of Bytes per sec vs bits per sec.5 × 10−2 = 2.5758 Q−1 6 × 10−3 = 2. great. If you can program it into your calculator. 1/s.3776 7 × 10−6 = 4.3439 8 × 10−6 = 4.1214 Q−1 1 × 10−3 = 3.5 × 10−6 = 4.5401 Q−1 3 × 10−4 = 3.7455 10−6 TABLE OF THE Q−1 (·) Q 1 × 10−4 = 3.5548 Q−1 7 × 10−2 = 1.5 × 10−4 = 3.3656 −1 FUNCTION: Q−1 1 × 10−2 = 2.4573 Q−1 8 × 10−3 = 2.0537 Q−1 3 × 10−2 = 1.0902 Q−1 1.5264 4 × 10−6 = 4.719 −1 Q 1.3263 Q−1 1.3145 9 × 10−6 = 4.775 9 × 10−5 = 3. To separate the eﬀect of Tb .9677 Q−1 2 × 10−3 = 2.5244 Q−1 4 × 10−1 = 0. this 0 is easy to calculate.8082 8 × 10−5 = 3.1701 Q−1 2 × 10−2 = 2.1075 3 × 10−5 = 4.2884 1 × 10−5 = 4. But the bit rate is 8/5 Mbps or 1. we often denote: C C/N0 Eb = Tb = N0 N0 Rb where Rb = 1/Tb is the bit rate.5 × 10−3 = 2. the following tables/plots of Q−1 (x) will appear on Exam 2. For your convenience.2 MBps or 200 kBps.5 × 10−5 = 4.
reduce your bit rate to conform to the BW constraint. Rb log2 M and then compute the required 4. Sometimes power is the limiting factor in determining the maximum achievable bit rate. Here is a stepbystep version of what you might need do in this case: Method A: Start with powerlimited assumption: 1. and the maximum BER are all given. 2. given the appropriate probability C/N0 Eb N0 constraint. ﬁnd the maximum bit rate.5 Inverse Q function. calculate the maximum symbol rate Rs = bandwidth using the appropriate bandwidth formula. Use the probability of error constraint to determine the of error formula for the modulation. Sometimes bandwidth is the limiting factor in determining the maximum achievable bit rate. Such links (or channels) are called power limited channels. Given the C/N0 constraint and the Eb N0 Eb N0 constraint. the link (or channel) is called a bandwidth limited channel. but be sure to express both in linear units. In this case.5 1 0. 3. You just need to try to solve the problem and see which one limits your system.5 2 1. the maximum bandwidth.5 3 2. Note that Rb = 1/Tb = . Q−1(x) 4 3. Compare the bandwidth at maximum Rs to the bandwidth constraint: If BW at Rs is too high.2 Power and Energy Limited Channels Assume the C/N0 . Given a maximum bit rate. then the system is bandwidth limited. Otherwise.ECE 5520 Fall 2009 108 5 4. your system is power limited.5 0 −7 10 10 −6 10 −5 10 −4 10 Value. Method B: Start with a bandwidthlimited assumption: . and your Rb is achievable. x −3 10 −2 10 −1 10 0 30.
27×106 )/4 = 851 kHz log2 M This is clearly lower than the maximum bandwidth of 1. and can operate with bit rate 2. (If BT had come out > 1.27×106 = 2.585 × 108 = 2.5.5(2. what is the maximum achievable bit rate on this link? Is this a power limited or bandwidth limited channel? Solution: 1. 4.5. So.) Eb N0 at the given bit rate by computing Eb N0 .27 Mbits/s 69.27 Mbits/s.5 MHz. C/N0 = 1082/10 = 1. your system is bandwidthlimited.5 MHz and with an available C/N0 = 82 dB Hz. Solving for Rb . Use the bandwidth constraint and the appropriate bandwidth formula to ﬁnd the maximum symbol rate Rs and then the maximum bit rate Rb . 2.ECE 5520 Fall 2009 109 1. Otherwise. The required bandwidth for this system is (1 + α)Rb BT = = 1. Find the probability of error at that using the appropriate probability of error formula. If the modulation is 16PSK using the SRRC pulse shape with α = 0. (Again.585 × 108 . Example: Rice 6. the system is power limited. what is the maximum achievable bit rate on the link? Is this a power limited or bandwidth limited channel? 2.5 MHz. Rb = C/N0 Eb N0 = 1.) . we would have needed to reduce Rb to meet the bandwidth limit. If the modulation is square 16QAM using the SRRC pulse shape with α = 0.84 (52) Converting C/N0 to linear. 1. If the computed P [error] is greater than the BER constraint. Eb N0 = C/N0 Rb . and you have found the correct maximum bit rate. then your system is power limited. The maximum bit error rate is 10−6 . Find the units. For M = 16 PSK. we can ﬁnd 10−6 = P [error] = 10−6 = Eb N0 Eb N0 = 2 Q 4 2 Eb N0 for the maximum BER: 2(log2 M ) sin2 (π/M ) Eb N0 2 2 Q log2 M Eb N0 2(4) sin2 (π/16) 1 Q−1 2 × 10−6 8 sin (π/16) = 69. Try Method A.67×105 Msymbols/s.33 Consider a bandpass communications link with a bandwidth of 1. Use the previous method to ﬁnd the maximum bit rate.84 and thus Rs = Rb / log2 M = 2.27×106 /4 = 5. make sure that everything is in linear 3.
Note that λ = c/fc where c = 3 × 108 meters per second is the speed of light.ECE 5520 Fall 2009 Eb N0 110 for the maximum BER: 3 log2 M Eb M − 1 N0 2.3 Computing Received Power We need to design our system for the power which we will receive. the received power is calculated from the Friis formula. How can we calculate how much power will be received? We can do this for both wireless and wired channels. this section of the class provides two examples. respectively. In free space.16 MHz log2 M This is greater than the maximum bandwidth of 1.585 × 108 = 5. we have a bandwidthlimited system with a bit rate of 4 MHz. 30.55 The required bandwidth for this bit rate is: BT = (1 + α)Rb = 1. but there are others. For M = 16 (square) QAM. the wavelength is nearly constant across the bandwidth.5 MHz = 4 MHz 1+α 1.5 In summary.1 Free Space ‘Free space’ is the idealization in which nothing exists except for the transmitter and receiver.75×106 )/4 = 2.75 Mbits/s 27. Wireless propagation models are required. Try Method A.3. But. • λ is the wavelength at signal frequency. so we just use the center frequency fc .5 MHz.5(5. so we must reduce the bit rate to Rb = BT log2 M 4 = 1. = √ 4 ( M − 1) √ = P [error] = Q log2 M M 4 (4 − 1) 3(4) Eb Q = 4 4 15 N0 15 −1 Q (4/3) × 10−6 12 2 = 27. For narrowband signals. PT GT GR λ2 (4πR)2 . we can ﬁnd 10 −6 10−6 Eb N0 Eb N0 Solving for Rb . C = PR = where • GT and GR are the antenna gains at the transmitter and receiver.55 (53) Rb = C/N0 Eb N0 = 1. 30. and can really only be used for deep space communications. this formula serves as a starting point for other radio propagation formulas.75×106 = 5.
My opinion is. For example. instead. is λ [C]dBm = [GT ]dB + [GR ]dB + [PT ]dBm + 20 log 10 − 20 log 10 R 4π This says the received power. 30. (This name that Rice chose is unfortunate because if it was positive. losses in a cable. other imperfections. The Rice book writes the Friis formula as [C]dBm = [GT ]dB + [GR ]dB + [PT ]dBm + [Lp ]dB where λ − 20 log 10 R 4π where [Lp ]dB is called the “channel loss”. antenna heights.) There are also typically other losses in a transmitter / receiver system. But. Typically people use decibels to express these numbers. etc. everything is in linear terms. because of shadowing caused by buildings. and L1m is the loss per meter. Rice lumps these in as the term L and writes: [Lp ]dB = +20 log10 [C]dBm = [GT ]dB + [GR ]dB + [PT ]dBm + [Lp ]dB + [L]dB 30. . it should be called “channel gain” and have a negative gain value. trees. [Lp ]dB = L0 − 10n log10 R. but it is typically very negative.ECE 5520 Fall 2009 • PT is the transmit power. it may be more like 1/Rn . Eﬀectively. whenever path loss is linear with the log of distance. C.3 Wired Channels Typically wired channels are lossy as well. for some constants n and L0 . given in dB. For example.. the average loss may increase more quickly than 1/R2 . but the loss is modeled as linear in the length of the cable. is linearly proportional to the log of the distance R. a very short summary is that radio propagation on Earth is diﬀerent than the Friis formula suggests.3.3. Lots of other formulas exist that approximate the received power as a function of distance. and we will write [PR ]dBm or [GT ]dB to denote that they are given by: [PR ]dBm = 10 log 10 PR [GT ]dB = 10 log 10 GT (54) In lecture 2. we showed that the Friis formula. [C]dBm = [PT ]dBm − R [L1m ]dB where PT is the transmit power and R is the cable length in meters. etc.2 Nonfreespace Channels We don’t study radio propagation path loss formulas in this course. etc. 111 Here. it would be a “gain”. type of environment.
