ECE 5520: Digital Communications

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 Discrete-Time Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14
3.2 Decibel Notation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14
4 Time-Domain Concept Review 15
4.1 Periodicity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
4.2 Impulse Functions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
5 Bandwidth 16
5.1 Continuous-time 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 Multi-Variate 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 Matched-Filter 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 Log-Likelihood 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 Equi-probable 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 M-ary 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 M-ary 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 Inter-symbol 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 Square-Root Raised Cosine Filtering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 72
23 M-ary Detection Theory in N-dimensional 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 M-QAM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79
24.6 Phase-Shift 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 Bound-based Probability of Error . . . . . . . . . . . . . . . . 84
ECE 5520 Fall 2009 5
26 QAM Probability of Error 85
26.1 Nearest-Neighbor 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 Non-coherent 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 M-ary Non-Coherent 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 / Non-linear 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 Non-free-space Channels . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
30.3.3 Wired Channels . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
30.4 Computing Noise Energy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112
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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 Early-late timing error detector (ELTED) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 120
33.4 Zero-crossing 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 Discrete-Time 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 well-written. 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 discrete-time, discrete-valued information across a physical channel.
Information sources might include audio, video, text, or data. They might be continuous-time (analog) signals
(audio, images) and even 1-D or 2-D. Or, they may already be digital (discrete-time, discrete-valued). 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., real-valued, continuous-time
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 7-layer 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 5-layer 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, mm-wave 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
Up-conversion
Down-
conversion
OtherComponents:
Digitalto Analog
Converter
AnalogtoDigital
Converter
Synchron-
ization
Figure 1: Block diagram of a single-user 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
continuous-time signals, and quantization of continuous-valued 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 digital-to-analog conversion which produces a continuous-time 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 source-channel 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 non-Gaussian noise distribution, or
non-white 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 time-delayed 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 time-invariant, 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.,
TCP-IP) can determine that an error occurred and then re-request 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 M-ary 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. Pre-requisites: (ECE 5510) Random Processes; (ECE 3500) Signals and Systems.
2. Signal Processing: (ECE 5530): Digital Signal Processing
3. Electromagnetics: EM Waves, (ECE 5320-5321) 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) Time-domain 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 non-zero 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 Discrete-Time 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 discrete-time function; whenever you see a function of t (or perhaps
τ) it is a continuous-time function. I’m sorry this is not more obvious in the notation.
For discrete-time 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 fiber-optic 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 Time-Domain Concept Review
4.1 Periodicity
Def ’n: Periodic (continuous-time)
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 (discrete-time)
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 continuous-time 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 discrete-time 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
Continuous-Time 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

_

ω=−∞
X(jω)e
jωt

Discrete-Time z-Transform
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

N
kn
X(e
jΩ
) =


n=−∞
x(n)e
−jΩn
x(n) =
1
N

N−1
k=0
X[k]e
j

N
nk
x(n) =
1

_
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
• 3-dB 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 square-root 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 Continuous-time Frequency Transforms
Notes about continuous-time 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

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

= 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

_
= 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

_

ω=−∞
[X(jω)[
2

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) =

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

1 +jω
, and X(jω) =
1
2
Z(jω/2).
So X(jω) =
1
2
e

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

R(jω).
Again, X(jω) =
1
2
e

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

1+jω
= −

2
1+jω
.
4. Let x(t) = y(−t), where y(t) = w(1 +t). Then
X(jω) = Y (−jω), where Y (jω) = e

W(jω) = e


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

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 low-pass 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

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) ⋆

T

n=−∞
δ
_
ω −
2πn
T
_
=

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 discrete-time 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 continuous-time signal? A: Using (10). We have that X
d
(e

) = X
sa
_

2πT
_
. Then
plugging into (10),
Solution:
X
d
(e
jΩ
) = X
sa
_
j

T
_
=

T

n=−∞
X
_
j
_

T

2πn
T
__
=

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, continuous-time 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 down-conversion 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 one-half the sampling rate).
Example: Bandpass Sampling
This is Example 2.6.2 (pp. 60-65) 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

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 discrete-time frequency is modulo 2π, so the
center frequency of the sampled signal is:

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 (continuous-time, real-valued) 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 ‘in-between’ 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 = |x||y| cos θ
where θ is the angle between vectors x and y. In words, the inner product is a measure of ‘same-direction-ness’:
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 cross-directions.
8.2 Inner Product of Functions
The inner product is not limited to vectors. It also is applied to the ‘space’ of square-integrable real functions,
i.e., those functions x(t) for which
_

−∞
x
2
(t)dt < ∞.
The ‘space’ of square-integrable 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

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

cos(4πt)
¸
¸
¸
¸
1
0
=
−1

(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 Walsh-Hadamard functions.
Example: Walsh-Hadamard 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: Walsh-Hadamard 2-length and 4-length functions in image form. Each function is a row of the image,
where crimson red is 1 and black is -1.
Figure 14: Walsh-Hadamard 64-length 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 half-time 1 and half-time -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 Walsh-Hadamard
This example is just for your information (not for any test). The set of 64-length Walsh-Hadamard sequences
ECE 5520 Fall 2009 32
is used in CDMA (IS-95) in two ways:
1. Reverse Link: Modulation: For each group of 6 bits to be send, the modulator chooses one of the 64
possible 64-length 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
Walsh-Hadamard functions are used in IS-95 reverse link). An orthonormal basis for an arbitrary signal set
tells us:
• how to build the receiver,
• how to represent the signals in ‘signal-space’, 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: Position-shifted 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 position-shifted 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: Amplitude-shifted 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 amplitude-shifted 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
• Self-interference
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 Walsh-Hadamard 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 Multi-Variate 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 non-negative. The CDF is non-negative and non-decreasing,
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 continuous-valued 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 continuous-valued.
Work-around: 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 non-negative 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

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 non-zero mean and non-unit-variance, 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 left-hand side is a unit-variance, zero mean Gaussian random variable, we can write the probability
of this event using the unit-variance, zero mean Gaussian CDF.
10.6.1 Complementary CDF
The probability that a unit-variance, 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


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


_

2x
0
e
−u
2
/2
du
= 2
_

2x
0
1


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 zero-mean 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 n-length 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

2
X
i
_
Section 4.3 spends some time with 2-D 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 2-D 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

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

2
_
I
0
_
rs
σ
2
_
, r ≥ 0
0, o.w.
where s
2
= µ
2
1

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: Wide-sense stationary (WSS)
A random process is wide-sense 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 wide-sense 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, wide-sense stationary random process from one
realization of a power-type 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 non-ergodic process of opinion polling. Consider what
happens when a pollster takes either a time-average 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 non-ergodic.
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 Matched-Filter 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 non-orthogonal,
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).
Noise-free 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 down-conversion,
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
X|H
0
(x[H
0
)P [H
0
] dx
+
_
x∈R
0
f
X|H
1
(x[H
1
)P [H
1
] dx
= P [H
0
] (19)
+
_
x∈R
0
_
f
X|H
1
(x[H
1
)P [H
1
] −f
X|H
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
X|H
0
(x[H
0
)P [H
0
] > f
X|H
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
X|H
1
f
X|H
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
X|H
0
(x[H
0
) and f
X|H
1
(x[H
1
), (b) joint p.d.f.s
f
X|H
0
(x[H
0
)P [H
0
] and f
X|H
1
(x[H
1
)P [H
1
], and (c) difference between the joint p.d.f.s, f
X|H
1
(x[H
1
)P [H
1
] −
f
X|H
0
(x[H
0
)P [H
0
], which is the integrand in (19).
ECE 5520 Fall 2009 55
Rearranging the terms,
f
X|H
1
(x[H
1
)
f
X|H
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 Log-Likelihood 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 log-likelihood ratio is
log
f
X|H
1
(x[H
1
)
f
X|H
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 log-likelihood ratio?
3. The decision regions for x?
Solution: What is the log of a Gaussian pdf?
log f
X|H
0
(x[H
0
) = log
_
_
1
_
2πσ
2
N
e

x
2

2
N
_
_
= −
1
2
log(2πσ
2
N
) −
x
2

2
N
(21)
The log f
X|H
1
(x[H
1
) term will be the same but with (x − 1)
2
instead of x
2
. Continuing with the log-likelihood
ratio,
log f
X|H
1
(x[H
1
) −log f
X|H
0
(x[H
0
) < log
P [H
0
]
P [H
1
]
x
2

2
N

(x −1)
2

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

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

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

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

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 Equi-probable 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 half-way 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, one-dimensional
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 equi-probable symbol case, and in the general case.
Lecture 9
Today: (1) Finish Lecture 8 (2) Intro to M-ary PAM
ECE 5520 Fall 2009 58
Sample exams (2007-8) 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)
M-ary 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 time-delayed 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 continuous-time and discrete-time realizations, respectively, of PAM
transmitters and receivers.
Example: Binary Bipolar PAM
Figure 22 shows a 4-ary 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 On-Off Keying (OOK). If the y-axis in Figure 22 was scaled such that the minimum was 0 instead
of −1, it would represent unipolar PAM.
Example: 4-ary PAM
A 4-ary 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 M-ary 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 M-ary PAM for different values of M.
15.2 Average Bit Energy in M-ary 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 4-ary PAM example.
Let ¦a
0
, . . . , a
M−1
¦ = −(M −1)A, −(M −3)A, . . . , (M −1)A, as described above to be the typical M-ary 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 1-3. It will be good to know the ‘tricks’ of how
each problem is done, since they will reappear.
Read over the lecture notes 1-7. 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 continuous-time 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 2-49). Three functions are shown in Figure P2-49.
(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 2-80) An RC low-pass filter is the circuit shown in Figure 24. Its transfer
function is
X(t) Y(t)
R
C
+ +
- -
Figure 24: An RC low-pass 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 low-pass
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 1-D 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 zero-mean Gaussian with variance 0.1, and n
1
is zero-mean 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 equally-likely 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 energy-type or power-type 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 zero-mean, 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) M-ary 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 1-D, 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
[

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 re-write 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) Multi-Hypothesis Detection Theory, (2) Probability of Error in M-ary
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
X|H
0
(x[H
0
)P [H
0
] and f
X|H
1
(x[H
1
)P [H
1
]
For M-ary 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
X|H
0
(x[H
0
)P [H
0
]
H
1
: f
X|H
1
(x[H
1
)P [H
1
]

H
M−1
: f
X|H
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 equi-probable 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
X|H
i
(x[H
i
) maximum, that is, maximum likelihood detection.
Symbol Decision = arg max
i
f
X|H
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
X|H
i
(x[H
i
) = −
1
2
log(2πσ
2
N
) −
(x −a
i
)
2

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: 4-ary 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 4-ary PAM.
ECE 5520 Fall 2009 67
20 M-ary 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 M-ary 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 M-ary 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 2-bits 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 8-ary PAM.
20.2.1 Bit Error Probabilities
How many bit errors are caused by a symbol error in M-ary 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 M-ary PAM - use (32). We will show that this
approximation is very good, in almost all cases. We will also show examples in multi-dimensional signalling
(e.g., QAM) when this is not a good approximation.
ECE 5520 Fall 2009 69
Lecture 13
Today: (1) Pulse Shaping / ISI, (2) N-Dim Detection Theory
21 Inter-symbol Interference
1. Consider the spectrum of the ideal 1-D PAM system with a square pulse.
2. Consider the time-domain 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 out-of-band
interference.
2. If we try (2) above and put symbols right next to each other in time, our own symbols experience
interference called inter-symbol 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 out-of-band interference requirements).
2. Channel: The wideband radio channel is a sum of time-delayed 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

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 line-of-sight 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: 2400-2480 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 frequency-shifted 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
-frequency-shifted 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: Do-it-yourself 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 Square-Root 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 zero-ISI 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 M-ary Detection Theory in N-dimensional 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 M-ary detection theory for 1-D signal vectors, and now
we will extend that to N-dimensional 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
X|H
i
(x[H
i
) (37)
Here,
f
X|H
i
(x[H
i
) =
1
(2πσ
2
N
)
N/2
exp
_

j
(x
j
−a
i,j
)
2

2
N
_
which can also be written as
f
X|H
i
(x[H
i
) =
1
(2πσ
2
N
)
N/2
exp
_

|x −a
i
|
2

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

2
N
_
ˆ
i = argmax
i

|x −a
i
|
2

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 3-D, or
subspace in N-D). 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 pair-wise 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 two-dimensional signalling method which uses the in-phase 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 non-zero within that window.
Previously, we’ve separated frequency up-conversion and down-conversion 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 unit-energy.
• 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)

0
¸
¸
¸
¸
T
2
T
1
= −c
2
cos(2ω
0
T
2
) −cos(2ω
0
T
1
)

0
= −c
2
cos(2ω
0
T
2
) −cos(2ω
0
T
1
)

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 ‘low-pass’
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 well-illustrated 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

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)

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, M-ary QAM is defined as an arbitrary signal set a
0
, . . . , a
M−1
, where each
signal space vector a
k
is two-dimensional:
a
k
= [a
k,0
, a
k,1
]
T
ECE 5520 Fall 2009 78
The signal corresponding to symbol k in (M-ary) 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 64-QAM.
• 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 M-QAM
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 Phase-Shift 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 phase-shift keying (BPSK) is the case of M = 2. Thus BPSK is the same as
bipolar (2-ary) 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 manufacturer-specific protocols. 6 GHz, and 11 GHz.
• Dial-up 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 (energy-consuming) 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 N-dimensional 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 N-D Gaussian
pdf across an area. We needed a Q function to get a tail probability for a 1-D 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
N-dimensional Gaussian pdf.
For example, consider M-ary PSK. Essentially, 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.
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

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: 4-QAM 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 ”2-amplitude 4-QAM” (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 4-QAM modulation,
c
av
= c
s
2(0.5) + 2(1.5)
4
= c
s
Thus there is no advantage to the two-amplitude QAM modulation in terms of average energy.
25.6.1 General Formula for Union Bound-based 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) M-ary PSK and QAM Analysis
26 QAM Probability of Error
26.1 Nearest-Neighbor 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 Q-function,
which in turn means a lower value of the Q-function. A little extra distance means a much lower value of the
Q-function. 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 step-by-step process for finding the probability of bit or symbol
error in arbitrary M-ary 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 pair-wise 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 nearest-neighbor 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 non-coherent. 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 phase-locked 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 equally-probable symbols). Also, having
M PLLs is a drawback.
27.5 Non-coherent 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 non-coherent 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 in-phase 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 non-coherent 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 non-coherent detector to find its analytical probability of error.
27.6 Receiver Block Diagrams
The block diagram of the the non-coherent 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) Phase-coherent demodulation of M-ary FSK signals.
ECE 5520 Fall 2009 92
Figure 45: (Proakis & Salehi) Demodulation of M-ary FSK signals for non-coherent detection.
Figure 46: (Proakis & Salehi) Demodulation and square-law 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 non-coherent 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 non-coherent 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
r|H
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
r|H
i
(r[H
i
) =
_

0
f
r,θm|H
i
(r, φ[H
i
)dφ
=
_

0
f
r|θm,H
i
(r[φ, H
i
)f
θm|H
i
(φ[H
i
)dφ
(49)
Note that f
r|θm,H
0
(r[φ, H
0
) is a 2-D 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
r|H
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 non-coherent
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 non-coherent) 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 M-ary Non-Coherent FSK
For M-ary non-coherent 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 non-coherent 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 Non-coherent M = 2 case
Use the above expressions to find the P [symbol error] and P [error] for binary non-coherent 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 non-coherent reception of M-ary 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 Shannon-Hartley
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 M-ary FSK, it tells us that the approximate
bandwidth is,
B
T
= (M −1)∆f + 2B
where B is the one-sided bandwidth of the digital baseband signal. For the null-to-null 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),
• Direct-sequence spread spectrum (DS-SS), or
• Frequency-hopping spread spectrum (FH-SS).
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
M-ary 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 K-ary 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
of4-aryPAM
Figure 49: Signal space diagram for K = 3 subchannel OFDM with 4-PAM 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 non-coherent 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
16-QAM 4.0 2.67 2.0
64-QAM 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 null-null 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 one-sided bandwidth of the digital baseband signal. For the null-to-null 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 non-coherent 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 8-PSK to achieve a probability of bit error of 10
−6
? What is the bandwidth
efficiency of 8-PSK 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. M-ary PAM, including Binary PAM or BPSK, OOK, DPSK.
2. M-ary PSK, including QPSK.
3. Square QAM
4. Non-square QAM constellations
5. FSK, M-ary 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 325-331.
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
M-PAM =
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
_
M-PSK ≤ 2Q
__
2(log
2
M) sin
2
(π/M)
E
b
N
0
_

1
log
2
M
P [symbol error]
Square M-QAM ≈
4
log
2
M
(

M−1)

M
Q
_
_
3 log
2
M
M−1
E
b
N
0
_
2-non-co-FSK =
1
2
exp
_

E
b
2N
0
_
same
M-non-co-FSK =

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]
2-co-FSK = Q
__
E
b
N
0
_
same
M-co-FSK ≤ (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 / Non-linear 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 band-pass filtered or tuned to the center frequency to force out spurious frequencies. Class C
non-linear amplifiers have power efficiencies around 90%. Can only amplify signals with a nearly constant
envelope.
In order to double the power efficiency, battery-powered 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
Q-channel
I-channel
OffsetQPSK:
Q-channel
I-channel
QPSKwithoutoffset:
(b) (a)
Figure 51: Constellation Diagram for (a) QPSK and (b) O-QPSK.
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
,

4
,

4
,

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 in-phase. 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 in-phase 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 (a-b) QPSK and (c-d) O-QPSK, 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 step-by-step version of what you might need do in this case:
Method A: Start with power-limited 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 bandwidth-limited 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 bandwidth-limited, 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 16-PSK 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 16-QAM 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 bandwidth-limited 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

−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

−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 Non-free-space 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 “point-to-point” 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 line-of-sight 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

−20 log
10
R −2dB −2dB −2dB −1dB
= 20dB + 20dB + 10dBW + 20 log
10
0.075m

−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) continuous-time and a (more) discrete-time
receiver. Figure 54 has a timing synchronization loop which controls the sampler (the ADC).
Figure 54: Block diagram for a continuous-time 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 discrete-time processing. This interpolation is the subject of this section.
ECE 5520 Fall 2009 114
cts-time
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 discrete-time 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 down-sampling (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 (2-4) 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

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

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).
Re-indexing 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 first-order 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 second-order 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 linear-phase 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 second-order polynomial, and get a linear-phase
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 re-write (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 half-way 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 discrete-time
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 discrete-time implementations:
1. Early-late timing error detector (ELTED)
ECE 5520 Fall 2009 120
2. Zero-crossing timing error detector (ZCTED)
We’ll use a continuous-time 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 eye-diagram of a RRC-shaped BPSK received signal (post matched filter), x(t). Here (a)
shows the signal x(t) and (b) shows its derivative ˙ x(t).
33.3 Early-late timing error detector (ELTED)
The early-late TED is a discrete-time implementation of the continuous-time “early-late 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 Zero-crossing timing error detector (ZCTED)
The zero-crossing 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 non-zero 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 re-written 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 zero-crossing 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 in-phase 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
modulo-1
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: Numerically-controlled 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 phase-locked 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 phase-locked 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 input-output 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 low-pass 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 Z-transform response as F(z).
33.6.3 VCO
A voltage-controlled 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 re-draw 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: Phase-equivalent and linear model for the continuous-time 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 phase-equivalent 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 proportional-plus-integrator 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 Discrete-Time 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 second-order filter, its implementation is quite efficient. See Figure C.2.1 (p 733, Rice book)
for diagrams of this discrete-time 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 Mon-Wed. The exam will be proctored. Please come to me with questions today or
tomorrow.
• Office hours: Today: 2-3:30. Friday 11-12, 3-4.
34 Exam 2 Topics
Where to study:
• Lectures covered: 8 (detection theory), 9, 11-19. Lectures 20,21 are not covered. No material from 1-7
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: 4-8. 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
• M-ary: PAM, QAM, PSK, FSK
4. Inter-symbol 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 non-standard modulation method, given signal space diagram.
• Exact expression, union bound, nearest-neighbor 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 1-2 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 2-3pm, Monday 4-5pm.
• Final homework: HW 10, which will be due April 28 at 5pm.
35 Source Coding
This topic is a semester-long 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.
379-423 and 623-656, July and October, 1948. http://www.cs.bell-labs.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 Lempel-Ziv 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 non-technical
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 log-base-2 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 base-2 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: Non-uniform 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 one-to-one 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 multi-variate 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 discrete-time 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 discrete-time 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 two-letter combination. This gives me the two-dimensional pmf shown in Figure
??(b). I calculate an entropy of 7.46, which is 2 3.73. For the three-letter combinations, the joint entropy was
10.04 = 3 3.35. For four-letter 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) two-letter 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 pulse-amplitude 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 n-dimensional 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 power-limited 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 two-sided PSD N
0
/2, by
P
N
= N
0
B
ECE 5520 Fall 2009 136
So finally we have the Shannon-Hartley 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 Shannon-Hartley 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 pulse-amplitude 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 n-dimensional signaling. So the received vector is y = [y
1
, . . . , y
n
], of length n. These might be
truly an n-dimensional 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 n-dimensional sphere (hypersphere) of radius

nE
N
with center x.
39.2.3 Introduction of Power Limitation
Shannon also formulated the problem as a energy-limited 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 Shannon-Hartley 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 log-y
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 Shannon-Hartley 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
c
y

(
b
p
s
/
H
z
)
4−FSK
16−FSK
64−FSK
256−FSK
1024−FSK
4−QAM
16−QAM
64−QAM
256−QAM
1024−QAM
8−PSK
16−PSK
32−PSK
64−PSK
Figure 71: From the Shannon-Hartley theorem bound with achieved bandwidth efficiencies of M-QAM, M-PSK,
and M-FSK.

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 Discrete-Time Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14 3.2 Decibel Notation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14 4 Time-Domain Concept Review 15 4.1 Periodicity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15 4.2 Impulse Functions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16 5 Bandwidth 5.1 Continuous-time 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

3 37 38 38 39 40 42 42 43 43 44 45 45

10 Multi-Variate 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 Matched-Filter 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 Log-Likelihood Ratio . . . . . . . . . . . . . . 14.5 Case of a0 = 0, a1 = 1 in Gaussian noise . . . 14.6 General Case for Arbitrary Signals . . . . . . 14.7 Equi-probable 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

. . . . . . . . . . . . .2 Transmitter and Receiver Filters . . . . . . . . . . . . . . . . . . . . 24. . . . . . . . . . . . . . . . . . . . .1 General Formula for Union Bound-based . . . 67 67 67 68 68 17 Additional Problems 17. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25. . . . . of Error . . . .4 Probability of Error in QPSK . . . 19 Detection with Multiple Symbols 20 M -ary PAM Probability of Error 20. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .6. .3 Binary PAM Error Probabilities . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69 21. . .1 Bit Error Probabilities . . PSD . . . . . . . . . . . . . 24. . . . . . . . . . . . . . . . . . . . . . . .6 Phase-Shift Keying . . . . . . . . . . . . . . . . . . . . . .2 BER Function of Distance. . . . . . . . . .1 Signal Distance . .5 Correlation / Matched Filter Receivers . . . . . .1. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24. . . . . . . . . . . . . . . . . . . . . .2 Sampling and Aliasing . . . . . . . . . . . . . . . . . . . . . . . 17. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3 Orthogonality and Signal Space . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 70 22 Nyquist Filtering 70 22. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .6 Application of Union Bound . . . . . . . . . . . . . . . . . . . . . . . . . . 72 23 M -ary Detection Theory in N -dimensional signal space 24 Quadrature Amplitude Modulation (QAM) 24. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17. . . . 18. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . Probability . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25.1 Symbol Error . . . 18 Probability of Error in Binary PAM 18. . . . . . . . .2. . . . . . . . . . . . .5 Average Energy in M-QAM . . . . 72 74 76 77 78 79 79 79 79 80 80 81 81 81 82 82 84 25 QAM Probability of Error 25. . . . . . . . . . . . . . . . . . . . . .4 Random Processes. . . . . . . . . . . . . . . . .1 Symbol Error Rate and Average Bit Energy 20. . . . . . . . . . . . . . . .1 Spectrum of Communication Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21 Inter-symbol Interference 69 21. . . . . . . . . . . . . . . . . . . .1 Showing Orthogonality . . . . . . . . . . . . . . . . . . . . 20. . . . . . . . . . . . . . 25. . . . . . . . . . . . . . . 24. . 20. . . . . . . . . . . . . . 18. . . . . . . . . . . . . . . . . . . . . . . . . . . . .3 Signal Constellations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4 Angle and Magnitude Representation . . . . . 72 22. . . . . . . . . . . . . . . . . 24. . . . . . . . . . . . . . . . . . . . . .3 Exact Error Analysis . . . . . . . . . . .1 Multipath Radio Channel . . . . . . . . . . . . . .7 Systems which use QAM . . . . . . . . . . . . . . . . . . . . . . . . . . .ECE 5520 Fall 2009 4 61 61 61 61 62 62 63 63 63 64 66 . . . . . .5 Union Bound . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1 Overview of Future Discussions on QAM . . . . 25. . . . . . . . . . . . . . 25. Noise PSD . . . . 25. . . . . . . . . . . . . . . . . . . . . . . . .2 Constellation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1 Raised Cosine Filtering . . . . . . . .2 Options for Probability of Error Expressions . . . . . . . . 24. . . . . .2 Bit Errors and Gray Encoding . . .2 Square-Root Raised Cosine Filtering . . . . . . . 17. . . 17. . . . . . . . . . . . . . . .

