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IET TELECOMMUNICATIONS SERIES 81

Applications of Machine
Learning in Wireless
Communications
Other volumes in this series:
Volume 9 Phase Noise in Signal Sources W.P. Robins
Volume 12 Spread Spectrum in Communications R. Skaug and J.F. Hjelmstad
Volume 13 Advanced Signal Processing D.J. Creasey (Editor)
Volume 19 Telecommunications Traffic, Tariffs and Costs R.E. Farr
Volume 20 An Introduction to Satellite Communications D.I. Dalgleish
Volume 26 Common-Channel Signalling R.J. Manterfield
Volume 28 Very Small Aperture Terminals (VSATs) J.L. Everett (Editor)
Volume 29 ATM: The broadband telecommunications solution L.G. Cuthbert and J.C. Sapanel
Volume 31 Data Communications and Networks, 3rd Edition R.L. Brewster (Editor)
Volume 32 Analogue Optical Fibre Communications B. Wilson, Z. Ghassemlooy and I.Z. Darwazeh
(Editors)
Volume 33 Modern Personal Radio Systems R.C.V. Macario (Editor)
Volume 34 Digital Broadcasting P. Dambacher
Volume 35 Principles of Performance Engineering for Telecommunication and Information
Systems M. Ghanbari, C.J. Hughes, M.C. Sinclair and J.P. Eade
Volume 36 Telecommunication Networks, 2nd Edition J.E. Flood (Editor)
Volume 37 Optical Communication Receiver Design S.B. Alexander
Volume 38 Satellite Communication Systems, 3rd Edition B.G. Evans (Editor)
Volume 40 Spread Spectrum in Mobile Communication O. Berg, T. Berg, J.F. Hjelmstad, S. Haavik
and R. Skaug
Volume 41 World Telecommunications Economics J.J. Wheatley
Volume 43 Telecommunications Signalling R.J. Manterfield
Volume 44 Digital Signal Filtering, Analysis and Restoration J. Jan
Volume 45 Radio Spectrum Management, 2nd Edition D.J. Withers
Volume 46 Intelligent Networks: Principles and applications J.R. Anderson
Volume 47 Local Access Network Technologies P. France
Volume 48 Telecommunications Quality of Service Management A.P. Oodan (Editor)
Volume 49 Standard Codecs: Image compression to advanced video coding M. Ghanbari
Volume 50 Telecommunications Regulation J. Buckley
Volume 51 Security for Mobility C. Mitchell (Editor)
Volume 52 Understanding Telecommunications Networks A. Valdar
Volume 53 Video Compression Systems: From first principles to concatenated codecs A. Bock
Volume 54 Standard Codecs: Image compression to advanced video coding, 3rd Edition
M. Ghanbari
Volume 59 Dynamic Ad Hoc Networks H. Rashvand and H. Chao (Editors)
Volume 60 Understanding Telecommunications Business A. Valdar and I. Morfett
Volume 65 Advances in Body-Centric Wireless Communication: Applications and state-of-the-
art Q.H. Abbasi, M.U. Rehman, K. Qaraqe and A. Alomainy (Editors)
Volume 67 Managing the Internet of Things: Architectures, theories and applications J. Huang
and K. Hua (Editors)
Volume 68 Advanced Relay Technologies in Next Generation Wireless Communications
I. Krikidis and G. Zheng
Volume 69 5G Wireless Technologies A. Alexiou (Editor)
Volume 70 Cloud and Fog Computing in 5G Mobile Networks E. Markakis, G. Mastorakis,
C.X. Mavromoustakis and E. Pallis (Editors)
Volume 71 Understanding Telecommunications Networks, 2nd Edition A. Valdar
Volume 72 Introduction to Digital Wireless Communications Hong-Chuan Yang
Volume 73 Network as a Service for Next Generation Internet Q. Duan and S. Wang (Editors)
Volume 74 Access, Fronthaul and Backhaul Networks for 5G & Beyond M.A. Imran, S.A.R. Zaidi
and M.Z. Shakir (Editors)
Volume 76 Trusted Communications with Physical Layer Security for 5G and Beyond
T.Q. Duong, X. Zhou and H.V. Poor (Editors)
Volume 77 Network Design, Modelling and Performance Evaluation Q. Vien
Volume 78 Principles and Applications of Free Space Optical Communications A.K. Majumdar,
Z. Ghassemlooy, A.A.B. Raj (Editors)
Volume 79 Satellite Communications in the 5G Era S.K. Sharma, S. Chatzinotas and D. Arapoglou
Volume 80 Transceiver and System Design for Digital Communications, 5th Edition Scott
R. Bullock
Volume 905 ISDN Applications in Education and Training R. Mason and P.D. Bacsich
Applications of Machine
Learning in Wireless
Communications
Edited by
Ruisi He and Zhiguo Ding

The Institution of Engineering and Technology

Published by The Institution of Engineering and Technology, London, United Kingdom

The Institution of Engineering and Technology is registered as a Charity in England & Wales
(no. 211014) and Scotland (no. SC038698).

© The Institution of Engineering and Technology 2019

First published 2019

This publication is copyright under the Berne Convention and the Universal Copyright
Convention. All rights reserved. Apart from any fair dealing for the purposes of research
or private study, or criticism or review, as permitted under the Copyright, Designs and
Patents Act 1988, this publication may be reproduced, stored or transmitted, in any
form or by any means, only with the prior permission in writing of the publishers, or in
the case of reprographic reproduction in accordance with the terms of licences issued
by the Copyright Licensing Agency. Enquiries concerning reproduction outside those
terms should be sent to the publisher at the undermentioned address:

The Institution of Engineering and Technology

Michael Faraday House
Six Hills Way, Stevenage
Herts, SG1 2AY, United Kingdom

www.theiet.org

While the authors and publisher believe that the information and guidance given in this
work are correct, all parties must rely upon their own skill and judgement when making
use of them. Neither the authors nor publisher assumes any liability to anyone for any
loss or damage caused by any error or omission in the work, whether such an error or
omission is the result of negligence or any other cause. Any and all such liability
is disclaimed.

The moral rights of the authors to be identified as authors of this work have been
asserted by them in accordance with the Copyright, Designs and Patents Act 1988.

British Library Cataloguing in Publication Data

A catalogue record for this product is available from the British Library

ISBN 978-1-78561-657-0 (hardback)

ISBN 978-1-78561-658-7 (PDF)

Typeset in India by MPS Limited

Printed in the UK by CPI Group (UK) Ltd, Croydon
Contents

Foreword xiii

1 Introduction of machine learning 1

Yangli-ao Geng, Ming Liu, Qingyong Li, and Ruisi He
1.1 Supervised learning 1
1.1.1 k-Nearest neighbours method 2
1.1.2 Decision tree 4
1.1.3 Perceptron 9
1.1.4 Summary of supervised learning 19
1.2 Unsupervised learning 20
1.2.1 k-Means 21
1.2.2 Density-based spatial clustering of applications with noise 23
1.2.3 Clustering by fast search and find of density peaks 24
1.2.4 Relative core merge clustering algorithm 27
1.2.5 Gaussian mixture models and EM algorithm 29
1.2.6 Principal component analysis 34
1.2.7 Autoencoder 37
1.2.8 Summary of unsupervised learning 40
1.3 Reinforcement learning 41
1.3.1 Markov decision process 42
1.3.2 Model-based methods 44
1.3.3 Model-free methods 46
1.3.4 Deep reinforcement learning 50
1.3.5 Summary of reinforcement learning 53
1.4 Summary 56
Acknowledgement 57
References 57

2 Machine-learning-enabled channel modeling 67

Chen Huang, Ruisi He, Andreas F. Molisch, Zhangdui Zhong, and Bo Ai
2.1 Introduction 67
2.2 Propagation scenarios classification 69
2.2.1 Design of input vector 70
2.2.2 Training and adjustment 71
2.3 Machine-learning-based MPC clustering 72
2.3.1 KPowerMeans-based clustering 73
2.3.2 Sparsity-based clustering 76
vi Applications of machine learning in wireless communications

2.3.3 Kernel-power-density-based clustering 78

2.3.4 Time-cluster-spatial-lobe (TCSL)-based clustering 82
2.3.5 Target-recognition-based clustering 82
2.3.6 Improved subtraction for cluster-centroid initialization 84
2.3.7 MR-DMS clustering 86
2.4 Automatic MPC tracking algorithms 89
2.4.1 MCD-based tracking 89
2.4.2 Two-way matching tracking 90
2.4.3 Kalman filter-based tracking 91
2.4.4 Extended Kalman filter-based parameters estimation
and tracking 92
2.4.5 Probability-based tracking 93
2.5 Deep learning-based channel modeling approach 95
2.5.1 BP-based neural network for amplitude modeling 96
2.5.2 Development of neural-network-based channel modeling 96
2.5.3 RBF-based neural network for wireless channel modeling 99
2.5.4 Algorithm improvement based on physical interpretation 101
2.6 Conclusion 103
References 103

3 Channel prediction based on machine-learning algorithms 109

Xue Jiang and Zhimeng Zhong
3.1 Introduction 109
3.2 Channel measurements 110
3.3 Learning-based reconstruction algorithms 111
3.3.1 Batch algorithms 111
3.3.2 Online algorithms 124
3.4 Optimized sampling 126
3.4.1 Active learning 126
3.4.2 Channel prediction results with path-loss measurements 127
3.5 Conclusion 130
References 131

4 Machine-learning-based channel estimation 135

Yue Zhu, Gongpu Wang, and Feifei Gao
4.1 Channel model 137
4.1.1 Channel input and output 138
4.2 Channel estimation in point-to-point systems 139
4.2.1 Estimation of frequency-selective channels 139
4.3 Deep-learning-based channel estimation 140
4.3.1 History of deep learning 140
4.3.2 Deep-learning-based channel estimator for orthogonal
frequency division multiplexing (OFDM) systems 142
Contents vii

4.3.3 Deep learning for massive MIMO CSI feedback 145

4.4 EM-based channel estimator 149
4.4.1 Basic principles of EM algorithm 149
4.4.2 An example of channel estimation with EM algorithm 152
4.5 Conclusion and open problems 156
References 157

5 Signal identification in cognitive radios using machine learning 159

Jingwen Zhang and Fanggang Wang
5.1 Signal identification in cognitive radios 159
5.2 Modulation classification via machine learning 161
5.2.1 Modulation classification in multipath fading channels via
expectation–maximization 162
5.2.2 Continuous phase modulation classification in fading
channels via Baum–Welch algorithm 170
5.3 Specific emitter identification via machine learning 178
5.3.1 System model 179
5.3.2 Feature extraction 181
5.3.3 Identification procedure via SVM 185
5.3.4 Numerical results 189
5.3.5 Conclusions 194
References 195

6 Compressive sensing for wireless sensor networks 197

Wei Chen
6.1 Sparse signal representation 198
6.1.1 Signal representation 198
6.1.2 Representation error 199
6.2 CS and signal recovery 200
6.2.1 CS model 200
6.2.2 Conditions for the equivalent sensing matrix 202
6.2.3 Numerical algorithms for sparse recovery 204
6.3 Optimized sensing matrix design for CS 206
6.3.1 Elad’s method 206
6.3.2 Duarte-Carvajalino and Sapiro’s method 208
6.3.3 Xu et al.’s method 209
6.3.4 Chen et al.’s method 210
6.4 CS-based WSNs 211
6.4.1 Robust data transmission 211
6.4.2 Compressive data gathering 213
6.4.3 Sparse events detection 214
6.4.4 Reduced-dimension multiple access 216
6.4.5 Localization 217
6.5 Summary 218
References 218
viii Applications of machine learning in wireless communications

7.9.2 Effect of increased network density 252

7.9.3 Effect of data rate 254
7.9.4 Effect of multi-hop 255
7.10 Conclusion 256
References 256

8 Machine-learning-based perceptual video coding in wireless

multimedia communications 261
Shengxi Li, Mai Xu, Yufan Liu, and Zhiguo Ding
8.1 Background 261
8.2 Literature review on perceptual video coding 264
8.2.1 Perceptual models 264
8.2.2 Incorporation in video coding 265
8.3 Minimizing perceptual distortion with the RTE method 267
8.3.1 Rate control implementation on HEVC-MSP 267
8.3.2 Optimization formulation on perceptual distortion 269
8.3.3 RTE method for solving the optimization formulation 270
8.3.4 Bit reallocation for maintaining optimization 274
8.4 Computational complexity analysis 275
8.4.1 Theoretical analysis 276
8.4.2 Numerical analysis 278
8.5 Experimental results on single image coding 279
8.5.1 Test and parameter settings 279
8.5.2 Assessment on rate–distortion performance 281
8.5.3 Assessment of BD-rate savings 287
8.5.4 Assessment of control accuracy 289
8.5.5 Generalization test 290
8.6 Experimental results on video coding 292
8.6.1 Experiment 296
8.7 Conclusion 300
References 302

9 Machine-learning-based saliency detection and its video decoding

application in wireless multimedia communications 307
Mai Xu, Lai Jiang, and Zhiguo Ding
9.1 Introduction 307
9.2 Related work on video-saliency detection 310
9.2.1 Heuristic video-saliency detection 310
9.2.2 Data-driven video-saliency detection 311
9.3 Database and analysis 312
9.3.1 Database of eye tracking on raw videos 312
9.3.2 Analysis on our eye-tracking database 313
9.3.3 Observations from our eye-tracking database 315
9.4 HEVC features for saliency detection 317
9.4.1 Basic HEVC features 317
x Applications of machine learning in wireless communications

9.4.2 Temporal difference features in HEVC domain 320

9.4.3 Spatial difference features in HEVC domain 321
9.5 Machine-learning-based video-saliency detection 322
9.5.1 Training algorithm 322
9.5.2 Saliency detection 324
9.6 Experimental results 325
9.6.1 Setting on encoding and training 325
9.6.2 Analysis on parameter selection 326
9.6.3 Evaluation on our database 329
9.6.4 Evaluation on other databases 332
9.6.5 Evaluation on other work conditions 334
9.6.6 Effectiveness of single features and learning algorithm 335
9.7 Conclusion 338
References 338

10 Deep learning for indoor localization based on bimodal CSI data 343
Xuyu Wang and Shiwen Mao
10.1 Introduction 343
10.2 Deep learning for indoor localization 345
10.2.1 Autoencoder neural network 345
10.2.2 Convolutional neural network 346
10.2.3 Long short-term memory 348
10.3 Preliminaries and hypotheses 348
10.3.1 Channel state information preliminaries 348
10.3.2 Distribution of amplitude and phase 349
10.3.3 Hypotheses 350
10.4 The BiLoc system 355
10.4.1 BiLoc system architecture 355
10.4.2 Off-line training for bimodal fingerprint database 356
10.4.3 Online data fusion for position estimation 358
10.5 Experimental study 359
10.5.1 Test configuration 359
10.5.2 Accuracy of location estimation 360
10.5.3 2.4 versus 5 GHz 362
10.5.4 Impact of parameter ρ 362
10.6 Future directions and challenges 364
10.6.1 New deep-learning methods for indoor localization 364
10.6.2 Sensor fusion for indoor localization using
deep learning 364
10.6.3 Secure indoor localization using deep learning 365
10.7 Conclusions 365
Acknowledgments 366
References 366
Contents xi

11 Reinforcement-learning-based wireless resource allocation 371

Rui Wang
11.1 Basics of stochastic approximation 371
11.1.1 Iterative algorithm 372
11.1.2 Stochastic fixed-point problem 373
11.2 Markov decision process: basic theory and applications 376
11.2.1 Basic components of MDP 378
11.2.2 Finite-horizon MDP 381
11.2.3 Infinite-horizon MDP with discounted cost 387
11.2.4 Infinite-horizon MDP with average cost 392
11.3 Reinforcement learning 394
11.3.1 Online solution via stochastic approximation 396
11.3.2 Q-learning 401
11.4 Summary and discussion 404
References 405

12 Q-learning-based power control in small-cell networks 407

Zhicai Zhang, Zhengfu Li, Jianmin Zhang, and Haijun Zhang
12.1 Introduction 407
12.2 System model 411
12.2.1 System description 411
12.2.2 Effective capacity 413
12.2.3 Problem formulation 414
12.3 Noncooperative game theoretic solution 414
12.4 Q-learning algorithm 415
12.4.1 Stackelberg game framework 416
12.4.2 Q-learning 417
12.4.3 Q-learning procedure 418
12.4.4 The proposed BDb-WFQA based on NPCG 420
12.5 Simulation and analysis 422
12.5.1 Simulation for Q-learning based on Stackelberg game 422
12.5.2 Simulation for BDb-WFQA algorithm 424
12.6 Conclusion 426
References 427

13 Data-driven vehicular mobility modeling and prediction 431

Yong Li, Fengli Xu, and Manzoor Ahmed
13.1 Introduction 431
13.2 Related work 434
13.3 Model 435
13.3.1 Data sets and preprocessing 435
13.3.2 Model motivation 436
13.3.3 Queue modeling 437
xii Applications of machine learning in wireless communications

13.4 Performance derivation 439

13.4.1 Vehicular distribution 440
13.4.2 Average sojourn time 441
13.4.3 Average mobility length 443
13.5 Model validation 443
13.5.1 Time selection and area partition 443
13.5.2 Arrival rate validation 445
13.5.3 Vehicular distribution 447
13.5.4 Average sojourn time and mobility length 449
13.6 Applications of networking 451
13.6.1 RSU capacity decision 452
13.6.2 V2I and V2V combined performance analysis 453
13.7 Conclusions 457
References 457

Index 461
Foreword

The technologies of wireless communications have been changed drastically in recent

years. The rapidly growing wave of wireless data is pushing against the boundary
of wireless communication system’s performance. Such pervasive and exponentially
increasing data present imminent challenges to all aspects of wireless communication
system’s design, and the future wireless communications will require robust intelligent
algorithms for different services in different scenarios. Contributions are needed
from multidisciplinary fields to enhance wireless system, such as computer science,
mathematics, control and many other science disciplines. The combined efforts from
scientists from different disciplines are important for the success of the wireless
communication industry.
In such an era of big data where data mining and data analysis technologies are
effective approaches for wireless system evaluation and design, the applications of
machine learning in wireless communications have received a lot of attention recently.
Machine learning provides feasible and new solutions for the complex wireless com-
munication system design. It has been a powerful tool and popular research topic with
many potential applications to enhance wireless communications, e.g. radio channel
modelling, channel estimation and signal detection, network management and per-
formance improvement, access control, resource allocation. However, most of the
current researches are separated into different fields and have not been well orga-
nized and presented yet. It is therefore difficult for academic and industrial groups
to see the potentialities of using machine learning in wireless communications. It is
now appropriate to present a detailed guidance of how to combine the disciplines of
wireless communications and machine learning.
In this book, present and future developments and trends of wireless communica-
tion technologies are depicted based on contributions from machine learning and other
fields in artificial intelligence. The prime focus of this book is given in the physical
layer and network layer with a special emphasis on machine-learning projects that are
(or are close to) achieving improvements in wireless communications. A wide vari-
ety of research results are merged together to make this book useful for students and
researchers. There are 13 chapters in this book, and we have organized them as follows:
● In Chapter 1, an overview of machine-learning algorithms and their applications
are presented to provide advice and references to fundamental concepts accessible
to the broad community of wireless communication practitioners. Specifically,
the materials are organized into three sections following the three main branches
of machine learning: supervised learning, unsupervised learning and reinforce-
ment learning (RL). Each section starts with an overview to illustrate the major
xiv Applications of machine learning in wireless communications

concerns and ideas of this branch. Then, classic algorithms and their last develop-
ments are reviewed with typical applications and useful references. Furthermore,
pseudocodes are added to provide interpretations and details of algorithms. Each
section ends by a summary in which the structure of this section is untangled and
relevant applications in wireless communication are given.
● In Chapter 2, using machine learning in wireless channel modelling is presented.
First of all, the background of the machine-learning-enabled channel modelling is
introduced. Then, four related aspects are presented: (i) propagation scenario clas-
sification, (ii) machine-learning-based multipath component (MPC) clustering,
(iii) automatic MPC tracking and (iv) deep-learning-based channel modelling.
The results in this chapter can provide references to other real-world measurement
data-based channel modelling.
● In Chapter 3, the wireless channel prediction is addressed, which is a key issue
for wireless communication network planning and operation. Instead of the
classic model-based methods, a survey of recent advances in machine-learning
technique-based channel prediction algorithms is provided, including both batch
and online methods. Experimental results are provided using the real data.
● In Chapter 4, new types of channel estimators based on machine learning are
introduced, which are different from traditional pilot-aided channel estimators
such as least squares and linear minimum mean square errors. Specifically, two
newly designed channel estimators based on deep learning and one blind estimator
based on expectation maximization algorithm are provided for wireless commu-
nication systems. The challenges and open problems for channel estimation aided
by machine-learning theories are also suggested.
● In Chapter 5, cognitive radio is introduced as a promising paradigm to solve the
spectrum scarcity and to improve the energy efficiency of the next generation
mobile communication network. In the context of cognitive radios, the necessity
of using signal identification techniques is first presented. A survey of signal
identification techniques and recent advances in this field using machine learn-
ing are then provided. Finally, open problems and possible future directions for
cognitive radio are briefly discussed.
● In Chapter 6, the fundamental concepts that are important in the study of com-
pressive sensing (CS) are introduced. Three conditions are described, i.e. the null
space property, the restricted isometry property and mutual coherence, that are
used to evaluate the quality of sensing matrices and to demonstrate the feasibility
of reconstruction. Some widely used numerical algorithms for sparse recovery are
briefly reviewed, which are classified into two categories, i.e. convex optimiza-
tion algorithms and greedy algorithms. Various examples are illustrated where
the CS principle has been applied to WSNs.
● In Chapter 7, the enhancement of the proposed IEEE 802.11p Medium Access
Control (MAC) layer is studied for vehicular use by applying RL. The purpose
of this adaptive channel access control technique is enabling more reliable, high-
throughput data exchanges among moving vehicles for cooperative awareness
purposes. Some technical background for vehicular networks is presented, as
well as some relevant existing solutions tackling similar channel sharing prob-
lems. Finally, some new findings from combining the IEEE 802.11p MAC with
Foreword xv

RL-based adaptation and insight of the various challenges appearing when

applying such mechanisms in a wireless vehicular network are presented.
● In Chapter 8, the advantage of applying machine-learning-based perceptual
coding strategies in relieving bandwidth limitation is presented for wireless
multimedia communications. Typical video coding standards, especially the state-
of-the-art high efficiency video coding (HEVC) standard, as well as recent
research progress on perceptual video coding, are included. An example that
minimizes the overall perceptual distortion is further demonstrated by mod-
elling subjective quality with machine-learning-based saliency detection. Several
promising directions in learning-based perceptual video coding are presented to
further enhance wireless multimedia communication experience.
● In Chapter 9, it is argued that the state-of-the-art HEVC standard can be used for
saliency detection to generate the useful features in compressed domain. There-
fore, this chapter proposes to learn the video saliency model, with regard to HEVC
features. First, an eye-tracking database is established for video saliency detec-
tion. Through the statistical analysis on our eye-tracking database, we find out
that human fixations tend to fall into the regions with large-valued HEVC fea-
tures on splitting depth, bit allocation and motion vector (MV). In addition, three
observations are obtained with the further analysis on our eye-tracking database.
Accordingly, several features in HEVC domain are proposed on the basis of split-
ting depth, bit allocation and MV. Next, a kind of support vector machine is
learned to integrate those HEVC features together, for video saliency detection.
Since almost all video data are stored in the compressed form, the proposed
method is able to avoid both the computational cost on decoding and the storage
cost on raw data. More importantly, experimental results show that the proposed
method is superior to other state-of-the-art saliency detection methods, either in
compressed or uncompressed domain.
● In Chapter 10, deep learning is incorporated for indoor localization based on
channel state information (CSI) with commodity 5GHz Wi-Fi. The state-of-the-
art deep-learning techniques are first introduced, including deep autoencoder
networks, convolutional neural networks and recurrent neural networks. The CSI
preliminaries and three hypotheses are further introduced, which are validated
with experiments. Then a deep-learning-based algorithm is presented to leverage
bimodal CSI data, i.e. average amplitudes and estimated angle of arrivals, in both
offline and online stages of fingerprinting. The proposed scheme is validated with
extensive experiments. Finally, several open research problems are examined for
indoor localization based on deep-learning techniques.
● In Chapter 11, the reinforcement-learning-based wireless resource allocation is
presented. First the basic principle of stochastic approximation is introduced,
which is the basis of the RL. Then how to formulate the wireless resource alloca-
tion problems via three forms of Markov decision process (MDP) is demonstrated,
respectively, namely finite-horizon MDP, infinite-horizon MDP with discount
cost and infinite-horizon MDP with average cost. One of the key knowledge to
solve MDP problem is the system state transition probability, which might be
unknown in practice. Hence, finally it is shown that when some system statistics
are unknown, the MDP problems can still be solved via the method of RL.
xvi Applications of machine learning in wireless communications

● In Chapter 12, by integrating information theory with the principle of effective

capacity, an energy efficiency optimization problem is formulated with statistical
QoS guarantee in the uplink of two-tier small cell networks. To solve the problem,
a Q-learning mechanism based on Stackelberg game framework is introduced, in
which a macro-user acts as a leader, and knows all small-cell-users’ transmit
power strategies, while the small-cell-users are followers and only communi-
cate with the microcell base station not with other small-cell base stations. In
a formulated Stackelberg game procedure, the macro-user selects the transmit
power level based on the best responses of the small-cell-users. Then those small-
cell-users find their best responses. And in order to improve the self-organizing
ability of femtocell, based on the non-cooperative game framework, a Boltzmann
distribution-based weighted filter Q-learning algorithm (BDB-WFQA) based on
Boltzmann distribution is proposed to realize power allocation. The simulation
results show the proposed distributed Q-learning algorithm has a better perfor-
mance in terms of convergence speed while providing delay QoS provisioning.
The proposed BDB-WFQA algorithm increases the achievable effective capac-
ity of macro-users and a better performance compared with other power-control
algorithm.
● In Chapter 13, the open Jackson queuing network models are used to model
the macroscopic level vehicular mobility. The proposed simple model can accu-
rately describe the vehicular mobility and then further predict various measures
of network-level performance like the vehicular distribution, and vehicular-level
performance like average sojourn time in each area and the number of sojourned
areas in the vehicular networks. Model validation based on two large-scale urban
city vehicular motion traces reveals that such a simple model can accurately pre-
dict a number of system metrics interested by the vehicular network performance.
Moreover, two applications are proposed to illustrate the proposed model is effec-
tive in the analysis of system-level performance and dimensioning for vehicular
networks.
The goal of this book is to help communications system designers gain an
overview of the pertinent applications of machine learning in wireless communi-
cations, and for researchers to assess where the most pressing needs for further work
lie. This book can also be used as a textbook for the courses dedicated for machine-
learning-enabled wireless communications. With contributions from an international
panel of leading researchers, this book will find a place on the bookshelves of
academic and industrial researchers and advanced students working in wireless com-
munications and machine learning. We hope that the above contributions will form an
interesting and useful compendium on applications of machine learning in wireless
communications.
Prof. Ruisi He
State Key Laboratory of Rail Traffic Control and Safety
Beijing Jiaotong University, China
and
Prof. Zhiguo Ding
School of Electrical and Electronic Engineering
The University of Manchester, UK
Chapter 1
Introduction of machine learning
Yangli-ao Geng1 , Ming Liu1 , Qingyong Li1 , and Ruisi He2

Machine learning, as a subfield of artificial intelligence, is a category of algorithms

that allow computers to learn knowledge from examples and experience (data), with-
out being explicitly programmed [1]. Machine-learning algorithms can find natural
patterns hidden in massive complex data, which humans can hardly deal with manu-
ally. The past two decades have witnessed tremendous growth in big data, which makes
machine learning become a key technique for solving problems in many areas such
as computer vision, computational finance, computational biology, business deci-
sion, automotive and natural language processing (NLP). Furthermore, our life has
been significantly improved by various technologies based on machine learning [2].
Facial-recognition technology allows social media platforms to help users tag and
share photos of friends. Optical character recognition technology converts images
of text into movable type. Recommendation systems, powered by machine learning,
suggest what films or television shows to watch next based on user preferences. Infor-
mation retrieval technology supports a search engine to return most related records
after users input some keywords. NLP technology makes it possible to filter out spam
from massive e-mails automatically. Self-driving cars that rely on machine learning
to navigate are around the corner to consumers.
In wireless communications, when you encounter a complex task or problem
involving a large amount of data and lots of variables, but without existing formula or
equation, machine learning can be a solution. Traditionally, machine-learning algo-
rithms can be roughly divided into three categories: supervised learning, unsupervised
learning and reinforcement learning (RL). In this chapter, we present an overview of
machine-learning algorithms and list their applications, with a goal of providing use-
ful advice and references to fundamental concepts accessible to the broad community
of wireless communications practitioners.

1.1 Supervised learning

Let us begin with an example to explain the basic idea of supervised learning. Imag-
ine that you are a weatherman and have access to historical meteorological data

1
School of Computer and Information Technology, Beijing Jiaotong University, China
2
State Key Laboratory of Rail Traffic Control and Safety, Beijing Jiaotong University, China
2 Applications of machine learning in wireless communications

(e.g. temperature, wind speed and precipitation for the past days). Now, given today’s
meteorological data, how to predict the weather of the next day? A natural idea is
to explore a rule from the historical meteorological data. Specifically, you need to
observe and analyse what the weather was under the meteorological data of the last
day. If you are fortunate enough to find a rule, then you will make a successful pre-
diction. However, in most cases, the meteorological data is too big to analyse for
humans. Supervised learning would be a solution to this challenge.
In fact, what you try to do in the above example is a typical supervised learning
task. Formally, supervised learning is a procedure of learning a function f (·) that maps
an input x (meteorological data of a day) to an output y (weather of the next day) based
on a set of sample pairs T = {(xi , yi )}ni=1 (historical data), where T is called a training
set and yi is called a label. If y is a categorical variable (e.g. sunny or rainy), then the
task is called a classification task. If y is a continuous variable (e.g. probability of
precipitation), then the task is called a regression task. Furthermore, for a new input
x0 , which is called a test sample, f (x0 ) will give the prediction.
In wireless communications, an important problem is estimating the channel
noise in a MIMO wireless network, since knowing these parameters are essential
to many tasks of a wireless network such as network management, event detection,
location-based service and routing [3]. This problem can be solved by using supervised
learning approaches. Let us consider the circumstances for the linear channel with
white added Gaussian noise MIMO environments with t transmitting antennas and r
receiving antennas. Assume the channel model is z = Hs + u, where s ∈ Rt , u ∈ Rr
and z ∈ Rr denote signal vector, noise vector and received vector, respectively. The
goal in the channel noise estimation problem is to estimate u given s and z. This
problem can be formulated as r regression tasks, and the target of the kth regression
task is to predict uk for 1 ≤ k ≤ r. In the kth regression task, a training pair is
represented as {[sT , zk ]T , uk }. We can complete these tasks using any regression model
(will be introduced later in this chapter). Once the model is well trained, uk can be
predicted when a new sample [s̄T , z̄k ]T comes. In this section, we will discuss three
practical technologies of supervised learning.

1.1.1 k-Nearest neighbours method

The k-nearest neighbours (k-NNs) method is a basic supervised learning method
which is applicable to both classification and regression tasks. Here, we will focus
on the classification since the regression shares similar steps with the classification.
Given a training set T = {(xi , yi )}ni=1 and a test sample x0 , the task is to predict the
category of x0 under the instruction of T . The main idea of k-NN is that first search
k-NNs of x0 in the training set and then classify x0 into the category which is most
common among the k-NNs (the majority principle). Particularly, if k = 1, x0 is simply
assigned to the class of its nearest neighbour.
Figure 1.1(a) shows an illustration for the main idea of k-NN. From Figure 1.1(a),
we observe that there are seven samples in the training set, four of which are labelled
as the first class (denoted by squares) and the others are labelled as the second
class (denoted by triangles). We intend to predict the category of a test sample
Introduction of machine learning 3

(denoted by a circle) using the k-NN method. When k = 3, as shown in Figure 1.1(b),
the test sample will be assigned to the first class according to the majority principle.
When k = 1, as shown in Figure 1.1(c), the test sample will be assigned to the second
class since its nearest neighbour belongs to the second class. A formal description of
k-NN is presented in Algorithm 1.1.
The output of the k-NN algorithm is related to two things. One is the distance
function, which measures how near two samples are. Different distance functions will
lead to different k-NN sets and thus different classification results. The most com-
monly used distance function is the Lp distance. Given two vectors x = (x1 , . . . , xd )T
and z = (z1 , . . . , zd )T , the Lp distance between them is defined as
d 1/p

Lp (x, z) = |xi − zi |p
. (1.1)
i=1

(a) (b) (c)

Figure 1.1 An illustration for the main idea of k-NN. (a) A training set there
consists of seven samples, four of which are labelled as the first class
(denoted by squares) and the others are labelled as the second class
(denoted by triangles). A test sample is denoted as a circle. (b) When
k = 3, the test sample is classified as the first class. (c) When k = 1,
the test sample is assigned as the second class

Algorithm 1.1: k-NN method

Input: number of neighbours k, training set T = {(xi , yi )}ni=1 , test sample x0
Output: label of test sample y0
1 Find the k-NN set of x in T and denote the set by Nk (x0 );
2 Determine the label of x according to the majority principle, i.e.

y0 = arg max I (yi = c),
1≤c≤m
xi ∈Nk (x0 )

where I (yi = c) equals 1 if yi = c and 0 otherwise;

4 Applications of machine learning in wireless communications

When p equals 2, the Lp distance becomes the Euclidean distance. When p equals 1,
the Lp distance is also called the Manhattan distance. When p goes to ∞, it can be
shown that
L∞ (x, z) = max |xi − zi |. (1.2)
i

Another useful distance is the angular distance, which is defined as

arccos xT z/(xz)
DA (x, z) = . (1.3)
π
As its name suggests, the angular distance measures the included angle between two
vectors, and thus it is independent of the length of the vectors. This property makes
the angular distance useful in the situation that we only concern the proportion of each
component of features. Readers can refer to [4] for more information about distances.
The other factor affecting the result of the algorithm is the value of k. As shown
in Figure 1.1, different values of k may lead to different results. The best choice of
k depends upon the data. Generally, smaller values of k can generate more accurate
result for a high-quality training set, but it is sensitive to noises. In other words, the
output for a test sample may be severely affected by the noise samples near to it. In
contrast, larger values of k reduce the effect of noise on the classification but make
boundaries between classes less distinct [5]. In practice, one popular way of choosing
the empirically optimal k is via cross validation [6].
The k-NN algorithm is easy to implement by computing the distance between
the test sample and all training samples, but it is computationally intensive for a
big training set. The acceleration strategy for searching k-NNs can be found in [7].
Some theoretical results about k-NN have been presented in [8,9]. References [10,11]
demonstrate two applications of the k-NN method to fall detection via wireless sensor
network data and energy enhancements for smart mobile devices, respectively.

1.1.2 Decision tree

Decision tree is a supervised learning model based on a tree structure, which is used
for both classification and regression tasks. It is one of the most popular models
in supervised learning due to its effectiveness and strong interpretability. As shown
in Figure 1.2, a decision tree consists of three parts: internal nodes, leaf nodes and
branches. Among them, each internal node defines a set of if–then rules; each leaf
node defines a category (or a target value for a regression task), and branches deter-
mine the topology structure of the tree. To predict the category (or target value) for a
given test sample, we should find a path from the root node to a leaf node following
the steps below. Starting from the root node, chose a branch according to that which
rules the test sample meet in each internal node. Then go to the next node along the
branch. Repeat the above two steps until arriving at a leaf node, then the category
(target value) is given by this leaf node.
For a given training set {(xi , yi )}ni=1 , we say a decision tree affirms it if the tree out-
puts a correct prediction yi for any xi (i = 1, . . . , n). Given a training set, there may
exist tremendous trees affirming it. However, only a few of them will achieve good
performance on test samples (we call these trees effective trees), but we cannot afford
Introduction of machine learning 5

Internal node

Leaf node

Figure 1.2 Decision tree structure

to enumerate all trees to find an effective tree. Thus, the key problem is how to con-
struct an effective tree in a reasonable span of time. A variety of methods have been
developed for learning an effective tree, such as ID3 [12], C4.5 [13] and classifica-
tion and regression tree (CART) [14]. Most of them share a similar core idea that
employs a top-down, greedy strategy to search through the space of possible decision
trees. In this section, we will focus on the common-used CART method and its two
improvements, random forest (RF) and gradient boosting decision tree (GBDT).

1.1.2.1 Classification and regression tree

CART [14] is a recursive partitioning method to build a classification or regression
tree. Different from other methods, CART constraints the tree as a binary tree, which
means there are only two branches in an internal node. In each internal node, a test
sample will go down the left or right branch according to whether it meets the rule
defined in the node or not.
Constructing CARTs share a similar process. The only difference between them
is the partition criterion in each internal node. For a classification tree, the partition
criterion is to minimize the Gini coefficient. Specifically, given a training set T of n
samples and k categories, with ni samples in the ith category, the Gini coefficient of
T is defined as
ni ni ni 2
k k
Gini(T ) = 1− =1− , (1.4)
i=1
n n i=1
n

where n = ki=1 ni . In the root node, CART will find a partition rule to divide a
training set T into two partitions, say T1 and T2 , which minimizes the following
function:
|T1 | |T2 |
Gini(|T1 |) + Gini(T2 ). (1.5)
|T | |T |
6 Applications of machine learning in wireless communications

The similar steps will be carried out recursively for T1 and T2 , respectively, until a
stop condition meets.
In contrast, a regression tree is to predict continuous variables and its partition
criteria is usually chosen as the minimum mean square error. Specifically, given a
training set T = {(xi , yi )}ni=1 , a regression tree will divide T into T1 and T2 such that
the following equation is minimized:
2
(yi − m1 )2 + yj − m 2 , (1.6)
(xi ,yi )∈T1 (xj ,yj )∈T2

where mj = (1/|Tj |) (xi ,yi )∈Tj yi ( j = 1, 2). For clarity, we summarize the construct-
ing process and the predicting process in Algorithms 1.2 and 1.3, respectively.
By using Algorithm 1.2, we can construct a decision tree. However, this tree is so
fine that it may cause overfitting (i.e. it achieves perfect performance on a training set
but bad predictions for test samples). An extra pruning step can improve this situation.
The pruning step consists of two main phases. First, iteratively prune the tree from the
leaf nodes to the root node and thus acquire a tree sequence Tree0 , Tree1 , . . . , Treen ,
where Tree0 denotes the entire tree and Treen denotes the tree which only contain

Algorithm 1.2: CART constructing tree

Input: training set T = {(xi , yi ) ∈ Rd+1 }ni=1 , stop number s

1 if |T | < s then
2 node.left = NULL, node.right = NULL;
3 if the task is classification then
4 set node.y as the category which is most common among T ;
5 if the task is regression
then
6 node.y = |T1 | yi ∈T yi
7 return node;
8 v̂ = ∞;
9 for j = 1 to d do
10 if the task is classification then
11 v̄, p̄ = minp |T|T
1 (p)|
|
Gini(|T1 (p)|) + |T2 (p)|
|T |
Gini(T2 (p));
12 if the task is regression then
2
13 v̄, p̄ = minp (xi ,yi )∈T1 (p) (yi − m1 )2 + (xj ,yj )∈T2 (p) yj − m2 ;
14 where T1 (p) = {(xi , yi )|xi [j] ≤ p}, T2 (p) = {(xi , yi )|xi [j] > p};
15 if v̄ < v̂ then
16 v̂ = v̄, node.dim = j, node.p = p̄;
17 node.left = Constructing Tree(T1 (node.p));
18 node.right = Constructing Tree(T2 (node.p));
19 return node;
Introduction of machine learning 7

Algorithm 1.3: CART predicting

Input: test sample x0 , root node node
Output: prediction y0
1 if node.left = NULL and node.right = NULL then
2 return node.y;
3 if x0 [node.dim] ≤ node.p then
4 return Predicting(x0 , node.left);
5 else
6 return Predicting(x0 , node.right);

the root node. Second, select the optimal tree from the sequence by using the cross
validation. For more details, readers can refer to [14]. References [15–17] demonstrate
three applications of CART in wireless communications.

1.1.2.2 Random forest

As discussed in Section 1.1.2.1, a tree constructed by the CART method has a risk
of overfitting. To meet this challenge, Breiman proposed the RF model in [18]. As
its name suggests, RF consists of many trees and introduces a random step in its
constructing process to prevent overfitting.
Suppose we are given a training set T . To construct a tree Treej , RF first generates
a training subset Tj by sampling from T uniformly and with replacement (Tj has
the same size with T ). Then, a construction algorithm will be carried out on Tj .
The construction algorithm is similar to the CART method but introduces an extra
random step. Specifically, in each internal node, CART chooses the optimal feature
from all d features, but RF first randomly select l features from all d features and
then chooses the optimal feature from the l features. The above construction process
will be repeated m times and thus a forest (which contains m trees) is constructed.
For a classification task, the output is determined by taking the majority vote in m
trees. For a regression task, the output is the mean of m outputs. The whole process
is summarized in Algorithm 1.4. In wireless communications, RF has been applied
in many fields, such as indoor localization [19] and device-free fall detection [20].

1.1.2.3 Gradient boosting decision tree

GBDT [21] is a special case of the famous boosting method [22] based on a tree
structure. Specifically, the model of GBDT is represented as a sum of some CART
trees, i.e.:

m
fm (x) = Treej (x; j ), (1.7)
j=1
8 Applications of machine learning in wireless communications

Algorithm 1.4: Random forest

Input: training set T = {(xi , yi ) ∈ Rd+1 }ni=1 , number of trees m, number of

categories k, test sample x0
Output: prediction y0
/* training */
1 for j = 1, . . . , m do
2 Tj ← ∅;
3 for i = 1, . . . , n do
4 randomly select a train sample (x, y) from T ;
5 Tj ← Tj ∪ {(x, y)};
6 based on Tj , construct a decision tree Treej using randomized CART;
/* testing */
7 if the task is classification then
m
8 y0 = arg max I (Treej (x0 ) = c);
1≤c≤k j=1

9 if the task is regression then

m
10 y0 = m1 Treej (x0 );
j=1

where Treej (x; j ) denotes the jth tree with parameter of j . Given a training set
{(x1 , y1 ), . . . , (xn , yn )}, the goal of GBDT is to minimize:

n
L( fm (xi ), yi ), (1.8)
i=1

where L(·, ·) is a differentiable function which measures the difference between fm (xi )
and yi and is chosen according to the task.
However, it is often difficult to find an optimal solution to minimize (1.8). As a
trade-off, GBDT uses a greedy strategy to yield an approximate solution. First, notice
that (1.7) can be written as a recursive form:

fj (x) = fj−1 (x) + Treej (x; j ) ( j = 1, . . . , m), (1.9)

where we have defined f0 (x) = 0. Then, by fixing the parameters of fj−1 , GBDT finds
the parameter set j by solving:

n
min L( fj−1 (xi ) + Treej (xi ; j ), yi ). (1.10)
j
i=1
Introduction of machine learning 9

Replacing the loss function L(u, v) by its first-order Taylor series approximation with
respect to u at u = fj−1 (xi ), we have

n
L( fj−1 (xi ) + Treej (xi ; j ), yi )
i=1
n
(1.11)
∂L( fj−1 (xi ), yi )
≈ L( fj−1 (xi ), yi ) + Tree(xi ; j ) .
i=1
∂fj−1 (xi )
Notice that the right side is a linear function with respect to Tree(xi ; j ) and its value
can decrease by letting Tree(xi ; j ) = −(∂L( fj−1 (xi ), yi )/∂fj−1 (xi )). Thus, GBDT n
trains Tree(·, j ) by using a new training set (xi , −(∂L( fj−1 (xi ), yi )/(∂fj−1 (xi )))) i=1 .
The above steps will be repeated for j = 1, . . . , m and thus a gradient boosting tree is
generated.
GBDT is known as one of the best methods in supervised learning and has been
widely applied in many tasks. There are many tricks in its implementation. Two
popular implementations, XGboost and LightGBM, can be found in [23] and [24],
respectively. References [25] and [26] demonstrate two applications of GBDT in
obstacle detection and quality of experience (QoE) prediction, respectively.

1.1.3 Perceptron
A perceptron is a linear model for a binary-classification task and is the foundation
of the famous support vector machine (SVM) and deep neural networks (DNNs).
Intuitively, it tries to find a hyperplane to separate the input space (feature space)
into two half-spaces such that the samples of different classes lie in the different half-
spaces. An illustration is shown in Figure 1.3(a). A hyperplane in Rd can be described
by an equation wT x + b = 0, where w ∈ RD is the normal vector. Correspondingly,

2 Hyperplane Positive sample

w Tx + b = 0 Negative sample

1.5 x1
w1

x2 w2
1 wT x + b > 0 w Tx + b

wTx + b < 0 wd
0.5 xd
b

0 +1
0 0.5 1 1.5 2
(a) (b)

Figure 1.3 (a) An illustration of a perceptron and (b) the graph representation of a
perceptron
10 Applications of machine learning in wireless communications

Algorithm 1.5: Perceptron learning

Input: training set {(x1 , y1 ), . . . , (xn , yn )}, learning rate η ∈ (0, 1]
Output: parameters of perceptron w, b
1 randomly initialize w, b;
2 flag = True;
3 while flag do
4 flag = False;
5 for i = 1, . . . , n do
6 if yi (wT xi + b) < 0 then
7 w = w + ηyi xi ;
8 b = b + ηyi ;
9 flag = True;

wT x + b > 0 and wT x + b < 0 represent the two half-spaces separated by the hyper-
plane wT x + b = 0. For a sample x0 , if wT x0 + b is larger than 0, we say x0 is in the
positive direction of the hyperplane, and if wT x0 + b is less than 0, we say it is in the
negative direction.
In addition, by writing wT x + b = [xT , 1] · [wT , b]T = di=1 xi wi + b, we can
view [xT , 1]T , [wT , b]T and wT x + b as the inputs, parameters and output of a per-
ceptron, respectively. Their relation can be described by a graph, where the inputs
and output are represented by nodes, and the parameters are represented by edges,
as shown in Figure 1.3(b). This graph representation is convenient for describing the
multilayer perceptron (neural networks) which will be introduced in Section 1.1.3.3.
Suppose we have a training set T = {(x1 , y1 ), . . . , (xn , yn )}, where xi ∈ Rd and
yi ∈ {+1, −1} is the ground truth. The perceptron algorithm can be formulated as

N
min L(w, b) − yi (wT xi + b), (1.12)
w∈Rd ,b∈R
i=1

where wT x + b = 0 is the classification hyperplane, and yi (wT xi + b) > 0 implies

that the ith sample lie in the correct half-space. Generally, the stochastic gradient
descent algorithm is used to obtain a solution to (1.12) and its convergence has been
shown in [27]. The learning process is summarized in Algorithm 1.5.

1.1.3.1 Support vector machine

SVM is a binary-classification model. SVM shares a similar idea with the perceptron
model, i.e. find a hyperplane to separate two classes of training samples. In general,
there may be several hyperplanes meeting the requirement. A perceptron finds any
one of them as the classification hyperplane. In contrast, SVM will seek the one that
maximizes the classification margin, which is defined as the distance from the hyper-
plane to the nearest training sample. As shown in Figure 1.4(a), three hyperplanes
which can separate the two classes of training samples are drawn in three different
Introduction of machine learning 11

2 2
Positive sample Positive sample
Negative sample Negative sample

1.5 1.5
Classification
margin
1 1

0.5 0.5

0 0
0 0.5 1 1.5 2 0 0.5 1 1.5 2
(a) (b)

Figure 1.4 (a) Three hyperplanes which can separate the two classes of training
samples and (b) the hyperplane which maximizes the classification
margin

styles, and each of them can serve as a solution to the perceptron. However, as shown
in Figure 1.4(b), only the hyperplane which maximizes the classification margin can
serve as the solution to SVM.
According to the definition of the classification margin, distance from any train-
ing sample to the classification hyperplane should be not less than it. Thus, given
the training set {(xi , yi )}ni=1 , the learning of SVM can be formulated as the following
optimization problem:
max γ
w∈Rd ,b∈R
(1.13)
w T xi b
s.t. yi + ≥ γ (i = 1, . . . , n)
w w
T
where (w xi /w) + (b/w) can be viewed as the signed distance from xi to the
classification hyperplane wT x + b = 0, and the sign of yi (wT xi /w) + (b/w)
denotes whether xi lies in the right half-space. It can be shown that problem (1.13)
is equivalent to
1
min w2
w∈Rd ,b∈R 2 (1.14)

s.t. yi wT xi + b − 1 ≥ 0 (i = 1, . . . , n).
Problem (1.14) is a quadratic programming problem [28] and can be efficiently solved
by several optimization tools [29,30].
Note that both the perceptron and SVM suppose that the training set can be
separated linearly. However, this supposition is not always correct. Correspondingly,
the soft-margin hyperplane and the kernel trick have been introduced to deal with
12 Applications of machine learning in wireless communications

the non-linear situation. Please refer to [31] and [32] for more details. In addi-
tion, SVM can also be used to handle regression tasks, which is also known as
support vector regression [33]. SVM has been widely applied in many fields of wire-
less communications, such as superimposed transmission mode identification [34],
selective forwarding attacks detection [35], localization [36] and MIMO channel
learning [3].

1.1.3.2 Logistic regression

First of all, we should clarify that logistic regression is a classification model and
its name is borrowed from the field of statistics. Logistic regression can be binomial
or multinomial depending on whether the classification task is binary or multi-class.
We will introduce the binomial case followed by the multinomial case.
Recall that in a perceptron, a hyperplane wT x + b = 0 is learned to discriminate
the two classes. A test sample x0 will be classified as positive or negative according
to whether wT x0 + b is larger than 0 or wT x0 + b is less than 0. Sometimes, however,
we want to know the probability of a test sample belonging to a class. In the example
shown in Figure 1.5(a), though both x1 and x2 are classified as the positive class, we
are more confident that x2 is the positive sample since x2 is farther from the decision
hyperplane than x1 (i.e. wT x2 + b is larger than wT x1 + b). The sample x3 is classified
as the negative sample since wT x3 + b is less than 0.
From the above example, we can infer that the sign of wT x + b decides the class
of x, and |wT x + b| gives the confidence level of the decision. However, the value
of wT x + b can be any one of (−∞, ∞) but we want a probability value. A nature
idea is that find a monotone increasing function g : (−∞, ∞) → (0, 1) such that

2 1

x2 0.8
1.5 Distance
g (wTx + b)

wTx + b = 0 0.6
1 (0,0.5)
x1
0.4
x3
0.5
0.2

0 0
0 0.5 1 1.5 2 –2 –1 0 1 2
(a) (b) w Tx + b

Figure 1.5 (a) A decision hyperplane wT x + b = 0 and three samples: x1 (blue

square), x2 (red triangle) and x3 (purple cross). (b) The graph of
function g(t) (exp(t)/(1 + exp(t))) and the probability of x1 , x2 and x3
belong to the positive class
Introduction of machine learning 13

g(wT x + b) represents the probability of x belonging to the positive class. Thus, by

choosing g(t) (exp (t)/(1 + exp (t))), we get the model of logistic regression:

exp wT x + b
P{ y = 1|x} = . (1.15)
1 + exp wT x + b
P{ y = 1|x} denotes the probability of x belonging to the positive class, and corre-
spondingly 1 − P{ y = 1|x} is the probability of x belonging to the negative class. As
shown in Figure 1.5(b), by the above transformation, the probability of x1 , x2 and x3
belonging to the positive class are around 0.6, 0.85 and 0.2, respectively.
Given a training set {(xi , yi )}ni=1 , the parameters w and b of the binomial
logistic regression model can be estimated by the maximum likelihood estimation.
Specifically, its log-likelihood is written as
⎛ ⎞
exp wT xi + b exp wT x + b
L(w, b) = log ⎝ · 1− ⎠
yi =1
1 + exp w Tx + b
i yi =−1
1 + exp w Tx + b

(1.16)

n

= w T xi + b − log 1 + exp wT xi + b .
yi =1 i=1

Because there is no closed-form solution to the problem of maximizing (1.16), the

gradient descent method and the quasi-Newton method are generally used to obtain
a numerical solution.
Following the binomial logistic regression, we can deduce the multinomial case.
For a k-classification task, the multinomial logistic regression model are given by

exp wjT x + bj
P( y = j|x) = k T ( j = 1, . . . , k), (1.17)
i=1 exp wi x + bi

where P( y = j|x) denotes the probability of x belonging to the jth class. The parameter
set {(wi , bi )}kj=1 can also be estimated by using the maximum likelihood estimation.
Another name of multinomial logistic regression is softmax regression, which is often
used as the last layer of a multilayer perceptron that will be introduced in the next
section. Logistic regression has been applied to predict device wireless data and
location interface configurations that can optimize energy consumption in mobile
devices [11]. References [37], [38] and [39] demonstrate three applications of logis-
tic regression to home wireless security, reliability evaluation and patient anomaly
detection in medical wireless sensor networks, respectively.

1.1.3.3 Multilayer perceptron and deep learning

A multilayer perceptron is also known as a multilayer neural network, which is also
called a DNN when the number of layers is large enough. As shown in Figure 1.6(a), a
multilayer perceptron is a perceptron with three or more layers rather than two layers
in the original perceptron. The leftmost and rightmost layers are called an input layer
and an output layer, respectively. Notice that the output layer can have more than one
14 Applications of machine learning in wireless communications

X 3
T
x1 yi = g(w1ix + b1i) Sigmoid
Tanh
y1 2 ReLU
T
x2 z= g(w2y + b2)
y2

g(t)
1
z

0
yh
xd

–1
+1 +1 –3 –2 –1 0 1 2 3
t
(a) (b)

Figure 1.6 (a) A simple neural network with three layers, where g(·) is a non-linear
activation function and (b) the curves of three commonly used
activation functions

node though we only use one for simplicity in this example. The middle layer is called
a hidden layer, because its nodes are not observed in the training process. Similar to
a Markov chain, the node values of each layer are computed only depending on the
node values of its previous layer.
Because the original perceptron is just a linear function that maps the weighted
inputs to the output of each layer, the linear algebra shows that any number of lay-
ers can be reduced to a two-layer input–output model. Thus, a non-linear activation
function g : R → R, which is usually monotonously increasing and differentiable
almost everywhere [40], is introduced to achieve a non-linear mapping. Here, we list
some commonly used activation functions in Table 1.1, and their curves are shown
in Figure 1.6(b). Notice that the sigmoid function is just a modification of the prob-
ability mapping (1.15) used in logistic regression. As shown in Figure 1.6(b), the
hyperbolic tangent (tanh) function shares similar curve trace with the sigmoid func-
tion except the output range being (−1,1) instead of (0, 1). Their good mathematical
properties make them popular in early research [41]. However, they encounter dif-
ficulties in DNNs. It is easy to verify that limt→∞ g (t) = 0 and |g (t)| is a small
value in most areas of the domain for both of them. This property restricts their use
in DNNs since training DNNs require that the gradient of the activation function is
around 1. To meet this challenge, a rectified linear unit (ReLU) activation function is
proposed. As shown in Table 1.1, the ReLU function is piece-wise linear function and
saturates at exactly 0 whenever the input t is less than 0. Though it is simple enough,
the ReLU function has achieved great success and became the default choice in
DNNs [40].
Now, we will have a brief discussion about the training process of the multi-
layer perceptron. For simplicity, let us consider the performance of a regression task
Introduction of machine learning 15

Table 1.1 Three commonly used activation functions

Name Abbreviation Formula Range

−t
Sigmoid Sigmoid g(t) = 1/(1 + e ) (0, 1)
Hyperbolic tangent Tanh g(t) = ((et − e−t )/(et + e−t )) (−1, 1)
Rectified linear unit ReLU g(t) = max(0, t) [0, +∞)

by using the model shown in Figure 1.6(a). For convenience, denote the parame-
ters between the input and hidden layers as a matrix W1 = ŵ11 , . . . , ŵ1h , ŵ1h+1 ∈
T
R(d+1)×(h+1) , where ŵ1i = w1iT , b1i ∈ Rd+1 and ŵ1h+1 = (0, 0, . . . , 1)T . Similarly,
the parameters between the hidden and output layers are denoted as a vector w2 =
(w21 , . . . , w2h , w2h+1 )T ∈ Rh+1 , where w2h+1 = b2 . Let x = (x1 , . . . , xd , 1)T ∈ Rd+1 ,
y = (y1 , . . . , yh , 1)T ∈ Rh+1 and z denote the input vector, the hidden vector, and the
output scalar, respectively. Then the relations among x = (x1 , . . . , xd , 1)T ∈ Rd+1 ,
y = (y1 , . . . , yh , 1)T ∈ Rh+1 and z can be presented as

y = g(W1T x),
(1.18)
z = g(w2T y),

where the activation function g will act on each element for a vector as input. Suppose
we expect that the model outputs z̄ for the input x, and thus the square error is given
by e = (1/2)(z − z̄)2 . We decrease this error by using the gradient descent method.
This means that ∂e/∂W1 and ∂e/∂w2 need to be computed. By the gradient chain
rule, we have
∂e
= (z − z̄),
∂z
(1.19)
∂e ∂e ∂z
= = (z − z̄)g (w2T y + b2 )y,
∂w2 ∂z ∂w2
and
∂e ∂e ∂z ∂y
= , (1.20)
∂W1 ∂z ∂y ∂W1
where we have omitted the dimensions for simplicity. Thus, to compute ∂e/∂W1 ,
we first need to compute:
∂z
= g (w2T y)w2 ,
∂y
(1.21)
∂y ∂yi h+1 T h+1
= = g (w1i x) · xeiT i=1 ,
∂W1 ∂W1 i=1
16 Applications of machine learning in wireless communications

where ei ∈ Rh+1 denotes the unit vector with its ith element being 1. By plugging
(1.21) into (1.20), we have
∂e ∂e ∂z ∂y h+1
= = (z − z̄)g (w2T y + b2 )w2 g (w1iT x) · xeiT i=1
∂W1 ∂z ∂y ∂W1
(1.22)

h+1
= (z − z̄)g (w2T y + b2 ) w2i g (w1iT x) · xeiT
i=1

Thus, we can update the parameters by using the gradient descent method to
reduce the error. In the above deduction, what we really need to calculate are just
∂e/∂z, ∂z/∂w2 , ∂z/∂y and ∂y/∂W1 . As shown in Figure 1.7(a), we find these terms
are nothing but the derivatives of the node values or the parameters of each layer with
respect to the node values of the next layer. Beginning from the output layer, ‘multiply’
them layer by layer according to the chain rule, and then we obtain the derivatives of
the square error with respect to the parameters of each layer. The above strategy is the
so-called backpropagation (BP) algorithm [42]. Equipped with the ReLU activation
function, the BP algorithm can train the neural networks with dozens or even hundreds
of layers, which constitutes the foundation of deep learning.
In the model shown in Figure 1.6(a), we can observe that every node of the input
layer is connected to every node of the hidden layer. This connection structure is
called fully connection, and the layer which is fully connected (FC) to the previous
layer is called the FC layer [42]. Supposing that the number of nodes of two layers
are m and n, then the number of parameters of fully connection will be m × n, which
will be a large number even when m and n are moderate. Excessive parameters will
slow down the training process and increase the risk of overfitting, which is especially
serious in DNNs. Parameter sharing is an effective technology to meet this challenge.
A representative example using the parameter sharing technology is convolution neu-
ral networks (CNNs), which are a specialized kind of neural networks for processing

∂y ∂z ∂z ∂e
∂W1 ∂y ∂w2 ∂z
a
x1
y1
b ag + bh + ci
x2 g
y2
c bg + ch + di
W1 w2 z h
d cg + dh + ei
yh i
xd e dg + eh + fi

+1 +1 f

(a) (b)

Figure 1.7 (a) The derivatives of each layer with respect to its previous layer and
(b) an example of the convolution operation performed on vectors
Introduction of machine learning 17

data that has a known grid-like topology [43], such as time-series data and matrix data.
The name of CNNs comes from its basic operation called convolution (which has a
little difference from the convolution in mathematics). Though convolution operation
can be performed on vectors, matrices and even tensors with arbitrary order, we will
introduce the vector case here for simplicity.
To perform the convolution operation on a vector x ∈ Rd , we first need a kernel,
which is also a vector k ∈ Rl with l ≤ d. Let x[i : j] denotes the vector generated by
extracting the elements from the ith position to the jth position of x, i.e. x[i : j]
T
xi , xi+1 , . . . , xj . Then the convolution of x and k is defined as
⎛ ⎞
x[1 : l], k
⎜ x[2 : l + 1], k ⎟
⎜ ⎟
xk ⎜ .. ⎟ = ŷ ∈ Rd−l+1 , (1.23)
⎝ . ⎠
x[d − l + 1 : d], k

where ·, · denotes the inner product of two vectors. See Figure 1.7(b) for an example
of convolution. The convolution operation for matrices and tensors can be defined
similarly by using a matrix kernel and a tensor kernel, respectively. Based on the
convolution operation, a new transformation structure, as distinct from the fully
connection, can be built as

y = g (x k), (1.24)

where k is the parameter that needs to be trained. The layer with this kind of trans-
formation structure is called the convolution layer. Compared with the FC layer, the
number of parameters has dramatically decreased for the convolution layer. Further-
more, the size of the kernel is independent to the number of nodes of the previous
layer.
It should be noted that we can set several kernels in a convolution layer to generate
richer features. For example, if we choose the convolution layer with M kernels as the
hidden layer in example shown in Figure 1.6(a), then m features will be generated as

g (x (k1 , . . . , km )) = (g (x k1 ) , . . . , g (x km )) = (y1 , . . . , ym ) . (1.25)

Similarly, we can continue to transform (y1 , . . . , ym ) by using more convolution layers.

In addition to the convolution operation, another operation widely used in CNNs
is the max-pooling operation. As a kernel is needed in the convolution operation,
the max-pooling operation needs a window to determine the scope of the operation.
Specifically, given a vector x ∈ Rd and a window of size l with d being divisible by
l, the max-pooling operation to x with the window is defined as
⎛ ⎞
max {x[1 : l]}
⎜ max {x[l + 1 : 2l]} ⎟
⎜ ⎟ d
max-pooling(x, l) ⎜ .. ⎟ = ŷ ∈ R l . (1.26)
⎝ . ⎠
max {x[d − l + 1 : d]}
18 Applications of machine learning in wireless communications

See Figure 1.8(a) for an example of max-pooling. The max-pooling operation for
matrices and tensors can be defined similarly by using windows with different dimen-
sions. Other layers such as normalization layers and average-pooling layers [44], we
do not detail here to stay focused.
Generally, a neural network may be constructed by using several kinds of layers.
For example, in a classical architecture, the first few layers are usually composed
alternate of the convolution layer and the max-pooling layer, and the FC layer is often
used as the last few layers. A simple example of architecture for classification with a
convolution network is shown in Figure 1.8(b).
The recent 10 years have witnessed earthshaking development of deep learning.
The state-of-the-art of many applications has been dramatically improved due to its
development. In particular, CNNs have brought about breakthroughs in processing
multidimensional data such as image and video. In addition, recurrent neural net-
works [42] have shone light on sequential data such as text and speech; generative
anniversary networks [45] are known as a class of models which can learn a mimic
distribution from the true data distribution to generate high-quality artificial samples,
such as images and speeches; deep RL (DRL) [46] is a kind of tool to solve control and
decision-making problems with high-dimensional inputs, such as board game, robot
navigation and smart transportation. Reference [42] is an excellent introduction to
deep learning. More details about the theory and the implementation of deep learning
can be found in [43]. For a historical survey of deep learning, readers can refer to [47].
Many open-source deep-learning frameworks, such as TensorFlow and Caffe, make
neural networks easy to implement. Readers can find abundant user-friendly tutorials
from the Internet. Deep learning has been widely applied in many fields of wireless
communications, such as network prediction [48,49], traffic classification [50,51],
modulation recognition [52,53], localization [54,55] and anomaly detection [56–58].
Readers can refer to [59] for a comprehensive survey of deep learning in mobile and
wireless networks.

Fully connected layer (output)

Fully connected layer

a
b Max-pooling layer
max(a,b)
c Convolution layer Data flow
Max-pooling ( , )= max(c,d)
d Max-pooling layer
max(e, f )
e Convolution layer
f Input
(a) (b)

Figure 1.8 (a) An example of max-pooling to x ∈ R6 with a window of size 2 and

(b) an simple example of architecture for classification with
convolutional network
Introduction of machine learning 19

1.1.4 Summary of supervised learning

In this section, we have discussed supervised learning. The main task of supervised
learning is to learn a function that maps an input to an output based on a training
set, which consists of examples of input–output pairs. According to whether the
predicted variable is categorical or continuous, a supervised learning task is called a
classification or regression task.
As shown in Figure 1.9, three popular technologies and their improvements have
been introduced in this section. Among them, the k-NNs method has the simplest form.
The k-NN method does not need explicit training steps and is very easy to implement.
In most cases, it can give a not-bad result. However, if your target is high accuracy, then
the latter two technologies will be better choices. Decision tree is a kind of supervised
learning algorithms based on tree structures. CART is known as an effective method
to construct a single tree. To acquire a better performance, RF constructs many trees
by using a randomized CART, and then a final prediction will be given by integrating
predictions of all trees. GBDT is a boosting method based on CART, and it is known as
one of the best methods in supervised learning. A perceptron is a linear classification
model as a foundation of SVM, logistic regression and multilayer perceptron. SVM
improves the performance of the perceptron by maximizing the classification margin.
Logistic regression is more robust to outliers than SVM and can give the probability of
a sample belongs to a category. Multilayer perceptron is a perceptron with multilayers,
which is also known as deep learning when the number of layers is large enough. Deep
learning, as a heavy tool in machine learning, can have millions of parameters and

k-Nearest neighbour
method
Classification and
regression tree

Decision tree Random forest

Supervised Gradient boosting

learning decision tree

Support vector
machine

Perceptron Logistic regression

Multilayer
perceptron and deep
learning

Figure 1.9 Structure chart for supervised learning technologies discussed in this
chapter
20 Applications of machine learning in wireless communications

Table 1.2 Summary of applications of supervised learning in wireless

communications

Method Function Application in

wireless communications

k-Nearest Classification/Regression Fall detection [10]

neighbours method Energy enhancements [11]
Classification and Classification/Regression Improving congestion control [15]
regression tree Monitoring animal behaviour [16]
Intrusion detection [17]
Gradient boosting Classification/Regression Indoor localization [19]
decision tree Fall detection [20]
Random forest Classification/Regression Obstacle detection [25]
QoE prediction [26]
Support vector Classification/Regression Transmission mode identification [34]
machine Attack detection [35]
Localization [36]
MIMO channel learning [3]
Logistic regression Classification Home wireless security [37]
Reliability evaluation [36]
Patient anomaly detection [39]
Energy enhancements [11]
Multilayer perceptron Classification/Regression Network prediction [48,49]
and deep learning Traffic classification [50,51]
Modulation recognition [52,53]
Localization [54,55]
Anomaly detection [56–58]

cost huge amount of computing resources for training. As a reward, it is state-of-the-

art method for most machine-learning tasks. Furthermore, many technologies from
hardware to software have been applied to accelerate its training stage. In Table 1.2,
we summarize the applications of supervised learning in wireless communications.
Zhang et al. [59] conduct a comprehensive survey regarding deep learning in wireless
communications. Readers can refer to [8] for more advanced approaches of supervised
learning.

1.2 Unsupervised learning

Unsupervised learning is a process of discovering and exploring for investigating
inherent and hidden structures from data without labels [60]. Unlike supervised learn-
ing where a training set {(xi , yi )}ni=1 is provided, we have to work with an unlabelled
data set {xi }ni=1 (there is no yi ) in unsupervised learning. Three common unsupervised
learning tasks are clustering, density estimation and dimension reduction. The goal
of clustering is to divide samples into groups (called clusters) such that the samples
in the same cluster are more similar to each other than to those in different clusters.
Introduction of machine learning 21

Rather than defining classes before observing the test data, clustering allows us to
find and analyse the undiscovered groups hidden in data. From Sections 1.2.1–1.2.4,
we will discuss four representative clustering algorithms. Density estimation aims to
estimate the distribution density of data in the feature space, and thus we can find the
high-density regions which usually reveal some important characteristics of the data.
In Section 1.2.5, we will introduce a popular density-estimation method: the Gaus-
sian mixture model (GMM). Dimension reduction pursues to transform the data in a
high-dimensional space to a low-dimensional space, and the low-dimensional repre-
sentation should reserve principal structures of the data. In Sections 1.2.6 and 1.2.7,
we will discuss two practical dimension-reduction technologies: principal component
analysis (PCA) and autoencoder.

1.2.1 k-Means
k-Means [61] is one of the simplest unsupervised learning algorithms which solve
a clustering problem. This method only needs one input parameter k, which is the
number of clusters we expect to output. The main idea of k-means is to find k optimal
points (in the feature space) as the representatives of k clusters according to an evalu-
ation function, and each point in the data set will be assigned to a clusterbased on the
n
distance between the point to each representative. Given a data set X = xi ∈ Rd i=1 ,
let Xi and ri denote the ith cluster and the corresponding representative, respectively.
Then, k-means aims to find the solution of the following problem:

k
min x − ri 2
ri , Xi
i=1 x∈Xi

k
(1.27)
s.t. Xi = X
i=1

Xi Xj = ∅ (i = j).

Notice that x∈Xi x − ri 2 measures how dissimilar the points in ith cluster to the
corresponding representative, and thus the object is to minimize the sum of these
dissimilarities.
However, the above problem has been shown to be an NP-hard problem [62],
which means the global optimal solution cannot be found efficiently in general cases.
As an alternative, k-means provides an iterative process to obtain an approximate
solution. Initially, it randomly selects k points as initial representative. Then it alter-
nately conducts two steps as follows. First, partitions the all points into k clusters in
which each point is assigned to the cluster with the nearest representative. Second,
take the mean points of each cluster as the k new representatives, which reveals the
origin of the name of k-means. The above steps will be repeated until the clusters
remain stable. The whole process is summarized in Algorithm 1.6.
Let us checkout the correctness of k-means step by step. In the first step, when
fixing k representatives, each point is assigned to the nearest representative and thus
the object value of (1.27) will decrease or remain unchanged. In the second step,
22 Applications of machine learning in wireless communications

Algorithm 1.6: K-means algorithm

n
Input: dataset xi ∈ Rd i=1 , number of clusters k
Output: clusters X1 , . . . , Xk
1 randomly select k points r1 , . . . , rk as representatives;
2 repeat
/* the first step */
3 for i = 1 to k do
4 Xi ← ∅;
5 for j = 1 to n do
6 î = arg min xj − ri 2 ;
1≤i≤k
7 Xî ← Xî ∪ {xj };
/* the second step */
8 for i = 1 to k do
x
ri ←
x∈Xi
9 |Xi |
;
10 until X1 , . . . , Xk do not change;

by fixing the k clusters, we can find the optimal solutions to the sub-problems of
(1.27), i.e.:

x∈Xi x
arg min x − r =
2
(i = 1, . . . , k). (1.28)
r∈R d
x∈X
|Xi |
i

Thus, the value of object function will also decrease or remain unchanged in the
second step. In summary, the two steps in k-means will decrease the object value or
reach convergence.
Because k-means is easy to implement and has short running time for low-
dimensional data, it has been widely used in various topics and as a preprocessing
step for other algorithms [63–66]. However, three major shortcomings are known for
the original k-means algorithm. The first one is that choosing an appropriate k is a non-
trivial problem. Accordingly, X -means [67] and G-means [68] have been proposed
based on the Bayesian information criterion [69] and Gaussian distribution. They can
estimate k automatically by using model-selection criteria from statistics. The second
one is that an inappropriate choice for the k initial representatives may lead to poor
performance. As a solution, the k-means++ algorithm [70] augmented k-means with
a simple randomized seeding technique and is guaranteed to find a solution that is
O(log k) competitive to the optimal k-means solution. The third one is that k-means
fails to discover clusters with complex shapes [71]. Accordingly, kernel k-means [72]
was proposed to detect arbitrary-shaped clusters, with an appropriate choice of the
kernel function. References [73], [74] and [75] present three applications of k-means
Introduction of machine learning 23

in collaborative signal processing, wireless surveillance systems and wireless hybrid

networks, respectively.

1.2.2 Density-based spatial clustering of applications with noise

k-Means is designed to discover spherical-shaped clusters. Though kernel k-means
can find arbitrary-shaped clusters with an appropriate kernel function, there is no
general guidelines for how to choose an appropriate kernel function. In contrast,
density-based clustering algorithms are known for the advantage of discovering
clusters with arbitrary shapes. In this section, we will introduce the most famous
density-based clustering algorithm named DBSCAN (density-based spatial clustering
of applications with noise).
DBSCAN [76] was proposed as the first density-based clustering algorithm in
1996. The main idea of DBSCAN can be summarized as three steps. First, DBSCAN
estimates the density of each point x by counting the number of points which belong
to the neighbourhood of x. Second, it finds core points as points with high density.
Third, it connects core points that are very close and their neighbourhood to form
dense regions as clusters. Next, we will detail the three steps.
To define density, DBSCAN introduces the concept of ε-neighbourhood, where
ε is a user-specified parameter. Given a data set X = {xi }ni=1 , the ε-neighbourhood of
a point x denoted by Nε (x) is defined as

Nε (x) = {xi |xi ∈ X , dist(x, xi ) ≤ ε} , (1.29)

where dist(·, ·) can choose any distance function according to the application. Then,
the density of x, denoted by ρ(x), is defined as the number of points belonging to the
neighbourhood of x, i.e.:

ρ(x) = |Nε (x)|. (1.30)

After the density being defined, DBSCAN introduces another user-specified

parameter MinPts to find core points. Specifically, if the density of a point x is not
less than MinPts, then x is called a core point. Furthermore, the set consisting of all
core points is denoted by O {x ∈ X |ρ(x) ≥ MinPts}.
To form clusters, DBSCAN defines the connected relation between core points.
For two core points x and y, we say they are connected if there exists a core-point
sequence x ≡ z1 , . . . , zt ≡ y such that zi+1 ∈ Nε (zi ) for 1 ≤ i < t and {zi }ti=1 ⊂ O.
An illustration is shown in Figure 1.10(a). Notice that the connected relation can be
verified as an equivalence relation on the set O. Thus, DBSCAN uses this equiv-
alence relation to divide core points into equivalence classes. Suppose k that core
points
are divided into k equivalence classes O 1 , . . . , Ok , where i=1 i = O and
O
Oi Oj = ∅ for i = j. These equivalence classes constitute the skeleton of clusters.
Then, DBSCAN forms the clusters C1 , . . . , Ck by letting:

Ci = Nε (Oi ) Nε (x) (i = 1, . . . , k). (1.31)
x∈Oi
24 Applications of machine learning in wireless communications

Non-core point
Non-core point Core point
Core point Outlier

Cluster C1
x ε
Cluster C2

(a) (b)

Figure 1.10 (a) An illustration for the connected relation, where the two core
points x and y are connected and (b) an illustration for clusters and
outliers. There are two clusters and seven outliers denoted by
four-pointed stars

Notice
that there
may exist some points which do not belong to any clusters, that is,
k
X\ i=1 Ci = ∅. DBSCAN assigns these points as outliers because they are far
from any normal points. An illustration is shown in Figure 1.10(b).
To this end, we have presented the three main steps of DBSCAN. Algorithm 1.7
summarizes the details of DBSCAN. Let the number of points be n = |X |. Finding
the ε-neighbourhood for points is the most time-consuming step with computational
complexity of O(n2 ). The other steps can be implemented within nearly linear com-
putational complexity. Thus, the computation complexity of DBSCAN is O(n2 ). The
neighbour searching step of DBSCAN can be accelerated by using spatial index tech-
nology [60] and groups method [77]. DBSCAN can find arbitrary-shaped clusters
and is robust to outliers. However, the clustering quality of DBSCAN highly depends
on the parameter ε and it is non-trivial to find an appropriate value for ε. Accordingly,
the OPTICS algorithm [78] provides a visual tool to help users find the hierarchical
cluster structure and determine the parameters. Some applications of DBSCAN in
wireless sensor networks can be found in [79–83].