075m. So Eb = 319. so: λ [C]dBW = [GT ]dB + [GR ]dB + [PT ]dBW + 20 log10 − 20 log10 R − 2dB − 2dB − 2dB − 1dB 4π 0. But in short. A pigeon in the lineofsight path causes an additional 2 dB loss. Teq can be kept low.52 × 10−21 J.76 × 10−18 J.84 × 106 bits/sec.38 × 10−23 J/K is Boltzmann’s constant and Teq is called the equivalent noise temperature in Kelvin.5 Examples Example: Rice 6. and so you are not responsible for that material here. to achieve P [error] = 10−8 .0 The noise power N0 = kTeq = 1. √ Solution: Starting with the modulation.4 dBW.13 × 10−11 W.) Both antenna gains are 20 dB and the transmit antenna power is 10 W. 30. C = (51.48dBW − 20 log 10 R (55) .0 × 5. and the pigeon. (Such links were the key component in the telephone company’s long distance network before ﬁber optic cables were installed. or 100. How far away can the two towers be if the bit error rate is not to exceed 10−8 ? Include the pigeon.38 × 10−23 (J/K)400(K) = 5. 255 N0 2 3 log2 M Eb M − 1 N0 255 Q−1 24 32 −8 10 15 = 319. Basically. Switching to ﬁnding an expression for C. The modulation is 51. M = 16). Teq is always going to be higher than the temperature of your receiver.plementation loss of 1 dB.ECE 5520 Fall 2009 112 30. all receiver circuits add noise to the received signal.84 Mbits/sec 256 square QAM with a carrier frequency of 4 GHz.36 Consider a “pointtopoint” microwave link. the wavelength is λ = 3 × 108 m/s/4 × 109 1/s = 0. implementation loss. Neal’s hint: Use the dB version of Friis formula and subtract these mentioned dB losses: atmospheric losses. M = 256 square QAM (log2 M = 8. With proper design.52 × 10−21 = 1.76 × 10−18 )1/sec = 9.075m − 20 log10 R − 7dB = 20dB + 20dB + 10dBW + 20 log10 4π = −1. incidental losses.4 Computing Noise Energy N0 = kTeq The noise energy N0 can be calculated as: where k = 1. Atmospheric losses are 2 dB and other incidental losses are 2 dB. 10 −8 = 10−8 = 10−8 = Eb N0 = √ 4 ( M − 1) √ Q log2 m M 4 15 3(8) Eb Q 8 16 255 N0 15 Q 32 24 Eb .84 × 106 )J(1. Since Eb = C/Rb and the bit rate Rb = 51. The receiver has an equivalent noise temperature of 400 K and an im. This is a topic covered in another course. the equivalent temperature is a function of the receiver design.
let’s compare receiver implementations for a (mostly) continuoustime and a (more) discretetime receiver.4 dBW = −1.3. Figure 55 shows a receiver with an ADC immediately after the downconverter. these are generally unknown to the receiver until timing synchronization is complete. This tk is the correct timing delay at each correlator for the kth symbol. Figure 54 has a timing synchronization loop which controls the sampler (the ADC).48dBW − 20 log 10 R and solving for R. φn (t) am (k) φn (t − tk ). then the correlation φn (t − tk ). As a start.1). including analog timing synchronization (from Rice book. For the correlation with n. φn (t − kTs − τ ) (57) Note that if tk = kTs + τ . or QPSK) can be written (assuming carrier synchronization) as r(t) = k m am (k)φm (t − kTs − τ ) (56) where ak (m) is the mth transmitted symbol. Here. The received complex baseband signal (for QAM. after the matched ﬁlter.3 km (about 55 miles) apart. Figure 8. But. an interpolator corrects the sampling time problems using discretetime processing. .3 km. (2) Interpolation Filters 31 Timing Synchronization At the receiver. φn (t − kTs − τ ) is highest and closest to 1. This interpolation is the subject of this section. Lecture 20 Today: (1) Timing Synchronization Intro. xn (k) = xn (k) = k m r(t). note the ADC has nothing controlling it. PAM. Thus microwave towers should be placed at most 88. The input signal is downconverted and then run through matched ﬁlters. we ﬁnd R = 88.ECE 5520 Fall 2009 113 Plugging in [C]dBm = −100. which correlate the signal with φn (t − tk ) for each n. a transmitted signal arrives with an unknown delay τ . Figure 54: Block diagram for a continuoustime receiver. Instead. and for some delay tk .
and then. since the sampling clock must operate at (at least) double the symbol rate. The idea is to “bring the ADC to the antenna” for purposes of reconﬁgurability. If downsampling (by 2) results in the symbol samples rk given by red squares. resulting in samples r(nT). because of the processing delay. Also. a BPSK receiver samples the matched ﬁlter output at a rate of twice per symbol. consider Figure 56. A ‘software radio’ follows the idea that as much of the radio is done in digital. we’ll talk about interpolation.5 −1 4 6 8 10 Time t 12 14 Figure 56: Samples of the matched ﬁlter output (BPSK. we will emphasize digital timing synchronization using an interpolation ﬁlter. after the signal has been sampled. But again. this digital and analog feedback loop can be problematic.ECE 5520 Fall 2009 r(nTsa) ADC ctstime bandpass signal Matched Filter Digital Interpolator Timing Synchronization 114 r(kTsy) x(t) p/2 PLL Timing Synch Control Optimal Detector 16 2cos(2pfct+f) bm 2sin(2pfct+f) ADC Matched Filter Digital Interpolator r(nTsa) r(kTsy) Figure 55: Block diagram for a digital receiver for QAM/PSK. For example. First. then sampling sometimes reduces the magnitude of the desired signal. we’ll consider the control loop. In this ﬁgure. but with a timing error. 32 Interpolation In this class. including discretetime timing synchronization. When we say ‘synchronized in rate’. Some example sampling clocks. which is not preferred. and reducing part counts and costs. unsynchronized with the symbol clock. this requires a DAC and feedback to the analog part of the receiver. Another implementation is like Figure 55 but instead of the interpolator. we mean within an integer multiple. . are shown in Figure 57. The industry trend is more and more towards digital implementations.5 Value 0 −0. 1 0. compared to the actual symbol clock. the timing synch control block is fed back to the ADC. RRC with α = 1) taken at twice the correct symbol rate (vertical lines). These are shown in degrees of severity of correction for the receiver.
This means that µ(k) is given by kTs + τ µ(k) = − m(k) T Example: Calculation Example Let Ts /T = 3. Calculate (m(k). and every 5 samples the delay until the next correct symbol sample will repeat.5 times per symbol.1 and τ /T = 1. kTs + τ is always µ(k)T after the most recent sampling instant. and 0 ≤ µ(k) < 1. m are integers. m(k) = kTs + τ T (58) where ⌊x⌋ is the greatest integer less than function (the Matlab floor function). µ(k)) for k = 1. 3. In other words. 2. T/Ts = 1/2. 32. if T/Ts = 2/5. we cannot count on them ever repeating. where T /Ts = 1/2 (the clocks are commensurate). Def ’n: Incommensurate Two clocks with rates T and Ts are incommensurate if the ratio T/Ts is irrational. the correct sampling times should be kTs + τ . the two clocks are commensurate and we sample exactly twice per symbol period.ECE 5520 Fall 2009 115 Figure 57: (1) Sampling clock and (24) possible actual symbol clocks. the highest integer such that nT < kTs + τ . For example. Instead. our receivers always deal with type (4) sampling clock error as drawn in Figure 57. Symbol clock may be (2) synchronized in rate and phase. (4) synchronized neither in rate nor phase. That is. As another example. the sampling clock has neither the same exact rate nor the same phase as the actual symbol clock. Since clocks are generally incommensurate. The situation shown in Figure 56 is case (3). but the sampling clock does not have the correct phase (τ is not equal to an integer multiple of T ). In general.8. but no samples were taken at those instants. . We can write that kTs + τ = [m(k) + µ(k)]T where m(k) is an integer. for both cases (3) and (4) in Figure 57. (3) synchronized in rate but not in phase. with sample clock. we sample exactly 2. In contrast. two clocks are commensurate if the ratio T/Ts can be written as n/m where n. where µ(k) is called the fractional interval.1 Sampling Time Notation In general.