. . 29. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30 Link Budgets and System Design 30. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .8 Probability of Error for Noncoherent Binary FSK 27. 29. . . . . . . 86 27 Frequency Shift Keying 27. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1 M -ary Non-Coherent FSK . . . .2 Benefits of Frequency Multiplexing . . . . . . . . . . . . . .2 Summary . . . . . . . . . . . .2 Power and Energy Limited Channels 30. . . . . . . . . . . . . . .5 Fidelity (P [error]) vs. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5 Non-coherent Reception . . . . . . . . . . . . . . .1 PSK. . . . . . . .2 Points for Comparison . . .7 Offset QPSK . . 27. . . . . . . . . . . . . . . .1 Linear / Non-linear Amplifiers . . . . . . . . 27. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .6. . . . . . . . .1 Frequency Selective Fading . . . . . . . . . . . . . . . Nb . . Part 2 . . . . . . . . . . . . . . . . . . . . .1 Orthogonal Frequencies . . . .8 Receiver Complexity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2 Summary and Examples . . . . . . 29. . . . . .1. .ECE 5520 Fall 2009 5 26 QAM Probability of Error 85 26. . . . . . . . . . . . . . . . . . . . . . . . . . . . 27. . . . . .9. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .6 Transmitter Complexity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .6 Receiver Block Diagrams . . . . . . . . . . . . . . . . . . . . . . .1. . . . 95 28. . . . . . .1 Free Space . . . . . . . . . . .7 Probability of Error for Coherent Binary FSK . . . . . . . 85 26. . . . . . . . . . . . . . . . . . . . . .3 Wired Channels . . . . 0 29. . . . .2 Transmission of FSK . . . . . . . . . . . . . . . . . . . . . 29. . . . . . . . . . . . . . . . . . . . .1 Nearest-Neighbor Approximate Probability of Error . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27. . . . . . . . . . . . . . . . . . .3 Computing Received Power . . . . . . . . . . . . . . . . . . . . . . . . . . E 29. . . .3. . . . . . . . . . . . . . . . . . . . . . . . . . . 30. 30. . . . . . . . . . . . . . . . . . . . . . . . . . 0 E 29. . . . . . . . . . . 30. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2 Non-free-space Channels . . PAM and QAM . . . . . . . . . . . . . . . 97 29 Comparison of Modulation Methods 29. . . . . . . . . .3 Reception of FSK . . . . . . . . . . . . . . . . . .4 Bandwidth Efficiency vs. . . . . . . . . . .9 FSK Error Probabilities. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30. . . . 98 98 99 99 100 100 101 101 101 102 102 103 103 104 105 105 106 108 110 110 111 111 112 . . . . . . . . . . . . . . . . . .1 Differential Encoding for BPSK . . . . . . . . . . . . . .3. . . . . . . . . . . . . . . . 88 88 89 90 90 90 91 93 93 94 94 95 95 28 Frequency Multiplexing 95 28. . . . 27. . . . . . . . . . . . . . . . . 27. . . . . . . . . . . . . . . . . . . .3. . . . . . . .3. . . . . . . . . . . .1 DPSK Transmitter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29. . . .3 Probability of Bit Error for DPSK 29. . . 30. . . . . 27. . . . .1. . . . . . . . . . . . Nb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96 28. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29. . .10Bandwidth of FSK . . . . . . . . . . . . . . . . . . . . . . . 27. . . . . . . . . . . . . . . . .3 Bandwidth Efficiency . . . . . . . . . . . . 29. .4 Computing Noise Energy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1 Link Budgets Given C/N0 . .3. . . .3 OFDM as an extension of FSK . . .9. . . . . . . . . . . 27. . . . . . . . . . . . . .4 Coherent Reception . . . . . . . . . . . . . . . . . . . . . . . . 27. . . . . . .2 DPSK Receiver . . . . . . . . . . . . . . . . 29. . . . . . . .2 FSK . . . . . . . . . .

. .2 Loop Filter . . 132 133 133 . .4. . . . . . . . . . . . . . . . . .1 Higher order polynomial interpolation filters . . . . 33. . 35. . . . . . . . . . .1 QPSK Timing Error Detection . . .2. . . . .4 Analysis .2 Final Results . . . 33. . . . . 37. . . . . .1 Noisy Channel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 133 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2. . .6. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4 Zero-crossing timing error detector (ZCTED) 33. . . . . . . . . . . . . . . . . . . 33. . . . . . . . . . . . . . . . . . . . . . . . . . . . .1 Phase Detector . 33. .6. . . . .2 C. . . . . . . . . . . . . . . . . . . . . 35. . . . . 124 . . . .2 Introduction of Latency . . . 131 . . . . . 119 . . . . . . .6. 33. . . . . . . . 112 31 Timing Synchronization 32 Interpolation 32. . . . . . . . . . . 128 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 135 . . . . . . . . . . . . . . . . . 124 . . . . . . 122 . . . . . . . . . . . . . . . . . . . . . . . . . . . .2 Joint Entropy . . . . . . . . . . . . . . . 33. 135 . . . . . . . . 32. . . . . . . . . . . . . .1 R. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4. . . . 33 Final Project Overview 33. . . . . . . . . . . . . . . . . . . . . 32. . . . . . . . . . 115 . . . . . . . . . . . . . . . . . . . . 134 . . . . . . . . . . . . . . . . . . . . . .3 Conditional Entropy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5 Examples . . . . . . . . . . 124 . . . . . .3 Approximate Interpolation Filters . . . . . . . . . . . .1 Entropy . . . . . . . 116 . . . . . . . . . . . 113 114 . . . .5 Voltage Controlled Clock (VCC) . 134 . . . . . 117 118 . . . . . . . . . . . . . . . . 122 . 134 . . . . . . . . .1 Returning to Hartley . . . . . . . . .3. . . . . . . . . . . . . . . . . . . . . . 120 . . . . . . . . . . . . . . . . . . 35. . . .6. . . . . . . 37. . . . . . . . . . . . . . . . 35. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 130 .6. . . . . . . . . . . . . . . . . 33. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37.2 Seeing Interpolation as Filtering .3 Early-late timing error detector (ELTED) . . . . . Shannon . . . 33. . 38 Review . . . . . . . 136 138 . . . . V. . . . . .5 Source Coding Theorem 36 Review 37 Channel Coding 37. . . . .1 Review of Interpolation Filters . . . . . . . . . . . . . . .2. . . . . . . . . . . . . . . . . . 32.6 Phase Locked Loops . . . 125 126 127 . . . . . . . . . . . . . . . . .3 VCO . . . . . . . . . . . . . . . . . . . . . . 123 . . . . . . . . . . . . Hartley . . . . . . . . . . . . . . . . . . . . 34 Exam 2 Topics 35 Source Coding 35. . . . . . . . . . . . . . . . . 119 . . . . . .4 Efficiency Bound . . 135 . E. . . . . . . . . . . . . . . 37. . . . . . . . . . . . 37. . . . . . . . . . . . . . . . . . . . . . 117 . .4 Implementations . . . . . . .5 Discrete-Time Implementations . .2 Timing Error Detection . . . . . . . . 121 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3 Introduction of Power Limitation 37. . . . . . 135 . . . . . . . . . L. . . . . . . .ECE 5520 Fall 2009 6 30. . . . . . . . 123 . . . . . 116 . . . . . 37. . . . . . . . . . . .1 Sampling Time Notation . . . . 130 . . . . 32. . . . . . . . . . . . . . . . . .4 Entropy Rate . 37. . . . . . . . . . . . 33. . . . . .3 Combining Two Results . . . . . . . . . .

Shannon . . . . 39. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3 Combining Two Results . . . .2 Introduction of Latency . . . .3. .2 C.1 R. . . . . E. 39. . . . . . . . . . . . . . . . . .1 Noisy Channel . . . . . . . . . . . . . . . . . . . . L. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39. . . . . . . . . . . . . . . . . . .ECE 5520 Fall 2009 7 138 138 139 139 139 139 140 140 141 141 39 Channel Coding 39. . . . . . . . . . . . . . . . .2. . . . . . . . . . . . . . . . . . . . . .4 Efficiency Bound . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3 Introduction of Power Limitation 39. . . . . . . . . . . . . . . . . 39. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3. . . . . . . . V. 39. . . . . . . . . . . 39. . . Hartley . . . . . . . . . . . . 39. . . . . . . . . . .1 Returning to Hartley . .2 Final Results .2. . . .2. . . . . . . . . . .

Communication Systems Engineering. images) and even 1-D or 2-D. But graduates today will almost always encounter / be developing solely digital communication systems. video. the past text was the ‘standard’ text in the area for an undergraduate course. These waveforms are from a discrete set of possible waveforms. 2001. 2nd edition. these are optional. Digital information on an analog medium: We can send waveforms. Or. Lecture Notes I type my lecture notes. So half of most textbooks are useless. not just watching. Our object is to convey the signals or data to another place (or time) with as faithful representation as possible. If I didn’t. and/or after lecture online. Salehi.G. What set of waveforms should we use? Why? . Students didn’t like that I had so many supplemental readings. discrete-valued information across a physical channel. continuous-time functions. on the channel (medium).. text. Taking notes is important: I find most learning requires some writing on your part. real-valued.e. you must accept these conditions: 1. I have taught ECE 5520 previously and have my lecture notes from past years. what we won’t cover. Proakis & Salehi. Unless specified. They might be continuous-time (analog) signals (audio. so that decoding the data (i. However. and does it in depth. Please take your own notes. and more importantly. I find it to be very well-written. they may already be digital (discrete-time. And. I will provide additional readings solely to provide another presentation style or fit another learning style.1 ”Executive Summary” Here is the one sentence version: We will study how to efficiently encode digital data on a noisy. For example. discrete-valued). bandwidthlimited analog medium. 2. Information sources might include audio. and that the other half is sparse and needs supplemental material.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. These can be available to you at lecture. The keys points stuffed into that one sentence are: 1. i. In this section we talk about what we’ll cover in this class. efficient. Proakis and M. there are few options in this area. J. I’m constantly updating my notes. even up to the lecture time. Prentice Hall. you could just watch a recording! 2 Introduction A digital communication system conveys discrete-time. or data. This year’s text covers primarily digital communications. 2. and I try to tailor my approach during the lecture.e. Finally. and highfidelity.. My written notes do not and cannot reflect everything said during lecture: I answer questions and understand your perspective better after hearing your questions. reception) at a receiver is simple.

by Proakis & Salehi. You want. 2. iPhone. Bandwidth. and Fidelity: Fidelity is the correctness of the received data (i. 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. bandwidth.ECE 5520 Fall 2009 9 2. to have the source impedance match the destination impedance. To manage complexity. and we build a layered network to handle the application. is as follows: • Application • Presentation (*) • Session (*) • Transport • Network . and fidelity? We all want high fidelity. Bluetooth.2 Why not Analog? The previous text used for this course. development of new systems will almost certainly be of digital communication systems. the opposite of error rate). WiFi. How do you make a decision about which waveform was sent? 3. You can look at this like an impedance matching problem from circuits. we study how to take ones and zeros from a transmitter. 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. What is the tradeoff between energy. has an extensive analysis and study of analog communication systems. The 7-layer OSI stack. send them through a medium. however. We typically start with an application.. Decoding the data: When receiving a signal (a function) in noise. wireless car key).g. Analog systems still exist and will continue to exist. none of the original waveforms will match exactly.e. and then (hopefully) correctly identify the same ones and zeros at the receiver.3 Networking Stack In this course. In digital comm. wireless keyboard. Efficiency. That is.. we (engineers) don’t try to build a system to do everything all at once. In the recent past. for power efficiency. Why? • Fidelity • Energy: transmit power. this means that we want our waveform choices to match the channel and receiver to maximize the efficiency of the communication system. this course would study both analog and digital communication systems. we study digital communications from bits to bits. which you would study in a CS computer networking class. such as radio and television broadcasting (Chapter 3). and low energy consumption and bandwidth usage (the costs of our communication system). There’s a lot more than this to the digital communication systems which you use on a daily basis (e.

the physical layer.ECE 5520 Fall 2009 • Link Layer • Physical (PHY) Layer 10 (Note that there is also a 5-layer model in which * layers are considered as part of the application layer.) ECE 5520 is part of the bottom layer.. including (top) transmitter. • Disk (data storage applications) 2. Notes: • Information source comes from higher networking layers. It may be continuous or packetized. Includes sampling of continuous-time signals. • Source Encoding: Finding a compact digital representation for the data source. light • Acoustic: ultrasound • Transmission lines. (middle) channel. 2. 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. Also includes compression of those sources (lossy. In fact. or lossless). optical fiber. mm-wave bands. coaxial cable.4 Channels and Media We can chose from a few media. waveguides. but we largely can’t change the properties of the medium (although there are exceptions).. . 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. and quantization of continuous-valued signals. and (bottom) receiver. the physical layer has much more detail. . Microwave.5 Encoding / Decoding Block Diagram Source Encoder Channel Encoder Modulator Up-conversion 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 single-user digital communication system. Here are some media: • EM Spectra: (anything above 0 Hz) Radio.

Wideband wireless channels display multipath. for mobile radio. we will focus primarily on the AWGN channel. people. We will not study channel encoding. Noise Transmitted Signal LTI Filter h(t) Received Signal Figure 2: Linear filter and additive noise channel model. 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 source-channel coding separation theorem. However. The noise comes from a variety of sources. He showed that (given enough time) we can consider them separately without additional loss. 2. And separation. The linear filtering of the channel result from the physics and EM of the medium. the TX and RX do not move or change. The filter may be constant. in real wireless channels. 2. due to multiple time-delayed reflections. thus it is referred to as additive white Gaussian noise (AWGN). It is analogous to impedance matching . or non-white noise spectral density. Even for stationary TX and RX. but we will mention what variations exist for particular channels. the channel may change very quickly over time. and scattering of the signal off of the objects in the environment. and white. or time-invariant.ECE 5520 Fall 2009 11 • Channel encoding refers to redundancy added to the signal such that any bit errors can be corrected. Thermal background noise: Due to the physics of living above 0 Kelvin. Typical models are additive noise. Modulation and demodulation will be the main focus of this course. because of the redundancy. reduces complexity to the designer.proper matching of a modulation to a channel allows optimal information transfer. • Modulation refers to the digital-to-analog conversion which produces a continuous-time signal that can be sent on the physical channel. .6 Channels A channel can typically be modeled as a linear filter with the addition of noise. Narrowband wireless channels experience fading that varies quickly as a function of frequency. Interference from other transmitted signals. etc. Well modeled as Gaussian. can correct some bit errors. like impedance matching ensured optimal power transfer. trees. like layering. if the medium. In this course. or linear filtering channel. in the environment may change the channel slowly over time. movement of cars. These other transmitters whose signals we cannot completely cancel. but predominantly: 1. we lump into the ‘interference’ category. but it is a topic in the (ECE 6520) Coding Theory. All of these can be modeled as linear filters. A channel decoder. attenuation in telephone wires varies by frequency. These may result in non-Gaussian noise distribution. For example. diffractions. and how they are addressed. • Channels: See above for examples.

. There is a tradeoff between frequency requirements and time requirements. and are sampled.ECE 5520 Fall 2009 12 2. We have to tolerate errors. in short. We can also employ multiple orthogonal signals in a single transmitter and receiver. We’ll use signal spaces to show graphically the results. This means. we need to show that they are orthogonal. The Fourier transform of our modulated.. phase.9 Topic: Orthogonality and Signal spaces To show that signals sharing the same channel don’t interfere with each other. for (a) M = 8 and (b) M = 16. attenuation. This may be tolerated. which will be a major part of this course. we often split the signals by frequency. transmitted signal is used to show that it meets the spectrum allocation limits of the FCC. We will study orthogonal signals. The concept of sharing a channel is called multiple access (MA). 2. and fading • Device frequency. Signals in different frequency bands are orthogonal. this is controlled by the FCC (in the US) and called spectrum allocation. and learn an algorithm to take an arbitrary set of signals and output a set of orthogonal signals with which to represent them. Optimal receiver design is something that we study using probability theory. M=8 M=16 Figure 3: Example signal space diagram for M -ary Phase Shift Keying.8 Topic: Frequency Domain Representations To fit as many signals as possible onto a channel. as the example in Figure 36. 2. in order to provide multiple independent means (dimensions) on which to modulate information. For the wireless channel. Each point is a vector which can be used to send a 3 or 4 bit sequence. or a higher layer networking protocol (eg. TCP-IP) can determine that an error occurred and then re-request the data.7 Topic: Random Processes Random things in a communication system: • Noise in the channel • Signal (bits) • Channel filtering. Separating signals by frequency band is called frequencydivision multiple access (FDMA). We want to build the best receiver possible despite the impediments. that a receiver can uniquely separate them. and timing offsets These random signals often pass through LTI filters. Noise and attenuation of the channel will cause bit errors to be made by the demodulator and even the channel decoder.

Use the units: energy is measured in Joules (J). dB (2) Time-domain concepts (3) Bandwidth. (ECE 5411) Fiberoptic Systems 4.10 Related classes 1. A signal x(t) has energy defined as E= ∞ −∞ |x(t)|2 dt For some signals.ECE 5520 Fall 2009 13 2. such as x(t). E will be infinite because the signal is non-zero for an infinite duration of time (it is always on). and starts the review of tools to analyze frequency content and bandwidth. . 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. (ECE 5720) Analog IC Design 6. Electromagnetics: EM Waves. so |x(t)|2 is the power dissipated at time t (since power is equal to the voltage squared divided by the resistance). Pre-requisites: (ECE 5510) Random Processes. (CS 5480) Computer Networks 7. (ECE 6540): Estimation Theory Lecture 2 Today: (1) Power. Fourier Transform Two of the biggest limitations in communications systems are (1) energy / power . (ECE 3500) Signals and Systems. Networking: (ECE 5780) Embedded System Design. power is measured in Watts (W) which is the same as Joules/second (J/sec). (ECE 5320-5321) Microwave Engineering. Energy. as voltage signals (V). 3 Power and Energy Recall that energy is power times time. Today’s lecture provides some tools to deal with power and energy. Signal Processing: (ECE 5530): Digital Signal Processing 3. Advanced Classes: (ECE 6590) Software Radio. Breadth: (ECE 5325) Wireless Communications 5. and (2) bandwidth. Also. recall that our standard in signals and systems is define our signals. Devices and Circuits: (ECE 3700) Fundamentals of Digital System Design. 2. When we want to know the power of a signal we assume it is being dissipated in a 1 Ohm resistor. (ECE 6520) Information Theory and Coding. (ECE 5324) Antenna Theory.

or it referred to a name ‘Bell’ so it was capitalized. the output of the channel has 100 times less power than the input to the channel. For discrete-time signals.they can apply to other unitless quantities as well. so we’ll follow the Rice text notation. 3.2 Decibel Notation We often use a decibel (dB) scale for power. And maybe this one: • 1 dB: This means the number is a little over 25% more (multiply by 5/4) in linear terms. the channel has a loss of 20 dB. then [P ]dBW = 10 log 10 Plin Decibels are more general . we refer to discrete samples of the sampled signal x as x(n). [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. • 10 dB: This means the number is ten times in linear terms.1 Discrete-Time Signals In this book. 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. • Only use 20 as the multiplier if you are converting from voltage to power. 20 in the dB formula. Essentially. whenever you see a function of n (or k. it was capitalized. But. Remember these two dB numbers: • 3 dB: This means the number is double in linear terms. and 1.ECE 5520 Fall 2009 14 3. so when the ‘bel’ was first proposed as log10 (·). Example: Convert dB to linear values: (4) .e. In particular. l. you can quickly convert losses or gains between linear and dB units without a calculator. You may be more familiar with the x[n] notation. taking the log10 of a voltage and expecting the result to be a dB power value. I’m sorry this is not more obvious in the notation. So if we say. such as a gain (loss) L(f ) through a filter H(f ). Note that (3) could also be written as: [L(f )]dB = 20 log10 |H(f )| Be careful with your use of 10 vs.. whenever you see a function of t (or perhaps τ ) it is a continuous-time function. m). i. Matlab uses parentheses also. Our standard is to consider power gains and losses. we use the unit decibel 10 log 10 (·) which is then abbreviated as dB in the SI system. for example. If Plin is the power in Watts. With these three numbers. not voltage gains and losses. Just convert any dB number into a sum of multiples of 10. In either case. this refers to a loss in power. it is a discrete-time function.

lin t.lin L−d cable. Pr. 30 dBW 2. Periodic signals have Fourier series representations. The smallest positive integer N0 is the period. The smallest such constant T0 > 0 is the period. and Lcable. as defined in Rice Ch.lin t. 33 dBm 3. These last two are what we will need in Section 6. where d is the cable length (typically in units of km).lin is a loss in a 1 km cable. Def ’n: Periodic (discrete-time) A DT signal x(n) is periodic if x(n) = x(n + N0 ) for some integer N0 = 0. for all integers n. If a signal is not periodic it is aperiodic. and Gt. 0. 4 dB Example: Convert linear values to dB: 1. d is the path length (m).lin (4πd)2 are the linear gains in the antennas.lin and Gt.lin . when we discuss link budgets.lin is the received power in a fiber-optic link.lin 2.lin = 100Px. .4. 2. Po.ECE 5520 Fall 2009 15 1.1 Time-Domain Concept Review Periodicity Def ’n: Periodic (continuous-time) A signal x(t) is periodic if x(t) = x(t + T0 ) for some constant T0 = 0 for all t ∈ R. 1. This is the Friis free space path loss formula.lin is the gain in any connectors.lin . The main idea is that we have a limited amount of power which will be available at the receiver. Py. 3.2 W 2. Pt. where λ is the wavelength (m). G G λ2 4 4. Gconnector.lin is the transmit power (W) and Pr.lin = Gconnector. -20 dB 4. 40 mW Example: Convert power relationships to dB: Convert the expression to one which involves only dB terms.lin is the received power (W).lin = Pt. where Po.

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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 continuous-time 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 discrete-time 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 Continuous-Time 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 z-Transform 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 Discrete-Time 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 • 3-dB 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 square-root 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

Continuous-time Frequency Transforms

Notes about continuous-time 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

ECE 5520 Fall 2009 19 See Figure 4(a).) 2. Duality property: x(jω) = F {X(−t)} x(−jω) = F {X(t)} (Confusing. Question: What if Y (jω) was a rect function? What would the inverse Fourier transform y(t) be? 5. It says is that you can go backwards on the Fourier transform.3 in the Rice book. and (b) Power vs. 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 . 20 log10 (X(j2πf )/Ts ).4. Important properties of the Fourier transform: 1. Time shift property: F {x(t − t0 )} = e−jωt0 X(jω) . just remember to flip the result around the origin.1 Fourier Transform Properties frequency See Table 2.1. Assume that F {x(t)} = X(jω).

check your answer by doing both. Parceval’s theorem: The energy calculated in the frequency domain is equal to the energy calculated in the time domain. ∞ ∞ ∞ 1 |X(jω)|2 dω |X(f )|2 df = |x(t)|2 dt = 2π ω=−∞ f =−∞ t=−∞ So do whichever one is easiest! Or. F {x(at)} = ω 1 X j |a| a 4. 5. Y (jω) = X(jω) · H(jω) 5. the output signal is y(t) = h(t) ⋆ x(t) Using the above convolution property.ECE 5520 Fall 2009 20 3. additionally y(t) has Fourier transform X(jω). F {x(t) ⋆ y(t)} = X(jω) · Y (jω) 5.2 Linear Time Invariant (LTI) Filters If a (deterministic) signal x(t) is input to a LTI filter with impulse response h(t). Convolution property: if. Scaling property: for any real a = 0. x(t) = 2 ∂w(t) ∂t 4.3 Examples Example: Applying FT Properties If w(t) has the Fourier transform W (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) = w(1 − t) Solution: To be worked out in class. jω 1 + jω Lecture 3 Today: (1) Bandpass Signals (2) Sampling Solution: From the Examples at the end of Lecture 2: . Modulation property: 1 1 F {x(t)cos(ω0 t)} = X(ω − ω0 ) + X(ω + ω0 ) 2 2 6.

Equivalently. 6. Alternatively. where B/2 < fc . . Then X(jω) = Y (−jω). Then Z(ω) = e−jω W (jω). there will be some ‘sidelobes’ out of the main band. Let x(t) = y(−t). let x(t) = r(t + 1) where r(t) = w(2t). 2 2. where y(t) = w(1 + t). B X(f) B -fc fc Figure 5: Bandpass signal with center frequency fc . X(jω) = 1 ejω 1+jω/2 . X(jω) will not be exactly zero outside of the bandwidth. Let x(t) = z(2t) where z(t) = w(t + 2). 1 + jω 2 jω/2 So X(jω) = 1 ejω 1+jω/2 . a bandpass signal x(t) has Fourier transform X(f ) = 0 for all f such that |f ± fc | > B/2. where W/2 < ωc . Then 2 1 R(jω) = W (jω/2). and X(jω) = ejω R(jω). X(jω) = 2jω 1+jω = − 1+jω . Realistically. where Y (jω) = ejω W (jω) = ejω −jω So X(jω) = e−jω 1−jω . as shown in Fig. 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. Then Z(jω) = ej2ω 1 jω .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. 2 4. jω 2ω 3. Let x(t) = e−jt z(t) where z(t) = w(t − 1). So j(ω+1) X(jω) = e−j(ω−1) W (j(ω − 1)) = e−j(ω+1) 1+j(ω+1) . 6. and X(jω) = Z(jω/2). 2 jω/2 Again. and X(jω) = Z(j(ω − 1)).ECE 5520 Fall 2009 21 1.

7. 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. Example: Modulated Square Wave We dealt last time with the square wave x(t) = rect(t/Ts ). Note that I use “rect” as follows: 1.ECE 5520 Fall 2009 22 xC(t) cos(2pfct) Figure 6: Modulator block diagram. Multiply (again) by the carrier 2 cos(ωc t). . o.w. 6. −1/2 < t ≤ 1/2 rect(t) = 0. 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). Input to a low-pass filter (LPF) with cutoff frequency ωc . This time.2 Downconversion of Bandpass Signals How will we get back the desired baseband signal xC (t) at the receiver? 1. 2. X(f) -fc fc Figure 7: Plot of X(j2πf ) for modulated square wave example. This is shown in Fig. 8.

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. how would you find the maximum of x(t)? • This is only precise when X(jω) = 0 for all |ω| ≥ 2πB. . (Does the LPF need to be ideal?) X2 (jω) = XC (jω) So x2 (t) = xC (t). • Given {xn }∞ n=−∞ . But the theorem really has to do with signal reconstruction from a sampled signal. where T ≥ 2B to yield the ∞ sequence {xc (nT )}n=−∞ .e. Notes: • This is an interpolation procedure. Assume the general modulated signal was. 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.. Then xc (t) = 2BT ∞ n=−∞ xc (nT ) sin(2πB(t − nT )) . Xc (jω) = 0 for all |ω| ≥ 2πB. Theorem: (Nyquist Sampling Theorem. Let xc (t) be sampled at multiples of T . continuous signal with bandwidth B (in 1 Hz).) Let xc (t) be a baseband.ECE 5520 Fall 2009 23 x(t) 2cos(2pfct) LPF Figure 8: Demodulator block diagram. i.

What observations would you make about Figure 10(b). 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. 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 ). 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.1 Aliasing Due To Sampling ∞ n=−∞ Essentially. and indicate what the spectrum is of the sampled signals. Also consider a parabola pulse (this doesn’t . 9. round pulse shape Consider the square pulse considered before. compared to Figure 10(a)? Example: Square vs. as shown in Fig. The Fourier transform of the sampled signal is many copies of X(jω) strung at integer multiples of 2π/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. Figure 10 shows what happens when the Nyquist theorem is applied to the each signal (whether or not it is valid).ECE 5520 Fall 2009 24 7.12.

ECE 5520 Fall 2009

25
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 discrete-time 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 continuous-time 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, continuous-time 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 down-conversion 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 one-half the sampling rate). Example: Bandpass Sampling This is Example 2.6.2 (pp. 60-65) 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).