1.2.3 Clustering by fast search and find of density peaks

In 2014, Rodriguez and Laio proposed a novel-density-based clustering method,
named Fast search-and-find of Density Peaks (FDP) [84]. FDP has received extensive
attention due to its brilliant idea and the capacity to detect clusters with complex point
distribution. As a density-based clustering algorithm, FDP shares similar steps with
Introduction of machine learning 25

Algorithm 1.7: DBSCAN

Input: dataset X = {xi }ni=1 , ε, MinPts
Output: clusters C1 , . . . , Ck , outliers set A
1 for i = 1 to n do
2 find Nε (xi );
3 ρ(xi ) = |Nε (xi )|;
4 define O {x ∈ X |ρ(x) ≥ MinPts};
5 k = 0;
6 repeat
7 k = k + 1;
8 randomly select a core point o from O;
9 use the depth-first-search algorithm to find the set
Ok {x ∈ O|x is connected to o};
10 define Ck Nε (Ok );
11 O = O\Ok ;
12 until O = ∅;

k
13 define A X \ i=1 C i ;

DBSCAN, that is, estimating density, finding core points and forming clusters. How-
ever, there are two differences between them. First, FDP detects core points based on
a novel criterion, named delta-distance, other than the density. Second, FDP forms
cluster by using a novel concept, named higher density nearest neighbour (HDN),
rather than the neighbourhood in DBSCAN. Next, we will introduce the two novel
concepts followed by the details of FDP.
To begin with, FDP shares the same density definition with DBSCAN.
Specifically, given a data set X = {xi }ni=1 , the density of a point x is computed as
ρ(x) = |Nε (x)|, (1.32)
where Nε (x) denotes the ε-neighbourhood of x (see (1.29)). After computing the
density, FDP defines the HDN of a point x, denoted by π (x), as the nearest point
whose density is higher than x, i.e.:
π (x) arg min dist(y, x). (1.33)
y∈X ,ρ(y)>ρ(x)

Specially, for the point with the highest density, its HDN is defined as the farthest
point in X . Then, FDP defines the delta-distance of a point x as
δ(x) = dist(x, π (x)). (1.34)
Note that the delta-distance is small for most points and only much larger for a point
being either a local maxima in the density or an outlier because the HDN of a outlier
may be far from it. In FDP, a local maxima in the density is called a core point.
26 Applications of machine learning in wireless communications

2.6 0.25
2.55
0.2 Core points
2.5
0.15
2.45 Outliers

δ
x2

0.1
2.4

2.35 0.05

2.3 0
1.65 1.7 1.75 1.8 1.85 1.9 1.95 2 0 20 40 60 80 100 120 140
x1 ρ
(a) (b)

Figure 1.11 (a) A simple data set distributed in a two-dimensional space. There
are two clusters and three outliers (denoted by green ‘x’). The core
point of each cluster is denoted by a pentagram. (b) The decision
graph corresponding to the example shown in (a), which is the plot
of δ as a function of ρ for all points

This observation is illustrated by a simple example shown in Figure 1.11.

Figure 1.11(a) shows a data set distributed in a two-dimensional space. We can find
that there are two clusters, and the core point of each cluster is denoted by a penta-
gram. In addition, three outliers are shown by green ‘×’. Figure 1.11(b) shows the
plot of δ as a function of ρ for all points. This representation is called a decision graph
in FDP. As shown in the decision graph, though the density of the ordinary points
(the blue dots) fluctuates from around 20 to 120, the low δ values situate them in the
bottom of the graph. The outliers have higher δ values than the ordinary points, but
they locate in the left of the graph due to their low densities. In contrast, the two core
points, which have higher densities and much larger δ values, locate in the top-right
area of the ρ–δ plane.
Thus, by using the decision graph, core points and outliers can be found visually.
FDP recognizes the cluster number as the number of core points. After that, each
remaining point is assigned to the same cluster as its HDN. An illustration is shown in
Figure 1.12. There are three core points and thus three clusters, and different clusters
are distinguished by different colours.
To this end, we have presented the main steps of FDP. The detail of FDP is
summarized in Algorithm 1.8. The clustering quality of FDP is largely dependent
on the parameter ε, and it is non-trivial to choose an appropriate ε. To reduce this
dependence, one can use the kernel density [85] instead of (1.32). Compared with
DBSCAN, FDP is more robust to the clusters with various densities, and they have the
same time complexity of O(n2 ). In addition, FDP provides a brilliant idea, the HDN.
Many algorithms have been proposed inspired by this idea, such as [86–90]. Refer-
ences [91] and [92] present two applications of FDP to optimal energy replenishment
and balancing the energy consumption in wireless sensor networks, respectively.
Introduction of machine learning 27

1 Core points
1

1 3 3
2 1
3 5 4
1 1
1 1
2
2
2 3
1 1
3
2 2 1

2
1

Figure 1.12 An illustration for assigning remaining points. The number in each
point denotes its density. The HDN of each point is specified by an
arrow. The three core points are denoted by pentagrams

Algorithm 1.8: FDP

Input: dataset X = {xi }ni=1 , ε
Output: clusters C1 , . . . , Ck , outliers set A
1 for i = 1 to n do
2 find Nε (xi );
3 ρ(xi ) = |Nε (xi )|;
4 for i = 1 to n do
5 find π (xi ) = arg miny∈X ,ρ(y)>ρ(x) dist(y, x);
6 δ(xi ) = dist(xi , π (xi ));
7 draw the decision graph, that is, the plot of δ as a function of ρ for all points;
8 Find core point O set and outlier set A by using the decision graph visually;
9 Suppose O = {o1 , . . . , ok };
10 Create clusters C1 = {o1 }, . . . , Ck = {ok };
11 for x ∈ X \(O ∪ A) do
12 assign x to a cluster according to π (x);

1.2.4 Relative core merge clustering algorithm

In real applications, clusters within a data set often have various shapes, densities
and scales. To detect clusters with various distributions, Geng et al. proposed a
RElative COre MErge (RECOME) clustering algorithm [90]. The core of RECOME is
a novel density measure, i.e. relative k-NN kernel density (NKD) (RNKD). RECOME
recognizes core samples with unit RNKD and divides noncore samples into atom
clusters by using the HDN relation as mentioned in Section 1.2.3. Core samples and
28 Applications of machine learning in wireless communications

their corresponding atom clusters are then merged through α-reachable paths on a
k-NNs graph. Next, we will introduce RNKD followed by the details of RECOME.
The RNKD is based on an NKD. For a sample x in a data set X = {xi }ni=1 , the
NKD of x is defined as

dist(x, z)
ρ(x) = exp − , (1.35)
z∈N (x)
σ
k

where Nk (x) denotes the k-NNs set of x in X , and σ is a constant which can be
estimated from the data set. NKD enjoys some good properties and allows easy
discrimination of outliers. However, it fails to reveal clusters with various densities.
To overcome this shortcoming, RNKD is proposed with the definition of:
ρ(x)
ρ ∗ (x) = . (1.36)
max {ρ(z)}
z∈Nk (x)∪{x}

Intuitively, RNKD is a ratio of densities of two neighbouring samples, and thus it

is robust to the change of density since the densities of two neighbouring samples
are always at the same level. Figure 1.13 shows a comparison between NKD and
RNKD for a simple data set. We can observe that RNKD successfully detects three
dense clusters and five sparse clusters which NKD fails to reveal. For more detailed
discussions about RNKD, readers can refer to [90].
From the definition of RNKD, we know that a sample with unite RNKD has a
local maximal density. Thus, the samples with unite RNKD are good candidates for
cluster centres and are called core samples in RECOME. In particular, the set of core
samples is denoted by O {x|x ∈ X , ρ ∗ (x) = 1}.
Inspired by the idea from FDP, RECOME defines a directed graph G = (X , A)
with the arc set A = {x, π(x)|x ∈ X \O}, where π (x) was defined in (1.33). It can be
shown that, starting from any noncore sample and following the arcs, a core sample
will be reached eventually. In fact, G consists of many trees with disjoint samples,

(a) (b)

Figure 1.13 (a) The heat map of NKD for a two-dimensional data set. (b) The heat
map of RNKD for a two-dimensional data set. Figures are quoted
from [90]
Introduction of machine learning 29

and each tree is rooted at a core sample (similar to the relation shown in Figure 1.12).
Furthermore, a core sample and its descendants in the tree are called an atom cluster
in RECOME. Atom clusters form the basis of final clusters; however, a true cluster
may consist of several atom clusters. This happens when many local maximal exist
in one true cluster. Thus, a merging step is introduced to combine atom clusters into
true clusters.
RECOME treats each core sample as the representative of the atom cluster that it
belongs to and merges atom clusters by merging core samples. To do that, it defines
another graph with undirected edges, k-NN graph, as

Gk = (X , E), E = {x, z|x ∈ Nk (z) ∧ z ∈ Nk (x)}. (1.37)

Furthermore, on the k-NN graph, two samples x and z are called α-connected if there
exists a path x, w1 , . . . , ws , z in Gk such that ρ ∗ (wi ) > α for i = 1, . . . , s, where
α is a user-specified parameter. It can be verified that the α-connected relation is
an equivalence relation on the core sample set. RECOME divides core samples into
equivalence classes by using this relation. Correspondingly, atom clusters associated
with core samples in the same equivalent class are merged into a final cluster. For
clarity, we summarize the details of RECOME in Algorithm 1.9.
In RECOME, there are two user-specified parameters k and α. As discussed
in [90], the clustering quality of √ RECOME
√ is not sensitive to k, and it is recom-
mended to tune k in the range [ n/2, n]. On the other hand, the clustering result
of RECOME largely depends on parameter α. In particular, as α increases, cluster
granularity (i.e. the volume of clusters) decreases and cluster purity increases. In [90],
authors also provide an auxiliary algorithm to help users to tune α fast. RECOME
has been shown to be effective on detecting clusters with different shapes, densities
and scales. Furthermore, it has nearly linear computational complexity if the k-NNs
of each sample are computed in advance. In addition, readers can refer to [93] for an
application to channel modelling in wireless communications.

1.2.5 Gaussian mixture models and EM algorithm

In this section, we will introduce GMM, which is used for density estimation. In
machine learning, the goal of density estimation is to estimate an unobservable
underlying probability density function, based on a finite observed data set. Once
a probability density function is obtained, we can learn a lot of valuable information
based on it. GMM has been widely used for density estimation due to its simple form
but strong capacity in data representation. Next, we will introduce the formulation of
GMM followed by the estimation for parameters of GMM.
As the name suggests, the distribution of GMM can be written as a weighted
linear combination of several Gaussian distributions. Specifically, the probability
density of a vector x ∈ Rd is given by

k
f (x) = φi N (x|μi , i ), (1.38)
i=1
30 Applications of machine learning in wireless communications

Algorithm 1.9: RECOME clustering

Input: dataset X = {xi }ni=1 , parameters k, α
Output: clusters C1 , . . . , Ct
1 for i = 1 to n do
2 find Nk (xi );

3 ρ(xi ) = exp − dist(x σ
i ,z)
;
z∈Nk (xi )

4 for i = 1 to n do
5 ρ ∗ (xi ) = ρ(xi )
max {ρ(z)}
;
z∈Nk (xi )∪{xi }

6 find O {x|x ∈ X , ρ ∗ (x) = 1};

7 for x ∈ X \O do
8 find π(x) arg min dist(y, x);
y∈X ,ρ(y)>ρ(x)

9 construct directed graph G = (X , A), where A = {x, π (x)|x ∈ X \O};

10 for o ∈ O do
11 find atom cluster Co = {o} ∪ {x|x is connected to o in G};
12 construct k-NN graph Gk = (X , E), where E = {x, z|x ∈ Nk (z) ∧ z ∈ Nk (x)};
13 t = 0;
14 repeat
15 t = t + 1;
16 randomly select a core sample o from O;
17 use the depth-first-search algorithm to find set
Ot {x ∈ O|x
is α-connected to o in Gk };
18 define Ct = x∈Ot Cx ;
19 O = O\Ot ;
20 until O = ∅;

k
where φ1 , . . . , φk are non-negative with i=1 φi = 1, and

1 1 1
N (x|μi , i ) = exp − (x − μi )T −1
i (x − μi ) (1.39)
(2π) | i |
d/2 1/2 2
is the Gaussian distribution with mean vector μi and covariance matrix i . Parameter
k controls the capacity and the complexity of GMM. Considering a two-dimensional
data set shown in Figure 1.14(a), Figure 1.14(b) shows the fitted distribution by using
GMM with k = 5, where we observe that only a fuzzy outline is reserved and most
details are lost. In contrast, Figure 1.14(c) shows the case for k = 20, where we
find the fitted distribution reflects the main characteristic of the data set. In fact, by
increasing k, GMM can approximate any continuous distribution to some desired
degree of accuracy. However, it does not mean that the larger k is better, because a
large k may lead to overfitting and a huge time cost for the parameter estimation.
In most cases, k is inferred experientially from the data.
Introduction of machine learning 31

×10–6
350 350 16
300 300 14
12
250 250
10
200 200
8
150 150
6
100 100 4
50 50 2
0 0
0 100 200 300 400 500 600 700 0 100 200 300 400 500 600 700
(a) (b)
×10–5
350
2
300 1.8
1.6
250
1.4
200 1.2
1
150
0.8
100 0.6
0.4
50 0.2
0
0 100 200 300 400 500 600 700
(c)

Figure 1.14 (a) A two-dimensional data set, (b) the fitted distribution by using
GMM with k = 5 and (c) the fitted distribution by using GMM with
k = 20

Now, we discuss the parameter estimation for GMM. Given a data set {xj }nj=1 , the
log-likelihood of GMM is given by
k

n
n
L= ln f (xj ) = ln φi N (xj |μi , i ) . (1.40)
j=1 j=1 i=1

Thus, the maximum likelihood estimation is to solve:

max L
{φi },{μi },{ i }

s.t. i 0 (i = 1, . . . , k),
φi ≥ 0 (i = 1, . . . , k), (1.41)

k
φi = 1.
i=1
32 Applications of machine learning in wireless communications

where i 0 means that i should be a positive semi-definite matrix. Unfortunately,

there is no effective ways to find the global optimal solution to (1.41). Next, we
will introduce the expectation–maximization (EM) algorithm to find an approximate
solution to (1.41).

1.2.5.1 The EM algorithm

The EM algorithm is an iterative method to solve the problem of maximum likelihood
estimation for the model which depends on unobserved latent variables. To explain
the meaning of latent variables, let us consider the aforementioned GMM. Samples
obeying the distribution of GMM can be generated by the following two steps:
1. Randomly select one from the k Gaussian models with the probability that
P{the ith model being chosen} = φi (i = 1, . . . , k).
2. Supposing that the zth model is selected in the first step, then generate a sample
x by using the zth Gaussian model.
By the above steps, we know that the joint probability distribution of (z, x) is given
by p(z, x) = φz N (x|μz , z ). However, in the parameter-estimation of GMM, what we
are given is only the value of x, and the value of z is unobservable to us. Here, z is
called a latent variable.
Now, let us consider the maximum likelihood estimation of a model with latent
variables. Supposing that an observable sample set X = {xj }nj=1 is given and the cor-
responding latent sample set Z = {zj }nj=1 is unobservable to us, the goal is to estimate
the parameter set θ of a probability distribution p(x|θ). Notice that we regard Z as a
random variable since it is unknown to us. By denoting p(X |θ ) = nj=1 p(xj |θ ), the
problem of maximum likelihood estimation can be written as

max ln p(X |θ) ≡ ln p(X , Z|θ ). (1.42)
θ
Z

However, solving (1.42) is usually a difficult problem since the unknown Z. As

an alternative, the EM algorithm can generate an approximate solution by using an
iterative process. Initially, it initializes the parameter set θ to some random values θ̄ .
Then, the following two steps will be alternately executed until convergence.
1. E-step: Compute the expectation of ln p(X , Z|θ ) with respect to Z with the
distribution p(Z|X , θ̄), that is,

EZ|X ,θ̄ ln p(X , Z|θ ) = p(Z|X , θ̄ ) ln p(X , Z|θ). (1.43)
Z

2. M-step: Update the parameter θ̄ by the solution of maximizing (1.43), that is,

θ̄ = arg max p(Z|X , θ̄ ) ln p(X , Z|θ ). (1.44)
θ
Z

Finally, the resulting θ̄ will be the estimated parameter set. Here, we do not discuss
the correctness of the EM algorithm to stay focused. Readers can refer to [94] for
more details.
Introduction of machine learning 33

1.2.5.2 The EM algorithm for GMM

We are ready to apply the EM algorithm to solve the parameter-estimation problem for
GMM. As mentioned in Section 1.2.5.1, let zj be a latent variable which denotes the
index of the Gaussian model that generates xj . To begin with, initialize the parameters
¯ i for i = 1, . . . , k.
φi , μi , i to some random values φ̄i , μ̄i ,
In the E-step, we first need to compute:
¯ l)
φ̄l N (xj |μ̄l ,
p(zj = l|xj , θ̄ ) = k . (1.45)
¯
i=1 φ̄i N (xj |μ̄i , i )

Since p(zj = l|xj , θ̄ ) is a constant in an iteration, we denote it as γjl for simplicity.

Then, compute the exception:

EZ|X ,θ̄ ln p(X , Z|θ ) = p(Z|X , θ̄ ) ln p(X , Z|θ )
Z

n
= p(zj |xj , θ̄) ln p(xj , zj |θ )
j=1 zj

n
k (1.46)
= p(zj = l|xj , θ̄ ) ln (φl N (xj |μl , l ))
j=1 l=1

n
k
= γjl ln (φl N (xj |μl , l )).
j=1 l=1

In the M-step, we need to solve:

n k
1 1
max γjl ln φl − ln | l | − (xj − μl )T −1
l (x j − μ l )
{φl },{μl },{ l } 2 2
j=1 l=1

s.t. l 0 (l = 1, . . . , k),
(1.47)
φl ≥ 0 (l = 1, . . . , k),

k
φl = 1,
l=1

where we have omitted unrelated constants. By using the Karush–Kuhn–Tucker

conditions [28], the optimal solution to (1.47) is given by
nl
φl = ,
n
1
n
μl = γjl xj ,
nl j=1 (1.48)

1
n
l = γjl (xj − μl )(xj − μl )T ,
nl j=1
34 Applications of machine learning in wireless communications

Algorithm 1.10: The EM algorithm for GMM

Input: dataset X = {xj }nj=1 , number of Gaussian models k
¯ l }kl=1
Output: estimated parameters {φ̄l }kl=1 , {μ̄l }kl=1 , {
1 k ¯
randomly initialize {φ̄l }l=1 , {μ̄l }l=1 , { l }l=1 ;
k k

2 repeat
/* E-step */
3 for l = 1 to k do
4 for i = 1 to n do
φ̄ N (x |μ̄ ,¯ )
5 γjl = k l φ̄ Nj (xl |μ̄ l,¯ ) ;
i=1 i j i i

6 nl = nj=1 γjl ;
/* M-step */
7 for l = 1 to k do
8 φ̄l = nnl ;

9 μ̄l = n1l nj=1 γjl xj ;
10 for l = 1 to
k do
11 ¯ l = 1 nj=1 γjl (xj − μ̄l )(xj − μ̄l )T ;
nl

12 until convergence;

where we have defined nl = nj=1 γjl . We conclude the whole procedure in
Algorithm 1.10.
In addition to fitting the distribution density of data, GMM can also be used for
clustering data. Specifically, if we regard the k Gaussian models as the ‘patterns’ of
k clusters, then the probability that a sample xi comes from the lth pattern is given by
¯ l)
φl N (xj |μ̄l ,
p(zj = l|xj ) = k . (1.49)
¯
i=1 φi N (xj |μ̄i , i )

Thus, l ∗ arg maxl p(zj = l|xj ) gives the index of the cluster that xj most likely
belongs to. Furthermore, low p(zj = l ∗ |xj ) may imply that xj is an outlier. Reference
[95] shows a diffusion-based EM algorithm for distributed estimation of GMM in
wireless sensor networks. References [96] and [97] present two applications of GMM
in target tracking and signal-strength prediction, respectively.

1.2.6 Principal component analysis

In real applications, we usually encounter data with high dimensionality, such as
speech signals or digital images. In order to handle such data adequately, we need to
reduce its dimensionality. Dimension reduction is a process of transforming original
high-dimensional data into low-dimensional representation which reserve meaningful
characteristics of the data. Among all dimension-reduction methods, PCA is the most
Introduction of machine learning 35

0.6 0.6
0.4 0.4
0.2 0.2
0
Z

Z
0
–0.2 –0.2
–0.4 –0.4
–0.6 –0.6
1 0.8 1 1 1
0.6 0.8 0.8 0.8
0.4 0.6 0.6 0.6
0.2 0.4 0.4 0.4
Y 0.2 X Y 0.2 0.2 X
0 0 0 0
(a) (b)

1.2
1
0.8
0.6
0.4
0.2
0
–0.2
–0.2 0 0.2 0.4 0.6 0.8 1 1.2
(c)

Figure 1.15 (a) A three-dimensional data set, (b) the points are distributed near a
two-dimensional plane and (c) projection of the points onto the plane

popular one. PCA can be derived from the perspectives of both geometry and statistics.
Here we will focus on the former perspective since it meets our intuition better.
To begin with, let us consider a three-dimensional data set as shown in
Figure 1.15(a). Though all points lie in a three-dimensional space, as shown in
Figure 1.15(b), they are distributed near a two-dimensional plane. As shown in Fig-
ure 1.15(c), after projecting all points onto the plane, we can observe that, in fact,
they are distributed in a rectangle. In this example, we find that the low-dimensional
representation captures the key characteristic of the data set.
Reviewing the above example, the key step is to find a low-dimensional plane
near all points. In PCA, this problem is formalized as an optimization problem by
using linear algebra. Specifically, given a data set {xi ∈ Rd }ni=1 , PCA intends to find
a t-dimensional (t < d) plane1 that minimizes the sum of the square of the distance
between each point and its projection onto the plane. Formally, a t-dimensional plane
can be described by a semi-orthogonal matrix B = (b1 , . . . , bt ) ∈ Rd×t (i.e. BT B = I)
and a shift vector s ∈ Rd . By linear algebra, {Bz|z ∈ Rt } is a t-dimensional subspace.

1
A formal name should be affine subspace. Here we use ‘plane’ for simplicity.
36 Applications of machine learning in wireless communications

{s + Bz│z ∈ R2}

s
b1

b2
B = (b1, b2)

Figure 1.16 A two-dimensional plane {s + Bz|z ∈ R2 } in R3

Then, we shift the subspace by s and get a plane {s + Bz|z ∈ Rt }. An illustration is

shown in Figure 1.16. Furthermore, for any point x ∈ Rd , the projection of it to the
plane is given by
s + BBT (x − s). (1.50)
Correspondingly, the distance from x to its projection is given by (I − BBT )(x − s) .
Thus, denoting the data matrix as X = (x1 , . . . , xn ) ∈ Rd×n , the goal of PCA is
to find the optimal solution to

n
2
min (I − BBT )(xi − s)
s,B
i=1 (1.51)
s.t. BT B = I.
n
By taking gradient with respect to s, the optimal s is given by μ = (1/n) i=1 xi .
Replacing s by μ, we still need to solve:

n
2
min (I − BBT )(xi − μ)
B
i=1 (1.52)
s.t. B B = I.
T

Noticing that
2
(I − BBT )(xi − μ) = xi − μ2 − (xi − μ)T BBT (xi − μ), (1.53)
(1.52) is equivalent to

n

max (xi − μ)T BBT (xi − μ) = Tr X̄T BBT X̄ = Tr BT X̄X̄T B
B
i=1 (1.54)
s.t. BT B = I.
Introduction of machine learning 37

Algorithm 1.11: Principal component analysis

Input: data set X = {xi ∈ Rd }ni=1 , target dimension t (t < d)

Output: coordinate matrix in dimension-reduction space
Y = (y 1 , . . . , yn ) ∈ R
t×n

1 compute μ = 1n ni=1 xi ;
2 define X̄ = (x1 − μ, . . . , xn − μ) ∈ Rd×n ;
3 compute the first t orthogonal eigenvectors p1 , . . . , pt of X̄X̄T ;
4 define P = (p1 , . . . , pt ) ∈ Rd×t ;
5 Y = PT X̄;

where X̄ = (x1 − μ, . . . , xn − μ). Suppose that p1 , . . . , pt are t orthogonal eigen-

vectors corresponding to the t largest eigenvalues of X̄X̄T . By denoting P =
(p1 , . . . , pt ) ∈ Rd×t , the optimal solution to (1.54) is given by

B∗ = P (1.55)

according to Corollary 4.3.39 in [98]. Thus, the coordinate of xi in the dimension

reduction space is given by PT (xi − μ) = yi ∈ Rt . Finally, we conclude the whole
process in Algorithm 1.11.
PCA is effective when data distribute near a low-dimensional plane. However,
sometimes data may be approximately embedded in a non-linear structure, such as
an ellipsoid or a hyperboloid. To handle the non-linear case, kernel PCA [99] is
proposed by using the kernel trick. In addition, Generalized PCA [100] is proposed
to deal with the case where data distribute in multi low-dimensional planes. When data
is contaminated by a few noises, Robust PCA [101] has been shown to be effective
in this case. References [102] and [103] demonstrate two applications of PCA to
multivariate sampling for wireless sensor networks and wireless capsule endoscopy,
respectively.

1.2.7 Autoencoder
In this section, we will introduce another dimension-reduction method autoencoder.
An autoencoder is a neural network (see Section 1.1.3.3) used to learn an effective
representation (encoded mode) for a data set, where the transformed code has lower
dimensions compared with the original data.
As shown in Figure 1.17, an autoencoder consists of two parts, i.e. an encoder
f (·|η) and a decoder g(·|θ ). Each of them is a neural network, where η and θ denote the
parameter sets of the encoder and the decoder, respectively. Given an input x ∈ Rd ,
the encoder is in charge of transforming x into a code z ∈ Rt , i.e. f (x|η) = z, where
t is the length of the code with t < d. In contrast, the decoder tries to recover the
original feature x from the code z, i.e. g(z|θ ) = x̄ ∈ Rd such that x̄ ≈ x. Thus, given
38 Applications of machine learning in wireless communications

Input Encoding Code Decoding Output

f (x│η) = z g(z│θ) = x–

Squared
error
││x – –x││2
–
X f (.│η) Z g (.│θ) X X

Figure 1.17 The flowchart of an autoencoder. First, the encoder f (·|η) encodes an
input x into a code z with short length. Then the code z is transformed
into an output x̄ being the same size as x, by the decoder g(·|θ ). Given
a data set {xi }ni=i , the object of training is to learn
the parameter set
such that minimize the sum of squared error ni=1 xi − x̄i 2

a data set {xi }ni=1 , the training process of the autoencoder can be formulated as the
following optimization problem:

n
min xi − g( f (xi |η)|θ)2 . (1.56)
η,θ
i=1

By limiting the length of code, minimizing the object function will force the code to
capture critical structure of input features and ignore trivial details such as sparse
noises. Thus, besides dimension reduction, an autoencoder can also be used to
de-noise.
In Figure 1.18, we show a specific implementation for the dimension-reduction
task on the MNIST data set of handwritten digits [41], where each sample is a grey-
scale image with the size of 28 × 28. For simplicity, stacking the columns of each
image into a vector, thus the input has a dimension of 28 × 28 = 784. As we see,
the encoder consists of three FC layers, where each layer is equipped with a sigmoid
activation function. The first layer non-linearly transforms an input vector with 784
dimensions into a hidden vector with 256 dimensions, and the second layer continues
to reduce the dimension of the hidden vector from 256 to 128. Finally, after the
transformation of the third layer, we get a code of 64 dimensions, which is far less
than the dimension of the input vector. On the other hand, the decoder shares a same
structure with the encoder except that each FC layer transforms a low-dimensional
vector into a high-dimensional vector. The encoder tries to reconstruct the original
input vector with 784 dimensions from the code with 64 dimensions. In addition,
a sparse constraint on the parameters should be added as a regularization term to
achieve better performance. After training the autoencoder using the BP algorithm
Introduction of machine learning 39

Encoder Decoder
FC layer FC layer FC layer FC layer FC layer FC layer
784×256 256×128 128×64 64×128 128×256 256×784
Inputs Outputs
784 (28×28) 784 (28×28)

Codes
64 (8×8)

Figure 1.18 A specific implementation for the dimension-reduction task on the

MNIST data set. The encoder consists of three fully connected layers
(FC layers), where each layer is equipped with a sigmoid activation
function. The decoder shares a same structure with the encoder except
that each FC layer transforms a low-dimensional vector into a
high-dimension vector

Inputs:

Codes:

Outputs:

Figure 1.19 Partial results on the MNIST data set

(see Section 1.1.3.3), we can obtain the result shown in Figure 1.19. From this figure,
we observe that an image can be reconstructed with high quality from a small-size
code, which indicates that the main feature of the original image is encoded into the
code.
We can use an encoder to compress a high-dimensional sample into a low-
dimensional code and then use a decoder to reconstruct the sample from the code. An
interesting problem is that, can we feed a randomly generated code into the decoder
to obtain a new sample. Unfortunately, in most cases, the generated samples either
are very similar to the original data or become meaningless things. Inspired by this
idea, a variational autoencoder [104] is proposed. Different from the autoencoder, the
variational autoencoder tries to learn an encoder to encode the distribution of original
data rather than the data itself. By using a well-designed object function, the distribu-
tion of original data can be encoded into some low-dimensional normal distributions
through an encoder. Correspondingly, a decoder is trained to transform the normal
40 Applications of machine learning in wireless communications

k-Means

DBSCAN
Clustering
FDP

Unsupervised RECOME
learning
Gaussian mixture
Density estimation
model

PCA
Dimension
reduction
Autoencoder

Figure 1.20 Structure chart for unsupervised learning technologies discussed in

this chapter

distributions into real data distribution. Thus, one can first sample a code from the
normal distributions and then feed it to the decoder to obtain a new sample. For more
details about a variational autoencoder, please refer to [104]. In wireless communi-
cations and sensor networks, autoencoder has been applied in many fields, such as
data compression [105], sparse data representation [106], wireless localization [107]
and anomaly detection [108].

1.2.8 Summary of unsupervised learning

In this section, we have discussed unsupervised learning. In contrast to supervised
learning, unsupervised learning needs to discover and explore the inherent and hidden
structures of a data set without labels. As shown in Figure 1.20, unsupervised learning
tasks can be mainly divided into three categories, i.e. clustering, density estimation
and dimension reduction.
Clustering is the area studied most in unsupervised learning. We have intro-
duced four practical clustering methods, i.e. k-means, DBSCAN, FDP and RECOME.
k-means has the simplest form among four methods. It is very easy to implement and
has short running time for low-dimensional data. But the original k-means prefers to
divide data into clusters with convex shapes. DBSCAN is a density-based clustering
algorithm. It is apt at detecting clusters with different shapes. But the clustering result
of DBSCAN is sensitive to the choice of its parameters. FDP introduces a brilliant
idea, HDN, and based on it, a clustering algorithm is constructed. FDP is easy to
implement and can generate satisfactory results with cluster centres being selected
visually. RECOME is based on a novel density measure, relative NKD. It has been
Introduction of machine learning 41

Table 1.3 Summary of applications of unsupervised learning in wireless

communications

Method Function Application in

wireless communications

k-Means Clustering Collaborative signal processing [73]

Wireless surveillance systems [74]
Wireless hybrid networks [75]
DBSCAN Clustering Localization [79]
Wireless agriculture [80]
Anomaly detection [81,83]
Power backup [82]
FDP Clustering Wireless energy strategy [91,92]
RECOME Clustering MIMO channel learning [93]
Gaussian mixture Density estimation Target tracking [96]
models Signal-strength prediction [97]
Principal component Dimension reduction Multivariate sampling [37]
analysis Wireless capsule endoscopy [103]
Autoencoder Dimension reduction Wireless data representation [105,106]
Localization [107]
Anomaly detection [108]

shown to be effective on detecting clusters with various shapes, densities and scales.
In addition, it also provides an auxiliary algorithm to help users selecting parameters.
Density estimation is a basic problem in unsupervised learning. We have pre-
sented the GMM, which is one of most popular models for density estimation. The
GMM can approximate any continuous distribution to some desired degree of accu-
racy as soon as the parameter k is large enough, but accordingly, time cost for the
estimation of its parameters will increase. Dimension reduction plays an important
role in the compression, comprehension and visualization of data. We have introduced
two dimension-reduction technologies, PCA and autoencoder. PCA can be deduced
from an intuitive geometry view and has been shown to be highly effective for data
distributed in a linear structure. However, it may destroy non-linear topological rela-
tions in original data. Autoencoder is a dimension-reduction method based on neural
networks. Compared with PCA, autoencoder has great potential to reserve the non-
linear structure in original data, but it needs more time to adjust parameters for a
given data set. In Table 1.3, we summarize the applications of supervised learning in
wireless communications. For more technologies of unsupervised learning, readers
can refer to [8].

1.3 Reinforcement learning

So far, we have discussed two kinds of machine-learning methods: supervised learn-
ing, which is adapted to handle a classification or regression task, and unsupervised
42 Applications of machine learning in wireless communications

Agent

Reward rt
State st Action at
Next state st+1

Environment

Figure 1.21 Markov decision process

learning, which is used to learn underlying hidden structure of data. However, in

wireless communications, we sometimes encounter some problems of real-time con-
trolling, which is hard to be solved by supervised or unsupervised learning methods.
For example, in radio access networks, we need to dynamically turn on/off some
base stations according to the traffic load variations so as to improve energy effi-
ciency [109]. As a solution, RL is a powerful tool to deal with these real-time control
problems. In this section, we will introduce the main idea and classic approaches
of RL.

1.3.1 Markov decision process

In RL, a real-time control problem is simplified as a system where an agent and
an environment interact over time. As illustrated in Figure 1.21, at time step t, the
environment is in a state st (e.g. the traffic load variations in radio access networks).
Then, the agent takes an action at (e.g. turn on/off some base stations) according to
the state st . After that, the environment will return a reward rt (e.g. the saved energy
cost) to the agent and turn into the next state st+1 on the basis of st and at . Since the
rule of states and rewards is determined by the environment, what the agent controls
is choosing actions in accordance with states to maximize total rewards in a long
period.
The above idea can be formulated as a Markov decision process (MDP). Formally,
an MDP, represented by a tuple S, A, P, R, γ , consists of five parts:
● S is a finite set of states.
● A is a finite set of actions.
● P is a state transition probability function. P(·|s, a) gives the distribution over
next state given a pair (s, a), where s ∈ S and a ∈ A.
● R : S × A → R is a reward function.2 R(s, a) gives the reward after the agent
takes an action a in a state s.
● γ ∈ [0, 1] is a discount factor, which is introduced to discount the long-period
reward.3

2
Here we suppose the reward function is deterministic for simplicity though it can be a random function.
3
People often pay more attention to the short-term reward.
Introduction of machine learning 43

In addition, the strategy of the agent taking actions is defined as a policy π , where
π(a|s) gives the probability of the agent taking an action a in a state s. In other words,
a policy fully defines the behaviour of an agent. Given an initial state s0 and a policy
π, an MDP can ‘run’ as follows:
For t = 0, 1, 2, . . .
at ∼ π(·|st );
(1.57)
rt = R(st , at );
st+1 ∼ P(·|st , at ).
Our
∞ objective is to find a policy π ∗ that maximizes cumulative discounted award
t
t=0 γ rt on average.
To smooth the ensuing discussion, we need to introduce two functions, i.e. a value
function and a Q-value function. The value function with the policy π is defined as
the expectation of cumulative discounted reward, i.e.:
!∞ #
"
"
V π (s) E γ t rt "s0 = s , s ∈ S. (1.58)
t=0

The Q-value function (also called action-value function) is defined as

!∞ #
"
t "
Q (s, a) E
π
γ rt "s0 = s, a0 = a , s ∈ S, a ∈ A. (1.59)
t=1

Intuitively, the value function and the Q-value function evaluate how good a state and
a state-action pair are under a policy π, respectively. If we expand the summations in
the value function, we have
!∞ #
"
"
V π (s) = E γ t rt "s0 = s
t=0
! #

∞ "
"
= E r0 + γ t rt "s0 = s
t=1
! # (1.60)

∞ "
"
= π(a|s) R(s, a) + γ P(s |s, a)E γ t−1 rt "s1 = s
a∈A s ∈S t=1

= π(a|s) R(s, a) + γ P(s |s, a)V (s )
π

a∈A s ∈S

Similarly, for Q-value function, we have

Qπ (s, a) = R(s, a) + γ P(s |s, a) π (a |s )Qπ (s , a ). (1.61)
s ∈S a ∈A

Equations (1.60) and (1.61) are so-called Bellman equations, which are foundations
of RL.
44 Applications of machine learning in wireless communications

On the other hand, if we fix s and a, V π (s) and Qπ (s, a) in fact evaluate how good
a policy π is. Thus, a policy that maximizes V π (s) (Qπ (s, a)) will be a good candidate
for π ∗ though s and a are fixed. An arising problem is that there exists a policy that
maximizes V π (s) (Qπ (s, a)) for any s ∈ S (and a ∈ A). The following theorem gives
a positive answer:

Theorem 1.1. [110] For any MDP, there exists an optimal policy π ∗ such that
∗
V π (s) = max V π (s) ∀s ∈ S
π

and
∗
Qπ (s, a) = max Qπ (s, a) ∀s ∈ S and ∀a ∈ A.
π

According to Theorem 1.1, we can define the optimal value function and the
optimal Q-value function as
∗ ∗
V ∗ (·) V π (·) and Q∗ (·, ·) Qπ (·, ·) (1.62)

respectively, which are useful in finding the optimal policy. Furthermore, if V ∗ (·) and
Q∗ (·, ·) have been obtained, we can construct the optimal policy π ∗ by letting:
⎧
⎨ 1 if a = arg maxa∈A Q∗ (s, a)
∗
π (a|s) = = arg maxa∈A R(s, a) + γ s ∈S P(s |s, a)V ∗ (s) (1.63)
⎩
0 otherwise.

In other word, there always exists a deterministic optimal policy for any MDP. In
addition, we have the Bellman optimality equations as follows:

V ∗ (s) = max R(s, a) + γ P(s |s, a)V ∗ (s) (1.64)
a
s ∈S

and

Q∗ (s, a) = R(s, a) + γ P(s |s, a) max

Q∗ (s , a ). (1.65)
a ∈A
s ∈S

MDP and the Bellman equations are theoretical cornerstones of RL, based on
which many RL algorithms have been derived as we will show below. For more results
regarding MDP, readers can refer to [111].

1.3.2 Model-based methods

In this subsection, we will discuss model-based methods, where the term ‘model-
based’ means that the model of MDP (i.e. S, A, P, R, γ ) have been given as known
information. There are two typical model-based algorithms. The one is policy itera-
tion, and the other is value iteration. We will introduce the former followed by the
latter.
Introduction of machine learning 45

Algorithm 1.12: Computing value function

Input: MDP M = S, A, P, R, γ , policy π
Output: value function V π
1 Initialize V0π randomly;
2 for i = 1, 2, . . . do
3 for s ∈ S do
4

Viπ (s) ← π(a|s) R(s, a) + γ P(s |s, a)Vi−1
π
(s ) ;
a∈A s ∈S

5 if Viπ converges then

6 break;
7 V π ← Viπ

The policy iteration takes an iterative strategy to find the optimal policy π ∗ . Given
an MDP M = S, A, P, R, γ and an initial policy π , the policy iteration alternatively
executes the following two steps:

1. Computing the value function V π based on M and π .

2. Improve the policy π according to V π .

How to compute a value function? Given an MDP and a policy, the corresponding
value function can be evaluated by Bellman equation (1.60). This process is described
in Algorithm 1.12. The function sequence in Algorithm 1.12 can be proved to converge
to V π .
How to improve a policy? Given an MDP and the value function of a policy, the
policy can be improved by using (1.63). As a result, we conclude the policy iteration
algorithm in Algorithm 1.13.
The value iteration, as its name suggests, iteratively updates a value function until
it achieves the optimal value function. It has a very brief form since it just iterates
according to the optimal Bellman equation (1.64). We present the value iteration
algorithm in Algorithm 1.14. After obtaining the optimal value function, we can
construct the optimal policy by using (1.63).
In summary, when an MDP is given as known information, we can use either
of the policy and the value iteration to find the optimal policy. The value iteration
has more simple form, but the policy iteration usually converges more quickly in
practice [111]. In wireless communications, the policy and the value iteration methods
have been applied to many tasks, such as heterogeneous wireless networks [112],
energy-efficient communications [113] and energy harvesting [114].
46 Applications of machine learning in wireless communications

Algorithm 1.13: Policy iteration

Input: MDP M = S, A, P, R, γ
Output: optimal policy π ∗
1 Initialize π randomly;
2 repeat
3 Compute value function V π by using Algorithm 1.12;

1 if a = arg maxa∈A R(s, a) + γ s ∈S P(s |s, a)V π (s )
π̄(a|s) =
0 otherwise.

π ← π̄;
4 until π converges;

Algorithm 1.14: Value iteration

Input: MDP M = S, A, P, R, γ
Output: optimal policy V ∗
1 Initialize V ∗ randomly;
2 repeat
3 V̄ ∗ (s) = maxa∈A R(s, a) + γ s ∈S P(s |s, a)V ∗ (s );
4 V ∗ ← V̄ ∗ ;
5 until V ∗ not change;

1.3.3 Model-free methods

In practice, we often encounter a problem where the MDP model behind it is
unknown to us, and thus model-based algorithms are prohibited in this situation.
In this subsection, we will discuss two kinds of model-free methods: Monte Carlo
(MC) methods, temporal-difference (TD) learning, which can be applied when the
MDP model is unobservable.

1.3.3.1 Monte Carlo methods

The MC approach is a general idea derived from the law of large numbers, i.e. the
average of the results obtained from numerous samples will be close to the expected
value. It performs estimations of the value or Q-value function based on the experience
of the agent (samples).
For example, consider the value function with a policy π:
!∞ #
"
"
V π (s) = E γ t rt "s0 = s .
t=0
Introduction of machine learning 47

Algorithm 1.15: Incremental Monte Carlo estimation

Input: experience trajectory s0 , a0 , r0 , . . . , policy π , discount factor γ ,
truncation number n, iteration number m
Output: Q-value function Qπ
1 for s ∈ S do
2 for a ∈ A do
3 N (s, a) ← 0;
4 Qπ (s, a) ← 0;

5 set R = nt=0 γ t rt ;
6 for t = 0, 1, . . . , m do
7 N (st , at ) ← N (st , at ) + 1;
8 Qπ (st , at ) ← Qπ (st , at ) + N (s1t ,at ) (R − Qπ (st , at ));
9 R ← R−r γ
t
+ γ n rt+1+n ;
10 if Qπ converges then
11 break;

It is defined by the expectation of all trails starting from s. Now, suppose that
we independently conduct l experiments' by applying the policy π , and thus ( we
(i) (i) (i) (i) (i) (i)
obtain l trajectories {τi }li=1 , where τi = s ≡ s0 , a0 , r0 , s1 , a1 , r1 , . . . , rn(i)i . Let
i t (i)
R(i) = nt=0 γ rt . Then, according to the law of large numbers, we have
!∞ #
" 1 (i)
l
t "
V (s) = E
π
γ rt "s0 = s ≈ R
t=0
l i=1

1 (i) 1 1 (i)
l−1 l−1
= R + R −
(l)
R
l − 1 i=1 l l − 1 i=1

when l is large enough. Therefore, the value function can be estimated if we afford
numerous experiments. Similarly, the Q-value function can also be estimated by
using the MC method. However, in practice, we often face an online infinity trajec-
tory: s0 , a0 , r0 , . . . . In this situation, we can update the Q-value (or value) function
incrementally as shown in Algorithm 1.15. The truncation number n in Algorithm 1.15
is used to shield negligible remainders.
Once we estimate the Q-value function for a given policy π , the optimal policy
can be obtained iteratively as presented in Algorithm 1.16. Here ε is introduced to
take a chance on small probability events.
MC methods are unbiased and easy to implement. However, they often suffer from
high variance in practice since the MDP model in real world may be so complicated
that a huge amount of samples are required to achieve a stable estimation. This restricts
the usage of MC methods when the cost of experiments is high.
48 Applications of machine learning in wireless communications

Algorithm 1.16: Incremental Monte Carlo policy iteration

Input: discount factor γ , truncation number n, iteration number m, ε
Output: optimal policy π ∗
1 Initialize π randomly;
2 repeat
3 Apply π to generate a trail s0 , a0 , r0 , . . . , rn+m ;
4 Estimate Q-value function Qπ by using Algorithm 1.15;
5

(1 − ε) if a = arg maxa∈A Qπ (s, a)
π̄(a|s) =
ε/(|S| − 1) otherwise.
π ← π̄ ;
6 until π converges;

1.3.3.2 Temporal-difference learning

Like the MC approach, TD learning also tries to estimate the value or Q-value function
from the experience of the agent. But it performs an incremental estimation based on
the Bellman equations besides MC sampling.
l
To begin with, suppose that we obtain a sample set s, a(i) , r (i) , s(i) i=1 by apply-
ing a policy π. Then, by applying MC sampling to the Bellman equation of the value
function, we have
) *
V π (s) = E R(s, a) + γ V π (s )|π

1 (i)
l
≈ r + γ V π (s(i) )
l i=1
(1.66)
1 (l)
= μl−1 + r + γ V π (s(l) ) − μl−1 ,
l
1 (l)
≈ V π (s) + r + γ V π (s(l) ) − V π (s) ,
l

where μl−1 = (1/(l − 1)) l−1 i=1 r + γ V (s ). Therefore, to acquire an estimation
(i) π (i)
π
of V (s), we can update it by the fixed point iteration [111]:
1 (l)
V π (s) ← V π (s) + r + γ V π (s(l) ) − V π (s) . (1.67)
l
In practice, 1/l in (1.67) is usually replaced by a monotonically decreasing sequence.
So far, we have presented the main idea of the TD learning. The detailed steps of TD
learning are concluded in Algorithm 1.17, where the learning rate sequence should
satisfy ∞ t=0 α t = ∞ and ∞
t=0 αt < ∞.
2

Similarly, the Q-value function w.r.t. a policy can be estimated by using Algo-
rithm 1.18, which is also known as the Sarsa algorithm. Based on the Sarsa algorithm,
Introduction of machine learning 49

Algorithm 1.17: TD learning

Input: experience trajectory s0 , a0 , r0 , s1 . . . w.r.t. policy π , discount factor γ ,
learning rate sequence α0 , α1 , . . .
Output: value function V π
1 for s ∈ S do
2 V π (s) ← 0;
3 for t = 0, 1, . . . do
4 V π (st ) ← V π (st ) + αt (rt + γ V π (st+1 ) − V π (st ));

Algorithm 1.18: Sarsa algorithm

Input: experience trajectory s0 , a0 , r0 , s1 . . . w.r.t. policy π , discount factor γ ,
learning rate sequence α0 , α1 , . . .
Output: Q-value function V π
1 for s ∈ S do
2 for a ∈ A do
3 Qπ (s, a) ← 0;

4 for t = 0, 1, . . . do
5 Qπ (st , at ) ← Qπ (st , at ) + αt (rt + γ Qπ (st+1 , at+1 ) − Qπ (st , at ));

we can improve the policy alternatively by using Algorithm 1.16, where the Q-value
function is estimated by the Sarsa algorithm.
On the other hand, if choosing the optimal Bellman equation (1.65) as the
iteration strategy, we can derive the famous Q-learning algorithm as presented in
Algorithm 1.19.
In summary, TD learning, Sarsa and Q-learning are all algorithms based on
the Bellman equations and MC sampling. Among them, the goal of TD learning
and Sarsa is to estimate the value or Q-value function for a given policy, while
Q-learning aims at learning the optimal Q-value function directly. It should be noted
that, by using TD learning, we can only estimate the value function, which is not
enough to determine a policy because the state transition probability is unknown to
us. In contrast, a policy can be derived from the Q-value function, which is estimated
by Sarsa and Q-learning. In practice, Sarsa often demonstrates better performance
than Q-learning. Furthermore, all of the three methods can be improved to converge
more quickly by introducing the eligibility trace. Readers can refer to [111] for
more details.
Moreover, TD learning, Sarsa and Q-learning have been widely applied in wire-
less communications. References [115] and [116] demonstrate two applications of TD
learning in energy-aware sensor communications and detection of spectral resources,
50 Applications of machine learning in wireless communications

Algorithm 1.19: Q-learning

Input: discount factor γ , learning rate sequence α0 , α1 , . . .
Output: optimal Q-value function Q∗
1 for s ∈ S do
2 for a ∈ A do
3 Q∗ (s, a) ← 0;
4 Initialize s0 ;
5 for t = 0, 1, . . . do
6 at ∼ π (·|st ), where

(1 − ε) if a = arg maxa∈A Q∗ (s, a)
π (a|s) =
ε/(|S| − 1) otherwise.

Take action at , observe rt and st+1 ;

7 Q∗ (st , at ) ← Q∗ (st , at ) + αt (rt + γ maxa∈A Q∗ (st+1 , a) − Q∗ (st , at ));

respectively. References [117], [118] and [119] show three applications of Sarsa in
channel allocation, interference mitigation and energy harvesting, respectively. Ref-
erences [120], [121] and [122] present three applications of Q-learning in routing
protocols, power allocation and caching policy, respectively.

1.3.4 Deep reinforcement learning

So far we have discussed both model-based and model-free methods in RL. All of
these methods need to store one or two tables with size |S| (for the value function)
or |S| × |A| (for the Q-value function and the policy). In practice, however, we often
encounter the situation where |S| is very large or even infinite. In this case, it is
impractical to store a table whose size is proportional to |S|. DNNs, as discussed in
Section 1.1.3.3, have a strong ability in representation and can be used to approximate
a complex function. As a result, DNNs have been applied to approximate the value
function, the Q-value function and the policy. In this subsection, we will discuss these
approximation ideas.
1.3.4.1 Value function approximation
First of all, we consider the problem of approximating the value function by using
a DNN. For simplicity, let V̂ (s, W) denote a DNN which is with parameter W and
receives an input s ∈ S. For a given policy π, our goal is turning W so as to V̂ (s, W) ≈
V π (s), ∀s ∈ S. Unfortunately, since the true V π (·) is unknown to us, we cannot learn
W directly. Alternatively, let us consider minimizing the difference between V̂ (·, W)
and V π (·) in expectation, i.e.:
1 2
min E V̂ (s, W) − V π (s) . (1.68)
W 2 s
Introduction of machine learning 51

Algorithm 1.20: Value function approximation

Input: a sample set D = {(s(i) , r (i) , s (i) )}li=1 , batch size m, learning rate α
Output: approximate value function V̂ (·, W)
1 Initialize W;
2 repeat
3 Randomly sample a subset {(s( j) , r ( j) , s ( j) )}mj=1 from D;

4 W ← W + mα mj=1 V̂ (s( j) , W) − (r ( j) + V̂ (s ( j) , W) ∇W V̂ (s( j) , W);
5 until convergence;

As mentioned in Section 1.1.3.3, we use the gradient descent method to update W.

Taking the gradient of (1.68) w.r.t. W, we have
1 2
∇W E V̂ (s, W) − V π (s) = E V̂ (s, W) − V π (s) ∇W V̂ (s, W). (1.69)
2 s s

By applying the Bellman equation, (1.69) can be transformed into

π
E V̂ (s, W) − (r + V (s )) ∇W V̂ (s, W). (1.70)
s,r,s

However, since the true V π (s ) is unknown, we substitute V̂ (s , W) for V π (s ) and get

E V̂ (s, W) − (r + V̂ (s , W) ∇W V̂ (s, W). (1.71)
s,r,s

Now, if we have obtained a finite sample set {(s(i) , r (i) , s (i) )}li=1 from the experience,
(1.71) can be estimated as

1 (i)
l
V̂ (s , W) − (r (i) + V̂ (s , W) ∇W V̂ (s(i) , W).
(i)
(1.72)
l i=1
Thus, we can use (1.72) to update W until convergence. The value function
approximation via DNNs is summarized in Algorithm 1.20.
On the other hand, for the Q-value function, we can approximate it with a similar
way as described in Algorithm 1.21. After the value function or the Q-value function
is approximated, we can work out the optimal policy by using the policy iteration
(Algorithms 1.13 or 1.16). However, given a large size problem, a more smarter way
is to parametrize the policy by using another DNN, which will be discussed in the
following part.

1.3.4.2 Policy gradient methods

Similar to value or Q-value functions, a policy can be parametrized by using DNNs
too. However, it is non-trivial to estimate a gradient to improve the parametrized
policy. Accordingly, a policy gradient has been proposed to solve this problem. In this
part, we will discuss the policy gradient and its derivation.
52 Applications of machine learning in wireless communications

Algorithm 1.21: Q-value function approximation

Input: a sample set D = {(s(i) , a(i) , r (i) , s (i) , a (i) )}li=1 , batch size m, learning
rate α
Output: approximate value function Q̂(·, ·, U)
1 Initialize U;
2 repeat
3 Randomly sample a subset {(s( j) , a( j) , r ( j) , s ( j) , a ( j) )}mj=1 from D;
4 U←

U + mα mj=1 Q̂(s( j) , a( j) , U) − (r ( j) + Q̂(s ( j) , a ( j) , U) ∇U Q̂(s( j) , a( j) , U);
5 until convergence;

To begin with, let π̂ (s, a, θ ) denote a DNN which is with parameter θ and receive
two inputs s ∈ S and a ∈ A. Our goal is to learn the parameter θ such that the
expectation of the total reward is maximized, i.e.:
!∞ # +
"
"
max J (θ) E γ rt "π̂ (·, ·, θ) = g(τ )P{τ |π̂ (·, ·, θ )}dτ ,
t
(1.73)
θ
t=0 τ

where τ = s0 , a0 , r0 , s1 , a1 , r1 , . . . and g(τ ) = ∞ t
t=0 γ rt denote a trajectory and its
reward, respectively. To update θ , we need to take the gradient w.r.t. θ, that is
+
∇θ J (θ ) = g(τ )∇θ P{τ |π̂ (·, ·, θ)}dτ. (1.74)
τ

But the gradient in (1.74) is hard to be estimated since it relies on the probability.
Fortunately, this difficulty can be resolved by using a nice trick as follows:
+
∇θ J (θ ) = g(τ )∇θ P{τ |π̂ (·, ·, θ)}dτ
τ
+
∇θ P{τ |π̂ (·, ·, θ )}
= g(τ )P{τ |π̂(·, ·, θ )} dτ
P{τ |π̂ (·, ·, θ )} (1.75)
τ
+
= g(τ )P{τ |π̂ (·, ·, θ)}∇θ log P{τ |π̂(·, ·, θ)}dτ
τ
, " -
"
= E g(τ )∇θ log P{τ |π̂(·, ·, θ )}"π̂(·, ·, θ ) .
Introduction of machine learning 53

Moreover, we have

∞
∇θ log P{τ |π̂ (·, ·, θ)} = ∇θ log P(s0 ) π̂(st , at , θ )P(st+1 |st , at )
t=0

∞
∞
= ∇θ log P(s0 ) + log P(st+1 |st , at ) + log π̂ (st , at , θ )
t=0 t=0

∞
= ∇θ log π̂ (st , at , θ ). (1.76)
t=0

Plugging (1.76) into (1.75), we have

! #

∞ "
"
∇θ J (θ ) = E g(τ ) ∇θ log π̂ (st , at , θ )"π̂ (·, ·, θ ) . (1.77)
t=0
Equation (1.77) can be estimated by the MC approach in principle. In practice, how-
ever, it suffers from high variance because credit assignment is really hard [123].
A way to reduce the variance is to replace (1.77) by the following equation [124]:
!∞ #
"
"
∇θ J (θ ) ≈ E (Q π̂ (·,·,θ )
(st , at ) − V π̂ (·,·,θ )
(st ))∇θ log π̂ (st , at , θ )"π̂ (·, ·, θ )
t=0

1
l
≈ (Qπ̂(·,·,θ ) (s(i) , a(i) ) − V π̂ (·,·,θ ) (s(i) ))∇θ log π̂ (s(i) , a(i) , θ ), (1.78)
l
i=1
where {(s , a )}i=1
(i) (i) l
is a sample set from the experience under the policy π̂(·, ·, θ ).
So far, a remaining problem is that Qπ̂(·,·,θ) and V π̂ (·,·,θ ) are unknown to us. The
answer would be using the value and Q-value function approximations as described
in Section 1.3.4.1. We summary the whole process in Algorithm 1.22. This algo-
rithm is known as the famous actor–critic (AC) algorithm (also known as the A3C
algorithm), where actor and critic refer to the policy DNN and the value (Q-value)
DNN, respectively.
It is worth mentioning that the AC algorithm has an extension named asyn-
chronous advantage AC algorithm [125]. The A3C algorithm has better convergence
and became a standard starting point in many recent works [126].
DRL is popular with current wireless communications research. For example,
Q-value function approximation has been applied in mobile edge computing [127],
resource allocation [128] and base station control [129]. In addition, [130], [131]
and [132] demonstrate three applications of actor–critic algorithm in quality of service
(QoS) driven scheduling, bandwidth intrusion detection and spectrum management,
respectively.

1.3.5 Summary of reinforcement learning

In this section, we have discussed RL, which is an effective tool to solve real-time
control problems in various fields. As a theoretical basis of RL, the MDP theory has
provided essential concepts for the RL algorithm design, such as Bellman equations,
(optimal) value function, (optimal) Q-value function and optimal policy. As shown in
54 Applications of machine learning in wireless communications

Algorithm 1.22: Actor–critic algorithm

Input: sampling sizes l, m, learning rates α1 , α2 , α3
Output: approximate optimal policy π̂(·, ·, θ), approximate optimal value
function V̂ (·, W), approximate Q-value function Q̂(·, ·, U)
1 Initialize θ , U, Q̂(·, ·, U);
2 repeat
3 Generate a sample set D = {(s(i) , a(i) , r (i) , s (i) , a (i) )}li=1 by using policy
π̂(·, ·, θ);
4 Update W by using Algorithm 1.20 with D, m, α1 , W (without parameter
initialization);
5 Update U by using Algorithm 1.21 with D, m, α2 , U (without parameter
initialization);
6

α3
l
θ ←θ+ (Q(s(i) , a(i) , U) − V (s(i) , W))∇θ log π̂ (s(i) , a(i) , θ );
l i=1

7 until convergence;

Figure 1.22, we have introduced three parts of RL: model-based methods, model-free
methods and DRL.
Model-based methods assume that the MDP model is given as prior information.
Based on the model information and Bellman equations, this kind of algorithms try
to learn the (optimal) value function, the (optimal) Q-value function and the optimal
policy. In general, model-based algorithms have a better effect and a faster conver-
gence than model-free algorithms provided that the given MDP model is accurate.
However, model-based algorithms are rarely used in practice, since MDP models in
real world are usually too complicated to be estimated accurately.
Model-free methods are designed for the case where information of hidden MDP
is unknown. Model-free algorithms can be further divided into two subclasses: MC
methods and TD learning. Based on the law of large numbers, MC methods try to
estimate the value or Q-value function from an appropriate number of samples gener-
ated from experiments. MC methods are unbiased, but they suffer from high variance
in practice since MDP models in real world are usually complex such that it needs
massive samples to achieve a stable result. On the other hand, TD learning integrates
the Bellman equations and MC sampling in its algorithms design. By introducing
the Bellman equations, TD learning reduces the estimation variance compared with
MC methods, though its estimation may be biased. TD learning has shown a decent
performance in practice and provides basic ideas for many subsequent RL algorithms.
DRL is proposed to deal with the condition where the number of states is
extremely large or even infinite. DRL applies DNNs to approximate the value
function, the Q-value function and the policy. Among them, the update rule of the value
Introduction of machine learning 55

Policy iteration
Model-based
methods
Value iteration

Monte Carlo
estimation

Monte Carlo policy

iteration

Model-free
TD learning
methods
Reinforcement
learning
Sarsa algorithm

Q-learning

Value function
approximation

Deep
Q-value function
reinforcement
approximation
learning

Actor-critic
algorithm

Figure 1.22 Structure chart for reinforcement-learning technologies discussed

in this chapter

Table 1.4 Summary of applications of reinforcement learning

in wireless communications

Method Application in wireless communications

Policy iteration Energy harvesting [114]

Value iteration Heterogeneous wireless networks [112]
Energy-efficient communications [113]
TD learning Energy-aware sensor communications [115]
Detection of spectral resources [116]
Sarsa Channel allocation [117]
Interference mitigation [118]
Energy harvesting [119]
Q-learning Routing protocols [120]
Power allocation [121]
Caching policy [122]
Q-value function Mobile edge computing [127]
approximation Resource allocation [128]
Base station control [129]
Actor–critic algorithm QoS-driven scheduling [130]
Bandwidth intrusion detection [131]
Spectrum management [132]
56 Applications of machine learning in wireless communications

(Q-value) approximation is very similar to that of TD learning except replacing a

table with a DNN in the learning process. In this case, the parameter of the DNN
can be trained conveniently by using the gradient descent method. In contrast, the
policy approximation is more difficult since its gradient cannot be estimated directly.
Accordingly, the policy gradient is introduced which provides an approximate gradient
to update parameters. Based on the policy gradient, actor–critic (AC) algorithm is
proposed where both the actor and the critic are realized by DNNs. AC algorithm is
very practical and has become a framework of many cutting-edge RL techniques.
In Table 1.4, we summarize the applications of RL in wireless communications.
A historical survey of RL can be seen in [133], and the new developments in DRL
can be seen in [126].

1.4 Summary
In this chapter, we have reviewed three main branches of machine learning: super-
vised learning, unsupervised learning and RL. Supervised learning tries to learn a
function that maps an input to an output by referring to a training set. A supervised
learning task is called a classification task or a regression task according to whether
the predicted variable is categorical or continuous. In contrast, unsupervised learning
aims at discovering and exploring the inherent and hidden structures of a data set
without labels. Unsupervised learning has three main functions: clustering, density
estimation and dimension reduction. RL is commonly employed to deal with the opti-
mal decision-making in a dynamic system. By modelling the problem as the MDP, RL
seeks to find an optimal policy. An RL algorithm is called a model-based algorithm or
a model-free algorithm depending on whether the MDP model parameter is required
or not. Furthermore, if an RL algorithm applies DNNs to approximate a function, it
also called a deep RL method.
There is no doubt that machine learning is achieving increasingly promising
results in wireless communications. However, there are several essential open-
research issues that are noteworthy in the future [59]:
1. In general, supervised models require massive training data to gain satisfying
performance, especially for deep models. Unfortunately, unlike some popular
research areas such as computer vision and NLP, there still lacks high-quality
and large-volume-labelled data sets for wireless applications. Moreover, due
to limitations of sensors and network equipment, wireless data collected are
usually subjected to loss, redundancy, mislabelling and class imbalance. How
to implement supervised learning with limited low-quality training data is a
significant and urgent problem in the research of wireless learning.
2. On the other hand, wireless networks generate large amounts of data every
day. However, data labelling is an expensive and time-consuming process. To
facilitate the analysis of raw wireless data, unsupervised learning is increas-
ingly essential in extracting insights from unlabelled data [134]. Furthermore,
recent success in generative models (e.g. variational autoencoder and generative
adversarial networks) greatly boosts the development of unsupervised learning.
Introduction of machine learning 57

It will be worthwhile and beneficial to employ these new technologies to handle

unsupervised tasks in wireless communications.
3. Currently, many wireless network control problems have been solved by con-
strained optimization, dynamic programming and game theory approaches.
These methods either make strong assumptions about the objective functions
(e.g. linearity or convexity) or sample distribution (e.g. Gaussian or Poisson dis-
tributed), or endure high time and space complexity. Unfortunately, as wireless
networks are getting increasingly complex, such assumptions become unrealistic
sometimes. As a solution, DRL is a powerful tool to handle complex control
problems. Inspired by its remarkable achievements in self-driving [135] and the
game of Go [136], a few researchers start to apply DRL to solve the wireless
network control problems. However, this work only demonstrates a small part of
DRL’s advantages, and its potential in wireless communications remains largely
unexplored.

Acknowledgement
This work is supported in part by the National Natural Science Foundation of China
(Grant No. 61501022).

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Chapter 2
Machine-learning-enabled channel modeling
Chen Huang1 , Ruisi He2 , Andreas F. Molisch3 ,
Zhangdui Zhong2 , and Bo Ai2

In this chapter, we present an introduction to the use of machine learning in wireless

propagation channel modeling. We also present a survey of some current research
topics that have become important issues for 5G communications.

2.1 Introduction
Channel modeling is one of the most important research topics for wireless com-
munications, since the propagation channel determines the performance of any
communication system operating in it. Specifically, channel modeling is a process of
exploring and representing channel features in real environments, which reveals how
radio waves propagate in different scenarios. The fundamental physical propagation
processes of the radio waves, such as reflections, diffractions, are hard to observe
directly, since radio waves typically experience multiple such fundamental interac-
tions on their way from the transmitter to the receiver. In this case, channel modeling
is developed to characterize some effective channel parameters, e.g., delay dispersion
or attenuation, which can provide guidelines for the design and optimization of the
communication system.
Most channel models are based on measurements in representative scenarios.
Data collected during such measurement campaigns usually are the impulse response
or transfer function for specific transmit and receive antenna configurations. With
the emergence of multiple-input–multiple-output (MIMO) systems, directional char-
acteristics of the channels can be extracted as well. In particular for such MIMO
measurements, high-resolution parameter estimation (HRPE) techniques can be
applied to obtain high-accuracy characteristics of the multipath components (MPCs).
Examples for HRPE include space-alternating generalized expectation-maximization
(SAGE) [1], clean [2], or joint maximum likelihood estimation (RiMAX) [3].

1
School of Computer and Information Technology, Beijing Jiaotong University, China
2
State Key Lab of Rail Traffic Control and Safety, Beijing Jiaotong University, China
3
Department of Electrical Engineering, University of Southern California, USA
68 Applications of machine learning in wireless communications

Machine learning, as an important branch of artificial intelligence, is considered

to be a powerful tool to analyze measurement data, understand propagation processes,
and create models. This is especially true for learning the principles and properties
in channel measurement data from modern measurement campaigns, since the data
volume and dimensionality of the measurement data have increased rapidly with the
advent of massive MIMO systems. Therefore, machine-learning-based channel mod-
eling has become a popular research topic. Typical applications of machine learning
in channel modeling include

● Propagation scenario classification. Classification of the scenarios is an impor-

tant part for the channel modeling and communication system deployment, since
the channel models, or at a minimum their parameterizations, depend on the con-
sidered environment and scenario. For example, most models show a difference
between line-of-sight (LOS) and non-line-of-sight (NLOS) scenarios. Current
solutions for LOS/NLOS classification are generally based on different met-
rics, e.g., the K-factor [4], the root-mean-square delay spread and mean excess
delay [5], or the Kurtosis of the channel state information [6]. However, the clas-
sification of the scenarios by using a binary hypothesis test based on single metric
is not accurate enough for variable environments in wireless communications. On
the other hand, some machine-learning techniques, e.g., support vector machines
(SVM) and deep learning, which have a great advantage for extracting data fea-
tures, can be used for scenario classification as well. In this case, learning and
extracting the difference of channel properties in the different scenarios helps to
automatically separate the measured data into different scenarios and discover
the scenario features for resource allocation, system optimization, or localiza-
tion. In this chapter, we present some first result on the machine-learning-based
LOS/NLOS scenarios identification.
● Machine-learning-based MPC clustering. A large body of MIMO measure-
ments has shown that the MPCs occur in groups, also known as clusters, such
that the MPCs within one group have similar characteristics but have significantly
different characteristics from the MPCs in other clusters. Separately characteriz-
ing the intra-cluster and intercluster properties can allow to significantly simplify
channel models without major loss of accuracy. Therefore, many channel mod-
els have been proposed and developed based on the concept of clusters, e.g.,
Saleh–Valenzuela (SV) [7], COST 259 [8,9] COST 2100 [10], 3GPP spatial
channel model [11], and WINNER model [12]. In the past, visual inspection has
been widely used for cluster identification, which is inapplicable for extensive
measurement data, and also subjective. Automated algorithms, in particular the
KPowerMeans algorithm [13], have gained popularity in recent years but still suf-
fer from the use of arbitrary thresholds and/or a priori assumption of the number
of clusters. Moreover, many clustering approaches require to extract the MPCs
via HRPE methods before clustering. These algorithms have generally high com-
putational complexity, which makes real-time operation in time-varying channels
difficult. Hence, automatic clustering for MPCs based on machine-learning
algorithms has drawn a lot of attention.
Machine-learning-enabled channel modeling 69

● Automatic MPC tracking. Time-variation of the propagation is relevant for many

applications such as high-speed railways and vehicular communications. How-
ever, in time-varying channels, the MPCs need to be not only clustered but also
tracked to characterize their dynamic features. Machine-learning-based tracking
algorithms can be adopted or developed to track the MPCs during consecutive
snapshots, e.g., Kalman filters [14,15], or the Kuhn–Munkres algorithm [16].
However, the MPCs’ behaviors during the time, e.g., split, merge, and lifetime,
are still not fully utilized in the current tracking procedures. Hence, how to track
the MPCs more accurately and efficiently is still an open question.
● Deep-learning-based channel modeling approach. The main goal of channel
modeling is to find the interconnections among the transmitted signals, envi-
ronments, and the received signals and model them by appropriate functions.
Meanwhile, with the dramatic development of artificial intelligence, neural-
network-based deep learning has shown great performance on characterizing
data and extracting the mapping relationship between system input and out-
put [17]. Therefore, many studies pay attention to modeling the channels by
using neural network. For example, a back-propagation (BP) network is used for
modeling the amplitude in propagation channels in [18], a radial-basis-function
(RBF)-based neural network is used for modeling the Doppler frequency shift
in [19]. Moreover, some other data-mining approaches can be adopted to pre-
process the measured data, which makes the data easier to be analyzed and
processed.

In this chapter, we introduce recent progress of the above applications for machine
learning in channel modeling. The results in this chapter can provide references to
other real-world measurement data-based channel modeling.

2.2 Propagation scenarios classification

In this section, machine-learning-based propagation scenario classification is intro-
duced. Generally, different channel models are used for different typical scenarios.
In particular, most models are developed based on different propagation assumptions
and parameter settings. At the same time, some machine-learning algorithms are able
to learn the interconnections and features of different training data and then refine
them into classification principles, which can automatically classify the input data
in applications. Due to the good accuracy of the classification, the machine-learning
approaches are expected to extract the features and properties of the different channels
and automatically distinguish the propagation scenarios. There are many machine-
learning algorithms that have been widely used for classification, e.g., SVM or deep
learning.
A particularly important example for classification problems is the identification
of LOS/NLOS scenarios, which is a binary classification problem. The SVM is one
of the promising solutions for such binary classification problems and offers a good
trade-off between accuracy and complexity compared to deep learning. Therefore, we
70 Applications of machine learning in wireless communications

investigate in the following in more detail the application of the SVM to distinguish
LOS/NLOS scenarios based on the channel properties.
The main goal of the algorithm described in the following is to use the machine-
learning tool, i.e., the SVM, to learn the internal features of the LOS/NLOS
parameters, which can be obtained by using parameters estimation algorithms, e.g.,
beamformers, and build an automatic classifier based on the extracted features. Con-
sequently, there are two main steps of the proposed algorithm: (i) develop the input
vector for the SVM method and (ii) adjust the parameters of the SVM method to
achieve a better accuracy of classification.

2.2.1 Design of input vector

In the described algorithm, the power-angle-spectrums (PASs) of the LOS/NLOS
scenarios obtained by using the Bartlett beamformer [20] are used as the training
database. Figure 2.1(a) and (b) depicts example PASs of the LOS and NLOS scenario,
respectively, where the data are collected from 30×4 MIMO measurements at a carrier
frequency of 5.3 GHz and estimated by using the Bartlett beamformer [20]. Since the
SVM can only use a vector as the input for training, the design of the input vector is

70
–55
60
Elevation (degree)

50
–60
40
30
–65
20
10
–70
20 40 60 80 100 120 140 160 180
(a) Azimuth (degree)
70
–67
60 –68
Elevation (degree)

50 –69
40 –70
–71
30
–72
20
–73
10 –74

20 40 60 80 100 120 140 160 180

(b) Azimuth (degree)

Figure 2.1 Power angle spectrum of (a) LOS and (b) NLOS scenarios, which are
estimated by using Bartlett beamformer
Machine-learning-enabled channel modeling 71

400
LOS
NLOS
300
Occurrence

200

100

0
–75 –70 –65 –60 –55
Power (dB)

Figure 2.2 Histograms of the power distribution of the LOS and NLOS scenarios,
respectively

crucial for the performance of the SVM. In the described algorithm, the SVM is used
to learn the difference between the LOS and NLOS from the classified data (training
data) and distinguish the LOS and NLOS condition of the unclassified data (test data).
In this case, an input vector that is able to most clearly present the physical features
of the LOS/NLOS data can achieve the best classification accuracy.
In order to design an appropriate input vector, we first consider the main differ-
ence of physical features between the MPCs in the LOS and NLOS scenarios. First,
the average power is usually different, where the LOS scenario usually has higher
power. Second, the power distribution is another noteworthy difference between the
LOS and NLOS scenarios. Since the LOS path is blocked in the NLOS scenario, the
impact of MPCs undergoing reflections, scatterings, and diffusions is more signifi-
cant in the NLOS case. In other words, even if all such indirect MPCs are exactly
the same in LOS and NLOS scenarios, the existence of the direct MPC changes the
power distribution.
From the above, it follows that the histogram of the power is a characteristic that
can be used to distinguish the LOS/NLOS scenarios, where the abscissa represents
different power intervals, and the ordinate represents how many elements in the PAS
distribute in the different power intervals. Furthermore, to simplify the feature vector,
the number of power intervals is set at 100, with a uniform distribution in the range
of the power of the PAS, as shown in Figure 2.2. In this case, the histogram of the
power is considered as the input vector X, which can be expressed as
X = {x1 , x2 , . . . , x100 } (2.1)

2.2.2 Training and adjustment

The thus-obtained input vector can now be fed into the SVM method. Nevertheless,
the typical linear-kernel-function-based SVM cannot achieve the best accuracy of
classification, considering that the physical features of the LOS/NLOS scenario are
72 Applications of machine learning in wireless communications

generally complicated to characterize. Consequently, we use here the RBF as the

kernel function, which can be expressed as

x − xi 2
k(x, xi ) = exp − (2.2)
δ2

By using the RBF kernel function, the training data are projected to a higher dimen-
sion, in which the difference between the LOS and NLOS data can be observed more
easily.
In this case, the histogram of the power in each PAS is considered as the feature
vector for the input of the SVM to distinguish the LOS and NLOS scenarios. Based
on our experiments, the described solution achieves nearly 94% accuracy on the
classification.
In addition, the angle (azimuth/elevation) distribution of the power is also gener-
ally considered to be different in the LOS and NLOS scenarios. Since there is no LOS
component in the NLOS scenario, it will more concentrate on reflections and scat-
terings in the environment, which leads to a lower average power and smaller power
spread in the histograms. Therefore, utilizing the angle distribution in the feature
vector may also increase the classification accuracy of the solution.