8⌋ = 4. r(t) = n r(nT)hI (t − nT) sin(πt/T) . πt/T (59) where hI (t) = Why is this so? What are the conditions necessary for this representation to be accurate? If we wanted the signal at the correct sampling times. where Ts is the actual symbol period used by the transmitter. r([m(k) + µ(k)]T) = i r([m(k) − i]T)hI ([i + µ(k)]T).1) + 1. (60) This is a ﬁlter on samples of r(·). using the (m(k).2 Seeing Interpolation as Filtering Consider the output of the matched ﬁlter.1) + 1. m(2) = ⌊2(3.ECE 5520 Fall 2009 116 Solution: m(1) = ⌊3.8⌋ = 11.4. Instead. this requires an inﬁnite sum over i from −∞ to ∞. Note: Good illustrations are given in M.13. and Figure 8. r([m(k) + µ(k)]T) = n r(nT)hI ([m(k) − n + µ(k)]T).8⌋ = 8. we could have it – we just need to calculate r(t) at another set of times (not nT). Reindexing with i = m(k) − n.4. Plugging these times in for t in (59). The problem is that in general. r(t) as given in (57).3 Approximate Interpolation Filters Clearly. . because the sinc function has inﬁnite support. (60) is a ﬁlter. in which we draw a straight line between the two nearest sampled values to approximate the values of the continuous signal between the samples. we use polynomial approximations for hI (t): • The easiest one we’re all familiar with is linear interpolation (a ﬁrstorder polynomial). since kTs + τ = [m(k) + µ(k)]T. µ(k)) notation. The desired sample at [m(k) + µ(k)]T is calculated by adding the weighted contribution from the signal at each sampling time.1 + 1. Rice.9 µ(1) = 0 µ(1) = 0. The analog output of the matched ﬁlter could be represented as a function of its samples r(nT). and in between samples 11 and 12. 32. µ(1) = 0. This isn’t so great of an approximation. at sample 8. Call the correct symbol sampling times as kTs + τ for integer k.12. we have that r(kTs + τ ) = n r(nT)hI (kTs + τ − nT) Now. 32. where the ﬁlter coeﬃcients are dependent on µ(k).1 m(3) = ⌊3(3. Thus your interpolation will be done: in between samples 4 and 5. Figure 8.
24. As µ(k) → 1. To see results for diﬀerent order polynomial ﬁlters. . the three point ﬁt does not result in a linearphase ﬁlter. Thus we could rewrite (60) as r([m(k) − i]T)hI (i. given µ(k). 0 r([m(k) + µ(k)]T) = i=−1 r([m(k) − i]T)hI (i. • Finally. µ(k)1 . consider the linear interpolator. (61) r([m(k) + µ(k)]T) = i That is.1 Higher order polynomial interpolation ﬁlters p In general. µ(k)). they are a weighted sum of µ(k)0 . and one on the other. of AT&T Bell Labs. we should take the r([m(k) + 1]T) sample exactly. Essentially. we form a weighted average of the two nearest samples. we should take the r(m(k)T) sample exactly. Given three points. .a parabola) is actually a very good approximation. we can see that the ﬁlter coeﬃcients are a function of µ(k). µ(k)) = µ(k) and hI (0. This is temporal asymmetry. Example: First order polynomial interpolation For example.4 Implementations Note: These polynomial ﬁlters are called Farrow ﬁlters and are named after Cecil W. Rice handout. . Rice Figure 8. Note that the i indices seem backwards. we could use a cubic interpolation ﬁlter. the ﬁlter is hI (i) but its values are a function of µ(k). Farrow. Four points determine a 3rd order polynomial. we can use four points to ﬁt a secondorder polynomial. From (60).) Instead.e. 32. • However. µ(k)) = bl (i)µ(k)l l=0 A full table of bl (i) is given in Table 8. hI (i. As µ(k) → 0. note in the time domain that two samples are on one side of the interpolation point. µ(k)) What are the ﬁlter elements hI for a linear interpolation ﬁlter? Solution: r([m(k) + µ(k)]T) = µ(k)r([m(k) + 1]T) + [1 − µ(k)]r(m(k)T) Here we have used hI (−1. see M. . . 32.ECE 5520 Fall 2009 117 • A secondorder polynomial (i.1 of the M. µ(k)p for a pth order polynomial ﬁlter. The ﬁlter coeﬃcients are a polynomial function of µ(k). µ(k)) = 1 − µ(k). and get a linearphase ﬁlter. who has the US patent (1989) for the “Continuously variable digital delay circuit”. that is. (To see this. These Farrow ﬁlters started to be used in the 90’s and are now very common due to the dominance of digital processing in receivers. the fractional interval.4. one can determine a parabola that ﬁts those three points exactly. and result in a linear ﬁlter. µ(k)2 .
25.5.125 − 0.125 − 0. It has been shown by simulation that α = 0. 0. there is an extra degree of freedom – you can select parameter α to be in the range 0 < α < 1. µ = 0.125 hI (−1. Filter Timing e(n) Loop Error Filter Detector s [ n] h[n]=p(nT) 2 cos(W0n) v(n) symbol strobe VCC m Interp.625 (62) Does this make sense? Do the weights add up to 1? Does it make sense to subtract a fraction of the two more distant samples? Example: Matlab implementation of Farrow Filter My implementation is called ece5520 lec20. Filter MF h[n]=p(nT) 2 sin(W0n) N/2 rI(k) rQ(k) xQ(n) Symbol Decision All Bits Preamble Detector DATA Bits Figure 58: Flow chart of ﬁnal project.43 is best. as my implementation uses a loop. . Example: 2nd order Farrow ﬁlter What is the Farrow ﬁlter for α = 0. µ = 0.125 hI (0.5) = −αµ2 + (α − 1)µ + 1 = −0.125 − 0. but people tend to use α = 0.25 = −0.5) = αµ2 − αµ = 0. timing synchronization for QPSK. and division by two is extremely easy in digital ﬁlters. Since µ2 = 0.25 = −0. 0. rather than vector processing. we can calculate that hI (−2. Be careful.m and is posted on WebCT.ECE 5520 Fall 2009 118 For the 2nd order Farrow ﬁlter. 0.5) = αµ2 − αµ = 0.75 = 0.5 which interpolates exactly halfway between sample points? Solution: From the problem statement.625 hI (1.5 because it is only slightly worse. µ0 = 1.5.25 + 1 = 0. 0. Lecture 21 Today: (1) Timing Synchronization (2) PLLs 33 Final Project Overview xI(n) MF N/2 Interp.125 + 0.5) = −αµ2 + (1 + α)µ = −0.
Loop Filter 4. It adds these blocks to your QPSK implementation: 1. where n is the integer part and µ is the fractional oﬀset. you can interpolate between samples to ﬁnd the desired sample. Implementations may be continuous time. Timing Error Detector (TED) 3. the samples may be at incorrect times.ECE 5520 Fall 2009 119 For your ﬁnal project.4. you have the advantage of having access to the the Rice book. which has output denoted xI (n). There are several possible timing error detection (TED) methods.4. it works nearly as well.1. you will implement the above receiver. • Problem: How do you know when the correct symbol sampling time should be? • Solution: Use a timing locked loop. Section 8. Voltage Controlled Clock (VCC) 5. We focus on the discrete time solutions. analogous to a phased locked loop. • Problem: After the matched ﬁlter. as related in Rice 8.1 Review of Interpolation Filters Timing synchronization is necessary to know when to sample the matched ﬁlter output. you will need to understand how and why they work. and in modern discretetime implementations. which contains much of the code for a BPSK implementation. 33. or a mix. However this solution leads to new problems: • Problem: True interpolation requires signiﬁcant calculation – the sinc ﬁlter has inﬁnite impulse response. Compared to last year’s class. The job of the timing error detector is to produce a voltage error signal e(n) which is proportional to the correct fractional oﬀset between the current sampling oﬀset (ˆ) and the correct µ sampling oﬀset. in order to ﬁx any bugs. Often. Earlylate timing error detector (ELTED) . We want to sample at times [n + µ]Ts . we leave out the Ts and simply talk about the index n or fractional delay µ. 33. This is a good source for detail on many possible methods. Preamble Detector These notes address the motivation and overall design of each block. there may be no analog feedback to the ADC. When you put these together and implement them for QPSK.4. Interpolation Filter 2. • Solution: From samples taken at or above the Nyquist rate. We will talk through two common discretetime implementations: 1. • Solution: Approximate the sinc function with a 2nd or 3rd order polynomial interpolation ﬁlter.2 Timing Error Detection We consider timing error detection blocks that operate after the matched ﬁlter and interpolator in the receiver. discrete time.