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 bandwidth is 2π 140 = π/7 radians/sample of sampled spectrum. The discrete-time frequency is modulo 2π. 8 Orthogonality From the first lecture. x = x.  . xn yn xT y = i xi y i Note the T transpose ‘in-between’ an inner product. we’ll start with something you’re probably more familiar with: vector multiplication. It is also determined by orthogonality. which you’re familiar with from vectors (as the dot product). y . Sample at 1000 samples/sec: Ωc = 2π × 10 Hz 2π = π/50 radians/sample. If we have two vectors x and y. 1000 samples/sec? 2. 8. x .it occupies only 2. Finally.  . This is the critical concept needed to understand digital receivers. the idea is to determine which waveforms were sent and in what quantities. The inner product is also denoted x. the norm of a vector is the square root of the inner product of a vector with itself. 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 (continuous-time.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 455 500 = 455 500 π. Loosely. This choice of waveforms is partially determined by bandwidth. which we have covered.1 Inner Product of Vectors Then we can take their inner product as: We often use an idea called inner product. The signal is very sparse . To build a better framework for it.   . real-valued) channel. orthogonality of waveforms means that some combination of the waveforms can be uniquely separated at the receiver. At the receiver.  y= .  . 140 samples/sec? Solution: 1.     x1 y1  x2   y2      x= . 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.

e. 1. 1]T . 0 < t ≤ 1 0. y ? • z. they’re at cross-directions. 0. the two vectors are pointing in a similar direction.w. o. when negative. What are: • x. In words. 0]T .. when zero. y(t) x(t). 8. 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). A norm is always indicated with the Example: Example Inner Products Let x = [1.e. i. z = [1. 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.ECE 5520 Fall 2009 · 29 symbols outside of the vector. 1. y = [0.2 Inner Product of Functions The inner product is not limited to vectors. 1]T . The ‘space’ of square-integrable complex functions. i. y(t) As a result. 0 < t ≤ 1 0. sin(2πt).. they are pointing in somewhat opposite directions. those functions x(t) for which ∞ −∞ ∞ −∞ x2 (t)dt < ∞. those functions x(t) for which |x(t)|2 dt < ∞ also have an inner product. the inner product is a measure of ‘same-direction-ness’: when positive. .w. These norms are: x(t). It also is applied to the ‘space’ of square-integrable real functions. o.

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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 Walsh-Hadamard functions. Example: Walsh-Hadamard 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: Walsh-Hadamard 2-length and 4-length functions in image form. Each function is a row of the image, where crimson red is 1 and black is -1.

Figure 14: Walsh-Hadamard 64-length 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 half-time 1 and half-time -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 Walsh-Hadamard This example is just for your information (not for any test). The set of 64-length Walsh-Hadamard sequences

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is used in CDMA (IS-95) in two ways: 1. Reverse Link: Modulation: For each group of 6 bits to be send, the modulator chooses one of the 64 possible 64-length 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).

. for which si (t) ∈ Span{B} for every i = 0. If you can find one basis. Def ’n: Orthonormal Basis The orthonormal basis is an orthogonal basis in which each basis function has norm 1.2 Synthesis Consider one of the signals in our signal set. {−φk (t)}k=0 as another orthonormal basis. the orthogonal basis is an orthogonal set of the smallest size N . . Notes: N −1 • The orthonormal basis is not unique.k φk (t) The particular constants ai.. .. 9. Then the signal is equal to the sum of its projections onto the basis functions.aN−1 ∈R Def ’n: Orthogonal Basis For any arbitrary set of signals S = {si (t)}M −1 . Why is this? . • All orthogonal bases will have size N : no smaller basis can include all signals si (t) in its span. φk (t) φk (t). • how to represent the signals in ‘signal-space’. M − 1. φk (t) The projection of one signal i onto basis function k is defined as ai. (As the Walsh-Hadamard functions are used in IS-95 reverse link). it can be represented as a linear combination of the basis functions.ECE 5520 Fall 2009 33 9.k are calculated using the inner product: ai. .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. and • how to quickly analyze the error rate of the scheme. for example. si (t) = N −1 k=0 ai. An orthonormal basis for an arbitrary signal set tells us: • how to build the receiver. si (t). i=0 N −1 B = {φi (t)}i=0 .k = si (t).. and a larger set would not be a basis by definition. Given that it is in the span of our basis B. you can use. The value of N is called the dimension of the signal set. The signal set might be a set of signals we can use to transmit particular bits (or sets of bits).k φk (t) = si (t). The span is referred to as Span{B} and is Span{B} = N −1 k=0 ak φk (t) a0 . .

s1 . Generally.5 in Rice (page 254). and s2 ? Solution: They are s0 = [1. s0 . log2 M bits of information. . ai. φj (t) = N −1 k=0 ai. which shows a block diagram of how a transmitter would synthesize one of the M signals to send. now we can now represent a signal by a vector. By choosing one of the M signals. We can also synthesize any of our M signals in the signal set by adding the proper linear combination of the N bases. we convey information.k φk (t) N −1 k=0 N −1 k=0 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. ai. φj (t) = ai. specifically. Example: Position-shifted pulses Plot the signal space diagram for the signals. φj (t) si (t). .j So. See Figure 5. and s2 = [1. .N −1 ]T This and the basis functions completely represent each signal. 0]T . this space that we’re plotting in is called signal space. φj (t) si (t).k φk (t). we choose M to be a power of 2 for simplicity). (Generally. 1]T . 1]T . 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. Plotting {si }i in an N dimensional grid is termed a constellation diagram.ECE 5520 Fall 2009 N Solution: Since si (t) ∈ Span{B}.1 .k φk (t). φj (t) = ai. s1 = [0.0 .k }k −1 such that 34 si (t) = Taking the inner product of both sides with φj (t). we know there are some constants {ai. φ0 (t) = u(t) − u(t − 1) What are the signal space vectors. si (t).k Proof? . . They are plotted in the signal space diagram in Figure 15. si = [ai. based on an input bitstream.

there will be additional functions added to the signal function we sent. s3 = [−1.5]. • Although different ON bases can be used. 0. M − 1}. di.5]. 9. This is the task of analysis. Energies are just the squared magnitude of the vector: 2.5]. 0. . .5[u(t) − u(t − 1)] s3 (t) = −1.25. j in {0. See Figure 16. .5[u(t) − u(t − 1)] s2 (t) = −0.3 Analysis At a receiver. s1 = [0. Example: Amplitude-shifted signals Now consider s0 (t) = 1.25.5[u(t) − u(t − 1)] N −1 k=0 (ai. -1 0 1 f1 Figure 16: Signal space diagram for amplitude-shifted signals example. and 2. .25. .5[u(t) − u(t − 1)] φ1 (t) = u(t) − u(t − 1) and the orthonormal basis.ECE 5520 Fall 2009 35 f2 1 f1 1 Figure 15: Signal space diagram for position-shifted signals example. It turns out that an orthonormal bases makes our analysis very straightforward and elegant.5].k − aj. • 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. What are the signal space vectors for the signals {si (t)}? What are their energies? Solution: s0 = [1.j = for i. our job will be to analyze the received signal (a function) and to decide which of the M possible signals was sent.25. respectively. s2 = [−0. We won’t receive exactly what we sent . the energy and distance between points will not change.k )2 s1 (t) = 0.

. . ˆ x = [x0 . ˆ argmin r (t)∈Span{B} ˆ ∞ −∞ r |ˆ(t) − r(t)|2 dt (13) Solution: Since r(t) ∈ Span{B}. r (t) = ˆ −1/2. N ..e. i. . xk = r(t). . x1 .w. it can be represented as a vector in signal space. what is r (t) ∈ Span{B} such that the energy of the ˆ difference between r(t) and r(t) is minimized. What is the best approximation to r(t) in the signal space? Specifically. Let s0 (t) = φ0 (t) and s1 (t) = φ1 (t). 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. you’d ˆ see that the minimum error is at ∞ r(t)φk (t)dt xk = −∞ that is.ECE 5520 Fall 2009 • Thermal noise • Interference from other users • Self-interference 36 We won’t get into these problems today.e. i. But. Lecture 6 Today: (1) Multivariate Distributions . 1 ≤ t < 2  o. Example: Analysis using a Walsh-Hadamard 2 Basis See Figure 17. What is r(t)? ˆ Solution:  0≤t<1  1. . 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). . so r(t) would not be in Span{B} either. 0. . for k = 0. But w(t) might not (probably not) be in the span of our basis B. φk (t) . . . sm (t) from our signal set. xN −1 ]T . we might say that if we send signal m. and then find the minimum with respect to each xk .

15 should be: fX (t)dt −∞ FX (x) = P [X ≤ x] To find the probability of an event. • Joint pdf: fX1 . Note: Two errors in Rice book dealing with the definition of the CDF. x2 ) = P [{X1 ≤ x1 } ∩ {X2 ≤ x2 }] It is the probability that both events happen simultaneously.−∞ FX1 .X2 (x1 . Eqn 4. x2 ) = ∂2 ∂x1 ∂x2 FX1 . The CDF is non-negative and non-decreasing. x2 )dx1 dx2 (x1 . For example.1/2 r(t) 2 1 -2 2 2 -1 -2 Figure 17: Signal and basis functions for Analysis example. you integrate.X2 (x1 . x2 ) fX1 .X2 (x1 . • Joint CDF: FX1 . for event B ∈ S.X2 (x1 . x2 ) = 0 and limx1 . x2 ) The pdf and pmf integrate / sum to one.ECE 5520 Fall 2009 37 f1(t) 1/2 1 . with limxi →.X2 )∈B PX1 .x2 →. • Joint pmf: PX1 .+∞ FX1 .X2 (x1 .X2 (x1 . • Discrete case: P [B] = • Continuous Case: P [B] = (X1 . x2 ) = 1.x2 )∈B .9 should be: x FX (x) = P [X ≤ x] = and Eqn 4. and are non-negative.1/2 r(t) 2 1 2 2 f2(t) 1/2 1 . 10 Multi-Variate Distributions For two random variables X1 and X2 .X2 (x1 .X2 (x1 . x2 ) = P [{X1 = x1 } ∩ {X2 = x2 }] It is the probability that both events happen simultaneously.

. Xn . The conditional pmf of X1 given X2 = x2 .. where PX2 (x2 ) > 0.1 Random Vectors Def ’n: Random Vector A random vector (R. .Xn (x1 .X2 (x1 .w.. 10. X is PX (x) = PX1 . X = [X1 . The pmf of a discrete R. .. (X1 . The CDF of R.X2 (x1 . where fX2 (x2 ) > 0. P [B] 0. x2 ) = PX1 (x1 )PX2 (x2 ) • fX1 . . xn ) = ∂n ∂x1 ···∂xn FX (x). X is FX (x) = FX1 . x2 ) = fX2 (x2 )fX1 (x1 ) 10.v. . X2 ) ∈ B o. .. .. fX1 . . X2 .x2 ) . the joint probability conditioned on event B is • Discrete case: PX1 .s: 1.V. 0.V.. .X2 (x1 . X2 ) ∈ B o. .X2 |B (x1 . . .) is a list of multiple random variables X1 . . P [B] (X1 .X2 (x1 . . The pdf of a continuous R. is fX1 |X2 (x1 |x2 ) = fX1 . • PX1 . 3.X2 (x1 . .Xn (x1 .X2 (x1 .V. .ECE 5520 Fall 2009 38 The marginal distributions are: • Marginal pmf: PX2 (x2 ) = • Marginal pdf: fX2 (x2 ) = x1 ∈SX1 x1 ∈SX1 PX1 .. . .. . xn ) = P [X1 ≤ x1 . x2 ) = Given r.V. . Xn = xn ].. .s X1 and X2 . . x2 ) fX1 . is PX1 |X2 (x1 |x2 ) = PX1 ..x2 ) . X is fX (x) = fX1 .w. . .X2 (x1 .X2 (x1 . x2 ) = • Continuous Case: fX1 . . xn ) = P [X1 = x1 . x2 )/fX2 (x2 ) PX1 .Xn (x1 . ...X2 |B (x1 . Xn ]T Here are the Models of R. . . x2 )/PX2 (x2 ) • Continuous Case: The conditional pdf of X1 given X2 = x2 . 2. x2 )dx1 Two random variables X1 and X2 are independent iff for all x1 and x2 .V. . Xn ≤ xn ]. • Discrete case.2 Conditional Distributions Given event B ∈ S which has P [B] > 0. X2 .

v. x2 ) = fX2 |X1 (x2 |x1 )fX1 (x1 ) or fX1 |X2 (x1 |x2 ) = fX2 |X1 (x2 |x1 )fX1 (x1 ) fX2 (x2 ) 39 10. 1.d. say N bits through a model of the communication system.3 Simulation of Digital Communication Systems A simulation of a digital communication system is often used to estimate a bit error rate. Let S = N Ei . What is the pmf of S? 3. 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 assume that {Ei } are independent and identically distributed i=1 (i.X2 (x1 .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 S? 2.. This trial Ei is in error (E1 = 1) with probability pe (the true bit error rate) and correct (Ei = 0) with probability 1 − pe . and count the number of bits that are in error. and S is called a Binomial r. 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) .i. It is written either as: fX1 .v. Each bit can either be demodulated without error. What is the mean and variance of S? Solution: Ei is called a Bernoulli r. What type of random variable is Ei ? Simulations run many bits. Thus the simulation of one bit is a Bernoulli trial. or with error.

Any probability integral will return the proper value. and X2 is the received signal voltage. then we would write a pdf with the probabilities as amplitudes of a dirac delta function centered at the value that has that probability. we won’t have a very good idea of the bit error rate.. What type of random variable is T1 ? 2. So. 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 . our estimate of pe will have relatively high variance. In a digital system. What is the mean of T1 ? Solution: T1 is a Geometric r. 10. then X2 will be continuous-valued. For example. although it was in the Yates & Goodman textbook. if we run a simulation and get zero bit errors. so the standard deviation for very low pe is almost the same e as the expected value.5 < x1 < 0. Work-around: We will sometimes use a pdf for a discrete   0. PX1 (x1 ) = 0.5.4. random variable. and non-negative values for all x1 . 1. {1.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. which is modeled as X2 = aX1 + N where N is a continuous-valued random variable representing the additive noise of the channel. For instance. e. X1 may take a discrete set of values. We’ll often have X1 discrete and X2 continuous. Example: Digital communication system in noise Let X1 is the transmitted signal voltage. Gaussian with mean µ and variance σ 2 . For example.4δ(x1 − 1) This pdf has integral 1.v. As long as σ 2 > 0. But the noise N may be continuous. e.g. What is the pmf of T1 ? 3. the probability that −0.6δ(x1 ) + 0.g.4 Mixed Discrete and Continuous Joint Variables This was not covered in ECE 5510.5.5.6. even if we run an experiment until the first bit error. 0. if x1 = 0 x1 = 1 o. fX1 (x1 ) = 0.6.5}. and a is a constant which represents the attenuation of the channel.5 would be an integral that would return 0.  0. Let T1 be the time (number of bits) up to and including the first error. .w.. −0. −1.

X2 ) ? Since X1 and X2 are NOT independent. What is the pdf of X2 ? 2. X2 ) ? Solution: 1. X2 Consider the channel model X2 = X1 + N . and the r.X2 (x1 . When two independent r. to get fX2 (x2 ) = 0. Use whichever seems more convenient for you. fX1 . x2 ) =    fX1 . x2 ) =  down the pdf of X2 . of (X1 . we cannot simply multiply the marginal pdfs together. .X2 (x1 .v. X1 is independent of N with   0. What is the joint p. x2 ) = 0.f.5fN (x2 − 1).X2 (x1 . See Figure 18.5fN (x2 − 1) 1 2 2 2 2 √ e−x2 /(2σ ) + e−(x2 −1) /(2σ ) fX2 (x2 ) = 2 2 2πσ 2.w.X2 (x1 .5.d. x2 ) = fX2 |X1 (x2 |x1 )fX1 (x1 ) Given a value of X1 (either 0 or 1) we can write   fX1 . of (X1 .5δ(x1 ) + 0.5δ(x1 − 1) we convolve this with fN (n) above. Writing fX1 (x1 ) = 0. x1 = 0 PX1 (x1 ) = 0.5.ECE 5520 Fall 2009 41 Example: Joint distribution of X1 . and fN (n) = √ 1 2πσ 2 e−n 2 /(2σ 2 ) where σ 2 is some known variance. o. o.d. 0. fX1 . 1.w. o. x1 = 0 fX2 |X1 (x2 |1)fX1 (1).s are added.5fN (x2 ).5fN (x2 )δ(x1 ) + 0. x1 = 0 0.  0. x1 = 1 0. x1 = 1 . What is the joint p.w.5fN (x2 ) + 0.v. x1 = 1 0.f. It is necessary to use Bayes’ Law. So break this into two cases: fX2 |X1 (x2 |0)fX1 (0).5fN (x2 − 1)δ(x1 − 1) These last two lines are completely equivalent. the pdf of the sum is the convolution of the pdfs of the inputs.

v. X with mean µX and variance σX . X2 ) of random variables X1 and X2 is given by. Discrete: E [g(X1 . x2 ) g(X1 . X2 ) = X1 or g(X1 . Then. X2 ) = (X1 − µX1 )(X2 − µX2 ). X2 ) are: • Mean of X1 or X2 : g(X1 . So. 10. X2 ) = (X1 − µX1 )2 or g(X1 . 2 2 Often denoted σX1 and σX2 . we can write its CDF as FX (x) = P [X ≤ x] = Φ x − µX σX . 10.6 Gaussian Random Variables 2 For a single Gaussian r.5 Expectation Def ’n: Expected Value (Joint) The expected value of a function g(X1 . the input and output (respectively) of the example additive noise communication system. x2 ) 2. • Expected value of the product of X1 and X2 . X2 ) = (X2 − µX2 )2 . Continuous: E [g(X1 . X2 )PX1 . also called the ‘correlation’ of X1 and X2 : g(X1 . X2 )] = Typical functions g(X1 . • Covariance of X1 and X2 : g(X1 .X2 (x1 . 1. X2 ) = X1 X2 . we have the pdf. fX (x) = 1 2 2πσX e− (x−µX )2 2σ 2 Consider Y to be Gaussian with mean 0 and variance 1. • Variance (or 2nd central moment) of X1 or X2 : g(X1 . which has non-zero mean and non-unit-variance. for X. when σ = 1. X2 )fX1 . the CDF of Y is denoted as FY (y) = P [Y ≤ y] = Φ(y). X2 )] = X1 ∈SX1 X1 ∈SX1 X2 ∈SX2 X2 ∈SX2 g(X1 . X2 ) = X2 will result in the means µX1 and µX2 .X2 (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 .

and in Matlab. 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 left-hand side is a unit-variance. Q(x). it is given its own name. 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. X exceeds some value x is one minus the CDF. 10. √ 1 1 Φ( 2x) = erf(x) + 2 2 Finally let y = √ 2x.1 Complementary CDF The probability that a unit-variance. zero mean Gaussian CDF.6. zero mean Gaussian r.6. 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). Q(x) = P [X > x] = 1 − Φ (x) What is Q(x) in integral form? Q(x) = x 2 For an Gaussian r. This is so common in digital communications. that is. we can write Φ(·) in terms of the erf(·) function.2 Error Function x − µX σX =1−Φ x − µX σX In math. the Q(x) function is not used. 1 − Φ(x). in some texts. so that 1 Φ(y) = erf 2 y √ 2 + 1 2 0 √ 2x .v. zero mean Gaussian random variable. ∞ 1 2 √ e−w /2 dw 2π P [X > x] = Q 10. we can write the probability of this event using the unit-variance.v. Instead.

• If X2 > 1/3. X1 ∈ {0. 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 . Given that X1 = 0. 1} with equal probabilities and N is independent of X1 and zero-mean Gaussian with variance 1/4. what is the probability that the receiver decides that a ‘1’ was sent? Solution: 1.* erf(y. Then the probability that the receiver decides ‘0’ is the probability that X2 ≤ 1/3.7 Examples Example: Probability of Error in Binary Example As in the previous example. P [error|X1 = 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. the probability that the receiver decides ‘1’ is the probability that X2 > 1/3. Given that X1 = 1. what is the probability that the receiver decides that a ‘0’ was sent? 2. P [error|X1 = 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 ‘0’.1/2 ./sqrt(2)). we have a model system in which the receiver sees X2 = X1 + N .ECE 5520 Fall 2009 44 Or in terms of Q(·). with mean 1 and variance 1/4. Here. Given that X1 = 1. it is clear that X2 is also a Gaussian r. 1. since X2 = X1 + N .v. then decide that the transmitter sent a ‘1’. 10. Given that X1 = 0. The receiver decides as follows: • If X2 ≤ 1/3.

Lecture 7 Today: (1) Matlab Simulation (2) Random Processes (3) Correlation and Matched Filter Receivers . fR (r) = r σ2 +s exp − r 2σ2 2 2 I0 rs σ2 0.ECE 5520 Fall 2009 45 10.w.3 spends some time with 2-D Gaussian random vectors. Gaussian with mean 0 and variance σ 2 . Both pdfs are sometimes needed to derive probability of error formulas.w. r≥0 o. 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. If the elements of X were independent random variables. This is the Rayleigh distribution. 0). An n-length R. X2 ). and covariance matrix CX if it has the pdf. X is multivariate Gaussian with mean µX .V. Rician) distribution. the pdf of R is fR (r) = r σ2 r exp − 2σ2 . if we have a matrix A and we let a new random vector Y = AX.V. the envelope R would now have a Rice (a.d.a. Any linear combination of jointly Gaussian random variables is another jointly Gaussian random variable.1 Envelope The envelope of a 2-D random vector is its distance from the origin (0.i. and CX is the inverse of the covariance matrix. 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. For example.k. If instead X1 and X2 are independent Gaussian with means µ1 and µ2 . . 10. and I0 (·) is the zeroth order modified Bessel function of the first kind. then Y is also a Gaussian random vector with mean AµX and covariance matrix ACX AT .8. o. and identical variances σ 2 . if the vector is (X1 . the envelope R is R= 2 2 X1 + X2 (15) If both X1 and X2 are i. r ≥ 0 2 0. 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. which is the dimension with which we spend most of our time in this class.8 Gaussian Random Vectors Def ’n: Multivariate Gaussian R. Specifically. where s2 = µ2 + µ2 . respectively.

We can estimate the autocorrelation function of an ergodic. RX (n. s) is a function of time t and outcome (realization) s. 2. An example that helps demonstrate the difference between time averages and ensemble averages is the non-ergodic process of opinion polling. s) is µX (t) = E [X(t. k) depends only on k and not on n. For an ergodic random process. but we now must define “ergodic”. respectively. For example. τ ) depends only on the time difference τ and not on t. RX (t. we omit the “s” when writing the name of the random process or random sequence. Consider what happens when a pollster takes either a time-average or a ensemble average: . multiple receivers would record different noisy signals even of the same transmission.1 Autocorrelation and Power Def ’n: Mean Function The mean function of the random process X(t. We then denote the autocorrelation function as RX (τ ). 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. Typically. Outcome or realization is included because there could be ways to record X even at the same time. 11. k) = E [X(n)X(n − k)] Def ’n: Wide-sense stationary (WSS) A random process is wide-sense stationary (WSS) if 1. 2. you don’t have anything to average to get the mean function µX (t). The power of a signal is given by RX (0). A random process is wide-sense stationary (WSS) if 1. wide-sense stationary random process from one realization of a power-type signal x(t). We then denote the autocorrelation function as RX (k). s) is a sequence of random variables indexed by time index n and realization s ∈ S. A random sequence X(n. its time averages are equivalent to its ensemble averages. and abbreviate it to X(t) or X(n). µX = µX (t) = E [X(t)] is independent of t. µX = µX (n) = E [X(n)] is independent of n.ECE 5520 Fall 2009 46 11 Random Processes A random process X(t. If you record one signal over all time t. s)] Note the mean is taken over all possible realizations s. 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. Def ’n: Autocorrelation Function The autocorrelation function of a random process X(t) is RX (t.

i.5. 1015 Hz). on the same day.. but probably not.2) has a good analysis of the physics of thermal noise. m = n.e. For continuous time signals. i=1 This transmission conveys to us which of M messages we want to send. The Rice book (4. because as frequency goes very high (e. log2 M bits of information. whether or not they will vote for person X. the power spectral density goes to zero. white noise is a commonly used approximation for thermal noise. every day for N days.ECE 5520 Fall 2009 47 • Time Average: The pollster asks the same person.. 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. This makes the process non-ergodic. thermal noise is not constant in frequency.1. whether or not she will vote for person X.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. 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. • Ensemble Average: The pollster asks N different people. At the receiver. The time average is taken by dividing the total number of “Yes”s by N . SX (f ) = F {RX (τ )} SX (ejΩ ) = DTFT {RX (k)} 11. Will they come up with the same answer? Perhaps. • 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). it is uncorrelated with X(m). that is. The ensemble average is taken by dividing the total number of “Yes”s by N . 12 Correlation and Matched-Filter Receivers We’ve been talking about how our digital transmitter can transmit at each time one of a set of signals {si (t)}M . 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? .g.

if n(t) = 0 and b = 1 then x = ai where ai is the signal space vector for si (t). N . Define nk = n(t). . .6 in Rice (page 226). the inner product is a correlation: r(t).1. 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’. If si (t) are non-orthogonal.1 Correlation Receiver ∞ −∞ What we’ve been talking about it the inner product.ECE 5520 Fall 2009 48 12. As notation. Noisy Channel Now. {φk (t)}N . . then we can reduce the amount of correlation (and thus multiplication and addition) done in the receiver by instead. correlating with the basis functions. φk = ∞ n(t)φk (t)dt. φk (t) = for k = 1 . x = [x1 . shown in Figure 5.k + n(t)φk (t)dt = ai. xN ]T Now that we’ve done these N correlations (inner products). . let b = 1 but consider n(t) to be a white Gaussian random process with zero mean and PSD SN (f ) = N0 /2. nk is zero mean: E [nk ] = ∞ −∞ E [n(t)] φk (t)dt = 0 .k + nk First. Noise-free channel Since r(t) = bsi (t) + n(t). . . In other terms. Correlation with the basis functions gives k=1 xk = r(t). −∞ 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.

k )2 2(N0 /2) fx (x) = k=1 N fXk (xk ) (x −a ) 1 − k i. 1 − e N/2 [2π(N0 /2)] PN 2 k=1 (xk −ai. nN are i.ECE 5520 Fall 2009 49 Next we can show that n1 . .d.i.m = 0.k e 2(N0 /2) 2π(N0 /2) 2 = k=1 = An example is in ece5510 lec07.k ) 2(N0 /2) 12. then xk are also independent. since {Xk } are independent. by calculating the autocorrelation Rn (m. Then since nk is fXk (xk ) = And. . . 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 . (Why?) What is the pdf of xk ? What is the pdf of x? Solution: Gaussian.w. .k is a deterministic constant. m=k δk.m. 2 Is {nk } WSS? Is it a Gaussian random sequence? Since the noise components are independent. k). 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 .k + nk and ai. N 1 2π(N0 /2) e − (xk −ai. Rn (m. o. xk are independent because xk = ai.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 .

• This works at time T . . . have a physical filter with the impulse response φk (T − t) and thus it is easy to do a matched filter implementation. . that is. 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. Try out the Matlab code. taken at time T .1.m. and detection.4 Review At a digital transmitter. as shown in Figure 19. A real channel attenuates! The job of our receiver will be to amplify the signal by approximately 1/b. . At the receiver. textbooks will assume that the automatic gain control (AGC) works properly. We might. we assume we receive signal si (t) corrupted by noise: r(t) = si (t) + n(t) How do we decide which signal i ∈ {0. This effectively multiplies the noise. • These are just two different physical implementations.ECE 5520 Fall 2009 50 where hk (t) = φk (T − t). correlation and matched filter rx. 12.) This is the output of a convolution. • It may be easier to ‘see’ why the correlation receiver works. Notes: • The xk can be seen as the output of a ‘matched’ filter at time T . gain control. for example. log2 M bits of information. correlation or matched filter reception. (Plug in (T − t) in this formula and the T s cancel out and only positive t is left. This transmission i=0 conveys to us which of M messages we want to send. .4. xk = r(t) ⋆ hk (t)|t=T Or equivalently xk = r(t) ⋆ φk (T − t)|t=T This is Rice Section 5. we can transmit at each time one of a set of signals {si (t)}M −1 . M − 1} was transmitted? We split the task into down-conversion. which is posted on WebCT. The output for other times will be different in the correlation and matched filter. In general.3 Amplitude Before we said b = 1. and then will assume the noise signal is multiplied accordingly. Lecture 8 Today: (1) Optimal Binary Detection 12.