2.3 Machine-learning-based MPC clustering

As outlined in the introduction, modeling of inter- and intra-cluster properties, instead
of the properties of the individual MPCs, offers an attractive trade-off between accu-
racy and complexity of the models and is thus widely used in the literature [21,22].
The basic requirement for such models is to identify clusters in measured data. In the
past, visual inspection has been widely used to recognize the clusters. However, it is
inapplicable to the analysis of large amounts of high-dimensional measurement data,
which commonly are encountered in particular in MIMO measurement campaigns.
Besides, visual inspection is a subjective approach, thus different inspectors may
provide different clustering results, which makes comparisons between results from
different groups difficult.
On the other hand, clustering is one of the most fundamental applications for
machine learning. Therefore, machine-learning-based clustering algorithms have
become a hot topic and are expected to be able to automatically cluster MPCs with high
accuracy. The main challenges of automatic clustering of MPCs include the follow-
ing: (i) the definition of MPCs’ cluster has not been addressed clearly; (ii) the ground
truth of MPCs’ clusters is generally unknown, which makes it difficult to validate the
clustering result; (iii) the number of clusters, which is required for many machine-
learning-clustering methods, is usually unknown; and (iv) the dynamic changes of
MPCs that occur in time-varying channels are difficult to utilize in many clustering
algorithm. To provide a benchmark, in the following we describe some widely used
classical MPC clustering algorithms.
Machine-learning-enabled channel modeling 73

2.3.1 KPowerMeans-based clustering

The KPowerMeans algorithm described in [13] is one of the most popular clustering
approaches for MPCs in the radio channels. The key idea of KPowerMeans is based
on the conventional KMeans method, which is a typical hard partition approach and
clusters data objects based on the distance among each other. Similar to KMeans, the
KPowerMeans requires the number of clusters as prior information, an indeterminate
cluster number may have an impact on the performance of clustering. While a number
of different methods have been described, the most straightforward way is to compute
results with different cluster numbers and compare the results. The main idea of
KPowerMeans is summarized in the following subsection.

2.3.1.1 Clustering
Figure 2.3(a)–(d) shows the four stages in the iteration of clustering. The dots and
blocks in (a) present the input MPCs and initialed cluster-centroids, respectively,
whereas the different colors of the dots in (b)–(d) represent different categories
of clusters. The KPowerMeans algorithm requires the number of clusters as prior

Cluster-
MPC centroid
AoA

AoA

Cluster-
centroid
(a) AoD (b) AoD
AoA
AoA

(c) AoD (d) AoD

Figure 2.3 The clustering framework of the KPowerMeans algorithm, where

(a)–(d) is four stage in the iteration of clustering. The dots and blocks in
(a) present input objects and initialed cluster-centroids, respectively,
whereas the different colors of the dots in (b)–(d) represent different
categories of clusters
74 Applications of machine learning in wireless communications

information, e.g., the blue and red blocks in Figure 2.3(a) and then clusters the MPCs
preliminarily to the closest cluster-centroid, as shown in Figure 2.3(b). To accurately
measure the similarity between MPCs/clusters, the MPCs distance (MCD) is used
to measure the distance between MPCs and cluster-centroids, where the angle of
arrival (AoA), angle of departure (AoD) and delay of the MPCs/cluster-centroids are
considered. The MCD between the ith MPC and the jth MPC can be obtained as

MCDij = MCDAoA,ij 2 + MCDAoD,ij 2 + MCD2τ ,ij (2.3)

where
⎛ ⎞ ⎛ ⎞
sin(θi ) cos(ϕi ) sin(θj ) cos(ϕj )

1 ⎜ ⎟ ⎜ ⎟
MCDAoA/AoD,ij = ⎝ sin(θi ) sin(ϕi ) ⎠ − ⎝ sin(θj ) sin(ϕj ) ⎠ , (2.4)
2
cos(θi ) cos(θj )

|τi − τj | τstd
MCDτ ,ij = ζ · · , (2.5)
τmax τmax
with τmax = maxi,j {|τi − τj |} and ζ as an opportune delay scaling factor; various
ways to select this scaling factor have been described in the literature. After the
MPCs are clustered preliminarily, the cluster-centroids are recomputed, as shown in
Figure 2.3(c). Then, the cluster members and the cluster-centroids are alternately
recomputed in each iteration, until the data converge to stable clusters or reach the
restraint of a preset running time.

2.3.1.2 Validation
To avoid the impact of an indeterminate number of the clusters, [13] develops the
CombinedValidate method based on the combination of the Calinski–Harabasz (CH)
index and the Davies–Bouldin criterion (DB). The basic idea of CombinedValidate is
to restrict valid choices of the optimum number of clusters by a threshold set in the
DB index. Subsequently, the CH index is used to decide on the optimum number out
of the restricted set of possibilities.

2.3.1.3 Cluster pruning—ShapePrune

After successfully finding the optimum number of the clusters, the ShapePrune clus-
ter pruning algorithm is adopted for discarding outliers. The basic idea of ShapePrune
is to remove data points that have the largest distance from their own cluster-centroid
with the constraint that cluster power and cluster spreads must not be changed sig-
nificantly. In this case, the features of clusters can be more easily observed, where
the clusters’ properties can also be preserved as much as possible. Figure 2.4 shows
the extracted MPCs from the MIMO measurement data in [13], where the power of
MPCs is color coded. By applying the KPowerMeans, the MPCs can be automati-
cally clustered without human interaction, as shown in Figure 2.5. Compared with
Machine-learning-enabled channel modeling 75

–40

29.5
–45
29
28.5
Delay (ns)

P (dB)
–50
28
27.5
27 –55

26.5

2 –60
2
0
0
AoD (rad) –2 –2 AoA (rad)

Figure 2.4 The unclustered MIMO measurement data in LOS scenario from [13],
where the power of MPCs is color coded

29.5
29
28.5
Delay (ns)

28
27.5
27
26.5
2
0 2
0
AoD (rad) –2 –2
AoA (rad)

Figure 2.5 The result of clustering from [13], where the weak MPCs are removed

the visual inspection, the KPowerMeans can well identify the clusters closest to each
other, e.g., the clusters in red, yellow, blue, and green in Figure 2.4.

2.3.1.4 Development
It is noteworthy that the initial parameters, e.g., cluster number and position of ini-
tial cluster-centroid, have a great impact on the performance of KPowerMeans. In
KPowerMeans, the validation method is applied to select the best estimation of the
76 Applications of machine learning in wireless communications

number of the clusters; thus the performance of the validation method also affects
the performance and efficiency of clustering. In [23], a performance assessment of
several cluster validation methods is presented. There it was found that the Xie–Beni
index and generalized Dunnes index reach the best performance, although the result
also shows that none of the indices is able to always predict correctly the desired
number of clusters. Moreover, to improve the efficiency of clustering, [24] devel-
ops the KPowerMeans by using the MPCs that have the highest power as the initial
cluster-centroids. On the other hand, the study in [25] claims that as a hard partition
approach, KMeans is not the best choice for clustering the MPCs, considering that
some MPCs are located near the middle of more than one cluster, and thus cannot be
directly associated with a single cluster. Therefore, instead of using hard decisions as
the KPowerMeans, a Fuzzy-c-means-based MPC clustering algorithm is described
in [25], where soft information regarding the association of multipaths to a centroid
are considered. As the result, the Fuzzy-c-means-based MPC clustering algorithm
performs a robust and automatic clustering.

2.3.2 Sparsity-based clustering

In this subsection, the framework of a sparsity-based MPC clustering algorithm [26]
is introduced, which was described to cluster channel impulse responses (CIRs) con-
sisting of multiple groups of clusters. The key idea of the described algorithm is to
use a sparsity-based optimization to recover the CIRs from measured data and then
use a heuristic approach to separate the clusters from the recovered CIRs. The main
idea can be summarized as follows [26,27]:
The CIRs are assumed to follow the SV model [7], i.e., the power of MPCs
generally decreases with the delays as follows:

2 2
α l,k = α 0,0 · exp − Tl · exp − τl,k (2.6)

l

A1 A2

where A1 and A2 denote the intercluster and intra-cluster power decay, respectively;
α 0,0 2 denotes the average power of the first MPC in the first cluster. and
l are
the cluster and MPC power decay constants, respectively.
Then, the measured power delay profile (PDP) vector P is considered as the
given signal, and the convex optimization is used to recover an original signal vec-
tor P̂, which is assumed to have the formulation (2.6). Furthermore, re-weighted l1
minimization [28], which employed the weighted norm and iterations, is performed
to enhance the sparsity of the solution.
Finally, based on the enhanced sparsity of P̂, clusters are identified from the curve
of P̂. Generally, each cluster appears as a sharp onset followed by a linear decay, in
the curve of the P̂ on a dB-scale. Hence, the clusters can be identified based on this
feature, which can be formulated as the following optimization problem:

min P − P̂22 + λ2 · 1 · P̂0 (2.7)

P̂
Machine-learning-enabled channel modeling 77

where · x denotes the lx norm operation, and the l0 norm operation returns the
number of nonzero coefficients. λ is a regularization parameter, and 1 is the finite-
difference operator, which can be expressed as

⎛ τ τ ⎞
− 0 ··· ··· 0
⎜ |τ1 − τ2 | |τ1 − τ2 | ⎟
⎜ ⎟
⎜ ⎟
⎜ τ τ ⎟
⎜ 0 − · · · · · · 0 ⎟
⎜ |τ − τ | |τ − τ | ⎟
⎜ 2 3 2 3 ⎟
⎜ .. .. .. .. .. .. ⎟
⎜
1 = ⎜ ⎟
. . . . . . ⎟
⎜ ⎟
⎜ .. τ τ ⎟
⎜ 0 0 . − 0 ⎟
⎜ |τN −2 − τN −1 | |τN −2 − τN −1 | ⎟
⎜ ⎟
⎜ ⎟
⎝ τ τ ⎠
0 0 ··· ··· −
|τN −1 − τN | |τN −1 − τN | (N −1)×N

(2.8)

where N is the dimension of P and P̂, τ is the minimum resolvable delay difference
of data. 2 is used to obtain the turning point at which the slope changes significantly
and can be expressed as
⎛ ⎞
1 −1 0 ··· ··· 0
⎜0 1 −1 · · · · · · ⎟
0
⎜ ⎟
⎜ .. . . .. .. .. ⎟
..
2 = ⎜
⎜. . . . . ⎟
⎟. (2.9)
⎜ .. ⎟
⎝0 0 . 1 −1 0⎠
0 0 ··· ··· 1 −1 (N −2)×(N −1)

Note that λ2 · 1 · P̂0 in (2.7) is used to ensure that the recovered P̂ conform
with the anticipated behavior of A2 in (2.6). In this case, even a small number of
clusters can be well identified by using the described algorithm. Moreover, [26] also
incorporates the anticipated behavior of A1 in (2.6) into P̂ by using a clustering-
enhancement approach.
The details of the implementation of the sparsity-based clustering algorithm can
be found in [26]. To evaluate the performance, Figure 2.6(a) gives the cluster identifi-
cation result by using the sparsity-based algorithm, while (b) and (c) give the results
by using KMeans and KPowerMeans approaches, respectively. It can be seen that
the clusters identified by the sparsity-based algorithm show more distinct features,
where each cluster begins with a sharp power peak and ends with a low power valley
before the next cluster. This feature well conforms to the assumption of the cluster
in the SV model. On the other hand, as shown in Figure 2.6(b) and (c), the KMeans
and KPowerMeans tend to group the tail of one cluster into the next cluster, which
may lead to the parameterized intra-cluster PDP model having a larger delay spread.
More details and further analysis can be found in [26].
78 Applications of machine learning in wireless communications

–60 –60 –60

–70 –70 –70
PDP (dB)

PDP (dB)

PDP (dB)
–80 –80 –80
–90 –90 –90
–100 –100 –100
–110 –110 –110
–120 –120 –120
0 100 200 300 0 100 200 300 0 100 200 300
(a) Delay (ns) (b) Delay (ns) (c) Delay (ns)

Figure 2.6 Example plots of PDP clustering in [26], where (a) gives the cluster
identification result by using the sparsity-based algorithm, (b) and (c)
give the results by using KMeans and KPowerMeans approaches,
respectively. Different clusters are identified by using different colors,
where the magenta lines represent the least squared regression of
PDPs within clusters

2.3.3 Kernel-power-density-based clustering

In this section, the framework of the Kernel-power-density (KPD)-based algorithm is
introduced, which was described in [29,30] to cluster the MPCs in MIMO channels.
In this algorithm, the Kernel density of MPCs is adopted to characterize the modeled
behavior of MPCs, where the powers of MPCs are also considered in the clustering
process. Moreover, the relative density is considered, using a threshold to determine
whether two clusters are density reachable.
To better elaborate, an example [29] is given in Figure 2.7, where (a) shows the
measured MPCs, (b) shows the estimated density ρ, (c) shows the estimated density
ρ ∗ , and (d) gives the final clustering results by using the KPD algorithm. The details
are introduced as follows:

1. The KPD-based algorithm identifies clusters by using the Kernel density to iden-
tify the clusters; therefore, the density needs to be calculated first. For each MPC
x, the density ρ with the K nearest MPCs can be obtained as follows:

|τx − τy |2 |T ,x − T ,y |
ρx = exp(αy ) × exp − × exp −
y∈Kx
(στ )2 σT

|T ,x − T ,y | |R,x − R,y |
× exp − × exp − (2.10)
σT σR

where y is an arbitrary MPC (y = x). Kx is the set of the K nearest MPCs for the
MPC x. σ(·) is the standard deviation of the MPCs in the domain of (·). Specif-
ically, past studies have modeled with good accuracy the intra-cluster power
angle distribution as Laplacian distribution [31]; therefore, the Laplacian Kernel
density is also used for the angular domain in (2.10).
Machine-learning-enabled channel modeling 79

Measurement Density ρ

300 300
Delay (ns)

Delay (ns)
200 200 15
–10
100 –20 100 10
5
0 –30 0
50 50
100 100
0 0 0 0
AoD (degree) –50 –100 AoA (degree) AoD (degree) –50 –100 AoA (degree)
(a) (b)
Relative density ρ* Clustering results

300 1 300
Delay (ns)

5 Delay (ns)
200 2 200
0.8
8
100 3 0.6 100
0.4
0 0.2 0
50 4 50
7 100 100
0 6 0 0 0
AoD (degree) –50 –100 AoA (degree) AoD (degree) –50 –100 AoA (degree)
(c) (d)

Figure 2.7 Illustration of KPD clustering using the measured MPCs: Part (a)
shows the measured MPCs, where the color bar indicates the power of
an MPC. Part (b) plots the estimated density ρ, where the color bar
indicates the level of ρ. Part (c) plots the estimated density ρ ∗ , where
the color bar indicates the level of ρ ∗ . The eight solid black points
are the key MPCs with ρ ∗ = 1. Part (d) shows the clustering results
by using the KPD algorithm, where the clusters are plotted with
different colors

2. In the next step, the relative density ρ ∗ also needs to be calculated based on the
obtained density ρx , which can be expressed as
ρx
ρx∗ = . (2.11)
maxy∈Kx ∪{x} {ρy }
Figure 2.7 shows an example plot of the relative density ρ ∗ . Specifically, the
relative density ρ ∗ in (2.11) can be used to identify the clusters with relatively
weak power.
3. Next, the key MPCs need to be obtained. An MPC x will be labeled as key MPC
x̂ if ρ ∗ = 1:
ˆ = {x|x ∈ , ρx∗ = 1}.
(2.12)
80 Applications of machine learning in wireless communications

In the described algorithm, the obtained key MPCs are selected as the initial
cluster-centroids. Figure 2.7(c) gives an example of the key MPCs, which are
plotted as solid black points.
4. The main goal of the KPD algorithm is to cluster MPCs based on the Kernel den-
sity, therefore, for each non-key MPC x, we define its high-density-neighboring
MPC x̃ as
x̃ = arg min d(x, y) (2.13)
y∈,
ρy >ρx

where d represents the Euclidean distance. Then, the MPCs are connected based
on their own high-density-neighboring x̃ and the connection is defined as
px = {x → x̃} (2.14)
and thus a connection map ζ1 can be obtained as follows:
ζ1 = {px |x ∈ }. (2.15)
In this case, the MPCs that are connected to the same key MPC in ζ1 are grouped
as one cluster.
5. For each MPC, the connection between itself and its K nearest MPCs can be
expressed as follows:
qx = {x → y, y ∈ Kx } (2.16)
where another connectedness map ζ2 can be obtained, as follows:
ζ2 = {qx |x ∈ }. (2.17)
In this case, two key MPCs clusters will be merged into a new cluster if the following
criteria are met:
● The two key MPCs are included in ζ2
● Any MPC belonging to the two key MPCs’ clusters has ρ ∗ > χ
where χ is a density threshold. As shown in Figure 2.7(c), clusters 2 and 3, 6 and 7
meet the conditions and are merged into new clusters, respectively.
To validate the performance of the clustering result, the F-measure is used in [29],
where the precision and recall of each cluster are considered. It is noteworthy that the
validation by using F-measure requires the ground truth of the cluster members. Gen-
erally, the ground truth is unavailable in measured channels; hence, the F-measure
can be only applied for the clustering result of simulated channel, for which the
(clustered) MPC generation mechanism, and thus the ground truth, is known. The
3GPP 3D MIMO channel model is used to simulate the channels in [29], and 300
random channels are simulated to validate the performance of the KPD-based algo-
rithm, where the conventional KPowerMeans [13] and DBSCAN [32] are shown as
comparisons. Figure 2.8 depicts the impact of the cluster number on the F-measure,
where the described algorithm shows better performance than the others, especially
in the scenarios containing more clusters, and the clustering performances of all three
reduce with the increasing number of clusters.
Machine-learning-enabled channel modeling 81

0.9

0.8

0.7

0.6 KPD
F measure

KPM
0.5
DBSCAN
0.4

0.3

0.2

0.1

0
4 6 8 10 12 14 16 18 20 22 24
Cluster number

Figure 2.8 Impact of cluster number on the F measure in [29]

1
KPD
0.9
KPM
0.8 DBSCAN

0.7
Silhouette coefficient

0.6

0.5

0.4

0.3

0.2

0.1

0
4 6 8 10 12 14 16 18 20 22 24
Cluster number

Figure 2.9 Impact of cluster angular spread on the F measure in [29]

Figure 2.9 shows the impact of the cluster angular spread on the F-measure of the
three algorithms. It is found that the F-measure generally decreases with the increasing
cluster angular spread, where the KPD-based algorithm shows the best performance
among the three candidates. Further validation and analysis can be found in [29].
82 Applications of machine learning in wireless communications

2.3.4 Time-cluster-spatial-lobe (TCSL)-based clustering

This section describes the time-cluster (TC)-spatial-lobe (SL) (TCSL) algorithm
described in [33] for 3D-millimeter-wave statistical channel models, where the
NYUSIM channel simulator [34] is implemented. A comparison to measured field
data yielded a fit to 2D and 3D measurements using the TCSL algorithm, and it is
found that TCSL can well fit the measurement data from urban NYC at mmWave using
directional antennas with a lower complexity structure compared to other classical
joint time-space modeling approaches [33,35–38]. The TCSL approach uses a fixed
intercluster void interval representing the minimum propagation time between likely
reflection or scattering objects. The framework of the TCSL algorithm is described
in the following subsection.

2.3.4.1 TC clustering
In [33], the TCs are defined as a group of MPCs that have similar runtime and sepa-
rated from other MPCs by a minimum interval, but which may arrive from different
directions. Specifically, the minimum intercluster void interval is set to 25 ns. In other
words, the MPCs whose inter-arrival time is less than 25 ns are considered as one
TC; otherwise, they are considered as different TCs. Besides, the propagation phases
of each MPC can be uniform between 0 and 2π. The choice of different intercluster
voids results in different number of clusters in the delay domain; to be physically
meaningful, this parameter needs to be adapted to the environment of observation.

2.3.4.2 SL clustering
Meanwhile, SLs are defined by the main directions of arrival/departure of the signal.
Since the TCSL is based on measurements without HRPE of the MPCs, the angular
width of an SL is determined by the beamwidth of the antenna (horn or lens or phased
array) and measured over several hundred nanoseconds. A −10 dB power threshold
with respect to the maximum received angle power is set in [33] to obtain the SLs
(again, different thresholds might lead to different clusterings).
By applying the TCs and SLs, the MPCs in the time-space domain are decoupled
into temporal and spatial statistics. Since the SLs and TCs are obtained individually,
it is possible that a TC contains MPCs which belong to different SLs. On the contrary,
an SL may contain many MPCs which belong to different TCs. These cases have
been observed in real-world measurements [35–37], where the MPCs in the same TC
may be observed in different SLs, or the MPCs in the same SL may be observed in
different TCs.
The TCSL-clustering approach has low complexity, and some of its parameters
can be related to the physical propagation environment [33]. However, it requires
some prior parameters, such as the threshold to obtain the SLs, the delays and power
levels of TC.

2.3.5 Target-recognition-based clustering

As we mentioned before, many current clustering algorithms for channel modeling
are based on the characteristics of the MPCs, which are extracted by using HRPE
Machine-learning-enabled channel modeling 83

algorithms, e.g., SAGE or CLEAN. However, performing the high-resolution esti-

mation is time-consuming, and usually requires selection of algorithm parameters
such as model order. Hence, some research focuses on the alternative approaches that
require much lower computational effort.
For example, a power angle spectrum-based clustering and tracking algorithm
(PASCT) is described in [39]. The PASCT algorithm first obtains the PAS by using a
Bartlett (Fourier) beamformer [20], as shown in Figure 2.10. In this case, the cluster is
defined as an “energy group,” which has obvious higher power than the background,
in the azimuth-elevation domain.
Generally, to recognize the clusters from the PAS, we need to distinguish between
clusters and background. Clusters close to each other tend to be identified as one big
target (called a target object), which contains one or more clusters. In this case, to
further identify clusters in the big target objects, a density-peak-searching method is
developed to divide the clusters. The details of the clustering process are as follows.
To recognize the target objects in PAS, the maximum-between-class-variance
method [40] is applied to automatically determine a selection threshold of power for
the elements in the PAS. This can separate the target objects from background noise
at first stage. The between-class-variance of the average power levels of background
noise and target objects can be expressed by
δ 2 (αT ) = pB (αT )(eB (αT ) − E(αi ))2 + pO (αT )(eO (αT ) − E(αi ))2 , (2.18)
where αT is the separation threshold between the clusters and background noise,
pB (αT ) and pO (αT ) are the probabilities of the background noise and target objects
occurrence in the current PAS, respectively, eB (αT ) and eO (αT ) are the average power
levels of background noise and target objects, respectively, and E(αi ) is the total mean
power level of all the elements in the PAS.
The difference between background noise and groups of clusters can be max-
imized by maximizing the between-class-variance, and the best selection threshold
αT ∗ can be therefore expressed as
∗
αT = arg{max δ 2 (αT )|α1 ≤ αT < αL }. (2.19)

120 –67
Elevation (degree)

100 –69
80
–71
60
–73
40
20 –75

50 100 150 200 250 300 350 dB

Azimuth (degree)

Figure 2.10 PAS obtained by using Bartlett beamformer in [39]

84 Applications of machine learning in wireless communications

Since the number of the power levels is limited, αT ∗ can be easily found by a sequential
search.
Nevertheless, the signal to interference plus noise ratio of the PAS has a strong
impact on the performance of the target recognition. In an LOS scenario, the clusters
are generally easily discernible, with strong power and low background noise, i.e.,
the targets can be easily recognized and detected. However, in many NLOS scenar-
ios, the power distribution of the PAS is more complicated with high background
noise, and many small clusters caused by reflections and scatterings interfere with
the recognition process. In this case, the targets that contain many small clusters are

difficult to be separated. To avoid this effect, an observation window αW is set so that

only the elements having the power level between [αL − αW , . . . , αL ] are processed
in the target recognition approach. In this case, the best selection threshold αT ∗ is
obtained by
∗
αT = arg{max δ 2 (αT )|αL − αW

≤ αT < αL }. (2.20)
By using the observation window, the recognition process can focus on the ele-
ments with stronger power compared to the noise background. Moreover, a heuristic

sequential search is used to select an appropriate observation window size αW as fol-

lows. Parameter αW is initialized to 0.1αL at the beginning of the searching process
and keeps increasing until the following constraints are no longer satisfied:
● Size of recognized targets: S < Smax
● Power gap of each single target: A < Amax
where S is the size of the recognized targets indicating how many elements the target
consists of and Smax is the upper limit of size. Specifically, to avoid the interference
caused by the small and fragmental targets, the lower limit of the size is also consid-
ered: only a target bigger than Smin is counted, whereas the target smaller than Smin is
considered as noise rather than clusters. Parameter A is the gap between the highest
power and the mean power of each target. In each iteration, S and A are updated
based on the recognized target objects by using the new αT ∗ from (2.20), until the
above constraints are no longer satisfied.
Examples for the clustering results in an LOS and NLOS scenarios are given in
Figure 2.11(a) and (b), respectively. In the experiments in [39], the PASCT algorithm
is able to well recognize the clusters in time-varying channels without using any
high-resolution estimation algorithm.

2.3.6 Improved subtraction for cluster-centroid initialization

As mentioned in Section 2.3.1, the initial values of the positions of cluster-centroids
have a great impact on the clustering results. Hence, a density-based initialization
algorithm is developed in [41] to find an appropriate number and positions of the
initial cluster-centroids. Once those are determined, the KPowerMeans algorithm
[13] can be initiated to cluster the MPCs, and the position of the cluster-centroid is
updated in each iteration. It is noteworthy that the MPC closest to an initial cluster-
centroid is considered the initial power weighted centroid position in KPowerMeans.
Machine-learning-enabled channel modeling 85

120

Elevation (degree)
100 –66

80
60 –70

40
–74
20

50 100 150 200 250 300 350 dB

(a) Azimuth (degree)
–65
120
–67
Elevation (degree)

100
80 –66

60
–71
40
20 –73

50 100 150 200 250 300 350 dB

(b) Azimuth (degree)

Figure 2.11 Cluster recognition results of the (a) LOS scenario and (b) NLOS
scenario, respectively, in [39]

To accurately measure the distance between MPCs, the BMCD (balanced multipath
component distance) [41] is used here. The main difference between the BMCD and
MCD is that the BMCD introduces additional normalization factors for the angular
domains. The normalization factors are calculated as
stdj (dMCD,AoD/AoA (xj , x̄))
δAoD/AoA = 2 · , (2.21)
max2j (dMCD,AoD/AoA (xj , x̄))
where stdj is the standard deviation of the MCD between all MPC positions xj and
the center of data space x̄, and maxj is the corresponding maximum.
The concrete steps of the improved subtraction are expressed as follows:
1. Calculate the normalized parameter β:
N
β = N , (2.22)
j=i dMPC (xj , x̄)
where N is the total number of MPCs and dMPC (xj , x̄) is the BMCD between xj
and x̄.
86 Applications of machine learning in wireless communications

2. Calculate the density value for each MPC xi :

N
Pjm = exp(−mT · β · dMPC (xj , x̄)) (2.23)
j=1

where mT · β scales the actual influence of neighboring MPCs and its inverse is
called neighborhood radius. For measurement data, it is more practical to find
the appropriate radii for DoA, DoD, and delay dimension separately. Hence, both
m and d vectors contain three components:
dMPC (xi , xj ) = [dMPC,DoA ((xi , xj )), dMPC,DoD ((xi , xj )), dMPC,delay ((xi , xj ))]T . (2.24)
3. The points xk with the highest density value are selected as the new cluster-
centroids if their density value is above a certain threshold. Stop the iteration if
all density values are lower than the threshold.
4. Subtract the new centroid from the data by updating the density values:
Pim = Pim − Pkm · exp(−η · mT · β · dMPC (xi , xk )), (2.25)
where η ∈ (0, 1] is a weight parameter for the density subtraction. Return to
step 3.
Then, the number and position of the initial cluster-centroids can be determined,
and the KPowerMeans can be initialized with these values.
Specifically, to find a proper neighborhood radius, the correlation self-
comparison method [41] is used. The detailed steps are
1. Calculate the set of density values for all MPCs P ml for an increasing ml , where
ml ∈ {1, 5, 10, 15, . . .}, and the other components in m are set to be 1.
2. Calculate the correlation between P ml and P ml+1 . If the correlation increases
above a preset threshold, ml here is selected as the value for m in this dimension.

2.3.7 MR-DMS clustering

MR-DMS (multi-reference detection of maximum separation) [42] is developed based
on the hierarchical cluster method, which first clusters all elements into one single
cluster and then further separates the cluster into more smaller clusters. Specifically,
the distances between all MPCs of a cluster seen from multiple reference points are
measured, and the MPC group with the biggest distance is separated into two clusters.
Besides, the BMCD introduced in Section 2.3.6 is used in [42] to measure the distance
between MPCs and reference points. In this study, the optimum number of the clusters
can be obtained by different ways: (i) using cluster validation indices, e.g., Xie–Beni
index [43], to validate different clustering results or (ii) predefine a threshold for the
separation process.
2.3.7.1 Cluster the MPCs
The concrete steps of the MR-DMS are as follows:
1. Spread N reference points over the data space. (e.g., N = 16).
2. Cluster all MPCs as single one cluster C1 .
Machine-learning-enabled channel modeling 87

3. Compare the current cluster number CN and the maximum cluster number CN ,max ,
if CN < CN ,max : for each recent cluster Ck , and calculate the BMCDs between
all MPCs xi in the current cluster and the reference points rn according to

dkn (i) = dMPC (xi , rn ). (2.26)

4. Sort the obtained BMCDs dkn (i) in ascending order.

5. Calculate the derivative (dkn (i)) , which is the real distance between MPCs in the
kth cluster and the nth reference point.
6. Separate the MPCs in the current cluster, which have maximum BMCD over all
MPCs and the reference max (dkn (i)) .
k,n,i
7. Update the number of current clusters CN and return to step 3.

2.3.7.2 Obtaining the optimum cluster number

The optimum cluster number can be determined by using the cluster validation
indices or a predefined threshold during the separation stage. The implementation
of the cluster validation indices is introduced in Section 2.3.1; thus, the threshold to
automatically detect a proper cluster number is explained in the below.
During the separation of MPCs in step 6, only the MPCs whose maximum deriva-
tive (dkn (i)) exceeds a certain threshold for at least one reference point are considered
for separation. In [42], an example of a dynamic threshold is defined by considering
the distribution of (dkn (i)) :

thkn = mean((dkn (i)) ) + α · std(((dkn (i)) )), (2.27)

where α is a weight parameter. Consequently, for each cluster, only the MPCs which
have a BMCD significantly larger than the others in the same cluster are considered
to be separated. The separation is stopped if all MPCs in the clusters are below the
threshold.
Figure 2.12 compares the accuracy of detecting the cluster number by using the
improved subtraction algorithm in [41] and the MR-DMS in [42]. In the validation,
over 500 drops of the WINNER channel model scenario “urban macro cell” (C2)
are tested. In addition, two different scenarios are used where the cluster angular
spread of arrival (ASA) is varied (ASA ={6◦ , 15◦ }). From the results, the MR-DMS
achieves better performance in detecting the correct cluster number than the improved
subtraction algorithm.
Moreover, Figure 2.13 gives azimuth of AoA/AoD and delay domain clustering
results based on a MIMO measurement campaign in Bonn, where Figure 2.13(a) and
(b) is obtained by using the improved subtraction algorithm together with a run of
KPowerMeans and the MR-DMS algorithm, respectively. Details of the measurement
campaign can be found in [44], and the MPCs are extracted by using the RiMAX
algorithm.
100

Probability of detection
60

40
Sub. ASA 6°
Sub. ASA 15°
20 MR-DMS ASA 6°
MR-DMS ASA 15°
0
5 10 15 20
Simulated # of cluster

Figure 2.12 Probability of correctly detecting the number of clusters by using the
improved subtraction algorithm in [41] and the MR-DMS in [42], vs.
cluster angular spread of arrival (ASA), where ASA = {6◦ , 15◦ }

4
τ (μs)

2 –200
1 –100
–200 0
–100
0 100 ϕd [°]
100
ϕa [°] 200 200
(a)

5
4
τ (μs)

3
2 –200
1 –100
–200 0
–100
0 100 ϕd [°]
a
100
(b) ϕ [°] 200 200

Figure 2.13 Clustering results based on a MIMO measurement campaign in Bonn,

where (a) and (b) are clustered by using the improved subtraction
algorithm, together with one run of KPowerMeans and the MR-DMS
algorithm, respectively
Machine-learning-enabled channel modeling 89

2.4 Automatic MPC tracking algorithms

To accurately model time-varying channels, the dynamic changes of the MPCs and
clusters need to be characterized. To achieve this, the dynamic MPCs need to be not
only clustered but also tracked over time to model the channel [45,46]. Unfortunately,
the true moving paths of the MPCs can never be obtained since the MPCs are indi-
vidually extracted from each snapshot if single-snapshot evaluation algorithms such
as SAGE or CLEAN are used. Therefore, many automatic tracking algorithm have
been described for searching the most likely moving paths among MPCs/clusters. In
this section, we present some machine-learning-based tracking algorithms used for
channel modeling.

2.4.1 MCD-based tracking

A tracking method for MPCs needs to capture the moving feature of the clusters/MPCs
considering the trade-off between tracking accuracy and computational complexity.
The MCD-based tracking method [47,48] aims to track the MPCs in time-varying
channels by measuring the MCD between each combination of MPC/cluster in two
consecutive snapshots. Note that the MCD between different clusters is defined as
the MCD between the cluster-centroids. The basic idea of the MCD-based tracking
algorithm is expressed as follows:
1. Preset a threshold PT based on measured data.
2. Measure the distance between MPCs:
DMPCi ,MPCj = MCD(MPCi , MPCj ), MPCi ∈ St , MPCj ∈ St+1 (2.28)
where St is the set of MPCs in the snapshot st , and St+1 is the set of MPCs in the
next snapshot.
3. Associate MPCs in the snapshots st and st+1 based on the DMPCi ,MPCj and PT ,
where
● if DMPCi ,MPCj < PT , the two MPCs are considered as the same MPC, as shown
as the MPCi and MPCj in Figure 2.14(a);
● if DMPCi ,MPCj > PT , the two MPC clusters are considered as different MPCs,
as shown as the MPCi and MPCk in Figure 2.14(a);
● if there are more than two MPCs in St+1 that are close to MPCi in St , MPCi
is considered to be split in the next snapshot, as shown in Figure 2.14(b);
● if there are more than two MPCs in St close to MPCi in St , MPCi and MPCj
are considered to be merged in the next snapshot, as shown in Figure 2.14(c);
One of the advantages of the MCD-based tracking algorithm is its low compu-
tational complexity, which can be used for complicated scenarios containing many
MPCs. Moreover, the behavior of the dynamic MPCs is properly considered in the
MCD-based tracking algorithm, including split, merge, and birth–death of MPCs,
which all correspond to realistic physical behavior. On the other hand, the value of
the preset threshold has a great impact on the tracking results of the algorithm. Hence,
the subjective threshold may cause unreliable performance.
90 Applications of machine learning in wireless communications

MPCj in MPCk in
snapshot t+1 snapshot t+1
Tracking
result Preset
AoA

AoA
threshold
Preset MPCj in
MPCi in
threshold snapshot t MPCi in snapshot t+1
snapshot t
MPCk in
snapshot t+1
AoD AoD
(b)
(a)

MPCj in
snapshot t
MPCk in
AoA

snapshot t+1
Preset
threshold
MPCi in
snapshot t

AoD
(c)

Figure 2.14 MCD-based tracking algorithm (a) gives the principle of tracking
process, whereas (b) and (c) shows the cases of split and mergence,
respectively

2.4.2 Two-way matching tracking

The two-way matching tracking algorithm is proposed in [45,46] for time-varying
channels. It requires the estimated MPCs, which can be extracted by using an HRPE
algorithm, and uses the MCD to measure the difference between MPSs. In addition,
this tracking algorithm introduces a two-way matching process between two consec-
utive snapshots to improve the tracking accuracy. The main steps of the two way
matching can be expressed as follows:
1. Obtain the MCD matrix D by calculating the MCD between each MPC during
two consecutive snapshots. For two snapshots s and s + 1, the MCD matrix D
can be expressed as
⎡ ⎤
D1,1 . . . D1,N (s+1)
⎢ . ⎥
D=⎢ ⎣ ..
..
.
..
.
⎥
⎦ (2.29)
DN (s),1 . . . DN (s),N (s+1)
where N (s) and N (s + 1) is the number of the MPCs in snapshots s and s + 1,
respectively.
Machine-learning-enabled channel modeling 91

2. MPCs x in snapshot s and y in snapshot s + 1 are considered as the same MPC

if the following condition is satisfied:

Dx,y ≤ ε (2.30)

x = arg min(Dx∈s,y ) (2.31)

y = arg min(Dx,y∈s+1 ) (2.32)

where ε is a preset threshold to determine whether the two MPCs in consecutive

snapshots could be the same MPC.
3. Matching all MPCs between snapshots s and s + 1, the matched MPC pairs are
considered as the same MPC; the remaining MPCs in snapshots s and s + 1 are
considered as dead and new born MPC, respectively.
4. Repeat the preceding steps 1, 2, and 3 for the current and next snapshots.

One of the advantages of the two-way matching is low computation complexity,

which makes it easy to implement for massive data, e.g., V2V channel measure-
ment data. The described algorithm is implemented in [45] for V2V measurements,
where it is found that only the MPCs with similar delay and angular characteristic are
considered as the same MPCs.

2.4.3 Kalman filter-based tracking

Kalman filtering is one of the most popular machine-learning methods used for target
tracking. Therefore, [15] described a cluster-tracking algorithm based on Kalman
filters. It is noteworthy that the Kalman filter-based tracking algorithm is used for
tracking cluster-centroids, instead of MPCs. The framework of the Kalman filtering
is given in Figure 2.15, where μ(n)c is the cluster-centroid position in the angle domain
or angle-delay domains, xc(n) are the tracked objects in the input data: angle-delay
vector (X(n) ), power (P(n) ), and n is the index of the snapshot.
For each iteration, the position of the cluster-centroid in the next snapshot is
predicted by the Kalman filter based on the current position, and the predicted
cluster-centroids are used for the clustering method, e.g., KPowerMeans, for the
next snapshot.

Kalman Next X(n),P(n) Initial Clustering μc (n)

Kalman
prediction xc(n|n–1) snapshot xc(n|n–1) guess algorithm xc(n|n–1) update

xc(n|n)

Figure 2.15 Framework of the Kalman filter, where xc(n) are tracked objects in the
input data (X(n) , P(n) )
92 Applications of machine learning in wireless communications

145

Delay (ns) 140

135

130
0

0.5
–1.5 –1
AoA (rad) –2
1 –3 –2.5
–3.5
AoD (rad)

Figure 2.16 Illustration of the tracking result in [15]

Figure 2.16 gives the tracking result in [15]. For a single target, the Kalman
filter-based tracking algorithm can achieve high tracking accuracy. To track multiple
targets, the Kalman filter can be replaced by a particle filter [49].

2.4.4 Extended Kalman filter-based parameters estimation

and tracking
In this section, we discuss an extended Kalman filter-based parameters estimation
and tracking algorithm [14,50,51] to capture the dynamics of the channel parameters
in time with a low computational complexity. A state-space model is proposed in [51],
which is based on the assumption that the parameters are evolved slowly over time
and correlated in consecutive time instances.
In the state-space model, the state vector consists of normalized delay μτk,p , the
normalized AoA μϕk,p , and the path weight including the real part γk,p
Re
and imaginary
Im
part γk,p . Consequently, the state model of the pth path at time k can expressed as
T
θk, p = μτk,p , μϕk,p , γk,p
Re Im
, γk,p . (2.33)

It is noteworthy that this model can be extended to contain additional parameters.

Let the θk denote the state model of all MPCs at time k, then the state model can be
rewritten as
θk = θk + νk (2.34)
yk = s(θk ) + ny,k (2.35)
where yk is the observation vector, ny,k is the complex vector containing dense MPCs
and noise, s(θk ) is the mapping function between θk and yk , is the state transition
Machine-learning-enabled channel modeling 93

matrix. Specifically, all the parameters in the state model are assumed to be uncorre-
lated with each other. In other words, each parameter evolves independently in time
and the state noise, which is additive real white Gaussian while the observation noise
is circular complex white Gaussian, is also assumed to be uncorrelated with each other
and the state. For each path, the covariance matrix of the state noise is represented by
Qθ, p = diag{σμ2(τ ) , σμ2(ϕ) , σγ2Re , σγ2Im }, whereas the covariance matrix of the observation
noise is denoted by Ry .
Considering the estimated parameters are real, the EKF equations can be
expressed as

θ̂(k|k−1) = θ̂(k−1|k−1) (2.36)

P(k|k−1) = P(k−1|k−1) + QθT
(2.37)
−1
P(k|k) = J(θ̂ , Rd ) + P−1
(k|k−1) (2.38)
T
R{R −1 D(k) }
y
K(k) = P(k|k−1) I − J(θ̂ , Rd )P(k|k) · (2.39)
I {Ry−1 D(k) }
T
R{yk − s(θ̂(k|k−1) )}
θ̂(k|k) = θ̂(k|k−1) + K(k) (2.40)
I {yk − s(θ̂(k|k−1) }

where R(•) and I (•) denotes the real and imaginary parts of •, respectively, P(k|k)
is the estimated error covariance matrix, J(θ̂, Rd ) = R{DH(k) Ry−1 D(k) }, and D is the
Jacobian matrix. For P paths containing L parameters, D can be expressed as

∂ ∂ ∂
D(θ) = T s(θ ) = s(θ ) · · · s(θ ) . (2.41)
∂θ ∂θ1T ∂θLPT

Apparently, the initialization value of parameters for the state transition matrix
and the covariance matrix of the state noise Qθ are crucial to the performance of the
following tracking/estimation for the EKF. Therefore, it is suggested in [51] to employ
another HRPE algorithm, e.g., SAGE [1], RiMAX [3], for this purpose.

2.4.5 Probability-based tracking

In this section, we discuss a probability-based tracking algorithm [16] to track the
dynamic MPCs in time-varying channels. The algorithm aims to (i) identify the mov-
ing paths of the MPCs in consecutive snapshots and (ii) cluster these MPCs based
on the relationships of moving paths. To track the MPCs, a novel-probability-based
tracking process is used, which is conducted by maximizing the total sum probability
of all moving paths.
In this algorithm, the number of MPCs is assumed to be time-invariant to reduce
the complexity. For each MPC, four parameters are considered: AoD φ D , AoA φA ,
delay τ and power α. Let A1 , . . . , Am and B1 , . . . , Bm represent the MPCs in the snap-
shots Si and Si+1 , respectively. l represents an ordered pair of the MPCs in consecutive
94 Applications of machine learning in wireless communications

snapshots, i.e., lAx ,By is the moving path from Ax to By , between Si and Si+1 , as shown
in Figure 2.17(a). In the probability-based tracking algorithm, each moving path lA,B
is weighed by a moving probability P(Ax , By ), as shown in Figure 2.17(b).
In the probability-based tracking algorithm, the moving paths are identified
by maximizing the total probabilities of all selected moving paths, which can be
expressed as

L∗ = arg max P(Ax , By ) (2.42)
L⊂L
(Ax ,By )∈L

where L is the selected set of the moving paths and L is the set of all moving
paths. Then, the moving probability P(Ax , By ) is obtained by using the normalized

MPC A1 l1 MPC B1
in Si in Si+1
l2
Delay
(ns)
l3 MPC B2
l4 in Si+1

MPC A2
in Si

O Azimuth (degree)
(a)

Snapshot Si Snapshot Si+1

P(A1, B1)
A1 B1
)
, B1
P(A 2

P(A ,
1 B
2)
A2 B2
P(A2, B2)

(b)

Figure 2.17 Illustration of the moving paths between two consecutive snapshots
in [16], where (a) is delay and azimuth domain and (b) is bipartite
graph domain
Machine-learning-enabled channel modeling 95

Euclidean distance DAx ,By of the vector of parameters [φ D , φA , τ , α], which can be
expressed as
⎧
⎪
⎪ 1 DAx ,By = 0,
⎪
⎪
⎨0 DAx ,Bz = 0, y = z,
P(Ax , By ) = (2.43)
⎪
⎪ 1
⎪
⎪ M others.
⎩
DAx ,By z=1 DA−1x ,Bz

To identify the sets of true moving paths L∗ , the Kuhn–Munkres algorithm is

executed, which is usually used to find the maximum weighted perfect-matching
in a bipartite graph of a general assignment problem. In the bipartite graph, every
node in two subsets links to each other and every link has its own weight. In this
algorithm, the MPCs in two successive snapshots are considered as the two subsets in
the bipartite graph, and the moving paths between each snapshot are considered as the
links between two subsets. which is weighted by P(Ax , By ), as shown in Figure 2.17(b).
In this case, the true moving paths can be recognized.
After obtaining the moving paths of all MPCs, a heuristic approach is developed
to cluster these MPCs with the purpose of comparing the moving probability of the
MPCs in the same snapshot with a preset threshold PT . The basic idea of the clustering
process is to group the MPCs with similar moving probabilities, which means their
moving patterns are close to each other, e.g., if P(Ax , By ) and P(Ax , Bz ) are greater
than PT , it indicates that the MPCs By and Bz are fairly similar and estimated to belong
to the same cluster. The clustering process can be expressed as

Kx = {By |P(Ax , By ) > PT , A ∈ Si , B ∈ Si+1 }. (2.44)

According to the simulations in [16], PT is suggested to be set to 0.8. From

(2.44), different Ax in Si may lead to different clustering results. In this case, the result
with the most occurrences is selected, e.g., if K1 = {B1 , B2 }, K2 = {B1 , B2 , B3 }, K3 =
{B1 , B2 , B3 }, then K = {B1 , B2 , B3 }.

2.5 Deep learning-based channel modeling approach

In this section, we present some learning-based channel-modeling algorithms. An
important advantage of many machine-learning methods, especially artificial neural
networks, is to automatically obtain the inherent features of input data and to find
the mapping relationship between input and output. On the other hand, the purpose
of wireless channel modeling is to accurately model MPCs in wireless channels,
and on some level, it aims to find the mapping relationship between the channel
parameters and the scenarios. Inspired by this, a number of papers have investigated
channel-modeling using neural networks.
96 Applications of machine learning in wireless communications

2.5.1 BP-based neural network for amplitude modeling

As early as 1997, [18] adaptively modeled nonlinear radio channels by using the odd
and even BP algorithm for multilayer perceptron (MLP) neural networks. The BP
algorithm is a classical algorithm for data training in neural networks; its flowchart
is given in Figure 2.18.
The SV model [7] is adopted in [18] as a comparison, the simulation result is
given in Figure 2.19.
Considering the advantages of neural network on regression problems, a multi-
layer neural network is used to find the mapping relationship between frequency and
amplitude, where the architecture of the neural network is shown in Figure 2.20. For
most of the neural networks, there are three parts in the framework: (i) input layer,
(ii) hidden layer, and (iii) output layer. The measured data are used as training data,
where the system input and output of the neural network are frequency and amplitude,
respectively, and the sub-output of each layer is the sub-input of the next layer in the
hidden layers. Through several iterations, the weight parameters of each layer can
be obtained; thus, the mapping function between the system input and output can be
modeled.

2.5.2 Development of neural-network-based channel modeling

Due to the good performance of finding mapping relationships, neural-network-based
channel modeling has drawn a lot of attention. For a classical architecture of the neural
network, the performance is related to the number of layers, and each layer is designed
to classify basic elements. Theoretically, a neural network with more layers can thus
achieve better performance. However, a neural network with too many layers leads
to another problem: vanishing gradient. In the training process of a neural network,
the difference between the system output and training data is assessed and fed back
to the upper layer to adjust the weight parameters.
In a multilayer network, the feedback obviously has more influence on the layers
closest to the system output and has a fairly limited influence on the layers at the
front end. Nevertheless, the layers at the front end usually have a great effect on
the final performance. As a result, the feedback in each iteration cannot be well
transmitted to the front layers, as shown in Figure 2.21. Hence, the performance
of the multilayer neural network can suffer. Besides, a multilayer neural network
has high computational complexity, which was prohibitive given the limitations of
the hardware at that time of that paper. Therefore, neural-network-based channel
modeling gradually disappeared from public view.
To avoid the vanishing gradient, [52] proved that a three-layer MLP neural net-
work can be multidimensionally approximated as an arbitrary function to any desired
accuracy, which is important and useful for modeling a mapping function. Since there
are only three layers in this network, the vanishing gradient does not impact the per-
formance during the training process. In this case, a three-layer MLP neural network
can be adopted for many channel-modeling applications without the limitation of
vanishing gradients.
Machine-learning-enabled channel modeling 97

Initialized

Give a input vector and

a target output

Calculate every unit output

of the hidden layer and the
output layer

Calculate warp value which

subtracted the target value from
the actual output

Whether warp value No

is satisfied with the
desire

Whether all warp No

values are satisfied
with the desire
Calculate the unit
Yes error of the hidden
layer

End

Calculate the error

grades

Weight value
learning

Figure 2.18 Flowchart of BP algorithm

98 Applications of machine learning in wireless communications

1.5
Measured data

1 Hetrakul–Taylor model

MSE = 1.60×10–3
0.5
Saleh model

MSE = 5.20×10–4
0
NN model

MSE = 2.63×10–4
–0.5

–1

–1.5
–6 –4 –2 0 2 4 6
Input amplitude

Figure 2.19 Simulation result of the comparison of SV model and ANN-based

model in [18]

Input layer Hidden layer Output layer

Figure 2.20 Illustrations of the common architecture of the neural network

Machine-learning-enabled channel modeling 99

Feed back

Hidden layer

Figure 2.21 Demonstration of the vanishing gradient in training process of the

neural network

In addition, there are some other novel frameworks for neural networks which
can avoid the vanishing gradient, e.g., the restricted Boltzmann machines framework
and the deep-learning method described in [53]. Hence, there are some propagation
channel modeling studies, e.g., [54], where the amplitude frequency response of
11 paths is modeled by an MLP neural network.

2.5.3 RBF-based neural network for wireless channel modeling

Reference [19] introduces a RBF neural network for modeling a single nonfading
path with additive white Gaussian noise (AWGN). The RBF neural network can
approximate any arbitrary nonlinear function with any accuracy, just like an MLP
network. However, it has a number of advantages: it only has one hidden layer, and
the number of hidden layer nodes can be adaptively adjusted in the training stage,
whereas the numbers of the hidden layer and the hidden layer nodes for MLP network
is not easily determined. The main framework of the RBF-based neural network is
expressed as follows.
1. RBF neural network: A RBF neural network is a three-layer feedforward net-
work, which contains one input layer, one hidden layer, and one output layer.
The input layer obtains the training data and transmits to the hidden layer. The
hidden layer consists of a group of RBF, and the corresponding center vectors
and width are the parameters of the RBF, where the Gaussian function is usually
100 Applications of machine learning in wireless communications

adopted as the basis function. At the end of the network, the out layer receives
the outputs of the hidden layer, which are combined with linear weighting. The
mapping function between the input and output layer can be expressed as

m
m x − c
i
y = f (x) = ωi φ(x − ci , σi ) = ωi exp − (2.45)
i=1 i=1
2σi2

where the vector x = (x1 , x2 , . . . , xm ) represents the input data of the network, ci
and σi are the mean and standard deviation of a Gaussian function, respectively,
m is the number of hidden layer neurons, ωi is the weight between the link of the
ith basis function and the output node, and • is the Euclidean norm.
The training process adjusts, through iterations, the parameters of the network
including the center and width of each neuron in the hidden layer, and the weight
vectors between the hidden and output layer.
2. Channel modeling: To model the radio channel by using a neural network,
the mapping relationship/function between the input, i.e., transmit power and
distance, and the output, receive power and delay, is usually a nonlinear function.
Hence, the goal of the neural-network-based channel modeling is to use the
network to approximate the transmission system, as shown in Figure 2.22.
In [19], the number of RBFs is set to the number of MPCs, to simulate the
transmit signal with different time delays. The output layer gives the received signal.
Besides, the width of the RBF network is obtained by
d
σ =√ (2.46)
2M
where d is the maximum distance and M is the number of RBF nodes. In this case,
once the nodes and width of RBF network are determined, the weights of the output
layer can be obtained by solving linear equations.

b1
W1i
W21

W22 b2
τ1
Y(KTs)
∑
τ2 W23

W24
τN

Input layer Hidden layer Output layer

Figure 2.22 The wireless channel based on RBF neural network

Machine-learning-enabled channel modeling 101

Furthermore, the BP-based network model is compared with the RBF-based

network model in [19]. A multipath channel with AWGN is simulated, where the
results are given in Figure 2.23. From the simulation results, the RBF-based network
model generally shows better accuracy than the BP-based network model.
Similarly, a neural network is also used in [55] to model the path loss in a mine
environment, where the frequency and the distance are considered as the input data,
and the received power is considered as output. The framework of the neural network
is given in Figure 2.24, which contains two input nodes, 80 hidden nodes, and one
output node. W1f and W2f are the weight parameters between the input layer and
hidden layer, whereas wjk and wNk are the weight parameters between the hidden and
output layers.

2.5.4 Algorithm improvement based on physical interpretation

The high accuracy and flexibility make neural networks a powerful learning tool for
channel modeling. Due to good learning ability and adaptability, neural networks are
expected not only to model the channel of one single scenario but also to model
the channels of multiple scenarios, as shown in Figure 2.25. Despite these high
expectations, there are still many problems that need to be discussed and studied.
Based on past research, artificial intelligence has shown great power for the
development of channel modeling, whether as a preprocessing tool or a learning tool.
However, the difference between the two fields of artificial intelligence and chan-
nel modeling still needs to be considered. Although machine-learning approaches

2
Desired output
1.5 BP neural network
RBF neural network
1

0.5
Output value

–0.5

–1

–1.5

–2
0 2 4 6 8 10
Sample numbers

Figure 2.23 Simulation results of AWGN channel containing two pathways with
doppler frequency shifts in [19]
102 Applications of machine learning in wireless communications

W1j
f Wjk

Output

W2j WNk
.
.
.

Input layer Hidden layer Output layer

Figure 2.24 Neural network used in [55], which contains two input nodes, 80
hidden nodes, and one output node. The two input nodes correspond
to frequency and distance, whereas the output node corresponds to the
power of the received signal

Neural network
Measured data (AoA, AoD and
delay, etc.) in Scenario n

Measured data (AoA, AoD and delay, etc.) in

Scenario 1
Input Output Comprehensive
Training
Clustering based on channel model
machine-learning method
Sub-neural
network n
Training
Iteration

Model 1 Sub-neural
network 1
Data
Iteration

Figure 2.25 Framework of the learning-based channel modeling for multiple

scenarios

generally have good performance at data processing, these approaches can be further
improved by considering the physical characteristics of channel parameters. For exam-
ple, KMeans is a good conventional clustering algorithm for data processing, and the
KPowerMeans in [13] is further developed by combining the physical interpretation
Machine-learning-enabled channel modeling 103

of the power of MPCs with KMeans, thus improving performance for clustering of
MPCs than merely using the KMeans. Moreover, the development of MCD is another
example, where the physical characteristics of the MPCs are considered, and the MCD
thus is a more accurate measure of the differences of MPCs than using the Euclidean
distance for channel modeling. As for neural networks, the physical interpretation is
also important to build an appropriate network, e.g., the description of the CIR needs
to be considered while constructing the activation function for the neural network.
In addition, the disadvantages of the adopted machine-learning methods cannot be
neglected, e.g., the KMeans is sensitive to initial parameters; this feature also appears
in the KPowerMeans, e.g., the clustering result is sensitive to the assumed number of
clusters and the position of cluster-centroids. Using the physical meaning of param-
eters of the approaches is a possible way to evade these disadvantages. Hence, the
potential relationship between parameters of machine-learning techniques and phys-
ical variables of the radio channels needs to be further incorporated into the adopted
algorithms to improve accuracy.

2.6 Conclusion
In this chapter, we presented some machine-learning-based channel modeling algo-
rithms, including (i) propagation scenarios classification, (ii) machine-learning-based
MPC clustering, (iii) automatic MPC tracking, and (iv) neural network-based channel
modeling. The algorithms can be implemented to preprocess the measurement data,
extract the characteristics of the MPCs, or model the channels by directly seeking the
mapping relationships between the environments and received signals. The results in
this chapter can provide references to other real-world measurement-based channel
modeling.

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Chapter 3
Channel prediction based on machine-learning
algorithms
Xue Jiang1 and Zhimeng Zhong 2

In this chapter, the authors address the wireless channel prediction using state-of-
the-art machine-learning techniques, which is important for wireless communication
network planning and operation. Instead of the classic model-based methods, the
authors provide a survey of recent advances in learning-based channel prediction
algorithms. Some open problems in this field are then proposed.

3.1 Introduction

Modern wireless communication networks can be considered as large, evolving dis-

tributed databases full of context and information available from mobile devices, base
stations, and environment. The wireless channel data in various scenarios including
large-scale and small-scale parameters are one of the important and useful data that
could be used for analyzing and making predictions.
A coverage map is often given as a set of radio measurements over discrete
geographical coordinates and is typically obtained by drive tests. Accurate coverage
maps are crucial for enabling efficient and proactive resource allocation. However, it
is nearly impossible to obtain these maps completely from measurements. Thus, the
coverage loss maps should be reconstructed with the available measurements. A reli-
able reconstruction of current and future coverage maps will enable future networks
to better utilize the scarce wireless resources and to improve the quality-of-service
experienced by the users. Reconstructing coverage maps is of particular importance
in the context of (network-assisted) device-to-device (D2D) communication where
no or only partial measurements are available for D2D channels [1].
As one of the hottest topics all over the world, machine-learning techniques have
been applied in various research fields in recent years including reconstruction of cov-
erage maps. These learning-based reconstruction approaches can be divided into two
categories: batch algorithms and online algorithms. The former one mainly includes

1
School of Electronic Information and Electrical Engineering, Shanghai Jiao Tong University, China
2
Huawei Technologies Ltd., China
110 Applications of machine learning in wireless communications

support vector machines (SVM) [2,3], artificial neural networks (ANN) [2,4], and
matrix completion with singular value thresholding (SVT) [5]. Aside from that,
Gaussian processes [6] and kriging-based techniques [7] have recently been suc-
cessfully used for the estimation of radio maps. In [8], kriging-based techniques
have been applied to track channel gain maps in a given geographical area. The pro-
posed kriged Kalman filtering algorithm allows to capture both spatial and temporal
correlations. These studies use batch schemes as well. In [9], an adaptive online
reconstruction methodology is proposed with adaptive projected subgradient method
(APSM) [10], which is the unique online coverage map reconstruction algorithm
having been employed so far.
This chapter mainly generalizes and reviews the learning-based coverage maps
reconstruction mentioned above. The rest of this survey is organized as follows. Sec-
tion 3.2 introduces methodologies of obtaining measurements. Section 3.3 discusses
the respective traits of batch algorithms and online algorithms. The corresponding
approaches will be studied as well. Section 3.4 comprises of the techniques applied
to label the measurements to get more accurate results. The final section draws the
conclusion of the survey.

3.2 Channel measurements

Before discussing the learning-based algorithms, a set of radio measurements over
discrete geographical coordinates should be obtained. In general, the methodology of
achieving measurements can be divided into two types:

● Conventional drive test: Conventional drive test is a manual process. To collect

network quality information, an operator often needs to send engineers directly
to the concerning area and obtain radio measurements in a hand-operated man-
ner. Typically, a measurement vehicle equipped with specially developed test
terminals, measurement devices, and a global-positioning system receiver to
obtain geographical location is used to check coverage outdoors [11]. With such
measurement vehicle, engineers would perform test calls in the car and record
measurement results along the drive route.
● Minimization of drive test (MDT): The main concept of MDT is to exploit
commercial user equipment (UE) measurement capabilities and geographically
spread nature for collecting radio measurements [12]. This methodology exploits
information from the so-called crowdsourcing applications. With crowdsourc-
ing, a user installs an application on an off-the-shelf smart phone and returns
measurements to a database [13].

Conventional drive test is simple and stable. Nevertheless, this methodology consumes
significant time and human efforts to obtain reliable data, and the cost will ascend
vastly with the studied area getting larger. Thus, it would be more suitable to small-
scale area. MDT is a relatively cost-efficient way to get the measurements, sacrificing
for its stability. The application on users’ smart phones would reduce their data budget
Channel prediction based on machine-learning algorithms 111

and the battery lifetime of the mobile device. Wide fluctuation of the involved smart
phones’ function might also result in the systematic measurement errors.

3.3 Learning-based reconstruction algorithms

Over the past years, path-loss models in various types of networks have been proposed
and analyzed. Despite the fact that path-loss modeling is useful in many applications,
the deviation between the path loss, measured in a real propagation environment,
and the one given by the model can be large [14]. For this reason, learning-based
reconstruction has been extensively studied since it can be tailored to the specific
environment under consideration and give more accurate results. As mentioned in
the first section, learning-based reconstruction algorithms can be divided into two
categories: batch and online algorithms (Table 3.1).

3.3.1 Batch algorithms

Batch algorithms assume the complete data to be available before performing the
reconstruction algorithm. The performance of batch algorithm is excellent, and the
cost is much cheaper than online algorithms, while storage is needed to store the sam-
ples. The batch algorithms, which have been employed in coverage loss reconstruction,
includes SVM, ANN, and SVT.

3.3.1.1 Support vector machine

SVM was first introduced by Vapnik [16]. One of the main advantages of SVM over
other classical machine-learning techniques (e.g., neural networks) is the absence
of local minima in the optimization problems, the possibility of enforcing sparse
solutions, and the capacity for controlling error margins in the prediction. SVMs
were initially developed for classification tasks, but they have proven to be a powerful
tool for regression problems (i.e., for function approximation problems), so they are
natural candidates for the task of coverage map estimation [2]. In particular, in [2],
an extended feature vector is used to train the SVM, and this feature vector includes
environmental information about transmitters, receivers, buildings, and the transmit
frequency, among others. To counter the curse of dimensionality owing to the high
number of input features, the authors of [3] use a principal component analysis (PCA)

Table 3.1 Representative algorithms for channel

map reconstruction

Type Methods

Batch SVM [2,3], ANN [2,4], matrix completion [5]

Online APSM [10], multi-kernel [15]
112 Applications of machine learning in wireless communications

based on dimensionality-reduction techniques. This operation is performed before

applying the SVMs.
The estimate usually assumes the following form by SVM:

m
ỹ = ωj φj (x) (3.1)
j=1

where {φj (x)}mj=1 is a set of m nonlinear basis functions. The loss function [17] used
for determining the estimate is given by

| ỹ − y| − ε, | ỹ − y| > ε
Lε ( ỹ, y) = (3.2)
0, otherwise
with ε being a small value. The problem can be formally stated as

1
N
min Lε ( ỹi , yi )
N i=1 (3.3)
s.t. ω ≤ α
where ω ∈ Rm and α ∈ R+ is an arbitrarily chosen constant parameter. It is possible,
by introducing some slack variables, to reformulate problem (3.3) as follows:

1 N
min ω2 + C {ξ i + ξ̄ i }
ξ i ,ξ̄ i ,ω 2 i=1

s.t. yi − ωT φ(xi ) ≤ ε + ξ i i = 1, . . . , N (3.4)

ω φ(x ) − y ≤ ε + ξ̄
T i i i
i = 1, . . . , N
ξ i , ξ̄ i ≥ 0 i = 1, . . . , N .
Then the dual problem of (3.4) can be considered as

N
N
max Q(α i , ᾱ i ) = yi (α i − ᾱ i ) − ε (α i + ᾱ i )
α i ,ᾱ i
i=1 i=1

1
N
N
− (α i − ᾱ i )(α j − ᾱ j )K(xi , x j )
2 i=1 j=1
(3.5)

N
s.t. y i (α i − ᾱ i ) = 0
i=1

0 ≤ αi ≤ C i = 1, . . . , N
0 ≤ ᾱ i ≤ C i = 1, . . . , N
where ε and C are arbitrarily chosen constants, and K(xi , x j ) is the inner-product
kernel:
K(xi , x j ) = φ(xi )T φ(x j ) (3.6)
Channel prediction based on machine-learning algorithms 113

defined in accordance with the Mercer’s condition [16]. Once problem (3.4) is solved,
α i , ᾱ i can be used to determine the approximating function:

N
f (x, ω) = (α i − ᾱ i )K(x, xi ). (3.7)
i=1

Data points for which α i − ᾱ i = 0 are defined as support vectors. Parameters ε and C
control in some way the machine complexity for whom control in nonlinear regression
is a very tough task, which impacts directly on the performance of SVM.