3 Earlylate timing error detector (ELTED) Since the derivative x(t) is close to zero at the correct sampling time. and increases away from the correct ˙ sampling time. ˙ 33. to 1) the slope is close to zero. Figure 59 shows that the derivative is a good indication of timing error when the bit is changing from 1. the error e(n) is determined by the slope and the sign of the sample x(n). In general. When the bit is constant (e. x(t). to 1. In ˙ both.5 Figure 59: A sample eyediagram of a RRCshaped BPSK received signal (post matched ﬁlter). at time n. Zerocrossing timing error detector (ZCTED) We’ll use a continuoustime realization of a BPSK received matched ﬁlter output. x(t).5 1 Time from Symbol Sample Time 1. When the bit sequence is The earlylate TED is a discretetime implementation of the continuoustime “earlylate gate” timing error detector. 1 0.5 Slope of x(t) 0 −5 −1. to 1.. Figure 59(a) shows an example eye diagram of the signal x(t) and Figure 59(b) shows its derivative x(t).5 −1 −1. to illustrate the operation of timing error detectors. Here (a) shows the signal x(t) and (b) shows its derivative x(t).5 −1 (b) −0.g.ECE 5520 Fall 2009 120 2.5 1 Time from Symbol Sample Time 1.5 0 0. hopefully with some samples taken at exactly the correct symbol sampling times.5 −1 −4 (a) x 10 5 −0. time 0 corresponds to a correct symbol sampling time. we can use it to indicate how far from the sampling time we might be.5 0 0.5 x(t) Value 0 −0. 1. consider for BPSK the value of x(t)sgn {x(t)} ˙ . You can imagine that the interpolator output xI (n) is a sampled version of x(t). In particular. to 1.
which will be 2 when the sign changes. that indicates a symbol transition. if it was taken at the correct sampling time. and then multiply it by 2 to ﬁnd the gain Kp of your ZCTED. and when the bit is changing from 1. but would mean that we’re ahead for a 1 symbol. For a discrete time system. to 1. which is a function of the excess bandwidth of the RRC ﬁlter used. These are imperfections in the ELTED which (hopefully) average out over many bits. This error detector assumes that the sampling rate is SP S = 2. In particular. In particular see Figure 8. • In the opposite manner. This factor contributes to the gain Kp of this part in the loop. That is. you get only an approximation of the slope unless the sampling rate SP S (or N . the intermediate sample n − 1 should be approximately zero. e(k) = x((k − 1/2)Ts + τ )[ˆ(k − 1) − a(k)] ˆ a ˆ (63) where the a(k) term is the estimate of the kth symbol (either +1 or −1). then the error e(n) = 0. . to 1.ECE 5520 Fall 2009 121 from 1.7 of the Rice book. The signum function sgn {x(t)} just corrects for the sign: • A positive slope would mean we’re behind for a +1 symbol. a negative slope would mean we’re ahead for a +1 symbol. The factor of two comes from the diﬀerence of signs in (66). 33. The error is. Basically. as it is called in the Rice book) is high. In terms of n.4. we don’t divide by the time duration 2Ts because it is just a scale factor. ˆ ˆ (64) (65) The error detector is nonzero at symbol sampling times. If there was a symbol transmission.17 of Rice.4. and we just need something proportional to the timing error.1 of the Rice book.4 Zerocrossing timing error detector (ZCTED) The zerocrossing timing error detector (ZCTED) is also described in detail in Section 8. because every second sample is a symbol sampling time. if the symbol strobe is not activated at sample n. (63) can be rewritten as e(n) = xI (n − 1)[sgn{xI (n − 2)} − sgn{xI (n)}] (66) This operation is drawn in Figure 60.4. we can send alternating bits during the synchronization period in order to help the ELTED more quickly converge. In the project.8 of the Rice book. ˆ ˆ a(k) = sgn {x(kTs + τ )} . to 1. to 1. The theoretical gain in the ZCTED is twice that of the ELTED. you will need to look up Kp in Figure 8. This timing error detector has a theoretical gain Kp which shown in Figure 8. the slope will be somewhat negative even when the sample is taken at the right time. if the sign changes between n − 2 and n. the slope will be somewhat positive even when the sample is taken at the right time. but would mean that we’re behind for a 1 symbol. We’d estimate x(t) from the samples out of the interpolator and see ˙ that e(n − 1) = {xI (n) − xI (n − 2)} sgn {xI (n − 1)} Here. ˆ a(k − 1) = sgn {x((k − 1)Ts + τ )} . It is described here for BPSK systems.
NCO(n2) NCO(n) .. 1 when positive and 1 when negative. When this happens. both the inphase and quadrature signals are used to calculate the error term. and the symbol has switched. (The choice of incrementing or decrementing is arbitrary. The error is simply the sum of the two errors calculated for each signal. .4.100) in the Rice book for the exact expression. where sgn indicates the signum function. 33. The input strobe signal activates the sampler when it is the correct symbol sampling time. the output is 2xI (n − 1).ECE 5520 Fall 2009 122 Timing Error Detector sgn sgn e(n) xI(n) z1 xI(n1) z1 xI(n2) symbol strobe Figure 60: Block diagram of the zerocrossing timing error detector (ZCTED). which would be approximately zero if the sample clock is synchronized. This is shown in Figure 61.5 Voltage Controlled Clock (VCC) We’ll use a decrementing VCC in the project. 33. See (8.) An example trace of the numerically controlled oscillator (NCO) which is the heart of the VCC is shown in Figure 62.1 QPSK Timing Error Detection When using QPSK.. Strobe high: change m to m=NCO(n1)/W(n) 1/2 v(n) W(n) symbol strobe m underflow? symbol strobe z 1 modulo1 NCO(n1) Voltage Controlled Clock (VCC) Figure 61: Block diagram of decrementing VCC with control voltage v(n). NCO(n1) m W(n) Figure 62: Numericallycontrolled oscillator signal N CO(n).
ahead of the carrier. the NCO decrements by 1/SP S + v(n). ˆ The job of the the PLL is to estimate φ(t). It is a function of time. . For example. The function g(θe ) may look something like the one shown in Figure 64. behind the carrier. For those of you familiar with PLLs.6 Phase Locked Loops Carrier synchronization is done using a phaselocked loop (PLL). µ(n) is kept the same.. t. µ(n) = N CO(n − 1) W (n) (67) When the strobe is not activated. the phase detector produces a voltage to correct the phase estimate. Acos(2pfct+q(t)) Phase Detector g(q(t)q(t)) Loop Filter v(t) cos(2pfct+q(t)) VCO t q(t)=k0òv(x)dx ¥ Figure 63: General block diagram of a phaselocked loop for carrier synchronization [Rice]. Once per symbol. then g(θe ) = 0 and VCO output phase remains constant. If the phase estimate is too low. θe (t) = 0. meaning that a sample must be taken. When the symbol strobe is activated. If the phase estimate is too high. the operation of the timing synchronization loop may be best understood by analogy to the continuous time carrier synchronization loop. At each sample n. let’s ignore the loop ﬁlter. 33. 2. because it may not be constant. This is the eﬀect of the ‘S’ curve function g(θe ) in Figure 64. You will prove that (67) is the correct form for µ(n) in the HW 9 assignment. µ(n) = µ(n − 1). The incoming signal is a carrier wave. it may include a frequency oﬀset. the phase oﬀset is a ramp.e. A cos(2πfc t + θ(t)) where θ(t) is the phase oﬀset. Consider a generic PLL block diagram in Figure 63. 1. (on average) the NCO voltage drops below zero.6. We call this estimate θ(t). then g(θe ) > 0 and the VCO increases the phase of the VCO output. In this case. the strobe is set to high. Initially. It is. If the phase estimate is exactly correct. that is. 3. 33. This section is included for reference. that is. When this happens.1 Phase Detector If the estimate of the phase is not perfect. and then θ(t) = ∆f t + α. the fractional interval is also recalculated. then g(θe ) < 0 and the VCO decreases the phase of the VCO output.ECE 5520 Fall 2009 123 The NCO starts at 1. The error in the phase estimate is ˆ θe (t) = θ(t) − θ(t) If the PLL is perfect. i.