. . ai. Gaussian. Both result in the same output x. We simulated this in the Matlab code correlation and matched filter rx. ai.d. as shown in Figure 20. x and ai should be close since i was actually sent. 13. . Optimal detection uses rules designed to minimize the probability that a symbol error will occur. then the decision will be.5 Correlation Receiver Our signal set can be represented by the orthonormal basis functions. -1 0 X 1 f1 Figure 20: Signal space diagram of a PAM system with an X to mark the receiver output x. 1]T and receiving r(t) (and thus x) in noise. because: • xk = ai. {φk (t)}K−1 . . In an example Matlab simulation. then we’ll need to shift the decision boundaries. . 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.m. we simulated sending ai = [1. . .k + nk for k = 0 . 13 Optimal Detection We receive r(t) and use the matched filter or correlation receiver to calculate x. . x1 .1 . . Given the signal space diagram of the transmitted signals and the received signal space vector x. .m. n1 . . . pick the i with ai closest to x. how exactly do we decide which si (t) was sent? Consider this in signal space.K−1 ]T n = [n0 . nK−1 ]T Hopefully.k ) 2(N0 /2) (16) We showed that there are two ways to implement this: the correlation receiver.i. ece5520 lec07. Correlation with the basis k=0 functions gives xk = r(t). .k + nk • ai. φk (t) = ai. and the matched filter receiver. K − 1. what rules should we have to decide on i? This is the topic of detection. xK−1 ]T ai = [ai. In general.ECE 5520 Fall 2009 51 12.1 Overview We’ll show two things: • If each symbol i is equally probable.k is a deterministic constant • The {nk } are i. . Now. We denote vectors: x = [x0 . the pdf of x is multivariate Gaussian with each component xk independent.0 . • If symbols aren’t equally probable. .

and S is the complete event space. When we talk about them as random variables. motion detection. we mean that we want the smallest probability of error. for example. Then using Bayes’ Law. • Radar applications: Obstruction detection. the symbols that could have been sent. The event listing is.2 Bayesian Detection When we say ‘optimal detection’ in the Bayesian detection framework. and n. we list the events that could have occurred. The probability of error is denoted P [error] By error. the events H0 ∪ H1 ∪ · · · ∪ HM −1 = S. we mean that a different signal was detected than the one that was sent. We need to decide from X whether s0 (t) or s1 (t) was sent. 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’. a0 and a1 . instead of vectors x. 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.e. That is. Further. • 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. H0 : H1 : Equivalently. detection is applicable in a wide variety of problems. P [error] = P [error|H0 ] P [H0 ] + P [error|H1 ] P [H1 ] (17) X = a0 + N X = a1 + N r(t) = s0 (t) + n(t) r(t) = s1 (t) + n(t) . i. At the start of every detection problem.. So even though it is somewhat theoretical. signal detection. we’ll have only scalars: x. where the ∪ means union.ECE 5520 Fall 2009 52 Detection theory is a major learning objective of this course. a0 and a1 . 13. and n. Then. it is very applicable in digital receivers. and only one basis function φ0 (t). 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). We follow all detection and statistics textbooks and label these classes Hi .

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. Over a different set R1 of values.3 Selecting R0 to Minimize Probability of Error We can see what the integrand looks like. the difference between the joint densities. 14. Figure ?? shows the joint densities (the conditional pdfs multiplied by the bit probabilities P [H0 ] and P [H1 ]. then this is the region in which X is more probable given H0 than given H1 . we’ll decide that H0 happened (s0 (t) was sent). so • There is no overlap: R0 ∩ R1 = ∅. Everything else is determined by the time we get to this point. Figure ?? shows the full integrand of (19). Figure 21 shows the conditional probability density functions. 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 . we’ll decide H1 occurred (that s1 (t) was sent). but the only thing we can change is the region R0 . We can’t be indecisive. . P [error] = P [H0 ] − + x∈R0 x∈R0 fX|H0 (x|H0 )P [H0 ] dx fX|H1 (x|H1 )P [H1 ] dx (19) fX|H1 (x|H1 )P [H1 ] − fX|H0 (x|H0 )P [H0 ] dx = P [H0 ] + x∈R0 We’ve got a lot of things in the expression in (19).we just say what region of x. So the question is. how do you pick R0 to minimize (19)? 14.1 Decision Region We’re making a decision based only on X.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 . The objective is to minimize the probability of error. • There are no values of x disregarded: R0 ∪ R1 = S. Then R0 is the area in which fX|H0 (x|H0 )P [H0 ] > fX|H1 (x|H1 )P [H1 ] If P [H0 ] = P [H1 ]. and the integral in (19) will integrate over it.ECE 5520 Fall 2009 53 14. We can pick R0 however we want . Over some set R0 of values of X. since the two are complementary sets. Finally.

(b) joint p.d.f.2 −0.d. .2 0 −3 −2 −1 (a) 0 r 1 2 3 (b) 0.f.6 −3 −2 −1 (c) 0 r 1 2 3 Figure 21: The (a) conditional p.f.4 0.ECE 5520 Fall 2009 54 1 f 0. fX|H1 (x|H1 )P [H1 ] − fX|H0 (x|H0 )P [H0 ].s fX|H0 (x|H0 )P [H0 ] and fX|H1 (x|H1 )P [H1 ].s.8 f X|H X|H 1 2 Probability Density 0.s (likelihood functions) fX|H0 (x|H0 ) and fX|H1 (x|H1 ).4 f 0. and (c) difference between the joint p.2 X∩H −f 2 X ∩H 1 Integrand 0 −0.d.4 −0.6 0. which is the integrand in (19).

Whenever x indicates that the likelihood ratio is less than the threshold.5 Case of a0 = 0. Continuing with the log-likelihood ratio. 14.e. n ∼ N (0. we’ll decide H1 . that s1 (t) was sent.ECE 5520 Fall 2009 55 Rearranging the terms. then we’ll decide H0 .. What is: 1. Equation (20) is a very general result. applicable no matter what conditional distributions x has. Otherwise. fX|H1 (x|H1 ) P [H0 ] < fX|H0 (x|H0 ) P [H1 ] (20) The left hand side is called the likelihood ratio. The right hand side is a threshold. Both sides are positive. the log-likelihood ratio is fX|H1 (x|H1 ) P [H0 ] < log log fX|H0 (x|H0 ) P [H1 ] 14. the math gets much easier if we take the log of both sides. Why can we do this? Solution: 1. The log-likelihood ratio? 3. The log() function is strictly increasing. that s0 (t) was sent. The log of a Gaussian pdf? 2. 2. In addition. Now.e. i. i. assume for a minute that a0 = 0 and a1 = 1. σN ).. The decision regions for x? Solution: What is the log of a Gaussian pdf? log fX|H0 (x|H0 ) = log   1 2 2πσN e 2 − x2 2σ N   (21) 1 x2 2 = − log(2πσN ) − 2 2 2σN The log fX|H1 (x|H1 ) term will be the same but with (x − 1)2 instead of x2 .4 Log-Likelihood Ratio For the Gaussian distribution. a1 = 1 in Gaussian noise 2 In this example. log fX|H1 (x|H1 ) − log fX|H0 (x|H0 ) < 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 .

then x = 1 and the logarithm of the fraction is zero. but now. decide that s0 (t) was sent. Rather than writing both inequalities each time. which direction will the optimal threshold move. P [H1 ] = 0. towards a0 or towards a1 ? Solution: Towards a1 . and P [H0 ] = 0.6 General Case for Arbitrary Signals If. a1 = 1. we use the following notation: x H1 > < H0 1 P [H0 ] 2 + σN log 2 P [H1 ] This completely describes the detector receiver.ECE 5520 Fall 2009 56 In the end result.if it is below the decision threshold. As long as a0 < a1 . The boundary is exactly half-way in between the two signal space vectors. This receiver is also called a maximum likelihood detector. 2 Example: Let a0 = −1.1. So then a0 + a1 2 The decision above says that if x is closer to a0 . P [H0 ] 1 2 x > + σN log 2 P [H1 ] decide H1 . σN = 0. we had arbitrary values for them (the signal space representations of s0 (t) and s1 (t)). decide H0 .8 Examples Example: When H1 becomes less likely. γ= 2 σN P [H0 ] a0 + a1 + log 2 a1 − a0 P [H1 ] (23) 14. If it is above the decision threshold. decide that s1 (t) was sent. we also write x H1 > < H0 γ where γ= 1 P [H0 ] 2 + σN log 2 P [H1 ] (22) 14. we could have derived the result in the last section the same way.7 Equi-probable Special Case P [H1 ] P [H0 ] H1 > < H0 If symbols are equally likely. What is the decision threshold for x? . For simplicity. we’d still have r H1 > < H0 γ. there is a simple test for x .4. because we only decide which likelihood function is higher (neither is scaled by the prior probabilities P [H0 ] or P [H1 ]. instead of a0 = 0 and a1 = 1. 14. P [H1 ] = P [H0 ].6. And if x is closer to a1 .

calculate the probability of error from (18). one-dimensional signalling.ECE 5520 Fall 2009 57 Solution: From (23). 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 > γ|H0 ] P [H0 ] + P [x < γ|H1 ] P [H1 ] 2 What is the first probability. and P [H0 ]. Example: In this example.4 Example: Can the decision threshold be higher than both a0 and a1 in this binary. • We used the log operator to show that for the Gaussian error case the decision regions are separated by a single threshold. and in the general case.1 log = 0. • We found the decision regions which minimized the probability of error. both in the equi-probable symbol case.0203 2 0. given that x|H1 is Gaussian with mean a1 and variance σN of error? Solution: P [x > γ|H0 ] = Q γ − a0 σN γ − a1 a1 − γ P [x < γ|H1 ] = 1 − Q =Q σN σN γ − a0 P [error] = P [H0 ] Q + P [H1 ] Q σN a1 − γ σN 14. you should be able to calculate the optimal decision threshold γ. P [H1 ]. a1 . γ =0+ 0.5 ≈ 0.9 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. given all of the above constants and the optimal threshold γ.05 log 1. receiver? 2 Given a0 . • We showed the formula for that threshold. Lecture 9 Today: (1) Finish Lecture 8 (2) Intro to M-ary PAM . given that r|H0 is Gaussian with mean a0 and variance σN ? What is the second 2 ? What is then the overall probability probability. σN .6 0.

. it is on the probability topic. 15. since there’s only one basis function. That is. Note: Keep in mind that when s(t) is to be transmitted on a bandpass channel. −1 ≤ t ≤ 1 and x1 (t) = 0.w. for i = 0. on the RHS of the given expressions should be “t”s. . −1 ≤ t ≤ 1 . . and ai instead of ai. 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. the “x”s o. by putting these sample exams up. we are sending one symbol every Ts units of time. you know the actual exam won’t contain the same exact problems. That makes our symbol rate as 1/Ts symbols per second. 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 .k. • Problem 3 is not on the topic of detection theory. There is a typo: fN (n) should be fN (t). • Problem 6 typo: x1 (t) = Notes on 2007 sample exam: • Problem 2 should have said “with period Tsa ” rather than “at rate Tsa ”. 0. o. . • The book that year used ϕi (t) instead of φi (t) as the notation for a basis function.ECE 5520 Fall 2009 58 Sample exams (2007-8) and solutions posted on WebCT. • Problem 6 is on the topic of detection theory. and Ts is the symbol period (not the sample period!). M where ai is the amplitude of waveform i. aM −1 } is the nth symbol transmitted. you’ll just add each time-delayed signal to the transmitted waveform. x(t) = ℜ s(t)ej2πfc t . “symbol”) conveys k = log2 M bits of information. t. Notes on 2008 sample exam: • Problem 1(b) was not material covered this semester. both for simplicity. There is no guarantee this exam will be the same level of difficulty of either past exam. .1 Baseband Signal Examples When you compose a signal of a number of symbols. . 1 2 2 (3t − 1). and αi instead of si as the signal space vector for signal si (t). Note we’re calling it p(t) instead of φ0 (t). Each sent waveform (a. .w.a. Instead of just sending one symbol. Of course. it is modulated with a carrier cos(2πfc t). si (t) = ai p(t).0 . .

4 Time t 0. The average energy per symbol has decreased.2. It shows a signal set using square pulses. Typical M -ary PAM (for all even M ) uses the following amplitudes: −(M − 1)A. √ 1/ Ts . Example: Binary Bipolar PAM Figure 22 shows a 4-ary PAM signal set using amplitudes a0 = −A. Example: 4-ary PAM A 4-ary PAM signal set using amplitudes a0 = −3A. . respectively.2 0. . If the y-axis in Figure 22 was scaled such that the minimum was 0 instead of −1. Es = E = E ∞ −∞ 2 ai a2 p2 (t)dt = i ∞ −∞ E a2 φ2 (t)dt i 0 (24) . A.2 Average Bit Energy in M-ary PAM Assuming that each symbol si (t) is equally likely to be sent. . 15. Example: Binary Unipolar PAM Unipolar binary PAM uses the amplitudes a0 = 0.6 show continuous-time and discrete-time realizations.ECE 5520 Fall 2009 59 Rice Figures 5.1 shows the signal space diagram of M-ary PAM for different values of M . . . a2 = A. a1 = −A. of PAM transmitters and receivers.2. −A. +(M − 1)A Rice Figure 5. √ 1/ Ts . o. a3 = 3A is shown in Figure 23.6 0. 0 ≤ t < Ts p(t) = 0.w. +(M − 3)A. This is also called On-Off Keying (OOK). 1 0. we want to calculate the average bit energy Eb .w. which is 1/ log 2 M times the symbol energy Es . .2. . −(M − 3)A. 0 ≤ t < Ts φ0 (t) = p(t) = 0. It shows a signal set using square pulses. it would represent unipolar PAM.5 Value 0 −0.8 1 Figure 22: Example signal for binary bipolar PAM example for A = √ Ts . a1 = A. a1 = A. . o.3 and 5.5 −1 0 0.

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 . Sampling and aliasing 4. as described above to be the typical M-ary PAM case. .8 1 Figure 23: Example signal for 4-ary PAM example. Correlation and Covariance 6. Fourier transform of signals. Bandpass signals 3. properties (12) 2.4 Time t 0.ECE 5520 Fall 2009 60 3 2 1 Value 0 −1 −2 −3 0 0. .6 0. . Correlation / matched filter receivers . 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 . (M − 1)A.2 0. Signal space representation 7. Orthogonality / Orthonormality 5. . −(M − 3)A. . . aM −1 } = −(M − 1)A. . .

If anything is confusing.W. Read over the lecture notes 1-7. You will be provided with Table 2. Problem 2-49).w. (a) Show that these functions are orthogonal over the interval (−4.ECE 5520 Fall 2009 61 Make sure you know how to do each HW problem in HWs 1-3.8. Couch 2007. Bateman Problem 1. (b) Find the scale factors needed to scale the functions to make them into an orthonormal set over the interval (−4. 2.  0. The spectrum of x(t) is bandlimited to −W ≤ ω ≤ W .   1. Three functions are shown in Figure P2-49. (from L. 0 ≤ t ≤ 4 0.100 What was the original continuous-time signal x(t)? Or.w.39. 4). It will be good to know the ‘tricks’ of how each problem is done. its samples Xn are. When x(t) is sampled at a rate n = ±1 n=0 o.3 Orthogonality and Signal Space 1.4. (c) Express the waveform w(t) = 1.9: 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.5×11 sheet of paper for notes. 2.4. Rice 2. if x(t) cannot be determined.4 and Table 2. 4. πt Ts . . why not? 17. Prove that the spectrum of y(t) is bandlimited to −2W ≤ ω ≤ 2W . 2. since they will reappear. How does this affect the spectrum of the output signal? 2.25. (b) Find the Fourier transform of x(t). 4). Ts 2 17. in signal space using the orthonormal set found in part (b).52 cos 0. Haykin & Moyer Problem 2. You may have one side of an 8. Consider the pulse shape x(t) = (a) Draw a plot of the pulse shape x(t). ask me.w.99. Assume that x(t) is a bandlimited signal with X(f ) 1 T = 2W . 3. Xn = 2. 1.1 Additional Problems Spectrum of Communication Signals 1. 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). − Ts < t < 2 o. o. 17 17.2 Sampling and Aliasing = 0 for |f | > W . Rice 2.

i. (a) Is s1 (t) an energy-type or power-type 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. and n0 is zero-mean Gaussian with variance 0.5. 5. πt T T 2 T 2 − cos 0. . πt T . H0 : H1 : r = a0 + n0 r = a1 + n1 where α0 = −1 and α1 = 1. find the value of RC to satisfy the design specifications.4 Random Processes. A binary communication system uses the following equally-likely signals. and n1 is zero-mean Gaussian with variance 0.e.. 3. 2. Let a binary 1-D communication system be described by two possible signals. (From Couch 2007 Problem 2-80) An RC low-pass filter is the circuit shown in Figure 24. 5.1. a signal x(t) = si (t) + n(t) is received. .e. and noise with different variance depending on which signal is sent.5 Correlation / Matched Filter Receivers 1.w. At the receiver. s1 (t) = s2 (t) = cos 0. −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. design an RC low-pass filter that will attenuate this signal by 20 dB at 15 kHz. Rice 5.1 on Page 219. 5. i. Rice Example 5.14.16. (a) What is the conditional pdf of r given H0 ? (b) What is the conditional pdf of r given H1 ? 17. Its transfer function is + R C + X(t) - Y(t) - Figure 24: An RC low-pass filter. PSD 1.w.1.13. 5. 5. SX (f ) = 1.23.2. −T ≤ t ≤ 2 o.3.26 17. 5. That is.ECE 5520 Fall 2009 62 2.

M 1/2 di.1 Signal Distance We mentioned. In the Gaussian noise case (with equal variance of noise in H0 and H1 ). Noise PSD We have consistently used a1 > a0 . a distance between vectors. 3.k − aj.1. we came up with threshold γ = (a0 + a1 )/2. there is only d1.0 = |a1 − a0 | (26) 18.ECE 5520 Fall 2009 63 2. Suppose that two signal waveforms s1 (t) and s2 (t) are orthogonal over the interval (0. white noise process is correlated with s1 (t) and 2 (t) to yield. How are N0 /2 and σN related? Recall from lecture 7 that the variance of nk came out to be N0 /2.16. T ). Prove that when a sinc pulse gT (t) = sin(πt/T )/(πt/T ) is passed through its matched filter. Proakis & Salehi 7. A sample function n(t) of a zero-mean. We had 2 also called the variance of noise component nk as σN . two signal case. 2 (25) 18. when talking about signal space diagrams.2 BER Function of Distance. So: 2 σN = N0 .2 of Rice.0 .8. the output is the same sinc pulse. and P [error] = P [x > γ|H0 ] P [H0 ] + P [x < γ|H1 ] P [H1 ] .k ) 2 d1. k=1 (ai. Lecture 11 Today: (1) M-ary PAM Intro from Lecture 9 notes (2) Probability of Error in PAM 18 Probability of Error in Binary PAM This is in Section 6.j = ai − aj = For the 1-D. Proakis & Salehi 7. T n1 = 0 T s1 (t)n(t)dt s2 (t)n(t)dt 0 n2 = Prove that E [n1 n2 ] = 0. and with P [H0 ] = P [H1 ].

when A0 = 0 and A1 = A. our signals s0 (t) = a0 p(t) and s1 (t) = a1 p(t) are just s0 = a0 and s1 = a1 . and again with P [H0 ] = P [H1 ] = 1/2. This expression. when A0 = −A and A1 = A. we’d have seen that in general.  A2 2N0   P [error] = Q  .3 Binary PAM Error Probabilities For binary PAM (M = 2) it takes one symbol to encode one bit. Thus ‘bit’ and ‘symbol’ are interchangeable. we have P [error] = Q d0. For bipolar signaling.ECE 5520 Fall 2009 64 Which simplified using the Q function.     2 2 4A  2A  = Q P [error] = Q  2N0 N0 For unipolar signaling.   2 d0. using (25) and (26). the following formula works: |a1 − a0 | P [error] = Q 2σN Then. so if we had also calculated for the case a1 < a0 . This will be important as we talk about binary PAM. P [x > γ|H0 ] = Q γ − a0 σN γ − a1 P [x < γ|H1 ] = 1 − Q σN =Q a1 − γ σN (27) Thus P [x > γ|H0 ] = Q P [x < γ|H1 ] = Q So both are equal.1 2 N0 /2 = Q  d2 0.1 2N0   (30) Is one that we will see over and over again in this class.1  Q 2N0 18. Now we can use (30) to compute the probability of error. P [error] = Q (a1 − a0 )/2 σN (29) (a1 − a0 )/2 σN (a1 − a0 )/2 σN (28) We’d assumed a1 > a0 . In signal space.

  2 4A  2Eb =Q P [error] = Q  2N0 N0 For unipolar signaling. the bit error rate of bipolar PAM beats unipolar PAM. • Even for the same bit energy Eb . 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. Using (24). Since half of the bits are exactly zero. For the unipolar signalling. this doesn’t take into consideration that the unipolar signaling method uses only half of the energy to transmit. For bipolar signaling. A2 = A2 always. P [error] = Q Discussion 2Eb 2N0 =Q Eb N0 These probabilities of error are shown in Figure 25. • 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. we re-write the probability of error expressions in terms of Eb . (2) Probability of Error in M-ary PAM . (1) Multi-Hypothesis Detection Theory.ECE 5520 Fall 2009 65 However. Thus Eb = Es = 1 A2 . so Eb = Es = A2 . dB b 0 12 14 Figure 25: Probability of Error in binary PAM signalling. 2 Now. they take no signal energy. A2 m m equals A2 with probability 1/2 and zero with probability 1/2. what is the average energy of bipolar and unipolar signaling? Solution: For the bipolar signalling.

Essentially. maximum likelihood detection. we’ll only consider the case of equally probable signals. this is a (squared) distance between x and ai . The decision between two neighboring signal vectors ai and ai+1 will be r Hi+1 > < Hi γi. .v. For this class. it is better to maximize the log of the likelihood rather than the likelihood directly. a1 g12 a2 g23 a3 g34 a4 r Figure 26: Signal space representation and decision thresholds for 4-ary PAM. So. we’ll see that we need to consider the highest of all M possible events Hm . (While equi-probable signals is sometimes not the case for M = 2 binary detection. the decision is. Symbol Decision = arg max fX|Hi (x|Hi ) i 2 For Gaussian (conditional) r.i+1 = ai + ai+1 2 As an example: 4-ary PAM. H0 : H1 : HM −1 : which have joint probabilities. it is very rare in higher M communication systems. that is.ECE 5520 Fall 2009 66 19 Detection with Multiple Symbols When we only had s0 (t) and s1 (t). H0 : H1 : HM −1 : ··· fX|H0 (x|H0 )P [H0 ] fX|H1 (x|H1 )P [H1 ] fX|HM −1 (x|HM −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. find the ai which is closest to x.) If P [H0 ] = · · · = P [HM −1 ] then we only need to find the i that makes the likelihood fX|Hi (x|Hi ) maximum. we saw that our decision was based on which of the following was higher: fX|H0 (x|H0 )P [H0 ] and fX|H1 (x|H1 )P [H1 ] For M -ary signals. See Figure 26. so (x − ai )2 1 2 log fX|Hi (x|Hi ) = − log(2πσN ) − 2 2 2σN This is maximized when (x − ai )2 is minimized.s with equal variances σN .

ECE 5520 Fall 2009 67 20 20. the decision threshold is always A away from the symbol values. P (symbol error|Hi ) = Q So overall. For the symbols i in the ‘middle’. 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. P (symbol error|Hi ) = 2Q For the symbols i on the ‘sides’.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 . Also the noise σN = N0 /2. Here. as discussed above. .1 M-ary 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. P (symbol error) = 20. log2 M 3 How does this relate to the average bit energy Eb ? From last lecture. Assuming neighboring symbols ai are spaced by 2A. dB b 0 12 14 Figure 27: Probability of Symbol Error in M -ary PAM. each ai = Ai . Eb = A= So P (symbol error) = Equation (31) is plotted in Figure 27.1.

. “110” “111” “101” “100” “000” “001” “011” “010” f1 -1 0 2 1 3 -3 -2 Figure 29: Gray encoding for 8-ary PAM. QAM) when this is not a good approximation. If Gray encoding is used.2.use (32). bits and symbols are not synonymous. At least at high Eb /N0 . . Example: Bit coding of M = 4 symbols While the two options shown in Figure 28 both assign 2-bits to each symbol in unique ways. as in Option 2 in Figure 28. For M -ary PAM. one will lead to a higher bit error rate than the other. 20. we need to study further the probability that x will jump more than one decision region. 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). The neighbors of ai will be more likely than distant signal space vectors. Thus Gray encoding will only change one bit across boundaries. up to log2 M in the worst case. As of now .1 Bit Error Probabilities How many bit errors are caused by a symbol error in M -ary PAM? • One.2 Bit Errors and Gray Encoding For binary PAM. there are only two symbols. The key is to recall the model for noise.if you have to estimate a bit error rate 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. we have to carefully assign bit codes to symbols 1 . the errors will tend to be just one bit flipped. P (error) ≈ 1 P (symbol error) log2 M (32) • Maybe more.ECE 5520 Fall 2009 68 20. Then.g. It will tend to leave the received signal x close to the original signal ai . We will also show examples in multi-dimensional signalling (e. more than multiple bits flipped. Solution: Here is one solution. When you make one symbol error (decide H0 or H1 in error) then it will cause one bit error. . Instead. It will not shift a signal uniformly across symbols. M . one will be assigned binary 0 and the other binary 1. in almost all cases. . We will show that this approximation is very good.