3.3.1.2 Neural networks

Machine-learning algorithms have been proven to be effective tools for solving regres-
sion problems, and they have been effectively applied to the task of wave propagation
prediction [2,4]. Channel prediction can be viewed as a regression problem related
to wave propagation, where the input consists of information about transmitters,
receivers, buildings, frequencies, among others, and the corresponding radio mea-
surements represent the output to be calculated, so we can pose the radio estimation
problem as that of finding a suitable input vector x to be used as the argument of a
function f that best approximates (in some sense) the radio measurements. Stated in
these general terms, we are in the typical setting of a machine-learning problem that
can be naturally addressed with ANNs.
In more detail, ANNs are methods motivated by the way biological nervous
systems, such as the human brain, process information. Their power lies in the fact
that they learn representations of the input data that are suitable for the prediction of
the output produced by possibly unseen inputs. In general, ANNs consist of several
elementary-processing units called neurons, which are located in different layers and
interconnected by a set of weighted edges. Neurons map their input information into
an output information by means of nonlinear functions, from which a variety exists,
each of them having its own estimation properties, but a principled means of choosing
the functions remains an open research problem.
Very early attempts to path-loss prediction via ANNs have been made in [18]
and [19]. These studies show that ANNs can give good estimates of path loss in rural
environments by using discretized information about land cover and topography, the
frequency of the radio waves, and the antenna height. These studies are built upon the
well-known empirical Okumura–Hata’s model for rural areas. In [18,19], the approx-
imation of Okumura–Hata’s model is carried out using a three-layer neural network
with four input units corresponding to these four parameters. This approach shows
a good predictive power, thus demonstrating the feasibility of neural networks for
the task of path-loss prediction. In [20], the authors consider a semiempirical model
for path-loss prediction. They use a semiempirical model field strength prediction
combined with theoretical results from propagation loss algorithms and neural net-
works. They obtain good results for the case of dense urban areas and show that
neural networks are efficient empirical methods, able to produce good models that
integrate theoretical and experimental data. A similar approach is taken in [21],
114 Applications of machine learning in wireless communications

where neural networks are used to correct the biases generated by unknown envi-
ronmental properties and algorithmic simplifications of path-loss estimations that
are common in ray-tracing techniques. The considered neural networks show to sub-
stantially improve the results obtained by classic ray-tracing tools. In [22], radial basis
function (RBF) neural networks are used instead of the classic multilayer perceptron
(MLP). One of the major advantages of RBF neural networks is that they tend to learn
much faster than MLP neural networks, because their learning process can be split
into two stages for which relatively efficient algorithms exist. More specifically, a
two-stage learning approach is taken, where the first stage is composed of an unsu-
pervised clustering step via the rival penalized competitive learning approach. Then
the centers of the radial basis function are adjusted, and, once fixed, the weights are
then learned in a supervised fashion by using the celebrated recursive least squares
algorithm.
In [23], a one-layer backpropagation ANN is proposed to gauge the perfor-
mance of kriging-based coverage map estimation. A new distance measure that takes
obstacles between two points into consideration is introduced, and it is defined as

di,j = (xi − xj )2 + ( yi − yj )2 + 10E (3.8)

where E = (10c)(−1) r∈Wi,j Lr with Wi,j representing the set of obstacles between
point i and j, Lr being the path loss of the respective obstacles, and c being the free
space parameter. The first term, involving the square root, is simply the Euclidean
distance between points i and j. The term 10E expresses the path loss caused by
obstacles. For an example, if one assumes that the path-loss factor of a wall between
two points is 5 dB and the free space parameter c is 2 dB for the environment in
which the wall resides, then the path loss between these two points due to the wall
will equal to the free space path loss of 105/(10×2) . This increase of the path loss
can be equivalently represented by an increase of the effective distance between the
two points. This new measure for the distance improves the achievable estimation
accuracy for prediction tools based on both kriging and ANNs.
A common problem that arises in learning tasks is that in general we have no or
little prior knowledge of the relevance of the input data, and hence many candidate
features are generally included in order to equip algorithms with enough degrees of
freedom to represent the domain. Unfortunately, many of these features are irrelevant
or redundant, and their presence does not improve the discrimination ability. Further-
more, many inputs and a limited number of training examples generally lead to the
so-called curse of dimensionality, where the data is very sparse and provides a poor
representation of the mapping. (Deep neural networks do not perform well with lim-
ited training data.) As a remedy to this problem, dimensionality-reduction techniques
are applied to the data in practice, which transform the input into a reduced represen-
tation of features. Dimensionality-reduction techniques are usually divided into two
classes, linear methods (e.g., independent component analysis) and nonlinear meth-
ods (e.g., nonlinear PCA). In [2], a two-step approach using learning machines and
dimensionality-reduction techniques is proposed. SVMs and ANNs are used as the
learning tools, and they are combined with two dimensionality-reduction techniques,
Channel prediction based on machine-learning algorithms 115

namely, linear and nonlinear PCA. In more detail, in [2], the macrocellular path-loss
model is defined as follows:

L(dB) = L0 + αbuildings = 32.4 + 20 log(d) + 20 log( f ) + αbuildings (3.9)

where L0 is the free space path loss in dB, d is the radio path, f is the radio frequency,
and αbuildings is an attenuation term that depends on several parameters, such as height
of base stations and receivers, the distance between consecutive buildings, the height
of buildings. In [2], the function in (3.9) is learned by using a three-layerANN, with the
three parameters as input. The estimation using dimensionality-reduction techniques
has shown to improve substantially the prediction power over methods that use the
full dimensionality of the input. In addition, PCA-based prediction models provide
better prediction performance than nonlinear PCA-based models, and ANNs-based
models tend to perform slightly better than SVM-based predictors (in the scenarios
considered in the above mentioned studies).
The applications of neural network discussed in this topic are considered as
function approximation problems consisting of a nonlinear mapping from a set of
input variables containing information about potential receiver onto a single output
variable representing the predicted path loss. MLPs is applied to reconstruct the path
loss in [24]. Figure 3.1 shows the configuration for an MLP with one hidden layer
and output layer. The output of the neural network is described as

M
N
y = F0 woj Fh wji xi (3.10)
j=0 i=0

where woj represents the synaptic weights from neuron j in the hidden layer to the
single output neuron, xi represents the ith element of the input vector, Fh and F0 are
the activation function of the neurons from the hidden and output layers, respectively,
and wji are the connection weights between the neurons of the hidden layer and the
inputs. The learning phase of the network proceeds by adaptively adjusting the free

Wji
X0
Woj

X1
Y
X2

Xn–1

Input layer Hidden layer Output layer

Figure 3.1 The configuration of the multilayer perceptron

116 Applications of machine learning in wireless communications

parameters of the system based on the mean squares error described by (3.10), between
predicted and measured path loss for a set of appropriately selected training examples:
1
m
E= ( yi − di )2 (3.11)
2 i=1
where yi is the output value calculated by the network and di represents the expected
output.
When the error between network output and the desired output is minimized, the
learning process is terminated. Thus, the selection of the training data is critical to
achieve good generalization properties [25,26]. In coverage map reconstruction, the
neural networks are trained with the Levenberg–Marquardt algorithm, which provides
faster convergence rate than the backpropagation algorithm with adaptive learning
rates and momentum. The Levenberg–Marquardt rule for updating parameters is
given by
−1 T
W = J T J + μI J e (3.12)
where e is an error vector, μ is a scalar parameter, W is a matrix of networks weights,
and J is the Jacobian matrix of the partial derivations of the error components with
respect to the weights.
An important problem that occurs during the neural network training is the over
adaptation. That is, the network memorizes the train examples, and it does not learn
to generalize the new situation. In order to avoid over adaptation and to achieve good
generalization performances, the training set is separated in the actual training subset
and the validation subset, typical 10%–20% of the full training set [26].
3.3.1.3 Matrix completion
In radio map reconstruction, if the sampling rate of the area of interest is high enough,
classical signal-processing approaches can be used to reconstruct coverage maps.
However, dense sampling can be very costly or impracticable, and in general only
a subset of radio measurements of an area are available at a given time. By mak-
ing assumptions on the spatial correlation properties of radio measurements, which
are strongly related to structural properties of an area, and by fitting correspond-
ing correlation models, statistical estimators such as kriging interpolation are able
to produce precise estimates based on only few measurements. However, the price
for this precision is the high computational complexity and questionable scalability.
Nevertheless, the spatial correlation exploited by kriging approaches suggests that
coverage maps contain redundant information, so, if represented by a matrix, radio
maps can be assumed to be of low rank. This observation has led some authors to
propose the framework of low-rank matrix completion for coverage map estimation,
which is the topic of this section.
Matrix completion builds on the observation that a matrix that is of low rank or
approximately low rank can be recovered by using just a subset of randomly observed
data [27,28]. A major advantage of matrix completion is that it is able to recover a
matrix by making no assumption about the process that generates the matrix, except
that the resulting matrix is of low rank. In the context of radio map estimation, matrix
Channel prediction based on machine-learning algorithms 117

completion has been successfully applied in [13,29,30]. Although in [31] standard

matrix completion is used for radio map construction, the authors in [29] consid-
ered the non-consistency (continuity) observed in radio map constructed via matrix
completion. More specifically, they add a smoothness constraint to the reconstruc-
tion problem. Building upon the approaches in [29,31], the authors in [13] use SVT
techniques for the reconstruction. In order to increase the estimation quality, which
generally degrades in areas with low spatial correlation, the query by committee
(QbC) rationale is used in [13] to identify areas requiring more samples in order to
obtain accurate predictions. An online algorithm for matrix completion is introduced
by the same authors in a subsequent work [30], where they propose an alternating
least squares (ALSs) algorithm as an alternative to the popular stochastic gradient
descent approach, popularized in the context of the Netflix prize problem [27].
Matrix completion, i.e., low-rank matrix recovery with missing entries, has
attracted much attention in recent years because it plays an important role in informa-
tion retrieval and inference and has numerous applications in computer vision, data
mining, signal processing, bioinformatics, and machine learning. For the following
theoretical review on matrix completion, we denote by P ∈ Rm×n the matrix to be
recovered, which in the present case is a two-dimensional coverage map containing
path-loss values. Without any assumptions, it is impossible to recover reliably P with
a small number d mn, of measurements. However, for the case that the rank of
the matrix P is small enough compared to its dimensions, the matrix completion
framework shows that full recovery of P is possible with high probability. More pre-
cisely, full recovery is feasible with high probability from d ≥ cn6/5 r log(n) uniformly
random measurements, with r being the matrix rank r = rank(P) and n > m.
Due to the regular propagation of a radio wave in unobstructed environments,
pass-loss maps exhibit spatial correlation and smooth patterns. Hence, they can be well
approximated by low-rank matrices. For the specific case of coverage map estimation,
first a matrix P representing the area of interest is defined, and this matrix contains
measured values and the respective missing entries. This matrix is used to represent
the physical space, where each cell corresponds to a physical position. The values of
the matrix are either zero, for the case of a missing entry, or contain the measured
path loss at the given cell. The problem of estimating missing entries using the matrix
completion framework can be informally formulated as follows: compute a low-rank
matrix A that has entries equal to the observation matrix P at the positions containing
observed measurements.
Nuclear norm minimization-based methods
Let us denote by the set of observed entries. Formally, the matrix completion
problem is formulated as the following nonconvex optimization problem [27]:
min rank(A)
A
(3.13)
s.t. Aij = Pij , ∀i, j ∈
where Pij and Aij are the {i, j}th entry of P and A, respectively, {i, j} ∈ . Unfortunately,
the problem of rank minimization is NP-hard. Therefore, existing approaches in the
literature replace the intractable problem by a relaxed formulation that can be solved
118 Applications of machine learning in wireless communications

efficiently with convex optimization tools. (The relaxed problems are often analyzed
to check the number of measurements required to recover the solution to the original
NP-hard problem exactly, with high probability.) In particular, a common relaxation
of the rank minimization problem is formulated as
min A∗
A
(3.14)
s.t. Aij = Pij , ∀i, j ∈
where A∗ denotes the nuclear norm of the matrix A which is defined as

min(m,n)
A∗ = σk (A)
k=1

with σk (·) being the kth largest singular value of a matrix. Note that (3.14) can
be converted into a semidefinite programming (SDP) and hence can be solved by
interior-point methods. However, directly solving the SDP has a high complexity.
Several algorithms faster than the SDP-based methods have been proposed to solve
the nuclear norm minimization, such as SVT, fixed point continuation (FPC), and
proximal gradient descent [5]. In radio map reconstruction, the authors of [13] opt
for the SVT algorithm, which can be briefly described as follows. Starting from an
initial zero matrix Y0 , the following steps take place at each iteration:
Ai = shrink(Yi−1 , τ )
(3.15)
Yi = Yi−1 + μ (P − Xi )
with μ being a nonnegative step size. The operator (X ) is the sampling operator
associated with the set . Entries not contained in the index set are set to zero,
the remaining entries are kept unchanged. The shrink operator (·, τ ) is the standard
rank-reduction thresholding function, which sets singular values beneath a certain
threshold τ > 0 to be zero.
In [13], the authors also introduce a method to improve the path-loss reconstruc-
tion via matrix completion. The idea is to define a notion of “informative areas,” which
are regions in which samples are required to improve greatly the map reconstruction.
The motivation for this approach is that, in coverage maps, there may exist nonsmooth
transitions caused by abrupt attenuation of signals, which are common when radio
waves impinge on obstacles such as large buildings, tunnels, metal constructions.
Consequently, path loss in such areas exhibits low spatial correlation, which can lead
to reconstruction artifacts that can only be mitigated by increasing the sampling rate
in those regions. In order to identify those regions, which are mathematically rep-
resented by matrix entries, the authors of [13] resort to a family of active learning
algorithms, and, in particular, they employ the QbC rationale. The general approach
is to quantify the uncertainty of the prediction in each missing value in the matrix, so
only measurements corresponding to the most uncertain entries are taken. In the QbC
rationale, the missing matrix values are first estimated by means of many different
algorithms, and only a subset of the available data is used. Assuming that the available
data budget amounts to k measurements, first the coverage map is computed by only
Channel prediction based on machine-learning algorithms 119

using l < k of the available entries. Then, three different algorithms for matrix recon-
struction are compared, and the top K = k − l entries with the largest disagreement
are chosen. New measurements for those K entries are then gathered, and a new cov-
erage map is estimated by using the new samples. The three different reconstruction
algorithms used in [13] are the SVT, the K-nearest neighbors, and the kernel APSM.
In a subsequent work [30], the authors from [13] derive an online algorithm
based on the ALS method for matrix completion. They adopt the matrix factorization
framework in which the low-rank matrix A is replaced by the low-rank product LRT ,
with L ∈ Rm×ρ and RT ∈ Rn×ρ , and ρ is a prespecified overestimated of the rank
of A. Based on this framework, the rank-minimization objective is replaced by the
equivalent objective:

minL,R 1
2
(L2F + R2F )
s.t. LRT = A (3.16)
Aij = Pij , ∀i, j ∈

For noisy case, the objective function for matrix completion becomes in [30]:

minL,R P − LRT 2F + γ (L2F + R2F )

s.t. LRT = A (3.17)
Aij = Pij , ∀i, j ∈

with γ being a regularization parameter that controls the trade-off between the close-
ness to data and the nuclear norm of the reconstructed matrix. The ALS method is
a two-step iterative method in which the objective is consecutively minimized over
one variable by holding the other constant. Hence, two quadratic programs have to
be solved in each iteration step consecutively. This amounts to solving D least square
problems to find the optimum solution of each row of L and R. This, however, amounts
computing a (ρ × ρ) matrix inversion for each row, which might become prohibitive
with increased number of samples. Therefore, the authors in [30] propose an approx-
imation algorithm, in which the coefficients of the optimum row vector are computed
one by one, which significantly reduces the computational complexity, especially for
sparse datasets. In this mindset, the online version of the ALS is proposed in a way
that, with new incoming data, only the respective coefficients are updated via this
approximated update function.
In addition to the online reconstruction algorithm for matrix-completion-based
coverage map reconstruction, the authors in [30] also derive a new adaptive sampling
scheme, able to outperform the QbC rationale from their previous work. They assume
that coverage maps are in general smooth. Therefore, for two neighboring matrix
entries (i1 , j1 ) and (i2 , j2 ) that satisfy |i1 − i2 | ≤ 1 and | j1 − j2 | ≤ 1, the entry difference
should be bounded by

|Ai1 j1 − Ai2 j2 | ≤
120 Applications of machine learning in wireless communications

where is a small positive number. Under this assumption, incomplete or erroneous

reconstruction will very likely violate this condition. In order to detect eventual vio-
lations of the gradient bound, a two-dimensional edge-detector filter was proposed
in [30] with the following kernel:
⎡ ⎤
−1 −1 −1
1⎢ ⎥
f = ⎣ −1 8 −1 ⎦ .
9
−1 −1 −1
The data smoothness condition on A implies that each entry of its filtered version
Ā will be bounded by |Āi,j | ≤ (8/9). With Y being the current estimate of A, the
authors of that study propose to obtain measurements corresponding to entries for
which |Ȳi,j | is large. Since the filter is a bounded linear operator, if the reconstruction
is reliable, then we should have
Ā − Ȳ ≤ M A − Y ≤ ε
for ε small enough. By considering the triangular inequality, the following constraint
can be imposed on each coefficient |Ȳi,j |:
8
|Ȳi,j | ≤ |Ȳi,j − X̄i,j | ≤ ε +
9
which implies that coefficients of Ȳ should be small. However, if the entries Ȳi,j
are large, typically larger than , then |Ȳi,j − X̄i,j | should also be large, and the
matrix completion algorithm fails to reconstruct the coverage map correctly. As a
consequence, we should obtain measurements in the region of the respective entry
(i, j) in order to increase the accuracy of the estimation. The proposed online algorithm
and adaptive sampling scheme are tested based on the image data. A 150 × 150 gray-
scale image is considered. At first, only 8% of the entire dataset is considered for the
reconstruction. In steps of N = 20 entries, which are selected based on the proposed
adaptive sampling scheme and the QbC rationale, the reconstruction is refined. It
is shown that 100 selected entries based on the adaptive sampling schemes resulted
in better improvements than 1,000 randomly selected entries. The proposed adaptive
sampling scheme outperformed the QbC approach. Further, it is also shown that in
combination with the proposed online algorithm, the QbC approach, which is based
on batch algorithms, is also outperformed in terms of computational complexity.
An alternative approach for producing smooth coverage maps is proposed in
[29]. The low-rank minimization objective is here extended by adding a smoothness
constraint term. The revised low-rank model is formulated as follows [29]:
minL,R 1
2
(L2F + R2F ) + λs(LRT )
s.t. LR = A
T
(3.18)
Aij = Pij , ∀i, j ∈
where s(A) is the smoothing term, and the regularization term λ is a weight that
balances low rankness and smoothness. In [29], the smoothness constraint term is
Channel prediction based on machine-learning algorithms 121

defined via the diversity of the row-wise and column-wise difference of LRT = A, or,
in mathematical terms:
s(LRT ) = Dx (LRT )2F + Dy (LRT )2F
with the gradient operators Dx (A) and Dy (A) being defined as
Dx (i, j) = A(i, j + 1) − A(i, j)
Dy (i, j) = A(i + 1, j) − A(i, j)
The smoothed matrix completion objective from is then stated as

minL,R 21 (L2F + R2F ) + λ Dx (LRT )2F + Dy (LRT )2F
s.t. LRT = A (3.19)
Aij = Pij , ∀i, j ∈
For the solution of the minimum problem in (3.19), an alternating iteration algorithm
over L and R is adopted in [29]. At first, L and R are chosen at random, then L is fixed
and R is optimized by a linear least square method. Then R is fixed and the cost function
is optimized over L. This procedure is repeated until no progress is observed. In [29],
the proposed smoothed low-rank reconstruction method is compared with interpola-
tion methods such as radial basis interpolation and inverse distance weighting. The
smoothed low-rank reconstruction method shows to achieve similar reconstruction
properties with fewer samples compared to these methods.
Alternating projection methods
Note that these nuclear-norm-based algorithms require performing the full SVD of
an m × n matrix. When m or n is large, computing the full SVD is time-consuming.
Different from rank or nuclear norm minimization, a new strategy is adopted for
matrix completion. The basic motivation of alternating projection algorithm (APA)
is to find a matrix such that it has low rank and its entries over the sample set are
consistent with the available observations. Denote the known entries as P :

Pi,j , if (i, j) ∈
[P ]i,j = . (3.20)
0, otherwise
Then it can be formulated as the following feasibility problem [32]:
find A
(3.21)
s.t. rank(A) ≤ r, A = P ,
where r min(m, n) is the desired rank. It is obvious that (3.21) is only suitable for
noise-free case. For noisy case, we use:
find A
(3.22)
s.t. rank(A) ≤ r, A − P F ≤ ε2
to achieve robustness to the Gaussian noise. In the presence of outliers, we adopt:
find A
(3.23)
s.t. rank(A) ≤ r, A − P p ≤ εp
122 Applications of machine learning in wireless communications

where εp > 0 is a small tolerance parameter that controls the p -norm of the fitting
error and ·p denotes the element-wise p -norm of a matrix, i.e.:
⎛ ⎞1/p
p
A p = ⎝ [A]i,j ⎠ . (3.24)
(i,j)∈

Apparently, (3.23) reduces to (3.22) when p = 2. Also, (3.23) reduces to the noise-free
case of (3.21) if εp = 0.
By defining the rank constraint set
Sr := {A|rank(A) ≤ r} (3.25)
and the fidelity constraint set

Sp := A| A − P p ≤ εp , (3.26)
the matrix completion problem of (3.23) is formulated as finding a common point of
the two sets, i.e.:
find X ∈ Sr ∩ Sp . (3.27)
For a given set S , the projection of a point Z ∈
/ S onto it, which is denoted as
S (Z), is defined as
S (Z) := arg min X − Z2F . (3.28)
X ∈S

We adopt the strategy of alternating projection (AP) onto Sr and Se to find a common
point lying in the intersection of the two sets [32]. That is, we alternately project onto
Sr and Sp in the kth iteration as
Y k = Sr (Ak )
(3.29)
Ak+1 = Sp (Y k ).
The choice of p = 1 is quite robust to outliers. Other values of p < 2 may also be of
interest. The case of p < 1 requires to compute the projection onto a nonconvex and
nonsmooth p -ball, which is difficult and hence not considered here. The 1 < p < 2
involves the projection onto a convex p -ball, which is not difficult to solve but requires
an iterative procedure. Since the choice of p = 1 is more robust than 1 < p < 2 and
computationally simpler, we can use p = 1 for outlier-robust matrix completion.
By Eckart–Young theorem, the projection of Z ∈ / Sr onto Sr can be computed
via truncated SVD of Z:
r
Sr (Z) = σi ui viT (3.30)
i=1

where {σi }ri=1 ,

{ui }ri=1 ∈ Rm , and {vi }ri=1 ∈ Rn are the r largest singular values and
the corresponding left and right singular vectors of Z, respectively. Clearly, the AP
does not need to perform the full SVD. Only truncated SVD is required. That is, we
only calculate the r largest singular values and their corresponding singular vectors.
Channel prediction based on machine-learning algorithms 123

Without loss of generality, assuming n ≤ m, the computational cost of full SVD is

O(mn2 + n3 ), while that of truncated SVD is O(mnr). In practical applications, the
rank r could be much smaller than the matrix dimension. Therefore, the computational
cost of the AP is much lower than the nuclear norm minimization-based methods that
need full SVD.
We then investigate computing the projection onto Sp for p = 1 and p = 2. Note
that projection onto Sp only affects the entries indexed by . Other entries {Zi,j } with
/ will remain unchanged through this projection. Define p ∈ R|| , where
(i, j) ∈
|| is the cardinality of , as the vector that contains the observed entries of P, i.e.,
the nonzero entries of P . Also, a ∈ R|| is defined in a similar manner. Then the
set Sp of (3.26) has the equivalent vector form:

Bp := a ∈ R|| a − p p ≤ εp (3.31)
which is an p -ball with the observed vector p being the ball center. Now it is clear
that the projection for matrices is converted into one for vectors of length ||. We
consider the following three cases with different values of p and εp :
● For εp = 0, (3.31) reduces to the equality constraint of a = p . For any vector
z ∈ R|| , the projection is calculated as Bp (z) = p .
● For p = 2 and ε2 > 0, B2 is the conventional 2 -ball in the Euclidean space. For
any vector z ∈ / B2 , it is not difficult to derive the closed-form expression of the
projection onto B2 as
ε2 (z − p )
B2 (z) = p + . (3.32)
z − p 2
With a proper value of ε2 and p = 2, the robustness to Gaussian noise can be
enhanced.
● For p = 1 and ε1 > 0, B1 is an 1 -ball. For any vector z ∈
/ B1 , the projection
onto B1 is the solution of
1
min a − z22 , s.t. a − p 1 ≤ ε1 . (3.33)
a 2
Using the Lagrange multiplier method, we obtain the solution of (3.33):
[B1 (zz ) ]i = sgn([z − p ]i ) max(|[z − p ]i | − λ , 0) (3.34)
where i = 1, . . . , || and λ is the unique root of the nonlinear equation:
||

max(|[z − p ]i | − λ, 0) = ε1 (3.35)
i=1

in the interval (0, z − p ∞ ) using the bisection method, where ·∞ is the ∞ -
norm of a vector. The computational complexity of projection onto 1 -ball is
O(||), which is much lower than that of the projection onto Sr .
The selection of εp is critical to the performance of APA. In the absence of noise,
the optimum is εp = 0. For noisy case, εp is related to the noise level. Roughly
speaking, larger noise requires a larger εp . If the probability of the noise is known
124 Applications of machine learning in wireless communications

a priori, we can estimate the probability distribution of the p -norm of the noise.
Then a proper value of εp can be determined according to the probability such that
the true entries are located in the p -ball. If the probability of the noise is unknown,
one may resort to cross validation to determine a proper εp . Note that in the nuclear
norm regularized problem:
1
min A − P 2F + τ A∗ (3.36)
A 2

one also faces the issue of selecting the regularization parameter τ . Clearly, an advan-
tage of the proposed formulation is that it is not difficult to determine εp from the
a priori noise level but not easy for τ .
Remark: It should be pointed out that the APA is different from the iterative hard
thresholding (IHT) and its variants [33,34] although they all use a rank-r projection.
The IHT solves the rank constrained Frobenius norm minimization:
1
min f (A) := A − P 2F , s.t. rank(A) ≤ r (3.37)
A 2
using gradient projection with iteration step being:

Ak+1 = Sr Ak − μ∇f (Ak ) (3.38)
where μ > 0 is the step size and ∇f is the gradient of f . Determining the step size
with a line search scheme requires computing the projection Sr (·) for several times.
Thus, the computational cost of the IHT is several times of the APA per iteration.
Convergence of the alternating projection for finding a common point of two sets
was previously established for convex sets only [35]. Recently, the convergence of
APA for nonconvex sets, which satisfies a regularity condition has been investigated
[36,37]. Exploiting the fact that the rank constraint set of (3.25) satisfies the prox-
regularity and according to Theorem 5.2 of [36], we can establish the convergence of
the APA for matrix completion, as stated in the following proposition.
Proposition: The APA locally converges to a point in Sr ∩ Sp at a linear rate.

3.3.2 Online algorithms

The batch algorithms applied in coverage map reconstruction was widely studied as
mentioned above. However, batch algorithms need a great number of storage and is
poor in real-time performance. It is more practical that an algorithm is capable of
updating the corresponding base station’s current approximation of the unknown path
loss function in its cell. Thus, coverage map reconstruction needs to be an online
function itself. In [9], APSM-based [10] and multi-kernel learning techniques [38]
are adopted for their capability of coping with large-scale problems where the huge
number of measurements arrives to operators.
3.3.2.1 APSM-based algorithm
APSM is a recently developed tool for iteratively minimizing a sequence of convex
cost functions [10]. And it can be easily combined with kernel-based tools from
machine learning [10,39,40]. In particular, a variation of the APSM is proposed in
this study.
Channel prediction based on machine-learning algorithms 125

In more detail, at each iteration, n, q sets are selected from the collection
{S1 , . . . , Sn } with the approach described in [9]. The intersection of these sets is the
set Cn and the index of the sets chosen from the collection is denoted by
(n) (n)
In,q := {ir(n)
n
, irn −1 , . . . , irn −q+1 } ⊆ {1, . . . , n}, (3.39)

where n ≥ q, and rn is the size of dictionary. With this selection of sets, starting from
fˆ0 = 0, sequence { fˆn }n∈N ⊂ H by

fˆn+1 := fˆn + μn ωj,n PSj ( fˆn ) − fˆn , (3.40)
j∈In,q

where μn ∈ (0, 2Mn ) is the step size, Mn is a scalar given by

⎧ 2
⎪
⎪ wj,n PSj ( fn ) − fn
⎨ j∈In,q
2 , if fn ∈
/ ∩j∈In,q Sj ,
Mn = (3.41)
⎪
⎪ j∈In,q wj,n PSj ( fn ) − fn
⎩
1, otherwise,

and ωj,n > 0 are weights satisfying:

ωj,n = 1 (3.42)
j

The projection onto the hyperslab induced by measurement n is given by PSn ( f ) =

f + βf κ(x̃n , ·) where
⎧
⎪ y − f , κ(x̃n , ·) − ε
⎪
⎪ , if f , κ(x̃n , ·) − y < −ε,
⎪
⎪ κ(x̃n , x̃n )
⎨
βf = y − f , κ(x̃n , ·) + ε (3.43)
⎪
⎪ , if f , κ(x̃n , ·) − y > ε,
⎪
⎪ κ(x̃n , x̃n )
⎪
⎩
0, if | f , κ(x̃n , ·) − y| ≤ ε.

3.3.2.2 Multi-kernel algorithm

The choice of the kernel κ for a given estimation task is one of the main challenges for
the application of kernel methods. To address this challenge in the path-loss estimation
problem, we propose the application of the multi-kernel algorithm described in [38].
Briefly, this algorithm provides good estimates by selecting, automatically, both a
reasonable kernel (the weighted sum of a few given kernels) and a sparse dictionary.
The APSM-based online algorithm demonstrated above has good performance in
real-time and requires little storage. Nevertheless, its time complexity and accuracy
are inferior to common batch algorithms. What should be noticed is that so far only
the online algorithm mentioned in this section has been employed in coverage map
reconstruction. So there is still a great deal of open questions in this topic.
126 Applications of machine learning in wireless communications

3.4 Optimized sampling

Informative areas are areas from which we want to have samples, since such knowl-
edge can improve the path-loss reconstruction. Note that some regions can be
nonsmooth. This is the consequence of large buildings, obstacles, tunnels that can
abruptly attenuate the propagating radio wave. Due to these, the path loss in such
areas exhibits low spatial correlation and this can lead to poor reconstruction effects.
Consequently, in this case, optimized samplings are required.

3.4.1 Active learning

Active learning is a special case of semi-supervised machine learning in which a
learning algorithm is able to interactively query the user (or some other information
source) to obtain the desired outputs at new data points. In active learning systems, an
algorithm is able to choose its training data, as opposed to passively trained systems
that must learn a behavior from a set of random observations.

3.4.1.1 Query by committee

A QbC training strategy for selecting new data points uses the disagreement between
a committee of different algorithms to suggest new data points, which most rationally
complement existing data, that is, they are the most informative data points.
In the application of [13], assume that the available budget corresponds to k
measurements, coming from drive tests, the matrix is first completed using a number
of l < k observed entries. Subsequently, having access to a number of reconstructed
matrices, finding the top K := k − l entries with the largest “disagreement” according
to a certain criterion and obtaining measurements from them. Finally, drive tests are
performed to obtain the K samples indicated by the previous step and reconstruct the
path-loss map exploiting the newly obtained information.
In general, one can employ any number of algorithms to reconstruct the matrix.
These algorithms run in parallel using the same set of measurements as an input. After
the estimation of the missing entries, the entries with the largest disagreement can
be obtained according to the following simple rules. Supposing that three algorithms
are employed as committees, the entries obtained by these algorithms are denoted by
aij (ξ ), ξ = 1, 2, 3. Then the disagreement equals to

(1) (2) (12) (3) (1) (3)

dij = (aij − aij )2 + (aij − aij )2 + (aij − aij )2 . (3.44)

The K entries, which score the largest disagreement, are chosen, and we perform
drive tests to obtain the path loss.
QbC algorithm is a simple one which can be implemented easily. And this algo-
rithm enhances the accuracy of the reconstruction as illustrated in [13]. However,
for that larger than two algorithms required to run parallel, it cannot be one of the
most efficient algorithms. And the predicted results can be influenced greatly if the
algorithms employed are not stable enough.
Channel prediction based on machine-learning algorithms 127

3.4.1.2 Side information

In [9], in order to predict the informative areas, the approach based on side information
is applied to decide the weight of parallel projections. For instance, sets corresponding
to measurements taken at pixels farther away from the route of the user of interest
(UOI) should be given smaller weights than measurements of pixels that are close to
the user’s trajectory. The reason is that estimates should be accurate at the pixels the
UOI is expected to visit because these are the pixels of interest to most applications
(e.g., video caching based on channel conditions). Therefore, we assign large weights
to measurements close to the UOI’s route by proceeding as follows. Let χUOI ⊂ N 2
be the set of pixels that belong to the path of the UOI. Then, for each weight ωi,n :
1
ωi,n = (3.45)
dmin (x̃i , χUOI ) + εω
where dmin (x̃i , χUOI ) denotes the minimum distance of measurement x̃i to the area of
interest, and εω > 0 is a small regularization parameter. This distance can be obtained
for each pixel x̃i by considering the distances of every pixel in χUOI to x̃i and by taking
the minimum of these distances. Subsequently, the weights are normalized. Compared
to an equal choice of the weights, the proposed method provides fast convergence to
a given prediction quality for the UOI, but at the cost of degraded performance in
other areas of the map.

3.4.2 Channel prediction results with path-loss measurements

We implement the matrix completion via the AP algorithm for channel prediction,
whose performance is compared with the state-of-the-art method, i.e., SVT [5]. The
experimental data is collected in Tengfei Industrial Park, and the scenario of data
collection is illustrated in Figure 3.2. Specifically, the UE routes, including the

Figure 3.2 Measurement scenario in Tengfei industrial park

128 Applications of machine learning in wireless communications

line-of-sight (LOS) and non-LOS routes, are in red lines and yellow lines, respec-
tively. With the data points being sampled from these UE routes, our purpose is to
predict the values of other points and hence to achieve the path-loss reconstruction.
We devise an experiment to evaluate the prediction performance of both AP and
SVT. The experiment is based upon the fact that only the fractional sampling points
are viewed as the known points. And the rest part of sampling points is considered to
be unknown in advance. Hence, those points are supposed to be predicted. Assuming
that 1 is a subset, consisting of those predicted points, of the set and Pi,j , (i, j) ∈ 1
denotes the true value of the (i, j) point. We compare the predicted value Pi,j , (i, j) with
its true value. If the condition

Pi,j , (i, j) − Pi,j , (i, j) ≤ δ
is satisfied, the prediction with respect to the (i, j) point is assumed to be successful,
otherwise the prediction is failed. In our experiment, we set δ = 20 and investigate
the successful ratio of prediction with respect to the rank r. For each value of the
matrix rank, 100 trials are carried out to calculate the average result. The proportion
of known sampling points, which are randomly selected in each trial, is 85%. Hence,
the rest of 15% sampling points is viewed as predicted points.
Note that AP’s performance is affected by the parameter of the estimated rank r,
while SVT’s performance is determined by the parameter of the threshold of singular
value. Hence, we evaluate the performance of each algorithm with respect to different
parameters. In terms of the AP, Figure 3.3 plots the successful ratio of prediction
versus the rank r. It is observed that using r = 2 yields the best performance. When
r > 2, the successful ratio is successively decreased with the increase of the rank. This
phenomenon shows that only when tight rank constraint with quite small r is adopted,
the reasonable prediction can be obtained. With the SVT in contrast, Figure 3.4 plots
its successful ratio of prediction versus the threshold of singular value. While an
appropriate threshold of singular value yields the highest successful ratio, too small
or too large threshold value will result in a decreased successful ratio.

0.8

0.7

0.6
Successful ratio

0.5

0.4

0.3

0.2

0.1
2 3 4 5 6 7
Rank

Figure 3.3 Successful ratio of prediction versus rank r with ε2 = 0 via AP

Channel prediction based on machine-learning algorithms 129

Based upon the optimal parameters in two algorithms (rank r = 2 in the APA and
threshold of singular value equal to 1.5 × 105 in the SVT), we compare the highest
successful ratio that two algorithms can attain. Observe that the highest successful
ratio of the APA is 72.9% and that of the SVT is 67.8%. We can see that the AP
outperforms the SVT.
Then we evaluate the prediction errors of both algorithms by the root mean square
error (RMSE) which is defined as

RMSE = 10 log10 E{ P1 − P1 2F }. (3.46)
Figure 3.5 plots the RMSE versus the rank r of AP, which demonstrates that the best
performance can be achieved when r = 2. This RMSE result is consistent with the
above successful ratio result. In those two aspects of evaluation, the choice of r = 2

0.7

0.6
Successful ratio

0.5

0.4

0.3
0.1 0.2 0.5 1 1.5 2
Threshold of singular value × 105

Figure 3.4 Successful ratio of prediction versus threshold of singular value via SVT

10.5

10
RMSE

9.5

8.5

7.5
2 3 4 5 6 7
Rank

Figure 3.5 RMSE versus rank r with ε2 = 0 via AP

130 Applications of machine learning in wireless communications

10.5

9.5
RMSE

8.5

7.5
0.10.2 0.5 1 1.5 2
× 105
Threshold of singular value

Figure 3.6 RMSE versus threshold of singular value via SVT

derives the best successful ratio and prediction error. Therefore, we can conclude that
adopting r = 2 as the estimated rank for the AP yields the best prediction. In contrast,
Figure 3.6 plots the RMSE versus the threshold of singular value for the SVT. While
the SVT can attain the smallest RMSE value of 8.57 with threshold of singular value
equal to 1 × 105 , the AP can obtain the smallest RMSE value of 7.92 with rank r = 2.
This comparison proves the better performance of the AP than the SVT.

3.5 Conclusion

Numerous wireless communication applications in future will greatly depend on accu-

rate coverage loss map. Making sense of the coverage loss map reconstruction can
be a precarious task for the uninitiated research. In this survey, methodologies of
reconstruction are divided into three parts to discuss the following: approaches of
achieving measurements, learning-based reconstruction algorithms, and optimized
sampling of measurements. And different methodologies of each part are studied
and analyzed, respectively. Mainly, two approaches can be applied to achieve mea-
surements: conventional drive tests and MDTs. The former one is simple and stable,
while the cost of it will rapidly ascend with the measured area getting larger. MDTs
are a relatively cheap and efficient way without the consideration of absolute stabil-
ity. Then learning-based reconstruction algorithms which can be categorized into two
parts are discussed. Batch algorithms are usually efficient and cheap, but it requires
huge amount of storage and performs in real-time calculating. Among batch algo-
rithms, SVM is hard to implement for which resolving nonlinear regression should
be complicated; SVT is easy to be implemented, but it is too simple to adapt to var-
ious scenarios. Over adaptation may occur in the training procedure of ANN, which
manifests that it cannot be an adaptive algorithm. On the other hand, online algo-
rithm has great performance in real-time and storage is unnecessary. The only online
Channel prediction based on machine-learning algorithms 131

approach having been employed in path-loss reconstruction is APSM-based algorithm

with multi-kernel techniques. It successfully updates the prediction of coverage map
whenever new data comes in. Nevertheless, the time complexity and accuracy are
inferior to common batch algorithms. The last part is optimized sampling. The main
techniques applied in this part is active learning which is derived from machine learn-
ing. There are mainly two active learning algorithms in the procedure of optimized
sampling: QbC and side information. QbC determines the informative areas through
running various reconstruction algorithms in parallel. Its accuracy is considerable,
but the stability and time complexity are poor. Side information is applied in the
online algorithm mentioned above. Informative areas are selected through exerting
different weights. This is a simple and efficient way, but whether it is robust should
be studied more deeply. According to the generalization discussed above, learning-
based algorithms employed in coverage loss map reconstruction has a great area to
be explored, especially in online algorithm. How to revise an efficient and real-time
learning-based algorithm may be a meaningful topic to study on.

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Chapter 4
Machine-learning-based channel estimation
Yue Zhu1 , Gongpu Wang1 , and Feifei Gao2

Wireless communication has been a highly active research field [1]. Channel estima-
tion technology plays a vital role in wireless communication systems [2]. Channel
estimates are required by wireless nodes to perform essential tasks such as precoding,
beamforming, and data detection. A wireless network would have good performance
with well-designed channel estimates [3,4].
Recently, artificial intelligence (AI) has been a hot research topic which attracts
worldwide attentions from both academic and industrial circles. AI, which aims to
enable machines to mimic human intelligence, was first proposed and founded as
an academic discipline in Dartmouth Conference in 1956 [5]. It covers a series of
research areas, including natural language processing, pattern recognition, computer
vision, machine learning (ML), robotics, and other fields as shown in Figure 4.1.
ML, a branch of AI, uses statistical techniques to develop algorithms that can
enable computers to learn with data and make predictions or yield patterns. According
to different learning styles, ML can be divided into supervised learning, unsuper-
vised learning, semi-supervised learning, and reinforcement learning. Typical ML
algorithms include support vector machine (SVM) [6], decision tree, expectation-
maximization (EM) algorithm [7], artificial neural network (NN), ensemble learning,
Bayesian model, and so on.
Currently, one of the most attractive branches of ML is deep learning proposed
by Geoffrey Hinton in 2006 [8]. Deep learning is a class of ML algorithms that can
use a cascade of multiple layers of nonlinear processing units for feature extraction
and transformation. Its origin can be traced back to the McCulloch–Pitts (MP) model
of neuron in the 1940s [9]. Nowadays, with the rapid development in data volume
and also computer hardware and software facilities such as central processing unit,
graphic processing unit, and TensorFlow library, deep learning demonstrates powerful
abilities such as high recognition and prediction accuracy in various applications.
In short, ML is an important branch of AI, and deep learning is one key family
among various ML algorithms. Figure 4.2 depicts a simplified relationship between
AI, ML, and deep learning.

1
School of Computer and Information Technology, Beijing Jiaotong University, China
2
Tsinghua National Laboratory for Information Science and Technology, Tsinghua University, China
136 Applications of machine learning in wireless communications

Supervised learning
Nature language processing Classification Unsupervised learning
of learning
styles Semi-supervised learning
Pattern recognition
Reinforcement learning
Machine learning
AI SVM
Computer vision
ANN Deep learning
•••

Traditional
machine Decision tree
Robotics learning

•••
algorithm
EM

Ensemble learning

Figure 4.1 The research branches of AI and ML

AI ML DP

Figure 4.2 The relationship between AI, ML, and deep learning

In recent years, a range of ML algorithms have been exploited in wireless com-

munication systems to address key issues. Reference [10] has proposed a Bayesian
channel estimator with a substantial improvement over the conventional estimators
in the presence of pilot contamination. Besides, a blind estimator based on EM algo-
rithm [11] has been introduced which requires no training symbols and outperforms
the existing training-aided estimators. Some deep-learning methods [12–14] have also
been exploited to enhance channel estimation and detection performance of wireless
communication systems. In addition, one new wireless communication architecture
on the basis of an ML-aided autoencoder has been suggested in [15].
In this chapter, we first review the channel model for wireless communication
systems and then describe two traditional channel estimation methods, and finally
introduce two newly designed channel estimators based on deep learning and one
EM-based channel estimator.
Machine-learning-based channel estimation 137

4.1 Channel model

The wireless channel is described by the response h(t, τ ) at time t to an impulse
transmitted at time t − τ . The channel consists of several independent paths. For this
multipath model, the general expression can be written as [16]:

h(τ , t) = ai (t)δ(τ − τi (t)), (4.1)
i

where ai (t) is the attenuation and τi (t) is the delay from the transmitter to the receiver on
the ith path. An example of a wireless channel with three paths is shown in Figure 4.3.
The general expression (4.1) is also known as a doubly selective channel since
there are several paths and the attenuations and delays are functions of time. The
following two special cases for h(t, τ ) are widely used:

● Time-invariant frequency-selective channel: This channel occurs when the trans-

mitter, receiver, and the environment are all stationary so that the attenuations
ai (t) and propagation delays τi (t) do not depend on time t. However, the delays
are significantly large compared to the symbol period.
● Time-varying (or time-selective) flat-fading channel: The delays τi (t) in this case
are all approximately constant and small compared to the symbol period. This
channel occurs when the transmitter or the receiver is mobile and when the symbol
period of the transmitted signal significantly exceeds any of all the delays.

Since the symbol period Ts decreases when the data rate increases, the channel
can be flat fading or frequency selective depending on the data rate. Moreover, the

Transmitter Delay
τ1(t)
a2(t)

a1(t) τ2(t)

a3(t)
τ3(t)

Receiver
t

Figure 4.3 Wireless channel model

138 Applications of machine learning in wireless communications

delay spread is another relevant parameter. Delay spread Td is defined as the difference
in propagation delay between the longest and shortest path:
Td = max |τi (t) − τj (t)|. (4.2)
i,j

When Ts is much larger than Td , the channel is flat fading. Otherwise, the channel is
frequency selective. For example, the typical delay spread in a wireless channel in
an urban area is 5 μs when the distance between transmitter and receiver is 1 km [1].
When the data rate is 1 kbps, the symbol period is 1 ms, and the channel is flat-fading
since the delay is negligible compared to the symbol period. If the data rate increases
to 1 Mbps, the symbol period Ts is 1 μs. Then the channel becomes frequency selective
due to the non-negligible delays.
Furthermore, the mobility of transmitter or receiver will induce a shift in radio
frequency, which is referred to as the Doppler shift Ds . Coherence time Tc , a parameter
related to the Doppler shift, is defined as
1
Tc = . (4.3)
4Ds
If the coherence time Tc is comparable to the symbol period, the channel is time-
varying. On the other hand, in time-invariant channels, the coherence time Tc is
much larger than the symbol period (i.e., the channel remains constant). For exam-
ple, if Doppler shift Ds = 50 Hz and the transmission data rate is 1 Mbps, then the
coherence time Tc = 2.5 ms is much larger than one symbol duration 1 μs. In this
case, the channel is time invariant.
The types of wireless channels are depicted in Table 4.1.

4.1.1 Channel input and output

In terms of the wireless channel h(t, τ ), the relationship between input s(t) and output
y(t) is given by
+∞
y(t) = h(t, τ )s(t − τ )dτ + w(t), (4.4)
−∞

where w(t) is an additive white Gaussian complex noise signal. The receiver is
required to recover data signal s(t) from received signal y(t); this process is called
data detection.
For data detection, the receiver requires the knowledge of h(t, τ ), which is referred
to as channel state information (CSI). To help the receiver estimate CSI, special

Table 4.1 Different types of wireless channels

Types of channel Characteristic

Time varying Tc Ts
Time invariant Tc Ts
Flat fading Td Ts
Frequency selective Td Ts
Machine-learning-based channel estimation 139

predefined symbols may be transmitted in addition to data symbols. These symbols

are called pilot symbols or training symbols. Pilot symbols are utilized by the channel
estimator at the receiver to obtain CSI.
In practice, channel estimation and data detection are done by using the discrete-
time baseband signals. Define the samples y(nTs ) = y(n) for n = 0, 1, . . . , N − 1.
The discrete-time baseband model equivalent to (4.4) can then be obtained as

L
y(n) = h(n, l)s(n − l) + w(n), (4.5)
l=0

where h(n, l) is the sampling version of h(t, τ ), i.e., h(n, l) = h(nTs , lTs ), and s(n − l)
is the sampling version of s(t), i.e., s(n − l) = s((n − l)Ts ), and L + 1 is the number of
multipaths and w(n) is complex white Gaussian noise with mean zero and variance σw2 .

4.2 Channel estimation in point-to-point systems

4.2.1 Estimation of frequency-selective channels

For a frequency-selective time-invariant channel where h(n, l) does not change with
time index n, i.e., h(n, l) = h(l), the model (4.5) can be simplified as

L
y(n) = h(l)s(n − l) + w(n). (4.6)
l=0

Define y = [y(0), y(1), . . . , y(N − 1)]T , w = [w(0), w(1), . . . , w(N − 1)]T , and h =
[h(0), h(1), . . . , h(L)]T , where N is the block length. We can further write (4.6) in the
following vector form:
y = Sh + w, (4.7)
where S is a N × (L + 1) circulant matrix with the first column s = [s(0), s(1), . . . ,
s(N − 1)]T . Note that the sequence s is the training sequence and depends on the
choice of pilots and their values.
Two linear estimators are often utilized to obtain the estimate of h from the
received signal y. The first one is least square (LS). It treats h as deterministic constant
and minimizes the square error. The LS estimate is [17]:
ĥ = (SH S)−1 SH y. (4.8)
LS estimator can be derived as follows. The square error between the real value
and the estimate value is
J (h) = (y − Sh)H (y − Sh) = yH y − 2yH Sh + hH SH Sh (4.9)
To minimize the error, the gradient of J (h) with respect to h is derived as
∂J (h)
= −2SH y + 2SH Sh (4.10)
∂h
140 Applications of machine learning in wireless communications

Setting the gradient to be zero, we can then obtain the LS estimate (4.8). For simplicity,
denote (SH S)−1 SH as S† , the LS estimate can be rewritten as
ĥ = S† y, (4.11)
†
where (·) represents the pseudo inverse. It can be readily checked that the minimum
square error of LS is
Jmin = J (h) = yT (I − S(XT S)−1 ST )y (4.12)
The second one is the linear minimum mean square error (LMMSE) estimator.
It treats h as a random vector and minimizes the mean square error.
Define Ryy = E(yyH ), Rhh = E(hhH ), and Rhy = E(hyH ), where E(x) is the
expected value of a random variable x. The LMMSE estimator can be expressed as
ĥ = Rh SH (SRh SH + σw2 I)−1 y (4.13)
LMMSE estimator can be derived as follows. As a linear estimator, the estimate
ĥ can be given as linear combination of the received signal y:
ĥ = Ay. (4.14)
LMMSE estimator aims to minimize the mean square error through choosing the
linear combinator A, i.e.:
A = arg min E(h − ĥ2 ) = arg min E(h − Ay2 ). (4.15)
A A

The mean square error can be further obtained as

E(h − Ay2 ) = E(tr{(h − A(Sh + e))(h − A(Sh + e))H }) (4.16)
= tr{Rh } − tr{Rh S A } − tr{ASRh } + tr{W(SRh S +
H H H
σw2 I)AH }
(4.17)
where tr{A} denotes the trace operation of the matrix A and σw2 is the noise variance.
Setting the derivative of MSE to the parameter A as zero, we can derive:
A = Rh SH (SRh SH + σw2 I)−1 . (4.18)
Substituting (4.18) into (4.14) will generate (4.13).
The LS estimator is simpler compared with the LMMSE estimator. But the
LMMSE estimator outperforms the LS estimator because it exploits the statistics
of h.

4.3 Deep-learning-based channel estimation

4.3.1 History of deep learning

Deep learning, suggested by Geoffrey Hinton in 2006, is rooted from NN. The ear-
liest idea about NN originated from the MP neuron model [9] proposed by Warren
McCulloch and Walter Pitts in 1943. Interestingly, we can find that there exists three
up-and-down tides of NN development during the past 70 years.
Machine-learning-based channel estimation 141

The first tide, as illustrated in Figure 4.4, took place from 1940s to 1960s [18]. The
MP neuron model, created in 1943, laid the foundation for the research of NN. Then in
1958, Frank Rosenblatt created the first machine referred to as perceptron [19] which
exhibited the ability of simple image recognition. The perception aroused huge inter-
ests and large investments during its first decade. However, in 1969, Marvin Minsky
discovered that perceptrons were incapable of realizing the exclusive OR function.
He also pointed out that the computers, due to the limited computing ability at that
time, cannot effectively complete the large amount of computation work required by
large-scale NN [20], such as adjusting the weights. The two key factors leaded to the
first recession in the development of NN.
The second wave started from the 1980s and ended at the 1990s. In 1986, David
Rumelhart, Geoffrey Hinton, and Ronald J. Williams successfully utilized the back
propagation (BP) algorithm [21] and effectively solved the nonlinear problems for
NN with multiple layers. From then on, BP algorithms gained much popularization,
which resulted in the second upsurge of NN. Unfortunately, in the early 1990s, it was
pointed out that there existed three unsolved challenges for BP algorithms. The first is
that the optimization method obtains the local optimal value, instead of global, when
training the multilayer NN. The second is the vanishing gradient problem that the
neuron weights closed to the inputs have little changes. The third is the over-fitting
problem caused by the contradiction between training ability and prediction results.
In addition, the data sets for training the NN and the computing capability at the time
also cannot fully support the requirements from the multilayer NN. Besides, SVM [6]
attracted much attentions and became one hot research topic. These factors led to a
second winter in the NN development.
The third wave emerged in 2006 when Geoffrey Hinton proposed deep belief
networks [8] to solve the problem of gradient disappearance through pretraining and
supervised fine tuning. The term deep learning became popular ever since then. Later
the success of ImageNet in 2012 provided abundant pictures for training sets and

1986, backpropagation

2006, deep learning

1958, perceptron

Mid-1990s, winter

1943, MP model 1969, winter

Time

Figure 4.4 Development trend of deep learning

142 Applications of machine learning in wireless communications

set a good example for deep-learning research. So far, the third wave is still gaining
momentums.

4.3.2 Deep-learning-based channel estimator for orthogonal

frequency division multiplexing (OFDM) systems
Figure 4.5 illustrates the functional diagram and the basic elements of a digital
orthogonal frequency division multiplexing (OFDM) communication system. At the
transmitter, the source bits X (k) follows the operations of modulation, inverse discrete
Fourier transform (IDFT), and adding cyclic prefix (CP), respectively.
Denote the multipath fading channels as h(0), h(1), . . . , h(L − 1). The signal
arrived at the receiver is
y(n) = x(n) ⊗ h(n) + w(n), (4.19)
where x(n) and w(n) indicate the transmitted signal and the noise, respectively,
and ⊗ represents the circular convolution. After removing CP and performing DFT
operation, the received signals can be obtained as
Y (k) = X (k)H (k) + W (k) (4.20)
where Y (k), X (k), H (K), and W (k) are the DFT of y(n), x(n), h(n), and w(n) respec-
tively. Finally, the source information is recovered from Y (k) through frequency
domain equalization and demodulation. Generally, the traditional OFDM receiver
first estimates the CSI H (k) using the pilot and then detects the source signal with
the channel estimates Ĥ (k).
Different from the classical design shown in Figure 4.5, a deep-learning-based
transceiver is proposed in [12] which can estimate CSI implicitly and recover the
signal directly. This approach considers the whole receiver as a black box, takes the
received signal as input of a deep NN (DNN), and outputs the recovered source bits
after calculation and transformation in the hidden layers of the NN.
The OFDM transmission frame structure is shown in Figure 4.6. One OFDM
frame consists of two OFDM blocks: one for pilot symbols and the other for data
symbols. Assume that the channel parameters remain unchanged in each frame and
may vary between frames.

x(n)
X(k)
Modulation IDFT Insert CP

Noise
Channel h(n) w(n)

y(n)
ˆ
X(k)
Demodulation FDE DFT Remove CP

Figure 4.5 Basis elements of an OFDM system

Machine-learning-based channel estimation 143

Frame 1 Frame 2 Frame 3

CP Pilot CP Data
First OFDM block Second OFDM block

Figure 4.6 OFDM frame structure for deep-learning-based transceiver

±1±1j

IFFT
0/1 sequence Pilot Add CP
QPSK
Pilot:128 bit Data:128 bit CP Pilot CP Data
Data

Channel

Signal y Remove CP Received signal

Pilot Data CP Pilot CP Data

Figure 4.7 The data-generation process

The data-generation process is depicted in Figure 4.7. Suppose the OFDM system
has 64 subcarriers and the length of CP is 16. In each frame, the first block contains
fixed pilot symbols, and the second data block consists of 128 random binary bits.
After QPSK modulation, IDFT, and CP insertion, the whole frame data is con-
volved with the channel vector. The channel vector is randomly selected from the
generated channel parameter sets based on the WINNER model [22]. The maximum
multipath delay is set as 16.
At the receiver side, the received signals including noise and interference in one
frame will be collected as the input of DNN after removing CP. The DNN model aims
to learn the wireless channel parameters and recover the source signals.
As illustrated in Figure 4.8, the architecture of the DNN model has five layers:
input layer, three hidden layers, and output layer. The real and imaginary parts of the
signal are treated separately. Therefore, the number of neurons in the first layer is 256.
The number of neurons in the three hidden layers are 500, 250, and 120, respectively.
The active function of the hidden layers are rectified linear unit (ReLU) function
and that of the last payer is Sigmoid function. Every 16 bits of the transmitted data
are detected for one model which indicates the dimension of the output layer is 16.
For example, the model in Figure 4.8 aims to predict the 16th–31st data bits in the
second block. Since the data block contains 128 binary bits, eight DNN models are
needed to recover the whole transmitted data part as indicated in Figure 4.9.
144 Applications of machine learning in wireless communications

Hidden Hidden Hidden Label

Input layer 1 layer 2 layer 3 Output (transmitted
Received (256) (500) (250) (120) (16) bits)
signal

16–31bit
Pilot

Data:0–127 bit
...

...
...
...
...
Data

...

w4, b4
w3, b3
w2, b2
w1, b1

Figure 4.8 Architecture of one DNN model

Recovered data signal Data:0/1 sequence

Recovered bit 0–15 bit 16–31 bit 32–47 bit 48–63 bit 64–79 bit 80–95 bit 96–111 bit 112–127 bit

Output(16)

Hidden layer(...)

Input
(256) Pilot Data Pilot Data Pilot Data Pilot Data Pilot Data Pilot Data Pilot Data Pilot Data

Received signal
Pilot Data

Figure 4.9 Eight DNN models

The objective function for optimization is L2 loss function and optimal parameters
is obtained with the root mean square prop (RMSProp)1 optimizer algorithm, where
Python2 environment and TensorFlow3 architecture are utilized. Table 4.2 lists some
key parameters for training the DNN.
Figure 4.10 illustrates the bit error rate performance of the DNN method and
traditional estimators: LS and LMMSE. It can be seen that the LS method performs
worst and that the DNN method has the same performance as the LMMSE at low SNR.
When the SNR is over 15 dB, the LMMSE slightly outperforms the DNN method.

1
RMSProp is a stochastic gradient descent method with adapted learning rates.
2
Python is a high-level programming language.
3
TensorFlow is an open-source software library for dataflow programming.
Machine-learning-based channel estimation 145

Table 4.2 Key parameters for training

the DNN

Parameters Value

Epoch 60
Batch size 2,000
Batch 400
Learning rate 0.001
Test set 10,000

100
Deep learning
LMMSE
LS

10–1
BER

10–2

10–3

10–4
5 10 15 20 25
SNR

Figure 4.10 DNN, LMMSE, and LS performance

4.3.3 Deep learning for massive MIMO CSI feedback

Massive multiple-input and multiple-output (MIMO) wireless communication sys-
tems have attracted enormous attentions from both academy and industry. For massive
MIMO with frequency division duplex mode, the user equipment (UE) estimates the
downlink channel information and returns it to the base station (BS) via the feed-
back link. The main challenge of this CSI feedback mechanism is the large overhead.
Existing systems usually utilize compressed sensing (CS)-based methods to obtain the
sparse vector and then restructure the matrix as the estimate. These methods require
that the channel should be sparse in some bases.
Different from the CS method, a feedback mechanism based on DNN is proposed
in [13]. The suggested deep-learning-based CSI network (CsiNet) can sense and
recover the channel matrix. The workflow of the CsiNet is shown in Figure 4.11.
146 Applications of machine learning in wireless communications

BS UE

Downlink

Feedback Channel
Decoder Encoder estimation

CsiNet based on deep learning

Figure 4.11 Deep-learning-based feedback mechanism

DFT Truncation Encoder

~ –
H H H S

Feedback
IDFT Completion Decoder
~ –
H H Ĥ S

Figure 4.12 CSI feedback approach

Suppose the BS has Nt transmit antennas and UE has one, and the OFDM system
c carriers. Denote the estimated downlink channel matrix as H
has N ∈ C Nt ×Nc . Once
it will apply the following DFT and then obtain:
the UE estimates the channel H,
Ha
H̄ = Fd HF (4.21)
where Fd and Fa are N c × Nc and Nt × Nt DFT matrices, respectively. Next the UE
selects the first Nc rows of H since the CSI are mainly included in these rows. Let H
represent the truncated matrix, i.e., H = H̄( :, 1 : Nc ). Clearly, the matrix H contains
N = Nc × Nt elements, which indicates that the number of the feedback parameters
is cut down to N .
Based on the deep-learning method, the CsiNet designs an encoder to convert
H to a vector s that only has M elements. Next the UE sends the codeword s to the
BS. The BS aims to reconstruct the original channel matrix H with the code word s.
The compression ratio is γ = M /N . Then the decoder in CsiNet can recover s to Ĥ.
After completing Ĥ to H̄, IDFT is used to obtain the final channel matrix. In summary,
the CSI feedback approach is shown as Figure 4.12.
The CsiNet is an autoencoder model based on convolution NN. Figure 4.13 shows
the architecture of the CsiNet which mainly consists of an encoder (Figure 4.14) and
a decoder (Figure 4.15).
The detailed structure of the encoder is shown in Figure 4.14. It contains two
layers. The first layer is a convolutional layer and the second is a reshape layer. The
real and imaginary parts of the truncated matrix H with dimensions 8 × 8 is the input
of the convolutional layer. This convolutional layer uses a 3 × 3 kernel to generate two
feature maps, which is the output of the layer, i.e., two matrices with dimensions 8 × 8.
Then the output feature maps are reshaped into a 128 × 1 vector. The vector enters
Machine-learning-based channel estimation 147

CsiNet

Input Encoder Compress Decoder Output

2 model representations model 2
(codeword)
8
8 8
× CNN × CNN ×
8 1 8
s

H Ĥ

Figure 4.13 Architecture of CsiNet

Input Encoder model Compress representation

(codeword)
Reshape
Convolutional layer layer
2 2 Fully connected layer

8 8 128 8
×
× × ×
1
8 8 1 s
H

Figure 4.14 Structure of the encoder

Compress
representation Decoder model Output
(codeword)
Fully RefineNet RefineNet
connected
layer Input Convolutional Convolutional Convolutional Input Convolutional Convolutional Convolutional
layer layer layer layer layer layer layer layer 2
16 2 8 16
(Reshape) 2 8 2 2
8 8
128 8 8 8 8 8 8 8 8
× ×
× × × × × × × × ×
1 8
1 8 8 8 8 8 8 8 8
s
Ĥ

Figure 4.15 Structure of the decoder

into a fully connected layer and the output of the connected layer is the compressed
codeword s with eight elements.
The goal of the decoder is to recover the codeword to the matrix H. The detailed
structure of the decoder is shown in Figure 4.15. The decoder is comprised of three
main parts: a fully connected layer (also referred to as dense layer) and two RefineNet.
The first fully connected layer transforms the codeword s into a 128 × 1 vector. And
148 Applications of machine learning in wireless communications

then the vector is reshaped into two 8 × 8 matrices which is considered as the initial
estimate of H. Next, two RefineNets are designed to refine the estimates.
Each RefineNet has four layers: one input layer and three convolutional layers.
The input layer has two feature maps, and the three convolutional layers have eight,
sixteen, and two feature maps, respectively. All the feature maps are of the same size
as the input channel matrix size 8 × 8.
It is worth noting that in each RefineNet, there is a direct data flow from the input
layer to the end of last convolutional layer so as to avoid the gradient disappearance.
Each layer of the CsiNet executes normalization and employs a ReLU function
to activate the neurons. After two RefineNet units, the refined channel estimates will
be delivered to the final convolutional layer and the sigmoid function is also exploited
to activate the neurons.
The end of the second RefineNet in the decoder will output two matrices with
size 8 × 8, i.e., the real and imaginary parts of Ĥ, which is the recovery of H at
the BS.
MSE is chosen as the loss function for optimization, and the optimal parameters
are obtained through ADAM algorithm. Simulation experiments are carried out in
Python environment with TensorFlow and Keras4 architecture. The key parameters
needed to train the network are listed in Table 4.3.
Here, we provide one example of H. The UE obtain channel estimate and trans-
form it into H. The real and image parts of H are as shown in Tables 4.4 and 4.5,
respectively.
The UE inputs H to CsiNet. The encoder of the CsiNet then generates a 8 × 1
codeword:
s = [−0.17767, −0.035453, −0.094305, −0.072261,
−0.34441, −0.34731, 0.14061, 0.089002] (4.22)

Table 4.3 The parameters for

training CsiNet

Parameters Value

Training set 100,000

Validation set 3,000
Test set 2,000
Epoch 1,000
Batch size 200
Batch 50
Learning rate 0.01

4
Keras is an open-source neural network library which contains numerous implementations of commonly
used neural network building blocks.
Machine-learning-based channel estimation 149

Table 4.4 Real parts of the channel matrix H

h(0) h(1) h(2) h(3) h(4) h(5) h(6) h(7)

0.49238 0.48270 0.57059 0.48917 0.50353 0.50847 0.46970 0.46497

0.49181 0.47943 0.53719 0.51973 0.50382 0.51015 0.47846 0.50859
0.49114 0.47463 0.52339 0.51216 0.50416 0.51259 0.48357 0.50004
0.49035 0.46690 0.51880 0.50956 0.50456 0.51650 0.48655 0.49873
0.48941 0.45239 0.51662 0.50631 0.50503 0.52370 0.48838 0.49633
0.48828 0.41570 0.51161 0.50951 0.50561 0.54108 0.48948 0.49984
0.48687 0.16565 0.51290 0.50482 0.50634 0.62954 0.48995 0.49621
0.48507 0.81859 0.50914 0.50794 0.50726 0.33268 0.49038 0.49959

Table 4.5 Image parts of the channel matrix H

h(0) h(1) h(2) h(3) h(4) h(5) h(6) h(7)

0.50008 0.50085 0.49364 0.44990 0.49925 0.49954 0.48900 0.34167

0.50012 0.50115 0.49413 0.51919 0.49924 0.49975 0.49007 0.57858
0.50016 0.50161 0.49481 0.50429 0.49924 0.50010 0.49050 0.55735
0.50022 0.50237 0.49508 0.50391 0.49924 0.50074 0.49014 0.50956
0.50029 0.50383 0.49503 0.50028 0.49926 0.50203 0.48907 0.50658
0.50037 0.50749 0.49465 0.50244 0.49929 0.50539 0.48700 0.50755
0.50049 0.52458 0.49375 0.49802 0.49933 0.52117 0.48301 0.50263
0.50064 0.44605 0.49326 0.49966 0.49941 0.46087 0.48067 0.50367

The decoder can utilize this codeword s to reconstruct the channel matrix Ĥ. Define the
distance between H and Ĥ as d = H − Ĥ22 . In this case, we obtain d = 3.98 × 10−4 .
The compression ratio is γ = ((8 × 8 × 2)/8) = 1/16.

4.4 EM-based channel estimator

4.4.1 Basic principles of EM algorithm

EM algorithm [7] is an iterative method to obtain maximum likelihood estimates
of parameters in statistical models which depend on unobserved latent or hidden
variables. Each iteration of the EM algorithm consists of two steps: calculating the
expectation (E step) and performing the maximization (M step).
Next, we will introduce the principles of the EM algorithm in detail. And for
clarity and ease of understanding EM algorithm, the notations used throughout this
section are illustrated in Table 4.6.
In probabilistic models, there may exist observable variables and latent variables.
For example, in the wireless communication systems, the received signals can be
given as
y(i) = hx(i) + w(i), i = 1, 2, . . . , N (4.23)
150 Applications of machine learning in wireless communications

Table 4.6 Parameters description

Notations Description Corresponding item

in the model (4.23)

y Observed variable [y(1), y(2), . . . , y(N )]T

z Latent variable (hidden variable) [x(1), x(2), . . . , x(N )]T
θ Parameter to be estimated h
θ ( j) The jth iterative estimate of parameter θ h( j)
L(θ) The log-likelihood function about parameter θ L(h)
N
P(y; θ ) The probability with the parameter θ i=1 P(y(i); h)
P(y, z; θ) The joint probability for variables y, z with the P(y(i), xk ; h)
parameter θ
P(z|y; θ ( j) ) The conditional probability given y and P(xk |y(i); h( j) )
parameter
LB(θ , θ ( j) ) The lower bound of log likelihood function LB(h, h( j) )
Q(θ , θ ( j) ) The expected value of the log likelihood Q(h, h( j) )
function of θ given the current estimates
of the parameter θ
Q(θ ( j) , θ ( j) ) The value of the log likelihood function Q(h( j) , h( j) )
when θ is equal to θ ( j)

where h is the flat-fading channel to be estimated, and x(i) is the unknown modulated
BPSK signal, i.e., x(i) ∈ {+1, −1}. In this statistical model (4.23) that aims to esti-
mate h with unknown BPSK signals x(i), the received signals y(i) are the observable
variables and the transmitted signals x(i) can be considered as latent variables.
Denote y, z, and θ as the observed data, the latent variable, and the parameter to be
estimated, respectively. For the model (4.23), we can have y = [y(1), y(2), . . . , y(N )]T ,
z = [x(1), x(2), . . . , x(N )]T , and θ = h.
If the variable z can be available, the parameter θ can be estimated by maximum
likelihood approach or Bayesian estimation. Maximum likelihood estimator solves
the log-likelihood function:
L(θ) = lnP(y; θ ) (4.24)
where z is a parameter in the probability density function P(y; θ ). Clearly, there is
only one unknown parameter θ in the log-likelihood function L(θ|y), and therefore
the estimate of θ can be obtained through maximizing5 L(θ):
θ̂ = max L(θ ). (4.25)
θ

However, if the variable z are unknown, we cannot find the estimate θ̂ from (4.25)
since the expression of L(θ ) contains unknown parameter z. To address this problem,
EM algorithm was proposed in [7] in 1977. EM algorithm estimates the parameter θ

5
One dimensional search or setting the derivative as zero can obtain the optimal value of θ .
Machine-learning-based channel estimation 151

iteratively. Denote the jth estimate of θ as θ ( j) . The basic principle of EM algorithm

is as follows.
Noting that the relationship between the marginal probability density function
P(y) and the joint density function P(y, z) is

P(y) = P(y, z). (4.26)
z

Hence, we can rewrite the likelihood function L(θ ) as

L(θ) = lnP(y; θ ) = ln P(y, z; θ ) (4.27)
z

= ln P(z; θ )P(y|z; θ) , (4.28)
z

where the Bayesian equation P(y, z) = P(z)P(y|z) is utilized in the last step in (4.28).
Equation (4.28) is often intractable since it contains not only the logarithm oper-
ation of the summation of multiple items and also the unknown parameter z in the
function P(y|z; θ).
To address this problem of the unknown parameter z, EM algorithm rewrites the
likelihood function L(θ ) as

( j) P(z; θ )P(y|z; θ)
L(θ) = ln P(z|y; θ ) (4.29)
z
P(z|y; θ ( j) )

where P(z|y; θ ( j) ) is the probability distribution function of the latent variable z

given y. Since the distribution P(z|y; θ ( j) ) can be readily obtained, it is possible
to generate a likelihood function that only contains one unknown parameter θ.
Using Jensen’s inequality [23]:
f (E[x]) ≥ E[ f (x)], (4.30)
where x is a random variable, f (x) is a concave function, and E[x] is the expected
value of x, we can have
ln(E[x]) ≥ E[ ln(x)]. (4.31)
Thus, we can find

P(z; θ )P(y|z; θ)
L(θ ) ≥ E ln
P(z|y; θ ( j) )
P(z; θ )P(y|z; θ )
= P(z|y; θ ( j) ) ln ( j) )
= LB(θ , θ ( j) ), (4.32)
z
P(z|y; θ

where LB(θ, θ ( j) ) is defined as the lower bound of the likelihood function L(θ ).
We can further simplify LB(θ, θ ( j) ) as
P(y, z; θ )
LB(θ, θ ( j) ) = P(z|y; θ ( j) ) ln . (4.33)
z
P(z|y; θ ( j) )
152 Applications of machine learning in wireless communications

It is worth noting that there is only one unknown parameter θ in the above
expression (4.33) of LB(θ , θ ( j) ). Therefore, we can find the ( j + 1)th iterative estimate
θ ( j+1) through:
θ ( j+1) = arg max LB(θ, θ ( j) ) (4.34)
θ

= arg max P(z|y; θ ( j) )(ln P(y, z; θ) − ln P(z|y; θ ( j) )) (4.35)
θ
z

which is the focus of the M step.

Since the item P(z|y; θ ( j) ) does not contain the variable θ , it can be negligible in
the maximization optimization operation (4.35). Therefore, we can further simplify
(4.35) as

θ ( j+1) = arg max P(z|y; θ ( j) ) ln P(y, z; θ ) (4.36)
θ
z

Q(θ ,θ ( j) )

where the function Q(θ , θ ( j) ) is defined as the corresponding item.

Interestingly, we can find the function Q(θ , θ ( j) ) can be written as

Q(θ , θ ( j) ) = P(z|y; θ ( j) ) ln P(y, z; θ) (4.37)
z

= Ez|y;θ ( j) [ ln P(y, z; θ)], (4.38)

which indicates that the function Q(θ , θ ( j) ) is the expected value of the log likelihood
function ln P(y, z; θ ) with respect to the current conditional distribution P(z|y; θ ( j) )
given the observed data y and the current estimate θ ( j) . This is the reason for the name
of the E step.
Next the M step is to find the parameter θ which can maximize the function
Q(θ, θ ( j) ) found on the E step, and we set the optimal value of θ as θ ( j+1) . That is,
θ ( j+1) = arg max Q(θ , θ ( j) ). (4.39)
θ

Till now, the jth iteration ends. The estimate θ ( j+1) is then used in the next round
of iteration (the E step and the M step).
The termination condition of the iterative process is
θ ( j+1) − θ ( j) < ε, (4.40)
or
Q(θ ( j+1) , θ ( j) ) − Q(θ ( j) , θ ( j) ) < ε, (4.41)
where ε is a predefined positive constant.

4.4.2 An example of channel estimation with EM algorithm

In this section, we will use the EM method to estimate the channel h in the signal trans-
mission model (4.23) without training symbols which indicates that the BPSK signals
Machine-learning-based channel estimation 153

x(i) are unknown to the receiver. We assume that the BPSK signals are equiprobable,
i.e., P(x(i) = +1) = P(x(i) = −1) = 1/2, i = 1, 2, . . . , N .
Suppose x1 = +1 and x2 = −1, and clearly the BPSK signals x(i) can be either
x1 or x2 . The conditional probability density function of received signal y(i) given
x(i) = xk , k = 1, 2 can be expressed as

1 (y(i) − hxk )2
P(y(i)|xk ; h) = √ exp , (4.42)
2πσw −2σw2

And the joint probability density function of xk and y(i) is

P(y(i), xk ; h) = P(y(i)|xk ; h)P(xk )

1 (y(i) − hxk )2
= √ exp . (4.43)
2 2πσw −2σw2

The core of the EM algorithm is to iteratively calculate the function Q(h, h( j) )

where h is the channel to be estimated and h( j) denoted the jth estimate of h. Accord-
ing to (4.38), we need the joint distribution P(y(i), xk ; h) and also the conditional
distribution P(xk |y(i); h( j) ) to derive the function Q(h, h( j) ). Since the joint distribu-
tion P(y(i), xk ; h) is given in (4.43), our next focus is to calculate the conditional
probability P(xk |y(i); h( j) ).
The conditional probability P(xk |y(i); h( j) ) can be derived as

P(xk , y(i); h( j) )
P(xk |y(i); h( j) ) =
P(y(i); h( j) )
P(y(i)|xk ; h( j) )P(xk )
= 2
P(y(i)|xm ; h( j) )P(xm )
m=1
√
(1/(2 2πσw )) exp (y(i) − h( j) xk )2 /−2σw2
= 2 √
m=1 (1/(2 2πσw )) exp (y(i) − h xm ) /−2σw
( j) 2 2

exp (y(i) − h( j) xk )2 /−2σw2
= 2 (4.44)
m=1 exp (y(i) − h xm ) /−2σw
( j) 2 2

Subsequently, in the E step, the expectation of ln P(y(i), xk ; h) with respect to the

current conditional distribution P(xk |y(i); h( j) ) given y(i) can be found as

N
2
Q(h, h ) =
( j)
P(xk |y(i); h( j) ) ln P(y(i), xk ; h) (4.45)
i=1 k=1
154 Applications of machine learning in wireless communications

Substituting (4.43) and ( 4.44) into (4.45), we can further obtain

N
2
√ (y(i) − hxk )2
Q(h, h( j) ) = P(xk |y(i); h( j) ) −ln 2 2πσw −
i=1 k=1
−2σw2

N 2
exp (y(i) − h( j) xk )2 /−2σw2 √ (y(i) − hxk )2
= 2 −ln 2 2π σ w −
i=1 k=1 m=1 exp (y(i) − h xm ) /−2σw
( j) 2 2 −2σw2
(4.46)
( j)
It is worth noting that the expression (4.46) of Q(h, h ) only contains one
unknown parameter h. Therefore, the ( j + 1)th estimate of the channel h can be cal-
culated through setting the derivative of (4.46) with respect to h as zero. Accordingly,
it can be readily obtained that
N 2
k=1 P(xk |y(i); h )y(i)xk
( j)
h( j+1) = i=1
N 2
k=1 P(xk |y(i); h )xk
( j) 2
i=1
N 2
2
i=1 exp (y(i) − h( j) xk )2 /−2σw2 y(i)xk /
k=1 m=1 exp (y(i) − h xm ) /−2σw
( j) 2 2
= N 2 2
i=1 k=1 exp (y(i) − h( j) xk )2 /−2σw2 xk2 / m=1 exp (y(i) − h xm ) /−2σw
( j) 2 2

(4.47)
In conclusion, the EM algorithm preset a value for θ and then calculates h( j+1)
iteratively according to (4.47) until the convergence condition (4.40) or (4.41) is
satisfied.
In the following part, we provide simulation results to corroborate the proposed
EM-based channel estimator. Both real and complex Gaussian channels are simu-
lated. For comparison, Cramér–Rao Lower Bound (CRLB) is also derived. CRLB
determines a lower bound for the variance of any unbiased estimator. First, since N
observations are used in the estimation, the probability density function of y is
N
1 (y(i) − hxk )2
P(y; h) = √ exp , (4.48)
i 2πσw −2σw2
and its logarithm likelihood function is
1
N
N
ln P(y; h) = − ln 2πσw2 − (y(i) − hx(i))2 . (4.49)
2 2σw2 i
The first derivative of ln P(y; h) with respect to h can be derived as

1
N
∂ln P(y; h)
= 2 (y(i) − hx(i))x(i). (4.50)
∂h σw i
Thus, the CRLB can be expressed as [17]
1 σ2 σw2
var(h) ≥ = wN = = CRLB (4.51)
−E (∂ 2 ln p(y; h))/∂ 2 h E[ i xi2 ] NPx
where Px is the average transmission power of the signals x(n).
Machine-learning-based channel estimation 155

Figures 4.16 and 4.17 depict the MSEs of the EM estimator versus SNR for
the following two cases separately: when the channel h is generated from N (0, 1),
i.e., real channel; and when it is generated from CN (0, 1), i.e., Rayleigh channel.
The observation length is set as N = 6. For comparison with the EM estimator, the
MSE curves of LS method are also plotted when the length of pilot is N /2 and N ,

100
EM
LS(N/2)
LS(N)
CRLB

10–1
MSE

10–2

10–3
0 2 4 6 8 10 12 14 16 18 20
SNR

Figure 4.16 Real channel estimation MSEs versus SNR

100
EM
LS(N/2)
LS(N)
CRLB

10–1
MSE

10–2

10–3
0 2 4 6 8 10 12 14 16 18 20
SNR

Figure 4.17 Complex channel estimation MSEs versus SNR

156 Applications of machine learning in wireless communications

100
EM-SNR = 3 dB
LS(N)-SNR = 3 dB
EM-SNR = 20 dB
LS(N)-SNR = 20 dB
10–1
MSE

10–2

10–3

10–4
0 2 4 6 8 10 12
N

Figure 4.18 Channel estimation MSEs versus N

respectively. The CRLBs are also illustrated as benchmarks. It can be seen from
Figures 4.16 and 4.17 that the EM-based blind channel estimator performs well and
approaches CRLB at high SNR. It can also be found that the EM-based blind channel
estimator with no pilots exhibits almost the same performance with the LS estimator
with N pilots and fully outperforms the LS estimator with N /2 pilots.
Figure 4.18 demonstrates the MSEs of LS method and EM algorithm versus the
length of signal N when SNR = 3 and 20 dB, respectively. As we expected, the MSEs
of the two estimator witness a downward trend when the length N increases. It can
be also found that the EM algorithm without pilot has almost consistent performance
with the LS method with N pilots when SNR is 20 dB.

4.5 Conclusion and open problems

In this chapter, we introduce two traditional, one EM-based, and two deep-learning-
based channel estimators. Exploiting ML algorithms to design channel estimators is
a new research area that involves many open problems [24]. For example, devel-
oping new estimators in the relay or cooperative scenarios is worthy of further
studies. Besides, design of new deep-learning-based channel estimators for wire-
less communications on high-speed railways is another interesting challenge since
the railway tracks are fixed and a large amount of historical data along the railways
can be exploited [25] through various deep-learning approaches to enhance estimation
performance.
Machine-learning-based channel estimation 157

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Chapter 5
Signal identification in cognitive radios
using machine learning
Jingwen Zhang1 and Fanggang Wang1

As an intelligent radio, cognitive radio (CR) allows the CR users to access and share
the licensed spectrum. Being a typical noncooperative system, the applications of
signal identification in CRs have emerged. This chapter introduces several signal
identification techniques, which are implemented based on the machine-learning
theory.
The background of signal identification techniques in CRs and the motivation
of using machine learning to solve signal identification problems are introduced
in Section 5.1. A typical signal-identification system contains two parts, namely, the
modulation classifier and specific emitter identifier, which are respectively discussed
in Sections 5.2 and 5.3. Conclusions are drawn in Section 5.3.5.