as we know. phaseequivalent perspective Q(s) kp F(s) V (s ) Q(s) k0 s Figure 65: Phaseequivalent and linear model for the continuoustime PLL. but leave the frequency the same.6. we model the ‘S’ curve (Figure 64) as a line. . 33. This curve is called an ‘S’ Curve. Note that in addition to the A cos(2πfc t + θ(t)) signal term we also have channel noise coming into the system. we can redraw Figure 63 as it is drawn in Figure 65. This linear approximation is generally ﬁne when the phase error is reasonably small. so we could also write the Ztransform response as F (z). This phaseequivalent model of the PLL allows us to do analysis.3 VCO A voltagecontrolled oscillator (VCO) simply integrates the input and uses that integral as the phase of a cosine wave.ECE 5520 Fall 2009 124 g(qe) qe Figure 64: Typical phase detector inputoutput relationship. 33.4 Analysis To analyze the loop in Figure 63. we write just Θ(s) and Θ(s) even though the physical process. Note that if you just raise the voltage for a short period and then reduce it back (a constant plus rect function input). 33. you change the phase of the output. is the cosine of 2πfc plus that phase. we end up making simpliﬁcations. i. The input is essentially the gas pedal for a race car driving around a track . ˆ In the left half of Figure 65. If we do this.6. We represent the loop ﬁlter in the Laplace or frequency domain as F (s) or F (f ).increase the gas (voltage).6.e. for continuous time.. We will be interested in discrete time later. In particular. in the Laplace domain [Rice]. and the engine (VCO) will increase the frequency of rotation around the track. because the shape of the curve looks like the letter ‘S’ [Rice].2 Loop Filter The loop ﬁlter is just a lowpass ﬁlter to reduce the noise. θe = 0. respectively.
and in case of a frequency oﬀset. The Rice Section C. its implementation is quite eﬃcient. or ramp input.6. given the desired natural frequency θn . 33. the normalized bandwidth Bn T .1 (p 733. k1 . Desired response to particular phase error models. Designing a ﬁlter to respond to these. in Homework 9. For example. See Figure C. Here those goals include: 1. phase error models might be: step input. In this case there are four parameters to set. Note: A accelerating frequency change wouldn’t be handled properly by the given ﬁlter. k2 . which responds to the input during phase oﬀset. will ramp up the phase at the proper rate. damping coeﬃcient ζ. design the particular loop ﬁlter to use in your ﬁnal project. The ﬁlter also has an integrator part (k2 /s) which integrates the phase input.ECE 5520 Fall 2009 125 We can write that ˆ Θe (s) = Θ(s) − Θ(s) = Θ(s) − Θe (s)kp F (s) k0 s To get the transfer function of the ﬁlter (when we consider the phase estimate to be the output). if the temperature of the transmitter or receiver varied quickly. is a challenge. Rice book) for diagrams of this discretetime PLL. then some manipulation shows you that Ha (s) = = ˆ kp F (s) ks0 Θ(s) = Θ(s) 1 + kp F (s) ks0 k0 kp F (s) s + k0 kp F (s) You can use Ha (s) to design the loop ﬁlter F (s) and gains kp and k0 to achieve the desired PLL goals. F (s) = k1 + k2 s This ﬁlter has a proportional part (k1 ) which is just proportional to the input. this is not typically fast enough to be a problem in most radios.57 (p 736) show how to calculate the constants K1 and K2 .2 details how to set these parameters to achieve a desired bandwidth and a desired damping factor (to avoid ringing. and kp .56 and C. Equations C. but not to AWGN noise (as much as possible). 2. However. but to converge quickly). T 1 + z −1 1 → s 2 1 − z −1 Because it is only a secondorder ﬁlter. The Rice book recommends a 2nd order proportionalplusintegrator loop ﬁlter. the frequency of the crystal will also change quickly. The latter deserves more explanation. You will. Desired bandwidth. and other gain constants K0 and Kp . k0 .2. . For example.5 DiscreteTime Implementations You can alter F (s) to be a discrete time ﬁlter using Tustin’s approximation.
• Rice book: Chapters 5 and 6. FSK 4. bipolar vs. No material from 17 will be directly tested. • Homeworks covered: 48. Intersymbol interference.g. • Standard modulation method. This means that the loop ﬁlter eﬀectively averages over many samples when setting the input to the VCO.005). QAM. Lecture 22 Today: (1) Exam 2 Review • Exam 2 is Tue Apr 14. 1119. 9. Please come to me with questions today or tomorrow. 0. • A nonstandard modulation method. union bound.ECE 5520 Fall 2009 126 Note that the bandwidth Bn T is the most critical factor. • I will be out of town MonWed. (e. given signal space diagram. • Oﬃce hours: Today: 23:30. Probability of Error vs. Both know the formula and know how it can be derived. The exam will be proctored. Digital modulation methods: • Binary. 34. Diﬀerential encoding and decoding for BPSK 6. It is typically much less than one.21 are not covered. Detection theory 2. unipolar • M ary: PAM. Nyquist ﬁltering. 34 Exam 2 Topics Where to study: • Lectures covered: 8 (detection theory). . nearestneighbor approximation. Please review these homeworks and know the correct solutions. Friday 1112. Lectures 20. Signal space 3. Gray encoding 8. What topics: 1. Eb N0 . Energy detection for FSK 7. but of course you need to know some things from the start of the semester to be able to perform well on the second part of this course. • Exact expression.. PSK. SRRC pulse shaping 5.
“A mathematical theory of communication. which will be due April 28 at 5pm.. . vol. pp. E. True / false or multiple choice. Short answer. audio.g. Text ﬁles can be presented in diﬀerent ways. Example: What is the probability of bit error vs. Work out solution.html We have done a lot of counting of bits as our primary measure of communication systems. 27. Bandwidth assuming SRRC pulse shaping for PAM/QAM/PSK. System design: Here are some engineering requirements for this communication system. Modulation schemes’ bandwidth eﬃciency is measured in bits per Hertz.ECE 5520 Fall 2009 127 9. email. Eb N0 for this signal space diagram? 4.belllabs. You can use two sides of an 8. or in bits per second. but there may be many wrong answers (or right answer / wrong reason combinations).)? Information can often be represented in many diﬀerent ways. Lecture 23 Today: (1) Source Coding • HW 9 is due today at 5pm.” Bell System Technical Journal. Images and sound can be encoded in diﬀerent ways. and the plot of Qinv. OH today 23pm. 35 Source Coding This topic is a semesterlong course at the graduate level in itself. video. July and October. Everything is measured in bits! But how do we measure the bits of a source (e.cs. but the basic ideas can be presented pretty quickly. and energy eﬃciency is energy per bit over noise PSD. 379423 and 623656. Monday 45pm. 1948. • Final homework: HW 10.. There may be multiple correct answers. with a 12 sentence limit. Shannon. Here are two misconceptions: .5 x 11 sheet of paper. which one would you recommend? What bandwidth and bit rate would it achieve? You will be provided the table of the Qinv function. 2. http://www. 3. One good reference: • C. Comparison of digital modulation methods: • Need for linear ampliﬁers (transmitter complexity) • Receiver complexity • Bandwidth eﬃciency Types of questions (format): 1.. concept of frequency multiplexing 10. FSK. Our information source is measured in bits. From this list of possible modulation.com/cm/ms/what/shannonday/ paper.
. • If you are told the value of a very “random” r. • Lossy: e. For example. So what is the intrinsic measure of bits of text.) above? Sometimes. but truly black or white). JPEG..v. .tar ﬁles represent the exact same information that was contained in the original ﬁles. in units of bits. that doesn’t vary that much. Randomness and information.v.. Using the ﬁle size tells you how much information is contained within the ﬁle. This would introduce errors into the image. Two types of compression algorithms: • Lossless: e. but with fewer bits. are just two perspectives on the same thing: • If you are told the value of a r.g. For example. 2. Then the entropy of X. consider a digital black & white image (not grayscale.} is an ordering of the possible values in SX . H [X] = − pi log2 pi (68) i Notes: . 1. audio.g. Take the log2 of the number of diﬀerent messages you could send. and x ∈ SX . 2. How many bits would be used in these two representations? What would make you decide which one is more eﬃcient? You can see that equivalent representations can use diﬀerent number of bits.. that telling conveys very little information to you.all black pixels). . This is the idea behind source compression. You could store it as a list of pixels.1 Entropy Entropy is a measure of the randomness of a random variable. there is a ﬁnite or countably inﬁnite set SX . in nontechnical language. We will shorten the notation by using pi as follows: pi = pX (xi ) = P [X = xi ] where {x1 . What if we had a ﬁxed number of bits to send any image.g. Each pixel has two possibilities (possible messages). MP3 Note: Both “zip” and the unix “compress” commands use the LempelZiv algorithm for source compression. that telling conveys quite a bit of information to you. Our technical deﬁnition of entropy of a random variable is as follows. and we used the sparse B&W image coding scheme (2. x2 .zip or . Here. thus we could encode it in log2 2 or one bit per pixel. Def ’n: Entropy Let X be a discrete random variable with pmf pX (xi ) = P [X = x]. You could simply send the coordinates of the pixels of one of the colors (e.ECE 5520 Fall 2009 128 1. or video? 35. . an image. is deﬁned as. the number of bits in the compressed image would exceed what we had allocated. Zip or compress.