2. consider the effect of filtering in the path between transmitter and receiver. 1. then r(t) = s(t) ⋆ h(τ. 3. noise power entering receiver. 2. In reality. Filters will be seen in 1. If we try (1) above and use FDMA (frequency division multiple access) then the interference is out-of-band interference. . We don’t want either: (1) occupies too much spectrum. The amplitudes of the impulses tend to decay over time. Now. t) = l=1 αl ejφl δ(τ − τl ). Typically. 2. Transmitter: Baseband filters come from the limited bandwidth. t) will be received. or scattering adds an additional impulse to (33) with a particular time delay (depending on the extra path length compared to the line-of-sight path) and complex amplitude (depending on the losses and phase shifts experienced along the path). diffraction. RF filters are used to limit the spectral output of the signal (to meet out-of-band interference requirements). call its impulse response h(τ. the radio channel is modeled as L h(τ. we want to compromise between (1) and (2) and experience only a small amount of both.ECE 5520 Fall 2009 69 Lecture 13 Today: (1) Pulse Shaping / ISI. our own symbols experience interference called inter-symbol interference. If s(t) is transmitted. The measured radio channel is a filter. speed of digital circuitry. Consider the time-domain of the signal which is close to a rect function in the frequency domain. If we try (2) above and put symbols right next to each other in time. and τl is its time delay. but multipath with delays of hundreds of nanoseconds often exist in indoor links and delays of several microseconds often exist in outdoor links. Consider the spectrum of the ideal 1-D PAM system with a square pulse.1 Multipath Radio Channel (Background knowledge). Note that the ‘matched filter’ receiver is also a filter! 21. (2) N-Dim Detection Theory 21 Inter-symbol Interference 1. (33) where αl and φl are the amplitude and phase of the lth multipath component. and (2) occupies too much time. Receiver: Bandpass filters are used to reduce interference. Waveguides have bandwidth limits (and modes). t). each reflection. Essentially. Channel: The wideband radio channel is a sum of time-delayed impulses. Wired channels have (nonideal) bandwidth – the attenuation of a cable or twisted pair is a function of frequency.

Key insight: .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.25 0.1 0. 22 Nyquist Filtering • We don’t need the leakage to be zero at all time.2 0. At this symbol rate. 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 0.05 0 0 100 (a) (b) 200 Delay.15 0. and (b) 14 to 43. each symbol would fully bleed into the next symbol.1 0.2µs is 1/ 5 MHz. ns 300 400 0.25 Amplitude Delay Profile 0.15 0. 0.2 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). 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. ns 300 400 Figure 30: Multiple delay profiles measured on two example links: (a) 13 to 43. Example: 2400-2480 MHz Measured Radio Channel The ‘Delay Profile’ is shown in Figure 30 for an example radio channel.05 0 0 Amplitude Delay Profile 100 200 Delay.

that there is an inner product of zero. are orthogonal to each other.−1. shown in Figure 32 accomplishes this. exactly constant within − 2Ts < f < 1 2Ts band and zero outside.w. symmetric about f = 1/(2Ts ). X f+ m Ts = Ts Proof: on page 833. for integer n. Basically.. forms an orthonormal basis. See Figure 31.. Where have you seen this condition before? This condition. we denote the difference between the ideal rect function and our X(f ) as ∆(f ). What we need are pulse shapes (waveforms). o. i. at the matched filter output. See Figure 31 .0. of Rice book.. ∆ 1 −f 2Ts =∆ 1 +f 2Ts 1 Ts + X(f ) + X f + 1 Ts + ··· for −1/Ts ≤ f < 1/Ts . the sum is constant.e. we need p(t) such that {φn (t) = p(t − nTs )}n=.. Essentially.. Appendix A. making the sum of ‘leakage’ or ISI from one symbol be exactly 0 at all other sampling times nTs . X(f ) = 0 for all |f | > Ts ).t. In more mathematical language. ∆(f ) = |X(f ) − rect(f Ts )| then we can rewrite the Nyquist Filtering condition as. 1 If X(f ) only bleeds into one neighboring channel (that is. that when shifted in time by Ts .ECE 5520 Fall 2009 • The value out of the matched filter (correlator) is taken only at times nTs . is the condition for orthogonality of two waveforms. n = 0 0. Thus X(f ) is allowed to bleed into other frequency bands – but the neighboring frequency-shifted copy of X(f ) must be lower s. • It may bleed into the next ‘channel’ but the sum of ··· + X f − must be constant across all f . X(f ) could be: 1 • X(f ) = rect(f Ts ). 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... The square root raised cosine.1. But lots of other functions also accomplish this. 71 Nyquist filtering has the objective of. The Nyquist filtering theorem provides an easy means to come up with others.

o. we need |H(f )|2 to meet the Nyquist filtering condition. one at the transmitter and one at the receiver. |f | − 1−α 2Ts . Leave a tiny bit so that it is not completely disconnected into two halves. and the vertical axis it X(f ). 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. .1 Raised Cosine Filtering HRC (f ) =   Ts . In other words.w. Unfold.5 × 1/Ts .  √ 0 ≤ |f | ≤ 1−α  Ts . a0 . Take a sheet of paper and fold it in half. which in series. . a modulation with two dimensional signal vectors. .2 Square-Root Raised Cosine Filtering   0. ≤ |f | 1−α 2Ts ≤ 1+α 2Ts (35) But we usually need do design a system with two identical filters. Setup: • Transmit: one of M possible symbols. 1−α ≤ |f | ≤ 1+α 2 α 2Ts 2Ts 2Ts    0. and then in half the other direction. Drawing a horizontal line for the frequency f axis. produce a zero-ISI filter.  Ts 2 1 + cos πTs α 22.w. and now we will extend that to N -dimensional signal vectors. and later even higher dimensional signal vectors. 2Ts   Ts (36) HRRC (f ) = 1 + cos πTs |f | − 1−α . . the middle is 0. 0 ≤ |f | ≤ 1−α 2Ts o.” Activity: Do-it-yourself Nyquist filter. .5/Ts .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 -frequency-shifted copies of X(f ) must add up to a constant. 22. We have developed M -ary detection theory for 1-D signal vectors. Cut a function in the thickest side (the edge that you just folded). aM −1 . 23 M-ary Detection Theory in N -dimensional signal space We are going to start to talk about QAM.

• Question: What are the optimal decision regions? When symbols are equally likely.05 −6 −5 −4 −3 −2 −1 i 0 1 2 Time. fX|Hi (x|Hi ) = which can also be written as fX|Hi (x|Hi ) = 1 2 (2πσN )N/2 exp − exp − j (xj 2 2σN − ai. are orthogonal to each other.05 0 −0. each component ni is independent with zero mean and variance σN = N0 /2. • Receive: the symbol plus noise: H0 : HM −1 : ··· X = a0 + n X = aM −1 + n ··· • Assume: n is multivariate Gaussian.25 Basis Function φ (t) 0.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) .15 0. the optimal decision turns out to be given by the maximum likelihood receiver.ECE 5520 Fall 2009 73 0. ˆ = argmax log fX|H (x|Hi ) i (37) i i Here. t/Tsy 3 4 5 6 7 Figure 32: Two SRRC Pulses. • Assume: Symbols are equally likely.2 0. delayed in time by nTs for any integer n.1 0.

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). This definition of the orthonormal basis specifically considers a pulse shape at a frequency ω0 . Previously. The combined decision region of Ri is the space which is the intersection of all pair-wise decision regions.ECE 5520 Fall 2009 74 Again: Just find the ai in the signal space diagram which is closest to x. 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 .) Example: Optimal Decision Regions See Figure 33. Draw a line segment connecting ai and aj . or subspace in N -D). respectively) as the two dimensions. (All conditions must be satisfied. Draw a point in the middle of that line segment. That is. 2. 3. p(t) is only non-zero within that window. cancel. Draw the perpendicular bisector of the line segment through that point. Thus QAM uses two basis functions. and reorganize to find a linear equation in terms of x. To find this line: 1. (2) QAM Probability of Error 24 Quadrature Amplitude Modulation (QAM) Quadrature Amplitude Modulation (QAM) is a two-dimensional signalling method which uses the in-phase and quadrature (cosine and sine waves. Decision Regions Each pairwise comparison results in a linear division of space. Lecture 14 Today: (1) QAM & PSK. 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 3-D. We include it here because . we’ve separated frequency up-conversion and down-conversion from pulse shaping. This is left as an exercise.

.ECE 5520 Fall 2009 75 (a) (b) (c) (d) Figure 33: Example signal space diagrams. Draw the optimal decision regions.

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). show that φ0 (t). consider the constant p(t) = c for a constant c. 24. it has low frequency content compared to ω0 t. let p(t) = c for a constant c . φ0 (t). φ1 (t) = c2 sin(ω0 (T2 + T1 )) sin(ω0 (T2 − T1 )) ω0 Solution: First. 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). so φ0 (t). with the same pulse shape p(t). • p(t) is ‘low pass’. Are the two bases orthogonal? First. for T1 ≤ t ≤ T2 . φ1 (t) We can see that there are two cases: = c2 sin(ω0 (T2 + T1 )) sin(ω0 (T2 − T1 )) ω0 . that is. for T1 ≤ t ≤ T2 . (This is not intuitive!) We have two restrictions on p(t) that makes these two basis functions orthonormal: • p(t) is unit-energy. 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.ECE 5520 Fall 2009 76 it is critical to see how. 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. we can have two orthogonal basis functions.

and so the correlation is exactly zero. The only remainder is the remainder (partial cycle) period. in the same way that the earlier integral was zero. we can attempt the proof for the case of arbitrary pulse shape p(t). How many times can it be divided by π/ω0 ? Let the integer number be (T2 − T1 )ω0 . [T1 + (i − 1)π/ω0 . T1 + iπ/ω0 ]. the right sin is zero. Then. 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. M -ary QAM is defined as an arbitrary signal set a0 . φ1 (t) ω0 ω0 which for very large ω0 . Certainly. Otherwise. Within each of these cycles. 2. That is.ECE 5520 Fall 2009 77 1. In this case. we’re going to get a very small inner product. Thus φ0 (t). In this figure. aM −1 .0 . we use the ‘low-pass’ assumption that the maximum frequency content of p(t) is much lower than 2πω0 .1 ]T . because it is a sinusoid. when we integrate p2 (t) sin(2ω0 t) across one cycle. Zooming in on any few cycles. In this case. For example. [T1 + Lπ/ω0 . . we’re going to end up with approximately zero. L. That is. you can see that the pulse p2 (t) is largely constant across each cycle. . The pulse duration T2 − T1 is an integer number of periods. This assumption allows us to assume that p(t) is nearly constant over the course of one cycle of the carrier sinusoid. for i = 1. when we divide by numbers on the order of 106 or 109 . Thus. where each signal space vector ak is two-dimensional: ak = [ak. assume p2 (t) is nearly constant. . 24. the inner product is bounded on either side: − p2 (T1 + Lπ/ω0 ) p2 (T1 + Lπ/ω0 ) ≤ φ0 (t).2 Constellation With these two basis functions. is nearly zero. this part of the integral is zero here. Proof? Solution: How many cycles are there? Consider the period [T1 . − c2 c2 ≤ φ0 (t).15 in the Rice book (page 298). . . This is well-illustrated in Figure 5. φ1 (t) ≤ ω0 ω0 Typically. Finally. we see a pulse modulated by a sine wave at frequency 2πω0 . T2 ]. T2 ]. . ak. For engineering purposes. ω0 (T2 − T1 ) = πk for some integer k. . ω0 is a large number. frequencies 2πω0 could be in the MHz or GHz ranges. the numerator bounded above and below by +1 and -1. L= π Then each cycle is in the period. φ0 (t) and φ1 (t) are orthogonal. .

. .1 ]T for k = 0. but you should be able to read other books and know what they’re talking about.0 2p(t) cos(ω0 t) + ak.1 e−jω0 t s(t) = √ = 2p(t)R e−jω0 t (ak. These constellations use M = 2a for some even integer a.3 Signal Constellations M=64 QAM Figure 34: Square signal constellation for 64-QAM. you will see them write a QAM signal in shorthand as sCB (t) = p(t)(ak.1 2p(t) sin(ω0 t) √ 2p(t) [ak. M − 1 24. 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. • Figure 5.0 φ0 (t) + ak.18 shows examples of constellations which use M = 2a for some odd integer a.0 cos(ω0 t) + ak. ak.0 e−jω0 t + jak. . and arrange the points in a grid. One such diagram for M = 64 square QAM is also given here in Figure 34. we can see that the signal space representation ak is given by ak = [ak.0 + jak. and arrange the points in a grid. • See Figure 5. or use squares with the corners cut out.0 + jak. . These are either rectangular grids. Then.ECE 5520 Fall 2009 78 The signal corresponding to symbol k in (M -ary) QAM is thus s(t) = ak. .1 φ1 (t) √ √ = ak.1 ) This is called ‘Complex Baseband’.1 ) (39) In many textbooks.17 for examples of square QAM.0 .1 sin(ω0 t)] = Note that we could also write the signal s(t) as √ 2p(t)R ak.

e. wikipedia. 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. for equally probable symbols. and is shown in Figure 35(a). Thus BPSK is the same as bipolar (2-ary) PAM. various manufacturer-specific protocols. 24. i. binary phase-shift keying (BPSK) is the case of M = 2. Some examples are shown in Figure 35. G.Lite uses up to 128 carriers..DMT uses multicarrier (up to 256 carriers) methods (OFDM). and the Rice book: • Digital Microwave Relay. each with up to 28 = 256 QAM. ..0 .5 Average Energy in M-QAM 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.ECE 5520 Fall 2009 79 24. . What is the probability of error in BPSK? The same as in bipolar PAM. Note how QPSK is the same as M = 4 square QAM. • DSL. We’ll work in class some examples of finding Eb in different constellation diagrams. |a0 | = |a1 | = · · · = |aM −1 | The points ai for i = 0.768 QAM). it can send up to 215 QAM (32.e. . M − 1 are uniformly spaced on the unit circle.1 ak.7 Systems which use QAM See [Couch 2007]. and on each narrowband (4.0 k. 24. • Dial-up modems: use a M = 16 or M = 8 QAM constellation.6 Phase-Shift Keying Some implementations of QAM limit the constellation to include only signal space vectors with equal magnitude. Note that the rotation of the signal space diagram doesn’t matter. G. 6 GHz. . P [error] = Q N0 QPSK M = 4 PSK is also called quadrature phase shift keying (QPSK).3kHz) carrier. 2Eb . .4 Angle and Magnitude Representation a2 + a2 k.1 ak. so both ‘versions’ are identical in concept (although would be a slightly different implementation). and 11 GHz. BPSK For example. i.

25 25. you should realize that there are two main considerations when a constellation is designed for a particular M : 1. 2. uses up to 64 QAM.11g: Adaptive modulation method. The highest magnitude points (furthest from the origin) strongly (negatively) impact average bit energy. Points with equal distances separating them and their neighboring points tend to reduce probability of error. • 802.1 QAM Probability of Error Overview of Future Discussions on QAM Before we get into details about the probabilities of error. • Digital Video Broadcast (DVB): APSK used in ETSI standard.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. There are also some practical considerations that we will discuss 1. • Cable modems.11a and 802. Some constellations are more difficult (energy-consuming) to amplify at the transmitter. Upstream: 6 MHz bandwidth channel. Downstream: QPSK or 16 QAM. Some constellations have more complicated decision regions which require more computation to implement in a receiver. . 2. with 64 QAM or 256 QAM.

Also.4 Probability of Error in QPSK In QPSK. 25. the probability of error is analytically tractable. π minus the area in the sector within ± M of the correct angle φi . We needed a Q function to get a tail probability for a 1-D Gaussian pdf. The decisions are decoupled – x2 has no impact on the decision about bit one. Similarly. and x1 has no impact on the decision . This is a provable upper bound on the probability of error. For example. there is an exact expression for P [error] in an AWGN environment. The decision is made using only one dimension. You have already calculated the decision regions for each symbol. when Gray encoding is used.2 Options for Probability of Error Expressions In order of preference: 1. now consider the decision region for the first bit. Now we need not just a tail probability.. Exact formula. This is. P (symbol error) = 1 − r∈Ri 1 − e 2πσ 2 r−α i 2 2σ 2 (40) This integral is a double integral.ECE 5520 Fall 2009 81 25. Essentially. They require an integration of a N -D Gaussian pdf across an area. specifically x1 . This is a way to get a solution that is analytically easier to handle. Union bound. but more like a section probability which is the volume under some part of the N -dimensional Gaussian pdf. 2. 0 25. the second bit decision is made using only x2 . It can be used for “worst case” analysis which is often very useful for engineering design of systems. 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. and we don’t generally have any exact expression to use to express the result in general. the noise contribution to each element is independent. In a few cases. of the received signal vector x.3 Exact Error Analysis Exact probability of error formulas in N -dimensional modulations can be very difficult to find. E Typically these approximations are good at high Nb . Nearest Neighbor Approximation. consider M -ary PSK.. 3. This is because our decisions regions are more complex than one threshold test. It is not an approximation. Consider the QPSK constellation diagram.

. n P [symbol error] ≤ P [Ei ] i=1 (44) The union bound gives a conservative estimate. 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. the bandwidth of QPSK is identical to that of BPSK. This can be useful for quick initial study of a modulation type. If you know one inequality. let’s calculate the union bound on the probability of error. the bit error rate does not increase. 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. (This holds for any events E and F !) Then. know this one. E2 . we have that P [E ∪ F ] ≤ P [E] + P [F ] .5 Union Bound From 5510 or an equivalent class (or a Venn diagram) you may recall the probability formula. Furthermore. P [symbol error] = 1 − 1−Q 2Eb N0 2 (45) In contrast. . but in theory.ECE 5520 Fall 2009 82 on bit two. using the above formula. 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. when some other symbol not equal to i was actually sent. as shown in Figure 37(a). from (42) it is straightforward to show that for any list of sets E1 . Example: QPSK First. We define the two error events as: . En we have that n n (42) P i=1 Ei ≤ P [Ei ] i=1 (43) This is called the union bound. As we will show later.6 Application of Union Bound In this class. let’s study the union bound for QPSK. In this case. 25. 25. The event Ei may represent the event that we decide Hi . Assume s1 (t) is sent. It is useful when the overlaps Ei ∩ Ej are small but difficult to calculate. . our events are typically decision events. and it is very useful across communications. and the first axiom of probability.

• E0 . P [symbol error|H1 ] ≤ P [E2 ] + P [E0 ] These two probabilities are just the probability of error for a binary modulation. The Rice book will not tell you to do this! His strategy is to ignore redundancies. and in fact. In (b) a1 = −a3 = E2 a2 E4 a0 Figure 37: Union bound examples. Either way produces an upper bound. Is E2 ∪ E0 ∪ E3 = E2 ∪ E0 ? Why or why not? Because of the answer to the question (2. we use the union bound. and both are identical. because we are only looking for an upper bound any way. it is perfectly acceptable (and in fact a better bound) to remove redundancies and then apply the bound. P [symbol error|H1 ] = P [E2 ∪ E0 ] But don’t be confused. compared to the exact expression.). Then we write P [symbol error|H1 ] = P [E2 ∪ E0 ∪ E3 ] Questions: 1. • E2 . so P [symbol error|H1 ] ≤ 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. Only at very low Eb /N0 is there any noticeable difference! Example: 4-QAM with two amplitude levels √ This is shown (poorly) in Figure 37(b). the event that r falls on the wrong side of the decision boundary with a0 .ECE 5520 Fall 2009 83 a1 a1 E2 a2 (a) a3 E4 a0 (b) a3 √ Es . 3Es /2]T and a2 = −a0 = [ Es /2. both produce nearly identical numerical results. Then. the event that r falls on the wrong side of the decision boundary with a2 . However. Is this equation exact? 2. 0]T . (a) is QPSK with symbols of equal energy [0. The amplitudes of the top and bottom symbols are 3 times the .

6. For the twoamplitude 4-QAM modulation. given H1 ? Solution: Given symbol 1. the union bound on probability of symbol error is P [symbol error|H2 ] ≤ 3·2+2·2 Q 4 2Eb N0 How about average energy? For QPSK. What is the union bound on the probability of symbol error. these two distances between symbol a1 and a2 or a0 are the same: 2Es /N0 .) I am calling this ”2-amplitude 4-QAM” (I made it up). 2(0. the union bound. What is the union bound on the probability of symbol error. Thus the formula for the union bound is the same.1 General Formula for Union Bound-based Probability of Error This is given in Rice 6. overall. it is 2Eb N0 P [symbol error|H2 ] ≤ 3Q So. This means the actual P [symbol error] will be at most this value. Thus Eav = Es . (They are positioned to keep the distance between points in the signal space equal to 2Es /N0 . given H2 ? Solution: Now. Defining E2 and E0 as above. 25. It is probably less than this value. the symbol energies are all equal. What do we mean? If I draw any .76: P [symbol error] ≤ 1 M M −1 M −1 m=0 n=0 n=m P [decide Hn |Hm ] Note the ≤ sign. dB 5 6 Figure 38: For QPSK.5) + 2(1. This is a general formula and is not necessarily the best upper bound.5) = Es Eav = Es 4 Thus there is no advantage to the two-amplitude QAM modulation in terms of average energy. the probability is the same as above.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. amplitude of the symbols on the right and left. the exact probability of symbol error expression vs.

m=n m = 0. P [decide Hn |Hm ].n . . Proof of (46)? Lecture 14 Today: (1) QAM Probability of Error. We will talk about the concept of “neighboring” symbols to symbol i. and n = 0.1 QAM Probability of Error Nearest-Neighbor Approximate Probability of Error As it turns out. Section 2.6.n dm. since Es will be proportional to 2 A2 and dm. and do not need to be included. I prefer Es to make clear we are talking about Joules per symbol. In particular. I have drawn an upper bound. .ECE 5520 Fall 2009 85 function that is always above the actual P [symbol error] formula. M − 1.n Es  = Q  (46) P [decide Hn |Hm ] = Q  2N0 2Es N0 where dm. . But I could draw lots of functions that are upper bounds. M − 1. the factors A will cancel out of the expression. The Rice book uses Eb where I use Eb to denote average bit energy. for some of the error events.n . If we use a constant A when describing the signal space vectors (as we usually do). decide Hn |Hm may be redundant. .n means a higher argument in the Q-function. Given this.1) with the smallest dm. . These neighbors are the ones that are necessary to include in the union bound.   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. . This is because higher dm. is given by:     2 2 dm. . So approximately.n will be proportional to A2 . (2) Union Bound and Approximations Lecture 15 Today: (1) M-ary PSK and QAM Analysis 26 26. which in turn means a lower value of the Q-function. . the pairwise probability of error. and Nmin is the number of pairs of symbols which are separated by distance dmin . then. the probability of error is often well approximated by the terms in the Union Bound (Lecture 14. Note that the Rice book uses Eavg where I use Es to denote average symbol energy. . A little extra distance means a much lower value of the Q-function.n is the Euclidean distance between the symbols m and n in the signal space diagram. some higher than others.

5. . Calculate pair-wise distances between pairs of symbols which contribute to the decision boundaries. Calculate the union bound on probability of symbol error using the formula derived in the previous lecture. Convert distances to be in terms of Eb /N0 . using the answer to step (1. an error region which is completely contained in a larger error region. 3. 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. if they are not already. j) which are that distance apart. Es . • The Union bound.n  2N0  7. i)!). 2. It is not necessary to include redundant boundary information. These steps include: 1. You should calculate: • An exact expression if one should happen to be available.2 Summary and Examples We’ve been formulating in class a pattern or step-by-step process for finding the probability of bit or symbol error in arbitrary M-ary PSK and QAM modulations. Convert probability of symbol error into probability of bit error if Gray encoding can be assumed. Draw decision boundaries. • The nearest-neighbor approximation. Calculate the average energy per bit. that is. using Eb = Es / log2 M . Eb . 8.j = dmin (don’t forget to count both (i. 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. di. j) and (j. Calculate the average energy per symbol. and Nmin is the number of ordered pairs (i. 6. 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. as a function of any amplitude parameters (typically we’ve been using A). 4.ECE 5520 Fall 2009 86 26.).

.ECE 5520 Fall 2009 87 Figure 39: Signal space diagram for some example (made up) constellation diagrams.

. fM −1 }. but we don’t want to waste spectrum.1 Orthogonal Frequencies We will want our cosine wave at these M different frequencies to be orthonormal. i. probability of symbol error. first. Frequency modulation in general changes the center frequency over time: Now. (2) FSK 27 Frequency Shift Keying x(t) = cos(2πf (t)t). in frequency shift keying. which for example. they are orthogonal if 2π(k − m)∆f Ts is a multiple of π. 27. ∆f = n fsy 1 =n 2Ts 2 . . symbols are selected to be sinusoids with frequency selected among a set of M different frequencies {f0 . f1 . o. (They are also approximately orthogonal if ∆f is realy big.e.w. The plots are in terms of amplitude A. could be a rectangular pulse: √ 1/ Ts . 0 ≤ t ≤ Ts p(t) = 0. . 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. . and thus √ (48) φk (t) = 2p(t) cos(2πfc t + 2πk∆f t) where p(t) is a pulse shape.) For general k = m. but all probability of error expressions should be written in terms of Eb /N0 .ECE 5520 Fall 2009 88 on. Lecture 16 Today: (1) QAM Examples (Lecture 15) . • Does this function have unit energy? • Are these functions orthogonal? Solution: φk (t). and then also probability of bit error. this requires that ∆f Ts = n/2. Consider fk = fc + k∆f .

0] . . . . Def ’n: Voltage Controlled Oscillator (VCO) A sinusoidal generator with frequency that linearly proportional to an input voltage. . . SRRC pulses. no they’re not. FSK can be seen as a switch between M different carrier signals. . bandwidth will be lower when we send less steep transitions. A. as seen in Figure 42. . . For M = 2 and M = 3 these vectors are plotted in Figure 40. . aM −1 = [0. 0. . . for example. . Signal space vectors ai are given by a0 = [A.) 1 Thus we need to plug into (48) for ∆f = n 2Ts for some integer n in order to have an orthonormal basis. 27. Signal f1(t) Signal f2(t) Switch FSK Out Figure 41: Block diagram of a binary FSK transmitter. as shown in Figure 41. Note that we don’t need to sent square wave input into the VCO. M=2 FSK M=3 FSK Figure 40: Signal space diagram for M = 2 and M = 3 FSK modulation. . 0. . What n? We will see that we either use an n of 1 or 2. But usually it is generated from a single VCO. A] √ What is the energy per symbol? Show that this means that A = Es . .ECE 5520 Fall 2009 89 for integer n.2 Transmission of FSK At the transmitter. (Otherwise. In fact. 0] a1 = [0.

This is more difficult than it sounds. Also. In non-coherent reception. though – we have a fundamental problem. there are M possible carrier frequencies. If we only correlate it with one (the sine wave. the carrier cos(2πfk t + θk ) is sent with the transmitted signal. 1/M of the time (assuming equally-probable symbols). there are two orthogonal functions.4 Coherent Reception FSK reception can be done via a correlation receiver. . . one for each symbol frequency: cos(2πfc t + θ0 ) cos(2πfc t + 2π∆f t + θ1 ) . so we’d need to know and be synchronized to M different phases θi . which are beyond the scope of this course.5 Non-coherent Reception Notice that in Figure 40. to aid in demodulation (at the expense of the additional energy). respectively. ˆ Each phase θk is estimated to be θk by a separate phase-locked loop (PLL).3 Reception of FSK FSK reception is either phase coherent or phase non-coherent. the phase of the output signal φk (t) does not change instantaneously at symbol boundaries iTs for integer i. we don’t know whether or not the energy at frequency fk correlates highly with the cosine wave or with the sine wave. As M gets high. Since we will not know the phase of the received signal. 27. lets call them xI using the capital I to denote in-phase and k xQ with Q denoting quadrature. This will give us two inner products. 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. 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 . cosine and sine (see QAM and PSK). These M PLLs must operate even though they can only synchronize when their symbol is sent. coherent detection becomes difficult. As we know. In one flavor of CPFSK (called Sunde’s FSK). the sign or phase of the sinusoid is not very important – only one symbol exists in each dimension. we just measure the energy in each frequency.ECE 5520 Fall 2009 90 Def ’n: Continuous Phase Frequency Shift Keying FSK with no phase discontinuity between symbols. k . There are a variety of flavors of CPFSK. for every frequency. VCO Frequency Signal f(t) FSK Out Figure 42: Block diagram of a binary FSK transmitter. 27. cos(2πfc t + 2π(M − 1)∆f t + θM −1 ) 27. just as we’ve seen for previous modulation types. for example). In other words. Here. This is the only case where I’ve heard of using coherent FSK reception.