5.1 Signal identification in cognitive radios

CR was first proposed by Joseph Mitola III in 1999 [1], with its original definition
as a software-defined radio platform, which can be fully configured and dynamically
adapt the communication parameters to make the best use of the wireless channels.
In 2005, Simon Haykin further developed the concept of CR to spectrum sharing [2],
where the CR users are allowed to share the spectrum with the licensed users and hence
mitigates the scarcity problem of limited spectrum. With the advent of the upcoming
fifth generation (5G) cellular-communication systems, the challenges faced by 5G are
neglectable. In particular, the explosion of mobile data traffic, user demand, and new
applications contradicts with the limited licensed spectrum. The existing cellular net-
work is built based on the legacy command-and-control regulation, which in large part
limits the ability of potential users to access the spectrum. In such a case, CR provides
a promising solution to address the above bottleneck faced by 5G cellular system.
In general, CR network is a noncooperative system. Within the framework, the
CR users and the licensed users work in separate and independent networks. For
a CR user, it has little a prior knowledge of the parameters used by the licensed

1
State Key Laboratory of Rail Traffic Control and Safety, Beijing Jiaotong University, China
160 Applications of machine learning in wireless communications

user and other CR users and barely knows the identities (whether they are legal or
evil) of other users in the network. Hence, signal identification plays a key role in
CRs in order to successfully proceed the received signals and to guarantee the safety
and fairness of the networks. In this chapter, two signal identification techniques
are introduced, modulation classification and specific emitter identification (SEI).
Figure 5.1 illustrates the diagram of a typical signal-identification system. The signal
identification system is capable of solving two typical problems that the CR networks
are confronted with: one is that the CR users have little information of the parameters
used by the licensed users and/or other CR users; the other is that with the ability
that allows any unlicensed users to access the spectrum, a certain user lacks the
identity information of other users in the network. Modulation classification can be
adopted to solve the unknown parameter problem and has vital applications in CRs.
For the case when the licensed and cognitive users share the same frequency band
for transmission and reception, the received signal at the cognitive receiver is the
superposition of signals from the licensed transmitter and its own transmitter, which
implies that the signal from the licensed user can be treated as an interference with
higher transmission power. By applying the modulation classification techniques, the
CR receiver can blindly recognize the modulation format adopted by the licensed
signal and is capable of demodulating, reconstructing, and canceling the interference
caused by the licensed user, which is the basis to proceed its own signal. Furthermore,
to solve the problem that the CRs is exposed to high possibility of being attacked
or harmed by the evil users, SEI offers a way to determine the user’s identity and
guarantees the safety and fairness of the CR networks.
The task of signal identification is to blindly learn from the signal and the envi-
ronment to make the classification decision, behind which is the idea of the artificial
intelligence. As an approach to implement the artificial intelligence, machine learn-
ing has been introduced in signal identification for the designing of the identification
algorithms. With little knowledge of the transmitted signal and the transmission

Signal reception

Unknown
parameter
Modulation
Licensed Signal processing
classification
and
Cognitive cognitive
radios users
spectrum
sharing Specific emitter identification
Unknown
identity

Identity recognition

Figure 5.1 The diagram of a typical signal identification system

Signal identification in cognitive radios using machine learning 161

environments, blind signal identification remains a challenging task for conventional

methods. Machine learning provides a novel and promising solution for the signal-
identification problem. In general, a signal identifier/classifier consists of two parts,
one is the feature extraction subsystems to extract distinguishable features and the
other one is the classifier, which implements the classification task and makes deci-
sions by training and learning from the extracted features. In the remaining sections,
details of two signal identification techniques, i.e., the modulation classification and
SEI algorithms, based on machine learning is described.

5.2 Modulation classification via machine learning

The automatic modulation classification, hereinafter referred to as modulation clas-
sification, is a technique adopted at an intelligent receiver to automatically determine
the unknown modulation format used by a detected signal of interest. As an indis-
pensable process between signal detection and demodulation, it finds its applications
in military communications, along with CRs and adaptive systems.
Depending on the theories that used, typical modulation classification techniques
can be categorized into two classes, i.e., the decision-theoretic and the pattern-
recognition algorithms. The decision-theoretic algorithms are based on the likelihood
theory, where the modulation-classification problem is formulated as a multiple-
hypothesis test [3,4]. On the other hand, the pattern-recognition algorithms are based
on the pattern-recognition and machine-learning theory, where certain classification
features are extracted from the received signal and then inputted into the classifiers
to decide the modulation format [5,6].
More recently, the modulation classification investigation has focused on a more
challenging task of implementing the classification problem in the realistic environ-
ment, e.g., in the presence of complicated channel conditions and with no knowledge
of many transmission parameters. The expectation–maximization (EM) algorithm is
a commonly known algorithm in machine learning, which can be widely adopted to
clustering and dimension reduction. For a probabilistic model with unobserved latent
variables, the EM algorithm provides a feasible way to obtain the maximum likelihood
estimates (MLEs) of the unknown parameters.
In this section, we introduce two modulation classifiers, one is for determining
the typical constellation-based modulation formats (e.g., quadrature amplitude modu-
lation (QAM) and phase-shift keying (PSK)), which are widely adopted in the mobile
communication systems long term evolution (LTE) and new radio (NR), and the other
is for classifying continuous phase modulation (CPM) types, which have critical appli-
cations in satellite-communication systems. Both of the two classification problems
are considered in unknown fading channels, where the EM-based algorithms are pro-
posed to obtain the MLEs of the unknown channel parameters and further determine
the modulation formats. Note that in Section 5.2.2, the exact algorithm used in the
classification problem is the Baum–Welch (BW) algorithm. However, it is clarified
that the idea behind the BW algorithm is using the EM algorithm to find the MLEs
of the unknowns in a hidden Markov model (HMM).
162 Applications of machine learning in wireless communications

5.2.1 Modulation classification in multipath fading channels via

expectation–maximization
In this section, we investigate the classification problem of classifying QAM/PSK
modulations in the presence of unknown multipath channels. QAM/PSK is the most
commonly used modulation formats in the existing LTE systems and the upcoming
NR systems, of which the classification problem has been thoroughly investigated
for decades. However, the classification problem in real-world scenarios, such as in
the unknown multipath fading channels, is still challenging and needs further study.
A hybrid maximum likelihood (also known as hybrid likelihood ratio test) based
classifier is proposed to solve this problem. Specifically, the likelihood function is
computed by averaging over the unknown transmitted constellation points and then
maximizing over the unknown channel coefficients. Solutions to this problem cannot
be obtained in a computationally efficient way, where the EM algorithm is developed
to compute the MLEs of the unknowns tractably.

5.2.1.1 Problem statement

We consider a typical centralized cooperation system with one transmitter, K
receivers, and one fusion center; the fusion center collects data from the K receivers
to enhance the classification performance.1 The transmit signal undergoes a multi-
path channel with L resolvable paths. Then, the received signal at the kth receiver is
written as

L−1
yk (t) = ak (l)ejφk (l) xs (n)g(t − nT − lT ) + wk (t), 0 ≤ t ≤ T0 (5.1)
n l=0

where T and T0 are the symbol and observation √ intervals, respectively, with T0 T ,
g(·) is the real-valued pulse shape, j = −1, ak (l) > 0 and φk (l) ∈ [0, 2π ) are the
unknown amplitude and phase of the lth path at the kth receiver, respectively, wk (t)
is the complex-valued zero-mean white Gaussian noise process with noise power σk2 ,
and xs (n) is the nth transmit constellation symbol drawn from an unknown modulation
format s. The modulation format s belongs to a modulation candidate set {1, . . . , S},
which is known at the receivers. The task of the modulation classification problem
is to determine the correct modulation format to which the transmit signal belongs
based on the received signal.
The maximum likelihood algorithm is adopted as the classifier, which is optimal
when each modulation candidate is equally probable. Let Hs denote the hypothesis
that transmit symbols are drawn from the modulation format s, the likelihood function
under the hypothesis Hs is given by

ps (y|θ ) = ps (y|xs , θ ) p (xs |θ) (5.2)
xs

1
Multiple receivers are considered to obtain the diversity gain, while the proposed algorithm is applicable
to the case with only one receiver.
Signal identification in cognitive radios using machine learning 163

where yT = [yT1 , . . . , yTK ], with yTk as the vector representation of yk (t), θ T =

[θ T1 , . . . , θ TK ], with θ k = [ak (0), φk (0), . . . , ak (L − 1), φk (L − 1)]T as the vector of
unknown parameters, and xs = [xs (1), . . . , xs (N )]T is the vector representation of the
transmit symbol, with N = (T0 /T ), and (·)T as the transpose of a vector/matrix. We
assume that the multiple receivers are spatially divided; therefore, the received signals
at multiple receivers are independent. Then, ps (y|xs , θ ) is expressed as

K

ps (y|xs , θ ) = ps yk |xs , θ k
k=1
⎧ ⎫
⎨ K 0
|yk (t) − fk (xs (n), t)|2 dt ⎬
T

∝ exp − (5.3)
⎩ σk2 ⎭
k=1 0

where

N
L−1
fk (xs (n), t) = ak (l)ejφk (l) xs (n)g(t − nT − lT ).
n=1 l=0

Define xs,i (n) as the nth transmit symbol that maps to the ith constellation point
under the hypothesis Hs and assume that each constellation symbol has equal prior
probability, i.e., p(xs,i (n)) = (1/M ), with M as the modulation order of modulation
format s. The log-likelihood function Ls (θ ) is then obtained by
⎛ ⎧ ⎫⎞
⎨ K 0 2 ⎬
T
M
1 |yk (t) − fk (xs,i (n), t)| dt ⎠
Ls (θ) = ln ps (y|θ ) = ln⎝ exp − . (5.4)
M ⎩ σk2 ⎭
i=1 k=1 0

The classifier makes the final decision on the modulation format by

ŝ = arg max Ls (θ ()
s ) (5.5)
s

s is the MLE of the unknown parameters under the hypothesis Hs , which

where θ ()
can be obtained by

s = arg max Ls (θ ).
θ () (5.6)
θ

It should be noted that (5.26) is a high-dimensional non-convex optimization problem

with no closed-form solutions. Essentially, the computation of the MLEs suffers from
high-computational complexity, which is impractical in applications of modulation
classification.
5.2.1.2 Modulation classification via EM
In this section, an EM-based algorithm is proposed to solve the problem in (5.26) in
a tractable way. The expectation step (E-step) and maximization step (M-step) under
the hypothesis Hs can be mathematically formulated as [7]:

s ) = Ez|y,θ (r) [ln p(z|θ )]

E-step: J (θ |θ (r) (5.7)
s

M-step: θ (r+1)
s = arg max J (θ |θ (r)
s ) (5.8)
θ
164 Applications of machine learning in wireless communications

where z is the complete data, which cannot be directly observed at the receivers.
Instead, the complete data is related to the observations, i.e., the received signals,
by y = K(z), where K(·) is a deterministic and non-invertible transformation. The
non-invertible property of K(·) implies that there exits more than one possible defini-
tions of the complete data to generate the same observations. It should be noted that
these choices have great impact on the complexity and convergence result of the EM
algorithm, bad choices of the complete data make the algorithm invalid.
In our problem, the received signal that undergoes multipath channels is equiv-
alent to a superposition of signals from different independent paths; therefore, the
complete data can be defined as

zkl (t) = ak (l)ejφk (l) xs (n)g(t − nT − lT ) + wkl (t) (5.9)
n

where wkl (t) is the lth noise component, which is obtained by arbitrarily decompos-
ing the
total noise wk (t) into L independent and identically distributed components,
i.e., L−1
l=0 wkl (t) = wk (t). Assume that wkl (t) follows the complex-valued zero-mean
Gaussian process with power σkl2 . The noise power σkl2 is defined as σkl2 = βkl σk2 ,
where βkl is a positive real-valued random noise decomposition factor following
L−1
l=0 βkl = 1 [8]. Hence, we can rewrite the transmission model in (5.1) as

yk (t) = K(zk ) = 1L zTk

where 1L is a L × 1 vector with all entries equal to 1, and zk is the vector representation
of zkl (t). Since the multiple receivers are assumed to be independent, we reduce the
E-step in (5.27) to

K
(r)
s ) =
J (θ|θ (r) Jk (θ |θ s,k )
k=1

K
= Ez (r) [ln p(zk |θ k )] (5.10)
k |yk ,θ s,k
k=1

which indicates that J (θ |θ (r)

s ) can be computed locally at each receiver.
In order to derive the E-step and M-step, the posterior expectations of the
(r)
unknown transmit symbols should be estimated first. Define ρs,i (n) as the poste-
rior probability of the nth transmit symbol mapping to the ith constellation point
under the hypothesis Hs at iteration r, which is given by
(r)
ρs,i (n)

= ps xs (n) = Xs,i |y, θ (r)
s

ps y|xs (n) = Xs,i , θ (r)
s p(xs (n) = Xs,i |θ (r)s )
= M
j=1 ps y|xs (n) = Xs,j , θ s p(xs (n) = Xs,j |θ (r)
(r)
s )
2
K L−1 (r) (r)

exp − k=1 yk (n) − l=0 as,k (l)e xs (n − l) /σk
jφs,k (l) (r) 2

= 2 (5.11)
M K
L−1 (r)
as,k (l)e s,k xs (n − l) /σk
(r) (r)
j=1 exp − k=1 yk (n) −
jφ (l) 2
l=0
Signal identification in cognitive radios using machine learning 165

where the last equality is computed with the assumption that each symbol has equal
prior probability, i.e., p xs (n) = Xs,i |θ (r)
s = (1/M ), yk (n) is the nth received sym-
bol at discrete time nT , Xs,i is the ith constellation point for modulation format s.
(r)
By obtaining the posterior probability ρs,i (n), we can compute xs(r) (n) as

M
(r)
xs(r) (n) = ρs,i (n)Xs,i . (5.12)
i=1

We define z̄kl (t) = n ak (l)ejφk (l) xs (n)g(t − nT − lT ). By computing (5.12), xs (n)
turns to a deterministic symbol. Furthermore, since flat fading is assumed for each
path, the channel amplitude ak (l) and phase φk (l) are treated as unknown deterministic
parameters. Thus, we have that z̄kl (t) is an unknown deterministic signal. Note that
wkl (t) is a zero-mean white Gaussian noise process, ln p(zk |θ k ) is then given by [9]:
L−1
T0
1
ln p(zk |θ k ) = C1 − |zkl (t) − z̄kl (t)|2 dt (5.13)
l=0 0
σkl2

where C1 is a term irrelevant to the unknowns. Taking the conditional expectation of

(r) (r)
(5.13) given yk and θ s,k , Jk (θ |θ s,k ) is obtained by [10]:
L−1
1 (r) 2
T0
(r)
(r)
Jk (θ|θ s,k ) = C2 − 2 ẑs,kl
(t) − z̄s,kl (t) dt (5.14)
l=0 0
σkl

where C2 is a term independent of the unknowns, and

(r) (r)

L−1
(r)
ẑs,kl (t) = z̄s,kl (t) + βkl yk (t) − z̄s,kl (t) . (5.15)
l=0
(r)
It is noted from (5.14) that the maximization of Jk (θ|θ s,k ) with respect to θ is equivalent
to the minimization of each of the L summations. Hence, the E-step and M-step in
(5.10) and (5.28) are respectively simplified as
E-step: for l = 0, . . . , L − 1 compute

(r) (r)

L−1
(r)
ẑs,kl (t) = z̄s,kl (t) + βkl yk (t) − z̄s,kl (t) (5.16)
l=0

M-step: for l = 0, . . . , L − 1 compute:

2
T0 ẑ (r) (t) − z̄ (r) (t) dt
(r+1) s,kl s,kl
θ s,k (l) = arg min . (5.17)
θ k (l) σkl2
0

By taking the derivative of (5.17) with respect to as,k (l) and setting it to zero, we can
obtain that
⎧ ⎫
N ⎨ T0 ⎬
1 (r)
xs(r) (n)∗ e−jφs,k (l) ẑs,kl (t)g ∗ (t − nT − lT )dt
(r+1) (r)
as,k (l) = (r) (5.18)
E n=1 ⎩ ⎭
0
166 Applications of machine learning in wireless communications
∞
where E (r) = Eg Nn=1 |xs(r) (n)|2 , with Eg = −∞ g 2 (t)dt as the pulse energy, {·} rep-
resents the real component of a complex variable, and (·)∗ denotes the conjugation
of a variable. Apparently, the second derivative of (5.17) with respect to as,k (l) is a
negative definite matrix, which implies that (5.18) is the optimal estimate of as,k (l).
By substituting (5.18) into (5.17), with the assumption that E (r) is independent of
(r)
φs,k (l), the M-step in (5.17) is rewritten as
M-step: for l = 0, . . . , L − 1 compute
H
(r) (r)

xs,l ẑ s,kl
−1
(r+1)
φs,k (l) = tan H (5.19)
(r) (r)
xs,l ẑ s,kl

1 (r) ∗ −jφs,k
N
(r+1)
(r+1)
as,k (l) = (r)
xs (n) e (l)
E n=1
⎫
T0 ⎬
ẑs,kl (t)g ∗ (t − nT − lT )dt
(r)
× (5.20)
⎭
0

(r) !T
where xs,l = 0Tl , xs(r) (1), . . . , xs(r) (N − l) , with 0l as a l × 1 vector with all elements
" #T
(r) (r) (r)
equal zero, ẑ s,kl = ẑs,kl (1), . . . , ẑs,kl (N ) ,
{·} represents the imaginary component
of a complex variable, and (·)H denotes the conjugate transpose of a vector/matrix.
It should be noted from (5.16), (5.19), and (5.20) that by employing the EM
algorithm and properly designing the complete data, the multivariate optimization
problem in (5.26) is successfully decomposed into L separate ones, where only one
unknown parameter is optimized at each step, solving the original high-dimensional
and non-convex problem in a tractable way.
Fourth-order moment-based initialization
The most prominent problem for the EM algorithm is how to set proper initialization
points of the unknowns, from which the EM algorithm takes iterative steps to converge
to some stationary points. Since the EM algorithm has no guarantee of the convergence
to the global maxima, poor initializations enhance its probability to converge to the
local maxima. In general, the most commonly adopted initialization schemes for the
EM algorithm include the simulated annealing (SA) [11] and random restart. However,
since our problem considers multipath channels and multiple users, where a (2 × K ×
L)-dimensional initial value should be selected, it is computationally expensive for
the SA and random restart algorithms to find proper initials.
In this section, we employ a simple though effective method to find the initial
values of the unknown fadings. A modified version of the fourth-order moment-based
estimator proposed in [12] is applied to roughly estimate the multipath channels,
which are then used as the initialization points of the EM algorithm. The estimator is
expressed as
y
m4k (p, p, p, l)
ĥk (l) = y (5.21)
m4k (p, p, p, p)
Signal identification in cognitive radios using machine learning 167
y
where m4k (τ1 , τ2 , τ3 , τ4 ) = E{yk (n + τ1 )yk (n + τ2 )yk (n + τ3 )yk (n + τ4 )} is the fourth-
order moment of yk (n), and hk (p) denotes the coefficient of the dominant path of
the channel between the transmitter and kth receiver. Without loss of generality, the
dominant path is assumed to be the leading path, i.e., p = 0.
The overall modulation classification algorithm is summarized as follows:

EM-based modulation classifier

1. Set the stopping threshold and the maximum number of iterations I ;
2. FOR s = 1, . . . , S;
3. Set r = 0;
(0)
4. Initialize the unknown parameters θ s,k (l) using the fourth-order moment-
based estimator;
5. For n = 1, . . . , N , compute xs(r) (n) according to (5.12);
6. Compute the likelihood function Ls (θ (r) s ) according to (5.4);
7. Set r = r + 1;
(r+1) (r+1)
8. Perform over (5.16), (5.19) and (5.20) to estimate φs,k (l) and as,k (l);
(r+1)
9. Compute
the likelihoodfunction L s (θ s ) with the new estimates;
10. If Ls (θ s ) − Ls (θ s ) /Ls (θ s ) > or r ≤ I , go to Step 5; otherwise, set
(r+1) (r) (r)

s = θs
θ (∗) (r+1)
, and continue;
11. ENDFOR
12. Final decision is made by ŝ = arg max Ls (θ (∗) s ).
s

5.2.1.3 Numerical results

In this section, various numerical experiments are provided to examine the clas-
sification performance of the proposed algorithm, using the probability of correct
classification Pc as a performance metric. The number of paths is set to L = 6, where
without loss of generality, the coefficient of the leading path hk (0) = ak (0)ejφk (0) is
set to 1, and the remaining channel coefficients follow zero-mean complex Gaussian
distribution with parameter ςk2 = 0.1. The number of receivers is set to K = 3, and
we assume that the noise power at all receivers is the same. The number of samples
per receiver is set to N = 500. For the EM algorithm, we set the stopping threshold
= 10−3 and the maximum number of iterations I = 100.
Since the convergence of the EM algorithm is highly dependent on the initial-
izations of the unknowns, we first evaluate the impact of the initial values of the
unknowns on the classification performance of the proposed algorithm, which are
formulated as the true values plus bias. Let φk (l) and ak (l) denote the true values,
respectively, and δφk (l) and δak (l) denote the maximum errors for each unknown param-
eter. We randomly take the initial values of the unknown channel phase and amplitude
within [φk (l) − δφk (l) , φk (l) + δφk (l) ] and [0, ak (l) + δak (l) ], respectively [13].
Figure 5.2(a) and (b) shows the classification performance of the proposed
algorithm for QAM and PSK, respectively, with curves parameterized by differ-
ent initialization points of the unknowns. For QAM modulations, the candidate set
is {4, 16, 64}-QAM, and for PSK modulations, it is {QPSK, 8-PSK, 16-PSK}.
168 Applications of machine learning in wireless communications

0.9

0.8
Pc

0.7

0.6
Upper bound
0.5 δφk(l) = π/20, δak(l) = 0.1
δφk(l) = π/10, δak (l) = 0.3
δφk(l) = π/5, δak (l) = 0.5
0.4
0 5 10 15 20
(a)
SNR (dB)
1

0.9

0.8
Pc

0.7

0.6
Upper bound
δφk(l) = π/20, δak(l) = 0.1
0.5
δφ (l) = π/10, δa (l) = 0.3
k k
δφ (l) = π/5, δa (l) = 0.5
k k
0.4
0 5 10 15 20
(b) SNR (dB)

Figure 5.2 (a) Impact of the initial values of unknowns on the proposed algorithm
for QAM and (b) impact of the initial values of unknowns on the
proposed algorithm for PSK

Three sets of maximum errors are examined, namely, (δφk (l) = π/20, δak (l) = 0.1),
(δφk (l) = π/10, δak (l) = 0.3), and (δφk (l) = π/5, δak (l) = 0.5). Moreover, the classifi-
cation performance is compared to the performance upper bound, which is obtained
by using the Cramér–Rao lower bounds of the estimates of the unknowns as the vari-
ances. It is apparent that the classification performance decreases with the increase of
the maximum errors. To be specific, for both QAM and PSK modulations, the clas-
sification performance is not sensitive to the initial values for the first two sets with
smaller biases, especially for the PSK modulation, where the classification perfor-
mance is more robust against smaller initialization errors, while for the third set with
Signal identification in cognitive radios using machine learning 169

larger bias, the classification performance degrades, especially in the low signal to
noise ratio (SNR) region. In our problem, we consider a complicated case with multi-
ple receivers in the presence of multipath channels; therefore, the likelihood function
contains large amounts of local extrema. Then, the EM algorithm easier converges to
local extrema when the initial values are far away from the true values. In addition,
we can see from Figure 5.2(a) and (b) that, in the high SNR region, the classification
performance with smaller maximum errors is close to the upper bounds. It indicates
that with proper initialization points, the proposed algorithm can provide promising
performance.
Next, we consider the classification performance of the proposed algorithm using
the fourth-order moment-based initialization scheme, as shown in Figure 5.3(a) and
(b) for QAM and PSK modulations, respectively. Figure 5.3(a) depicts that the clas-
sification performance of the proposed algorithm for QAM modulations using the
fourth-order moment-based initialization method attains Pc ≥ 0.8 for SNR > 10 dB.
When compared to that taking the true values plus bias as the initial values, we can
see that the classification performance of the proposed algorithm is comparable in the
SNR region ranges from 6 to 10 dB. The results show that the fourth-order moment-
based method is feasible in the moderate SNR region. In addition, we also compare the
classification performance of the proposed algorithm with the cumulant-based meth-
ods in [14,15]. The number of samples per receiver for the cumulant-based methods is
set to Nc = 2, 000. Note that the difference of cumulant values between higher order
modulation formats (e.g., 16-QAM and 64-QAM) is small; therefore, the classifica-
tion performance of the cumulant-based approaches is limited, which saturates in the
high SNR region. It is apparent that the classification performance of the proposed
algorithm outperforms that of the cumulant-based ones. Meanwhile, it indicates that
the propose algorithm is more sample efficient than the cumulant-based ones. Similar
results can be seen from Figure 5.3(b) when classifying PSK modulations. The advan-
tage in the probability of correct classification of the proposed algorithm is obvious
when compared to that of cumulant-based methods.
On the other hand, however, we can see from Figure 5.3(a) and (b) that the
classification performance of the proposed algorithm decreases in the high SNR
region. The possible reason is that the likelihood function in the low SNR region
is dominated by the noise, which contains less local extrema and is not sensitive to
the initialization errors. In contrast, the likelihood function in the high SNR region
is dominated by the signal, which contains more local extrema. In such a case, the
convergence result is more likely to be trapped at the local extrema, even when the
initial values are slightly far away from the true values. In addition, when comparing
Figure 5.3(a) with (b), it is noted that using the moment-based estimator, the PSK
modulations are more sensitive to initialization errors in the high SNR region.
To intuitively demonstrate the impact of the noise decomposition factor βkl on the
classification performance, we evaluate the probability of correct classification versus
the SNR in Figure 5.4, with curves parameterized by different choices of βkl . The lines
illustrate the classification performance with fixed βkl = 1/L, and the markers show
that with random βkl . As we can see, different choices of the noise decomposition
factor βkl does not affect the classification performance of the proposed algorithm.
170 Applications of machine learning in wireless communications

0.9

0.8

0.7
Pc

0.6

0.5 EM upper bound

EM with moment-based initialization
EM with δφ (l) = π/10, δa (l) = 0.3
0.4 k k
Cumulant-based method in [14]
Cumulant-based method in [15]
0.3
0 5 10 15 20
(a)
SNR (dB)
1

0.9

0.8

0.7
Pc

0.6

0.5 EM upper bound

EM with moment-based initialization
0.4 EM with δφ (l) = π/10, δa (l) = 0.3
k k
Cumulant-based method in [14]
Cumulant-based method in [15]
0.3
0 5 10 15 20
(b) SNR (dB)

Figure 5.3 (a) The classification performance of the proposed algorithm for QAM
with the fourth-order moment-based initialization scheme and (b) the
classification performance of the proposed algorithm for PSK with the
fourth-order moment-based initialization scheme

5.2.2 Continuous phase modulation classification in fading

channels via Baum–Welch algorithm
In contrast to constellation-based modulation formats (e.g., QAM and PSK) in the
previous section, CPM yields high spectral and power efficiency, which are widely
adopted in wireless communications, such as the satellite communications. Most of
the existing literature that classifies CPM signals is merely investigated in additive
white Gaussian noise (AWGN) channels [16] [17], while the effective algorithms in
the presence of the unknown fading channels have not been adequately studied yet.
Signal identification in cognitive radios using machine learning 171

0.9

0.8
Pc
0.7

0.6
δφ (l) = π/20, δa (l) = 0.1 with fixed β
k k
0.5 δφ (l) = π/20, δa (l) = 0.1 with rand β
k k
Moment-based initialization with fixed β
Moment-based initialization with rand β
0.4
0 5 10 15 20
SNR (dB)

Figure 5.4 The classification performance of the proposed algorithm for QAM
with curves parameterized by different choices of βkl

In this section, we consider a classification problem of CPM signals under

unknown fading channels, where the CPM signal is formulated as an HMM corre-
sponding to the memorable property of the continuous phases. A likelihood-based
classifier is proposed using the BW algorithm, which obtains the MLEs of the
unknown parameters of the HMM based on the EM algorithm.

5.2.2.1 Problem statement

Consider a typical centralized cooperation system with one transmitter and K
receivers, and a fusion center is introduced to fuse data from the K receivers to enhance
the classification performance. At the kth receiver, the discrete-time transmission
model is given by

yk,n = gk xn + wk,n , n = 1, . . . , N , k = 1, . . . , K (5.22)

where gk is the unknown complex channel fading coefficient from the transmitter to
the kth receiver, wk,n is circularly symmetric complex Gaussian with the distribution
CN (0, σk2 ), and xn is the transmit CPM signal. The continuous-time complex CPM
signal is expressed as
$
2E j(t;I )
x(t) = e (5.23)
T
where E is the energy per symbol, T is the symbol duration, and (t; I ) is the time-
varying phase. For t ∈ [nT , (n + 1)T ], the time-varying phase is represented as

n−L
n
(t; I ) = π h Il + 2π h Il q(t − lT ) (5.24)
l=−∞ l=n−L+1
172 Applications of machine learning in wireless communications

Sn Sn+1
0 κ1 0 +1
κ2 –1

κ3

2 κ4 2

κ5

κ6

κ7
3 3
2 κ8 2

Figure 5.5 A trellis of state transition for CPM with parameter

{M = 2, h = (1/2), L = 1}. In this example, the number of states is
Q0 = 4, and the information data is drawn from {+1, −1}. The
transition from state Sn to Sn+1 , denoted as κq , q = 1, . . . , 8, is
represented by solid line, if +1 is transmitted, and dotted line, if −1 is
transmitted

where h is the modulation index, Il is the lth information symbol drawn from the set
{±1, . . . , ±(M −1)}, with M as the symbol level, q(t) is the integral of the pulse shape
t
u(t), i.e., q(t) = 0 u(τ )dτ , t ≤ LT , and L is the pulse length. From (5.24), we can
see that a CPM format is determined by a set of parameters, denoted as {M , h, L, u(t)}.
Basically, by setting different values of these parameters, infinite number of CPM
formats can be generated. Let Sn = {θn , In−1 , . . . , In−L+1 } be the state of the CPM signal
at t = nT , where θn = πh n−L l=−∞ Il . The modulation index h is a rational number,
which can be represented by h = (h1 /h2 ), with h1 and h2 as coprime numbers. Then,
we define h0 as the number of states of θn , which is given by

h2 , if h1 is even
h0 =
2h2 , if h1 is odd.

Hence, we can obtain that the number of states of Sn is Q0 = h0 M L−1 . A trellis of state
transition for CPM with parameter {M = 2, h = (1/2), L = 1} is shown in Figure 5.5.
Let S be the set of CPM candidates, which is known at the receivers. The classifi-
cation task is to identify the correct CPM format s ∈ {1, . . . , S} based on the received
signals. Let y = [y1 , . . . , yK ], where yk is the received signal at the kth receiver,
g = {gk }Kk=1 , and x = {xn }Nn=1 , which are the observations, unknown parameters, and
hidden variables of the HMM, respectively. We formulate this problem as a multiple
composite hypothesis testing problem, and the likelihood-based classifier is adopted
to solve it. For hypothesis Hs , meaning that the transmit signal uses the CPM format s,
Signal identification in cognitive radios using machine learning 173

the log-likelihood function ln ps (y|g) is computed. The classifier makes the final
decision on the modulation format by

ŝ = arg max ln ps (y|g(†)

s ) (5.25)
s∈S

s is the MLEs of the unknown parameters g under hypothesis Hs , which can

where g(†)
be computed by

s = arg max ln ps (y|g).

g(†)
g
(5.26)

Unlike the case for constellation-based modulation formats, where the likelihood
function ps (y|g) is
obtained by averaging over all the unknown constellation symbols
A, i.e., ps (y|g) = x∈A ps (y|x, g)ps (x|g), since the CPM signal is time correlated, its
likelihood function cannot be calculated in such a way.

5.2.2.2 Classification of CPM via BW

In this section, the CPM signal is first formulated as an HMM regarding to its mem-
orable property, a likelihood-based classifier is then proposed based on the BW
algorithm, which utilizes the EM algorithm to calculate the MLEs of the unknown
parameters in the HMM [18].

HMM description for CPM signals

According to the phase memory of the CPM signal, it can be developed as an HMM.
We parameterize the HMM by λ = (A, B, π), which are defined as follows

1. A denotes the state probability matrix, the element αij = Pr{Sn+1 = j|Sn = i} is
expressed as

1
, if i → j is permissible
αij = M
0, otherwise.

2. B represents the conditional probability density function vector, the element

βi (yk,n ) = p(yk,n |Sn = i, gk ) is given by

1 |yk,n − gk x(Sn = i)|2
βi (yk,n ) = exp −
πσk2 σk2

with x(Sn = i) as the transmit CPM signal at t = nT corresponding to the state Sn .

3. π is the initial state probability vector, the element is defined as

1
πi = Pr{S1 = i} = .
Q0
174 Applications of machine learning in wireless communications

BW-based modulation classifier

The BW algorithm provides a way to compute the MLEs of the unknowns in the HMM
by using the EM algorithm. To be specific, under the hypothesis Hs , the E-step and
M-step at iteration r are written as

s , g) =
E-step: J (g(r) ps (y, x|g(r)
s ) log ps (y, x|g) (5.27)
x

M-step: g(r+1)
s = arg max J (g(r)
s , g). (5.28)
g

As shown in Figure 5.5, we denote κq as the transition from state Sn to Sn+1 , where
κq , q = 1, . . . , Q is drawn from the information sequence, with Q = Q0 M . Let
(xn , κq ) denote the transmit signal at t = nT that corresponding to κq . To simplify the
notifications, denote x1:n−1 and xn+1:N as the transmit symbol sequences {x1 , . . . , xn−1 }
and {xn+1 , . . . , xN }, respectively; we then can rewrite (5.27) as

s , g) =
J (g(r) s ) (log ps (x|g) + log ps (y|x, g))
ps (y, x|g(r)
x

K
N
= 1 + log ps (yk,n |xn , gk ) ps (y, x|g(r)
s )
k=1 n=1 x

K
N
Q
= 1 + log ps (yk,n |(xn , κq ), gk )
k=1 n=1 q=1

× ps (y, x1:n−1 , (xn , κq ), xn+1:N |g(r)
s )
x\xn

K
N
Q
= 2 − ps (y, (xn , κq )|g(r)
s )
k=1 n=1 q=1

1
× 2 |yk,n − gk x(Sn = z(q, Q0 ))|2 (5.29)
σk
where z(q, Q0 ) defines the remainder of (q/Q0 ).
Forward–backward algorithm: Define ηs (n, q) = ps (y, (xn , κq )|g(r)s ). Note that
(xn , κq ) is equivalent to the event {Sn = i, Sn+1 = j}, we can then derive ηs (n, q) as

K
ηs (n, q) = ps (yk , Sn = i, Sn+1 = j|gk )
k=1

K
= ps (yk |Sn = i, Sn+1 = j, gk )ps (Sn = i|gk )
k=1

×ps (Sn+1 = j|Sn = i, gk )

K
= υk,n (i)ωk,n+1 (j)ps (Sn+1 = j|Sn = i, gk ) (5.30)
k=1
Signal identification in cognitive radios using machine learning 175

where υk,n (i) = ps (yk,1:n |Sn = i, gk )ps (Sn = i|gk ) and ωk,n+1 (j) = ps (yNk,n+1 |Sn+1 =
j, gk ) are the forward and backward variables, respectively, which can be inductively
obtained by performing the forward–backward procedure as follows:
● Compute forward variable υk,n (i)
Initialize: υk,1 (i) = πi
Induction: υk,n (i) = Qj=1 υk,n−1 (j)αji βi (yk,n )
● Compute backward variable ωk,n+1 (j)
Initialize: ωk,N +1 (j) =1
Induction: ωk,n (j) = Qi=1 ωk,n+1 (i)αij βi (yk,n ).
Finally, by taking the derivative of (5.29) with respect to g and setting it to zero,
the unknown channel fading can be estimated by
N Q ∗
(r+1) n=1 q=1 ηs (n, q)yk,n x(Sn = z(q, Q0 ))
gs,k = N Q . (5.31)
n=1 q=1 ηs (n, q) x(Sn = z(q, Q0 ))

The proposed BW-based modulation classifier is summarized in the follow-

ing box:

BW-based modulation classifier for CPM

1. Set stopping threshold ε and maximum number of iterations Nm ;

2. FOR s = 1, . . . , |S|;
3. Set r = 0;
(0)
4. Initialize the unknown parameters gs,k ;
5. For n = 1, . . . , N , q = 1, . . . , Q, perform the forward–backward procedure
to compute ηs (n, q) in (5.30);
6. Compute J (g(r) s , g) according to (5.29);
7. Set r = r + 1;
(r+1)
8. Compute gs,k according to (5.31);
9. Compute
J (g(r+1)
s , g) using the new estimates;

10. If (J (gs , g) − J (g(r)
(r+1) (r)
s , g))/J (gs , g) > ε or r ≤ Nm , go to Step 5; other-
wise, set g(opt)
s = g (r+1)
s , and continue;
11. ENDFOR
12. Final decision is made by ŝ = arg max J (g(opt) s , g).
s

5.2.2.3 Numerical results

In this section, the classification performance of the proposed algorithm is examined
through various numerical simulations. The probability of correct classification Pc is
adopted as the measure metric. To comprehensively evaluate the proposed algorithm,
two experiments with different CPM candidate sets are considered and is compared
to the conventional approximate entropy (ApEn)-based method in [17]:
Experiment 1: Denote the CPM parameters as {M , h, L, u(t)}, four CPM candi-
dates are considered, namely, {2, 1/2, 1, rectangular pulse shape}, {4, 1/2, 2, raised
176 Applications of machine learning in wireless communications

cosine pulse shape}, {2, 3/4, 3, Gaussian pulse shape}, and {4, 3/4, 3, Gaussian
pulse shape}. For the Gaussian pulse shape, the bandwidth-time product is set to
B = 0.3.
Experiment 2: Consider the case where various values of M , h, and L are set,
with M = {2, 4}, h = {1/2, 3/4 }, L = {1, 2, 3}. The pulse shapes for L = 1, 2, 3
are set to rectangular, raised cosine, and Gaussian, respectively. In such a case,
12 different CPM candidates are considered.
For the proposed algorithm, we assume that the number of symbols per receiver is
N = 100, the stopping threshold is ε = 10−3 , and maximum iterations is Nm = 100.
For the ApEn-based algorithm, the simulation parameters are set according to [17].
Without loss of generality, we assume that the noise power at all receivers is the same.
Comparison with approximate entropy-based approach
We first consider the scenario with one receiver in the presence of AWGN and fading
channels. Figure 5.6 evaluates the classification performance of the proposed algo-
rithm when considering experiment 1 and is compared to that of the ApEn-based
algorithm in [17] as well. From Figure 5.6, we can see that in AWGN channels, the
proposed algorithm attains an acceptable classification performance, i.e., a classifica-
tion probability of 0.8, when SNR = 5 dB, and it achieves an error-free classification
performance at SNR = 10 dB.
Furthermore, we consider the classification performance in the presence of fad-
ing channels. We first initiate the unknown fading channel coefficients with the true
values with bias for the proposed algorithm [13]. Let ak and φk denote the true values
of the magnitude and phase of the fading channel, respectively, and let ak and φk
denote the maximum errors of the magnitude and phase, respectively. The initial val-
ues of the unknown magnitude and phase are arbitrarily chosen within [0, ak + ak ]

0.9
Probability of correct classification

0.8

0.7

0.6

0.5

0.4

0.3 Proposed, AWGN

Approximate entropy, AWGN
Proposed, fading w small bias
0.2 Proposed, fading w large bias
Approximate entropy, fading
0.1
0 5 10 15 20
SNR (dB)

Figure 5.6 The classification performance of the proposed algorithm under AWGN
and fading channels
Signal identification in cognitive radios using machine learning 177

and [φk − φk , φk + φk ], respectively. Two sets of initials of the unknowns are eval-
uated, whose maximum errors are set to (ak , φk ) = (0.1, π/20) and (0.3, π/10),
respectively. It is noted that the classification performance of the proposed algorithm
outcomes that of the ApEn-based algorithm in the fading channels. In particular, the
proposed algorithm provides classification probability of 0.8 for SNR > 15 dB, while
the probability of correct classification of the ApEn-based algorithm saturates around
0.6 in the high SNR region, which is invalid for the classification in fading channels.
Impact of initialization of unknowns
Note that the BW algorithm uses the EM algorithm to estimate the unknowns; there-
fore, its estimation accuracy highly relies on the initial values of the unknowns. In
Figure 5.7, we examine the impact of the initializations of the unknowns on the
classification performance of the proposed algorithm. Both experiments 1 and 2 are
evaluated. We consider multiple receivers to enhance the classification performance,
and the number of receivers is set to K = 3. In such a case, the unknown fadings are
first estimated at each receiver independently, the estimations are then forwarded to a
fusion center to make the final decision. The initial values of the unknown parameters
are set as the true values with bias, as previously described. It is seen from Figure 5.7
that, with smaller bias, the classification performance of the proposed algorithm pro-
vides promising classification performance. For experiment 1, the proposed classifier
achieves Pc > 80% when SNR > 10 dB, and for experiment 2, Pc > 80% is obtained
when SNR = 14 dB. Apparently, the cooperative classification further enhances the
classification performance when compared to that with a single receiver. Further-
more, with large bias, it is noticed that the classification performance of the proposed
algorithm degrades in the high SNR region. This phenomenon occurs when using the
EM algorithm [19]. The main reason is that the estimation results of the EM are not

0.9
Probability of correct classification

0.8

0.7

0.6

0.5

0.4

0.3
Multiple Rx w small bias : Experiment 1
Multiple Rx w large bias : Experiment 1
0.2 Multiple Rx w small bias : Experiment 2
Multiple Rx w large bias : Experiment 2
0.1
0 5 10 15 20
SNR (dB)

Figure 5.7 The impact of different initial values of the unknowns on the
classification performance under fading channels
178 Applications of machine learning in wireless communications

0.9
Probability of correct classification
0.8

0.7

0.6

0.5

0.4

0.3

0.2 Multiple Rx SA ini

Multiple Rx w small bias
Multiple Rx w large bias
0.1
0 2 4 6 8 10 12 14 16 18 20
SNR (dB)

Figure 5.8 The classification performance under fading channels with simulated
annealing initialization method

guaranteed to converge at the global maxima and are sensitive to the initialization
points of the unknowns. In the high SNR region, the likelihood function is dominated
by the signal and it has more local maxima than that in the low SNR region. Thus,
with large bias, the proposed scheme is more likely to converge at a local maxima,
which causes the degradation of the classification performance.
Performance with simulated annealing initialization
Next, we evaluate the classification performance of the proposed algorithm with the
SA initialization method, as illustrated in Figure 5.8. The parameters of the SA method
are set as in [13]. Experiment 1 is considered. Figure 5.8 shows that, using the SA
method to generate the initial values of the unknowns, the classification performance
of the proposed algorithm monotonically increases in the low-to-moderate SNR region
(0–10 dB). It implies that the SA scheme can provide appropriate initializations for
the proposed algorithm. Apparently, a gap is noticed between the classification per-
formance with the SA scheme and that with the true values of the unknowns plus bias.
However, note that how to determine proper initials could be an interesting topic for
future research, which is out of the scope of this chapter.

5.3 Specific emitter identification via machine learning

SEI, motivated by the identification and tracking task of unique emitters of interest
in military communications, is a technique to discriminate individual emitters by
extracting the identification feature from the received signal and comparing it with
Signal identification in cognitive radios using machine learning 179

a categorized feature set [20]. As a key technology in military communications, the

SEI has been intensively investigated over the past few decades. More recently, with
the advent of CRs and adaptive techniques, it has become increasingly important for
commercial applications.
In general, the SEI technique is based on the machine-learning theory, where
a simplified SEI system includes three main parts, namely, signal preprocessing,
feature extraction, and identification classifier. In the feature-extraction subsystem,
identification features are extracted from the received signal, and then the features are
input to the identification classifier to determine the class of the emitters. Hence, the
most important and challenging part of SEI based on machine learning is to design
proper and robust identification features, which have strong separation capability
among different emitters and are robust against the real-world scenarios.
In this section, we introduce three SEI algorithms, which extract identification
features via adaptive time–frequency analysis and implement classification task using
the support vector machine (SVM). We investigate the SEI approaches under realistic
scenarios. The three SEI algorithms are applicable to both single-hop and relaying
systems and are robust against several nonideal channel conditions, such as the non-
Gaussian and fading channels.

5.3.1 System model

5.3.1.1 Single-hop scenario
A time-division multiple-access communication system is considered with K emitters/
transmitters and one receiver, as shown in Figure 5.9(a). The number of the emitters
is assumed to be a priori known at the receiver, which can be achieved by algorithms
for estimating the number of transmitters. Since the target of the SEI is to distinguish
different emitters by extracting the unique feature carried by the transmit signals, a
model that describes the difference between emitters is first introduced below.
For an emitter, a power amplifier is the main component, whose nonlinear system
response characteristic is one of the principle sources of the specific feature, as known
as the fingerprint of an emitter. We use the Taylor series to describe the nonlinear
system [21,22], and let Ls denote the order for the Taylor series. For emitter k, the
system response function of the power amplifier is then defined as

Ls
[k]
[k] (x(t)) = αl (x(t))l (5.32)
l=1

where x(t) = s(t)ej2π nfT is the input signal at the power amplifier—with s(t) as the
baseband-modulated signal, f as the carrier frequency, and T as the sampling period—
[k]
{αl } denotes the coefficients of the Taylor series, and [k] (x(t)) denotes the output
signal at the power amplifier of the kth emitter, i.e., the transmit signal of the kth
emitter. Apparently, for emitters with the same order Ls , their different coefficients
represent the specific fingerprints, which are carried through the transmit signals
[k] (x(t)).
180 Applications of machine learning in wireless communications

At the receiver, the received signal is given by

[k]
r(t) = Hsd [k] (x(t)) + w(t), k = 1, . . . , K (5.33)
[k]
where Hsd is the unknown channel fading coefficient between the kth emitter and
the receiver, and w(t) is the additive noise. By substituting (5.32) into (5.33), we can
rewrite the received signal as

[k]

Ls
[k]
r(t) = Hsd αl (x(t))l + w(t). (5.34)
l=1

5.3.1.2 Relaying scenario

Next, we expand the single-hop scenario to the multi-hop one. Note that two-hop
communication systems are the most commonly adopted, such as the satellite com-
munications; therefore, we focus on the two-hop communication system with an
amplify-and-forward relay, as shown in Figure 5.9(b). The received signal at the relay
is expressed as
y(t) = Hsr[k] [k] (x(t)) + η(t), k = 1, . . . , K (5.35)
where Hsr[k]is the unknown channel fading from the kth emitter to the relay, and η(t)
is the additive noise.

1
[1]
H sd
[2]
2 H sd
D

[K]
H sd

(a)

1
[1]
H sr
[2]
2 H sr
Hrd
R D

[K]
H sr

K
(b)

Figure 5.9 (a) The system model of the single-hop scenario and (b) the system
model of the relaying scenario
Signal identification in cognitive radios using machine learning 181

Then, the received signal at the receiver, which is forwarded by the relay, is
written as
r(t) = Hrd (y(t)) + υ(t)

= Hrd Hsr[k] [k] (x(t)) + η(t) + υ(t) (5.36)
where (·) denotes the system response characteristic of the power amplifier of the
relay, Hrd is the unknown channel fading coefficient from the relay to the receiver,
and υ(t) is the additive noise. Similarly, we use the Taylor series to define (·), which
is given by

Lr
(y(t)) = βm (y(t))m (5.37)
m=1

where Lr denotes the order of Taylor series for the power amplifier of the relay,
and {βm } represent the fingerprint of the relay. Hence, the received signal is further
expressed as

Lr
r(t) = Hrd βm (y(t))m + υ(t) (5.38)
m=1
m

Lr
Ls
[k]
= Hrd βm Hsr[k] αl (x(t)) + η(t)
l
+ υ(t). (5.39)
m=1 l=1

It is obvious that the features carried by the received signal are the combinations of
the fingerprint of both the emitter and the relay, meaning that the fingerprint of the
emitter is contaminated by that of the relay, which causes negative effect on SEI.

5.3.2 Feature extraction

In this section, three feature-extraction algorithms are introduced, namely, the entropy
and first- and second-order moments-based algorithm (EM2 ), the correlation-based
(CB) algorithm, and the Fisher discriminant ratio (FDR)-based algorithm. All the
three algorithms extract identification features from the Hilbert spectrum, which is
obtained by employing the Hilbert–Huang transform (HHT).

5.3.2.1 Hilbert–Huang transform

The HHT is a powerful and adaptive tool to analyze nonlinear and nonstationary
signals [23]. The principle of the HHT being adaptive relies on the empirical mode
decomposition (EMD), which is capable of decomposing any signals into a finite
number of intrinsic mode functions (IMFs). By performing the Hilbert transform on
the IMFs, we can obtain a time–frequency–energy distribution of the signal, also
known as the Hilbert spectrum of the signal.
Empirical mode decomposition
The target of the EMD is to obtain the instantaneous frequency with physical meaning,
therefore, the IMF should satisfy two conditions [23]: (1) the number of extrema and
182 Applications of machine learning in wireless communications

the number of zero-crossings should either be equal, or the difference is one at most;
(2) at any point, the sum of the upper and lower envelopes, respectively, defined by
the local maxima and minima, should be zero.
Let z(t) be the original signal, the EMD uses an iteration process to decompose
the original signal into the IMFs, which is described as follows [23]:
1. First, identify all local maxima and minima, then employing the cubic spline
fitting to obtain the upper and lower envelopes of the signal;
2. Compute the mean of the upper and lower envelopes, denoted by μ10 (t). Subtract
μ10 (t) from z(t) to obtain the first component z10 (t), i.e., z10 (t) = z(t) − μ10 (t);
3. Basically, since the original signal is complicated, the first component does not
satisfy the IMF conditions. Thus, steps 1 and 2 are repeated p times until z1p (t)
becomes an IMF:
z1p (t) = z1(p−1) (t) − μ1p (t), p = 1, 2, . . . , (5.40)
where μ1p (t) is the mean of the upper and lower envelopes of z1(p−1) (t). We
define that

Ts
|z1(p−1) (t) − z1p (t)|2
ξ= 2
(5.41)
t=0
z1(p−1) (t)
where Ts is the length of the signal. Then, the stopping criterion of this shifting
process is when ξ < ε. Note that an empirical value of ε is set between 0.2
and 0.3.
4. Denote c1 (t) = z1p (t) as the first IMF. Subtract it from the z(t) to obtain the
residual, which is
d1 (t) = z(t) − c1 (t). (5.42)
5. Consider the residual as a new signal, repeat steps 1 to 4 on all residuals dq (t),
q = 1, . . . , Q, to extract the remaining IMFs, i.e.:
d2 (t) = d1 (t) − c2 (t),
··· (5.43)
dQ (t) = dQ−1 (t) − cQ (t)
where Q is the number of IMFs. The stopping criterion of the iteration procedure
is when dQ (t) < ε, or it becomes a monotonic function without any oscillation.
From (5.42) and (5.43), we can rewrite z(t) as

Q
z(t) = cq (t) + dQ (t). (5.44)
q=1

Hilbert spectrum analysis

We omit the residual dQ (t) in (5.44), and implement the Hilbert transform on the
extracted IMFs, the original signal is then expressed as
⎛ ⎞
Q
z(t) = ⎝ aq (t) exp j ωq (t)dt ⎠ (5.45)
q=1
Signal identification in cognitive radios using machine learning 183
√
where
% (·) denotes the real component of a complex variable, j = −1, aq (t) =
cq2 (t) + ĉq2 (t) and ωq (t) = (dθq (t)/dt) are the instantaneous amplitude and frequency
∞
of the IMF cq (t), respectively, in which ĉq (t) = π1 −∞ (cq (τ )/(t − τ ))dτ is the Hilbert
transform, and θq (t) = arctan (ĉq (t)/cq (t)) is the phase function.
With the instantaneous amplitude and frequency of the IMFs, we can obtain the
Hilbert spectrum of the original signal, denoted by H(ω, t). In this section, we use
the squared value of the instantaneous amplitude, therefore, the Hilbert spectrum
provides the time–frequency–energy distribution.
5.3.2.2 Entropy and first- and second-order moments-based algorithm
The EM2 algorithm extracts features by measuring the uniformity of the Hilbert
spectrum. The feature contains three-dimensional data, which are the energy entropy,
the first-order and second-order moments, respectively.
Energy entropy
We use the definition of information entropy to define the energy entropy of the
Hilbert spectrum. First, the Hilbert spectrum is divided into several time–frequency
slots. Denote Hij (ω, t), i = 1, . . . , Gt , j = 1, . . . , Gω as the (i, j)th time–frequency
slot, where Gt is the number of time slots with resolution t and Gω is the number
of frequency slots with resolution ω. By using the expression of the information
entropy [24], the energy entropy of the Hilbert spectrum is defined as
Gt Gω
I =− pij log pij (5.46)
i=1 j=1
where pij = Eij /E is the proportion of the energy of each time–frequency slot, with
E as the total energy of the Hilbert spectrum, and Eij as the energy of the (i, j)th
time–frequency slot, given by
it jω
Eij = Hij (ω, t)dωdt. (5.47)
(i−1)t (j−1)ω

First- and second-order moments

The first- and second-order moments adopt the concept of color moments in image
processing, which measure the color distribution of an image. To compute the first-
and second-order moments, we first map the Hilbert spectrum into a gray scale image,
where the Hilbert spectrum elements are described by shades of gray with intensity
information, which is
& '
Hm,n
Bm,n = (2ζ − 1) × (5.48)
max{m,n} Hm,n
where Bm,n is the (m, n)th value of the gray scale image matrix B, Hm,n is the (m, n)th
element of the Hilbert spectrum matrix H,2 and · is the floor function, which equals

2
As a two-dimensional spectrum, we represent the Hilbert spectrum through a matrix, referred to as the
Hilbert spectrum matrix, where the indices of the columns and rows correspond to the sampling point and
instantaneous frequency, respectively, and the elements of the matrix are combinations of the instantaneous
energy of the IMFs.
184 Applications of machine learning in wireless communications

the nearest lower integer value. Taking ζ -bit gray scale as an example, the largest value
of the Hilbert spectrum is converted to the gray scale (2ζ − 1), while other values are
linearly scaled.
The first- and second-order moments of the gray scale image are, respectively,
defined as

1
M N
μ= Bm,n (5.49)
NH m=1 n=1
1/2
1 2
M N
ς = Bm,n − μ (5.50)
NH m=1 n=1

where NH = M × N is the total number of pixels (elements) of the gray scale image
matrix. Note that the first-order moment interprets the average intensity of the gray
scale image, and the second-order moment describes the standard deviation of the
shades of gray.

5.3.2.3 Correlation-based algorithm

It is observed that the shape of the time–frequency–energy distribution of signals
from the same emitter are similar, while that from different emitters are diverse;
therefore, the correlation coefficients between Hilbert spectra can be extracted as the
identification features.
Let Hi and Hj , i, j = 1, . . . , NR represent the Hilbert spectrum matrices of the
ith and jth training sequence, respectively, where NR is the total number of train-
ing sequences over K classes. The correlation coefficient between Hi and Hj is
expressed as

n Hi,m,n − E(Hi ) Hj,m,n − E(Hj )
ρ (i,j) = $ m
(5.51)
2 2
m n Hi,m,n − E(Hi ) m n Hj,m,n − E(Hj )

where Hi,m,n (Hj,m,n ) denotes the (m, n)th element of the Hilbert spectrum matrix
Hi (Hj ), and E(·) is the mean of the elements. Equation (5.51) depicts the linear
dependence between Hi and Hj ; larger ρ (i,j) implies that Hi and Hj are more likely
from the same emitter; otherwise, ρ (i,j) close to zero indicates that Hi and Hj are
from diverse emitters.

5.3.2.4 Fisher’s discriminant ratio-based algorithm

It should be noted that large number of elements in a Hilbert spectrum are featureless,
meaning that they have little discrimination. In contrast to the EM2 and CB algorithms
that exploits all the elements of a Hilbert spectrum, the FDR algorithm selects the
elements of a Hilbert spectrum that provides well separation between two classes.
Let (k1 , k2 ) be a possible combination of two classes arbitrarily selected from K
classes, k1 = k2 . Note that the total number of possible combinations (k1 , k2 ) for all K
Signal identification in cognitive radios using machine learning 185

classes is C = K(K − 1)/2. For (k1 , k2 ), we define the FDR at time–frequency spot
(ω, t) as
2
[k ] [k ]
Ei Hi 1 (ω, t) − Ei Hi 2 (ω, t)
F (k1 ,k2 ) (ω, t) = (5.52)
[k]
k=k1 ,k2 Di Hi (ω, t)

[k]
where Hi (ω, t), i = 1, . . . , N̄0 is the Hilbert spectrum of the ith training sequence
of the kth class at (ω, t), with
N̄0 as the number of training sequences for each class,
[k] [k]
and Ei Hi (ω, t) and Di Hi (ω, t) denote the mean and variance of the training
sequences of class k at (ω, t), respectively. From (5.52), we can see that the FDR
F (k1 ,k2 ) (ω, t) measures the separability of the time–frequency spot (ω, t) between
classes k1 and k2 . It indicates that the time–frequency spot (ω, t) with larger FDR
provides larger separation between the mean of two classes and smaller within-class
variance, which shows stronger discrimination.
(k ,k ) (k ,k )
For each combination (k1 , k2 ), we define = {F1 1 2 (ω, t), . . . , FNH1 2 (ω, t)}
as the original FDR sequence. Sort in descending order and denote the new
FDR sequence as ˜ = {F̃1(k1 ,k2 ) (ω, t), . . . , F̃N(k1 ,k2 ) (ω, t)}, i.e., F̃1(k1 ,k2 ) (ω, t) ≥ · · · ≥
H
(k ,k )
F̃NH1 2 (ω, t). Let {(ω̃1 , t̃1 ), . . . , (ω̃NH , t̃NH )} be the time–frequency slots which cor-
respond to the rearranged FDR sequence . ˜ Then, we select the time–frequency
spots that correspond to the S largest FDR as optimal time–frequency spots, denoted
as Z (c) = {(ω̃s(c) , t̃s(c) ), s = 1, . . . , S, c = 1, . . . , C}. The total set of optimal time–
(
frequency spots is defined as the union of Z (c) , i.e., Z = Cc=1 Z (c) . For the same
(ω̃s , t̃s ) between different combinations (k1 , k2 ), only one is retained in order to avoid
duplication, i.e., Z = {(ω̃1 , t̃1 ), . . . , (ω̃D , t̃D )}, where D is the number of optimal
time–frequency spots without duplication, with D ≤ S × ((K(K − 1))/2).

5.3.3 Identification procedure via SVM

By using the feature extraction algorithms proposed in the previous section to extract
the identification features, the SVM is then adopted to implement the identification
procedure. SVM is a widely used supervised learning classifier in machine learning,
which is originally designed for two-class classification problems. Input a set of
labeled training samples, the SVM outputs an optimal hyperplane, which can classify
new samples.
Linear SVM: Suppose that {vi , ιi } is the training set, with vi as the training vector
and ιi ∈ {1, −1} as the class label. We first consider the simplest linear case, where
the labeled data can be separated by a hyperplane χ (v) = 0. The decision function
χ (v) is expressed as [25]:

χ (v) = wT v + b (5.53)

where w is the normal vector to the hyperplane and (b/ w ) determines the per-
pendicular offset of the hyperplane from the origin, with · as the Euclidean
norm.
186 Applications of machine learning in wireless communications

Given a set of training data, labeled as positive and negative ones, we define
the closest distances from the positive and negative points to the hyperplane as md,+
and md,− , respectively. Then, the optimization task of the hyperplane is to make the
margin, md = md,+ + md,− , the largest. To simplify the derivation, two hyperplanes
that bound the margin are defined as
wT v + b = 1 (5.54)
w v + b = −1.
T
(5.55)
Note that the distance between the two hyperplanes is md = (2/ w ), the original
problem of maximizing md can be converted to a constrained minimization problem,
which is [25,26]:
1
min w 2 (5.56)
2

s.t. ιi wT vi + b ≥ 1, i = 1, . . . , N̄ (5.57)

where N̄ is the number of training examples. By introducing nonnegative Lan-

grage multipliers λi ≥ 0, (5.56) is transformed to a dual quadratic programming
optimization problem, given by [25,26]:

N̄
1
N̄
max λi − λi λj ιi ιj vi , vj (5.58)
λ
i=1
2 i,j=1

s.t. λi ≥ 0, i = 1, . . . , N̄ (5.59)

N̄
λ i ιi = 0 (5.60)
i=1
N̄
where w and b are represented by w = i=1 λi ιi vi and b = −(1/2)( max wT vi +
i:ι1 =−1
min wT vi ), respectively.
i:ι1 =1
The decision function is obtained by solving the optimization problem in (5.58):

N̄
χ (v) = λi ιi vi , v + b (5.61)
i=1

where ·, · denotes the inner product. The decision criterion of the correct classifica-
tion is ιl χ(ul ) > 0, i.e., the testing example ul that satisfies χ (ul ) > 0 is labeled as 1;
otherwise, it is labeled as −1.
Nonlinear SVM: For the case where vi cannot be simply distinguished by the linear
classifier, a nonlinear mapping function φ is utilized to map vi to a high-dimensional
space F, in which the categorization can be done by a hyperplane. Similarly, the
decision function is expressed as [25]:

N̄
χ(v) = λi ιi φ(vi ), φ(v) + b. (5.62)
i=1
Signal identification in cognitive radios using machine learning 187

In such a case, a kernel function κ(vi , v) is defined to avoid the computation of the inner
product φ(vi ), φ(v), which is generally intractable in high-dimensional spaces [27].3

Using w = N̄i=1 λi ιi vi and the kernel function, we rewrite (5.62) as

N̄
χ(v) = λi ιi κ(vi , v) + b. (5.63)
i=1

The decision rule is the same as that for the linear classifier.
Multi-class SVM: Next, we consider the case of multiple classes. The multi-
class classification problem is solved by reducing it to several binary classification
problems. Commonly adopted methods include one-versus-one [28], one-versus-
all [28] and binary tree architecture [29] techniques. In this section, we employ the
one-versus-one technique for the multi-class problem, by which the classification is
solved using a max-win voting mechanism, and the decision rule is to choose the
class with the highest number of votes.
The training and identification procedures of the three proposed algorithms using
the SVM are summarized as follows:

Training and identification procedures of the EM2 algorithm

Training procedure: Let Hi , i = 1, . . . , N̄ , denote the Hilbert spectrum
matrix of the training sequence i, with N̄ as the total number of training sequences
over all K classes.
1. From (5.46), we compute the energy entropy of Hi , denoted as Īi .
2. Map Hi to Bi using (5.48), with Bi as the Hilbert gray scale image matrix
of the training sequence i. From (5.49) and (5.50), we compute the first- and
second-order moments, denoted as (μ̄i , ς̄i )T , respectively.
T
3. Generate the three-dimensional training vector, i.e., vi = Īi , μ̄i , ς̄i .
4. Let {vi , ιi } be the training set, where ιi ∈ {1, . . . , K} is the label of each class.
Use the labeled training set to train the optimal hyperplane χ (v).
Identification procedure: Let Hl , l = 1, . . . , N , denote the Hilbert spectrum
matrix of the lth test sequence of an unknown class, where N is the number of test
sequences.
1. Use (5.46)–(5.50) to compute the energy entropy Il , the first- and second-order
moments (μl , ςl )T of Hl .
2. Generate the test vector, denoted as ul = (Il , μl , ςl )T .
3. The identification task is implemented by employing the SVM classifier
defined in the training procedure. For K = 2, ul which satisfies χ(ul ) > 0
is labeled as class 2; otherwise, it is labeled as class 1. For K > 2, the one-
versus-one technique is applied, where the decision depends on the max-win
voting mechanism, i.e., the class with the highest number of votes is considered
as the identification result.

Typical kernel functions include the Gaussian radial-base function (RBF), κ(x, y) = e− x−y
3 2 /2γ 2
, and the
polynomial kernel, κ(x, y) = (x, y)d , with d as the sum of the exponents in each term.
188 Applications of machine learning in wireless communications

Training and identification procedures of the CB algorithm

Training procedure: Let Hi and Hj denote the Hilbert spectrum matrices of
the training sequences i and j, respectively, where i, j = 1, . . . , N̄ , with N̄ as the
total number of the training sequences over all K classes.
1. From (5.51), we calculate the correlation coefficient ρ̄ (i,j) between Hi and Hj .
For the training sequence i, the training vector is N̄ -dimensional, denoted as
" #T
ρ̄ i = ρ̄ (i,1) , . . . , ρ̄ (i,N̄ ) .
) *
2. Let ρ̄ i , ιi be the set of training data with ιi ∈ {1, . . . , K} as the label of each
class. Then, input the labeled training set to the SVM classifier to optimize
the decision hyperplane χ (ρ̄).
Identification procedure: Let Hl , l = 1, . . . , N , be the Hilbert spectrum
matrix of the test sequence of an unknown class, where N is the number of test
sequences.
1. For the test sequence l, the correlation coefficient ρ (l,i) between Hl and Hi
is calculated from (5.51), the N̄ -dimensional test vector is denoted as ρ l =
[ρ (l,1) , . . . , ρ (l,N̄ ) ]T .
2. Classify the test sequence by employing the SVM classifier. For K = 2, ρ l
which satisfies χ (ρ l ) > 0 is labeled as class 2; otherwise, it is labeled as
class 1. For K > 2, the one-versus-one technique is applied, where the decision
depends on the max-win voting mechanism, i.e., the class with the highest
number of votes is considered as the identification result.

Training and identification procedures of the FDR algorithm

Training procedure: Let Hi (ω, t) denote the Hilbert spectrum of the training
sequence i at time–frequency spot (ω, t), where i = 1, . . . , N̄ , with N̄ as the total
number of the training sequences over all K classes.
1. For the combination (k1 , k2 ), we compute the original FDR sequence =
(k ,k ) (k ,k )
{F1 1 2 (ω, t), . . . , FNH1 2 (ω, t)} from (5.52);
2. Obtain the FDR sequence (descending order of elements) ˜ =
(k1 ,k2 ) (k1 ,k2 )
{F̃1 (ω, t), . . . , F̃NH (ω, t)};
3. Select the time–frequency spots that correspond with the S largest FDR
in ˜ to form the optimal time–frequency spots set Z (c) = {(ω̃s(c) , t̃s(c) ), s =
1, . . . , S, c = 1, . . . , C};
4. Repeat steps 1–3 over all C = K(K − 1)/2 possible combinations ( (k1 , k2 ) to
obtain the total set of optimal time–frequency spots, Z = Cc=1 Z (c) . For the
same (ω̃s , t̃s ) between different combinations (k1 , k2 ), only one of them is
retained in order to avoid duplication;
5. For the training sequence i, the Hilbert spectrum elements corresponding to
(ω̃1 , t̃1 ), . . . , (ω̃D , t̃D ) are extracted to form a D-dimensional training vector,
!T
expressed as vi = Hi (ω̃1 , t̃1 ), . . . , Hi (ω̃D , t̃D ) ;
Signal identification in cognitive radios using machine learning 189

6. Let {vi , ιi } be the set of training data with ιi ∈ {1, . . . , K} as the label of each
class. Then, the data is input into the SVM classifier for training, i.e., to obtain
the optimal w and b of the decision hyperplane χ (v).
Identification procedure: Let Hl (ω, t), l = 1, . . . , N , denote the Hilbert
spectrum of test sequence l at time–frequency spot (ω, t) of a unknown class,
where N is the number of test sequences.
1. For the test sequence l, extract the elements corresponding to the D opti-
mal time–frequency spots as the test vector, i.e., ul = [Hl (ω̃1 , t̃1 ), . . . ,
Hl (ω̃D , t̃D )]T ;
2. Utilize the SVM classifier to identify the test sequence. For K = 2, ul which
satisfies χ(ul ) > 0 is labeled as class 2; otherwise, it is labeled as class 1.
For K > 2, one-versus-one technique is applied, where the decision depends
on the max-win voting mechanism, i.e., the class with the highest number of
votes is considered as the identification results.

5.3.4 Numerical results

In this section, we provide various numerical experiments based on the probability of
correct identification Pc , to evaluate the identification performance of the proposed
algorithms, and compare it with that of some conventional approaches. The identifi-
cation performance of the proposed algorithms are evaluated under three scenarios,
namely, the AWGN channels, the flat-fading channels, and the non-Gaussian noise
channels. It is noted that distinguishing the identity of different emitters is a difficult
task even in the presence of the AWGN channels, the results in AWGN channels
provides the baseline of the proposed algorithms, which show that they can obtain
better identification performance than the conventional methods. Furthermore, we
examine the identification performance of the proposed algorithms in the presence of
flat-fading channels and non-Gaussian noise channels to illustrate that the proposed
algorithms are robust against more realistic and complicated scenarios.
For the power amplifiers of both the emitters and the relay, we assume that
the order of the Taylor polynomial is Ls = Lr = 3. In the simulations, we con-
sider three cases, i.e., the number of emitters are set to K = 2, 3 and 5. For the
power amplifier
of the emitters, denote the coefficients of the Taylor polynomial
[k] [k]
as α = α1 , . . . , αLs , and each coefficient is set to α [1] = (1, 0.5, 0.3)T , α [2] =
[k]

(1, 0.08, 0.6)T , α [3] = (1, 0.01, 0.01)T , α [4] = (1, 0.01, 0.4)T and α [5] = (1, 0.6, 0.08)T ,
respectively.
To be specific, for the case that K = 2, the coefficient
matrix is
AT2 = α [1]; α [2] ; for the case that K = 3, it is A3 = α [1] ; α [2] ; α [3] ; and for K = 5,
it is A5 = α [1] ; α [2] ; α [3] ; α [4] ; α [5] . For the coefficient matrix of the power amplifier
Taylor polynomial model of the relay, we set it to BT = (1, 0.1, 0.1). The SVM classi-
fier is implemented by using the LIBSVM toolbox, in which we adopt the Gaussian
RBF, κ(x, y) = e− x−y /2γ , as the kernel function with parameter γ = 0.1. For each
2 2

class, we set the number of training and test sequences as N̄0 = N0 = 50.
● Algorithms performance in AWGN channel
190 Applications of machine learning in wireless communications

In the AWGN channels, we first evaluate the identification performance of the

EM2 algorithm for K = 2 and K = 3 in the single-hop and relaying scenarios, shown
as Figure 5.10. For K = 2, the acceptable identification performance (Pc ≥ 0.8) is
attained for SNR ≥ 0 dB and SNR > 2 dB under the single-hop and relaying scenarios,
respectively. For K = 3, Pc achieves 0.8 at SNR = 6 dB in the single-hop scenario and
SNR = 8 dB in the relaying scenario. When compared to the conventional algorithm
in [30], the advantage of the EM2 algorithm is apparent. In particular, Pc enhances
more than 30% at SNR = 8 dB for both K = 2 and K = 3 in both single-hop and
relaying scenarios. In addition, it is seen that in high SNR region, the identification
accuracy of the EM2 algorithm in the single-hop and relaying scenarios is similar,
while the conventional algorithm presents a gap of 10% between these cases. It implies
that the EM2 algorithm can effectively combat the negative effect caused by the relay
on the identification of emitters at high SNR.
We then consider the identification performance of the EM2 algorithm versus
the number of training and test samples. In Figure 5.11, Pc is plotted as a function of
the number of training sequences for each class N̄0 , with curves parameterized by the
number of test sequences for each class N0 . We take the case that K = 2 and K = 3
in the relaying scenario at SNR = 14 dB as an example. It is noted that the number
of training and test samples barely influences the identification performance.