This is true in the limit of x log x as x → 0+ .v. .w. not another random variable.8 Entropy H[X] 0. o. X to have pmf 1/16. x = 1 1/2.audio). What is the entropy H [X] as a function of s? Solution: Entropy is given by (68) and is: H[X] = −s log2 s − (1 − s) log2 (1 − s) The solution is plotted in Figure 66. they express information in “nats”. x = 2 1/4. x = −2 0. x = 0 0. • Use that 0 log 0 = 0. x = 0 pX (x) = 1/8.g. x=1 s.w. Model the r. Example: Binary r. not a function of a random variable.v.8 1 Figure 66: Entropy of a binary r. This it is like E [X]. . when theorists use loge or the natural log. • The sum will be from i = 1 .4 0. another operator on a random variable. It returns a (deterministic) number. • All “log” functions are logbase2 in information theory unless otherwise noted. x = −1 1/16.ECE 5520 Fall 2009 129 • H [X] is an operator on a random variable.v.6 P[X=1] 0.v.2 0 0 0. pX (x) = 1 − s. A binary (Bernoulli) r. Keep this in mind when reading a book on information theory. Nothing else is needed.6 0. Actually. short for “natural” digits. has pmf. o. The “reason” the units are bits is because of the base2 of the log. .2 0. Example: Nonuniform source with ﬁve messages Some signals are more often close to zero (e. 1 0.4 0. • Entropy of a discrete random variable X is calculated using the probability values of the pmf of X. pi . N when SX  = N < ∞.
random variables has N times the entropy of any one of them. XN ] = −N pX1 (x1 ) log2 pX1 (x1 ) = N H(X1 ) x1 ∈SX1 The entropy of N i. the joint entropy of N r. .. . X2 compared just to one of them? This is often an important question because it answers the question... X1 . because of the dependence or correlation.. .v. random variables? You can show that H[X1 . (71) . “How much additional information do I get from both.ECE 5520 Fall 2009 130 What is its entropy H [X]? Solution: H [X] = = Other questions: 1. entropy is H[X1 .s is just the sum of the entropy of each individual r.X2 (x1 . In addition. XN . . if you know some of them. . XN ] = − ··· pX1 . . since pixels are correlated in space. 35.v. . . For example. . .. . the entropy of any N independent (but possibly with diﬀerent distributions) r.s is less than N times the entropy of one of them.i.XN (x1 .v. X2 ] = − pX1 . . X1 ] − H[X1 ]. x2 ) log2 pX1 . .s are not independent. When r.. . Would multiplying X by 2 change its entropy? 3. . Would an arbitrary onetoone function change the entropy of X? 1 1 1 1 log2 2 + log2 4 + log2 8 + 2 log2 16 2 4 8 16 15 bits 8 (69) 35. .d. Intuitively. X2 with event sets SX1 and SX2 is deﬁned as H[X1 . . x2 ) x1 ∈SX1 x2 ∈SX2 (70) For N joint random variables. .d.i.v.. xN ) xN ∈SXN x1 ∈SX1 What is the entropy for N i. the joint r. .3 Conditional Entropy How much additional entropy is in the joint random variables X1 . xN ) log2 pX1 . Do you need to know what the symbol set SX is? 2. of several neighboring pixels will have less entropy than the sum of the individual pixel entropies.v. .2 Joint Entropy Def ’n: Joint Entropy The joint entropy of two random variables X1 . compared to just one of them?”.X2 (x1 .. H[X2 X1 ]: H[X2 X1 ] = H[X2 . We call this diﬀerence the conditional entropy.XN (x1 . the rest that you don’t know become less informative. the B&W image.
given the values of the N − 1 previous random variables. XN ]. . we are more interested in the rate..k. x2 ) + x1 ∈SX1 x2 ∈SX2 x1 ∈SX1 pX1 (x1 ) log2 pX1 (x1 ) pX1 . source output).X2 (x1 . N →∞ N Example: Entropy of English text Let Xi be the ith letter or space in a common English sentence.X2 (x1 . x2 ) + x1 ∈SX1 x2 ∈SX2 x1 ∈SX1 x2 ∈SX2 x1 ∈SX1 x2 ∈SX2 pX1 . N →∞ It can be shown that entropy rate can equivalently be written as H = lim 1 H[X1 . 35. . . the joint entropy of all of them may go to inﬁnity as N → ∞. . x1 ) xN−1 ∈SXN−1 x1 ∈SX1 which is the additional entropy (or information) contained in the N th random variable.ECE 5520 Fall 2009 131 What is an equation for H[X2 X1 ] as a function of the joint probabilities pX1 . X2 . H[X2 X1 ] = − = − = − = − = − pX1 . x2 ) pX1 (x1 ) pX1 .. in the limit. For this case. X1 ] and H[X1 ].. . . x2 ) log2 pX1 (x1 ) x1 ∈SX1 x2 ∈SX2 x1 ∈SX1 x2 ∈SX2 pX1 .v. .X2 (x1 .X2 (x1 .X2 (x1 . x2 ) log2 pX2 X1 (x2 x1 ) (72) Note the asymmetry – there is the joint probability multiplied by the log of the conditional probability.4 Entropy Rate Typically. X1 ].X2 (x1 .. . X2 . . X1 ] = − ··· pX1 . x2 ) log2 pX1 . is deﬁned as H = lim H[XN XN −1 .a.XN (x1 . H[XN XN −1 . . we’re interested in discretetime random processes. in which we have random variables X1 .. .X2 (x1 .. How many additional bits. This is not like either the joint or the marginal entropy. x2 ) log2 pX1 . We could also have multivariate conditional entropy. . . . .X2 (x1 . as N → ∞? Def ’n: Entropy Rate The entropy rate of a stationary discretetime random process.X2 (x1 . . in units of bits per random variable (a.X1 (xN xN −1 . x2 ) (log2 pX1 . Solution: Plugging in (68) for H[X2 . . . x2 ) and the conditional probabilities pX2 X1 (x2 x1 ). .X2 (x1 . What is the sample space SXi ? Is Xi uniform on that space? .... xN ) log2 pXN XN−1 . x2 ) log2 pX1 . Since there are inﬁnitely many of them. are needed for the average r. x2 ) − log2 pX1 (x1 )) pX1 .X2 (x1 .
04 = 3 · 3.5 Source Coding Theorem The key connection between this mathematical deﬁnition of entropy and the bit rate that we’ve been talking about all semester is given by the source coding theorem.3 bits/letter [taken from Proakis & Salehi Section 6.3. For fourletter combinations.2]. What is H[Xi . which is 2 · 3. 35. I calculate an entropy of 7. The Proakis & Salehi book mentions that this value for general English text is about 4. Notes: • Here. See Shannon’s original 1948 paper.35.01 0.015 0.1 0.ECE 5520 Fall 2009 132 What is H[Xi ]? Solution: I had Matlab read in the text of Shakespeare’s Romeo and Juliet.46. It is one of the two fundamental theorems of information theory.99. independent of the complexity of the encoder and the decoder employed.025 0. I calculated an entropy of H = 4. This gives me the twodimensional pmf shown in Figure ??(b). an ‘error’ occurs when your compressed version of the data is not exactly the same as the original. See Figure 67(a).1199.02 0. Theorem: A source with entropy rate H can be encoded with arbitrarily small error probability. I calculated the entropy of the joint pmf of each twoletter combination.15 u p k f Poccurrence 0. For the threeletter combinations.05 0 (a) sp f k p Space or Letter u z (b) sp sp f k p u Second Space or Letter z Figure 67: PMF of (a) single letters and (b) twoletter combinations (including spaces) in Shakespeare’s Romeo and Juliet. Example: B&W images. H? Solution: For N = 10. the joint entropy was 10. and was introduced by Claude Shannon in 1948. Conversely. • What is the minimum possible rate to encode English text (if you remove all punctuation)? • The theorem does not tell us how to do it – just that it can be done. For this pmf.98 = 4 · 2. • Theorem fact: Information measure (entropy) gives us a minimum bit rate. You can see that the average entropy in bits per letter is decreasing quickly. if R < H.73.03 First Space or Letter 0. we have H = 1. the joint entropy was 11.2 z 0. • R is our 1/Tb . Xi+1 ]? Solution: Again.005 0. the error probability will be bounded away from zero. Proof: Proof: Using typical sequences. at any rate R (bits / source output) as long as R > H. using Matlab on Shakespeare’s Romeo and Juliet. 0. . What is the entropy rate.