Compare this to the coherent FSK receiver in Figure 44. 1. and calculating the squared length k k of the vector [xI . . . that is. Q Efk = (xI )2 + (xk )2 k is calculated for each 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 non-coherent 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. fk . but the optimal detector (Maximum likelihood detector) is then to decide upon the frequency with the maximum energy Efk . 27. xQ ]T .6 Receiver Block Diagrams The block diagram of the the non-coherent FSK receiver is shown in Figures 45 and 46 (copied from the Proakis & Salehi book). Figure 44: (Proakis & Salehi) Phase-coherent demodulation of M -ary FSK signals. Is this a threshold test? We would need new analysis of the FSK non-coherent detector to find its analytical probability of error. . k k The energy at frequency fk . . . k = 0. You could prove this. M −1.

ECE 5520 Fall 2009 92 Figure 45: (Proakis & Salehi) Demodulation of M -ary FSK signals for non-coherent detection. Figure 46: (Proakis & Salehi) Demodulation and square-law detection of binary FSK signals. .

1. f2 . Hi )fθm |Hi (φ|Hi )dφ (49) Note that fr|θm. which is assumed to be uniform between 0 and 2π.5 dB more energy per bit). but the symbols are spaced differently (more closely) as a function of Eb . but 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 f1 . to d0.d. 1. What do they do to prove this in Proakis & Salehi? They prove it for binary non-coherent FSK.θm|Hi (r.H0 (r|φ. so rm is termed the envelope.5 dB better than OOK (requires 1. φ|Hi )dφ fr|θm.ECE 5520 Fall 2009 93 27. 2 Question: What will rm equal when the noise is very small? As it turns out. and independent of the noise. binary FSK is about 1. What is the detection threshold line separating the two decision regions? 2. 27. Note that this depends on θm . sM . H0 ) is a 2-D Gaussian random vector with i. given the non-coherent receiver and rmc and rms .5 dB less energy per bit).Hi (r|φ. . let’s look at coherent detection of binary FSK. .5 dB worse than bipolar PAM (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. Now. We had that   d2 0. components. 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. 2.1  P [error]2−ary = Q  2N0 √ 2/2 compared to bipolar PAM. the envelope rm is an optimum (sufficient) statistic to use to decide between s1 . They formulate the prior probabilities fr|Hi (r|Hi ). .7 Probability of Error for Coherent Binary FSK First. 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.1 = Eb N0 √ 2Eb .i. 2π fr|Hi (r|Hi ) = = 0 2π 0 fr.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. It takes quite a bit of 5510 to do this proof.

section 7. 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. The proof is in Proakis & Salehi.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. and is presented since it does not appear in the Rice book. Proof Summary: Our non-coherent receiver finds the energy in each frequency.9. 5. 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 non-coherent) are important. 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”. Lecture 17 Today: (1) FSK Error Probabilities (2) OFDM 27. They formulate the joint probabilities fr∩H0 (r ∩ H0 ) and fr∩H1 (r ∩ H1 ). and it often is. The decisions in this last step. as well.1 FSK Error Probabilities.9.9 27. after manipulation of the pdfs.6. and you should make note of them. the derivation in the Proakis & Salehi book. Solution: 1 − 1 Eb P [symbol error] = P [bit error] = e 2 N0 2 M/2 P [symbol error] M −1 . You will use them to be able to design communication systems that use FSK. The expression for probability of error in binary non-coherent FSK is given by. They instead are either Ricean (for the transmitted frequency) or Rayleigh distributed (for the “other” M − 1 frequencies). Example: Probability of Error for Non-coherent M = 2 case Use the above expressions to find the P [symbol error] and P [error] for binary non-coherent FSK. page 430. it decides H1 . Section 7. which is posted on WebCT. Otherwise.6. Where the joint probability fr∩H0 (r ∩ H0 ) is greater than fr|H1 (r|H1 ). 94 4. An exact expression for the probability of error can be derived.ECE 5520 Fall 2009 3. The above proof is simply FYI. These energy values no longer have a Gaussian distribution (due to the squaring of the amplitudes in the energy calculation). the receiver decides H0 . Part 2 M -ary Non-Coherent FSK For M -ary non-coherent FSK.

27. so the sum of the energy is constant.6dB. For each band. because phone lines were not designed for higher frequency signals. it tells us that the approximate bandwidth is. 10 log 10 |H(f )|2 . In HW 7. B = (1 + α)/Ts . 28. so you don’t lose anything by this division. dB 10 12 0 Figure 47: Probability of bit error for non-coherent reception of M-ary FSK. your transmission energy is the same. 28 Frequency Multiplexing Frequency multiplexing is the division of the total bandwidth (call it BT ) into many smaller frequency bands.9.10 Bandwidth of FSK Carson’s rule is used to calculate the bandwidth of FM signals. But. you show that the total bit rate is the same in the multiplexed system. you now have BT /K bandwidth on each subchannel. we might send a PSK or PAM or QAM signal on that band. where H(f ) is an . Remember this for later . B = 1/Ts . the probability of error E goes to zero. Specifically. For the null-to-null bandwidth of raisedcosine pulse shaping. For example. 27. and sending a lower bit rate on each one. frequency dependent gains are also experienced in DSL. Note that in the multiplexed system. longer by a factor of K so that the bandwidth of that subchannel has narrower bandwidth.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. consider Figure 48. For M -ary FSK. The energy would be divided by K in each subchannel. which shows an example frequency selective fading pattern.it is related to the Shannon-Hartley 0 theorem on the limit of reliable communication. for the case when Nb > −1. Furthermore. BT = (M − 1)∆f + 2B where B is the one-sided bandwidth of the digital baseband signal. Note for square wave pulses.2 Summary The key insight is that probability of error decreases for increasing M .1 Frequency Selective Fading Frequency selectivity is primarily a problem in wireless channels. divide your BT bandwidth into K subchannels. each subchannel has a symbol period of Ts K. Our choice is arbitrary. As M → ∞.

or • Frequency-hopping spread spectrum (FH-SS). there are K parallel bitstreams. • Direct-sequence spread spectrum (DS-SS). has a very low SNR. which introduces ISI.ECE 5520 Fall 2009 BT/K 96 Severely faded channel Figure 48: An example frequency selective fading pattern. The H(f ) acts as a filter. example wireless channel. If the channel loss is too great. Wideband: The bandwidth is used for one channel.2 Benefits of Frequency Multiplexing Now.. where R is the total bit rate. The whole bandwidth is divided into K subchannels. We’re going to mention frequency multiplexing.e. Narrowband: Part of the bandwidth is used for one narrow channel. When designing for frequency selective fading. and each subchannel’s channel filter is mostly constant. In frequency multiplexing. designers may use a wideband modulation method which is robust to frequency selective fading. • Frequency multiplexing methods (including OFDM). For example.5 (the . • For stationary transmitter and receiver and stationary f . Problems: 1. 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. bit errors are not an ‘all or nothing’ game. while most experience little attenuation. 28. the channel stays constant over time. As a first order approximation. a subchannel either experiences a high SNR and makes no errors. Some channels experience gain. 2. (The f is relative. or is in a severe fade. take the actual frequency f = f + fc for some center frequency fc . than the error rate will be too high.) • You can see that some channels experience severe attenuation. at a lower bit rate. This H(f ) might be experienced in an average outdoor channel. ˜ i. then the loss 10 log10 |H(f ′ )|2 will not change throughout the communication. If you put down the transmitter and receiver and have them operate at a particular f ′ . and experiences a BER of 0. each of rate R/K.

OFDM is the combination: φ0.Q (t))ej2πk∆f t Ak (t)ej2πk∆f t k=1 2/Ts R (51) where Ak (t) = ak. . than you can see why it is possible to use an IFFT and FFT to generate the complex baseband signal.w.Q (t) = 0.Q (t) = φ1. In FSK. which has probability β that it will have a 0. 0 ≤ t ≤ Ts o. 0. 2/Ts sin(2πfc t). o.Q (t). 2/Ts cos(2πfc t + 2π(M − 1)∆f t). Compare this to a single narrowband channel. Does this look like an inverse discrete fourier transform? If yes.I (t) = φ0.I (t) = φM −1. 2/Ts cos(2πfc t). 0 ≤ t ≤ Ts o. The .5 probability of error.w. .I (t) + jak. 2/Ts cos(2πfc t + 2π∆f t). a wideband system with very high ISI might be completely unable to demodulate the received signal.Q (t) = . 0 ≤ t ≤ Ts 0.w.I (t) cos(2πfk t) + ak. This avoids having K independent transmitter chains and receiver chains. FFT implementation: There is a particular implementation of the transmitter and receiver that use FFT/IFFT operations. 0 ≤ t ≤ Ts 0. 1. the overall probability of error will be 0.w.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. φM −1. These are all orthogonal functions! We can transmit much more information than possible in M -ary FSK.ECE 5520 Fall 2009 97 worst bit error rate!). o. 28. In QAM.3 OFDM as an extension of FSK This is section 5. 1 where ∆f = 2Ts . 0. 2/Ts sin(2πfc t + 2π∆f t).w. (Note we have 2M basis functions here!) The signal on subchannel k might be represented as: xk (t) = 2/Ts [ak. 0 ≤ t ≤ Ts o.I (t) = φ1.I (t) + jak.5 in the Rice book. If β is the probability that a subchannel experiences a severe fade. 0 ≤ t ≤ Ts 0. we use a single basis function at each of different frequencies. 2/Ts sin(2πfc t + 2π(M − 1)∆f t).w. Frequency multiplexing is typically combined with channel coding designed to correct a small percentage of bit errors. we use two basis functions at the same frequency. o. Similarly.5β.

11a is 250k/sec. 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. the signal is like K-ary FSK. But. 3 subchannels of 4-ary PAM Figure 49: Signal space diagram for K = 3 subchannel OFDM with 4-PAM on each channel. . so effectively 48 subcarriers are used for data. 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. Each data subcarrier can be modulated in different ways. Example: 802. 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. Since the K carriers are orthogonal.11a IEEE 802. Four of the subcarriers are reserved for pilot tones. The symbol rate in 802. OFDM.ECE 5520 Fall 2009 98 FFT implementation (and the speed and ease of implementation of the FFT in hardware) is why OFDM is popular. 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.11a uses OFDM with 52 subcarriers.

0. we find bn by comparing the phase of xn to the phase of xn−1 . consider the bit sequence {bn }. k8 ]T will be sent? Solution: k = [1. 0. where bn is the nth bit that we want to send. kn = kn−1 . for differential BPSK. in differential encoding. How do we decide which phase to send? Prior to this. 0. we also include kn−1 . The sequence bn is a sequence of 0’s and 1’s. 1. π. decide bn = 1. What symbols k = [k0 . 1.1. we’ve said. this means transmitting a training sequence. is to measure the statistic ∗ ℜ{xn xn−1 } = |xn ||xn−1 | cos(∠xn − ∠xn−1 ) as the statistic – if it less than zero. 1. send a0 if bn = 0. while staying at the same angle indicates a bit 0. π. 0. 0. . we use differential encoding (at the transmitter) and decoding (at the receiver). 0]. 1. For non-coherent reception of PSK. 1. 0. Example: Differential encoding Let b = [1. Instead of setting k for ak only as a function of bn . 0. 29. 1. What our receiver does.ECE 5520 Fall 2009 99 For coherent reception of PSK.1. Typically. . ˆn = [1. show that the receiver will decode the original bitstream with differential decoding. 1. 0. Example: Differential decoding 1. 0. 1. a switch in the angle of the signal space vector from 0o to 180o or vice versa indicates a bit 1. 1]T These values of kn correspond to a symbol stream with phases: ∠ak = [π. π. 1. Basically. 0. if kn−1 = 0 then 1 − kn−1 = 1. 1. at the receiver. 0. and send a1 if bn = 1. Solution: Assuming φi0 = 0. decide bn = 0. 0. Assume b0 = 0. 0]T . Now. Note that we have to just agree on the “zero” phase. π. b . 1. . Assuming no phase shift in the above encoding example. bn = 1 Note that 1 − kn−1 is the complement or negation of kn−1 – if kn−1 = 1 then 1 − kn−1 = 0.2 DPSK Receiver Now. and if it is greater than zero. . 1. Typically k0 = 0. will always need some kind of phase synchronization in BPSK. bn = 0 1 − kn−1 . π]T 29.1 DPSK Transmitter Now.

0. 0. • Bandwidth efficiency Eb N0 . π. 0. 1. 1.1. 0. 1.ECE 5520 Fall 2009 100 2. Both are plotted in Figure 50. What will be decoded at the receiver? Solution: ˆn = [0. π. Now.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. 29. 0].2 Points for Comparison • Linear amplifiers (transmitter complexity) • Receiver complexity • Fidelity (P [error]) vs. assume that all bits are shifted π radians and we receive ∠x′ = [0. DPSK requires about 1 dB more of bit error Q 2Eb N0 than BPSK. π. 0. 0]. 0. 0. dB 10 12 14 0 Figure 50: Comparison of probability of bit error for BPSK and Differential BPSK. 10 0 Probability of Bit Error 10 −2 10 −4 10 −6 BPSK DBPSK 2 4 6 8 Eb/N0. 1. b Rotating all symbols by π radians only causes one bit error. which has probability . 29.

e. is the ratio of bits per second to bandwidth: η = Rb /BT Bandwidth efficiency depends on the definition of “bandwidth”.67 4. Since it is usually used for comparative purposes.0 α = 0. we can scale a system. 29. or BT = where Rb = 1/Tb is the bit rate. to increase the bit rate by decreasing the symbol period. and correspondingly increase the bandwidth. the bandwidth is largely determined by the pulse shape.3. BT = (M − 1)∆f + 2B . i.5 1.0 101 Table 2: Bandwidth efficiency of PSK and QAM modulation methods using raised cosine filtering as a function of α. Def ’n: Bandwidth efficiency The bandwidth efficiency. We’ve also talked about Hertz. Bandwidth efficiency is then η = Rb /BT = See Table 2 for some numerical examples.3. we just make sure we use the same definition of bandwidth throughout a comparison..0 2.5 2.. 29. i.67 1.5 0.0 α=1 0. The transmission bandwidth (for a bandpass signal) is BT = 1+α Ts Since Ts is seconds per symbol. the null-null 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. For root raised cosine filtering. This relationship is typically linearly proportional.33 2. typically denoted η.2 FSK (1 + α)Rb log2 M log2 M 1+α We’ve said that the bandwidth of FSK is. Typically. The key figure of merit: bits per second / Hertz.3 Bandwidth Efficiency We’ve talked about measuring data rate in bits per second. bps/Hz.0 3. the quantity of spectrum our signal will use.1 PSK. 29.ECE 5520 Fall 2009 η BPSK QPSK 16-QAM 64-QAM α=0 1.0 6.e. PAM and QAM In these three modulation methods.0 4.

Eb N0 Eb N0 : Main modulations which we have evaluated probability of error vs. 1. including Binary PAM or BPSK. including QPSK. M-ary PSK.5 to be about 14 dB. DPSK. Eb N0 For each modulation format. FSK. Nb for M = 8 PSK 0 E What is the required Nb for 8-PSK to achieve a probability of bit error of 10−6 ? What is the bandwidth 0 efficiency of 8-PSK when using 50% excess bandwidth? Solution: Given in Rice Figure 6. OOK. log2 M (M − 1)∆f Ts + (1 + α) log2 M M +α η= 29. So. 2B = (1 + α)/Ts . bandwidth efficiency) pairs. M-ary 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. For the null-to-null bandwidth of raisedcosine pulse shaping. we have quantities of interest: • Bandwidth efficiency. M-ary PAM. 2.3.6.ECE 5520 Fall 2009 102 where B is the one-sided bandwidth of the digital baseband signal. 3.4 Bandwidth Efficiency vs. See also Rice pages 325-331. Square QAM 4. . and 2. 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 non-coherent reception). E We can plot these (required Nb . Non-square QAM constellations 5. and E • Energy per bit ( Nb ) requirements to achieve a given probability of error. 0 29.3. 0 E Example: Bandwidth efficiency vs.5 Fidelity (P [error]) vs. See Rice Figure 6.

and Pout is the output signal power. Output signal is a scaled up version of the input signal. • Class B: linear amplifiers turn on for half of a cycle (conduction angle of 180o ) with maximum power efficiency of 78. Power is dissipated at all times. and as a result of their configuration.6. Amplifiers are separated into ‘classes’ which describe their configuration (circuits).ECE 5520 Fall 2009 Name BPSK OOK DPSK M-PAM QPSK M-PSK Square M-QAM 2-non-co-FSK M-non-co-FSK 2-co-FSK M-co-FSK = 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. their linearity and power efficiency.6 Transmitter Complexity What makes a transmitter more complicated? • Need for a linear power amplifier • Higher clock speed • Higher transmit powers • Directional antennas 29. • Class AB: Like class B. Two in parallel are used to amplify a sinusoidal signal. • Class A: linear amplifiers with maximum power efficiency of 50%. If PDC is the input power (e. 29. the power efficiency is rated as ηP = Pout /PDC . but each amplifier stays on slightly longer to reduce the “dead zone” at zero. the conduction angle is slightly higher than 180o .g.5%. .1 Linear / Non-linear Amplifiers An amplifier uses DC power to take an input signal and increase its amplitude at the output.. from battery) to the amplifier. That is. one for the positive part and one for the negative part.

Can only amplify signals with a nearly constant envelope. For offset QPSK (OQPSK). 29. . The new transmitted signal takes the same bandwidth and average power. which precludes use of a class C amplifier. you’ll see a circle. and has the same Eb /N0 vs. Class C non-linear amplifiers have power efficiencies around 90%. See Figure 52(b) and Figure 51 (a) to see this graphically. 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. if you look at the constellation diagram. For QPSK. but then is band-pass filtered or tuned to the center frequency to force out spurious frequencies. 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. the signal s(t) will go through zero. This means that. battery-powered transmitters are often willing to use Class C amplifiers. 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.ECE 5520 Fall 2009 104 • Class C : A class C amplifier is closer to a switch than an amplifier. . However.7 Offset QPSK QPSK without offset: Q-channel Offset QPSK: Q-channel I-channel I-channel (a) (b) Figure 51: Constellation Diagram for (a) QPSK and (b) O-QPSK. They can do this if their output signal has constant envelope. we delay the quadrature Ts /2 with respect to the in-phase. See Figure 52 for a comparison of QPSK and OQPSK. It is in a discrete set. The signal must never go through the origin (envelope of zero) or even near the origin. This generates high distortion. probability of bit error performance. . the envelope |s(t)| is largely constant. π 3π 5π 7π ∠a(t) ∈ { . } 4 4 4 4 and p(t) is the pulse shape. In order to double the power efficiency. we just need to delay the sampling on the quadrature half of a sample period with respect to the in-phase signal.

1 105 0.ECE 5520 Fall 2009 0.1 Imag 0 −0. High fidelity (low bit error rate) 3.1 0.8 Receiver Complexity What makes a receiver more complicated? • Synchronization (carrier.1 0.1 0 Real 0.1 0.05 0 −0.2 −0.1 0 1 2 Time t 3 4 Imag 0. High data rate 2.15 0.05 −0.2 0.1 0.2 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 1 2 Time t 3 4 Imag 0.1 −0.1 0 1 2 Time t 3 4 (b) −0.2 0. your mission (if you choose to accept it) is to achieve: 1.1 Real 0 −0.1 (c) 0 1 2 Time t 3 4 (d) −0.1 0 Real 0. 29.2 Real 0 −0.1 Imag 0 −0.15 0.15 −0.05 0 −0.2 Figure 52: Matlab simulation of (a-b) QPSK and (c-d) O-QPSK.15 −0. phase.1 (a) 0. compared to (b) that for QPSK. showing the (d) largely constant envelope of OQPSK.05 −0. Low transmit power .

and it cuts across ECE classes. optics. In this lecture.g. For example.. radio propagation.ECE 5520 Fall 2009 106 4. given the bandwidth limits. . Figure 53: Relationships between important system design variables (rectangular boxes). the operator is drawn in. for example. C/N0 is shown to have a divide by relationship with C and N0 . The effect of the choice of modulation impacts several functional relationships. which is drawn as a dotted line. I’d be able to determine the probability of error for several different modulation types..1 Link Budgets Given C/N0 The received power is denoted C. in particular circuits. otherwise it is left as an unnamed function f (). and to what they are related. it has units of Watts. we discuss this procedure. if I was given the bandwidth limits and C/N0 ratio. the received signal power and noise power. What is C/N0 ? It is received power divided by noise energy. because you can’t have everything at the same time. Typically some subset of requirements are given. but it summarizes what we need to know about the signal and the noise for the purposes of system design. Low transmitter/receiver complexity But this is truly a mission impossible. So the system design depends on what the desired system really needs and what are the acceptable tradeoffs. Low bandwidth 5. E e. but in the Rice book it is typically denoted C.g. 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. Functional relationships between variables are given as circles – if the relationship is a single operator (e. and bit error rate limit. 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). It is an odd quantity. This lecture is about system design. 0 30. • We often describe both the received power at a receiver PR . For example. antennas. what data rate can be achieved? Using which modulation? To answer this question. See Figure 53. and define each of the variables we see in Figure 53. the relationship between probability of bit error and Nb . division).

if it takes 5 seconds to transfer a 1MB file.8082 8 × 10−5 = 3.8461 7 × 10−5 = 3. and bandwidth.7478 Q−1 4 × 10−3 = 2.6708 2 × 10−6 = 4.25335 Q−1 5 × 10−1 = 0 .1559 Q−1 9 × 10−4 = 3.5 × 10−1 = 1. C/N0 = Hz.775 9 × 10−5 = 3. Given C/N0 .1075 3 × 10−5 = 4.1947 Q−1 8 × 10−4 = 3. 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× = 4.6153 −1 Q 2 × 10−4 = 3.4652 5 × 10−6 = 4. it’s really not a big deal to pull it off of a chart or table. In other words.2816 Q−1 1.6449 Q−1 6 × 10−2 = 1.4051 Q−1 9 × 10−2 = 1.3528 Q−1 5 × 10−4 = 3.9444 5 × 10−5 = 3. For your convenience.8906 6 × 10−5 = 3.1214 Q−1 1 × 10−3 = 3. The noise energy is N0 .5401 Q−1 3 × 10−4 = 3. Commonly.0537 Q−1 3 × 10−2 = 1.5264 4 × 10−6 = 4. then software often reports that the data rate is 1/5 = 0.4573 Q−1 8 × 10−3 = 2. great. 1/s.3145 9 × 10−6 = 4.3776 7 × 10−6 = 4.7534 1. CS people use Bps (kBps or MBps) when describing data rate. While you have Matlab. For example.5 × 10−2 = 2. We can write Eb = CTb . we often denote: C C/N0 Eb = Tb = N0 N0 Rb where Rb = 1/Tb is the bit rate. E Note: We typically use Q (·) and Q−1 (·) to relate BER and Nb in each direction. the following tables/plots of Q−1 (x) will appear on Exam 2.2884 1 × 10−5 = 4. bit rate.1701 Q−1 2 × 10−2 = 2.5758 Q−1 6 × 10−3 = 2.8782 Q−1 3 × 10−3 = 2. I am not picky about getting lots of correct decimal places. modulation.2 MBps or 200 kBps. we can now relate bit error rate.5 × 10−3 = 2.7507 Q−1 5 × 10−2 = 1.2905 Q−1 6 × 10−4 = 3.5548 Q−1 7 × 10−2 = 1.2649 1. since energy is power times time.6114 3 × 10−6 = 4.5121 Q−1 7 × 10−3 = 2.0364 Q−1 2 × 10−1 = 0. To separate the effect of Tb .719 −1 Q 1.3656 −1 FUNCTION: Q−1 1 × 10−2 = 2. If you can program it into your calculator.0902 Q−1 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 × 10−5 = 4.4316 Q−1 4 × 10−4 = 3.2389 Q−1 7 × 10−4 = 3.5244 Q−1 4 × 10−1 = 0.9677 Q−1 2 × 10−3 = 2.3263 Q−1 1.4758 Q−1 8 × 10−2 = 1. • Note that people often report C/N0 in dB Hz.6521 Q−1 5 × 10−3 = 2.ECE 5520 Fall 2009 107 E • We know the probability of bit error is typically written as a function of Nb .3408 Q−1 1 × 10−1 = 1.4089 Q−1 9 × 10−3 = 2. Otherwise.8808 Q−1 4 × 10−2 = 1.5 × 10−6 = 4.84162 Q−1 3 × 10−1 = 0.7455 10−6 TABLE OF THE Q−1 (·) Q 1 × 10−4 = 3. But the bit rate is 8/5 Mbps or 1.1735 2 × 10−5 = 4.5 × 10−4 = 3.4172 6 × 10−6 = 4.3439 8 × 10−6 = 4. The 0 bit energy is Eb .6 × 106 bps.0128 4 × 10−5 = 3. this 0 is easy to calculate.

2 Power and Energy Limited Channels Assume the C/N0 . Given the C/N0 constraint and the Eb N0 Eb N0 constraint. the link (or channel) is called a bandwidth limited channel. and your Rb is achievable. find the maximum bit rate.5 0 −7 10 10 −6 10 −5 10 −4 10 Value.5 3 2.5 1 0.5 2 1. x −3 10 −2 10 −1 10 0 30.5 Inverse Q function. Such links (or channels) are called power limited channels. Rb log2 M and then compute the required 4. your system is power limited. Otherwise. Q−1(x) 4 3. Use the probability of error constraint to determine the of error formula for the modulation. You just need to try to solve the problem and see which one limits your system. and the maximum BER are all given. 3. Sometimes bandwidth is the limiting factor in determining the maximum achievable bit rate.ECE 5520 Fall 2009 108 5 4. then the system is bandwidth limited. reduce your bit rate to conform to the BW constraint. 2. In this case. calculate the maximum symbol rate Rs = bandwidth using the appropriate bandwidth formula. Note that Rb = 1/Tb = . Method B: Start with a bandwidth-limited assumption: . Sometimes power is the limiting factor in determining the maximum achievable bit rate. given the appropriate probability C/N0 Eb N0 constraint. the maximum bandwidth. but be sure to express both in linear units. Given a maximum bit rate. Compare the bandwidth at maximum Rs to the bandwidth constraint: If BW at Rs is too high. Here is a step-by-step version of what you might need do in this case: Method A: Start with power-limited assumption: 1.