K=2 K=3
1 1

0.9 0.9

0.8 0.8

0.7 0.7
Pc
Pc

0.6 0.6

0.5 0.5

0.4 EM2 wo relay 0.4 EM2 wo relay

EM2 w/ relay EM2 w/ relay
[9] wo relay [9] wo relay
[9] w/ relay [9] w/ relay
0.3 0.3
0 5 10 15 20 0 5 10 15 20
SNR (dB) SNR (dB)

Figure 5.10 The identification performance of the EM2 algorithm in AWGN

channel
Signal identification in cognitive radios using machine learning 191

0.98
N0 = 25
0.96 N0 = 50
K=2 N0 = 75
0.94
N0 = 100
0.92
Pc

0.9

0.88

0.86

0.84

0.82
K=3
0.8
10 20 30 40 50 60 70 80 90 100
Number of training samples, N¯0

Figure 5.11 The identification performance of the EM 2 algorithm in AWGN

channel. The number of test sequences for each class is
N0 = 25, 50, 75, 100 and SNR = 14 dB

Next, we depict the identification performance of the CB algorithm in

Figure 5.12. where the identification performance for K = 2 and K = 3 in the single-
hop and relaying scenarios is illustrated. When K = 2, the CB algorithm reaches
Pc ≥ 0.8 for SNR ≥ 0 dB in both scenarios. When K = 3, the identification perfor-
mance attains Pc = 0.8 at SNR = 4 dB in the single-hop scenario and at SNR = 7 dB
in the relaying scenario. Moreover, we compare the identification performance of
the CB algorithm with that of the EM2 algorithm, and the results show that the CB
algorithm provides better identification performance for K = 2 and K = 3 in both
single-hop and relaying scenarios.
In Figure 5.13, the identification performance of the FDR algorithm is illustrated,
along with that of the EM2 and CB algorithms when classifying five emitters for
single-hop and relaying scenarios. Apparently, the FDR attains a reliable performance
when K = 5 and outperforms that of the EM2 and CB algorithms. Specifically, the
FDR algorithm obtains Pc ≥ 0.8 at SNR = 5 dB in the single-hop scenario and at
SNR = 7 dB in the relaying scenario. The reason is that the FDR algorithm extracts
elements with strong separability as identification features, which can provide better
identification performance.
Table 5.1 summarizes the identification performance of the three proposed algo-
rithms, along with that of the algorithm in [30], for SNR = 4, 12, and 20 dB in AWGN
channels. It can be seen that when K = 2 and K = 3, all the proposed algorithms
attain good identification performance in both single-hop and relaying scenarios.
192 Applications of machine learning in wireless communications

K=2 K=3
1 1

0.9 0.9

0.8 0.8

0.7 0.7
Pc

Pc
0.6 0.6

0.5 0.5

0.4 CB wo relay 0.4 CB wo relay

CB w/ relay CB w/ relay
[9] wo relay [9] wo relay
[9] w/ relay [9] w/ relay
0.3 0.3
0 5 10 15 20 0 5 10 15 20
SNR (dB) SNR (dB)

Figure 5.12 The identification performance of the CB algorithm in AWGN channel

0.9

0.8

0.7
Pc

0.6

0.5

0.4 FDR wo relay

FDR w relay
2
EM wo relay
0.3 EM2 w relay
CB wo relay
CB w/ relay
0.2
0 2 4 6 8 10 12 14 16 18 20
SNR (dB)

Figure 5.13 The identification performance of the FDR algorithm in AWGN

channel
Signal identification in cognitive radios using machine learning 193

Table 5.1 Identification performance Pc for SNR = 4,12,20 dB in AWGN channels

4 dB 12 dB 20 dB

EM2 CB FDR [30] EM2 CB FDR [30] EM2 CB FDR [30]

K = 2 Single-hop 0.93 0.97 0.99 0.55 0.98 0.99 1.00 0.70 0.99 1.00 1.00 0.94
Relaying 0.87 0.90 0.94 0.53 0.97 0.99 0.99 0.62 0.99 0.99 0.99 0.85
K = 3 Single-hop 0.79 0.81 0.97 0.41 0.88 0.93 0.99 0.56 0.90 0.96 0.99 0.81
Relaying 0.72 0.70 0.89 0.38 0.86 0.91 0.98 0.51 0.89 0.95 0.99 0.72
K = 5 Single-hop 0.56 0.56 0.77 0.25 0.63 0.66 0.92 0.28 0.65 0.71 0.93 0.33
Relaying 0.48 0.50 0.70 0.25 0.61 0.62 0.89 0.26 0.64 0.69 0.92 0.30

As expected, the FDR algorithm obtains the best identification performance, then
followed by the CB algorithm and finally the EM2 algorithm. The FDR algorithm
can effectively identify emitters when K = 5 since it extracts features with strong
separability. In addition, the identification performance of the proposed algorithms
outperforms that of the algorithm in [30], especially in the relaying scenario.
● Algorithms performance comparison in non-Gaussian noise channel
Next, the identification performance of the proposed algorithms is evaluated
under the non-Gaussian noise channels and compared to that in [30]. In the simula-
tions, we assume that the non-Gaussian noise follows the Middleton Class A model,
where the probability distribution function (pdf) is expressed as [31,32]:

∞
Am
fClassA (x) = e−A e−(x /2σm )
2 2
+ (5.64)
m=0 m! (2πσm2 )

where A is the impulse index, and σm2 = (((m/A) + )/(1 + )) is the noise variance,
with as the ratio of the intensity of the independent Gaussian component and the
intensity of the impulsive non-Gaussian component. We set A = 0.1 and = 0.05,
respectively. In addition, we assume that the number of terms in the Class A pdf M is
finite, i.e., m ∈ [0, M − 1], and set M = 500 [31]. No fading is considered.
Table 5.2 summarizes the identification accuracy of the proposed algorithms and
the algorithm in [30], at SNR = 4, 12 and 20 dB in the presence of the non-Gaussian
noise channels. Comparing the results with that in the AWGN channels, it is noticed
that for K = 2 and K = 3 in the single-hop scenario, the proposed algorithms have
little degradation in the identification performance, and in the relaying scenario, the
proposed algorithms effectively combat the negative effect of the non-Gaussian noise
in the high SNR region. The results indicate that the proposed algorithms are appli-
cable to non-Gaussian noise channels. Furthermore, it is obvious that all proposed
algorithms outperform that of the conventional method in [30], which performs poorly
especially in the relaying scenario.
● Algorithms performance comparison in flat-fading channel
194 Applications of machine learning in wireless communications

Table 5.2 Identification performance Pc for SNR = 4, 12, 20 dB in non-Gaussian

noise channels

4 dB 12 dB 20 dB

EM2 CB FDR [30] EM2 CB FDR [30] EM2 CB FDR [30]

K = 2 Single-hop 0.90 0.91 0.98 0.55 0.98 0.99 0.99 0.70 0.99 0.99 0.99 0.94
Relaying 0.82 0.83 0.86 0.53 0.96 0.97 0.99 0.62 0.98 0.99 0.99 0.85
K = 3 Single-hop 0.74 0.75 0.94 0.40 0.86 0.91 0.99 0.56 0.89 0.95 0.99 0.81
Relaying 0.60 0.62 0.81 0.38 0.85 0.87 0.97 0.51 0.88 0.94 0.99 0.72
K = 5 Single-hop 0.53 0.50 0.73 0.25 0.61 0.62 0.90 0.28 0.64 0.69 0.93 0.33
Relaying 0.41 0.43 0.63 0.25 0.57 0.57 0.88 0.26 0.62 0.67 0.91 0.30

Table 5.3 Identification performance Pc for SNR = 4, 12, 20 dB in fading channels

4 dB 12 dB 20 dB

EM2 CB FDR [30] EM2 CB FDR [30] EM2 CB FDR [30]

K = 2 Single-hop 0.81 0.90 0.97 0.51 0.96 0.98 0.99 0.56 0.99 0.99 0.99 0.68
Relaying 0.63 0.72 0.82 0.50 0.90 0.90 0.99 0.53 0.98 0.98 0.99 0.59
K = 3 Single-hop 0.65 0.73 0.91 0.36 0.86 0.89 0.98 0.40 0.90 0.93 0.99 0.54
Relaying 0.63 0.50 0.77 0.35 0.80 0.76 0.96 0.39 0.98 0.92 0.98 0.45
K = 5 Single-hop 0.50 0.49 0.71 0.21 0.60 0.61 0.89 0.23 0.62 0.66 0.92 0.29
Relaying 0.35 0.40 0.58 0.20 0.55 0.55 0.86 0.21 0.60 0.64 0.91 0.25

Next, the identification performance in the presence of flat-fading channels is

examined. We assume that the channel-fading coefficients are unknown at the receiver.
Table 5.3 summarizes the identification performance of all the proposed algo-
rithms and compared to that in [9], for SNR = 4, 12 and 20 dB in fading channels
for both single-hop and relaying scenarios. As expected, the fading degrades the per-
formance, especially for the CB algorithm in the relaying scenario. The main reason
lies in that the fading channel significantly corrupts the similarity between Hilbert
spectra, which leads to a severe identification performance loss for the CB algorithm.
In addition, it is noted that the FDR algorithm provides the most promising identifi-
cation performance among all the three proposed algorithms, which has a relatively
reduced performance degradation in the fading channels.

5.3.5 Conclusions
This chapter discusses two main signal identification issues in CRs, that is the
modulation classification and the SEI. New challenges to the signal identification
techniques have raised considering the real-world environments. More advanced
Signal identification in cognitive radios using machine learning 195

and intelligent theory is required to solve the blind recognition tasks. The machine-
learning theory-based algorithms are introduced to solve the modulation classification
and SEI problems. Numerical results demonstrate that the proposed algorithms
provide promising identification performance.

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Chapter 6
Compressive sensing for wireless
sensor networks
Wei Chen1

Over the past two decades, the rapid development of technologies in sensing, com-
puting and communication has made it possible to employ wireless sensor networks
(WSNs) to continuously monitor physical phenomena in a variety of applications, for
example, air-quality monitoring, wildlife tracking, biomedical monitoring and disas-
ter detection. Since the development of these technologies will continue to reduce the
size and the cost of sensors in the next few decades, it is believed that WSNs will be
involved more and more in our daily lives increasing the impact on the way we live
our lives.
A WSN can be defined as a network of sensor nodes, which can sense the physical
phenomena in a monitored field and transmit the collected information to a central
information-processing station, namely, the fusion center (FC), through wireless links.
A wireless sensor node is composed of three basic elements, i.e., a sensing unit, a
computation unit and a wireless communication unit, although the node’s physical
size and shape may differ in various applications. The rapid development of WSNs
with various types of sensors has resulted in a dramatic increase in the amount of
data that has to be transmitted, stored and processed. As number and resolution of the
sensors grow, the main constraints in the development of WSNs are limited battery
power, limited memory, limited computational capability, limited wireless bandwidth,
the cost and the physical size of the wireless sensor node. While the sensor node
is the performance bottleneck, the FC (or any back-end processor) usually has a
comparatively high-computational capability and power. The asymmetrical structure
of WSNs motivates us to exploit compressive-sensing (CS)-related techniques and to
incorporate those techniques into a WSN system for data acquisition.
CS, also called compressed sensing or sub-Nyquist sampling, was initially pro-
posed by Candès, Romberg and Tao in [1] and Donoho in [2], who derived some
important theoretical results on the minimum number of random samples needed to
reconstruct a signal. By taking advantage of the sparse characteristic of the nat-
ural physical signals of interest, CS makes it possible to recover sparse signals
from far fewer samples that is predicted by the Nyquist–Shannon sampling theorem.

1
State Key Laboratory of Rail Traffic Control and Safety, Beijing Jiaotong University, China
198 Applications of machine learning in wireless communications

The typical CS measurement of signals can be seen as a randomly weighted linear

combination of samples in some basis different from the basis where the signal is
sparse. Since the number of CS measurements is smaller than the number of elements
in a discrete signal, the task of CS reconstruction is to solve an underdetermined
matrix equation with a constraint on the sparsity of the signal in some known basis.
In contrast to the conventional data-acquisition structure, i.e., doing compression at
the sensor node and decompression at the FC, the CS process trade-off an increase in
the computational complexity of post-processing at the FC against the convenience
of a smaller quantity of data acquisition and lower demands on the computational
capability of the sensor node.
As indicated, the rapid development of WSNs with various types of sensors has
resulted in a dramatic increase in the amount of data that needs to be processed, trans-
ported and stored. CS is a promising technique to deal with the flood of data, as it
enables the design of new kinds of data-acquisition systems that combine sensing,
compression and data processing in one operation by using the so-called compressive
sensors, instead of conventional sensors. In addition, CS is a nonadaptive compres-
sion method, and the characteristics of the signals are exploited at the recovery stage.
Therefore, compressive sensors do not need the intra-sensor and inter-sensor correla-
tion information to perform compression, which facilitates distributed compression
in WSNs.
This chapter introduces the fundamental concepts that are important in the study
of CS. We present the mathematical model of CS where the use of sparse signal
representation is emphasized. We describe three conditions, i.e., the null space prop-
erty (NSP), the restricted isometry property (RIP) and mutual coherence, that are
used to evaluate the quality of sensing matrices and to demonstrate the feasibility of
reconstruction. We briefly review some widely used numerical algorithms for sparse
recovery, which are classified into two categories, i.e., convex optimization algo-
rithms and greedy algorithms. Finally, we illustrate various examples where the CS
principle has been applied to deal with various problems occurring in WSNs.

6.1 Sparse signal representation

6.1.1 Signal representation

Most naturally occurring signals, that people are interested in monitoring, have a
very high degree of redundancy of information. Therefore, by removing redundancy,
signals can be transformed to some compressed version that is convenient for storage
and transportation.
Transform compression reduces the redundancy of an n dimensional signal
f ∈ Rn by representing it with a sparse or nearly sparse representation x ∈ Rn in
some basis ∈ Rn×n , i.e.:
f = x (6.1)
Here, sparse means that all elements in vector x are zeros except for a small
number of them. Thus, we say the signal f is s-sparse if its sparse representation x
Compressive sensing for wireless sensor networks 199

has only s nonzero elements. Most naturally occurring signals are not exactly but
nearly sparse under a given transform basis, which means the values of the elements
in x, when sorted, decay rapidly to zero, or follow power-law distributions, i.e., the
ith element of the sorted representation x̀ satisfies:
|x̀i | ≤ c · i−p (6.2)
for each 1 ≤ i ≤ n, where c denotes a constant and p ≥ 1.
Transforming signals into a sparse domain has been widely used in data reduction.
For example, audio signals are compressed by projecting them into the frequency
domain, and images are compressed by projecting them into the wavelet domain and
curvelet domain. Furthermore, sometimes it is easier to manipulate or process the
information content of signals in the projected domain than in the original domain
where signals are observed. For example, by expressing audio signals in the frequency
domain, one can acquire the dominant information more accurately than by expressing
them as the amplitude levels over time. In this case, people are more interested in
the signal representation in the transformed domain rather than the signal itself in the
observed domain.

6.1.2 Representation error

By preserving a small number of the largest components of the representation, the
original signals are compressed subject to a tolerable distortion. If the signals are
exactly sparse, then perfect reconstruction of the signals is possible. However, as most
signals of interest are nearly sparse in the transformed domain, discarding the small
elements of the representations will result in a selective loss of the least significant
information. Therefore, the transform compression is typically lossy.
A representation error x − xs occurs if xs is used to recover the signal. Assume
x follows power-law distribution as in (6.2), then the squared representation error is
given by
n
n
−p 2
x − xs 2 ≤
2
c·i ≈c 2
z −2p dz
i=s+1 s+1
2
c
= ((s + 1)1−2p − n1−2p ) (6.3)
2p − 1
c0
<
(s + 1)2p−1
where c0 = c2 /(2p − 1) is a constant and p ≥ 1. Obviously, the more elements that
are kept in the representation, the smaller error the compressed signal has. It is noted
that the squared sparse representation error decays in the order of 1/(s2p−1 ).
Lossy compression is most commonly used to compress multimedia signals.
For example, the image in Figure 6.1 with 256 × 256 pixel can be represented by
coefficients of the discrete wavelet transform. By sorting the coefficients in order
of their magnitude, we note that only a small portion of the coefficients have a
large amplitude. We then choose to neglect all the coefficients whose amplitudes are
200 Applications of machine learning in wireless communications

Original image 1,000 Compressed image

800

Magnitude
600

400
Neglect 90%
200

0
0 2 4 6
DWT coef. index (sorted)
× 104

Figure 6.1 The original cameraman image vs. the compressed version

smaller than 10 and show the compressed image in the right-hand image in Figure 6.1.
The perceptual loss in quality of the compressed image compared to the original is
imperceptible.

6.2 CS and signal recovery

6.2.1 CS model
Given a signal f ∈ Rn , we consider a measurement system that acquires m (m ≤ n)
linear measurements by projecting the signal with a sensing matrix ∈ Rm×n . This
sensing system can be presented as
y = f , (6.4)
where y ∈ Rm denotes the measurement vector.
The standard CS framework assumes that the sensing matrices are randomized
and nonadaptive, which means each measurement is derived independently to the
previously acquired measurements. In some settings, it is interesting to design fixed
and adaptive sensing matrices which can lead to improved performance. More details
about the design of sensing matrices are given in [3–7]. For now, we will concentrate
on the standard CS framework.
Remembering that the signal f can be represented by an s-sparse vector x as
expressed in (6.1), the sensing system can be rewritten as
y = x = Ax, (6.5)
where A = denotes an equivalent sensing matrix. The simplified model with
the equivalent sensing matrix A will be frequently used in this dissertation unless
we need to specify the basis, not only to simplify nomenclature but also because
many important results are given in the product form of and . More gener-
ally, measurements are considered to be contaminated by some noise term n ∈ Rm
Compressive sensing for wireless sensor networks 201

owing to the sampling noise or the quantization process. Then the CS model can be
described as
y = Ax + n. (6.6)
In generally, it is not possible to solve (6.6) even if the noise term is equal to zero,
as there are infinite number of solutions satisfying (6.6). However, a suitable sparsity
constraint may rule out all the solutions except for the one that is expected. Therefore,
the most natural strategy to recover the sparse representation from the measurements
uses 0 minimization, which can be written as
min x0
x (6.7)
s.t. Ax = y.
The solution of (6.7) is the most sparse vector satisfying (6.5). However, (6.7) is a
combinatorial optimization problem and thus computationally intractable.
Consequently, as a convex relaxation of 0 minimization, 1 minimization is used
instead to solve the sparse signal representation, which leads to a linear program and
thus straight forward to solve [8]. Therefore, the optimization problem becomes:
min x1
x (6.8)
s.t. Ax = y.
This program is also known as basis pursuit (BP).
In the presence of noise, the equality constraint in (6.8) can never be satisfied.
Instead, the optimization problem (6.8) can be relaxed by using the BP de-noising
(BPDN) [9], which is
min x1
x (6.9)
s.t. Ax − y22 ≤ ε,
where ε is an estimate of the noise level. It has been demonstrated that only
m = O(s log(n/s)) measurements [10] are required for robust reconstruction in the
CS framework.
This standard CS framework only exploits the sparse characteristics of the sig-
nal to reduce the dimensionality required for sensing the signal. A recent growing
trend relates to the use of more complex signal models that go beyond the simple
sparsity model to further enhance the performance of CS. For example, Baraniuk
et al. [11] have introduced a model-based CS, where more realistic signal models
such as wavelet trees or block sparsity are leveraged in order to reduce the number of
measurements required for reconstruction. In particular, it has been shown that robust
signal recovery is possible with m = O(s) measurements in model-based CS [11].
Ji et al. [12] introduced Bayesian CS, where a signal statistical model instead is
exploited to reduce the number of measurements for reconstruction. In [13,14], recon-
struction methods have been proposed for manifold-based CS, where the signal is
assumed to belong to a manifold. Other works that consider various sparsity mod-
els that go beyond that of simple sparsity in order to improve the performance of
traditional CS include [15–18].
202 Applications of machine learning in wireless communications

6.2.2 Conditions for the equivalent sensing matrix

Another theoretical question in CS is what conditions should the equivalent sensing
matrix A satisfy in order to preserve the information in the sparse representation x.
In this subsection, three different conditions for the matrix A are presented, i.e., the
NSP, the RIP and mutual coherence. The information in the sparse representation x
is recoverable by CS if any property can be satisfied.

6.2.2.1 Null space property

For any pair of distinct s-sparse vectors, x and x , a proper equivalent sensing matrix
A must have Ax = Ax . Otherwise, it is impossible to differentiate each other from
the measurements y in conjunction with the s-sparse constraint. Since x − x could
be any 2s-sparse vector and A(x − x ) = 0, we can deduce that there exists a unique
s-sparse vector x satisfying Ax = y if and only if the null space of matrix A does not
contain any 2s-sparse vector.
This condition corresponds to the 0 norm constraint, which is used in (6.7).
However, as mentioned previously, it is painful to solve (6.7) directly. Therefore, it is
desirable to evaluate the quality of matrix A corresponding to the 1 norm operation
which is computationally tractable. Based on this consideration, we now give the
definition of the NSP as in [19].

Definition 6.1. A matrix A ∈ Rm×n satisfies the NSP in 1 of order s if and only if
the following inequality:

xJ 1 < xJ c 1 (6.10)

holds for all x ∈ Null(A) and x = 0 with J 0 = s.

The NSP highlights that vectors in the null space of the equivalent sensing matrix
A should not concentrate on a small number of elements. Based on the definition of
the NSP, the following theorem [19] guarantees the success of 1 minimization with
the equivalent sensing matrix A satisfying the NSP condition.

Theorem 6.1. Let A ∈ Rm×n . Then every s-sparse vector x ∈ Rn is the unique solution
of the 1 minimization problem in (6.8) with y = Ax if and only if A satisfies the NSP
in 1 of order s.

This theorem claims that the NSP is both necessary and sufficient for successful
sparse recovery by 1 minimization. However, it does not consider the presence of
noise as in (6.9). Furthermore, it is very difficult to evaluate the NSP condition for a
given matrix, since it includes calculation of the null space and testing all vectors in
this space.

6.2.2.2 Restricted isometry property

A stronger condition, named RIP, is introduced by Candès and Tao in [20].
Compressive sensing for wireless sensor networks 203

Definition 6.2. A matrix A ∈ Rm×n satisfies the RIP of order s with a restricted
isometry constant (RIC) δs ∈ (0, 1) being the smallest number such that
(1 − δs )x22 ≤ Ax22 ≤ (1 + δs )x22 (6.11)
holds for all x with x0 ≤ s.
The RIP quantifies the notion that the energy of sparse vectors should not be
scaled too much when projected by the equivalent sensing matrix A. It has been
established in [21] that the RIP provides a sufficient condition for exact or near exact
recovery of a sparse signal via 1 minimization.

Theorem 6.2. Let A ∈ Rm×n . Then the solution x∗ of (6.9) obeys:

x∗ − x2 ≤ c1 s−1/2 x − xs 1 + c2 ε, (6.12)
√ √ √ √
where c1 = (2 + (2 2 − 2)δ2s )/(1 − ( 2 + 1)δ2s ), c2 = (4 1 + δ2s )/(1 − ( 2 + 1)δ2s ),
and δ2s is the RIC of matrix A.

This theorem claims that with a reduced number of measurements, the recon-
structed vector x∗ is a good approximation to the original signal representation x. In
addition, for the noiseless case, any sparse representation x with support size no
√ larger
than s, can be exactly recovered by 1 minimization if the RIC satisfies δ2s < 2 − 1.
Improved bounds based on the RIP are derived in [22–24].
For any arbitrary matrix, computing the RIP by going through all possible sparse
signals is exhaustive. Baraniuk et al. prove in [10] that any random matrix whose
entries are independent identical distributed (i.i.d.) realizations of certain zero-mean
random variables with variance 1/m, e.g., Gaussian distribution and Bernoulli distri-
bution,1 satisfies the RIP with a very high possibility when the number of samples
m = O(s log(n/s)).
Note that the RIP is a sufficient condition for successful reconstruction, but it is
too strict. In practice, signals with sparse representations can be reconstructed very
well even though the sensing matrices do not satisfy the RIP.
6.2.2.3 Mutual coherence
Another way to evaluate a sensing matrix, which is not as computationally intractable
as the NSP and the RIP, is via the mutual coherence of the matrix [26], which is
given by
μ= max |AiT Aj |. (6.13)
1≤i, j≤n,i =j

Small mutual coherence means that any pair of vectors in matrix A has a low coher-
ence, which eases the difficulty in discriminating components from the measurement
vector y.

1
In most of the experiments we have conducted, we use random matrices with elements drawn from i.i.d.
Gaussian distributions, since it is the typical setting found in the literature and its performance is no worse
than one with elements drawn from i.i.d. Bernoulli distributions [25].
204 Applications of machine learning in wireless communications

Donoho, Elad and Temlyakov demonstrated in [26] that every x is the unique
sparsest solution of (6.7) if μ < 1/(2s − 1), and the error of the solution (6.8) is
bounded if μ < 1/(4s − 1). According to the relationship between the RIC and the
mutual coherence, i.e., δs ≤ (s − 1)μ [19], it is clear that if a matrix possesses a small
mutual coherence, it also satisfies the RIP condition. It means that the mutual coher-
ence condition is a stronger condition than the RIP. However, the mutual coherence is
still very attractive for sensing matrix design owing to its convenience in evaluation.

6.2.3 Numerical algorithms for sparse recovery

By applying CS, the number of samples required is reduced and the compression
operation is simpler than that for traditional compression methods. However, the
convenience of the compression operation leads to the increased complexity of the
decoding operation, i.e., the decoder requires sophisticated algorithms to recover
the original signal. Many approaches and their variants have been proposed in the
literature. Most of those algorithms can be classified into two categories—convex
optimization algorithms and greedy algorithms. Here, we briefly review some of
these convex optimization algorithms and greedy pursuit algorithms, and refer the
interested readers to the literature for other classes of algorithms including Bayesian
approaches [12,27–30] and nonconvex optimization approaches [31].

6.2.3.1 Convex optimization algorithms

Replacing the 0 norm with the 1 norm as in (6.9) is one approach to solve a convex
optimization problem that is computationally tractable. The reason for selecting the
1 norm is that the 1 norm is the closest convex function to the 0 norm, which is
illustrated in Figure 6.2. It is clear that any p norm with 0 < p < 1 is not convex,

4
L0
3.5 Lp (0<p<1)
L1
3
Lp (p>1)
2.5
L norm

1.5

0.5

0
−2 −1.5 −1 −0.5 0 0.5 1 1.5 2
x

Figure 6.2 Norm functions

Compressive sensing for wireless sensor networks 205

and the curve of the 1 norm is closer to the curve of the 0 norm than any other p
norms with p > 1.
Some equivalent formulations to (6.9) exist. For example, the least absolute
shrinkage and selection operator (LASSO) [32] instead minimizes the energy of
detection error with an 1 constraint:
min Ax − y22
x (6.14)
s.t. x1 ≤ η,
where η ≥ 0. Both BPDN and LASSO can be written as an unconstrained optimization
problem with some τ ≥ 0 for any η ≥ 0 in (6.14) and ε ≥ 0 (6.9):
1
min Ax − y22 + τ x1 . (6.15)
x 2
Note that the value of τ is an unknown coefficient to make these problems equivalent.
How to choose τ is discussed in [33].
There are several methods, such as the steepest descent and the conjugate gra-
dient, to search for the global optimal solution for these convex-relaxed problems.
Interior-point (IP) methods, developed in the 1980s to solve convex optimization, are
used in [9,34] for sparse reconstruction. Figueiredo, Nowak and Wright propose a
gradient projection approach with one level of iteration [35], while the IP approaches
in [9,34] have two iteration levels, and 1 -magic [9,34] has three iteration levels.
Other algorithms proposed to solve (6.15) include the homotopy method [36,37],
the iterative soft thresholding algorithm [38] and the approximately message passing
algorithm [39].
Generally, algorithms in this category have better performances than greedy algo-
rithms in terms of the number of measurements required for successful reconstruction.
However, their high computing complexity makes them unsuitable for applications
where high-dimensional signals are required to be reconstructed within a short time.

6.2.3.2 Greedy pursuit algorithms

While we have seen the success of convex optimization algorithms for sparse recov-
ery, greedy pursuit algorithms also attract much attention because of their simple
forms and ease of implementation. The main idea of greedy pursuit algorithms is
to iteratively compute the support of the sparse representation and so estimate the
representation corresponding to the identified support until some stopping criterion
is satisfied.
One main branch of those algorithms is known as matching pursuit (MP) [40].
The outline of MP can be described as follows:
● Step 0: Initialize the signal x = 0, the residual r = y and the index set J = ∅.
● Step 1: Find the largest coordinate i of AT r.
● Step 2: Add the coordinate to the index set J = J {i}, and calculate the
product u = rT Ai .
● Step 3: Update the residual r = r − uAi and the signal xi = xi + u.
● Step 4: Return to step 1 until the stopping criterion is satisfied.
206 Applications of machine learning in wireless communications

The disadvantage of MP is its poor performance although asymptotic conver-

gence is guaranteed. A modified version called orthogonal MPs (OMPs) is proposed
in [41]. OMP converges faster than MP by ensuring full backward orthogonality
of the error. Tropp and Gilbert claimed in [42] that OMP can reliably reconstruct
a signal with s nonzero entries out of n coordinates using O(s ln n) random linear
measurements of that signal. Comparing with BP, OMP requires more measurements
for reconstruction. Another shortcoming of OMP is that only one component of the
support of the signal is selected in each iteration. The number of iterations would
be large for recovering a signal having a lot of nonzero components. To overcome
this defect, Donoho et al. propose an alternative greedy approach, called stagewise
OMP (StOMP) [43]. Instead of choosing one component in each iteration, StOMP
selects all of the components whose values are above a specified threshold. Obviously,
StOMP is faster than OMP, and in some cases, it even outperforms BP. However, a
suitable threshold for StOMP is difficult to acquire in practice, which significantly
effects the reconstruction performance. Many other algorithms based on OMP have
been proposed in the literature such as regularized OMP [44], compressive sampling
MP [45] and subspace MP [46], all of which require prior knowledge of the number
of nonzero components.

6.3 Optimized sensing matrix design for CS

The sensing matrices used in CS play a key role for successful reconstruction in
underdetermined sparse-recovery problems. A number of conditions, such as the
NSP, the RIP and mutual coherence, have been put forth in order to study the quality
of the sensing matrices and recovery algorithms. These conditions are mainly used
to address the worst case performance of sparse recovery [25,47,48]. However, the
actual reconstruction performance in practice is often much better than the worst case
performance, so that this viewpoint can be too conservative. In addition, the worst case
performance is a less typical indicator of quality in signal-processing applications than
the expected-case performance. This motivates us to investigate the design of sensing
matrices with adequate expected-case performance. Furthermore, a recent growing
trend relates to the use of more complex signal models that go beyond the simple
sparsity model to further enhance the performance of CS. The use of additional signal
knowledge also enables one to replace the conventional random sensing matrices by
optimized ones in order to further enhance CS performance (e.g., see [3–7,49–53]).

6.3.1 Elad’s method

The mutual coherence denotes the maximum coherence between any pair of columns
in the equivalent sensing matrix A. In [49], Elad considered a different coherence
indicator, called t-averaged mutual coherence, which reflects the average coherence
between columns. The t-averaged mutual coherence is defined as the average of
all normalized absolute inner products between different columns in the equivalent
sensing matrix that are not smaller than a positive number t.
Compressive sensing for wireless sensor networks 207

Definition 6.3. For a matrix A ∈ Rm×n , its t-averaged mutual coherence is

T T

1≤i, j≤n and i =j 1( Ai Aj ≥ t) · Ai Aj
μt (A) = T , (6.16)
1( Ai Aj ≥ t)
1≤i, j≤n and i =j

where t ≥ 0, and the function 1(•) is equal to 1 if its input expression is true, otherwise
it is equal to 0.
If t = 0, the t-averaged mutual coherence is the average of all coherence between
columns. If t = μ, then the t-averaged mutual coherence μt is exactly equal to the
mutual coherence μ. Elad claimed that the equivalent sensing matrix A will have a
better performance if one can reduce the coherence of columns. Iteratively reducing
the mutual coherence by adjusting the related pair of columns is not an efficient
approach to do this since the coherence of all column pairs is not improved except
for the worst pair in each iteration. The t-averaged mutual coherence includes the
contribution of a batch of column pairs with high coherence. Thus, one can improve
the coherence of many column pairs by reducing the t-averaged mutual coherence.
Elad proposes an iterative algorithm to minimize μt (A) = μt () with respect to
the sensing matrix , assuming the basis and the parameter t are fixed and known.
In each iteration, the Gram matrix G = AT A is computed, and the values above t
are forced to reduce by multiplication with γ (0 < γ < 1), which can be expressed as
⎧
⎪
⎨ γ Gi, j |Gi, j | ≥ t
Ĝi, j = γ t · sign(Gi, j ) t > |Gi, j | ≥ γ t , (6.17)
⎪
⎩
Gi, j γ t > |Gi, j |

where sign(•) denotes the sign function. The shrunk Gram matrix Ĝ becomes full
rank in the general case due to the operation in (6.17). To fix this, the Gram matrix
Ĝ is forced to be of rank m by applying the singular value decomposition (SVD) and
setting all the singular values to be zero except for the m largest ones. Then one can
build the square root of Ĝ, i.e., ÃT Ã = Ĝ, where the square root Ã is of size m × n.
The last step in each iteration is to find a sensing matrix that makes closest to
Ã by minimizing Ã − 2F .
The outline of Elad’s algorithm is given as follows:
● Step 0: Generate an arbitrary random matrix .
● Step 1: Generate a matrix A by normalizing the columns of .
● Step 2: Compute the Gram matrix G = AT A.
● Step 3: Update the Gram matrix Ĝ by (6.17).
● Step 4: Apply SVD and set all the singular values of Ĝ to be zero except for the
m largest ones.
● Step 5: Build the square root m × n matrix Ã by ÃT Ã
= Ĝ. 2

● Step 6: Update the sensing matrix by minimizing Ã − .
F
● Step 7: Return to step 1 if some halting condition is not satisfied.
Elad’s method aims to minimize the large absolute values of the off-diagonal
elements in the Gram matrix and thus reduces the t-averaged mutual coherence. This
208 Applications of machine learning in wireless communications

method updates a number of columns at the same time in each iteration. Therefore, it
converges to a good matrix design faster than directly working on and updating the
mutual coherence iteratively. Empirical knowledge is required to determine the value
of t and γ , which affects the matrix quality and the convergence rate, respectively.

6.3.2 Duarte-Carvajalino and Sapiro’s method

In [50], Duarte-Carvajalino and Sapiro propose an algorithm to iteratively optimize
both the sensing matrix and the basis simultaneously. For any given basis, they propose
an m-step algorithm to optimize the sensing matrix. Their aim is to make the Gram
matrix of the equivalent sensing matrix as close as possible to an identity matrix,
which can be described as
G = AT A = T T ≈ In . (6.18)
T
By multiplying both sides of (6.18) with on the left and on the right,
we have
T T T ≈ T . (6.19)
Now can be decomposed to VV by eigen-decomposition, where V ∈ Rn×n
T T

is an orthonormal matrix and ∈ Rn×n is a diagonal matrix. Thus, (6.19) can be

rewritten as
VVT T VVT ≈ VVT , (6.20)
which is equivalent to
VT T V ≈ . (6.21)
After defining a matrix = V, they formulated the following optimization problem:
min − T 2F , (6.22)

and then let the sensing matrix = VT .

Duarte-Carvajalino and Sapiro select (6.18) as their optimal design without
giving a theoretical analysis, although the approach appears reasonable and the supe-
riority of the design is witnessed in their experimental results. As a closed form for its
solution cannot be determined, they propose an algorithm that requires m iterations
to determine a solution to (6.22).
The outline of their algorithm is given as follows:
● Step 0: Generate an arbitrary random matrix .
● Step 1: Apply eigen-decomposition T = VVT .
● Step 2: Let = V , Z = and i = 1.
● Step 3: Find the largest eigenvalue τ and corresponding eigenvector u of −
T + Zi ZiT √
2
F.
● Step 4: Let Zi = τ u and i = i + 1.
● Step 5: Update according to = Z−1 .
● Step 6: Return to step 2 unless i = m + 1.
● Step 7: Compute the sensing matrix = VT .
Compressive sensing for wireless sensor networks 209

6.3.3 Xu et al.’s method

Xu et al. considered the equiangular tight frame as their target design and proposed an
iterative algorithm to make the sensing matrix close to that design [51]. Tight frames
will be introduced in the next section, but for now, we only need to mention that the
equiangular tight frame is a class of matrix with some good properties, for example,
if the coherence value between any two columns of the equivalent sensing matrix A
is equal, then A is an equiangular tight frame.
Although the equiangular tight frame does not exist for any arbitrary selection of
dimensions and finding the equiangular tight frame for given dimensions is difficult,
the achievable lower bound of the Gram matrix of the equiangular tight frame, i.e.,
G = AT A, has been derived [54]:

n−m
|Gi, j | ≥ for i = j. (6.23)
m(n − 1)
Being aware of the difficulty in generating equiangular tight frames, Xu et al. propose
an optimization approach that iteratively makes the Gram matrix of the equivalent
sensing matrix close to the lower bound in (6.23). In each iteration, a new Gram
matrix is calculated by
⎧
⎪
⎪ 1 i=j
⎪
⎪
⎪
⎨ n−m
Gi, j |Gi, j | <
Ĝi, j = m(n − 1) . (6.24)
⎪
⎪
⎪
⎪ n − m n − m
⎪
⎩ · sign(Gi, j ) |Gi, j | ≥
m(n − 1) m(n − 1)
Then they update the Gram matrix by
G = αGprev + (1 − α)Ĝ, (6.25)

where Gprev denotes the Gram matrix in the previous iteration, and 0 < α < 1 denotes
the forgetting parameter. Then they update the sensing matrix as the matrix with
the minimum distance to the Gram matrix, given by

min T T − G2F , (6.26)

which can be solved using QR factorization with eigen-decomposition. As the Gram

matrix is forced to the bound of the equiangular tight frame in each iteration, they
expect the equivalent sensing matrix in turn to be close to an equiangular tight frame.
The outline of their algorithm is given as follows:
● Step 0: Generate an arbitrary random matrix .
● Step 1: Compute the Gram matrix G = T T .
● Step 2: Update the Gram matrix by (6.24).
● Step 3: Update the Gram matrix by (6.25).
● Step 4: Apply SVD and set all the singular values of G to be zero except for the
m largest ones.
210 Applications of machine learning in wireless communications

●
= G. 2
Step 5: Build the square root m × n matrix Ã by ÃT Ã

● Step 6: Update the sensing matrix by minimizing Ã − .
F
● Step 7: Return to step 1 if some halting condition is not satisfied.
Xu et al. make the equiangular tight frame lower bound as the target of their
design, as the equiangular tight frame has minimum mutual coherence [51]. However,
the lower bound can never be achieved if an equiangular tight frame for dimensions
m × n does not exist. Although the design target is based on knowledge of the bound,
an improved performance has been shown for arbitrary dimensions.

6.3.4 Chen et al.’s method

While previous work considers iterative methods, Chen et al. proposed noniterative
methods that use tight frame principles for CS sensing matrix design. For finite-
dimensional real spaces, a frame can be seen as a matrix ∈ Rm×n such that for any
vector z ∈ Rm :
az22 ≤ T z22 ≤ bz22 , (6.27)
where a > 0 and b > 0 are called frame bounds. Tight frames are a class of frames
with equal frame bounds, i.e., a = √ b. Any tight frame can be scaled with the frame
bound equal to 1 by multiplying 1/ a. A tight frame is called a Parseval tight frame
if its frame bound is equal 1, i.e.:
T z22 = z22 , (6.28)
for any z. A tight frame represents a matrix whose coherence matrix is as close as
possible to an orthonormal matrix corresponding to the Frobenius norm. Tight frames
have been widely used in many applications such as denoising, code division multiple
access systems and multi-antenna code design. Equal-norm tight frames require one
more condition than a general tight frame, i.e., the 2 norm of all columns are equal.
If all the columns of a tight frame are equal to 1, it is called a unit-norm tight frame.
In [4], Chen et al. proposed the use of unit-norm tight frames, which has been
justified from optimization considerations and has been shown to lead to MSE per-
formance gains when used in conjunction with standard sparse recovery algorithms.
Finding a matrix, which has minimum mutual coherence and columns with 2 norm
equal to 1, is equivalent to finding n points on the sphere in Rm so that the points are
as orthogonal to each other as possible. However, the equiangular tight frame does not
exist for any arbitrary selection of dimensions and finding the equiangular tight frame
for any given dimension is in general very difficult. An alternative approach is to find
the equilibrium of the points on the sphere under some applied “force.” Equilibrium
in this case means that in such a state, the points will return to their original positions
if slightly disturbed.
In [3], Chen et al. proposed a closed-form sensing matrix design, given an
over-complete dictionary . The proposed sensing matrix design is given by
ˆ = U ˆ ˆ T U
T ,
(6.29)
Compressive sensing for wireless sensor networks 211

where U ˆ is an arbitrary orthonormal matrix, = Diag (1/λ

m ), . . . , (1/λ1 )
Om×(n−m) and = Jn . It uncovers the key operations performed by this optimal
sensing matrix design. In particular, this sensing matrix design (i) exposes the modes
(singular values) of the dictionary; (ii) passes the m strongest modes and filters out the
n − m weakest modes and (iii) weighs the strongest modes. This is also accomplished
by taking the matrix of right singular vectors of the sensing matrix to correspond to
the matrix of left singular vectors of the dictionary and taking the strongest modes of
the dictionary. It leads immediately to the sensing matrix design, which is consistent
with the sensing cost constraint ˆ 2F = n, as follows:
√ √
nˆ n ˆ Jn U
T
= = . (6.30)
ˆ F
ˆ F
The sensing matrix design can be generated as follows:
● Step 0: For a given n × k dictionary , perform SVD, i.e., = U

V
T
(λ
1 ≥ · · · ≥ λn ).

● Step 1: Build an m × n matrix whose diagonal entries are λ i = (1/λi )

(i = 1, . . . , m) and all the other entries are zeros.

● Step 2: Build the sensing matrix = U ˆ U
T , where U ˆ is an arbitrary
orthonormal matrix. √
● Step 3: Normalize the sensing matrix energy using = ( n/F ).
In [5], Ding, Chen and Wassell proposed sensing matrix designs for tensor CS.
Previous study concerning CS implicitly assume the sampling costs for all samples
are equal and suggest random sampling as an appropriate approach to achieve good
reconstruction accuracy. However, this assumption does not hold for applications such
as WSNs which have significant variability in sampling cost owing to the different
physical conditions at particular sensors. To exploit this sampling cost nonuniformity,
Chen et al. proposed cost-aware sensing matrix designs that minimize the sampling
cost with constraints on the regularized mutual coherence of the equivalent sensing
matrix [6,7].

6.4 CS-based WSNs

The CS principle can be applied to various practical applications where the sensing
matrices represent different systems. In this section, we only illustrate five examples
where the CS principle is used to deal with different problems occurring in WSNs,
although CS can address a much wider range of applications.

6.4.1 Robust data transmission

Data loss in WSNs is inevitable due to the wireless fading channel. Various error-
detection and error-correction schemes have been proposed to fight against channel
fading in the physical layer. In addition, to guarantee reliable data transmission,
retransmission schemes are often applied in the application layer. Considering the
212 Applications of machine learning in wireless communications

poor computing capability of wireless sensor devices and the bandwidth overhead,
many error protection and retransmission schemes are not suitable for WSNs. In
addition, for small and battery-operated sensor devices, sophisticated source coding
cannot be afforded in some cases. However, as naturally occurring signals are often
compressible, CS can be viewed as a compression process. What is more interesting
is that these CS data with redundant measurements are robust against data loss, i.e.,
the original signal can be recovered without retransmission even though some data
are missing.
As shown in Figure 6.3, the conventional sequence of sampling, source coding
and channel coding is replaced by one CS procedure. Each CS measurement contains
some information about the whole signal owing to the mixture effect of the sensing
matrix. Thus, any lost measurement will not cause an inevitable information loss. With
some redundant measurements, the CS system can combat data loss and successfully
recover the original signal.
This CS-coding scheme has a low-encoding cost especially if random sampling
is used. All the measurements are acquired in the same way, and thus the number of
redundant measurements can be specified according to fading severity of the wireless
channel. In addition, one can still use physical layer channel coding on the CS mea-
surements. In this case, CS can be seen as a coding strategy that is applied at the appli-
cation layer where the signal characteristics are exploited, and also can be seen as a
replacement of traditional sampling and source-coding procedures. If channel coding
fails, the receiver is still able to recover the original signal in the application layer.
In [56], Davenport et al. demonstrate theoretically that each CS measurement
carries roughly the same amount of signal information if random matrices are used.
Therefore, by slightly increasing the number of measurements, the system is robust to
the loss of a small number of arbitrary measurements. Charbiwala et al. show that this
CS coding approach is efficient for dealing with data loss, and cheaper than several
other approaches including Reed–Solomon encoding in terms of energy consumption
using a MicaZ sensor platform [57].
Note that the fountain codes [58]—in particular random linear fountain codes—
and network coding [59] can also be used to combat data loss by transmitting mixed

Physical Source Channel Channel Channel Source

Sampling
signal coding coding decoding decoding

(a)

Physical Compressive Channel Compressive

signal sensing decoding

(b)

Figure 6.3 Conventional transmission approach vs. CS approach: (a) the

conventional sequence of source and channel coding and (b) joint
source and channel coding using CS
Compressive sensing for wireless sensor networks 213

symbols, which are “equally important.” The number of received symbols for both
approaches should be no smaller than the original number of symbols for decoding,
which is not necessary in the CS-based approach owing to the use of the sparse signal
characteristic.

6.4.2 Compressive data gathering

Typical WSNs consist of a large number of sensors distributed in the field to collect
information of interest for geographical monitoring, industrial monitoring, security
and climate monitoring. In these WSNs, signals sensed in the physical field usually
have high spatial correlations. Generally, it is difficult to compress the signals at
the sensor nodes due to their distributed structure. However, by exploiting the CS
principle, the signals can be gathered and transmitted in an efficient way.

6.4.2.1 WSNs with single hop communications

The proposed architecture in [60] for efficient estimation of sensor field data con-
siders single hop communications between n sensor nodes and an FC as shown in
Figure 6.4. Each sensor has a monitored parameter fi (i = 1, . . . , n) to report. The
conventional method to transmit the total number of parameters, n, requires n time
slots by allocating one time slot to each sensor node, while the new strategy only
needs m (m < n) time slots to transmit all the information.
In the new strategy, each sensor generates a random sequence i ∈ Rm (i =
1, . . . , n) by using its network address as the seed of a pseudorandom number gener-
ator. Each sensor sequentially transmits the product of the random sequence and the
sensed parameter fi in m time slots, while the transmission power is reduced to 1/m
of its default value in each time slot. All the sensors transmit in an analog fashion.
The received signal at the FC in m time slots can be written as

y = H ◦ f + n, (6.31)

Fusion center

Sensor nodes

Figure 6.4 A WSN with single hop communication

214 Applications of machine learning in wireless communications

where ∈ Rm×n denotes the random matrix corresponding to n different random

sequences, H ∈ Rm×n denotes the channel path gain matrix, and n denotes the noise
term. It is assumed that the channel path gain hi, j can be calculated using:
1
hj,i = α/2
, (6.32)
di
where di is the distance between the ith sensor and the FC, and α is the propagation
loss factor. It is also assumed that the FC knows the network address of each sensor
node and its distance to each sensor node.
Authors in [61] propose a method exploiting both the intra-sensor and inter-
sensor correlation to reduce the number of samples required for reconstruction of the
original signals. In [62], a sampling rate indicator feedback scheme is proposed to
enable the sensor to adjust its sampling rate to maintain an acceptable reconstruction
performance while minimizing the number of samples. In [6], authors propose a cost-
aware activity-scheduling approach that minimizes the sampling cost with constraints
on the regularized mutual coherence of the equivalent sensing matrix.
As the sensed field signal has high spatial correlation, it has a sparse representa-
tion in some basis. In [63], different 2-D transformations which sparsify the spatial
signals are discussed with real data. In [64], the authors use principal component
analysis to find transformations that sparsify the signal. Then, by exploiting the CS
principle, the FC can recover the original signal f from the received signal y.
Note that multiterminal source coding can also be used for joint decoding multi-
ple correlated sources, where statistical correlation models are considered typically.
However, the CS-based approach relies on a sparse transform, e.g., wavelet transform,
which is appropriate for a specific class of signals.

6.4.2.2 WSNs with multi-hop communications

Sensor readings in some WSNs are transmitted to the FC through multi-hop routing.
In [65], Luo et al. propose a compressive data gathering (CDG) scheme to reduce
the communication cost of WSNs using multi-hop transmissions. A chain-type WSN,
with n sensor nodes as shown in Figure 6.5(a), requires O(n2 ) total message transmis-
sions in the network and O(n) maximum message transmissions for any single sensor
node. On the other hand, in the CDG scheme shown in Figure 6.5(b), the jth sen-
sor nodetransmits m (m < n) messages that are the sum of the received message
j−1
vector i=1 φ i f i and its own message vector φ j fj generated by multiplying its
monitored element fj with a spreading code φ j ∈ Rm . By exploiting the CS principle,
the CDG scheme only requires O(mn) total message transmissions in the network and
O(m) maximum message transmissions for any single node.
This CDG scheme can also be applied to a tree-type topology [65]. Although the
transmission cost for the tree-type topology is different to the chain-type topology,
the superiority of the CDG scheme over the baseline scheme remains [65].

6.4.3 Sparse events detection

Another important usage of WSNs is to detect anomalies. For example, WSNs can be
used to detect poisonous gas or liquid. Not only an alarm concerning the leakage but
Compressive sensing for wireless sensor networks 215

also the leaking positions and the volumes of the leaks are reported to the monitoring
system. Other anomalies, such as abnormal temperature, humidity and so on, can
also be detected by WSNs. All of these anomaly detection problems can be analyzed
using the same model [66–69].
As shown in Figure 6.6, the n grid intersection points denote sources to be mon-
itored, the m yellow nodes denote sensors, and the s red hexagons denote anomalies.
The monitored phenomena is modeled as a vector x ∈ Rn where xi denotes the value at
the ith monitored position. The normal situation is represented by xi = 0 and xi = 0
represents the anomaly. The measurements of the sensors are denoted by a vector
y ∈ Rm where yj represents the jth sensor’s measurement. The relationship between
the events x and measurements y can be written as

y = Ax + n, (6.33)

f1
...
f1
( f1 ) f2 fn
... Fusion
center
1 2 3 n
(a)

Φ1 f1 Φ1 f1 + Φ2 f2 ∑ ni=1 Φi fi
... Fusion
1 2 3 n center

(b)

Figure 6.5 Chain-type WSNs: (a) baseline data gather and (b) compressive data
gather

Figure 6.6 A WSN for sparse event detection

216 Applications of machine learning in wireless communications

where n ∈ Rm is the noise vector. The channel response matrix is denoted by A ∈

Rm×n . The influence of the ith source to the jth sensor can be calculated as
|gi, j |
Ai, j = α/2
, (6.34)
di, j
where di, j is the distance from the ith source to the jth sensor, α is the propagation
loss factor and gi, j is the Rayleigh fading factor.
Assume that the total number of sources n is large while the number of anomalies
s is relatively very small, i.e., (s n). The number of sensors m follows s < m n.
Therefore, to solve x from a reduced number of measurements, y turns out to be a
typical sparse recovery problem. Consequently, WSNs are able to accurately detect
multiple events at high spatial resolution by using measurements from a small number
of sensor devices.
Various research have been performed using this framework. Meng, Li and Han
formulate the problem for sparse event detection in WSNs as a CS problem in [66],
where they assume the events are binary in nature, i.e., either xi = 1 or xi = 0. Ling
and Tian develop a decentralized algorithm for monitoring localized phenomena in
WSNs using a similar CS model [67]. Zhang et al. rigorously justify the validity of the
formulation of sparse event detection by ensuring that the equivalent sensing matrix
satisfies the RIP [68]. Liu et al. further exploit the temporal correlation of the events
to improve the detection accuracy [69].

6.4.4 Reduced-dimension multiple access

In WSNs, medium access control (MAC) plays an important role in data transmission,
where n sensor nodes share a wireless channel with m (m < n) degree of freedom.
Uncoordinated channel access from different sensor nodes will lead to packet colli-
sions and retransmissions, which reduces both the efficiency of bandwidth usage and
the lifetime of the sensor nodes. The simplest MAC protocols in WSNs are designed
to avoid simultaneous transmissions. For example, the 802.11 protocol protects a
communication link by disabling all other nodes using request-to-send and clear-
to-send messages. Simultaneous transmissions can be realized by using multiuser
communication techniques in the physical layer, which will result in an improved net-
work throughput performance. One of these techniques is multiuser detection, where
the received signal consists of a noisy version of the superposition of a number of
transmitted waveforms, and the receiver has to detect the symbols of all users simul-
taneously. However, multiuser detectors in general have high complexities, as the
number of correlators used at the receiver’s front-end is equal to the number of users
in the system.
In WSNs, the number of active sensor nodes s at any time is much smaller than
the total number of sensor nodes n, which can be exploited to reduce the dimension
in multiuser detection. We assume that the duration of the period of one time frame
is less than the coherence time of both the monitored signal and the wireless channel,
i.e., both the environment and the channel remain static in one time frame. Each time
frame can be divided into m time slots, in which sensor nodes can transmit one symbol.
Compressive sensing for wireless sensor networks 217

Each sensor node has a unique signature sequence φ i ∈ Rm , which is multiplied by

its transmitted symbol. The received signal y ∈ Rm in one time frame at the receiver
is given by

n
y= φ i hi fi + n, (6.35)
i=1

where hi and fi denote the channel gain and transmitted symbol corresponding to the ith
sensor node, and n ∈ Rm denotes a white Gaussian noise vector. We assume that both
the channel gains hi (i = 1, . . . , n) and sensor signature sequences φ i (i = 1, . . . , n)
are known at the receiver. It is in general impossible to solve f ∈ Rn with m received
measurements. However, as there are very few active sensor nodes in each time
frame, the transmitted symbols can be reconstructed exploiting CS reconstruction
algorithms.
This reduced-dimension MAC design has been proposed in WSN applications to
save channel resource and power consumption [70,71]. Various linear and nonlinear
detectors are given and analyzed by Xie, Eldar and Goldsmith [72]. In [73], in addition
to the sparsity of active sensor nodes, the authors exploit additional correlations that
exist naturally in the signals to further improve the performance in terms of power
efficiency.

6.4.5 Localization
Accurate localization is very important in many applications including indoor
location-based services for mobile users, equipment monitoring in WSNs and radio
frequency identification-based tracking. In the outdoor environment, the global posi-
tioning system (GPS) works very well for localization purpose. However, this solution
for the outdoor environment is not suitable for an indoor environment. For one thing,
it is difficult to detect the signal from the GPS satellites in most buildings due to the
penetration loss of the signal. For the other thing, the precision of civilian GPS is
about 10 m [74], while indoor location-based services usually requires a much higher
accuracy than GPS provides.
Using trilateration, the position of a device in a 2-D space can be determined by
the distances from the device to three reference positions. The precision of localization
can be improved by using an increased number of distance measurements, which are
corrupted by noises in real application. One localization technique considered for the
indoor environment uses the received signal strength (RSS) as a distance proxy where
the distance corresponding to a particular RSS value can be looked up from a radio
map on the server. However, the RSS metric in combination with the trilateration
is unreliable owing to the complex nature of indoor radio propagation [75]. Another
approach is to compare the online RSS readings with off-line observations of different
reference points, which is stored in a database. The estimated position of a device is a
grid point in the radio map. However, owing to the dynamic and unpredictable nature
of indoor radio propagation, accurate localization requires a large number of RSS
measurements. CS can be used to accurately localize a target with a small number of
RSS measurements, where the sparsity level of the signal representation is equal to 1.
218 Applications of machine learning in wireless communications

The efficiency of this CS-based localization system is demonstrated in [75].

Further improved localization systems and multiple target localization systems based
on the CS principle are proposed in [75–77].

6.5 Summary
This chapter reviews the fundamental concepts of CS and sparse recovery. Particularly,
it has been shown that compressively sensed signals can be successfully recovered
if the sensing matrices satisfy any of the three given conditions. The focus of this
chapter is on the applications in WSNs, and five cases in WSNs are presented where
the CS principle has been used to solve different problems. There are many new
emerging directions and many challenges that have to be tackled. For example, it
would be interesting to study better signal models beyond sparsity, computational
efficient algorithms, compressive information processing, data-driven approaches,
multidimensional data and so on.

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Chapter 7
Reinforcement learning-based channel sharing
in wireless vehicular networks
Andreas Pressas1 , Zhengguo Sheng1 , and Falah Ali1

In this chapter, the authors study the enhancement of the proposed IEEE 802.11p
medium access control (MAC) layer for vehicular use by applying reinforcement
learning (RL). The purpose of this adaptive channel access control technique is
enabling more reliable, high-throughput data exchanges among moving vehicles for
cooperative awareness purposes. Some technical background for vehicular networks is
presented, as well as some relevant existing solutions tackling similar channel sharing
problems. Finally, some new findings from combining the IEEE 802.11p MAC with
RL-based adaptation and insight of the various challenges appearing when applying
such mechanisms in a wireless vehicular network are presented.

7.1 Introduction
Vehicle-to-vehicle (V2V) technology aims to enable safer and more sophisti-
cated transportation starting with minor, inexpensive additions of communication
equipment on conventional vehicles and moving towards network-assisted fully
autonomous driving. It will be a fundamental component of the intelligent trans-
portation services and the Internet of Things (IoT). This technology allows for the
formation of vehicular ad hoc networks (VANETs), a new type of network which
allows the exchange of kinematic data among vehicles for the primary purpose of
safer and more efficient driving as well as efficient traffic management and other
third-party services. VANETs can help minimize road accidents and randomness in
driving with on-time alerts as well as enhance the whole travelling experience with new
infotainment systems, which allow acquiring navigation maps and other information
from peers.
The V2V radio technology is based on the IEEE 802.11a stack, adjusted for low
overhead operations in the dedicated short-range communications (DSRCs) spec-
trum (30 MHz in the 5.9 GHz band for Europe). It is being standardized as IEEE
802.11p [1]. The adjustments that have been made are mainly for enabling exchanges

1
Department of Engineering and Design, University of Sussex, UK
226 Applications of machine learning in wireless communications

without belonging in a basic service set (SS) (BSS). Consequently, communication

via IEEE 802.11p is not managed by a central access point (AP) as in typical wireless
LANs (WLANs). This allows faster, ad hoc communication, necessary for mission
critical applications.
Enabling such applications for various traffic scenarios to prevent safety issues
is a challenging topic to deal with in VANETs; since the bandwidth is limited, the
topology is highly mobile, and there is a lack of central coordination. Additionally,
a significant amount of data would have to be exchanged via DSRC links in dense,
urban scenarios. Every vehicle has in-car sensor and controller networks, collecting
kinematic, engine, safety, environmental and other information and passing some of
them to on-board units (OBUs) to be exchanged via DSRC links. In this chapter, we
look into the various subsystems for wireless vehicular networks and also suggest a
new RL-based protocol, first presented in [2] to efficiently share the DSRC control
channel (CCH) for safety communications among multiple vehicle stations. We begin
from how the data is generated, proceed to presenting the DSRC stack for transmitting
the vehicle’s information via wireless links, and then go on to present a self-learning
MAC protocol that is able to improve performance when network traffic becomes too
heavy for the baseline DSRC MAC (i.e. in urban scenarios, city centres).

7.1.1 Motivation
VANETs are the first large-scale network to operate primarily on broadcast transmis-
sions, since the data exchanges are often relevant for vehicles within an immediate
geographical region of interest (ROI) of the host vehicle. This allows the transmission
of broadcast packets (packets not addressed to a specific MAC address), so that they
can be received from every vehicle within range without the overhead of authentica-
tion and association with an AP. Broadcasting has always been controversial for the
IEEE 802.11 family of protocols [3] since they treat unicast and broadcast frames
differently. Radio signals are likely to overlap with others in a geographical area,
and two or more stations will attempt to transmit using the same channel leading
to contention. Broadcast transmissions are inherently unreliable and more prone to
contention since the MAC specification in IEEE 802.11 does not request explicit
acknowledgements (ACKs packets) on receipt of broadcast packets to avoid the ACK
storm phenomenon, which appears when all successful receivers attempt to send back
an ACK simultaneously and consequently congest the channel. This has not changed
in the IEEE 802.11p amendment.
A MAC protocol is part of the data link layer (L2) of the Open Systems Intercon-
nection model (OSI model) and defines the rules of how the various network stations
share access to the channel. The de facto MAC layer used in IEEE 802.11-based net-
works is called carrier sense multiple access (CSMA) with collision avoidance (CA)
(CSMA/CA) protocol. It is a simple decentralized contention-based access scheme
which has been extensively tested in WLANs and mobile ad hoc networks (MANETs).
The IEEE 802.11p stack also employs the classic CSMA/CA MAC. Although the
proposed stack works fine for sparse VANETs with few nodes, it quickly shows its
inability to accommodate increased network traffic because of the lack of ACKs.
Reinforcement learning-based channel sharing 227

The lack of ACKs not only makes transmissions unreliable but also does not provide
any feedback mechanism for the CSMA/CA backoff mechanism. So it cannot adapt
and resolve contention among stations when the network is congested.
The DSRC operation requires that L1 and L2 must be built in a way that they
can handle a large number of contenting nodes in the communication zone, on the
order of 50–100. The system should not collapse from saturation even if this number
is exceeded. Useful data for transportation purposes can be technical (i.e. vehicular,
proximity sensors, radars), crowd-sourced (i.e. maps, environment, traffic, parking)
or personal (i.e. Voice over Internet Protocol (VoIP), Internet radio, routes). We believe
that a significant part of this data will be exchanged throughV2V links, making system
scalability a critical issue to address. There is a need for an efficient MAC protocol
for V2V communication purposes that adapts to the VANET’s density and transmitted
data rate, since such network conditions are not known a priori.

7.1.2 Chapter organization

The chapter is organized as follows: Section 7.2 refers to the architecture and various
components of vehicular networks, from in-vehicle to V2V and vehicle-to-anything
(V2X). Section 7.3 introduces the DSRC networking stack for V2V and Section 7.4
gets into the intrinsics of the DSRC channel access control protocol. Section 7.5
is an overview of the congestion problem in such networks and methods of resolu-
tion. Section 7.6 is an introduction to learning-based networking protocols, Markov
decision processes (MDPs) and Q-learning. Section 7.7 describes the operation of a
novel-proposed Q-learning-based channel access control protocol for DSRC. Finally,
Sections 7.8 and 7.9 introduce a simulation environment for wireless vehicular net-
works and present the protocol performance evaluation regarding packet delivery and
achieved delay.

7.2 Connected vehicles architecture

The various electronic systems incorporated on a modern-vehicle-enable connected
vehicles, since these provide the data input (sensors), actuators as well as the local
processing and communication capabilities.

7.2.1 Electronic control units

Electronic control, rather than purely mechanical control, governs every function in
modern vehicles from activating various lights to adaptive cruise control. Electronic
control units (ECUs) are embedded systems that collect data from the vehicle’s sensors
and perform real-time calculations on these. Then they drive various electronic sys-
tems/actuators accordingly so which maximum driving efficiency can be achieved at
all times. Each unit works independently, running its own firmware, but cooperation
among ECUs can be done if needed for more complex tasks.
228 Applications of machine learning in wireless communications

7.2.2 Automotive sensors

Vehicle Sensors are components that enable autonomous vehicles applications as well
as advanced driver assistance/safety systems. One type of sensor that is commonly
deployed for such application is radar sensors, which use radio waves to detect nearby
objects and determine their relative position and velocity [4]. In actual applications,
usually an array of radar sensors is placed on the vehicle. Cameras are another option
for object detection and are used as sensing inputs for safety and driver-assistance
applications. Radars, as mentioned, have the ability to detect distance and velocity,
but cameras have a better angle of detection. These sensors are often used together to
provide more accurate-detection capabilities. Radars are usually employed as primary
sensors, with cameras extending the area of detection on the sides, for enhanced,
reliable object detection.

7.2.3 Intra-vehicle communications

The various electronic subsystems incorporated on a vehicle have to communi-
cate with each other so that sensor readings, control signals and other data can be
exchanged to perform complex tasks. This is a challenging research task, considering
the progressive advancements in the car electronics density. A typical car would fea-
ture around 8–10 ECUs in the early 1990s, around 50 ECUs at 2000, and today it is
common to have around 100 ECUs exchanging up to 2,500 signals between them [5].
One can understand the complexities involved in designing a network of ECUs that
dense, and the need for advancements in intra-vehicle networks. Table 7.1 presents
the most common in-vehicle communication protocols.

7.2.4 Vehicular ad hoc networks

The technological advances of the past few years in the field of communications,
regarding both software and hardware, are enablers of new types of networks targeted
for previously unexplored environments. The VANET is a type of wireless networks
that has received a lot of interest from researchers, standardization bodies and develop-
ers the past few years, since it has the potential to improve road safety, enhance traffic
and travel efficiency as well as make transportation more convenient and comfortable
for both drivers and passengers [6]. It is envisioned to be a fundamental building block
of intelligent transport services (ITS), the smart city as well as the IoT. VANETs are
self-organized networks composed of mobile and stationary nodes connected with

Table 7.1 Network protocols used in automotive networks

Protocol Max. bit-rate Medium Protocol

CAN 1 Mbps Shielded twisted pair CSMA/CD

LIN 20 kbps Single wire Serial
FlexRay 2×10 Mbps Twisted pair/fibre optic TDMA
MOST 150 Mbps Fibre optic TDMA
Ethernet 1 Gbps Coaxial/twisted pair/fibre optic CSMA/CD
HomePlug AV >150 Mbps Vehicular power lines CSMA/CA
Reinforcement learning-based channel sharing 229

wireless links [7]. They are a subclass of MANETs, but in this case, the mobile sta-
tions are embedded on vehicles and the stationary nodes are roadside units (RSUs).
There are more differences from classic MANETs, since VANETs are limited to road
topology while moving, meaning that potentially we could predict the future positions
of the vehicles to be used for, i.e. better routing and traffic management. Addition-
ally, since vehicles do not have the energy restrictions of typical MANET nodes, they
can feature significant computational, communication and sensing capabilities [8].
Because of these capabilities and opportunities, many applications are envisioned for
deployment on VANETs, ranging from simple exchange of status or safety messages
between vehicles to large-scale traffic management, Internet Service provisioning
and other infotainment applications.

7.2.5 Network domains

The VANET system architecture comprises three domains: the in-vehicle, the ad hoc
and the infrastructure domain, as seen in [9]. The in-vehicle domain (whose com-
ponents are described in Sections 7.2.1 and 7.2.3) is composed of an on-board
communication unit (OBU) and multiple control units. The connections between them
are usually wired, utilizing the protocols in Section 7.2.3, and sometimes wireless.
The ad hoc domain is composed of vehicles equipped with such OBUs and RSUs.
The OBUs can be seen as the mobile nodes of a wireless ad hoc network, and likewise
RSUs are static nodes. Additionally, RSUs can be connected to the Internet via gate-
ways, as well as communicate with each other directly or via multi-hop. There are two
types of infrastructure domain access, RSUs and hot spots (HSs). These provide
OBUs access to the Internet. In the absence of RSUs and HSs, OBUs could also
use cellular radio networks (GSM, GPRS, LTE) for the same purpose. The various
networking domains and their respective components can be seen in Figure 7.1.

7.2.6 Types of communication

In-vehicle communication refers to a car’s various electronic controllers communi-
cating within the in-vehicle domain. The in-vehicle communication system can detect
the vehicle’s performance regarding the internal systems (electrical and mechanical)
as well as driver’s fatigue and drowsiness [10], which is critical for driver and public
safety.
In the ad hoc domain, V2V communication can provide a data-exchange plat-
form for the drivers to share information and warning messages, so as to expand
driver assistance and prevent road accidents. Vehicle-to-road infrastructure (V2I)
communication enabled by VANETs allows real-time traffic updates for drivers, a
sophisticated and efficient traffic light system as well, as could provide environmental
sensing and monitoring.
Vehicle-to-broadband cloud communication means that vehicles may commu-
nicate via wireless broadband mechanisms such as 3G/4G (infrastructure domain).
As the broadband cloud includes more traffic information and monitoring data as well
as infotainment, this type of communication will be useful for active driver assistance
and vehicle tracking as well as other infotainment services [11]. Figure 7.2 presents
the various applications envisioned for the different types of communications.
230 Applications of machine learning in wireless communications

Infrastructure domain
Vehicular cloud Internet Cellular
management provider

V2V V2I In-vehicle

Ad hoc domain

V2I

V2V
RSU

In-vehicle domain

Sensors VPLC CAN bus

OBU ECU
Cameras
Cellular

Figure 7.1 Internet of Vehicles network domains

V2B
Remote data processing
Generation of new intelligence
Vehicle tracking

Intelligent
Infotainment traffic control
Road traffic V2I
V2V
information
Data collection
Spacing Environmental/road data
Driver assistance
Lane passing dissemination
Cooperative Geo-significant
driving Vehicle condition services
Obstacle Neighbouring
detection vehicle awareness

Local data processing

Engine control
Local sensing and decision making
In vehicle

Figure 7.2 Key functionality for every type of automotive communication

Reinforcement learning-based channel sharing 231

7.3 Dedicated short range communication

The primary functionality of VANETs is advanced active road safety via V2V and V2I
communication. A vehicular safety communication network is ad hoc, highly mobile
with a large number of contending nodes. The safety messages are very short as it is
their useful lifetime relevance and must be received with high probability [12]. The
key-enabling technology, specifying Layer 1 and 2 (L1, L2) of the protocol stack used
in V2X (ad hoc domain), is DSRC. The DSRC radio technology is essentially IEEE
802.11a adjusted for low overhead operations in the DSRC spectrum (70 MHz in the
5.9 GHz band for North America). It is being standardized as IEEE 802.11p [1].
Work in [13] shows that IEEE 802.11p exhibits lower latency and higher delivery
ratio than LTE in scenarios fewer than 50 vehicles. More specifically, for smaller net-
work densities, the standard allows end-to-end delays less than 100 ms and throughput
of 10 kbps which satisfies the requirements set by active road safety applications
and few of the lightweight cooperative traffic awareness applications. However,
as the number of vehicles increases, the standard is unable to accommodate the
increased network traffic and support performance requirements for more demanding
applications.

7.3.1 IEEE 802.11p

In the architecture of classic IEEE 802.11 networks, there are three modes of
operation:

● A BSS, which includes an AP node that behaves as the controller/master station

(STA).
● The (independent BSS) IBSS, which is formed by STAs without infrastructure
(AP/s). Networks formed like this are called ad-hoc networks.
● The (Extended Service Set), which is the union of two or more BSSs connected
by a distribution system [14].

The most suitable architecture for a VANET would be the IBSS. An STA(node)
within an IBSS acts as the AP and periodically broadcasts the SSID and other infor-
mation. The rest of the nodes receive these packets and synchronize their time and
frequency accordingly. Communication can only be established as long as the STAs
belong in the same SS.
The IEEE 802.11p amendment defines a mode called “Outside the context of
BSS” in its L2, that enables exchanging data without the need for the station to
belong in a BSS, and thus, without the overhead required for these association and
authentication procedures with an AP before exchanging data.
DSRC defines seven licenced channels, as seen in Figure 7.3, each of 10 MHz
bandwidth, six service channels (SCHs) and one CCH. All safety messages, whether
transmitted by vehicles or RSUs, are to be sent in the CCH, which has to be regularly
monitored by all vehicles. The CCH could be also used by RSUs to inform approaching
vehicles of their services, then use the SCH to exchange data with interested vehicles.
232 Applications of machine learning in wireless communications

SCH SCH SCH CCH SCH SCH SCH

5.855

5.865

5.875

5.885

5.895

5.905

5.915

5.925
Figure 7.3 The channels available for 802.11p

Non-safety
WAVE stack applications Safety applications

UDP/ WSMP
Transporting layer
TCP IEEE 1609.2 (security)
IEEE 1609.3 (networking services)
Networking layer IPv6

Upper link layer IEEE 802.2 (logical link control)

IEEE 1609.4 (multichannel operation)

Lower link layer
IEEE 802.11p (MAC)

Physical layer IEEE 802.11p (PHY)

Figure 7.4 The DSRC/WAVE Protocol Stack and Associated Standards

The explicit multichannel nature of DSRC necessitates a concurrent multichannel

operational scheme for safety and non-safety applications [15]. This need is facilitated
with a MAC protocol extension by the IEEE 1609 working group, which deals with
the standardization of the DSRC communication stack between the link layer and
applications.
The IEEE 802.11p and IEEE 1609.x protocols combined form the wireless access
in vehicular environments (WAVE) stack, and they aim to enable wireless communi-
cation between vehicles for safety (via the CCH) and other purposes (via the SCH).
The complete WAVE/DSRC stack is presented at Figure 7.4.

7.3.2 WAVE Short Message Protocol

So there are two stacks supported by WAVE, one being the classic Internet Protocol
version 6 (IPv6) and a proprietary one known as WAVE Short Message Protocol
(WSMP). The reason for having two variations in the upper layers is to distinguish
the messages as high-priority/time sensitive and less-demanding transmissions such
as User Datagram Protocol (UDP) transactions.
Vehicular safety applications do not require big datagram lengths or complex
packets to be transmitted rather than very strict probability of reception and little
latency. The overhead is 11 bytes, when a typical UDP-IPv6 packet has a minimum
overhead of 52 bytes [16]. WSMP enables sending short messages while directly
manipulating the physical layer characteristics such as the transmission power and
data rate so that nearby vehicles have a high probability of reception within a set
time frame. A provider service ID field is similar to a port number in Transmission
Reinforcement learning-based channel sharing 233

WSM version Security type Channel number Data rate TX Power PSID Length DATA
1 byte 1 byte 1 byte 1 byte 1 byte 4 bytes 2 bytes variable

Figure 7.5 The format of a WSMP packet

Control Protocol (TCP)/UDP, which acts as an identity and answers which application
is a specific WSMP heading towards. To reduce latency, WSMP exchanges do not
necessitate the formation of a BSS, which is a requirement for SCH exchanges. The
WSMP format can be seen at Figure 7.5.
However, WSMP is not able to support the classic Internet applications or
exchange of multimedia, and it does not need to, since such applications are more
tolerant to delay or fluctuations in network performance. By supporting the IPv6
stack, which is open and already widely deployed, third-party internet services are
easily deployable in a vehicular environment, and the cost of deployment would be
significantly lower for private investors.

7.3.3 Control channel behaviour

The CCH is the one to facilitate safety communications through the exchange of
safety messages, while having the following link layer characteristics:
● Most of the safety applications are based on single hop communication since
they are much localized. This means that the basic DSRC communication design
does not feature any networking (packet routing) capabilities, but there are sce-
narios where multi-hop communication is of use (warning for an accident/hazard
along a highway). Some rebroadcast schemes for enhancing the broadcasting per-
formance or reliability can be found in the literature [17] but are not considered
as proper multi-hop.
● As mentioned already, safety communications are of broadcast nature, which
means that it is targeted at vehicles for where they are rather than who they are.
Additionally, channel access is not centrally managed in DSRC. Vehicular safety
communication is fully distributed.
● A major concern for DSRC is that since all DSRC-enabled vehicles and infra-
structure continuously broadcast beacon messages and event-triggered safety
messages, such a system would require special design so that it can work reliably
and efficiently in a large scale. Although safety communications are mostly single-
hop, the system is unbounded which means that V2V communication can stretch
to great distances, unlike a bounded system (cells in mobile telephony) [15].
● The CCH is to facilitate the exchange of safety messages, complying with WSMP.
Occasionally, it is used for advertising non-safety applications (by RSUs) which
take place in one of the SCHs. These are called WAVE Service Advertisement
(WSA) messages. The receiving node would get informed of the existence of
such applications and tune in the appropriate channel if it needs to make use
of these. These advertisements are generally lightweight and their effect to the
CCH’s load is insignificant [15]. Consequently, the focus when designing the CCH
characteristics is towards safety applications.
234 Applications of machine learning in wireless communications

7.3.4 Message types

Two types of (safety) WSMP messages are to be sent through the CCH by every
DSRC-enabled vehicle:

● Periodic safety messages: Cooperative awareness messages (CAMs) are broad-

cast status messages (beacons) containing the location, direction, velocity and
other kinematic information of the transmitting vehicle. These messages are
meaningful for little time, so that the receivers can predict the movement of
the sender, and after a few seconds become irrelevant. RSUs also utilize these
beacons for traffic light status and other applications. The beaconing interval is
100 ms (Fbeacon = 10 Hz).
● Event-triggered messages: Changes in the vehicle dynamics (hard breaking) or
RSU status activate the broadcasting of emergency messages containing safety
information (i.e. road accident warning, unexpected breaking ahead, slippery
road), called decentralized environmental notification messages.
● There are also non-safety communications, that can happen for file transfers
(local map updates, infotainment) or transactions (toll collection) and others.
These take place in the SCHs but are often advertised through WSA messages in
the CCH, in which every DSRC-enabled vehicle is tuned in by default.

7.4 The IEEE 802.11p medium access control

Reliable transmission and reception of messages can be affected by packet collisions.
Two or more transmitters within the range of the same receiver sending a packet
simultaneously would lead to a packet collision at the receiver, and the receiver would
not receive any message. To tackle this problem, a MAC protocol, which serves the
purpose of allowing only one node to transmit at a time, would have to be designed
and implemented. Two nearby nodes transmitting frames at the same time means that
these frames will collide leading to wasted bandwidth. A MAC protocol is a set of
rules defined in L2 by which the radio [12] (physical layer – L1) decides when to
send data and when to defer from transmission.
Given that wireless vehicular networks are ad hoc in nature, TDMA, FDMA or
CDMA are difficult to realize since some short of centralized control (AP) would be
needed to dynamically allocate time slots, frequency channels or codes, respectively.
In addition to the infrastructure-less nature of VANETs, the high degree of mobil-
ity makes these MAC protocols unsuitable for such networks [12]. Random access
mechanisms are better suited to ad hoc networks, such as ALOHA or in the case of
VANETs, CSMA.

7.4.1 Distributed coordination function

The de facto technique for sharing access to the medium among multiple nodes without
central coordination in IEEE 802.11-based networks is the distributed coordination
Reinforcement learning-based channel sharing 235

function (DCF). It employs a CSMA/CA algorithm. DCF defines two access mech-
anisms to enable fair packet transmission, a two-way handshaking (basic mode) or a
four-way handshaking (request-to-send/clear-to-send (RTS/CTS)).
Under the basic access mechanism, a node wishing to transmit would have to
sense the channel for a DCF Interframe Space (IFS) (DIFS) interval. If the channel
is found busy during the interval, the node would not transmit but instead wait for an
additional DIFS interval plus a specific period of time known as the backoff interval,
and then try again. If the channel was not found busy for a DIFS interval, the node
would transmit.
Another optional mechanism for transmitting data packets is RTS/CTS reserva-
tion scheme. Small RTS/CTS packets are used to reserve the medium before large
packets are transmitted.