which is aﬀected by additive White Gaussian noise (AWGN). which says that a source with entropy rate H can be encoded with arbitrarily small error probability. for a ﬁnite source. . Hartley was particularly inﬂuenced by Nyquist’s result. Here he developed the “Hartley oscillator”. depending on how pulse amplitudes were chosen. 1 ≤ 2B. degree from the University of Utah in 1909. N →∞ N We showed that entropy can be used to quantify information.B. Lecture 24 Today: (1) Channel Capacity 36 Review H[X] = − pi log2 pi i Last time.ECE 5520 Fall 2009 133 • The theorem does not tell us how well it can be done if N is not inﬁnite. we turn to the noisy channel. Any lower rate than H would guarantee loss of information. That is. the rate may need to be higher. Hartley Ralph V. In particular. as described by Nyquist. Hartley assumed that the maximum amplitude available to the transmitter was A. we deﬁned entropy. Afterwards. if they . involved in radio telephony. When transmitting a sequence of pulses. Nyquist determined that the pulse rate was limited to two times the available channel bandwidth B. Given our information source X or {Xi }. XN ]. V. he published in the Bell System Technical Journal a paper on “Transmission of Information”. He worked as a researcher for the Western Electric Company. L. and entropy rate. . . 37. But. at any rate R (bits / source output) as long as R > H. 1 H[X1 . This discussion of entropy allows us to consider the maximum data rate which can be carried without error on a bandlimited channel. Hartley made the assumption that the communication system could discern between pulse amplitudes. 1888) received the A.1 R. L. at Bell Laboratories. Tsy In Hartley’s 1928 paper. he considered digital transmission in pulseamplitude modulated systems. In July 1928. X2 . Hartley (born Nov. he developed relationships useful for determining the capacity of bandlimited communication channels. . the source data. 30. the value of H[X] or H gives us a measure of how many bits we actually need to use to encode. Then. H = lim 37 Channel Coding Now. without loss. The pulse rate was limited to 2B. The major result was the Shannon’s source coding theorem. each pulse could represent more or less information. each of duration Tsy .
his capacity bound considers simultaneously demodulating sequences of received symbols. . In fact. .1 Noisy Channel Shannon did take into account an AWGN channel. .2. we need a law called the law of large numbers (ECE 6962 topic). yi = xi + zi where X is the transmitted signal and Z is the noise in the channel. as n → ∞. In other words. xn ] simultaneously.ECE 5520 Fall 2009 134 were at separated by at least a voltage spacing of Aδ . n 1 (yi − xi )2 ≤ PN n i=1 happens with probability one. yn ]. in bits per second. This late decision will decide all values of y = [x1 . of length n. Hartley used the ‘bit’ measure to quantify the data which could be encoded using M amplitude levels. This asymptotic limit as n → ∞ allows for a proof using the statistical convergence of a sequence of random variables. . 37. the capacity formula was for a particular type of PAM system. Shannon What was left unanswered by Hartley’s capacity formula was the relationship between noise and the minimum amplitude separation between symbols. assuming perfect syncronization) is yi . Next. Given that a PAM system operates from 0 to A in increments of Aδ . . The noise term zi is assumed to be i. 37.d. the measured value y will be located √ within an ndimensional sphere (hypersphere) of radius nPN with center x. . y = [y1 . All n symbols are received before making a decision. regardless of modulation type. Furthermore. and used statistics to develop a universal bound for capacity. Hartley quantiﬁed the data rate using Nyquist’s relationship to determine the maximum rate C. In this AWGN channel. log2 M = log2 1 + A Aδ Finally. Shannon’s proof considers the limiting case as n → ∞.2 C.2 Introduction of Latency Shannon’s key insight was to exchange latency (time delay) for reduced probability of error. as n → ∞. .i. In particular. 37. Gaussian with variance PN . the ith symbol sample at the receiver (after the matched ﬁlter. and did not deal well with negative amplitudes. . This law says that the following event. . C = 2B log2 1 + A Aδ (Note that the Hartley transform came later in his career in 1942). possible from the digital communication system.2. Engineers would have to be conservative when setting Aδ to ensure a low probability of error. and did not say anything fundamental about the relationship between capacity and bandwidth for arbitrary modulation. Further. E. the number of diﬀerent pulse amplitudes (symbols) is M =1+ A Aδ Note that early receivers were modiﬁed AM envelope detectors.
within a period of n sample times.1 Returning to Hartley Adjusting Hartley’s formula. Shannon’s formulation asks us how many multidimensional amplitudes xi √ be ﬁt can into a hypersphere of radius n(P + PN ) centered at the origin. by PN = N0 B . The result of this geometrical problem is that P ( n/2) (73) M = 1+ PN 37. show how we many diﬀerent symbols we could have uniquely distinguished. A] such that they are all separated by Aδ . That is. n 1 x2 ≤ P i n i=1 This combination of signal power limitation and noise power results in the fact that.3. together. 1 n As a result y 2 n i=1 2 yi ≤ 1 n n x2 + i i=1 1 n n i=1 (yi − xi )2 ≤ P + PN ≤ n(P + PN ) This result says that the vector y. This is shown in P&S in Figure 9.3. 584). Hartley asked how many symbol amplitudes could be ﬁt into [0. we can replace the noise variance PN with the relationship between noise twosided PSD N0 /2.3 Combining Two Results The two results.3 Introduction of Power Limitation Shannon also formulated the problem as a powerlimited case. 37. in which the average power in the desired signal xi was limited to P .ECE 5520 Fall 2009 135 37. with probability one as n → ∞.2.9 (pg. This number M is the number of diﬀerent messages that could have been sent in n pulses. if we could send M messages now in n pulses (rather than 1 pulse) we would adjust capacity to be: 2B C= log2 M n Using the M from (76) above.2 Final Results Finally. is contained within a hypersphere of radius n(P + PN ) centered at the origin. such that hyperspheres of radius nPN do not overlap. C = P 2B n log2 1 + n 2 PN P = B log2 1 + PN 37.
• The ﬁnal project is due Tue.4 Eﬃciency Bound Recall that Eb = P Tb . . Shannon also proved that any system which operates at a bit rate higher than the capacity C will certainly incur a positive bit error rate. Note that the ratio NPW is the signal power divided by the noise power. This relationship is shown in Figure 70. or signal to noise ratio (SNR). Thus 0 the capacity bound is also written C = W log2 (1 + SN R). • Everyone gets a 100% on the “Discussion Item” which I never implemented. 37. C is just a capacity limit. “Digital Terrestrial Television Broadcasting”. Any practical communication system must operate at Rb < C. is canceled. where Rb is the operating bit rate. so Rb Eb Rb ≤ log2 1 + W W N0 Deﬁning η = Rb W. May 5. Thus from (74). please the Robert Graves lecture 3:05 PM in Room 105 WEB. because of the late due date. set your deadline to May 4. 0 Lecture 25 Today: (1) Channel Capacity • Lecture Tue April 28. you can look at it as a bound on the bandwidth E eﬃciency as a function of the Nb ratio. C = W log2 1 + Eb /Tb N0 W Rb Eb W N0 Here. 5pm. That is. the energy per bit is the power multiplied by the bit duration. C = B log2 1 + Typically.ECE 5520 Fall 2009 136 So ﬁnally we have the ShannonHartley Theorem. η ≤ log2 1 + η Eb N0 This expression can’t analytically be solved for η. people also use W = B so you’ll see also C = W log2 1 + P N0 W (75) P N0 B (74) This result says that a communication system can operate at bit rate C (in a bandlimited channel with width W given power limit P and noise value N0 ). C = W log2 1 + or since Rb = 1/Tb . No late projects are accepted. However. If you think you might be late. Instead. We know that our bit rate Rb ≤ C. with arbitrarily low probability of error.
on a (a) linear plot and (b) log plot. bound on bandwidth eﬃciency. . η.ECE 5520 Fall 2009 137 14 12 Bits/second per Hertz 10 8 6 4 2 0 0 5 10 15 20 E /N ratio (dB) b 0 25 30 (a) 10 1 Bits/second per Hertz 10 0 10 −1 0 5 (b) 10 15 20 E /N ratio (dB) b 0 25 30 Figure 68: From the ShannonHartley theorem.