If the modulation is square 16-QAM using the SRRC pulse shape with α = 0. what is the maximum achievable bit rate on the link? Is this a power limited or bandwidth limited channel? 2.) . Use the bandwidth constraint and the appropriate bandwidth formula to find the maximum symbol rate Rs and then the maximum bit rate Rb . Rb = C/N0 Eb N0 = 1. the system is power limited. Otherwise. 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. So.) Eb N0 at the given bit rate by computing Eb N0 . Find the units. what is the maximum achievable bit rate on this link? Is this a power limited or bandwidth limited channel? Solution: 1. Eb N0 = C/N0 Rb . If the computed P [error] is greater than the BER constraint. Try Method A. Example: Rice 6.27×106 /4 = 5.27×106 = 2.33 Consider a bandpass communications link with a bandwidth of 1. The required bandwidth for this system is (1 + α)Rb BT = = 1. we would have needed to reduce Rb to meet the bandwidth limit.5. 2. 4. If the modulation is 16-PSK using the SRRC pulse shape with α = 0. Solving for Rb . 1. C/N0 = 1082/10 = 1.5 MHz and with an available C/N0 = 82 dB Hz.585 × 108 .585 × 108 = 2.27 Mbits/s.84 and thus Rs = Rb / log2 M = 2.5(2. (Again. Find the probability of error at that using the appropriate probability of error formula.5 MHz.84 (52) Converting C/N0 to linear. then your system is power limited.67×105 Msymbols/s. your system is bandwidth-limited.5.27×106 )/4 = 851 kHz log2 M This is clearly lower than the maximum bandwidth of 1. and can operate with bit rate 2.ECE 5520 Fall 2009 109 1. The maximum bit error rate is 10−6 . and you have found the correct maximum bit rate.27 Mbits/s 69. (If BT had come out > 1. make sure that everything is in linear 3. Use the previous method to find the maximum bit rate.5 MHz.

In free space. • λ is the wavelength at signal frequency. so we must reduce the bit rate to Rb = BT log2 M 4 = 1. Note that λ = c/fc where c = 3 × 108 meters per second is the speed of light. so we just use the center frequency fc . But. PT GT GR λ2 (4πR)2 . the received power is calculated from the Friis formula. this section of the class provides two examples.5 In summary.ECE 5520 Fall 2009 Eb N0 110 for the maximum BER: 3 log2 M Eb M − 1 N0 2.1 Free Space ‘Free space’ is the idealization in which nothing exists except for the transmitter and receiver. 30.55 The required bandwidth for this bit rate is: BT = (1 + α)Rb = 1. the wavelength is nearly constant across the bandwidth.5 MHz = 4 MHz 1+α 1. but there are others.55 (53) Rb = C/N0 Eb N0 = 1. Wireless propagation models are required.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.75 Mbits/s 27. 30.3. we can find 10 −6 10−6 Eb N0 Eb N0 Solving for Rb . For M = 16 (square) QAM. respectively. and can really only be used for deep space communications. this formula serves as a starting point for other radio propagation formulas. = √ 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 = 27.585 × 108 = 5.75×106 )/4 = 2.5 MHz. C = PR = where • GT and GR are the antenna gains at the transmitter and receiver.75×106 = 5. For narrowband signals. Try Method A.5(5.16 MHz log2 M This is greater than the maximum bandwidth of 1. we have a bandwidth-limited system with a bit rate of 4 MHz.

type of environment. is linearly proportional to the log of the distance R. given in dB. For example. (This name that Rice chose is unfortunate because if it was positive. but it is typically very negative. is λ [C]dBm = [GT ]dB + [GR ]dB + [PT ]dBm + 20 log 10 − 20 log 10 R 4π This says the received power.3 Wired Channels Typically wired channels are lossy as well. instead. etc. and L1m is the loss per meter. but the loss is modeled as linear in the length of the cable. 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. etc. 111 Here. Effectively. we showed that the Friis formula. 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”. My opinion is.) There are also typically other losses in a transmitter / receiver system. a very short summary is that radio propagation on Earth is different than the Friis formula suggests.3. because of shadowing caused by buildings.. C. antenna heights. But. other imperfections.ECE 5520 Fall 2009 • PT is the transmit power. 30. everything is in linear terms.2 Non-free-space Channels We don’t study radio propagation path loss formulas in this course. it would be a “gain”. Typically people use decibels to express these numbers. etc. it may be more like 1/Rn . For example. [C]dBm = [PT ]dBm − R [L1m ]dB where PT is the transmit power and R is the cable length in meters. . it should be called “channel gain” and have a negative gain value. [Lp ]dB = L0 − 10n log10 R.3. 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. trees. for some constants n and L0 . the average loss may increase more quickly than 1/R2 . Lots of other formulas exist that approximate the received power as a function of distance. whenever path loss is linear with the log of distance. losses in a cable.

Switching to finding an expression for C.76 × 10−18 J. or -100.38 × 10−23 (J/K)400(K) = 5. A pigeon in the line-of-sight path causes an additional 2 dB loss. The receiver has an equivalent noise temperature of 400 K and an im. M = 256 square QAM (log2 M = 8. 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 . Teq is always going to be higher than the temperature of your receiver. so: λ [C]dBW = [GT ]dB + [GR ]dB + [PT ]dBW + 20 log10 − 20 log10 R − 2dB − 2dB − 2dB − 1dB 4π 0. Basically.52 × 10−21 J. 255 N0 2 3 log2 M Eb M − 1 N0 255 Q−1 24 32 −8 10 15 = 319.13 × 10−11 W.84 × 106 )J(1. The modulation is 51. √ Solution: Starting with the modulation. incidental losses.075m − 20 log10 R − 7dB = 20dB + 20dB + 10dBW + 20 log10 4π = −1. C = (51.84 Mbits/sec 256 square QAM with a carrier frequency of 4 GHz. the wavelength is λ = 3 × 108 m/s/4 × 109 1/s = 0.plementation loss of 1 dB.ECE 5520 Fall 2009 112 30. Since Eb = C/Rb and the bit rate Rb = 51. This is a topic covered in another course. Teq can be kept low. all receiver circuits add noise to the received signal.36 Consider a “point-to-point” microwave link. With proper design. (Such links were the key component in the telephone company’s long distance network before fiber optic cables were installed.0 The noise power N0 = kTeq = 1. Atmospheric losses are 2 dB and other incidental losses are 2 dB.5 Examples Example: Rice 6. to achieve P [error] = 10−8 . How far away can the two towers be if the bit error rate is not to exceed 10−8 ? Include the pigeon. But in short.76 × 10−18 )1/sec = 9. M = 16).84 × 106 bits/sec. and so you are not responsible for that material here.4 Computing Noise Energy N0 = kTeq The noise energy N0 can be calculated as: where k = 1. So Eb = 319. Neal’s hint: Use the dB version of Friis formula and subtract these mentioned dB losses: atmospheric losses.4 dBW.52 × 10−21 = 1. the equivalent temperature is a function of the receiver design.) Both antenna gains are 20 dB and the transmit antenna power is 10 W.48dBW − 20 log 10 R (55) .075m. 30. implementation loss.0 × 5.38 × 10−23 J/K is Boltzmann’s constant and Teq is called the equivalent noise temperature in Kelvin. and the pigeon.

Figure 8. Lecture 20 Today: (1) Timing Synchronization Intro. and for some delay tk . a transmitted signal arrives with an unknown delay τ . xn (k) = xn (k) = k m r(t). The received complex baseband signal (for QAM. The input signal is downconverted and then run through matched filters. This interpolation is the subject of this section.3 km (about 55 miles) apart. Figure 55 shows a receiver with an ADC immediately after the downconverter. (2) Interpolation Filters 31 Timing Synchronization At the receiver. This tk is the correct timing delay at each correlator for the kth symbol. φn (t) am (k) φn (t − tk ).48dBW − 20 log 10 R and solving for R. which correlate the signal with φn (t − tk ) for each n.4 dBW = −1. As a start. these are generally unknown to the receiver until timing synchronization is complete. let’s compare receiver implementations for a (mostly) continuous-time and a (more) discrete-time receiver. φn (t − kTs − τ ) (57) Note that if tk = kTs + τ . For the correlation with n. PAM. an interpolator corrects the sampling time problems using discrete-time processing. we find R = 88. Figure 54 has a timing synchronization loop which controls the sampler (the ADC).3. then the correlation φn (t − tk ). Instead.1). But. Thus microwave towers should be placed at most 88.ECE 5520 Fall 2009 113 Plugging in [C]dBm = −100. note the ADC has nothing controlling it. φn (t − kTs − τ ) is highest and closest to 1. . Here. after the matched filter. including analog timing synchronization (from Rice book. Figure 54: Block diagram for a continuous-time receiver.3 km. 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.

a BPSK receiver samples the matched filter output at a rate of twice per symbol. are shown in Figure 57. Also. RRC with α = 1) taken at twice the correct symbol rate (vertical lines). The industry trend is more and more towards digital implementations. These are shown in degrees of severity of correction for the receiver. consider Figure 56. resulting in samples r(nT). But again. . this digital and analog feedback loop can be problematic. the timing synch control block is fed back to the ADC. since the sampling clock must operate at (at least) double the symbol rate. compared to the actual symbol clock. but with a timing error. First. because of the processing delay. The idea is to “bring the ADC to the antenna” for purposes of reconfigurability. we’ll consider the control loop. Another implementation is like Figure 55 but instead of the interpolator.ECE 5520 Fall 2009 r(nTsa) ADC cts-time 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. we will emphasize digital timing synchronization using an interpolation filter. this requires a DAC and feedback to the analog part of the receiver. If down-sampling (by 2) results in the symbol samples rk given by red squares. For example. which is not preferred. 32 Interpolation In this class. after the signal has been sampled. When we say ‘synchronized in rate’. and then. we mean within an integer multiple. unsynchronized with the symbol clock. 1 0. then sampling sometimes reduces the magnitude of the desired signal. In this figure. including discrete-time timing synchronization. Some example sampling clocks. and reducing part counts and costs. we’ll talk about interpolation.5 −1 4 6 8 10 Time t 12 14 Figure 56: Samples of the matched filter output (BPSK.5 Value 0 −0. A ‘software radio’ follows the idea that as much of the radio is done in digital.

1 Sampling Time Notation In general. (3) synchronized in rate but not in phase. Calculate (m(k). if T/Ts = 2/5.1 and τ /T = 1. we sample exactly 2. 32. and every 5 samples the delay until the next correct symbol sample will repeat. but no samples were taken at those instants. m are integers. but the sampling clock does not have the correct phase (τ is not equal to an integer multiple of T ). where µ(k) is called the fractional interval. 3. m(k) = kTs + τ T (58) where ⌊x⌋ is the greatest integer less than function (the Matlab floor function). This means that µ(k) is given by kTs + τ µ(k) = − m(k) T Example: Calculation Example Let Ts /T = 3. . the correct sampling times should be kTs + τ . the highest integer such that nT < kTs + τ . (4) synchronized neither in rate nor phase. two clocks are commensurate if the ratio T/Ts can be written as n/m where n. kTs + τ is always µ(k)T after the most recent sampling instant. We can write that kTs + τ = [m(k) + µ(k)]T where m(k) is an integer. In contrast. µ(k)) for k = 1. In other words. our receivers always deal with type (4) sampling clock error as drawn in Figure 57. The situation shown in Figure 56 is case (3). where T /Ts = 1/2 (the clocks are commensurate). 2. the sampling clock has neither the same exact rate nor the same phase as the actual symbol clock. with sample clock.8. For example. we cannot count on them ever repeating. Instead.ECE 5520 Fall 2009 115 Figure 57: (1) Sampling clock and (2-4) possible actual symbol clocks. Since clocks are generally incommensurate. As another example. T/Ts = 1/2. That is.5 times per symbol. for both cases (3) and (4) in Figure 57. Def ’n: Incommensurate Two clocks with rates T and Ts are incommensurate if the ratio T/Ts is irrational. Symbol clock may be (2) synchronized in rate and phase. In general. and 0 ≤ µ(k) < 1. the two clocks are commensurate and we sample exactly twice per symbol period.

r(t) = n r(nT)hI (t − nT) sin(πt/T) . because the sinc function has infinite support. m(2) = ⌊2(3.1) + 1. and Figure 8. r(t) as given in (57). The analog output of the matched filter could be represented as a function of its samples r(nT).4. Instead. we could have it – we just need to calculate r(t) at another set of times (not nT). (60) This is a filter on samples of r(·). The desired sample at [m(k) + µ(k)]T is calculated by adding the weighted contribution from the signal at each sampling time. (60) is a filter. where Ts is the actual symbol period used by the transmitter. The problem is that in general.1 m(3) = ⌊3(3. in which we draw a straight line between the two nearest sampled values to approximate the values of the continuous signal between the samples.13. Rice. we use polynomial approximations for hI (t): • The easiest one we’re all familiar with is linear interpolation (a first-order polynomial). r([m(k) + µ(k)]T) = n r(nT)hI ([m(k) − n + µ(k)]T). Plugging these times in for t in (59). 32.4.8⌋ = 11. Re-indexing with i = m(k) − n. and in between samples 11 and 12.1 + 1. Note: Good illustrations are given in M.8⌋ = 4. µ(1) = 0. Call the correct symbol sampling times as kTs + τ for integer k. using the (m(k). µ(k)) notation. at sample 8. This isn’t so great of an approximation.ECE 5520 Fall 2009 116 Solution: m(1) = ⌊3. since kTs + τ = [m(k) + µ(k)]T. Thus your interpolation will be done: in between samples 4 and 5. where the filter coefficients are dependent on µ(k). . we have that r(kTs + τ ) = n r(nT)hI (kTs + τ − nT) Now.1) + 1. r([m(k) + µ(k)]T) = i r([m(k) − i]T)hI ([i + µ(k)]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.9 µ(1) = 0 µ(1) = 0.2 Seeing Interpolation as Filtering Consider the output of the matched filter.8⌋ = 8. Figure 8. this requires an infinite sum over i from −∞ to ∞. 32.3 Approximate Interpolation Filters Clearly.12.

we can use four points to fit a second-order polynomial. . they are a weighted sum of µ(k)0 . we can see that the filter coefficients are a function of µ(k). Essentially. we could use a cubic interpolation filter. Rice handout. given µ(k). one can determine a parabola that fits those three points exactly.1 Higher order polynomial interpolation filters p In general. µ(k)) = bl (i)µ(k)l l=0 A full table of bl (i) is given in Table 8. . 32. As µ(k) → 0. and get a linear-phase filter. µ(k)2 . hI (i.4. This is temporal asymmetry. .24. 32. Farrow. note in the time domain that two samples are on one side of the interpolation point. we form a weighted average of the two nearest samples. the filter is hI (i) but its values are a function of µ(k). As µ(k) → 1. the three point fit does not result in a linear-phase filter. who has the US patent (1989) for the “Continuously variable digital delay circuit”. µ(k)). see M.1 of the M. and result in a linear filter. µ(k)) = 1 − µ(k). To see results for different order polynomial filters.ECE 5520 Fall 2009 117 • A second-order polynomial (i. • However. we should take the r([m(k) + 1]T) sample exactly.e. µ(k)p for a pth order polynomial filter. we should take the r(m(k)T) sample exactly. . µ(k)) = µ(k) and hI (0. of AT&T Bell Labs.) Instead. Note that the i indices seem backwards. 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. The filter coefficients are a polynomial function of µ(k).4 Implementations Note: These polynomial filters are called Farrow filters and are named after Cecil W. µ(k)1 . Rice Figure 8. the fractional interval. and one on the other. that is. 0 r([m(k) + µ(k)]T) = i=−1 r([m(k) − i]T)hI (i. (61) r([m(k) + µ(k)]T) = i That is. From (60). Thus we could re-write (60) as r([m(k) − i]T)hI (i. 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.a parabola) is actually a very good approximation. Four points determine a 3rd order polynomial. Given three points. consider the linear interpolator. . (To see this. Example: First order polynomial interpolation For example. • Finally.

there is an extra degree of freedom – you can select parameter α to be in the range 0 < α < 1.5. 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. It has been shown by simulation that α = 0.25 = −0.m and is posted on WebCT. 0.43 is best. as my implementation uses a loop.125 hI (0. Filter Timing e(n) Loop Error Filter Detector s [ n] h[n]=p(-nT) 2 cos(W0n) v(n) symbol strobe VCC m Interp.125 hI (−1. rather than vector processing.25 = −0.625 (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 lec20.25 + 1 = 0. but people tend to use α = 0. Be careful. and division by two is extremely easy in digital filters. 0. timing synchronization for QPSK.ECE 5520 Fall 2009 118 For the 2nd order Farrow filter.125 + 0.25. Lecture 21 Today: (1) Timing Synchronization (2) PLLs 33 Final Project Overview xI(n) MF N/2 Interp. .5. µ = 0.5) = αµ2 − αµ = 0. we can calculate that hI (−2.125 − 0.5) = αµ2 − αµ = 0.75 = 0. Example: 2nd order Farrow filter What is the Farrow filter for α = 0. µ0 = 1.625 hI (1.5 which interpolates exactly half-way between sample points? Solution: From the problem statement. µ = 0. 0. 0.5) = −αµ2 + (α − 1)µ + 1 = −0.5 because it is only slightly worse.5) = −αµ2 + (1 + α)µ = −0. Since µ2 = 0.125 − 0.125 − 0.

1 Review of Interpolation Filters Timing synchronization is necessary to know when to sample the matched filter output. it works nearly as well. We will talk through two common discrete-time implementations: 1. the samples may be at incorrect times. • Problem: After the matched filter. where n is the integer part and µ is the fractional offset. which has output denoted xI (n). Interpolation Filter 2. or a mix.2 Timing Error Detection We consider timing error detection blocks that operate after the matched filter and interpolator in the receiver. Timing Error Detector (TED) 3. • Solution: From samples taken at or above the Nyquist rate. 33. discrete time.4. We want to sample at times [n + µ]Ts . There are several possible timing error detection (TED) methods. • Problem: How do you know when the correct symbol sampling time should be? • Solution: Use a timing locked loop. We focus on the discrete time solutions.4. we leave out the Ts and simply talk about the index n or fractional delay µ. It adds these blocks to your QPSK implementation: 1. there may be no analog feedback to the ADC. you will implement the above receiver. as related in Rice 8. Preamble Detector These notes address the motivation and overall design of each block. • Solution: Approximate the sinc function with a 2nd or 3rd order polynomial interpolation filter. you will need to understand how and why they work. you can interpolate between samples to find the desired sample. Loop Filter 4. Early-late timing error detector (ELTED) . you have the advantage of having access to the the Rice book. Section 8.1. This is a good source for detail on many possible methods.ECE 5520 Fall 2009 119 For your final project. Compared to last year’s class. in order to fix any bugs. Often. Implementations may be continuous time. Voltage Controlled Clock (VCC) 5. and in modern discrete-time implementations. analogous to a phased locked loop. When you put these together and implement them for QPSK.4. which contains much of the code for a BPSK implementation. 33. However this solution leads to new problems: • Problem: True interpolation requires significant calculation – the sinc filter has infinite impulse response. 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.

You can imagine that the interpolator output xI (n) is a sampled version of x(t).5 Figure 59: A sample eye-diagram of a RRC-shaped BPSK received signal (post matched filter).5 1 Time from Symbol Sample Time 1. we can use it to indicate how far from the sampling time we might be. the error e(n) is determined by the slope and the sign of the sample x(n). x(t). hopefully with some samples taken at exactly the correct symbol sampling times.5 x(t) Value 0 −0.5 −1 −4 (a) x 10 5 −0. 1 0. In general. to -1.5 0 0. 1. x(t).5 Slope of x(t) 0 −5 −1. In ˙ both. When the bit is constant (e.ECE 5520 Fall 2009 120 2. to 1) the slope is close to zero. time 0 corresponds to a correct symbol sampling time. In particular. consider for BPSK the value of x(t)sgn {x(t)} ˙ .5 −1 (b) −0.g.5 1 Time from Symbol Sample Time 1..5 0 0.5 −1 −1. at time n. Here (a) shows the signal x(t) and (b) shows its derivative x(t). Figure 59 shows that the derivative is a good indication of timing error when the bit is changing from -1. ˙ 33. to illustrate the operation of timing error detectors. When the bit sequence is The early-late TED is a discrete-time implementation of the continuous-time “early-late gate” timing error detector. Zero-crossing timing error detector (ZCTED) We’ll use a continuous-time realization of a BPSK received matched filter output. Figure 59(a) shows an example eye diagram of the signal x(t) and Figure 59(b) shows its derivative x(t). to 1.3 Early-late 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. to 1.

17 of Rice. (63) can be re-written as e(n) = xI (n − 1)[sgn{xI (n − 2)} − sgn{xI (n)}] (66) This operation is drawn in Figure 60. the slope will be somewhat positive even when the sample is taken at the right time.1 of the Rice book.4. to -1. Basically. For a discrete time system. In the project. 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. The error is.ECE 5520 Fall 2009 121 from 1.4 Zero-crossing timing error detector (ZCTED) The zero-crossing timing error detector (ZCTED) is also described in detail in Section 8. and when the bit is changing from -1. if the sign changes between n − 2 and n. a negative slope would mean we’re ahead for a +1 symbol. if it was taken at the correct sampling time. you will need to look up Kp in Figure 8. the slope will be somewhat negative even when the sample is taken at the right time. This factor contributes to the gain Kp of this part in the loop. and then multiply it by 2 to find the gain Kp of your ZCTED. In particular see Figure 8. which will be 2 when the sign changes. if the symbol strobe is not activated at sample n. but would mean that we’re behind for a -1 symbol. to 1. The signum function sgn {x(t)} just corrects for the sign: • A positive slope would mean we’re behind for a +1 symbol. 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).4. ˆ ˆ a(k) = sgn {x(kTs + τ )} . you get only an approximation of the slope unless the sampling rate SP S (or N . that indicates a symbol transition. It is described here for BPSK systems. ˆ a(k − 1) = sgn {x((k − 1)Ts + τ )} . to -1. In terms of n.4. This error detector assumes that the sampling rate is SP S = 2. That is. and we just need something proportional to the timing error. The theoretical gain in the ZCTED is twice that of the ELTED. 33. to 1. the intermediate sample n − 1 should be approximately zero. we can send alternating bits during the synchronization period in order to help the ELTED more quickly converge. The factor of two comes from the difference of signs in (66). This timing error detector has a theoretical gain Kp which shown in Figure 8. In particular. we don’t divide by the time duration 2Ts because it is just a scale factor.7 of the Rice book. then the error e(n) = 0. These are imperfections in the ELTED which (hopefully) average out over many bits. as it is called in the Rice book) is high. which is a function of the excess bandwidth of the RRC filter used. . because every second sample is a symbol sampling time. • In the opposite manner. but would mean that we’re ahead for a -1 symbol. ˆ ˆ (64) (65) The error detector is non-zero at symbol sampling times. If there was a symbol transmission.8 of the Rice book.

(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. the output is 2xI (n − 1). The input strobe signal activates the sampler when it is the correct symbol sampling time.4.. where sgn indicates the signum function. See (8. and the symbol has switched. This is shown in Figure 61..5 Voltage Controlled Clock (VCC) We’ll use a decrementing VCC in the project. The error is simply the sum of the two errors calculated for each signal. NCO(n-1) m W(n) Figure 62: Numerically-controlled oscillator signal N CO(n). NCO(n-2) NCO(n) . . 33. Strobe high: change m to m=NCO(n-1)/W(n) 1/2 v(n) W(n) symbol strobe m underflow? symbol strobe z -1 modulo-1 NCO(n-1) Voltage Controlled Clock (VCC) Figure 61: Block diagram of decrementing VCC with control voltage v(n).1 QPSK Timing Error Detection When using QPSK. 33.100) in the Rice book for the exact expression. which would be approximately zero if the sample clock is synchronized. 1 when positive and -1 when negative.ECE 5520 Fall 2009 122 Timing Error Detector sgn sgn e(n) xI(n) z-1 xI(n-1) z-1 xI(n-2) symbol strobe Figure 60: Block diagram of the zero-crossing timing error detector (ZCTED). When this happens. both the in-phase and quadrature signals are used to calculate the error term.

The function g(θe ) may look something like the one shown in Figure 64. then g(θe ) < 0 and the VCO decreases the phase of the VCO output. µ(n) is kept the same. 3. the NCO decrements by 1/SP S + v(n). When the symbol strobe is activated.6. This section is included for reference. This is the effect of the ‘S’ curve function g(θe ) in Figure 64. 33. The incoming signal is a carrier wave. meaning that a sample must be taken. For example. behind the carrier. that is. In this case. the phase offset is a ramp.. If the phase estimate is too low. that is. µ(n) = N CO(n − 1) W (n) (67) When the strobe is not activated. the operation of the timing synchronization loop may be best understood by analogy to the continuous time carrier synchronization loop. 2. 33. θe (t) = 0. Initially. µ(n) = µ(n − 1). . When this happens. ˆ The job of the the PLL is to estimate φ(t). We call this estimate θ(t). Consider a generic PLL block diagram in Figure 63. A cos(2πfc t + θ(t)) where θ(t) is the phase offset. You will prove that (67) is the correct form for µ(n) in the HW 9 assignment. and then θ(t) = ∆f t + α. If the phase estimate is exactly correct. because it may not be constant. For those of you familiar with PLLs.ECE 5520 Fall 2009 123 The NCO starts at 1. At each sample n. The error in the phase estimate is ˆ θe (t) = θ(t) − θ(t) If the PLL is perfect.6 Phase Locked Loops Carrier synchronization is done using a phase-locked loop (PLL). let’s ignore the loop filter. ahead of the carrier. Once per symbol.e. (on average) the NCO voltage drops below zero. 1. then g(θe ) = 0 and VCO output phase remains constant. If the phase estimate is too high. t. the strobe is set to high. It is a function of time. the fractional interval is also recalculated.1 Phase Detector If the estimate of the phase is not perfect. It is. 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 phase-locked loop for carrier synchronization [Rice]. then g(θe ) > 0 and the VCO increases the phase of the VCO output. it may include a frequency offset. the phase detector produces a voltage to correct the phase estimate. i.

e. as we know. for continuous time. so we could also write the Z-transform response as F (z). 33. in the Laplace domain [Rice].4 Analysis To analyze the loop in Figure 63. we can re-draw Figure 63 as it is drawn in Figure 65. In particular. This phase-equivalent model of the PLL allows us to do analysis. 33. but leave the frequency the same.ECE 5520 Fall 2009 124 g(qe) qe Figure 64: Typical phase detector input-output relationship. θe = 0. This linear approximation is generally fine when the phase error is reasonably small. we model the ‘S’ curve (Figure 64) as a line. The input is essentially the gas pedal for a race car driving around a track . . We will be interested in discrete time later. This curve is called an ‘S’ Curve.6.6. If we do this. 33. because the shape of the curve looks like the letter ‘S’ [Rice]. you change the phase of the output. i. is the cosine of 2πfc plus that phase.6. and the engine (VCO) will increase the frequency of rotation around the track.increase the gas (voltage). ˆ In the left half of Figure 65..3 VCO A voltage-controlled oscillator (VCO) simply integrates the input and uses that integral as the phase of a cosine wave. we end up making simplifications. we write just Θ(s) and Θ(s) even though the physical process.2 Loop Filter The loop filter is just a low-pass filter to reduce the noise. We represent the loop filter in the Laplace or frequency domain as F (s) or F (f ). Note that in addition to the A cos(2πfc t + θ(t)) signal term we also have channel noise coming into the system. Note that if you just raise the voltage for a short period and then reduce it back (a constant plus rect function input). phaseequivalent perspective Q(s) kp F(s) V (s ) Q(s) k0 s Figure 65: Phase-equivalent and linear model for the continuous-time PLL. respectively.