7.4.2 Basic access mechanism

In a network like VANETs where many stations contend for the wireless medium, if
multiple stations sense the channel and find it busy, they would also find it being idle
virtually the same time and try to use it at that time instant. To avoid the collisions that
would occur that very moment, as seen in Figure 7.6, every station would have to wait
for a backoff interval, whose length is specified by the random backoff mechanism in
DCF. This interval is picked randomly from the uniform distribution over the interval
[0,CW] where CW is the current contention window. According to the IEEE 802.11p,
the CW can be a number between CWmin = 3 and CWmax = 255 [18].
For every DIFS interval that the node senses the medium to be idle, the backoff
timer is decreased. If the medium is used, the counter will freeze and resume when
the channel is again idle for a DIFS interval. The station whose backoff timer expires
(reaches 0) first will begin the transmission, and the other stations freeze their timers
and defer transmission. Once the transmitting station completes transmission, the
backoff process starts again and the remaining stations resume their backoff timers.

Figure 7.6 Collisions in a simulated environment

236 Applications of machine learning in wireless communications

For unicast packet transmissions, in the case of a successful reception, the desti-
nation will send an ACK to the source node after a short IFS (SIFS), so it (the ACK)
can be given priority (since SIFS<DIFS). If the source does not receive an ACK
within a set time frame, it reactivates the sending process after the channel remains
idle for an extended IFS. If two or more nodes decrease their backoff counter to 0
simultaneously, a collision occurs. For each retransmission attempt (because of colli-
sion and no ACK), the used CW is doubled, until it reaches CWmax . Upon successful
transmission, CW resets to CWmin . The operation of CSMA/CA for both unicast and
broadcast transmissions can be seen in Figure 7.7.

7.4.3 Binary exponential backoff

This mechanism of CW adaptation for unicast packets is called the binary exponential
backoff (BEB) algorithm. It is the CA part of CSMA/CA and specifies that for every
packet transmission, the station uniformly selects a random value for its backoff
counter within [0, Wi − 1], where Wi is the current CW and i is the number of failed
attempts to transmit this single packet:
Wi = 2i × CW for i ∈ [0, m], (7.1)
where the number of the backoff stages m is given by

CWmax
m = log2 (7.2)
CWmin
At the first transmission attempt for a packet:
W0 = CWmin = CW . (7.3)
If a unicast packet encounters a collision (meaning no ACK was received for a
set time frame), W1 = 2 × CW . Wi is doubled every time a collision happens, until
it reaches a Wm = CWmax = 2m CW . When Wi = Wmax , it maintains this value until a
successful transmission (ACK received). Wi will be reset to CWmin , and the process
will start again for the next unicast packet.

Unicast only
DIFS Backoff

DATA
Node A
SIFS DIFS Backoff

ACK
Node B
Time

Figure 7.7 A CSMA/CA cycle for both unicast and broadcast cases. It manages
channel access among transmitting nodes A and B
Reinforcement learning-based channel sharing 237

Two problems seem to appear with the BEB mechanism when trying to establish
unicast communication among many highly mobile nodes. First, in dense wireless
networks such as VANETs, there is higher probability that more than one node choose
the same CW, resulting to collisions. Second, every time a collision occurs, the CW
is doubled to avoid more collisions. But given that the network density for a VANET
can vary a lot over small time periods because of high mobility, a node with a large
CW (because of previous failed transmissions) will wait more than it needs to before
transmitting under lighter network conditions. This will result in unnecessary delay.

7.4.4 RTS/CTS handshake

The hidden terminal problem, shown in Figure 7.8, appears when node C wants
to transmit to node B but cannot hear that it is already occupied. This is due to
node C not being within the transmission range of node A. The RTS/CTS scheme
(also known as virtual carrier sensing) can help to reduce contention caused from
this phenomenon. An RTS packet is transmitted first from the sender, containing
the size of the upcoming, larger, data frame and the channel time which is required
for it to be transmitted. In the case that the receiving node is free to receive the
data frame, it sends a CTS packet back to the sender. The neighbouring nodes defer
from the medium until the channel becomes free again. This mechanism is helpful
when transmitting large data packets and tackles the hidden and exposed terminal
problem, but in the case of ad hoc networks it was found [19] not to be as effective.
The overhead associated with the exchange of RTS/CTS packets does not worth it
when the network has many, highly mobile stations, especially when targeting low
latency communication. Additionally, for VANETs, vehicle-stations positioned at the
edge of a station’s transmission range are not as depended on geo-significant safety
information transmitted from that station, as much as the vehicles of close proximity
to the transmitter.

A Tx range C Tx range

A B C
A transmitting

Nodes A and C are hidden from each

other with respect to B

Figure 7.8 Hidden terminal problem

238 Applications of machine learning in wireless communications

7.4.5 DCF for broadcasting

Frames exchanged in an 802.11 DCF-based network can be distinguished as unicast,
if sent to a single destination node, or broadcast if sent to all available nodes within
the transmission range. In both transmission cases, the MAC layer of the future-
transmitting node receives a request to transmit from the upper layers of the networking
stack, and then physical carrier sensing takes place to observe whether the channel
is unoccupied by another transmitter. The channel being found idle for more than
a DIFS interval means that the MAC layer of that station will begin the process of
transmission, for either unicast or broadcast packets.
For the unicast case, the DCF backoff mechanism uses multiple backoff stages,
as defined by (7.1) and (7.2). For every transmission that an ACK packet from the
destination is not received in time or at all, the transmitter’s CW is doubled, except
from the last stage in which CW stays at maximum value.
For broadcast packet transmission though, the algorithm can have only a single
backoff stage. The reason for this is that the 802.11 protocol does not require ACKs
from the destination nodes for broadcast transmission, for practical reasons. All the
receiving nodes sending an ACK on the reception of a message would impose an
overhead, causing even more collisions (among all the ACK packets) and generally
degrading the performance of the network for the given time.
So as usual, when the backoff counter expires for the first backoff stage, the
message is going to be sent but will not be acknowledged, which means there is
no definite way to know whether the packet actually reached the destination nodes.
Collisions would be unrecoverable in this case [20], since no intelligent retransmission
strategy is implemented by default for broadcasting. The different behaviours of the
backoff algorithm for unicast and broadcast traffic also lead to different contention
times [20].

7.4.6 Enhanced distributed channel access

When just the basic DCF scheme is employed, all nodes contend for access to the
medium using the CSMA/CA algorithm with the same parameters. Data packets
that are different regarding content, priority or delay-tolerance should be handled
differently, and quality of service (QoS) should be guaranteed [21]. Real-time traffic
information and collision-warning messages have strict delay requirements, while
applications such as map data downloading and Internet browsing are more time
tolerant.
In order to meet the different QoS requirements such as end-to-end delay and
throughput, traffic should be differentiated depending on such requirements. A way
of doing this service differentiation is by setting different contention parameters for
different classes of data, as seen in Table 7.2.
The IEEE 802.11p/WAVE stack can adopt the enhanced distributed channel
access from 802.11e in order to improve the QoS. It offers traffic classification
through four access categories (ACs). When packets have different ACs, they contend
internally and the winner will participate in external contention [22]. As shown on the
table below, highly important messages (safety broadcasts) fall in AC3 which has the
Reinforcement learning-based channel sharing 239

Table 7.2 Contention parameters for different access categories in 802.11p

AC Data class CW min CW max AIFS

3 Safety related 3 7 2
2 Voice 3 7 3
1 Best effect 7 15 6
0 Background traffic 15 1,023 9

Table 7.3 Typical DSRC QoS requirements as seen in [12]

Applications Packet size (bytes)/ Allowable Network Message Priority

bandwidth latency traffic range (m)
(ms) type

Intersection collision ∼100 ∼100 Event 300 Safety of

warning/avoidance life
Cooperative collision ∼100/∼10 kbps ∼100 Periodic 50–300 Safety of
warning life
Work zone warning ∼100/∼1 kbps ∼1,000 Periodic 300 Safety
Transit vehicle signal ∼100 ∼1,000 Event 300–1,000 Safety
priority
Toll collection ∼100 ∼50 Event 15 Non-safety
Service ∼100/2 kbps ∼500 Periodic 0–90 Non-safety
announcements
Movie download >20 Mbps N/A N/A 0–90 Non-safety
(2 hours of
MPEG 1): 10 min
download time

lowest arbitrary IFS (AIFS) and CW size, so they are more likely to win the internal
contention and affect transmission delay as little as possible (up to seven time slots
for unicast and up to three time slots for broadcast transmissions). The QoS require-
ments for various vehicular networking applications can be found at Table 7.3, taken
from [12].

7.5 Network traffic congestion in wireless vehicular networks

It is by now clear that the safety applications made possible through VANETs require
a low end-to-end delay and high packet delivery ratio (PDR). Additionally, since
the safety messages will be of broadcast nature, VANETs will be the first large-scale
networks where communication is based on broadcast rather than on unicast messages.
The choice of an IEEE 802.11-based technology for this kind of network raises some
issues [23]. The MAC protocol in this family of standards is well known for its
240 Applications of machine learning in wireless communications

inability to cope with large-scale broadcast communications, since it was designed

for a different use case, and it clearly favours unicast [20] communication.
Network traffic congestion in VANETs has a devastating impact on the perfor-
mance of ITS applications. Given the large number of contending nodes, especially
in an urban environment, it has been found [24] that the CSMA/CA algorithm, which
is the basic medium access scheme in the IEEE 802.11, is not reliable enough due
to high collision rates. This means channel congestion control and broadcast perfor-
mance improvements of the 802.11p MAC are of particular concern and need to be
addressed [15] in order to meet the QoS requirements of DSRC applications. The
basic reason for this is the non-adaptation of the CW size depending on sensed net-
work traffic. Work presented in [25] proves that throughput (derived from a simple
Markov chains model) is diminished with an increased number of competing nodes
exchanging broadcast packets.
The node density in a typical scenario can vary from very sparse connectivity
to more than 150 cars/lane/km [23], so VANETs have to be able to accommodate
the needs (channel-wise) of multiple simultaneous transmitters. The modifications
brought by the IEEE 802.11p amendment focused on the physical layer, while the
classic 802.11 MAC layer was enhanced for transmission of data outside BSS context
which will contribute towards the scalability goal by removing the association and
authentication overheads. But IEEE 802.11 was designed for unicast applications in
mind, so it comes as no surprise that the CCH operating under 802.11p can be saturated
solely by periodic broadcasting (beaconing), even for medium vehicle density [23].

7.5.1 Transmission power control

One idea on how to treat degrading performance on increasing vehicle density that
has been extensively studied is limiting the number of contending nodes, which can
be done by using TX power control mechanisms. When access to the wireless medium
becomes difficult, lowering the transmission power of a station reduces the interfer-
ence area [26]. WSMPs exploit this by providing the capability to set the transmission
power on a per-packet basis. There are, however, some limitations on the minimum
area that safety messages should reach.

7.5.2 Transmission rate control

Another solution, often combined with power control, is controlling the transmission
time of a beacon. Since the packets’ size is set by the application, only the data rate can
be adjusted. Higher data rate translates into higher transmission probability [27], but
also higher Signal-to-noise ratio (SNR) at the destination of the message, so the cov-
erage area is reduced. This solution suffers from the same limitation as power control.

7.5.3 Adaptive backoff algorithms

A way to operate on maximum coverage area and still avoid collisions and degrad-
ing performance would be an adaptive backoff mechanism. With a high number of
transmitting nodes, a large CW size is needed to avoid unnecessary collisions. On the
Reinforcement learning-based channel sharing 241

other hand, when the traffic load of the network is low, a small CW size is needed so
that potential senders can access the wireless medium with a short delay [18], thus
making more efficient use of channel bandwidth. Additionally, the time the channel
is idle because of nodes being in the backoff stage could be minimized. In an ideal
situation, there would be zero idle time (which is essentially lost and is a synonym of
bandwidth wastage) between messages with the exception of the DIFS [23].

7.6 Reinforcement learning-based channel access control

Machine-learning-based techniques have the potential to enter and improve every

layer of the network stack for the IoT and other applications. The focus of this chapter
is on RL [28] in the context of MAC for wireless V2V/V2X communication in the
ad-hoc domain of vehicular networks. An adaptive backoff algorithm based on RL
can help tackle the channel congestion when many stations are deployed, without
reducing the transmission range or data rate, or knowing any details about the network
beforehand.
RL is a general class of machine-learning algorithms fit for problems of sequential
decision-making and control. It can be used as a parameter-perturbation/adaptive-
control method for MDPs [29], a discrete time stochastic control formulation. It is
based on the idea that if an action is followed by a satisfactory state of affairs, or by
an improvement in the state of affairs (or a reward function), then the agent’s ten-
dency to produce that action is strengthened, i.e. reinforced. Specifically, we develop
and evaluate a solution based on Q-learning, a much-used model-free RL algorithm
that can solve MDPs with very little information from the dynamic VANET envi-
ronment, but still reveals effective solutions regarding contention control for various
network conditions. In addition, we employ a strategy for building self-improving
Q-learning controllers that yield instant performance benefits since the vehicle-
station’s deployment and always strive for optimum operation while online.

7.6.1 Review of learning channel access control protocols

When it comes to relevant work which is focused specifically on the MAC layer
issues, [30] uses the MDP formulation to design a MAC with deterministic backoff
for virtualized IEEE 802.11 WLANs. For V2V exchanges, the work presented in [31]
examines the IEEE 802.11p MAC regarding channel contention using the Markov
model from [32] and proposes a passive contention estimation technique by observing
the count of idle inter-frame slots.
RL is inspired by behaviourist psychology and deals with how software agents
should take actions in an environment while aiming to maximize their cumulative
reward. The problem, because of its generality, is studied in many disciplines such
as game theory, control systems, IT, simulation-based optimization, statistics and
genetic algorithms. There have been attempts to apply RL for optimizing the access
control layer of wireless networks. The protocol in [33] is targeted on wireless sensor
networks, optimizing battery-power node energy consumption. The protocol in [34]
242 Applications of machine learning in wireless communications

is targeted on wireless vehicular networks that operate on a unicast basis. It employs

CW adaptation [35] which is a proven technique to improve the network contention
because of interference in wireless networks. The premise is interesting, but the
proposed IEEE 802.11p is a broadcast-based protocol. The current literature does not
deal with the broadcasting issues within the context of contention resolution on the
MAC level.

7.6.2 Markov decision processes

In RL, the learning agents can be studied mathematically by adopting the MDP for-
malism. An MDP is defined as a (S, A, P, R) tuple, where S stands for the set of
possible states, As is the set of possible actions from state s ∈ S, Pa (s, s ) is the proba-
bility to transit from a state s ∈ S to s ∈ S by performing an action a ∈ A. Ra (s, s ) is
the reinforcement (or immediate reward), result of the transition from state s to state
s because of an action a, as seen in Figure 7.9. The decision policy π maps the state
set to the action set, π : S → A. Therefore, the MDP can be solved by discovering
the optimal policy that decides the action π(s) ∈ A that the agent will make when in
state s ∈ S.

7.6.3 Q-learning
There are, though, many practical scenarios such as the channel access control prob-
lem studied in this work, for which the transition probability Pπ (s) (s, s ) or the reward
function Rπ (s) (s, s ) are unknown, which makes it difficult to evaluate the policy π .
Q-learning [36,37] is an effective and popular algorithm for learning from delayed
reinforcement to determine an optimal policy π in the absence of transition probabil-
ity. It is a form of model-free RL which provides agents the ability to learn how to act
optimally in Markovian domains by experiencing the consequences of their actions,
without requiring maps of these domains.
In Q-learning, the agent maintains a table of Q[S, A], where S is the set of states
and A is the set of actions. At each discrete time step t = 1, 2, . . . , ∞, the agent
observes the state st ∈ S of the MDP, selects an action at ∈ A, receives the resultant
reward rt and observes the resulting next state st+1 ∈ S. This experience (st , at , rt , st+1 )
updates the Q-function at the observed state-action pair, thus providing the updated
Q(st , at ). The algorithm, therefore, is defined by the function (1) that calculates the

From s execute a with Pa(s,s')

Agent Environment
Agent in s

Transit to s'
Receive Ra(s,s')

Figure 7.9 Abstract MDP model

Reinforcement learning-based channel sharing 243

quantity of a state-action (s, a) combination. The goal of the agent is to maximize its
cumulative reward. The core of the algorithm is a value iteration update. It assumes
the current value and makes a correction based on the newly acquired information,
as in the following equation:
Q(st , at ) ← Q(st , at ) + α × [rt + γ × max Q(st+1 , at+1 ) − Q(st , at )], (7.4)
at+1

where the discount factor γ models the importance of future rewards. A factor of γ = 0
will make the agent “myopic” or short-sighted by only considering current rewards,
while a factor close to γ = 1 will make it strive for a high long-term reward. The
learning rate α quantifies to what extent the newly acquired information will override
the old information. An agent with α = 0 will not learn anything, while with α = 1, it
would consider only the most recent information. The maxat+1 ∈A Q(st+1 , at+1 ) quantity
is the maximum Q value among possible actions in the next state. In the following
sections, we present employing (7.4) as a learning, self-improving, control method
for managing channel access among IEEE 802.11p stations.

7.7 Q-learning MAC protocol

The adaptive backoff problem fits into the MDP formulation. RL is used to design a
MAC protocol that selects the appropriate CW parameter based on gained experience
from its interactions with the environment within an immediate communication zone.
The proposed MAC protocol features a Q-learning-based algorithm that adjusts the
CW size based on binary feedback from probabilistic rebroadcasts in order to avoid
packet collisions.

7.7.1 The action selection dilemma

The state space S contains the discrete IEEE 802.11p-compatible CW values rang-
ing from CWmin = 3 to CWmax = 255. The CW is adapted prior to every packet
transmission by performing one of the following actions:
a∈{(CWt −1)/2,CWt ,CWt ×2−1}
CWt+1 ←−−−−−−−−−−−−−−− CWt . (7.5)
RL differs from supervised learning in which correct input/output pairs are never
presented, nor suboptimal actions are explicitly corrected. In addition, in RL there is a
focus on online performance, which involves finding a balance between exploration of
uncharted territory and exploitation of current knowledge. This in practice translates
as a trade-off in how the learning agent in this protocol selects its next action for every
algorithm iteration. It can either randomly pick an action from (7.5) (exploration) so
that the algorithm can transit to a different (s, a) pair and get experience (reward) for
it or follow a greedy strategy (exploitation) and choose the action with the highest
Q-value for its current state given by
π (s) = argmax Q(s, a). (7.6)
a
244 Applications of machine learning in wireless communications

7.7.2 Convergence requirements

The RL algorithm’s purpose is to converge to a (near) optimum output, in terms
of CW. Watkins and Dayan [36] proved that Q-learning converges to the optimum
action-values with probability 1 as long as all actions are repeatedly sampled in all
states and the action-value pairs are represented discretely.
The greedy policy, with respect to the Q-values, tries to exploit continuously.
However, since it does not explore all (s, a) pairs properly, it fails satisfying the first
criterion. At the other extreme, a fully random policy continuously explores all (s, a)
pairs, but it will behave suboptimally as a controller. An interesting compromise
between the two extremes is the ε-greedy policy [28], which executes the greedy
policy with probability 1 − ε. This balancing between exploitation and exploration
can guarantee convergence and often good performance.
The proposed protocol uses the ε-greedy strategy to focus the algorithm’s explo-
ration on the most promising CW trajectories. Specifically, it guarantees the first
convergence criterion by forcing the agent to sample all (s, a) pairs over time with
probability ε. Consequently, the proposed algorithmic implementation satisfies both
convergence criteria, but further optimization is needed regarding convergence speed
and applicability of the system.
In practice, the Q-learning algorithm converges under different factors, depend-
ing on the application and complexity. When deployed in a new environment, the
agent should mostly explore and value immediate rewards and then progressively
show its preference for the discovered (near) optimal actions π (s) as it is becoming
more sure of its Q estimates. This can be achieved via the decay function shown in
the following equation:
Ntx
ε =α =1− for 0 ≤ Ntx ≤ Ndecay , (7.7)
Ndecay
where Ntx is the number of transmitted broadcast packets and Ndecay is a preset number
of packets that sets the decay period. This decay function is necessary to guarantee
convergence towards the last-known optimum policy in probabilistic systems such as
the proposed contention-based MAC since there is no known optimum final state. By
reducing the values of ε and α over time via (7.7), the agent is forced to progressively
focus on exploitation of gained experience and strive for a high long-term reward.
This way, when approaching the end of the decay period, the found (near) optimal
states-CW/s are revealed.

7.7.3 A priori approximate controller

The above strategy can be used to get instant performance benefits, starting from
the first transmission. This is done by preloading approximate controllers, pretrained
for different transmitted bit rates and number of neighbours via (7.7), to the station’s
memory. These controllers define an initial policy that positively biases the search
and accelerates the learning process.
The agent’s objective in this phase is to quickly populate its Q-table with values
(explore all the state-action pairs multiple times) and form an initial impression of the
Reinforcement learning-based channel sharing 245

environment. The lookup Q-table is produced by encoding this knowledge (Q-values)

for a set period of Ndecay a priori and can be used as an initial approximate controller,
which yields an instant performance benefit since the system is deployed.
Q-learning is an iterative algorithm so it implicitly assumes an initial condition
before the first update occurs. Zero initial conditions are used the very first time
the algorithm is trained on a set environment, except for some forbidden state-action
pairs with large negative values, so it does not waste iterations in which it would
try to increase/decrease the CW when it is already set on the upper/lower limit.
The algorithm is also explicitly programmed to avoid performing these actions on
exploration. The un-trained, initial Q-table is set as in (7.8), where the rows represent
the possible states – CW sizes and columns stand for the action space:
CW (CW − 1)/2 CW CW ×2 + 1
⎛ ⎞
3 −100 0 0
7 ⎜ ⎜ 0 0 0 ⎟
⎟
15 ⎜⎜ 0 0 0 ⎟
⎟
Q0 [7][3] = 31 ⎜⎜ 0 0 0 ⎟
⎟ (7.8)
63 ⎜⎜ 0 0 0 ⎟
⎟
127 ⎝ 0 0 0 ⎠
255 0 0 −100
Each station can employ various different learning, self-improving, controllers
and use the appropriate one depending on a combination of sensed density and
received bit rate. This is feasible because the station has the ability to sense the
number of one-hop neighbours since they all transmit heart-beat, status packets peri-
odically. It also does not have the memory constraints that typical sensor networks
have. An example of a controller’s table at the end of the ε decay period as in (7.7)
can be seen in (7.9). The controller has been trained a priori with γ = 0.7 and a
decay period lasting for 180 s in a 60-car network, where every car transmits 256
bytes every 100 ms. A trajectory leading to optimum/near-optimum CW/s is being
formed (depending on past experience) by choosing the maximum Q-value for every
CW-state, seen in bold font. The controller in (7.9) oscillates between the values 31
and 63 when exploiting the Q-table to find the optimum CW:
CW (CW − 1)/2 CW CW ∗ 2 + 1
⎛ ⎞
3 −100 −0.07218 0.2388
7 ⎜ ⎜ −0.076 −0.0325 0.6748 ⎟ ⎟
15 ⎜⎜ 0.198 0.28012 0.817 ⎟ ⎟
Qπ [7][3] ≈ 31 ⎜⎜ 0.2896 0.2985 0.4917 ⎟ ⎟ (7.9)
63 ⎜⎜ 0.4945 0.10115 0.2838 ⎟ ⎟
127 ⎝ 0.2043 −0.055 −0.0218 ⎠
255 0.1745 −0.86756 −100
Figure 7.10 shows the steps (in terms of CW size) the algorithm takes (explo-
ration/pretraining) until it converges to an optimal value, or more specifically
oscillates among two values (63–127) so that it can exploit (90% of the time) its
knowledge and yield performance benefits. Figure 7.11 shows how the proposed
246 Applications of machine learning in wireless communications

255 1.0
Contention window

127 0.8
63

Epsilon
0.6
31
15 0.4
7 0.2
3 0.0
0 50 100 150 200 250 300
Simulation time (s)

Figure 7.10 Trace of CW over time for a station in a 100-car network. The first
stage is the a priori controller training phase via (4) for 200 sec (or
Ndecay = 2000 original packets), then online stage for the remaining
time, with an exploration to exploitation ratio of 1:9

Mean contention window versus epsilon

127 1.0

0.8
Contention window

0.6

ε=α
63
6 × 101
0.4
0.2
4 × 101 50 cars
75 cars
100 cars
31 0.0
100 120 140 160 180 200
Simulation time (s)

Figure 7.11 Mean network-wide CW versus training time (second half) for
networks of different densities using the Q-learning-based MAC

MAC discovers the optimum CW size of the stations in three networks of different
densities.

7.7.4 Online controller augmentation

While the pretrained, preloaded, approximate controller is useful for speeding up the
learning process as well as getting an instant performance benefit, its drawback is that
by default it is not adaptive to changes in the environment while online. The online
efficiency of the Q-learning controller depends on finding the right balance between
exploitation of the station’s current knowledge and exploration for gathering new
information. This means that the algorithm must sometimes perform actions other
than the ones dictated by the current policy to update and augment that controller
with new information.
While the station is online, exploratory action selection is performed less fre-
quently (ε = 0.1) than in a priori learning (7.7) (ε starts from 1), primarily to
compensate for modelling errors in the approximate controller. This means that the
controller in its online operation uses the optimum Q-value 90% of the time and makes
Reinforcement learning-based channel sharing 247

exploratory CW perturbations 10% of the time in order to gain new experience. In

this way, the agent still has the opportunity to correct its behaviour based on new
interactions with the VANET and corresponding rewards.

7.7.5 Implementation details

In RL, the only positive or negative reinforcement an agent receives upon acting so
that it can learn to behave correctly in its environment comes in a form of a scalar
reward signal. Taking advantage of the link capacity for maximum packet delivery
(throughput) was of primary concern for this design, aiming to satisfy the requirements
of V2V traffic (frequent broadcasting of kinematic and multimedia information). For
this purpose, the reward function is based on the success of these transmissions.
Reward r can be either 1 or −1 for successful (ACK) and failed transmissions (no
ACK) correspondingly. A successful transmission from the same consecutive state –
CW – is not given any reward. The pseudo-code in Algorithm 7.1 summarizes the
operation of our proposed protocol.

Algorithm 7.1: Q-learning V2V MAC

1: Initialize Q0 (CW , A) at t0 = 0 as in (7.8)
2: procedure Action-selection(CWt ) ε-greedy
3: if pε ≤ ε then
4: at+1 ← random[ (CW2t −1) , CWt , CWt ∗ 2 − 1]
5: else if pε ≥ 1 − ε then
6: at+1 ← aπ Optimum a from (7.5)
7: end if
8: if A-priori Controller Learning then
9: ε = α → decay according to rule (7.7)
10: else if On-line Learning then
11: ε = α → constant
12: end if
13: CWt+1 ← CW at+1
14: end procedure
15: TX Broadcast Packet: MessageId Transmit
16: procedure Feedback(CWt+1 , at+1 ) Collect Reward
17: Initialize: RTT ← 0 s
18: if RX MessageId AND RTT < 0.1 s then
19: if at = (CWt+1 ←− CWt ) then
20: rt ← 1
21: end if
22: else if RTT ≥ 0.1 s then
23: rt ← −1
24: end if
25: end procedure
26: Update Q(CWt+1 , at+1 ) according to rule (7.4)
27: GOTO 2
248 Applications of machine learning in wireless communications

The first step of the MAC protocol would be to set the default CW of the station
to the minimum possible value, which is suggested by the IEEE 802.11p standard.
After that, the node makes an exploratory move with probability ε (exploration) or
picks the best known action to date (highest Q value) with probability 1 − ε.
Received packet rebroadcasts can be used as ACKs since some will definitely be
overheard from the source vehicle, even assuming that they move at the maximum
speed limit. These rebroadcasts can happen for forwarding purposes, and they enhance
the reliability of the protocol, since the original packet senders can detect collisions
as well as provide a means to reward them if they succeed in broadcasting a packet.
We use probabilistic rebroadcasting for simplicity, but various routing protocols can
be used instead.
Every time a packet containing original information is transmitted, a timer is
initiated which waits for a predefined time for an overheard retransmission of that
packet, which will have the same MessageId. These broadcast packets are useful
for a short lifetime, which is the period between refreshes. So a rebroadcast packet,
received after that period, is not considered to be a valid ACK because the information
will not be relevant any more, since the nodes in VANETs attempt to broadcast fresh
information frequently (i.e. 1–10 Hz).

7.8 VANET simulation modelling

A VANET simulation has two main components; a network component as described

above, which must have the capability to simulate the behaviour of communication
networks as well as a vehicular traffic component which provides accurate enough
mobility patterns for the nodes of such a network (vehicles/cars).

7.8.1 Network simulator

There are a few software environments for simulating a wireless network [38], of
which OMNeT++ 4.6 is chosen for its many available models, maturity and advanced
GUI capabilities. OMNeT++ [39] is a simulation platform written in C++ with a
component-based, modular and extensible architecture.
The basic entities in OMNeT++ are simple modules implemented in C++. Com-
pound modules can be built of simple modules as well as compound modules. These
modules can be hosts, routers, switches or any other networking devices. Modules
communicate with each other via message passing through gates. The connections
from one gate to another can have various channel characteristics such as error/data
rate or propagation delay.
Another important reason for choosing OMNeT++ to conduct simulation
experiments is the availability of third party libraries containing many protocol imple-
mentations for wireless networks. The INET [40] framework version 3.2.3 is used
for higher layer protocol implementations to achieve Internet connectivity for the
OBUs. The VEINS 4.4 (Vehicles in Network Simulation) framework is used for its
DSRC/IEEE 802.11p implementation and its ability to bind a network simulation
Reinforcement learning-based channel sharing 249

INET car_sensors_app 1. Safety applications

2. Non-safety applications

Networking
CoAP2
WSMP1
UDP2

IPv6
VEINS
IEEE 1609.4 MAC ext. Data-generating sensors
Link

VPLC Ethernet
IEEE 802.11p MAC
CAN LIN FlexRay
V2V
V2I IEEE 802.11p PHY Intra-vehicle communication

Figure 7.12 Network protocols used on all communication domains

with a live mobility simulation conducted by Simulation of Urban Mobility (SUMO)

v0.25. Figure 7.12 shows the network protocols made available by OMNeT++ to
enable in-vehicle and inter-vehicle communications.

7.8.2 Mobility simulator

Since vehicular traffic flow is very complex to model, researchers try to predict road
traffic using simulations. A traffic simulation introduces models of transportation
systems such as freeway junctions, arterial routes, roundabouts to the system under
study. SUMO [41] is an open-source microscopic and continuous road traffic sim-
ulation package which enables us to simulate the car flow in a large road network
such as the one in the city of Brighton. Microscopic traffic flow models, in contrast
to macroscopic, simulate single vehicle units, taking under consideration properties
such as position and velocity of individual vehicles.

7.8.3 Implementation
The simulation environment on which novel medium access algorithms are to be
evaluated uses SUMO and open data to reproduce accurate car mobility [42]. The
map is extracted off OpenStreetMap and converted to an XML file which defines
the road network. Then random trips are generated from this road network file, and
finally these trips are converted to routes and traffic flow. The resulting files are
used in SUMO for live traffic simulation as depicted in Figure 7.13. The vehicles are
dynamically generated with unique IDs shown in green labels.
Each node within OMNeT++, either mobile (car) or static (RSU) consists of a
network interface that uses the 802.11p PHY and MAC, and the application layer that
describes a basic safety message exchange and a mobility module. A car, chosen in
random fashion, broadcasts a periodic safety message, much like the ones specified
in the WSMP.
250 Applications of machine learning in wireless communications

Figure 7.13 Large scale simulation in the city of Brighton

Listing 7.1 SUMO scripts and parameters to produce the needed XML files

//map file to road network XML

$netconvert --osm city.osm
//random trips from XML; source and
//destination edge weighted by length "-l"
$randomTrips.py -n city.net.xml -l -e 800 -o
city.trips.xml
//routes using Shortest Path computation
$duarouter -n city.net.xml -t city.trips.xml
-o city.rou.xml

As well as safety message exchange, connected cars can provide extra function-
ality and enable driving assisting and infotainment systems, such as downloading
city map content from RSUs, exchanging video for extended driver vision or even
uploading traffic information to the cloud towards an efficient traffic light system.
The protocols used for such applications would be different from WSMP, such as
the Internet protocols (IPv6, UDP) for the pervasiveness of IP-based applications.
Figure 7.14 shows an example of V2V connectivity, where a car broadcasts a safety
message to neighbouring cars within range.
Reinforcement learning-based channel sharing 251

Figure 7.14 A car is broadcasting to neighbouring cars using IEEE 802.11p

in OMNeT++

7.9 Protocol performance

The MAC method of the vehicular communication standard IEEE 802.11p has been
simulated in a realistic vehicular traffic scenario with vehicle stations periodically
broadcasting packets. In order to evaluate the performance of the novel proposed
RL-based channel sharing protocol in comparison to the baseline IEEE 802.11p pro-
tocol, V2V simulations were carried out using OMNeT++ 5 simulator and the Veins
framework. Realistic mobility simulation is achieved by using SUMO coupled with
OMNeT++.

7.9.1 Simulation setup

All the cars within the area content for access to medium when trying to transmit
a packet or rebroadcast a copy of one. Retransmission probability is set so that a
proportion of nodes in the area of interest will rebroadcast the same information upon
receipt (i.e. for 100 cars it is set at 2%). We collect most of our results within a specific
ROI of ∼600 m × 500 m within the University of Sussex campus and set the power
to a high enough level within the DSRC limit, in order to not be influenced by border
252 Applications of machine learning in wireless communications

Figure 7.15 Campus map used in network simulations

effects (hidden/exposed terminals). The artificial campus map used for simulations
can be seen in Figure 7.15.
The achieved improvement on link-level contention was of primary concern, so a
multitude of tests were run for a single hop scenario, with every node being within the
range of the others. By eliminating the hidden terminal problem from the experiment
and setting an infinite queue size, packet losses from collisions can be accurately
measured. A multi-hop scenario is also presented, which makes the hidden terminal
effect apparent in the performance of the network.
The simulation run time for the proposed MAC protocol consists of two stages,
as seen in Figure 7.10. First is the approximate controller training stage, which lasts
for Ndecay = 1,800 transmitted packets (or 180 s with fb = 10 Hz). Then follows the
evaluation or online period which lasts for 120 s, in which the agent acts with an
ε = α = 0.1. During this time, we benchmark the effect of the trained controllers
regarding network performance as well as keep performing some learning for the
controller augmentation. For IEEE 802.11p simulations, only the evaluation stage is
needed, which lasts for the same time.
All cars in the network are continuously transmitting broadcast packets, such as
CAMs with a period Tb = (1/fb ) = 100 ms. The packets are transmitted using the
highest priority, voice traffic (AC_VO) AC. In VANETs, the network density changes
depending on location and time of the day. We test the performance of the novel MAC
against the standard IEEE 802.11p protocol for different number of cars. The data
rate is set at 6 Mbps so it can conveniently accommodate hundreds of vehicles within
the DSRC communication range. Simulation parameters can be found at Table 7.4.

7.9.2 Effect of increased network density

The scalability of the MAC protocols is evaluated against a varying number of vehicles
travelling in the simulated campus map shown in Figure 7.15. The packet size Lp
Reinforcement learning-based channel sharing 253

Table 7.4 Simulation parameters

Parameter Value

Evaluation time 120 s

A priori training time 180 s
Channel frequency 5.9 GHz
Transmission rate 6 Mbps
Transmission power 1-hop: 100 mW, 2-hop: 40 mW
Packet size Lp 256 bytes
Backoff slot time 13 μs
Broadcasting frequency fb 10 Hz
No of relays ≥2 cars (probabilistic)
Discount rate γ 0.7
Learning rate α Training: (7.4), on-line: 0.1
Epsilon ε training: (7.4), on-line: 0.1

1.0

0.8
Packet delivery ratio

0.6

0.4

0.2
IEEE 802.11p
Q-learning MAC
0.0 Delivery gain

20 40 60 80 100
Network density (cars)

Figure 7.16 PDR versus network density for broadcasting of 256-byte packets
with fb = 10 Hz

used in this scenario is 256 bytes, and the broadcasting frequency fb is set at 10 Hz.
Figure 7.16 shows the increase in goodput when using this novel MAC protocol,
expressed as a PDR. When using the standard IEEE 802.11p, PDR decreases in
denser networks due to the increased collisions between data packets.
The PDR for the proposed Q-learning MAC is measured after the initial,
exploratory phase (since the agent by then has gained significant experience). We
observed a 37.5% increase in performance (original packets delivered) in a network
formed of 80 cars when using the modified, “learning” MAC. There is a slight loss
in performance (4%) for 20-car networks. In such sparse networks, the minimum
CW is optimal, since with a big CW (waiting for more b time slots), transmission
opportunities can be lost and the channel access delay will increase. When using our
254 Applications of machine learning in wireless communications

35
IEEE 802.11p
30 Q-learning MAC

Average normalized RTT (ms)

Access delay overhead
25

0
20 40 60 80 100
Network density (cars)

Figure 7.17 Packet Return Time (delay) versus network density for broadcasting
of 256-byte packets with fb = 10 Hz

learning protocol, the agent still explores larger CW levels 10% of the time (ε = 0.1),
for better adaptability and augmentation of its initial controller. When the network
density exceeds 40 cars, the proposed learning MAC performs much better regarding
successful deliveries.
The round-trip time (RTT) shown in Figure 7.17 is defined as the length of time
it takes for an original broadcast packet to be sent plus the length of time it takes for
a rebroadcast of that packet to be received by the original sender. We can see that the
increased CW of the learning MAC adds to the channel-access delay time. The worst
case scenario simulated is for 100 simultaneous transceivers within the immediate
range of each other, in which the average RTT doubles to 32.8 ms when using the
Q-learning MAC. Given that both the transmission and heard retransmission are of
the same packet size, we can assume that the mean packet delivery latency is 16.4 ms
when using the learning MAC instead of 8 ms for baseline IEEE 802.11p, while PDR
is improved by 54%.

7.9.3 Effect of data rate

We also examine the performance of both the standard and enhanced protocol for
different data rates. PDR is measured for a network of 60 nodes without hidden
terminals. The broadcasting frequency is set at fb = 10 Hz, and the packet size Lp
varies from 64 to 512 bytes, as seen in Figure 7.18. For 512 byte packets, the mean
achieved goodput Tavg per IEEE 802.11p node from (7.10) is 16.925 kbps. For the
same settings, each learning MAC station achieves 29.218 kbps on average, yielding
to a 72.63% increase in goodput. It is clear that for larger packet transmissions the
Q-learning-based protocol will be much faster and more reliable:
Tavg = Lp × fb × 8 bit × PDR. (7.10)
Reinforcement learning-based channel sharing 255

1.0

0.8

Packet delivery ratio

0.6

0.4

0.2
IEEE 802.11p
Q-learning MAC
0.0 Delivery gain

64 128 256 384 512

Packet size (bytes)

Figure 7.18 PDR versus packet size for 60 vehicles broadcasting with fb = 10 Hz

1.0

0.8
Packet delivery ratio

0.6

0.4

0.2 IEEE 802.11p

Q-learning MAC
Delivery gain
0.0
30 60 90 120 150
Network density (cars)

Figure 7.19 PDR versus network density for broadcasting of 256-byte packets with
fb = 10 Hz in a two-hop scenario

7.9.4 Effect of multi-hop

In a network without fixed topology, the most common way to disseminate informa-
tion is to broadcast packets across the network. In VANETs, vehicles often cooperate
to deliver data messages through multi-hop paths, without the need of centralized
administration. In this scenario, we test the performance of the proposed protocol
when attempting to transmit two hops away. We evaluate performance for two-hop
transmissions by reducing the transmission power to 40 mW. As the network density
increases, the proposed MAC offers a valid delivery benefit for vehicle-stations con-
tenting for access on the same channel. The performance of both IEEE 802.11p and
the proposed learning MAC regarding two-hop packet reception ratio is compared in
Figure 7.19.
256 Applications of machine learning in wireless communications

We see that because the hidden terminal phenomenon appears, the performance
deteriorates compared to the single hop scenario, but the performance gain regarding
packet delivery is still apparent when using Q-learning to adapt the backoff. Packets
lost are not recovered since we are concerned with the performance of the link layer.

7.10 Conclusion
A contention-based MAC protocol for V2V/V2I transmissions was introduced in this
chapter. It relies on Q-learning to discover the optimum CW by continuously interact-
ing with the network. Simulations were developed to demonstrate the effectiveness
of this learning-based MAC protocol. Results prove that the proposed method allows
the network to scale better to increase network density and accommodate higher
packet delivery rates compared to the IEEE 802.11p standard. This translates to more
reliable packet delivery and higher system throughput, while maintaining acceptable
delay levels. Future work will be focused on how the learning MAC responds to dras-
tic changes in the networking environment via invoking the ε decay function while
online as well as improving fairness and transmission latency.

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Chapter 8
Machine-learning-based perceptual video coding
in wireless multimedia communications
Shengxi Li1 , Mai Xu2 , Yufan Liu3 , and Zhiguo Ding4

We present in this chapter the advantage of applying machine-learning-based per-

ceptual coding strategies in relieving bandwidth limitation for wireless multimedia
communications. Typical video-coding standards, especially the state-of-the-art high
efficiency video coding (HEVC) standard as well as recent research progress on
perceptual video coding, are included in this chapter. We further demonstrate an exam-
ple that minimizes the overall perceptual distortion by modeling subjective quality
with machine-learning-based saliency detection. We also present several promis-
ing directions in learning-based perceptual video coding to further enhance wireless
multimedia communication experience.

8.1 Background
At present, multimedia applications, such as Facebook and Twitter, are becoming
integral components in the daily lives of millions, leading to the explosion of big
data. Among them, videos are one of the largest types of big data [1], thus posing
a great challenge to the limited communication and storage resources. Meanwhile,
due to more powerful camera hardware, their resolutions are significantly increasing,
further intensifying the hunger on communication and storage resources. Aiming at
overcoming this resource-hungry issue, a set of video-coding standards have been
proposed to condense video data, e.g., MPEG-2 [2], MPEG-4 [3] VP9 [4], H.263 [5]
and H.264/AVC [6].
Most recently, as the successor of H.264/AVC, HEVC [7] was formally approved
in April, 2013. In HEVC, several new features, e.g., the quadtree-based coding

1
Department of Electrical and Electronic Engineering, Imperial College London, UK
2
Department of Electronic and Information Engineering, Beihang University, China
3
Institute of Automation, Chinese Academy of Sciences, China
4
School of Electrical and Electronic Engineering, The University of Manchester, UK
262 Applications of machine learning in wireless communications

structure and intra-prediction modes with 33 directions,1 were adopted. Consequently,

the HEVC Main Still Picture (HEVC-MSP) profile [8], which is designed for still
picture compression, achieves the best performance among all the state-of-the-art
standards on image compression, with an approximately 10% (over VP9) to 40% (over
JPEG) improvement in bit-rate savings [9]. However, all existing standards, including
HEVC-MSP, primarily focus on removing statistical redundancy by adopting various
techniques [10], e.g., intra-prediction and entropy coding. Further reducing statistical
redundancy may help to improve coding efficiency, but at the cost of extremely high
computational complexity.
Koch et al. [12] investigated that the bandwidth between the human eyes and
brain is approximately 8 Mbps, which is far insufficient to process the visual input
captured by millions of optical cells. Thus, the human eye is mostly at a quite low res-
olution, except for a small area at the fovea (visual angle of approximately 2◦ ), which
is called the region-of-interest (ROI) in the video-coding community. Meanwhile, as
pointed out by [13], human ROIs are similar across different individuals. It is also well
known [14] that the coding mechanism can be modified to cater to the human visual
system (HVS) by moving bits from non-ROIs to ROIs to achieve better subjective
quality. This is also illustrated in Figure 8.1(b) and (c). Perceptual video coding has
received a great deal of research effort from 2000 onwards, due to its great potential in
improving coding efficiency [15–18]. In H.263, a perceptual rate control scheme [15]
was proposed. In this scheme, a perceptual sensitive weight map of conversational
scene (i.e., scene with frontal human faces) is obtained by combining stimulus-driven
(i.e., luminance adaptation and texture masking) and cognition-driven (i.e., skin col-
ors) factors together. According to such a map, more bits are allocated to ROIs by
reducing QP values in these regions. Afterwards, for H.264/AVC, a novel resource
allocation method [16] was proposed to optimize the subjective rate–distortion
(R–D)-complexity performance of conversational video coding, by improving the
visual quality of face region extracted by the skin-tone algorithm. Moreover, Xu
et al. [19] utilized a novel window model to characterize the relationship between
the size of window and variations of picture quality and buffer occupancy, ensuring
a better perceptual quality with less quality fluctuation. This model was advanced
in [20] with an improved video quality metric for better correlation to the HVS. Most
recently, in HEVC, the perceptual model of structural similarity (SSIM) has been
incorporated for perceptual video coding [21]. Instead of minimizing mean squared
error (MSE) and sum of absolute difference, SSIM is minimized [21] to improve
the subjective quality of perceptual video coding in HEVC. However, through our
investigation, the substantial low quality in non-ROIs may also significantly degrade
image quality, as shown in Figure 8.1(d). Thus, how many bits should “move” from
non-ROIs to ROIs, together with accurate ROI detection, is crucial for compression.
In other words, we need to ensure that the detected ROIs are the regions that attract
human attention, and then bit allocation needs to be optimized according to ROIs,
targeting minimal overall perceptual distortion.

1
Planar and DC are two other intra-prediction modes.
Machine-learning-based perceptual video coding 263

CTU numbers Bit allocation (%)

6 58 16 84
0 10 20 30 40 50 60 0 10 20 30 40 50 60 70 80 90100
CTUs in ROIs CTUs in non-ROIs Bits in ROIs Bits in non-ROIs

(a) Heat map (b) No emphasis

Bit allocation (%) Bit allocation (%)

33 67 83 17
0 10 20 30 40 50 60 70 80 90 100 0 10 20 30 40 50 60 70 80 90 100
Bits in ROIs Bits in non-ROIs Bits in ROIs Bits in non-ROIs

(c) Well balanced (d) More emphasis

Figure 8.1 An example of HEVC-based compression for Lena image, with different
bit allocation emphasis on ROIs. Note that (a) is the heat map of eye
fixations; (b), (c) and (d) are compressed by HEVC-MSP at 0.1 bpp
with no, well balanced and more emphasis on face regions. The
difference mean opinion scores (DMOS) for (b), (c) and (d) are 63.9,
57.5 and 70.3, respectively [11]

The organization of this chapter is as follows. The literature review is first intro-
duced in Section 8.2, from perspectives of perceptual models and incorporations in
video coding. We then present in Section 8.3 the recursive Taylor expansion (RTE)
method for optimal bit allocation toward the perceptual distortion and also provide rig-
orous proofs. The computational analysis of the proposed RTE method is introduced in
264 Applications of machine learning in wireless communications

Section 8.4. For experimental validations, we first verify the proposed RTE method
on compressing one single image/frame in Section 8.5, followed by the results on
compressing video sequences in Section 8.6.

8.2 Literature review on perceptual video coding

Generally speaking, main parts of perceptual video coding are perceptual mod-
els, perceptual model incorporation in video coding and performance evaluations.
Specifically, perceptual models, which imitate the output of the HVS to specify
the ROIs and non-ROIs, need to be designed first for perceptual video coding.
Second, on the basis of the perceptual models and existing video-coding standards,
perceptual model incorporation in video coding from perceptual aspects needs to
be developed to encode/decode the videos, mainly through removing their percep-
tual redundancy. Rather than incorporating perceptual model in video coding, some
machine-learning-based image/video compression approaches have also proposed
during the past decade.

8.2.1 Perceptual models

Perceptual models can be classified into two categories: either manual or automatic
identifications.
8.2.1.1 Manual identification
This kind of perceptual models requires manual effort to distinguish important regions
which need to be encoded with high quality. In the early years, Geisler and Perry [22]
employed a foveated multi-resolution pyramid video encoder/decoder to compress
each image of varying resolutions into five or six regions in real-time, using a pointing
device. This model requires the users to specify which regions attract them most
during the video transmission. Thus, this kind of models may lead to transmission
and processing delay between the receiver and transmitter sides, when specifying
the ROIs. Another way [23] is to specify ROIs before watching, hence avoiding the
transmission and processing delay. However, considering the workload of humans,
these models cannot be widely applied to various videos.
In summary, the advantage of manual identification models is the accurate detec-
tion of ROIs. However, as the cost, it is expensive and intractable to extensively apply
these models due to the involvement of manual effort or hardware support. In addition,
for the models of user input-based selection, there exists transmission and processing
delay, thus making the real-time applications impractical.
8.2.1.2 Automatic identification
Just as its name implies, this category of perceptual models is to automatically
recognize ROIs in videos, according to visual attention mechanisms. Therefore,
visual attention models are widely used among various perceptual models. There are
two classes of visual attention models: either bottom-up or top-down models. Itti’s
model [24] is one of the most popular bottom-up visual attention models in perceptual
Machine-learning-based perceptual video coding 265

video coding. Mimicking processing in primate occipital and posterior parietal cortex,
Itti’s model integrates low-level visual cues, in terms of color, intensity, orientation,
flicker and motion, to generate a saliency map for selecting ROIs [17].
The other class of visual attention models is top-down processing [14,18,25–29].
The top-down visual attention models are more frequently applied to video applica-
tions, since they are more correlated with human attractiveness. For instance, human
face [16,18,26] is one of the most important factor that draws top-down attention,
especially for conversational video applications. Beyond, a hierarchical perceptual
model of face [18] has been established, endowing unequal importance within face
region. However, abovementioned approaches are unable to figure out the importance
of face region.
In this article, we quantify the saliency of face and facial features via learning the
saliency distribution from the eye fixation data of training videos, via conducting the
eye-tracking experiment. Then, after detecting face and facial features for automati-
cally identifying ROI [18], the saliency map of each frame of encoded conversational
video is assigned using the learnt saliency distribution. Although the same ROI is
utilized in [18], the weight map of our scheme is more reasonable for the perceptual
model for video coding, as it is in light of learnt distribution of saliency over face
regions. Note that the difference between ROI and saliency is that the former refers
to the place that may attract visual attention while the later refers to the possibility of
each pixel/region to attract visual attention.

8.2.2 Incorporation in video coding

The existing incorporation schemes can be mainly divided into two aspects: model-
based and learning-based approaches. The model-based approaches apply prior
models or the above perceptual models to the existing video-coding approaches,
while the learning-based approaches aim at discovering similarities among pixels or
blocks to reduce redundancy in video coding.

8.2.2.1 Model-based approaches

One category of approaches called preprocessing is to control the nonuniform dis-
tribution of distortion before encoding [30–32]. A common way for preprocessing
is spatial blurring [30,31]. For instance, the spatial blurring approach [30] separates
the scene into foreground and background. The background is blurred to remove
high-frequency information in spatial domain so that less bits are allocated to this
region. However, this may cause obvious boundaries between the background and
foreground.
Another category is to control the nonuniform distribution of distortion during
encoding, therefore called embedded encoding [16,18,33–35]. As it is embedded into
the whole coding process, this category of approaches is efficient in more flexibly
compressing videos with different demands. In [16], Liu et al. established importance
map at macro block (MB) level based on face detection results. Moreover, combin-
ing texture and nontexture information, a linear rate–quantization (R–Q) model is
applied to H.264/AVC. Based on the importance map and R–Q model, the optimized
266 Applications of machine learning in wireless communications

QP values are assigned to all MBs, which enhances the perceived visual quality of
compressed videos. In addition, after obtaining the importance map, the other encod-
ing parameters, such as mode decision and motion estimation search, are adjusted to
provide ROIs with more encoding resources. Xu et al. [18] proposed a new weight-
based unified R–Q (URQ) rate control scheme for compressing conversational videos,
which assigns bits according to bpw, instead of bpp in conventional URQ scheme.
Then, the quality of face regions is improved such that its perceived visual quality
is enhanced. The scheme in [18] is based on the URQ model [36], which aims at
establishing the relationship between bite-rate R and quantization parameters Q, i.e.,
R–Q relationship. However, since various flexible coding parameters and structures
are applied in HEVC, R–Q relationship is hard to be precisely estimated [37]. There-
fore, Lagrange multiplier λ [38], which stands for the slope of R–D curve, has been
investigated. According to [37], the relationship between λ and R can be better char-
acterized in comparison with R–Q relationships. This way, on the basis of R–λ model,
the state-of-the-art R–λ rate control scheme [39] has better performance than the
URQ scheme. Therefore, on the basis of the latest R–λ scheme, this article proposes
a novel weight-based R–λ scheme to further improve the perceived video quality
of HEVC.

8.2.2.2 Learning-based approaches

From the viewpoint of machine learning, the pixels or blocks from one image or several
images may have high similarity. Such similarity can be discovered by machine-
learning techniques and then utilized to decrease redundancy of video coding. For
exploiting the similarity within an image/video, image inpainting has been applied
in [40,41] to use the image blocks from spatial or temporal neighbors for synthe-
sizing the unimportant content, which is deliberately deleted at the encoder side. As
such, the bits can be saved as not encoding the missing areas of the image/video.
Beyond, rather than predicting the missing intensity information in [40,41] , several
approaches [42–45] have been proposed to learn to predict the color in an images using
the color information of some representative pixels. Then, only representative pixels
and grayscale image need to be stored, such that the image [43–45] or video [42] cod-
ing can be achieved. Most recently, the deep-learning technique has also been applied
to reduce coding complexity via an early-terminated coding unit (CU) partition
scheme [46].
For working on similarity across various images or frames of videos, dictio-
nary learning has been developed to discover the inherent patterns of image blocks.
Together with dictionary learning, sparse representation can be then used to effec-
tively represent an image for image [47] or video coding [48], instead of conventional
image transforms such as discrete cosine transform.
The above approaches primarily improve the fidelity of ROIs, but they may fail
in ensuring the overall subjective quality, as extremely low quality on non-ROIs can
also degrade the subjective quality. In the next section, we propose an approach to
optimize the overall subjective quality, different from the above approaches that only
increase bits in ROIs.
Machine-learning-based perceptual video coding 267

8.3 Minimizing perceptual distortion with the RTE method

In this section, we primarily focus on minimizing the perceptual distortion of
one image/frame compression for clarify and propose a closed-form bit allocation
approach to minimize the perceptual distortion [49]. It needs to point out that the
approach can also be applied in perceptual video compression, which is to be presented
in Section 8.6.
Specifically, the most recent work [17] has pointed out that eye-tracking weighted
peak signal-to-noise ratio (EWPSNR), which is the combination of eye-tracking fix-
ations and MSE, is highly correlated with subjective quality. Due to the unavailability
of eye-tracking data, we utilize the saliency weighted PSNR (SWPSNR) instead as
the perceptual distortion to approximate subjective quality. Automatic saliency detec-
tion is thus the first step of our approach for saliency-guided image compression. In
our approach, we leverage on our most recent face saliency-detection method [50]
for compressing face images and a latest saliency-detection method [51] for com-
pressing other generic images. Note that face and non-face images are automatically
classified using the face detector in [50]. Then, we propose a formulation to min-
imize perceptual distortion with reasonable bit allocation on compressed images.
Unfortunately, it is intractable to obtain a closed-form solution to the proposed opti-
mization formulation because the formulation is a high-order algebraic equation, and
its non-integer exponents vary across different coding tree units (CTUs). We thus
develop a new method, namely, RTE, to acquire the solution for optimal bit alloca-
tion in a closed-form manner. In the proposed RTE method, we iterate a third-order
Taylor expansion to reach the optimal solution for bit allocation. We also develop
an optimal bit reallocation process to alleviate the mismatch between the target and
actual bits, while maintaining perceptual distortion optimization. We further verify
via both theoretical and numerical analyses that little time cost is incurred by our
approach.
We first transplant the R–λ RC approach [37] into HEVC-MSP in Section 8.3.1.
Upon this, an optimization formulation is proposed in Section 8.3.2, which aims at
maximizing the SWPSNR at a given bit rate for each image. The RTE method is then
proposed in Section 8.3.3 to solve this formulation with a closed-form solution. In
this way, the perceptual distortion can be minimized via bit allocation. In addition, we
develop an optimal bit reallocation method in Section 8.3.4 to alleviate the mismatch
between the target and actual bit rates.

8.3.1 Rate control implementation on HEVC-MSP

The latest R–λ approach is proposed in [37] for RC in HEVC. Since we concentrate
on applying RC to image compression, the CTU level RC in one video frame is
discussed here. Specifically, for HEVC, it has been verified that the hyperbolic model
can better fit the R–D relationship [37]. Based on the hyperbolic model, an R–λ model
is developed for bit allocation in the latest HEVC RC approach, where λ is the slope
of the R–D relationship [38]. Assuming that di , ri and λi represent the distortion, bits
268 Applications of machine learning in wireless communications

and R–D slope for the ith CTU, respectively, the R–D relationship and R–λ model are
formulated as follows:
di = ci ri −ki , (8.1)
and
∂di
λi = − = ci ki · ri −ki −1 , (8.2)
∂ri
where ci and ki are the parameters that reflect the content of the ith CTU. In the R–λ
approach [37], ri is first allocated according to the predicted mean absolute difference,
and then its corresponding λi is obtained using (8.2). By adopting a fitting relationship
between λi and QP, the QPs of all CTUs within the frame can be estimated such that
RC is achieved in HEVC. For more details, refer to [37].
However, for HEVC-MSP, ci and ki cannot be obtained when encoding CTUs.
Thus, it is difficult to directly apply the R–λ RC approach to HEVC-MSP. In the work
of [52], the sum of the absolute transformed differences (SATD), calculated by the
sum of Hadamard transform coefficients, is utilized for HEVC-MSP. Specifically,
the modified R–λ model is
βi
si
λ i = αi , (8.3)
ri
where αi and βi are the constants for all CTUs and remain the same when encoding
an image. Moreover, si denotes the SATD for the ith CTU, which measures the CTU
texture complexity. Nevertheless, SATD is too simple to reflect image content, leading
to an inaccurate R–D relationship during RC.
To avoid the above issues, we adopt a preprocessing process in calculating ci and
ki . After pre-compressing, the pre-encoded distortion, bits and λ can be obtained for
the ith CTU, which are denoted as d̄i , r̄i and λ̄i , respectively. Then, the RC-related
parameters, ci and ki , can be estimated upon (8.1) and (8.2) before encoding the
ith CTU:
d̄i
ci = , (8.4)
−λ̄ ·r̄ /d̄
r̄i i i i

and
λ̄i · r̄i
ki = . (8.5)
d̄i
With the estimated ci and ki , the RC of the R–λ approach [37] can be implemented in
HEVC-MSP.
Here, a fast pre-compressing process is developed in our approach, which sets the
maximum CU depth to 0 for all CTUs. We have verified that the fast pre-compressing
process slightly increases the computational complexity by a 5% burden, which is
slightly larger than the 3% of the SATD-based method [52]. However, this process is
able to well reflect the R–D relationship, as to be verified Section 8.5.4.
Machine-learning-based perceptual video coding 269

8.3.2 Optimization formulation on perceptual distortion

The primary objective of this chapter is to minimize perceptual distortion for HEVC-
based image compression. In our approach, the SWPSNR is applied to measure the
perceptual distortion, as [53] has shown that SWPSNR is highly correlated with sub-
jective quality. For SWPSNR, the pixel-wise saliency values need to be detected as
the first step in our approach, and these values are used for weighting the MSE.
In this chapter, we utilize two state-of-the-art saliency-detection methods for
calculating SWPSNR. Specifically, the latest Boolean-map-based saliency (BMS)
method [51] is applied in modeling SWPSNR for generic images. Furthermore, for
face images, our most recent work [50] has better accuracy in saliency detection than
the BMS method. Thus, when computing the SWPSNR of face images, we use the
work of [50] to obtain the saliency values.
Here, we denote wi as the average saliency value within the ith CTU. Meanwhile,
we calculate distortion di by the sum of pixel-wise square error for the ith CTU. Then,
based on di and wi , the optimization on SWPSNR at a given target bit rate R can be
formulated as
M
i=1 wi di
min M
M
s.t. i=1 ri = R. (8.6)
i=1 wi
In (8.6), M denotes the number of CTUs in the image. By using the Lagrange multi-
plier λ, (8.6) can be turned to find the minimum value of R–D cost J [38], which is
defined as
M
i=1 wi di
J = M
+ λ · (i=1
M
ri ). (8.7)
i=1 wi
By setting the partial derivatives of (8.7) to zero, the minimum J can be found as
follows2 :
M
∂J ∂ i=1 M
wi di /i=1 wi + λ(i=1
M
ri )
=
∂ri ∂ri
wi ∂di
= M · +λ
i=1 wi ∂ri
= 0. (8.8)
Given (8.1) and (8.2), (8.8) is turned to
−(1/(ki +1))
λ · i=1
M
wi wi ai bi

ri = = , (8.9)
c i ki w i λ
where ai = ci ki and bi = (1/(ki + 1)) also reflect the image content for each CTU.
Moreover, wi = wi /(i=1
M
wi ) represents the visual importance for each CTU. Note
that with our pre-compressing process, ci and ki can be obtained in advance. Thus, ai

2
It needs to point out that J in (8.7) is convex with regard to ri and λ, which ensures the global minimum
of the problem (8.7).
270 Applications of machine learning in wireless communications

and bi are available before encoding the image. Once λ is known, ri can be estimated
using (8.9) for achieving the minimum J .
Meanwhile, there also exists a constraint on bit rate, which is formulated as

M
ri = R. (8.10)
i=1

According to (8.9) and (8.10), we need to find the “proper” λ and bit allocation ri to
satisfy the following equation:

M M

wi ai bi

ri = = R. (8.11)
i=1 i=1
λ

After solving (8.11) to find the “proper” λ, the target bits can be assigned to each
CTU with the maximum SWPSNR.
Unfortunately, since ai and bi vary across different CTUs, (8.11) cannot be solved
by a closed-form solution. Next, the RTE method is proposed to provide a closed-form
solution.

8.3.3 RTE method for solving the optimization formulation

For solving (8.11), we assume that ri (
λ)bi = ( ri and
wi ai )bi , where λ are the estimated
ri and λ, respectively. Then, (8.11) can be rewritten as

M M
M bi
i ai bi
w λ
ri = =
ri = R. (8.12)
i=1 i=1
λ i=1
λ

From (8.12), we can see that once ri → ri . As such, the optimiza-

λ → λ, there exists
tion formulation of (8.11) can be solved in our approach. However, we do not know

λ at the beginning. Meanwhile, λ of (8.12) is also unknown because it is intractable
to find the closed-form solution to (8.11). Therefore, a chicken-and-egg dilemma
exists between
λ and λ. To solve this dilemma, a possible
λ is initially set. In our RTE
method, the picture λ (denoted as λpic ) is chosen as the initial value of λ for quick
convergence. It is calculated by the R–λ model at the picture level [37,52]:
s βpic
pic
λpic = αpic , (8.13)
R

where αpic and βpic are the fitted constants (αpic = 6.7542 and βpic = 1.7860 in HM
16.0) and spic represents the SATD for the current picture. Recall that R denotes the
target bits allocated to the currently encoded picture.
In the following, the RTE method is proposed to iteratively update
λ for making

λ → λ.
Machine-learning-based perceptual video coding 271

Specifically, we preliminarily apply Taylor expansion on (λ/λ)bi of (8.12), and

then we discard the biquadratic and higher order terms. The process can be formulated
as follows:
M bi
λ
R=
ri
i=1
λ

ln ( (ln (
M
λ/λ) λ/λ))n n
=
ri 1 + bi + · · · + bi + · · ·
i=1
1! n!

ln ( (ln (
λ/λ))2 2 (ln (
M
λ/λ) λ/λ))3 3
≈
ri 1 + bi + bi + bi (8.14)
i=1
1! 2! 3!

In the following equation, we use

λ to denote the approximation solution to (8.14)
after discarding the biquadratic and higher order terms. Consequently, (8.12) can be
approximated to be a cubic equation with variable ln
λ:

ln (
λ/
(ln (
λ/
λ))2 2 (ln (
λ/

M
λ) λ))3 3
R=
ri 1 + bi + bi + bi
i=1
1! 2! 3!

M 3 M 2
bi bi b3
=−
ri ln3

λ+
ri λ ln2

+ i ln λ
i=1
6 i=1
2 2

A B

M
b3i
−
ri b2i ln
λ + bi + ln λ ln

2
λ
i=1
2

C

M
b2i 2 b3i 3
+
ri 1 + bi ln λ + ln λ + ln λ . (8.15)
i=1
2 6

D

By applying the Shengjin formula [54], this cubic equation is evaluated to

obtain the solution of

λ as
√

√
3 √
3 −F ± F 2 − 4EG
λ = e((−B−( Y1 + Y2 ))/3A) , Y1,2 = BE + 3A , (8.16)
2

where E = B2 − 3AC, F = BC − 9A(D − R) and G = C 2 − 3B(D − R). Since =

F 2 − 4EG > 0 in practical encoding, (8.16) has only one real solution [54]. Thus, the
value of

λ is unique for optimizing bit allocation. After further removing the cubic-
order term, (8.14) is turned to be a quadratic equation. We found that such a quadratic
equation may have no real solution or two solutions. Meanwhile, using only one term
may lead to large approximation error and slow convergence speed, while keeping
more than four terms probably makes the polynomial equations on ln
λ unsolvable.
272 Applications of machine learning in wireless communications

Therefore, discarding the biquadratic and higher order terms of the Taylor expansion
is the best choice for our approach.
However, due to the truncation of high-order terms in the Taylor expansion,
λ
estimated by (8.16) may not be an accurate solution to (8.12). Fortunately, as proven
in Lemma 8.1,
λ is more accurate3 than
λ when
λ < λ.

Lemma 8.1. Consider λ > λ > 0, bi > 0, and R > 0 for (8.12). When the solution of
λ to (8.12) is

λ, the following inequality holds for

λ:

λ − λ| < |
λ − λ|. (8.17)

Proof. As canbe seen in (8.15),

λ is the solution of λ to the third-order Taylor
expansion on M
r (λ/λ) bi
. Hence, the following equation exists:
i=1 i

M bi
λ
R =
ri
i=1
λ

M M
ln (
λ/
λ) M
(ln (
λ/
λ))2 2 (ln (
M
λ/
λ))3 3
=
ri +
ri bi + ri bi + ri bi . (8.18)
i=1 i=1
1! i=1
2! i=1
3!

In
fact, 0 < ( <
λ/λ)bi < 1 holds for 0 λ < λ and bi > 0. Besides, there exists
M
R = i=1 ri (
λ/λ) i in (8.18). Therefore, M
b
i=1
ri > R can be evaluated.

Next, assuming that λ ≤ λ, we have ln (λ/λ) ≥ 0. Due to M ri > R, ln (

λ/

i=1 λ) ≥
0, and bi > 0, the inequality below holds:

M M
ln (
λ/
λ) M
(ln (
λ/
λ))2 2 (ln (
M
λ/
λ))3 3

ri +
ri bi +
ri bi +
ri bi > R, (8.19)
i=1 i=1
1! i=1
2! i=1
3!

which is contradictory with (8.18). Therefore, it can be proven that

λ <
λ. Then,

given Lemma 8.2, λ < λ < λ can be obtained. As a result, |λ − λ| < |λ − λ| exists.
This completes the proof of Lemma 8.1.

Lemma 8.2. Consider λ > 0, λ > 0, bi > 0, λ =

λ and R > 0 for (8.12). If

λ is the
solution of λ to (8.12), then the following holds:

λ < λ. (8.20)

3
It is obvious that 0 < bi = 1/(ki + 1) < 1 and R > 0 in HEVC encoding.
Machine-learning-based perceptual video coding 273
M

i=1
bi
Proof. Toward the Taylor expansion of ri (λ/λ) in (8.12), we can obtain the
following equations:

M bi
λ
R=
ri
i=1
λ

M M
ln (
λ/
λ) M
(ln (
λ/
λ))2 2 (ln (
M
λ/
λ))3 3
=
ri +
ri bi +
ri bi +
ri bi
i=1 i=1
1! i=1
2! i=1
3!

M M
ln (
λ/λ) M
λ/λ))2 2 (ln (
(ln (
M
λ/λ))3 3
=
ri +
ri bi +
ri bi +
ri bi
i=1 i=1
1! i=1
2! i=1
3!

M
λ/λ))4 4 (ln (
(ln (
M
λ/λ))5 5 (ln (
M
λ/λ))6 6
+
ri bi +
ri bi +
ri bi + · · · .
i=1
4! i=1
5! i=1
6!
(8.21)

There exist two cases of

λ and λ:
● For > 0, we can obtain (ln (
λ > λ > 0 and bi λ/λ)) · bi > 0. It is known that
(8.21) holds with R > M
r i > 0 and ln (λ/λ) > 0 because of (λ/λ) > 1. Thus,
i=1

ln (λ/λ) > ln (λ/λ) > 0 exists such that λ < λ can be achieved.
● For λ > λ > 0 and bi > 0, we have
M
λ/λ))4 4 (ln (
(ln (
M
λ/λ))5 5 (ln (
M
λ/λ))6 6

ri bi +
ri bi +
ri bi + · · · > 0. (8.22)
i=1
4! i=1
5! i=1
6!

Then, with (8.21), the following inequality exists:

M
ln (
λ/
λ) M
(ln (
λ/
λ))2 2 (ln (
M
λ/
λ))3 3

ri bi +
ri bi +
ri bi
i=1
1! i=1
2! i=1
3!

M
ln (
λ/λ) M
λ/λ))2 2 (ln (
(ln (
M
λ/λ))3 3
>
ri bi +
ri bi +
ri bi . (8.23)
i=1
1! i=1
2! i=1
3!

Moreover, viewing
λ and λ as variable x, the inequality (8.23) can be analyzed
by (8.24). The function of (8.24) monotonously decreases to 0 along with the
increasing of variable x (until x ≤
λ):

M M
ln (
λ/x) M
λ/x))2 2 (ln (
(ln (
M
λ/x))3 3

ri +
ri bi +
ri bi +
ri bi . (8.24)
i=1 i=1
1! i=1
2! i=1
3!

By combining (8.23) and (8.24), we can obtain

λ < λ.
Therefore,

λ < λ holds for both cases. This completes the proof of Lemma 8.2.
274 Applications of machine learning in wireless communications

Remark 8.1. Given Lemma 8.2, for the subsequent iterations of the RTE method,
0 < λ < λ of Lemma 8.1 can be satisfied since the value of λ has been replaced
by that of

λ. Furthermore, as shown in Lemma 8.1, although both λ and

λ may be
inaccurate for estimating λ in (8.11),

λ, obtained through (8.12)–(8.16), is closer to

λ than λ. Therefore, we can iterate the Taylor expansion by using
λ as
λ to the next
iteration, which is the core of our RTE method. In this way, the closed-form solution
λ can be obtained by iteratively estimating λ.
Our RTE method is summarized in Table 8.1. For each iteration, the con-
vergence criterion is set according to the approximation error, Ea < 10−10 , where
Ea = |i=1M
ri − R|/R. As analyzed in Section 8.4, the approximation error of our
RTE method is able to converge to 10−10 , generally with no more than three itera-
tions. In other words, after three or fewer iterations, the RTE method is able to reduce
the difference between λ and λ to an extremely small range, meeting the convergence
criterion. Thus, λ can be output as the closed-form solution to (8.12) (as well as
(8.11)). Finally, we replace λ by λ in (8.9) to allocate the target bits to each CTU such
that SWPSNR can be maximized.
The physical explanation for the fast convergence speed of our RTE method is
as follows. Obviously, the approximation error for each iteration of the RTE method
is largely related to ln (λ/λ) in ((ln (
λ/λ))n /n!)bni of (8.14). To reduce the value of

ln (λ/λ) for small approximation error, our RTE method utilizes a more accurate solu-
tion

λ after each iteration to replace

λ for the next iteration, making ((ln (λ/λ))n /n!)bni
decrease sharply. Therefore, such a replacement not only provides a more accurate
input for the next iteration but also greatly reduces the values of the discarded terms
and the approximation error. In this way, the convergence speed can be accelerated
along with iterations. Moreover, keeping three terms for the Taylor expansion rather
than other terms is solvable and also contributes to the fast convergence speed of our
RTE method.

8.3.4 Bit reallocation for maintaining optimization

As we discussed in Section 8.3.3, bits are reasonably allocated in our approach to
minimize perceptual distortion. However, in practical encoding, a slight difference

Table 8.1 The RTE method for solving (8.12) [11]

• Input: ai , bi , wi for each encoding CTUs and target bits R.

• Output: reasonable bit allocation ri for each CTU on maximizing SWPSNR.
– Initialize
λ to be λpic .
– While λ does not meet the convergence criterion
1 Calculate A, B, C and D of (8.15) with λ.
2 Obtain
λ estimated by (8.16).
3 Update λ with the obtained
λ.
End
– Save the final λ.
– Apply it to bit allocation ri with (8.9).
– Return ri for each CTU.
Machine-learning-based perceptual video coding 275
Optimization on subjective quality Control on bit-rates Compression on image
Ti,i–1+K
wi, ci, ki λi Bit
ith CTU QPi
RTE method R – λ model λ – QP fitting HEVC-based encoding streams
Input Pre-
image compress
Optimal bit re-allocation
Actual bits
Saliency Ti+1, i+K
detection wi+1, ci+1, ki+1 Bit
λi+1 QPi+1
(i+1)th RTE method R – λ model λ – QP fitting HEVC-based encoding streams
CTU
Actual bits

Figure 8.2 The procedure of our approach on minimizing perceptual

distortion [11]

between the target and actual bits may exist for each CTU. This difference may degrade
RC accuracy. To overcome this, we develop a bit reallocation process to accurately
control bit rates, meanwhile maintaining the optimization for perceptual distortion.
Specifically, for compensating the bit-rate error after encoding the ith CTU, the
target bits for the incoming K CTUs (denoted as Ti+1,i+K ) are updated by
⎛ ⎞

j=i+K

j=M
Ti+1,i+K = rj + ⎝

T− rj ⎠ .
(8.25)
j=i+1 j=i+1

bit-rate error

In (8.25),
T is the remaining bits for encoding remaining CTUs, and rj represents
the target bits for the jth CTU by our RTE method. Recall that M denotes the total
number of CTUs. Obviously, as seen from (8.25), the bit error is compensated during
encoding the next K CTUs. Here, the RTE method of Section 8.3.3 is applied to
reallocate Ti+1,i+K to the next K CTUs. Note that we follow [52] and [37] to set
K = 4, which means that bits are reassigned in the next four CTUs. Moreover, note
that due to the fast convergence speed of our RTE method, the complexity increases
little for the bit reallocation process.
Finally, we summarize our HEVC-based image compression approach in
Figure 8.2. Specifically, we first transplant RC to HEVC-MSP with a simplified
pre-compression process, and the saliency values are detected for the input image.
Then, our RTE method obtains the target bits of each CTU, which can minimize per-
ceptual distortion at a given bit rate. Next, the QP value of each CTU is estimated
using the R–λ model and QP fitting. Note that the bits need to be reallocated in the
following CTUs to bridge the gap between the target and actual bits. In addition, as
to be verified in Section 8.4, little computational complexity cost is introduced in our
RTE method, further highlighting the efficiency of our approach.