he developed relationships useful for determining the capacity of bandlimited communication channels. . Nyquist determined that the pulse rate was limited to two times the available channel bandwidth B. the value of H[X] or H gives us a measure of how many bits we actually need to use to encode. he considered digital transmission in pulseamplitude modulated systems. we deﬁned entropy. Hartley was particularly inﬂuenced by Nyquist’s sampling theorem. Hartley (born Nov. . he published in the Bell System Technical Journal a paper on “Transmission of Information”. involved in radio telephony. In July 1928. as described by Nyquist. and did not deal well with negative amplitudes. log2 M = log2 1 + A Aδ . the number of diﬀerent pulse amplitudes (symbols) is M =1+ A Aδ Note that early receivers were modiﬁed AM envelope detectors. 39 Channel Coding Now. Given our information source X or {Xi }. depending on how pulse amplitudes were chosen. degree from the University of Utah in 1909. we turn to the noisy channel. 30. each of duration Ts . Hartley assumed that the maximum amplitude available to the transmitter was A. L. at Bell Laboratories. N We showed that entropy can be used to quantify information. if they were at separated by at least a voltage spacing of Aδ . X2 . When transmitting a sequence of pulses. Ts In Hartley’s 1928 paper. each pulse could represent more or less information. Hartley Ralph V. . Then. which is aﬀected by additive White Gaussian noise (AWGN). 39. Any lower rate than H would guarantee loss of information.B.1 R. Next. Afterwards. at any rate R (bits / source output) as long as R > H. Given that a PAM system operates from 0 to A in increments of Aδ . and entropy rate. The major result was the Shannon’s source coding theorem. which says that a source with entropy rate H can be encoded with arbitrarily small error probability. L. without loss. Hartley made the assumption that the communication system could discern between pulse amplitudes. In particular. 1888) received the A. H = lim N →∞ 1 H[X1 . XN ]. the source data. But. V. 1 ≤ 2B. .ECE 5520 Fall 2009 138 38 Review H[X] = − pi log2 pi i Last time. He worked as a researcher for the Western Electric Company. Hartley used the ‘bit’ measure to quantify the data which could be encoded using M amplitude levels. The pulse rate was limited to 2B. This discussion of entropy allows us to consider the maximum data rate which can be carried without error on a bandlimited channel.
In either case. his capacity bound considers ndimensional signaling.2 C. . Hartley quantiﬁed the data rate using Nyquist’s relationship to determine the maximum rate C. assuming perfect synchronization) is yi . This law says that the following event.3 Introduction of Power Limitation Shannon also formulated the problem as a energylimited case. the ith symbol sample at the receiver (after the matched ﬁlter.ECE 5520 Fall 2009 139 Finally. and did not say anything fundamental about the relationship between capacity and bandwidth for arbitrary modulation.. this late decision will decide all values of x = [x1 . and used statistics to develop a universal bound for capacity. . as n → ∞.g. n 1 2 xi ≤ E n i=1 . we need a law called the law of large numbers. Shannon uses all n dimensions in the constellation – the detector must use all n samples of y to make a decision. . FSK). . 39. C = 2B log2 1 + A Aδ 39. the measured value y will be located √ within an ndimensional sphere (hypersphere) of radius nEN with center x. or they might use multiple symbols over time (recall that symbols at diﬀerent multiples of Ts are orthogonal). The noise term zi is assumed to be i. In fact. These might be truly an ndimensional signal (e. yn ]. Shannon What was left unanswered by Hartley’s capacity formula was the relationship between noise and the minimum amplitude separation between symbols. Gaussian with variance EN = N0 /2. 39. So the received vector is y = [y1 . In other words.2. In this AWGN channel. Further. .d. . of length n. That is. . 1 n n i=1 (yi − xi )2 ≤ EN happens with probability one. Furthermore. Shannon’s proof considers the limiting case as n → ∞. yi = xi + zi where X is the transmitted signal and Z is the noise in the channel. In the multiple symbols over time. in bits per second. This asymptotic limit as n → ∞ allows for a proof using the statistical convergence of a sequence of random variables. . in which the maximum symbol energy in the desired signal xi was limited to E. E. the capacity formula was for a particular type of PAM system. In particular. regardless of modulation type.1 Noisy Channel Shannon did take into account an AWGN channel.2. Engineers would have to be conservative when setting Aδ to ensure a low probability of error. as n → ∞. 39.2. xn ] simultaneously.2 Introduction of Latency Shannon’s key insight was to exchange latency (time delay) for reduced probability of error. possible from the digital communication system.i.
ECE 5520 Fall 2009 140 This combination of signal energy limitation and noise energy results in the fact that. with probability one as n → ∞. 1 n As a result y 2 n i=1 2 yi ≤ 1 n n x2 + i i=1 1 n n i=1 (yi − xi )2 ≤ E + EN This result says that the vector y. This is shown in Figure 69. is contained within a hypersphere of radius n(E + EN ) centered at the origin.3. A] such that they are all separated by Aδ .1 Returning to Hartley Adjusting Hartley’s formula. show how we many diﬀerent symbols we could have uniquely distinguished. if we could send M messages now in n pulses (rather than 1 pulse) we would adjust capacity to be: 2B C= log2 M n Using the M from (76) above. together. Shannon’s formulation asks us how many multidimensional amplitudes xi √ be ﬁt can into a hypersphere of radius n(E + EN ) centered at the origin. radius nEN ) EN E+ n( Figure 69: Shannon’s capacity formulation simpliﬁes to the geometrical question of: how many hyperspheres of √ a smaller radius nEN ﬁt into a hypersphere of radius n(E + EN )? This number M is the number of diﬀerent messages that could have been sent in n pulses. The result of this geometrical problem is that E n/2 (76) M = 1+ EN 39.3 Combining Two Results The two results. ≤ n(E + EN ) 39. C= 2B n E log2 1 + n 2 EN = B log2 1 + E EN . within a period of n sample times. Hartley asked how many symbol amplitudes could be ﬁt into [0. such that hyperspheres of radius nEN do not overlap.
This relationship is shown in Figure 70. Shannon also proved that any system which operates at a bit rate higher than the capacity C will certainly incur a positive bit error rate. or signal to noise ratio (SNR). i.e. . the energy per bit is the maximum power multiplied by the bit duration. Thus from (77). Note that the ratio NEB is the signal power divided by the noise power.3. Thus 0 the capacity bound is also written C = B log2 (1 + SN R). with arbitrarily low probability of error.2 Final Results P Since energy is power multiplied by time.. C is just a capacity limit. C = B log2 1 + Rb Eb B N0 Here. 39. we have the ShannonHartley Theorem. E = P Ts = 2B where P is the maximum signal power and B is the bandwidth. Figure 71 is the plot on a logy 0 axis with some of the modulation types discussed this semester. Eb = P Tb . Be know that our bit rate Rb ≤ C. so Rb Rb Eb ≤ log2 1 + B B N0 Deﬁning η = Rb B (the spectral eﬃciency). η ≤ log2 1 + η Eb N0 This expression can’t analytically be solved for η. Eb /Tb C = B log2 1 + N0 B or since Rb = 1/Tb .ECE 5520 Fall 2009 141 39. where Rb is the operating bit rate. and EN = N0 /2. However. That is.4 Eﬃciency Bound Another way to write the maximum signal power P is to multiply it by the bit period and use it as the maximum energy per bit. Any practical communication system must operate at Rb < C. (77) This result says that a communication system can operate at bit rate C (in a bandlimited channel with width B given power limit E and noise value N0 ). C = B log2 1 + P N0 B . you can look at it as a bound on the bandwidth E eﬃciency as a function of the Nb ratio.
η. bound on bandwidth eﬃciency. 10 1 Bandwidth Efficiency (bps/Hz) 10 0 1024−QAM 256−QAM 64−QAM 64−PSK 16−QAM 32−PSK 16−PSK 8−PSK 4−QAM 4−FSK 16−FSK 64−FSK 10 −1 256−FSK 10 −10 −2 1024−FSK 0 10 Eb/N0 (dB) 20 30 Figure 71: From the ShannonHartley theorem bound with achieved bandwidth eﬃciencies of MQAM. and MFSK. MPSK. .ECE 5520 Fall 2009 142 14 12 Bits/second per Hertz 10 8 6 4 2 0 0 5 10 15 20 Eb/N0 ratio (dB) 25 30 Figure 70: From the ShannonHartley theorem.
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