In this case there are four parameters to set. For example. Desired bandwidth. The latter deserves more explanation. Designing a filter to respond to these. For example. damping coefficient ζ. Note: A accelerating frequency change wouldn’t be handled properly by the given filter. and other gain constants K0 and Kp . is a challenge. You will. F (s) = k1 + k2 s This filter has a proportional part (k1 ) which is just proportional to the input. or ramp input.1 (p 733. Here those goals include: 1.2 details how to set these parameters to achieve a desired bandwidth and a desired damping factor (to avoid ringing. However. and kp . T 1 + z −1 1 → s 2 1 − z −1 Because it is only a second-order filter. k0 . but to converge quickly). but not to AWGN noise (as much as possible).2.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). See Figure C. the frequency of the crystal will also change quickly.57 (p 736) show how to calculate the constants K1 and K2 . 33.56 and C. Rice book) for diagrams of this discrete-time 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. The filter also has an integrator part (k2 /s) which integrates the phase input. the normalized bandwidth Bn T . design the particular loop filter to use in your final project. k2 .6. in Homework 9. The Rice book recommends a 2nd order proportional-plus-integrator loop filter. The Rice Section C.5 Discrete-Time Implementations You can alter F (s) to be a discrete time filter using Tustin’s approximation. will ramp up the phase at the proper rate. this is not typically fast enough to be a problem in most radios. Desired response to particular phase error models. 2. k1 . Equations C. which responds to the input during phase offset. its implementation is quite efficient. phase error models might be: step input. given the desired natural frequency θn . and in case of a frequency offset. . if the temperature of the transmitter or receiver varied quickly.

unipolar • M -ary: PAM.005). FSK 4. Please review these homeworks and know the correct solutions. bipolar vs. • Rice book: Chapters 5 and 6. • I will be out of town Mon-Wed. QAM. PSK. Nyquist filtering. • Office hours: Today: 2-3:30. given signal space diagram. What topics: 1. Lecture 22 Today: (1) Exam 2 Review • Exam 2 is Tue Apr 14. 11-19. Please come to me with questions today or tomorrow. Gray encoding 8. Energy detection for FSK 7. Eb N0 .ECE 5520 Fall 2009 126 Note that the bandwidth Bn T is the most critical factor. . Differential encoding and decoding for BPSK 6. Signal space 3. Probability of Error vs. • Standard modulation method. 3-4. Friday 11-12. Detection theory 2. 0. (e. It is typically much less than one. nearest-neighbor approximation. Inter-symbol interference.g. 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: 4-8. union bound. Lectures 20.. No material from 1-7 will be directly tested.21 are not covered. 34 Exam 2 Topics Where to study: • Lectures covered: 8 (detection theory). SRRC pulse shaping 5. Digital modulation methods: • Binary. • A non-standard modulation method. This means that the loop filter effectively averages over many samples when setting the input to the VCO. The exam will be proctored. 9. • Exact expression. Both know the formula and know how it can be derived.

System design: Here are some engineering requirements for this communication system. Monday 4-5pm.bell-labs. pp.com/cm/ms/what/shannonday/ paper. Images and sound can be encoded in different ways.cs. concept of frequency multiplexing 10.. Text files can be presented in different ways. vol. 2. OH today 2-3pm. or in bits per second. 27. Here are two misconceptions: .. 35 Source Coding This topic is a semester-long course at the graduate level in itself.” Bell System Technical Journal.)? Information can often be represented in many different ways. but there may be many wrong answers (or right answer / wrong reason combinations). FSK. From this list of possible modulation. Short answer. video. True / false or multiple choice. which one would you recommend? What bandwidth and bit rate would it achieve? You will be provided the table of the Qinv function. Work out solution.html We have done a lot of counting of bits as our primary measure of communication systems. 3. There may be multiple correct answers. Comparison of digital modulation methods: • Need for linear amplifiers (transmitter complexity) • Receiver complexity • Bandwidth efficiency Types of questions (format): 1.g.. Our information source is measured in bits. 379-423 and 623-656. audio. Bandwidth assuming SRRC pulse shaping for PAM/QAM/PSK.ECE 5520 Fall 2009 127 9. “A mathematical theory of communication. http://www. and the plot of Qinv.5 x 11 sheet of paper. Eb N0 for this signal space diagram? 4. July and October. • Final homework: HW 10. You can use two sides of an 8. and energy efficiency is energy per bit over noise PSD. E. Modulation schemes’ bandwidth efficiency is measured in bits per Hertz. . which will be due April 28 at 5pm. Example: What is the probability of bit error vs. Everything is measured in bits! But how do we measure the bits of a source (e. 1948. email. Shannon. with a 1-2 sentence limit. One good reference: • C. but the basic ideas can be presented pretty quickly. Lecture 23 Today: (1) Source Coding • HW 9 is due today at 5pm.

Zip or compress. and x ∈ SX . . 2. 2. So what is the intrinsic measure of bits of text.1 Entropy Entropy is a measure of the randomness of a random variable.v. that telling conveys quite a bit of information to you. 1. Here.ECE 5520 Fall 2009 128 1. What if we had a fixed number of bits to send any image.g. JPEG. in non-technical language. are just two perspectives on the same thing: • If you are told the value of a r. . This would introduce errors into the image. there is a finite or countably infinite set SX .) above? Sometimes. H [X] = − pi log2 pi (68) i Notes: . MP3 Note: Both “zip” and the unix “compress” commands use the Lempel-Ziv algorithm for source compression. For example.v. the number of bits in the compressed image would exceed what we had allocated. • Lossy: e. consider a digital black & white image (not grayscale.. . x2 .zip or . We will shorten the notation by using pi as follows: pi = pX (xi ) = P [X = xi ] where {x1 . You could simply send the coordinates of the pixels of one of the colors (e. Def ’n: Entropy Let X be a discrete random variable with pmf pX (xi ) = P [X = x]. Using the file size tells you how much information is contained within the file.tar files represent the exact same information that was contained in the original files. Our technical definition of entropy of a random variable is as follows. an image. Take the log2 of the number of different messages you could send. that telling conveys very little information to you.g. You could store it as a list of pixels. or video? 35. in units of bits.g. . audio. and we used the sparse B&W image coding scheme (2. Then the entropy of X. that doesn’t vary that much. • If you are told the value of a very “random” r. but with fewer bits. thus we could encode it in log2 2 or one bit per pixel. For example.all black pixels). Two types of compression algorithms: • Lossless: e...} is an ordering of the possible values in SX . Randomness and information. but truly black or white). 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. Each pixel has two possibilities (possible messages). is defined as. This is the idea behind source compression.

they express information in “nats”.w.6 0. pi . Keep this in mind when reading a book on information theory. . x = 2   1/4. Example: Binary r. • All “log” functions are log-base-2 in information theory unless otherwise noted. N when |SX | = N < ∞. another operator on a random variable. short for “natural” digits. . not a function of a random variable. It returns a (deterministic) number. pX (x) = 1 − s.8 1 Figure 66: Entropy of a binary r. . o.4 0. Actually. Example: Non-uniform source with five messages Some signals are more often close to zero (e. • Use that 0 log 0 = 0. A binary (Bernoulli) r. x = 0 pX (x) =  1/8. This it is like E [X]. The “reason” the units are bits is because of the base-2 of the log. x = −1    1/16. x = 1     1/2. • The sum will be from i = 1 .audio). x = 0  0. not another random variable. o. Model the r.2 0 0 0.4 0.w. has pmf.v.6 P[X=1] 0.ECE 5520 Fall 2009 129 • H [X] is an operator on a random variable. x = −2    0. 1 0.  x=1  s. • Entropy of a discrete random variable X is calculated using the probability values of the pmf of X.8 Entropy H[X] 0. Nothing else is needed.2 0.v.v. This is true in the limit of x log x as x → 0+ . 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.g. X to have pmf   1/16. when theorists use loge or the natural log.v.

s is less than N times the entropy of one of them. the joint r.X2 (x1 .XN (x1 . . . X2 ] = − pX1 . compared to just one of them?”. XN ] = −N pX1 (x1 ) log2 pX1 (x1 ) = N H(X1 ) x1 ∈SX1 The entropy of N i.s are not independent. X1 . X1 ] − H[X1 ]. H[X2 |X1 ]: H[X2 |X1 ] = H[X2 .v. random variables has N times the entropy of any one of them.X2 (x1 . . entropy is H[X1 . .d. Intuitively. . the B&W image. . the rest that you don’t know become less informative.v. xN ) log2 pX1 . the joint entropy of N r. . xN ) xN ∈SXN x1 ∈SX1 What is the entropy for N i. . (71) . because of the dependence or correlation.. . We call this difference the conditional entropy.v. random variables? You can show that H[X1 .i. of several neighboring pixels will have less entropy than the sum of the individual pixel entropies.. x2 ) x1 ∈SX1 x2 ∈SX2 (70) For N joint random variables.XN (x1 . Would multiplying X by 2 change its entropy? 3. the entropy of any N independent (but possibly with different distributions) r..3 Conditional Entropy How much additional entropy is in the joint random variables X1 .2 Joint Entropy Def ’n: Joint Entropy The joint entropy of two random variables X1 .v. . . X2 with event sets SX1 and SX2 is defined as H[X1 .d. For example. XN . .ECE 5520 Fall 2009 130 What is its entropy H [X]? Solution: H [X] = = Other questions: 1. . .v.. since pixels are correlated in space. Do you need to know what the symbol set SX is? 2.s is just the sum of the entropy of each individual r. . In addition. . XN ] = − ··· pX1 . . 35.. x2 ) log2 pX1 .i. Would an arbitrary one-to-one 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. “How much additional information do I get from both. X2 compared just to one of them? This is often an important question because it answers the question.. if you know some of them.. . . When r. ..

. X1 ] = − ··· pX1 .X2 (x1 . in units of bits per random variable (a. . This is not like either the joint or the marginal entropy. is defined as H = lim H[XN |XN −1 . .X1 (xN |xN −1 . we’re interested in discrete-time random processes. x2 ) log2 pX1 .X2 (x1 .X2 (x1 .X2 (x1 . x2 ) log2 pX1 (x1 ) x1 ∈SX1 x2 ∈SX2 x1 ∈SX1 x2 ∈SX2 pX1 .4 Entropy Rate Typically. H[XN |XN −1 . x2 ) and the conditional probabilities pX2 |X1 (x2 |x1 ). . x2 ) + x1 ∈SX1 x2 ∈SX2 x1 ∈SX1 pX1 (x1 ) log2 pX1 (x1 )    pX1 .X2 (x1 . X2 . .. What is the sample space SXi ? Is Xi uniform on that space? . Solution: Plugging in (68) for H[X2 .X2 (x1 . x2 ) − log2 pX1 (x1 )) pX1 . X1 ]. XN ].. . x1 ) xN−1 ∈SXN−1 x1 ∈SX1 which is the additional entropy (or information) contained in the N th random variable. For this case. . How many additional bits. . the joint entropy of all of them may go to infinity as N → ∞.X2 (x1 .X2 (x1 . N →∞ N Example: Entropy of English text Let Xi be the ith letter or space in a common English sentence. x2 ) + x1 ∈SX1 x2 ∈SX2 x1 ∈SX1 x2 ∈SX2 x1 ∈SX1 x2 ∈SX2 pX1 . x2 ) (log2 pX1 . . . in the limit.X2 (x1 . N →∞ It can be shown that entropy rate can equivalently be written as H = lim 1 H[X1 . source output). . .v. H[X2 |X1 ] = − = − = − = − = − pX1 . x2 ) pX1 (x1 ) pX1 .k. we are more interested in the rate. Since there are infinitely many of them.. X2 .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..ECE 5520 Fall 2009 131 What is an equation for H[X2 |X1 ] as a function of the joint probabilities pX1 . in which we have random variables X1 .. . . are needed for the average r. x2 ) log2 pX1 .. We could also have multi-variate conditional entropy. X1 ] and H[X1 ]. . .X2 (x1 .a. as N → ∞? Def ’n: Entropy Rate The entropy rate of a stationary discrete-time random process. . given the values of the N − 1 previous random variables. . . x2 ) log2 pX1 . xN ) log2 pXN |XN−1 .XN (x1 . 35...

we have H = 1. See Figure 67(a).04 = 3 · 3.15 u p k f Poccurrence 0.2].025 0. which is 2 · 3.005 0.3 bits/letter [taken from Proakis & Salehi Section 6. Xi+1 ]? Solution: Again.35. For the three-letter combinations. • 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. • Theorem fact: Information measure (entropy) gives us a minimum bit rate. and was introduced by Claude Shannon in 1948. What is the entropy rate. the error probability will be bounded away from zero. For this pmf. I calculate an entropy of 7.05 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) two-letter combinations (including spaces) in Shakespeare’s Romeo and Juliet. if R < H. H? Solution: For N = 10.1199. Conversely.99. the joint entropy was 11. 0. For four-letter combinations. using Matlab on Shakespeare’s Romeo and Juliet.2 z 0. the joint entropy was 10. You can see that the average entropy in bits per letter is decreasing quickly. at any rate R (bits / source output) as long as R > H. • R is our 1/Tb . The Proakis & Salehi book mentions that this value for general English text is about 4. See Shannon’s original 1948 paper. I calculated the entropy of the joint pmf of each two-letter combination.1 0. Notes: • Here. What is H[Xi . an ‘error’ occurs when your compressed version of the data is not exactly the same as the original.3. Example: B&W images. I calculated an entropy of H = 4.5 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.02 0. Theorem: A source with entropy rate H can be encoded with arbitrarily small error probability.01 0.46.015 0. This gives me the two-dimensional pmf shown in Figure ??(b).ECE 5520 Fall 2009 132 What is H[Xi ]? Solution: I had Matlab read in the text of Shakespeare’s Romeo and Juliet. 35. Proof: Proof: Using typical sequences. It is one of the two fundamental theorems of information theory.98 = 4 · 2. .73. independent of the complexity of the encoder and the decoder employed.03 First Space or Letter 0.

at any rate R (bits / source output) as long as R > H.1 R. XN ]. as described by Nyquist. Tsy In Hartley’s 1928 paper. he developed relationships useful for determining the capacity of bandlimited communication channels. Afterwards. 30.B. for a finite source. . without loss. In July 1928. involved in radio telephony. Any lower rate than H would guarantee loss of information. X2 . Then. the value of H[X] or H gives us a measure of how many bits we actually need to use to encode. . . The major result was the Shannon’s source coding theorem. 1 H[X1 . V. He worked as a researcher for the Western Electric Company. 37. 1 ≤ 2B. Hartley was particularly influenced by Nyquist’s result. H = lim 37 Channel Coding Now. he considered digital transmission in pulse-amplitude modulated systems. Given our information source X or {Xi }. This discussion of entropy allows us to consider the maximum data rate which can be carried without error on a bandlimited channel. L. he published in the Bell System Technical Journal a paper on “Transmission of Information”. When transmitting a sequence of pulses. which says that a source with entropy rate H can be encoded with arbitrarily small error probability. we defined entropy. each pulse could represent more or less information. and entropy rate. Hartley assumed that the maximum amplitude available to the transmitter was A. the source data. Hartley made the assumption that the communication system could discern between pulse amplitudes. Here he developed the “Hartley oscillator”. In particular. degree from the University of Utah in 1909. That is. Nyquist determined that the pulse rate was limited to two times the available channel bandwidth B. the rate may need to be higher. each of duration Tsy . L. The pulse rate was limited to 2B. at Bell Laboratories. which is affected by additive White Gaussian noise (AWGN). But. we turn to the noisy channel. Hartley (born Nov. if they . 1888) received the A. . Lecture 24 Today: (1) Channel Capacity 36 Review H[X] = − pi log2 pi i Last time. depending on how pulse amplitudes were chosen. Hartley Ralph V.ECE 5520 Fall 2009 133 • The theorem does not tell us how well it can be done if N is not infinite. N →∞ N We showed that entropy can be used to quantify information.

d. n 1 (yi − xi )2 ≤ PN n i=1 happens with probability one. . possible from the digital communication system. Gaussian with variance PN . In fact. 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). . The noise term zi is assumed to be i. . y = [y1 . and did not say anything fundamental about the relationship between capacity and bandwidth for arbitrary modulation. Given that a PAM system operates from 0 to A in increments of Aδ . 37. In other words. and used statistics to develop a universal bound for capacity. Engineers would have to be conservative when setting Aδ to ensure a low probability of error.1 Noisy Channel Shannon did take into account an AWGN channel. Hartley used the ‘bit’ measure to quantify the data which could be encoded using M amplitude levels. in bits per second. of length n. In this AWGN channel. 37. . This late decision will decide all values of y = [x1 . Furthermore. This asymptotic limit as n → ∞ allows for a proof using the statistical convergence of a sequence of random variables. as n → ∞. Further. E. we need a law called the law of large numbers (ECE 6962 topic). log2 M = log2 1 + A Aδ Finally. In particular. the measured value y will be located √ within an n-dimensional sphere (hypersphere) of radius nPN with center x. yn ]. assuming perfect syncronization) is yi .ECE 5520 Fall 2009 134 were at separated by at least a voltage spacing of Aδ . and did not deal well with negative amplitudes. All n symbols are received before making a decision. . Next. Hartley quantified the data rate using Nyquist’s relationship to determine the maximum rate C. 37.i. Shannon What was left unanswered by Hartley’s capacity formula was the relationship between noise and the minimum amplitude separation between symbols. . xn ] simultaneously.2. the capacity formula was for a particular type of PAM system.2 Introduction of Latency Shannon’s key insight was to exchange latency (time delay) for reduced probability of error. yi = xi + zi where X is the transmitted signal and Z is the noise in the channel. Shannon’s proof considers the limiting case as n → ∞. . regardless of modulation type.2 C. as n → ∞. the number of different pulse amplitudes (symbols) is M =1+ A Aδ Note that early receivers were modified AM envelope detectors. . his capacity bound considers simultaneously demodulating sequences of received symbols. the ith symbol sample at the receiver (after the matched filter. .2.

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.3 Introduction of Power Limitation Shannon also formulated the problem as a power-limited case. A] such that they are all separated by Aδ . with probability one as n → ∞. n 1 x2 ≤ P i n i=1 This combination of signal power limitation and noise power results in the fact that. 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. 37. show how we many different symbols we could have uniquely distinguished. in which the average power in the desired signal xi was limited to P . Hartley asked how many symbol amplitudes could be fit into [0. This is shown in P&S in Figure 9. This number M is the number of different messages that could have been sent in n pulses. we can replace the noise variance PN with the relationship between noise two-sided PSD N0 /2.9 (pg.3. The result of this geometrical problem is that P ( n/2) (73) M = 1+ PN 37. That is. 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.1 Returning to Hartley Adjusting Hartley’s formula. such that hyperspheres of radius nPN do not overlap. C = P 2B n log2 1 + n 2 PN P = B log2 1 + PN 37. is contained within a hypersphere of radius n(P + PN ) centered at the origin.2.ECE 5520 Fall 2009 135 37.3 Combining Two Results The two results. 584).3.2 Final Results Finally. by PN = N0 B . within a period of n sample times.

That is. 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. so Rb Eb Rb ≤ log2 1 + W W N0 Defining η = Rb W. C = B log2 1 + Typically.ECE 5520 Fall 2009 136 So finally we have the Shannon-Hartley Theorem. because of the late due date. set your deadline to May 4. the energy per bit is the power multiplied by the bit duration. No late projects are accepted. or signal to noise ratio (SNR). “Digital Terrestrial Television Broadcasting”. 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 ).4 Efficiency Bound Recall that Eb = P Tb . C is just a capacity limit. 5pm. C = W log2 1 + Eb /Tb N0 W Rb Eb W N0 Here. Instead. where Rb is the operating bit rate. May 5. This relationship is shown in Figure 70. you can look at it as a bound on the bandwidth E efficiency as a function of the Nb ratio. Any practical communication system must operate at Rb < C. . • Everyone gets a 100% on the “Discussion Item” which I never implemented. C = W log2 1 + or since Rb = 1/Tb . Thus 0 the capacity bound is also written C = W log2 (1 + SN R). is canceled. with arbitrarily low probability of error. 0 Lecture 25 Today: (1) Channel Capacity • Lecture Tue April 28. If you think you might be late. please the Robert Graves lecture 3:05 PM in Room 105 WEB. However. η ≤ log2 1 + η Eb N0 This expression can’t analytically be solved for η. We know that our bit rate Rb ≤ C. Note that the ratio NPW is the signal power divided by the noise power. 37. Thus from (74). • The final project is due Tue.

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 Shannon-Hartley theorem. η. . on a (a) linear plot and (b) log plot. bound on bandwidth efficiency.

1 R. Given that a PAM system operates from 0 to A in increments of Aδ .ECE 5520 Fall 2009 138 38 Review H[X] = − pi log2 pi i Last time. as described by Nyquist. The pulse rate was limited to 2B. Hartley (born Nov. and did not deal well with negative amplitudes. if they were at separated by at least a voltage spacing of Aδ . which is affected by additive White Gaussian noise (AWGN). . depending on how pulse amplitudes were chosen. he developed relationships useful for determining the capacity of bandlimited communication channels. XN ]. Hartley assumed that the maximum amplitude available to the transmitter was A. at any rate R (bits / source output) as long as R > H. L. log2 M = log2 1 + A Aδ . Nyquist determined that the pulse rate was limited to two times the available channel bandwidth B. N We showed that entropy can be used to quantify information. Hartley was particularly influenced by Nyquist’s sampling theorem. we turn to the noisy channel. which says that a source with entropy rate H can be encoded with arbitrarily small error probability. Hartley used the ‘bit’ measure to quantify the data which could be encoded using M amplitude levels. This discussion of entropy allows us to consider the maximum data rate which can be carried without error on a bandlimited channel. the source data. 30. Then. we defined entropy. H = lim N →∞ 1 H[X1 . and entropy rate. 39. Any lower rate than H would guarantee loss of information. . each pulse could represent more or less information. without loss. Given our information source X or {Xi }. X2 . V. the number of different pulse amplitudes (symbols) is M =1+ A Aδ Note that early receivers were modified AM envelope detectors. involved in radio telephony. But. he published in the Bell System Technical Journal a paper on “Transmission of Information”. Next. each of duration Ts . Afterwards. The major result was the Shannon’s source coding theorem. He worked as a researcher for the Western Electric Company. L.B. In particular. Ts In Hartley’s 1928 paper. . he considered digital transmission in pulse-amplitude modulated systems. Hartley Ralph V. the value of H[X] or H gives us a measure of how many bits we actually need to use to encode. In July 1928. 1888) received the A. 1 ≤ 2B. Hartley made the assumption that the communication system could discern between pulse amplitudes. . at Bell Laboratories. 39 Channel Coding Now. degree from the University of Utah in 1909. When transmitting a sequence of pulses.

These might be truly an n-dimensional signal (e.i. of length n. This law says that the following event.2. the measured value y will be located √ within an n-dimensional sphere (hypersphere) of radius nEN with center x. Gaussian with variance EN = N0 /2. In the multiple symbols over time. his capacity bound considers n-dimensional signaling. in bits per second.d. or they might use multiple symbols over time (recall that symbols at different multiples of Ts are orthogonal). In fact. as n → ∞. the capacity formula was for a particular type of PAM system. yn ]. as n → ∞. we need a law called the law of large numbers. In other words. and did not say anything fundamental about the relationship between capacity and bandwidth for arbitrary modulation. Hartley quantified the data rate using Nyquist’s relationship to determine the maximum rate C. 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. and used statistics to develop a universal bound for capacity. 39.ECE 5520 Fall 2009 139 Finally. C = 2B log2 1 + A Aδ 39.2 Introduction of Latency Shannon’s key insight was to exchange latency (time delay) for reduced probability of error. Engineers would have to be conservative when setting Aδ to ensure a low probability of error. xn ] simultaneously.1 Noisy Channel Shannon did take into account an AWGN channel. possible from the digital communication system. Shannon What was left unanswered by Hartley’s capacity formula was the relationship between noise and the minimum amplitude separation between symbols. That is. . . 39.2. Shannon uses all n dimensions in the constellation – the detector must use all n samples of y to make a decision. . . Furthermore. Further. n 1 2 xi ≤ E n i=1 . assuming perfect synchronization) is yi . . E.2.2 C. In particular. Shannon’s proof considers the limiting case as n → ∞.3 Introduction of Power Limitation Shannon also formulated the problem as a energy-limited case. . In either case. So the received vector is y = [y1 . yi = xi + zi where X is the transmitted signal and Z is the noise in the channel. . 39. .. this late decision will decide all values of x = [x1 . FSK). In this AWGN channel. regardless of modulation type.g. the ith symbol sample at the receiver (after the matched filter. 1 n n i=1 (yi − xi )2 ≤ EN happens with probability one. The noise term zi is assumed to be i.

such that hyperspheres of radius nEN do not overlap. C= 2B n E log2 1 + n 2 EN = B log2 1 + E EN . 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. 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. 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 ≤ E + EN This result says that the vector y. The result of this geometrical problem is that E n/2 (76) M = 1+ EN 39.3 Combining Two Results The two results.1 Returning to Hartley Adjusting Hartley’s formula. 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. ≤ n(E + EN ) 39. Hartley asked how many symbol amplitudes could be fit into [0.ECE 5520 Fall 2009 140 This combination of signal energy limitation and noise energy results in the fact that. This is shown in Figure 69. with probability one as n → ∞. is contained within a hypersphere of radius n(E + EN ) centered at the origin. A] such that they are all separated by Aδ .3. within a period of n sample times.

Note that the ratio NEB is the signal power divided by the noise power. or signal to noise ratio (SNR).. you can look at it as a bound on the bandwidth E efficiency as a function of the Nb ratio.2 Final Results P Since energy is power multiplied by time. Figure 71 is the plot on a log-y 0 axis with some of the modulation types discussed this semester.ECE 5520 Fall 2009 141 39. so Rb Rb Eb ≤ log2 1 + B B N0 Defining η = Rb B (the spectral efficiency). i. 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. However.e. we have the Shannon-Hartley Theorem. Be know that our bit rate Rb ≤ C.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. C is just a capacity limit. C = B log2 1 + Rb Eb B N0 Here. C = B log2 1 + P N0 B . η ≤ log2 1 + η Eb N0 This expression can’t analytically be solved for η.3. Thus 0 the capacity bound is also written C = B log2 (1 + SN R). (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 ). That is. Thus from (77). 39. where Rb is the operating bit rate. This relationship is shown in Figure 70. . the energy per bit is the maximum power multiplied by the bit duration. Eb /Tb C = B log2 1 + N0 B or since Rb = 1/Tb . with arbitrarily low probability of error. E = P Ts = 2B where P is the maximum signal power and B is the bandwidth. Eb = P Tb . Any practical communication system must operate at Rb < C. and EN = N0 /2.

η. and M-FSK.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 Shannon-Hartley theorem. 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 Shannon-Hartley theorem bound with achieved bandwidth efficiencies of M-QAM. . M-PSK. bound on bandwidth efficiency.

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