8.4 Computational complexity analysis

In this section, we primarily focus on the computational complexity of our approach.
Since our approach adopts the RTE method to optimize perceptual distortion, the
276 Applications of machine learning in wireless communications

convergence speed of the RTE method is first discussed from both theoretical and
numerical perspectives. In the numerical analysis, we also provide the practical
computational time of our approach.

8.4.1 Theoretical analysis

For the theoretical analysis, we investigate the difference between
λ and λ alongside
the iterations of our RTE method. Here, we define λ as the difference between λ
and λ as

λ−λ
λ = . (8.26)
λ
If |λ| → 0, then it indicates that our RTE method is stably convergent. Therefore,
we take into consideration λ along with each iteration in our RTE method to analyze
its convergence speed.
In practice, ki (> 0) of (8.9) varies in a small range when encoding images
using HEVC-MSP. Therefore, we assume that bi (0 < bi = (1/(ki + 1)) < 1 in (8.9))
remains constant for simplicity. Based on this assumption, the convergence speed of
our RTE method can be determined with Lemma 8.3.

Lemma 8.3. Consider that λ > 0,

λ > 0, λ > 0, R > 0, and ∀i, bi = l ∈ (0,1). Recall

that
λ is the estimated λ of (8.12) before each iteration of our RTE method and that

λ is the solution of λ to (8.12) after each iteration of our RTE method. After each
iteration in our RTE method, λ is replaced by
λ. Then, there exists |λ| → 0 along
with iterations. Specifically, when −0.9 < λ < 0:
|λ| < 0.04 (8.27)
exists after two iterations.

ri = (ai /
Proof. Since ri = ri · (λ/
λ)bi and ri = (ai /λ)bi , we can obtain λ)bi . Then, by
combining (8.14) and (8.15), we obtain the following equation:

M bi
λ
R=
ri
i=1
λ

M M
ln (
λ/
λ) M
(ln (
λ/
λ))2 2 (ln (
M
λ/
λ))3 3
=
ri +
ri bi + ri bi + ri bi . (8.28)
i=1 i=1
1! i=1
2! i=1
3!
M M
Since ∀i, bi = l ∈ (0, 1) and i=1 ri = R, there exists ri = M
i=1 i=1 ri ·
λ)bi = R · (λ/
(λ/ λ)l . Next, we can rewrite (8.28) as
l 3 l 2
λ R · l3 λ λ R · l2 λ
· ln + · ln

λ 3!

λ λ 2!

λ
l
λ R·l λ λ l
+ · ln +R · = R. (8.29)

λ 1!

λ
λ
Machine-learning-based perceptual video coding 277

0
The second iteration: –0.0378
–0.2
( λ̂ – λ) / λ
–0.4
The first iteration: –0.5785 l=1
–0.6
l = 0.5

–0.8 l = 0.01

–0.9 –0.8 –0.7 –0.6 –0.5 –0.4 –0.3 –0.2 –0.1 0

Δλ

Figure 8.3 The relationship between (

λ − λ)/λ and λ for each iteration. Note
that the first (second) iteration shown in the figure represents the lowest
convergence speed, i.e., when l = 1 and initial |λ| = 0.9 [11]

By solving this cubic equation, we can obtain:

√ √

λ =
3 3
λ · e(1+2( Z1 + Z2 ))/l , (8.30)

where
⎛ ⎞
l 2l l
1 1 λ λ λ
Z1 , Z2 = − + · ⎝−3 +2± 9 −6 +2⎠ . (8.31)
8 8 λ λ λ

Given (8.30), the relationship between (

λ − λ)/λ (which is λ for the next
iteration) and λ (which is λ for the current iteration) is illustrated in Figure 8.3.
From this figure, we can determine that |λ| → 0 in a quite fast speed. On the other
hand, the convergence speed of |λ| → 0 depends on l. When l = 1, the lowest
convergence speed of |λ| holds. In this case, λ decreases at least to 0.58 for
one iteration (when the largest initial |λ| = 0.9) and to 0.038 for two iterations. For
other cases (e.g., l = 0.5 and l = 0.01), |λ| decreases at a considerably faster speed.
Therefore, |λ| < 0.04 exists after two iterations.
This completes the proof of Lemma 8.3.

As proven in Lemma 8.2, λ < λ of our RTE method holds after the first iteration,
which means that λ ∈ (−1, 0). Moreover, we empirically found that λ for all CTUs
is restricted to (−0.9, 0) after the first iteration in HEVC-MSP. Then, Lemma 8.3
indicates that |λ| can be reduced to below 0.04 in at most three iterations, quickly
approaching 0. This verifies the fast convergence speed of the RTE method in terms
of λ. Next, we numerically evaluate the convergence speed of our RTE method in
terms of Ea .
278 Applications of machine learning in wireless communications

8.4.2 Numerical analysis

In this section, the numerical analysis of the convergence speed of our approach is pre-
sented. Specifically, we utilize the approximation error Ea to verify the convergence
speed of the RTE method. Recall that Ea = |i=1 M
ri − R|/R (defined in Section 8.3.3).
Figure 8.4 shows Ea versus RTE iterations when applying our approach to image com-
pression in the HM 16.0 platform. As shown in this figure, with no more than three
iterations, Ea reaches below 10−10 , thereby reflecting the fast convergence speed of our
RTE method. This result is in accordance with the theoretical analysis of Section 8.4.1.
We further investigate the computational time for each iteration of the RTE
method. As shown in Table 8.1, the computational time for each iteration is indepen-
dent of the image content in our RTE method. Therefore, one image was randomly
chosen from our test set, and the average time of one iteration of our RTE method
was then recorded. The computer used for the test has an Intel Core i7-4770 CPU at

100
Approximation error Ea

10–5

10–10

10–15
1 2 3 4 5 6 7 8 9 10
Iteration times
(a) Lena 512 × 512

100
Approximation error Ea

10–5

10–10

10–15
1 2 3 4 5 6 7 8 9 10
Iteration times
(b) All images of our test set

Figure 8.4 Ea versus iteration times of the RTE method at various bit rates. Note
that for (a), the black dots represent λ for each CTU in Lena image.
For (b), all 38 images (from our test set of Section 8.5) were used to
calculate the approximation error Ea and the corresponding standard
deviation along with the increasing iterations [11]
Machine-learning-based perceptual video coding 279

3.4 GHz and 16 GB of RAM. From this test, we found out that one iteration of our
RTE method only consumes approximately 0.0015 ms for each CTU. Since it takes
at most three iterations to acquire the closed-form solution, the computational time
for our RTE method is less than 0.005 ms.
Our approach consists of two parts: bit allocation and reallocation with the RTE
method. For bit allocation, three iterations are sufficient for encoding one image, thus
consuming at most 0.005 ms. For bit reallocation, the computational time depends on
the number of CTUs of the image since each CTU requires at most three iterations
to obtain the reallocated bits. For a 1, 600 × 1, 280 image, the computational time of
our approach is approximately 2.5 ms because it includes 500 CTUs. This implies the
negligible computational complexity burden of our approach.

8.5 Experimental results on single image coding

In this section, experimental results are presented to validate the performance of our
approach. Specifically, the test and parameter settings for image compression are
first presented in Section 8.5.1. Then, the R–D performance is evaluated in Section
8.5.2. In Section 8.5.3, the Bjontegaard delta bit rate (BD-rate) savings are provided
to show how many bits can be saved in our approach for image compression. Then, the
accuracy of bit-rate control is discussed in Section 8.5.4. Finally, the generalization
of our approach is verified in Section 8.5.5.

8.5.1 Test and parameter settings

To evaluate the performance of our approach, we established a test set consisting of
38 images at different resolutions. Table 8.2 summarizes all 38 of these images in our
test set. Among these images, ten images have faces, and the other images have no
faces. Saliency for these images is first detected in our approach. Note that the face
and non-face images are automatically recognized by using the face detector in [50].
Specifically, the face detector is first utilized to determine whether there is any face in
the image. For the images with detected faces, we use [50] to predict saliency, and then
we calculate SWPSNR as the optimization objective in our approach. Otherwise, [51]
is utilized to predict saliency for SWPSNR for optimization.
Since the detected salient regions may deviate from the regions attracting human
attention, in our experiments, we measure the EWPSNR of compressed images, which
adopts the ground-truth eye fixations to weight MSE. The previous work of [17] has
also verified that the EWPSNR is highly correlated with subjective quality. To obtain
the ground-truth eye fixations4 for measuring EWPSNR, 21 subjects (12 males and
9 females) with either corrected or uncorrected normal eyesight participated in our
eye-tracking experiments by viewing all images of our test set. Note that only one
among the 21 subjects was an expert who worked in the research field of saliency

4
The ground-truth eye fixations, together with their corresponding images, can be obtained from our
website at https://github.com/RenYun2016/TMM2016.
Face
From
Images

Resolution
√
Tourist

Golf

[50]
√ √
Travel

1,920×1,080
√
Doctor

√
Woman

×
Cafe

Bike

Picture01

Picture06
Table 8.2 Details of our test set [11]

1,280×1,600
JPEG XR test set
× × × ×
Picture10

×
Picture14

×
×
Picture30

Kodim01
×

Kodim02
×

Kodim03
×

Kodim05
×

Kodim06
×

Kodim07
×

Kodim08
×

Kodim11

Kodim12
× ×

Kodim13
768×512
×

Kodim14
√

Kodim15

Kodim16
Kodak test set
× ×

Kodim20
×

Kodim21
×

Kodim22
×

Kodim23
×

Kodim24
√

Kodim04
×

Kodim09

Kodim10
× ×

Kodim17
512×768
√

Kodim18
×

Kodim19
√

Tiffany
512×512
√

Lena
Standard images
Machine-learning-based perceptual video coding 281

detection. The other 20 subjects did not have any background in saliency detection,
and they were naive to the purpose of the eye-tracking experiment. Then, a Tobii
TX60 eye tracker integrated with a monitor of a 23-in. LCD display was used to
record the eye movement at a sample rate of 60 Hz. All subjects were seated on an
adjustable chair at a distance of 60 cm from the monitor of the eye tracker. Before
the experiment, the subjects were instructed to perform the 9-point calibration for the
eye tracker. During the experiment, each image was presented in a random order and
lasts for 4 s, followed by a 2-s black image for a drift correction. All subjects were
asked to freely view each image. Overall, 9,756 fixations were collected for our 38
test images.
In our experiments, our approach was implemented in HM 16.0 with the MSP
configuration profile. Then, the non-RC HEVC-MSP [9], also on the HM 16.0 plat-
form, was utilized for comparison. The RC HEVC-MSP was also compared, the RC of
which is mainly based on [52]. Note that both our approach and the RC HEVC-MSP
have integrated RC to specify the bit rates, and the other parameters in the configu-
ration profile were set by default, the same as those of the non-RC HEVC-MSP. To
obtain the target bit rates, we encoded each image with the non-RC HEVC-MSP at
six fixed QPs, the values of which are 22, 27, 32, 37, 42 and 47. Then, the target bit
rates of our approach and the RC HEVC-MSP were set to be the actual bits obtained
by the non-RC HEVC-MSP. As such, high ranges of visual quality for compressed
images can be ensured.

8.5.2 Assessment on rate–distortion performance

Now, we assess the R–D performance of our approach and of the conventional non-
RC and RC HEVC-MSP approaches. The R–D curves for face and non-face images
are first plotted and analyzed. Subsequently, we present the results of image quality
improvement of our approach at different QPs, which are measured by the EWP-
SNR and SWPSNR increase of our approach over the conventional approaches.
Next, we evaluate how ROI detection accuracy affects the quality improvement in
our approach. Finally, the subjective quality is evaluated by calculating the DMOS,
as well as showing several compressed images.
R–D curve: The first ten figures of Figures 8.5 and 8.6 show the EWPSNR and
PSNR versus bit rates for all ten face images of our test set. As shown in these figures,
our approach is able to significantly improve the EWPSNR of compressed images,
despite the slight decrease in PSNR. Consequently, subjective quality can be dra-
matically improved by our approach. Moreover, The last eight figures of Figures 8.5
and 8.6 show the curves of EWPSNR and PSNR versus bit rates for eight non-face
images randomly selected from our test set. These figures show that our approach is
also capable of achieving superior subjective quality for non-face images.
EWPSNR assessment: To quantify the R–D improvement of our approach, we
tabulate in Table 8.3 the EWPSNR enhancement of our approach over conventional
approaches. We have the following observations with regard to the EWPSNR enhance-
ment. For face images, our approach achieves significant EWPSNR improvement, as
the increase over the non-RC HEVC-MSP and RC HEVC-MSP is 2.31±1.23 dB and
282 Applications of machine learning in wireless communications
Our and EWPSNR Our and PSNR Non-RC HEVC-MSP and EWPSNR Non-RC HEVC-MSP and PSNR
PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB)

PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB)

PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB)
45 50
Tourist Golf Travel
45 45
40
40 40
35
35
Average increase: 1.2595 dB Average increase: 2.368 dB Average increase: 3.8423 dB
30 35 30
BD rate saving: 24.9553% BD rate saving: 41.1194% BD rate saving: 46.4434%
30
0.2 0.4 0.6 0.8 1 0.05 0.1 0.15 0.2 0.25 0.2 0.4 0.6 0.8 1 1.2
Bit-rates (bpp) Bit-rates (bpp) Bit-rates (bpp)
45
45 Doctor 45 Woman Kodim15
40 40
40
35 35
35 Average increase: 2.6338 dB 30 Average increase: 4.0383 dB Average increase: 2.3264 dB
BD rate saving: 52.3495% BD rate saving: 58.9304% 30 BD rate saving: 40.8396%
30 25
0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 0.2 0.4 0.6 0.8 1
Bit-rates (bpp) Bit-rates (bpp) Bit-rates (bpp)
45
Kodim04 40 Kodim18 Tiffany
40 40
35
35 35
30
Average increase: 1.3689 dB Average increase: 1.7866 dB Average increase: 1.6407 dB
30 BD rate saving: 27.4917% 25 BD rate saving: 29.2705% 30 BD rate saving: 34.8788%
0.2 0.4 0.6 0.8 1 0.5 1 1.5 0.2 0.4 0.6 0.8 1
Bit-rates (bpp) Bit-rates (bpp) Bit-rates (bpp)

Lena 40 Bike Picture14

40 40
35
35 35
Average increase: 1.8311 dB 30 Average increase: 0.92624 dB Average increase: 1.0435 dB
30 30
BD rate saving: 35.5361% 25 BD rate saving: 14.8884% BD rate saving: 18.6092%
0.2 0.4 0.6 0.8 1 0.2 0.4 0.6 0.8 1 1.2 1.4 0.2 0.4 0.6 0.8 1
Bit-rates (bpp) Bit-rates (bpp) Bit-rates (bpp)
45
Kodim02 40 Kodim06 40 Kodim07
40
35 35
35
30
30 Average increase: 2.0685 dB Average increase: 1.8057 dB 30 Average increase: 1.2884 dB
BD rate saving: 38.528% 25 BD rate saving: 26.604% BD rate saving: 18.2286%
25 25
0.2 0.4 0.6 0.8 1 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8
Bit-rates (bpp) Bit-rates (bpp) Bit-rates (bpp)

40 Kodim10 Kodim16 40 Kodim24

40
35 35 35

30 Average increase: 1.118 dB 30 Average increase: 1.563 dB 30 Average increase: 0.87117 dB

BD rate saving: 19.3336% BD rate saving: 28.7228% 25 BD rate saving: 14.5608%
25 25
0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.2 0.4 0.6 0.8 1 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6
Bit-rates (bpp) Bit-rates (bpp) Bit-rates (bpp)

Figure 8.5 EWPSNR and PSNR versus bit rates for our approach and the non-RC
HEVC-MSP [11]

2.47±1.20 dB, respectively. In addition, the maximum increase of EWPSNR is 5.75

and 6.30 dB in our approach over the non-RC and RC HEVC-MSP approaches,
respectively, whereas the minimum increase is 0.39 and 0.71 dB for these two
approaches, respectively. For non-face images, the EWPSNR improvement of our
approach reaches 1.49 dB on average compared with the RC HEVC-MSP approach,
with a standard deviation of 0.70 dB. Compared to the non-RC HEVC-MSP approach,
our approach enhances the EWPSNR by 1.21 dB on average, and the standard devi-
ation of this enhancement is 0.61 dB. In a word, our approach dramatically improves
the EWPSNR over the conventional approaches for both face and non-face images.
SWPSNR assessment: Since the optimization objective of our approach is to
maximize SWPSNR, we further report in Table 8.3 the SWPSNR improvement of
our approach over the conventional approaches. As shown in Table 8.3, our approach
Machine-learning-based perceptual video coding 283
PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) Our and EWPSNR Our and PSNR RC HEVC-MSP and EWPSNR RC HEVC-MSP and PSNR

PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB)

PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB) PSNR and EWPSNR (dB)
45 50
Tourist Travel
45 Golf 45
40
40 40
35
35
Average increase: 1.6706 dB Average increase: 1.9525 dB Average increase: 3.7202 dB
30 35 30
BD rate saving: 30.8758% BD rate saving: 41.0559% BD rate saving: 45.9429%
30
0.2 0.4 0.6 0.8 1 0.05 0.1 0.15 0.2 0.25 0.3 0.2 0.4 0.6 0.8 1 1.2
Bit-rates (bpp) Bit-rates (bpp) Bit-rates (bpp)
45
45 Doctor 45 Woman Kodim15
40 40
40
35 35
35 Average increase: 3.1595 dB 30 Average increase: 4.0712 dB Average increase: 2.6205 dB
BD rate saving: 56.3626% BD rate saving: 54.5463% 30 BD rate saving: 46.817%
30 25
0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 0.2 0.4 0.6 0.8 1
Bit-rates (bpp) Bit-rates (bpp) Bit-rates (bpp)
45
Kodim04 40 Kodim18 Tiffany
40 40
35
35 35
30
Average increase: 1.3912 dB Average increase: 2.0058 dB Average increase: 1.9259 dB
30 BD rate saving: 30.7179% 25 30 BD rate saving: 37.5514%
BD rate saving: 40.3616%
0.2 0.4 0.6 0.8 1 0.5 1 1.5 0.2 0.4 0.6 0.8 1
Bit-rates (bpp) Bit-rates (bpp) Bit-rates (bpp)

Lena 40 Bike Picture14

40 40
35
35 35
Average increase: 2.2044 dB 30 Average increase: 1.295 dB Average increase: 1.1705 dB
30 30
BD rate saving: 41.0319% 25 BD rate saving: 16.7768% BD rate saving: 20.6045%
0.2 0.4 0.6 0.8 1 0.2 0.4 0.6 0.8 1 1.2 1.4 0.2 0.4 0.6 0.8 1 1.2
Bit-rates (bpp) Bit-rates (bpp) Bit-rates (bpp)
45
Kodim02 40 Kodim06 40 Kodim07
40
35 35
35
30
30 Average increase: 2.5219 dB Average increase: 2.1546 dB 30 Average increase: 1.605 dB
BD rate saving: 40.3722% 25 BD rate saving: 31.6989% BD rate saving: 21.0092%
25 25
0.2 0.4 0.6 0.8 1 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8
Bit-rates (bpp) Bit-rates (bpp) Bit-rates (bpp)

40 Kodim10 Kodim16 40 Kodim24

40
35 35 35

30 Average increase: 1.4464 dB 30 Average increase: 2.0432 dB 30 Average increase: 0.71958 dB

BD rate saving: 24.2348% BD rate saving: 34.7002% 25 BD rate saving: 12.8462%
25 25
0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.2 0.4 0.6 0.8 1 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6
Bit-rates (bpp) Bit-rates (bpp) Bit-rates (bpp)

Figure 8.6 EWPSNR and PSNR versus bit rates for our approach and the RC
HEVC-MSP [11]

also achieves significant improvements in SWPSNR at different QPs. Specifically,

compared with RC HEVC-MSP, our approach achieves an SWPSNR improvement
over all images, with up to a 4.14 dB SWPSNR enhancement for face images and
up to a 2.14 dB enhancement for non-face images. On average, for non-face images,
our approach increases the SWPSNR by 0.72 and 1.00 dB over non-RC and RC
HEVC-MSP, respectively. For face images, a more average SWPSNR gain is obtained
by our approach, which has 1.56 and 1.67 dB increase over non-RC and RC
HEVC-MSP.
Influence of ROI detection accuracy: Now, we investigate how the ROI detec-
tion accuracy influences the results of quality improvement in our approach. To this
end, we further implement our approach using EWPSNR (instead of SWPSNR) as the
optimization objective, which means that ROI detection is of 100% accuracy when
Table 8.3 EWPSNR and SWPSNR improvement of our approach over non-RC and RC HEVC-MSP approaches, for the 38 images [11]

Face Non-face

SWPSNR improvement EWPSNR improvement SWPSNR improvement EWPSNR improvement

Avg. ± Std. Max./Min. Avg. ± Std. Max./Min. Avg. ± Std. Max./Min. Avg. ± Std. Max./Min.

QP = 47 Over non-RC 1.10 ± 0.47 2.05/0.44 1.55 ± 0.79 2.93/0.39 0.44 ± 0.19 0.95/0.14 0.71 ± 0.43 1.91/0.04
Over RC 1.19 ± 0.52 2.21/0.65 1.67 ± 0.86 2.87/0.71 0.90 ± 0.40 1.84/0.25 1.15 ± 0.55 2.51/0.24
QP = 42 Over non-RC 1.21 ± 0.43 1.83/0.39 1.71 ± 0.79 2.84/0.47 0.58 ± 0.23 1.17/0.18 0.92 ± 0.42 2.13/0.15
Over RC 1.43 ± 0.55 2.43/0.55 1.99 ± 0.80 2.98/1.07 0.97 ± 0.45 1.74/0.31 1.34 ± 0.62 2.74/0.23
QP = 37 Over non-RC 1.29 ± 0.38 1.95/0.72 1.92 ± 0.93 3.64/0.67 0.71 ± 0.29 1.23/0.25 1.16 ± 0.46 2.21/0.31
Over RC 1.42 ± 0.50 2.40/0.90 2.16 ± 0.92 3.56/0.80 1.00 ± 0.50 1.96/0.25 1.47 ± 0.65 2.83/0.51
QP = 32 Over non-RC 1.51 ± 0.51 2.48/0.67 2.23 ± 1.08 4.20/1.04 0.81 ± 0.34 1.32/0.24 1.35 ± 0.54 2.40/0.27
Over RC 1.57 ± 0.52 2.49/0.95 2.38 ± 1.10 4.18/0.91 0.99 ± 0.50 1.99/0.21 1.56 ± 0.68 2.90/0.36
QP = 27 Over non-RC 1.90 ± 0.73 3.26/0.79 2.85 ± 1.37 5.41/1.66 0.86 ± 0.36 1.48/0.33 1.49 ± 0.61 2.66/0.10
Over RC 2.01 ± 0.65 3.14/1.01 2.98 ± 1.25 5.16/1.73 0.97 ± 0.47 2.13/0.36 1.58 ± 0.73 2.77/0.23
QP = 22 Over non-RC 2.38 ± 0.92 4.14/1.26 3.60 ± 1.21 5.75/2.17 0.92 ± 0.38 1.54/0.40 1.62 ± 0.69 3.07/0.12
Over RC 2.42 ± 1.05 4.14/1.17 3.65 ± 1.21 6.30/2.07 1.15 ± 0.51 2.14/0.39 1.85 ± 0.82 3.60/0.08
Overall Over non-RC 1.56 ± 0.73 4.14/0.39 2.31 ± 1.23 5.75/0.39 0.72 ± 0.34 1.54/0.14 1.21 ± 0.61 3.07/0.04
Over RC 1.67 ± 0.76 4.14/0.55 2.47 ± 1.20 6.30/0.71 1.00 ± 0.47 2.14/0.21 1.49 ± 0.70 3.60/0.08
Machine-learning-based perceptual video coding 285

Table 8.4 EWPSNR difference (dB) of our approach after replacing SWPSNR with
EWPSNR as the optimization objective [11]

QP 47 42 37 32 27 22 Overall

Face 0.72 0.77 0.67 0.66 0.57 0.45 0.64

Non-face 0.70 0.77 0.84 0.92 0.98 1.01 0.87

compressing images using our approach. Specifically, Table 8.4 shows the EWPSNR
difference averaged over all 38 test images when replacing SWPSNR with EWPSNR
as the optimization objective in our approach. This reflects the influence of ROI
detection accuracy on the quality improvement of our approach. We can see from
Table 8.4 that the EWPSNR of our approach can be enhanced by 0.64 and 0.87 dB on
average for face and non-face images after replacing SWPSNR by EWPSNR as the
optimization objective. Thus, visual quality can be further improved in our approach
when ROI detection is more accurate.
Subjective quality evaluation: Next, we compare our approach with the non-
RC HEVC-MSP using DMOS. Note that the DMOS of the RC HEVC-MSP is not
evaluated in our test because it produces even worse visual quality than the non-RC
HEVC-MSP. The DMOS test was conducted by the means of single stimulus contin-
uous quality score, which is processed by Rec. ITU-R BT.500 to rate the subjective
quality. The total number of subjects involved in the test is 12, consisting of 6 males and
6 females. Here, a Sony BRAVIA XDV-W600, with a 55-in. LCD, was utilized for dis-
playing the images. The viewing distance was set to be four times the image height for
rational evaluation. During the experiment, each image was displayed for 4 s, and the
order in which the images were displayed was random. Then, the subjects were asked
to rate after each image was displayed, i.e., excellent (100–81), good (80–61), fair
(60–41), poor (40–21) and bad (21–0). Finally, DMOS was computed to qualify the
difference in subjective quality between the compressed and uncompressed images.
The DMOS results for the face images are tabulated in Table 8.5. Smaller values
of DMOS indicate better subjective quality. As shown in Table 8.5, our approach
has considerably better subjective quality than the non-RC HEVC-MSP at all bit
rates. Note that for all images, the DMOS values of our approach at QP = 47 are
almost equal to those of the non-RC HEVC-MSP at QP = 42, which approximately
doubles the bit rates of QP = 47. This indicates that a bit rate reduction of nearly half
can be achieved in our approach. This result is also in accordance with the ∼40%
BD-rate saving of our approach (to be discussed in Section 8.5.3). We further show
in Figure 8.7 Lena and Kodim18 compressed by our and the other two approaches.
Obviously, our approach, which incorporates the saliency-detection method of [50],
is able to significantly meliorate the visual quality over face regions (that humans
mainly focus on). Consequently, our approach yields significantly better subjective
quality than the non-RC and RC HEVC-MSP for face images.
In addition, the DMOS results of those eight non-face images are listed in
Table 8.6. Again, our approach is considerably superior to the non-RC HEVC-MSP
Table 8.5 DMOS results for face images between our approach and the non-RC HEVC-MSP [11]

Tourist Golf Travel Doctor Woman Kodim15 Kodim04 Kodim18 Tiffany Lena

QP = 47 Bits (bpp) 0.04 0.02 0.04 0.02 0.04 0.03 0.03 0.05 0.03 0.05
Our 57.2 58.0 56.9 56.5 61.4 64.5 68.9 55.0 59.2 57.5
Non-RC 74.3 69.6 69.1 63.9 78.4 70.1 73.9 66.3 67.6 63.9
QP = 42 Bits (bpp) 0.08 0.03 0.10 0.03 0.13 0.06 0.06 0.16 0.06 0.09
Our 45.0 50.0 42.7 47.8 43.9 50.7 53.6 43.1 43.1 47.9
Non-RC 58.5 56.3 53.7 52.1 61.3 61.2 61.9 56.9 54.1 55.5
QP = 32 Bits (bpp) 0.27 0.08 0.36 0.10 0.56 0.29 0.31 0.76 0.26 0.28
Our 28.1 35.2 26.1 34.1 28.9 30.0 30.0 20.8 27.1 36.9
Non-RC 36.4 42.0 34.0 42.3 36.0 38.7 38.8 28.5 30.2 44.0

Note: The bold values mean the best subjective quality per test QP and test image.
Machine-learning-based perceptual video coding 287

Face Face Face

(a) (b) (c) (d)

Figure 8.7 Subjective quality of Lena and Kodim18 images at both 0.05 bpp
(QP = 47) for three approaches [11]: (a) human fixations, (b) non-RC
HEVC-MSP, (c) RC HEVC-MSP and (d) our

approach at all bit rates. Moreover, Figure 8.8 shows two images Kodim06 and
Kodim07 compressed by our approach and by the other two approaches. From this
figure, we can see that our approach improves the subjective quality of compressed
images, as the fixated regions are with higher quality.

8.5.3 Assessment of BD-rate savings

It is interesting to investigate how many bits can be saved when applying our approach
to image compression. In our experiments, BD-rates were calculated for this investi-
gation. To calculate the BD-rates, the six different bit rates, each of which corresponds
to one fixed QP (among QP = 22, 27, 32, 37, 42, and 47), were all utilized. Since the
above section has shown that the EWPSNR is more effective than the PSNR for eval-
uating subjective quality, the EWPSNRs of each image at six bit rates were measured
as the distortion metric. Given the bit rates and their corresponding EWPSNRs, the
BD rate of each image was achieved. Then, the BD-rate savings of our approach can
be obtained, with the non-RC or RC HEVC-MSP as an anchor.
Table 8.7 reports the BD-rate savings of our approach averaged over all 38
images of our test set. As shown in this table, a 24.3% BD-rate saving is achieved
in our approach for all images over the non-RC HEVC-MSP. The BD-rate saving
of our approach increases to 27.7%, when compared with the RC HEVC-MSP. In
Table 8.7, the results of BD-rate savings for face and non-face images are also listed.
Accordingly, we can see that our approach is able to save 39.1% and 42.5% BD-rates
over non-RC and RC HEVC-MSP, respectively. Note that compared with non-face
images, face images witness more gains in our approach. It is probably due to the fact
Table 8.6 DMOS results for non-face images between our approach and the non-RC HEVC-MSP [11]

Bike Picture14 Kodim02 Kodim06 Kodim07 Kodim10 Kodim16 Kodim24

QP = 47 Bits (bpp) 0.07 0.04 0.02 0.04 0.05 0.03 0.02 0.06
Our 53.3 59.6 65.5 62.0 56.8 63.0 71.1 67.1
Non-RC 57.2 63.1 69.9 72.1 67.0 68.1 79.2 70.2
QP = 42 Bits (bpp) 0.14 0.10 0.04 0.12 0.10 0.08 0.06 0.17
Our 36.8 50.3 50.0 52.7 50.1 54.5 56.2 55.4
Non-RC 38.9 54.2 53.4 57.6 56.3 58.7 62.1 59.3
QP = 32 Bits (bpp) 0.49 0.40 0.26 0.60 0.33 0.28 0.36 0.71
Our 30.3 31.7 33.5 34.8 36.3 34.7 35.6 32.6
Non-RC 30.8 32.6 35.2 35.6 38.0 37.9 40.8 33.8

Note: The bold values mean the best subjective quality per test QP and test image.
Machine-learning-based perceptual video coding 289

(a) (b) (c) (d)

Figure 8.8 Subjective quality of Kodim06 and Kodim07 image at 0.04 and
0.05 bpp (QP = 47) for three approaches [11]: (a) human fixations,
(b) non-RC HEVC-MSP, (c) RC HEVC-MSP and (d) our

Table 8.7 BD-rate savings and encoding time ratio of our

approach over non-RC and RC HEVC-MSP [11]

Over non-RC Over RC

HEVC-MSP HEVC-MSP

Face images (%) 39.18 42.50

Non-face images (%) 18.98 22.43
All generic images (%) 24.30 27.72
Encoding time (%) 108.3 105.2

that human faces are more consistent than other objects in attracting human attention.
Meanwhile, in our approach, the saliency of face images can be better predicted than
that of non-face images. Consequently, the ROI-based compression of face images
by our approach is more effective in satisfying human perception, resulting in larger
improvements in EWPSNR, BD-rate savings and DMOS scores.
As a result of BD-rate saving, the computational time of our approach increases,
which is also reported in Table 8.7. Specifically, our approach increases the encoding
time by approximately 8% and 5% over non-RC and RC HEVC-MSP, respectively.
The computational time of our approach mainly comes from three parts, i.e., saliency
detection, pre-compression and RTE optimization. As discussed above (Sections 8.3.1
and 8.4.2), our pre-compression process slightly increases the computational cost
by ∼3%, while our RTE method consumes negligible computational time. Besides,
saliency detection, which is the first step in our approach, consumes ∼2% extra time.

8.5.4 Assessment of control accuracy

The control accuracy is another factor in evaluating the performance of RC-related
image compression. Here, we compare the control accuracy of our approach and of
290 Applications of machine learning in wireless communications

the RC HEVC-MSP over all images in our test set. Since the bit reallocation process
is developed in our approach to bridge the gap between the target and actual bits, the
control accuracy of our approach with and without the bit reallocation process is also
compared. In the following, the control accuracy is evaluated from two aspects: the
CTU level and the image level.
For the evaluation of control accuracy at the CTU level, we compute the bit-rate
error of each CTU, i.e., the absolute difference between target and actual bits assigned
to one CTU. Then, Figure 8.9 demonstrates the heat maps of bit-rate errors at the CTU
level averaged over all images with the same resolutions from the Kodak and JPEG
XR sets. The heat maps of our approach and of the RC HEVC-MSP are both shown in
Figure 8.9. It can easily be observed that our approach ensures a considerably smaller
bit-rate error for almost all CTUs when compared with the RC HEVC-MSP. Note that
the accurate rate control at the CTU level is meaningful because it ensures that the
bit consumption follows the amount that it is allocated, satisfying the subjective R–D
optimization formulation of (8.6). As a result, the bits in our approach can be accu-
rately assigned to ROIs with optimal subjective quality. In contrast, the conventional
RC HEVC-MSP normally accumulates redundant bits at the end of image bitstreams,
resulting in poor performance in R–D optimization.
For the evaluation of control accuracy at the image level, the bit-rate error, defined
as the absolute difference between the target and actual bits of the compressed image,
is worked out. Figure 8.10 shows the bit-rate errors of all 38 images from our test set in
terms of maximum, minimum, average and standard deviation values. As shown in this
figure, our approach achieves smaller bit-rate error than the RC HEVC-MSP from the
aspects of mean, standard deviation, maximum and minimum values. This verifies
the effectiveness of our approach in RC and also makes our approach more practical
because the accurate bit allocation of our approach well meets the bandwidth or storage
requirements. Furthermore, Figure 8.10 shows that the bit-rate error significantly
increases from 1.43% to 6.91% and also dramatically fluctuates once bit reallocation
is disabled in our approach. This indicates the effectiveness of the bit-reallocation
process in our approach. Note that because a simple reallocation process is also
adopted in the RC HEVC-MSP, the bit-rate errors of RC HEVC-MSP are also much
smaller than those of our approach without bit reallocation.
In summary, our approach has more accurate RC at both the CTU and image
levels compared to the RC HEVC-MSP.

8.5.5 Generalization test

To verify the generalization of our approach, we further compare our approach and
conventional approaches on 112 raw images from 3 test sets grouped into 4 categories,
i.e., 22 face images, 41 non-face images, 4 graphics images and 45 aerial images. The
resolutions of these images range from 256 × 256 to 7,216 × 5,408. The experimental
results on these 112 images are reported in Table 8.8, including the mean, standard
deviation, maximum and minimum values of SWPSNR3 as well as bit-rate errors.
Due to space limitation, this table only shows the results of compression at QP = 32
and the overall results of compression at QP = 22, 27, 32, 37, 42 and 47.
Machine-learning-based perceptual video coding 291

Our RC HEVC-MSP
1 1 1
2 2 0.8
3 3
4 4 0.6
5 5
0.4
6 6
7 7 0.2
8 8
1 2 3 4 5 6 7 8 9 10 11 12 1 2 3 4 5 6 7 8 9 10 11 12
(a)
Our RC HEVC-MSP
1 1
2 2 0.6
3 3
4 4 0.5
5 5
6 6 0.4
7 7
8 8 0.3
9 9
0.2
10 10
11 11 0.1
12 12
1 2 3 4 5 6 7 8 1 2 3 4 5 6 7 8
(b)
Our RC HEVC-MSP
1 1
2 2 2.2
3 3
4 4 2
5 5
6 6 1.8
7 7
8 8
9 9 1.6
10 10
11 11 1.4
12 12
13 13 1.2
14 14
15 15 1
16 16
17 17 0.8
18 18
19 19 0.6
20 20
21 21
22 22 0.4
23 23
24 24 0.2
25 25
1 2 3 4 5 6 7 8 9 1011121314151617181920 1 2 3 4 5 6 7 8 9 1011121314151617181920

(c)

Figure 8.9 Heat maps of bit-rate errors at CTU level for our approach and RC
HEVC-MSP. Each block in this figure indicates the bit-rate error of one
CTU. Note that the bit-rate errors are obtained via averaging all
images compressed by our and the RC HEVC MSP at six different bit
rates (corresponding to QP = 22, 27, 32, 37, 42, 47) [11]. (a) Kodak
768 × 512, (b) Kodak 512 × 768, (c) JPEG XR 1,280 × 1,600
292 Applications of machine learning in wireless communications
20 Avg±Std: 2.33%±2.80% RC HEVC-MSP Our approach Our approach without bit re-allocation
Max/Min: 9.83% / 0.03%
Avg±Std: 1.43%±1.24% Avg±Std: 6.91%±4.92%
15 Max/Min: 4.98% / 0.03% Max/Min: 17.15% / 0.07%
Bit-rate error (%)

Ko im0
Ko im0
Ko m0
Ko m0
Ko m0
Ko m08
Ko m1
Ko im1
Ko m13
Ko m14
Ko im1
Ko m16
Ko im2
Ko m2
Ko im2
Ko m23
Ko m2
Ko m04
Ko im0
Ko m10
Ko im1
Ko m18
Ti im1
Le any
To
Go ist
Tr
Do el
W or
Ca an
Bi
Pi
Pi re0
Pi re0
Pi re1
Pi re1
Ko re30
Ko im0

ff 9
ctu
ctu 1
ctu 6
ctu 0
ctu 4
om
av

na
ke
ur

d 2
di 3
di 5
di 6
di 7
di
d 1
di 2
di
d
di 5
d
di 0
d 1
di 2
di
di 4
d
di 9
d
di 7
d
fé
lf

d
d 1

Figure 8.10 The bit-rate errors of each single image for our approach with and
without bit reallocation, as well as the RC HEVC-MSP. The maximum,
minimum, average and standard deviation values over all images are
also provided [11]

As shown in Table 8.8, our approach still dramatically outperforms the conven-
tional approaches across different categories of images in terms of both quality and RC
error. Specifically, the SWPSNR improvement on the newly added 112 images is sim-
ilar to that on the above 38 test images. In particular, when compressing face images
at 6 QPs, our approach has 1.50 ± 0.84 dB SWPSNR increase over the conventional
RC HEVC-MSP. Moreover, the average increase in SWPSNR at six QPs is 0.75 dB
for non-face images, 0.83 dB for graphic images and 0.60 dB for aerial images. For
control accuracy, the average bit-rate errors of our approach stabilize at 1.84%–3.74%
across different categories, while the conventional RC approach in HEVC fluctuates
from 4.08% to 12.40% on average with an even larger standard deviation. This result
validates that our approach can achieve a stable and accurate RC, compared to RC
HEVC-MSP. Finally, the generalization of our approach can be validated.

8.6 Experimental results on video coding

In this section, we present an implementation of the RTE method in perceptual video

coding, i.e., to optimize the subjective quality at a given bit rate for panoramic videos.
Several saliency-detection methods for panoramic videos can be employed in our
framework [55,56]. We adopt the S-PSNR of [55] in this chapter as an example.
Based on the definition of S-PSNR, the sphere-based distortion is the sum of square
error between pixels sampled from sphere:

di = (S(xn , yn ) − S (xn , yn ))2 , (8.32)
n∈Ci

where Ci is the set of pixels belonging to the ith CTU. Therefore, at target bit rate R,
the optimization on S-PSNR can be formulated by
M

M
min di s.t. ri = R. (8.33)
i=1 i=1
Table 8.8 Performance improvement of our approach over non-RC and RC HEVC-MSP approaches, for 112 test images belonging to
different categories [11]

QP SWPSNR improvement (dB) Face Non-face Graphics Aerial All

32 Over non-RC Avg. ± Std. 1.14 ± 0.03 0.54 ± 0.40 0.51 ± 0.28 0.35 ± 0.30 0.58 ± 0.50
Max./Min. 2.82/0.11 1.60/0.02 0.81/0.14 1.04/0.00 2.82/0.00
Over RC Avg. ± Std. 1.40 ± 0.76 0.73 ± 0.49 0.65 ± 0.13 0.59 ± 0.61 0.80 ± 0.66
Max./Min. 2.85/0.17 1.80/0.01 0.83/0.53 2.97/0.00 2.97/0.00
All Over non-RC Avg. ± Std. 1.25 ± 0.71 0.53 ± 0.45 0.50 ± 0.21 0.30 ± 0.33 0.58 ± 0.58
Max./Min. 3.30/0.01 3.35/0.00 0.90/0.14 2.28/0.01 3.35/0.00
Over RC Avg. ± Std. 1.50 ± 0.84 0.75 ± 0.59 0.83 ± 0.68 0.60 ± 0.52 0.84 ± 0.71
Max./Min. 4.59/0.01 3.13/0.01 2.88/0.06 2.97/0.01 4.59/0.01

Bit-rate error (%) Face Non-face Graphics Aerial Overall

32 RC HEVC-MSP Avg. ± Std. 2.40 ± 2.76 3.53 ± 9.11 6.43 ± 9.80 6.93 ± 9.09 4.78 ± 8.39
Max./Min. 10.9/0.06 53.65/0.01 20.99/0.47 35.07/0.02 53.65/0.01
Our Avg. ± Std. 2.72 ± 2.62 2.80 ± 4.15 1.89 ± 1.69 1.63 ± 3.34 2.28 ± 3.51
Max./Min. 12.11/0.36 25.45/0.03 4.42/0.85 20.38/0.06 25.45/0.03
All RC HEVC-MSP Avg. ± Std. 4.08 ± 5.51 7.96 ± 15.64 11.96 ± 21.20 12.40 ± 16.29 9.12 ± 15.07
Max./Min. 33.61/0.04 98.81/0.00 86.00/0.12 69.12/0.00 98.81/0.00
Our Avg. ± Std. 3.37 ± 3.63 3.74 ± 5.71 2.17 ± 1.85 1.84 ± 3.03 2.85 ± 4.37
Max./Min. 25.79/0.10 39.32/0.01 7.00/0.31 21.39/0.00 39.32/0.00
294 Applications of machine learning in wireless communications

In (8.33), rm is the assigned bits at the mth CTU, and M is the total number of CTUs
in the current frame. To solve the above formulation, a Lagrange multiplier λ is
introduced, and (8.33) can be converted to an unconstrained optimization problem:

M
min J = (di + λri ). (8.34)
{ri }M
i=1 i=1

Here, we define J as the value of R–D cost. By setting derivative of (8.34) to zero,
minimization on J can be achieved by

M
∂J ∂ i=1 (di + λri )
=
∂ri ∂ri
∂di
= +λ
∂ri
= 0. (8.35)

Next, we need to model the relationship between distortion di and bit rate ri , for
solving (8.35). Note that di and ri are equivalent to S-MSE and bpp divided by the
number of pixels in a CTU, respectively. Similar to [37], we use the hyperbolic model
to investigate the relationship between sphere-based distortion S-MSE and bit-rate
bpp, on the basis of four encoded panoramic video sequences. Figure 8.11 plots the
fitting R–D curves using the Hyperbolic model, for these four sequences. In this
figure, bpp is calculated by
R
bpp = , (8.36)
f ×W ×H
where f means frame rate, and W and H stand for width and height of video, respec-
tively. Figure 8.11 shows that the Hyperbolic model is capable of fitting on the
relationship between S-MSE [55] and bpp, and R-square for the fitting curves of
four sequences are all more than 0.99. Therefore, the hyperbolic model is used in our
RC scheme as follows:

di = ci · (ri )−ki , (8.37)

where cm and km are the parameters of the hyperbolic model that can be updated for
each CTU using the same way as [11].
The above equation can be rewritten by
∂di (−k −1)
− = ci · ki · ri i . (8.38)
∂ri
Given (8.35) and (8.38), the following equation holds:

ci ki (1/(ki +1))
ri = . (8.39)
λ
Machine-learning-based perceptual video coding 295
Fengjing_1 Tiyu_1
25
S-MSE vs. bpp S-MSE vs. bpp
60 Hyperbolic fitting Hyperbolic fitting

50 S-MSE = 0.2534 bpp–0.7312 20 S-MSE = 0.231 bpp

–0.5683

R-square = 0.9988 R-square = 0.9956

S-MSE

S-MSE
40
15
30
10
20

10
5
1 2 3 4 5 6 7 8 0.5 1 1.5 2 2.5
–3
bpp × 10 bpp × 10–3

Dianying Hangpai_2

S-MSE vs. bpp 70 S-MSE vs. bpp

Hyperbolic fitting Hyperbolic fitting
10
60
–0.7283 –0.6562
S-MSE = 0.03365 bpp S-MSE = 2.916 bpp
8 R-square = 0.9980 50 R-square = 0.9997
S-MSE
S-MSE

40
6
30
4 20

10
2
0.5 1 1.5 2 2.5 3 0.02 0.04 0.06 0.08 0.1 0.12
–3 bpp
bpp × 10

Figure 8.11 R–D fitting curves using the hyperbolic model. Note that these four
sequences are encoded by HM 15.0 with the default low delay P
profile. The bit rates are set as the actual bit rates when
compressing at four fixed QP (27, 32, 37, 42), to be described in
Section 8.6.1.1 [55]

Moreover, according to (8.33), we have the following constraint:

M
ri = R. (8.40)
i=1

Upon (8.39) and (8.40), the bit allocation for each CTU can be formulated as follows:
M
ci · ki (1/(ki +1))
M
ri = = R. (8.41)
i=1 i=1
λ
Therefore, once (8.41) is solved, target bit ri can be obtained for each CTU, with
maximization on S-PSNR. In this chapter, we apply the RTE method [49] in solving
(8.41) with the closed-form solution.
After obtaining the optimal bit-rate allocation, quantization parameter (QP) of
each CTU can be estimated using the method of [37]. Figure 8.12 summarizes the
overall procedure of our RC scheme for panoramic video coding. Note that our RC
scheme is mainly applicable for the latest HEVC-based panoramic video coding, and
296 Applications of machine learning in wireless communications

cm, km
Current frame t
rm QPm
Bit allocation QP estimation Encoding
Panoramic sequence
Bit stream
Update parameters 0 0 1 1 1 0 1 0

cm, km
Next frame t + 1
rm QPm
Bit allocation QP estimation Encoding

Figure 8.12 The framework of the proposed RC scheme for panoramic video
coding [55]

it can be extended to other video-coding standards by reinvestigating the hyperbolic

model of bit rate and distortion.

8.6.1 Experiment
In this section, experiments are conducted to validate the effectiveness of our RC
scheme. Section 8.6.1.1 presents the settings for our experiments. Section 8.6.1.2
evaluates our approach from aspects of R–D performance, BD-rate and Bjontegaard
delta S-PSNR (BD-PSNR). Section 8.6.1.3 discusses the RC accuracy of our scheme.

8.6.1.1 Settings
Due to space limitation, eight panoramic video sequences at 4K are chosen from
the test set of IEEE 1857 working group in our experiments. They are shown in
Figure 8.13. These sequences are all at 30 fps with duration of 10 s. Figure 8.13
shows that the contents of these sequences, which vary from indoor to outdoor scenes
and contain people and landscapes. Then, these panoramic video sequences are com-
pressed by the HEVC reference software HM-15.0. Here, we implement our RC
scheme in HM-15.0, and then compare our scheme with the latest R–λ RC scheme [37]
that is default RC setting of HM-15.0. For HM-15.0, the Low Delay P setting is applied
with the configuration file encoder lowdelay P main.cfg. The same as [37], we first
compress panoramic video sequences using the conventional HM-15.0 at four fixed
QPs, which are 27, 32, 37 and 42. Then, the obtained bit rates are used to set the
target bit rates of each sequence for both our and conventional [37] schemes. It is
worth pointing out that we only compare with the state-of-the-art RC scheme [37] of
HEVC for 2D video coding, since there exists no RC scheme for panoramic video
coding.

8.6.1.2 Evaluation on R–D performance

R–D curves. We compare the R–D performance of our and the conventional RC [37]
schemes using S-PSNR in Y channel. We plot in Figure 8.14 the R–D curves of
all test panoramic video sequences, for both our and the conventional RC schemes.
Machine-learning-based perceptual video coding 297

(a) (b) (c) (d)

(e) (f) (g) (h)

Figure 8.13 Selected frames from all test panoramic video sequences [55]:
(a) Fengjing_1 (4,096 × 2,048), (b) Tiyu_1 (4,096 × 2,048),
(c) Yanchanghui_2 (4,096 × 2,048), (d) Dianying (4,096 × 2,048),
(e) Hangpai_1 (4,096 × 2,048), ( f ) Hangpai_2 (4,096 × 2,048),
(g) AerialCity (3,840 × 1,920), (h) DrivingInCountry (3,840 × 1,920)

We can see from these R–D curves that our scheme achieves higher S-PSNR than [37]
at the same bit rates, for all test sequences. Thus, our RC scheme is superior to [37]
in R–D performance.
BD-PSNR and BD-rate. Next, we quantify R–D performance in terms of BD-
PSNR and BD-rate. Similar to the above R–D curves, we use S-PSNR inY channel for
measuring BD-PSNR and BD-rate. Table 8.9 reports the BD-PSNR improvement of
our scheme over [37]. As can be seen from this table, our scheme averagely improves
0.1613 dB in BD-PSNR over [37]. Such improvement is mainly because our scheme
aims at optimizing S-PSNR, while [37] deals with optimization on PSNR. Table 8.9
also tabulates the BD-rate saving of our RC scheme with [37] being an anchor. We
can see that our RC scheme is able to save 5.34% BD-rate in average, when compared
with [37]. Therefore, our scheme has potential in relieving the bandwidth-hungry
issue posed by panoramic videos.
Subjective quality. Furthermore, Figure 8.15 shows visual quality of one
selected frame of sequence Dianying, encoded by HM-15.0 with our and conven-
tional RC schemes at the same bit rate. We can observe that our scheme yields better
visual quality than [37], with smaller blurring effect and less artifacts. For exam-
ple, both regions of fingers and light generated by our scheme is much more clearer
than those by [37]. Besides, the region of the leg encoded with our RC scheme has
less blurring effect, compared to [37]. In summary, our scheme outperforms [37]
in R–D performance, evaluated by R–D curves, BD-PSNR, BD-rate and subjective
quality.

8.6.1.3 Evaluation on RC accuracy

Now, we evaluate the RC accuracy of our scheme. For such evaluation, Table 8.10
illustrates the error rate of actual bit rate with respect to target bit rate, for both our
and the conventional RC [37] schemes. We can see from this table that the average RC
error rate is less than 1‰, comparable to the error rate of [37]. Besides, the maximum
error rate for our RC scheme is 3.02‰ for sequence Tiyu 1, and the error rate of [37]
Fengjing_1 Tiyu_1
39 41
Averaged Y-SPSNR (dB)

Averaged Y-SPSNR (dB)

38
37 40
36 39
35 38
34
33 37
32 36
31 Conventional Conventional
30 35
Our Our
29 34
0 500 1,000 1,500 2,000 2,500 0 200 400 600 800
(a) Bit-rates (kbps) (b) Bit-rates (kbps)

Yanchanghui_2 Dianying
45

Averaged Y-SPSNR (dB)

41
44
40
39 43
38 42
37 41
36 40
35 39
Conventional Conventional
34 38
Our Our
33 37
0 500 1,000 1,500 2,000 0 200 400 600 800 1,000

(c) Bit-rates (kbps) (d) Bit-rates (kbps)

Hangpai_1 Hangpai_2
38
Averaged Y-SPSNR (dB)
Averaged Y-SPSNR (dB)

40
39 37
38 36
37 35
36 34
35 33
34 32
33 Conventional 31 Conventional
32 30
Our Our
31 29
0 2,000 4,000 6,000 8,000 10,000 0 5,000 10,000 15,000 20,000 25,000 30,000 35,000

(e) Bit-rates (kbps) (f) Bit-rates (kbps)

AerialCity DrivingInCountry
39.5 38
Averaged Y-SPSNR (dB)

Averaged Y-SPSNR (dB)

38.5 37
37.5 36
35
36.5
34
35.5 33
34.5 32
33.5 31
Conventional Conventional
32.5 30
Our Our
31.5 29
0 1,000 2,000 3,000 4,000 0 5,000 10,000 15,000 20,000

(g) Bit-rates (kbps) (h) Bit-rates (kbps)

Figure 8.14 R–D curves of all test sequences compressed by HM-15.0 with our
and conventional RC [37] schemes [55]: (a) Fengjing 1, (b) Tiyu 1,
(c) Yanchanghui 2, (d) Dianying, (e) Hangpai 1, ( f ) Hangpai 2,
(g) AerialCity, (h) DrivingInCountry
Table 8.9 BD-rate saving and BD-PSNR enhancement for each test panoramic video sequence [55]

Name Fengjing 1 Tiyu 1 Yanchanghui 2 Dianying Hangpai 1 Hangpai 2 AerialCity DrivingInCountry Average

BD-rate saving (%) −7.63 −4.39 −3.96 −4.81 −3.87 −4.04 −5.41 −8.63 −5.34
BD-PSNR (dB) 0.2527 0.1155 0.1619 0.1441 0.1143 0.1197 0.1356 0.2464 0.1613
300 Applications of machine learning in wireless communications

(a) (b)

Figure 8.15 Visual quality of Dianying compressed at 158 kbps by HM-15.0 with
our and conventional RC [37] schemes. Note that this figure shows
the 68th frame of compressed Dianying [55]. (a) Conventional RC
scheme and (b) our scheme

is up to 1.37‰. Although the RC accuracy of our scheme is smaller than [37], it

is rather high and very close to 100% accuracy. Therefore, our scheme is effective
and practical for controlling bit rate of HEVC-based panoramic video coding. More
importantly, our RC is capable of improving RC performance for panoramic video
coding.

8.7 Conclusion
In this chapter, we have proposed a novel HEVC-based compression approach that
minimizes the perceptual distortion. Benefiting from the state-of-the-art saliency
detection, we developed a formulation to minimize perceptual distortion, which main-
tains properly high quality at regions that attract attention. Then, the RTE method was
proposed as a closed-form solution to our formulation with little extra time for mini-
mizing perceptual distortion, followed by the bit allocation and reallocation process.
Consequently, we validated our approach in experiments of compressing both images
and videos.
There are two possible directions for future work: (1) our approach only takes
into account the visual attention in improving the subjective quality of compressed
images/videos. In fact, other factors of the HVS, e.g., JND, may also be integrated
into our approach for perceptual compression. (2) Our approach in its present form
only concentrates on minimizing perceptual distortion according to the predicted
visual attention of uncompressed frames. However, the distribution of visual attention
may be influenced by the distortion of compressed frames in reverse. A long-term
Table 8.10 S-PSNR improvement and RC accuracy of our RC scheme, compared with the conventional scheme [37,55]

Name Fixed RC error (‰) RC S-PSNR (dB) Name Fixed RC error (‰) RC S-PSNR (dB) Name Fixed RC error (‰) RC S-PSNR (dB)
QP (conventional) error (‰) improvement error (‰) QP (conventional) error (‰) improvement QP (conventional) error (‰) improvement
(our) (our) (our)

Fengjing 1 27 0.07 0.12 0.27 Tiyu 1 27 0.04 0.04 0.18 Yanchanghui 2 27 0.74 0.34 0.03
32 0.06 0.05 0.28 32 0.46 0.08 0.13 32 0.34 0.52 0.17
37 0.32 0.20 0.23 37 0.01 1.98 0.07 37 0.25 0.96 0.19
42 0.31 0.23 0.15 42 1.37 3.02 0.10 42 0.54 1.89 0.20
Dianying 27 0.02 1.68 −0.07 Hangpai 1 27 0.05 0.19 0.19 Hangpai 2 27 0.02 0.18 0.15
32 0.00 2.00 0.11 32 0.10 0.29 0.11 32 0.45 0.27 0.14
37 0.26 0.49 0.25 37 0.06 1.46 0.09 37 0.42 0.50 0.09
42 0.70 0.04 0.32 42 0.12 0.10 0.12 42 0.01 0.78 0.09
AerialCity 27 0.17 0.95 0.06 Driving 27 0.02 2.75 0.34 Average 27 0.14 0.78 0.14
InCountry
32 0.28 0.74 0.12 32 0.04 0.24 0.27 32 0.21 0.52 0.17
37 0.06 1.18 0.20 37 0.02 0.52 0.21 37 0.17 0.91 0.17
42 0.09 4.43 0.18 42 0.05 1.24 0.19 42 0.40 1.47 0.17
Overall average 0.23 0.92 0.16
302 Applications of machine learning in wireless communications

goal of perceptual compression should thus include the loop between visual attention
and perceptual distortion over compressed images/videos.

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Chapter 9
Machine-learning-based saliency detection and
its video decoding application in wireless
multimedia communications
Mai Xu1 , Lai Jiang1 , and Zhiguo Ding2

Saliency detection has been widely studied to predict human fixations, with various
applications in wireless multimedia communications. For saliency detection, we argue
that the state-of-the-art high-efficiency video-coding (HEVC) standard can be used to
generate the useful features in compressed domain. Therefore, this chapter proposes
to learn the video-saliency model, with regard to HEVC features. First, we establish an
eye-tracking database for video-saliency detection. Through the statistical analysis on
our eye-tracking database, we find out that human fixations tend to fall into the regions
with large-valued HEVC features on splitting depth, bit allocation, and motion vector
(MV). In addition, three observations are obtained from the further analysis on our eye-
tracking database. Accordingly, several features in HEVC domain are proposed on the
basis of splitting depth, bit allocation, and MV. Next, a support vector machine (SVM)
is learned to integrate those HEVC features together, for video-saliency detection.
Since almost all video data are stored in the compressed form, our method is able to
avoid both the computational cost on decoding and the storage cost on raw data. More
importantly, experimental results show that the proposed method is superior to other
state-of-the-art saliency-detection methods, either in compressed or uncompressed
domain.

9.1 Introduction

According to the study on the human visual system (HVS) [1], when a person looks
at a scene, he/she may pay much visual attention on a small region (the fovea) around
a point of eye fixation at high resolution. The other regions, namely, the peripheral
regions, are captured with little attention at low resolutions. As such, humans are
able to avoid the processing of tremendous visual data. Visual attention is therefore

1
School of Electronic and Information Engineering, Beihang University, China
2
School of Electrical and Electronic Engineering, The University of Manchester, UK
308 Applications of machine learning in wireless communications

a key to perceive the world around humans, and it has been extensively studied in
psychophysics, neurophysiology, and even computer vision societies [2]. Saliency
detection is an effective way to predict the amount of human visual attention attracted
by different regions in images/videos. Most recently, saliency detection has been
widely applied in wireless multimedia communications and other computer vision
tasks, such as, object detection [3,4], object recognition [5], image retargeting [6],
image-quality assessment [7], and image/video compression [8,9].
In earlier time, some heuristic saliency-detection methods are developed accord-
ing to the understanding of the HVS. Specifically, in light of the HVS, Itti and
Koch [10] found out that the low-level features of intensity, color, and orientation
are efficient in detecting saliency of still images. In their method, center-surround
responses in those feature channels are established to yield the conspicuity maps.
Then, the final saliency map can be obtained by linearly integrating conspicuity maps
of all three features. For detecting saliency in videos, Itti et al. [11] proposed to
add two dynamic features (i.e., motion and flicker contrast) into Itti’s image saliency
model [10]. Later, other advanced heuristic methods [12–18] have been proposed for
modeling video saliency.
Recently, data-driven methods [19–24] have emerged to learn the visual atten-
tion models from the ground-truth eye-tracking data. Specifically, Judd et al. [19]
proposed to learn a linear classifier of SVM from training data for image saliency
detection, based on several low, middle, and high-level features. For video-saliency
detection, most recently, Rudoy et al. [23] have proposed a novel method to predict
saliency by learning the conditional saliency map from human fixations over a few
consecutive video frames. This way, the inter-frame correlation of visual attention is
taken into account, such that the accuracy of video-saliency detection can be signif-
icantly improved. Rather than free-view saliency detection, a probabilistic multitask
learning method was developed in [21] for the task-driven video-saliency detection,
in which the “stimulus-saliency” functions were learned from the eye-tracking data
as the top-down attention models.
HEVC [25] was formally approved as the state-of-the-art video-coding standard
in April 2013. It achieves double coding efficiency improvement over the preced-
ing H.264/AVC standard. Interestingly, we found out that the state-of-the-art HEVC
encoder can be explored as a feature extractor to efficiently predict video saliency. As
shown in Figure 9.1, the HEVC domain features on splitting depth, bit allocation, and
MV for each coding tree unit (CTU), are highly correlated with the human fixations.
The statistical analysis of Section 9.3.2 verifies such high correlation. Therefore, we
develop several features in our method for video-saliency detection, which are based
on splitting depths, bit allocation, and MVs in HEVC domain. It is worth pointing
out that most videos exist in the form of encoded bitstreams and the features related
to entropy and motion have been well exploited by video coding at the encoder side.
Since [2] has argued that entropy and motion are very effective in video-saliency
detection, our method utilizes these well-exploited HEVC features (splitting depth,
bit allocation, and MV) at the decoder side to achieve high accurate detection on
video saliency.
Machine-learning-based saliency detection 309

(a) CTU structure (b) Bit allocation

Heatmap

0 1
Counts

(c) MV (d) Heat map of human fixations

Figure 9.1 An example of HEVC domain features and heat map of human fixations
for one video frame. Parts (a), (b), and (c) are extracted from the
HEVC bitstream of video BQSquare (resolution: 416 × 240) at
130 kbps. Note that in (c) only the MVs that are larger than 1 pixel are
shown. Part (d) is the heat map convolved with a 2D Gaussian filter
over fixations of 32 subjects

Generally speaking, the main motivation of using HEVC features in our saliency
detection method is 2-fold: (1) our method takes advantage from sophisticated
encoding of HEVC, to effectively extract features for video-saliency detection. Our
experimental results in this chapter also show that the HEVC features are quite effec-
tive in video-saliency detection. (2) Our method can efficiently detect video saliency
from HEVC bitstreams without completely decoding the videos, thus saving both the
computational time and storage. Consequently, our method is generally more effi-
cient than the aforementioned video-saliency detection methods at pixel domain (or
called uncompressed domain), which have to decode the bitstreams into raw data.
Such efficiency is also validated by our experiments.
There are only a few methods [26–28] proposed for detecting video saliency
in compressed domain of previous video-coding standards. Among these methods,
the block-wise discrete cosine transform (DCT) coefficients and MVs are extracted
in MPEG-2 [26] and MPEG-4 [27]. Bit allocation of H.264/AVC is exploited for
saliency prediction in [28]. However, all above methods do not take full advantage of
the sophisticated features of the modern HEVC encoder, such as CTU splitting [29]
and R–λ bit allocation [30]. More importantly, all methods of [26–28] fail to find out
the precise impact of each compressed domain feature on attracting visual attention.
310 Applications of machine learning in wireless communications

In fact, the relationship between compressed domain features and visual attention can
be learned from the ground-truth eye-tracking data. Thereby, this chapter proposes
to learn the visual attention model of videos with regard to the well-explored HEVC
features.
Similar in spirit, the latest work of [31] also makes use of HEVC features for
saliency detection. Despite conceptually similar, our method is greatly different from
[31] in two aspects. From the aspect of feature extraction, our method develops pixel-
wise HEVC features, while [31] directly uses block-based HEVC features with deeper
decoding (e.g., inverse DCT). Instead of going deeper, our method develops shallow
decoded HEVC features with sophisticated design of temporal and spatial difference
on these features, more unrestrictive than [31]. In addition, camera motion is detected
and then removed in our HEVC features, such that our features are more effective
in predicting attention. From the aspect of feature integration, compared with [31],
our method is data driven, in which a learning algorithm is developed to bridge the
gap between HEVC features and video saliency. Meanwhile, our data-driven method
benefits from thorough analysis of our established eye-tracking database.
Specifically, the main contributions of this chapter are listed in the following:
● We establish an eye-tracking database on viewing 33 raw videos of the latest data
sets, with the thorough analysis and observations on our database.
● We propose several saliency-detection features in HEVC domain, according to
the analysis and observations on our established eye-tracking database.
● We develop a data-driven method for video-saliency detection, with respect to the
proposed HEVC features.
The rest of this chapter is organized as follows: in Section 9.2, we briefly review
the related work on video-saliency detection. In Section 9.3, we present our eye-
tracking database as well as the analysis and observations on our database. In light
of such analysis and observations, Section 9.4 proposes several HEVC features for
video-saliency detection. Section 9.5 outlines our learning-based method, which is
based on the proposed HEVC features. Section 9.6 shows the experimental results to
validate our method. Finally, Section 9.7 concludes this chapter.

9.2 Related work on video-saliency detection

9.2.1 Heuristic video-saliency detection

For modeling saliency of a video, a great number of methods [11–18] have been
proposed. Itti et al. [11] started the initial work of video-saliency detection, by
adding two dynamic features of motion and flicker contrast into Itti’s image saliency
model [10]. Later, a novel term called surprise was defined in [14] to measure
how the visual change attracts human observers. With the new term surprise, [14]
developed a Bayesian framework to calculate the Kullback–Leibler (KL) divergence
between spatiotemporal posterior and prior beliefs, for predicting video saliency.
Some other Bayesian-framework-related methods, e.g., [15], were also proposed
Machine-learning-based saliency detection 311

for video-saliency detection. Most recently, some advanced video-saliency detection

methods [16–18] have been proposed. To be more specific, Guo et al. [16] applied
phase spectrum of quaternion Fourier transform (PQFT) on four feature channels
(two color channels, one intensity channel, and one motion channel) to detect video
saliency. Lin et al. [18] utilized earth mover’s distance to measure the center-surround
difference in spatiotemporal receptive filed, for producing the dynamic saliency maps
of videos. Inspired by sparse representation, Ren et al. [17] proposed to explore the
movement of a target patch for temporal saliency detection of videos. In their method,
the movement of the target patch can be estimated by finding the minimal reconstruc-
tion error of sparse representation regarding the patches of neighboring frames. In
addition to temporal saliency detection, the center-surround contrast needs be mod-
eled for spatial saliency detection. This is achieved through sparse representation with
respect to neighboring patches.
In fact, top-down visual cues play an important role in determining the saliency
of a scene. Thereby, the top-down visual attention models have been studied in [32,33]
for predicting the saliency of dynamic scenes in a video. In [32], Pang et al. proposed
to integrate the top-down information of eye-movement patterns (i.e., passive and
active states [13]) for video saliency detection. In [33], Wu and Xu found out that the
high-level features, such as face, person, car, speaker, and flash, may attract extensive
human attention. Thus, these high-level features are integrated with the bottom-up
model [16] for saliency detection of news videos.
However, the understanding of the HVS is still in its infancy, and saliency detec-
tion thus has a long way to go yet. In fact, we may rethink saliency detection by
taking advantage of the existing video-coding techniques. Specifically, the video-
coding standards have evolved for almost three decades, with HEVC being the latest
one. The evolution of video coding adopts several elegant and effective techniques to
produce several sophisticated features, for continuously improving coding efficiency.
For example, the state-of-the-art HEVC standard introduced fractional sample inter-
polation to represent MVs with quarter-sample precision, thus being able to precisely
model object motions. Moreover, HEVC proposes to partition CTUs into smaller
blocks using the tree structure and quadtree-like signaling [29], which can well reflect
the texture complexity of video frames. On the other hand, the HEVC features, which
are generated by the sophisticated process of the latest HEVC techniques, may be
explored for efficient video-saliency detection.

9.2.2 Data-driven video-saliency detection

During the past decade, data-driven methods have emerged as a possible way to learn
video-saliency model from ground-truth eye-tracking data, instead of the study on the
HVS. The existing data-driven video-saliency detection can be further divided into
task-driven [13,21,22,34,35] and free-view [20,23,24,36] methods.
For task-driven video-saliency detection, Peter and Itti [13] proposed to incorpo-
rate the computation on signatures of each video frame. Then, a regression classifier
is learned from the subjects’ fixations on playing video games, which associates the
different classes of signatures (seen as gist) with the gaze patterns of task-driven
312 Applications of machine learning in wireless communications

attention. Combined with 12 multi-scale bottom-up features, [13] has high accu-
racy in task-driven saliency detection. Most recently, a dynamic Bayesian network
method [35] has been proposed for learning top-down visual attention model of play-
ing video games. Besides the task of playing video games, a data-driven method [34]
on video-saliency detection was proposed with the dynamic consistency and align-
ment models, for the task of action recognition. In [34], the proposed models are
learned from the task-driven human fixations on large-scale dynamic computer vision
databases like Hollywood-2 [37] and UCF Sports [38]. In [21], Li et al. developed
a probabilistic multitask learning method to include the task-related attention mod-
els for video-saliency detection. The “stimulus-saliency” functions are learned from
the eye-tracking database, as the top-down attention models to some typical tasks
of visual search. As a result, [21] is “good at” video-saliency detection in multiple
tasks, more generic than other methods that focus on single visual task. However, all
task-driven saliency-detection methods can only deal with the specific tasks.
For free-view video-saliency detection, Kienzle et al. [20] proposed a nonpara-
metric bottom-up method to model video saliency, via learning the center-surround
texture patches and temporal filters from the eye-tracking data. Recently, Lee et al.
[24] have proposed to extract the spatiotemporal features, i.e., rarity, compactness,
center prior, and motion, for the bottom-up video-saliency detection. In their bottom-
up method, all extracted features are combined together by an SVM, which is learned
from the training eye-tracking data. In addition to the bottom-up model, Hua et al. [36]
proposed to learn the middle-level features, i.e., gists of a scene, as the top-down cue
for both video and image-saliency detection. Most recently, Rudoy et al. [23] have
proposed to detect the saliency of a video, by simulating the way that humans watch
the video. Specifically, a visual attention model is learned to predict the saliency map
of a video frame, given the fixation maps from the previous frames. As such, the inter-
frame dynamics of gaze transitions can be taken into account during video-saliency
detection.
As aforementioned, this chapter mainly concentrates on utilizing the HEVC fea-
tures for video-saliency detection. However, there is a gap between HEVC features
and human visual attention. From data-driven perspective, machine learning can be
utilized in our method to investigate the relationship between HEVC features and
visual attention, according to eye-tracking data. Thus, this chapter aims at learning an
SVM classifier to predict saliency of videos using the features from HEVC domain.

9.3 Database and analysis

9.3.1 Database of eye tracking on raw videos

In this chapter, we conducted the eye-tracking experiment to obtain fixations on
viewing videos of the latest test sets. Here, all 33 raw videos from the test sets [9,39],
which have been commonly utilized for evaluating HEVC performance, were included
in our eye-tracking experiment. We further conducted the extra experiment to obtain
the eye-tracking data on watching all videos of our database compressed by HEVC at
Machine-learning-based saliency detection 313

different quality. Through the data analysis, we found that visual attention is almost
unchanged when videos are compressed at high or medium quality (more than 30 dB).
This is consistent with the result of [40]. Compared with the conventional databases
(e.g., SFU [41] and DIEM [42]), the utilization of these videos benefits from the
state-of-the-art test sets in providing videos with diverse resolutions and content. For
the resolution, the videos vary from 1080p (1,920 × 1,080) to 240p (416 × 240). For
the content, the videos include sport events, surveillance, video conferencing, video
games, videos with the subscript, etc.
In our eye-tracking experiment, all videos are withYUV 4:2:0 sampling. Here, the
resolutions of the videos in Class A of [39] were down-sampled to be 1,280 × 800,
as the screen resolution of the eye tracker can only reach to 1,920 × 1,080. Other
videos were displayed in their original resolutions. In our experiment, the videos were
displayed in a random manner at their default frame rates to reduce the influence of
video-playing order on the eye-tracking results. Besides, a blank period of 5 seconds
was inserted between two consecutive videos, so that the subjects can have a proper
rest time to avoid eye fatigue.
There were a total of 32 subjects (18 male and 14 female, aging from 19 to
60) involved in our eye-tracking experiment. These subjects were selected from the
campuses of Beihang University and Microsoft ResearchAsia. All subjects have either
corrected or uncorrected normal eyesight. Note that only two subjects were experts,
who are working in the research field of saliency detection. The other 30 subjects
did not have any research background in video-saliency detection, and they were also
native to the purpose of our eye-tracking experiment.
The eye fixations of all 32 subjects over each video frame were recorded by
a Tobii TX300 eye tracker at a sample rate of 300 Hz. The eye tracker is inte-
grated with a monitor of 23-inch LCD screen, and the resolution of the monitor
was set to be 1,920 × 1,080. All subjects were seated on an adjustable chair at a
distance of around 60 cm from the screen of the eye tracker, ensuring that their
horizontal sight is in the center of the screen. Before the experiment, subjects were
instructed to perform the 9-point calibration for the eye tracker. Then, all subjects
were asked to free-view each video. After the experiment, 392,163 fixations over
13,020 frames of 33 videos were collected. Here, the eye fixations of all subjects
and the corresponding MATLAB® code for our eye-tracking database are available
online: https://github.com/remega/video_database.

9.3.2 Analysis on our eye-tracking database

Figure 9.1 has shown that the HEVC features, i.e., splitting depth, bit allocation, and
MV, are effective in predicting human visual attention. It is therefore interesting to
statistically analyze the correlation between these HEVC features and visual attention.
From now on, we concentrate on the statistical analysis on our eye-tracking database
to show the effectiveness of the HEVC features on the prediction of visual attention.
This is a new finding, which reveals the correlation between HEVC features and
visual attention.
314 Applications of machine learning in wireless communications

For all videos of our database, the features on splitting depth, bit allocation, and
MV were extracted from the corresponding HEVC bitstreams. Then, the maps of
these features were generated for each video frame. Note that the configuration to
generate the HEVC bitstreams can be found in Section 9.6. Afterwards, a 2D Gaussian
filter was applied to all three feature maps of each video frame. For each feature map,
after sorting pixels in the descending order of their feature values, the pixels were
equally divided into ten groups according to the values of corresponding features.
For example, the group of 0%–10% stands for the set of pixels, the features of which
rank top 10%. Finally, the number of fixations belonging to each group was counted
upon all 33 videos in our database.
We show in Figure 9.2 the percentages of eye fixations belonging to each group, in
which the values of the corresponding HEVC features decrease alongside the groups.
From this figure, we can find out that extensive attention is drawn by the regions
with large-valued HEVC features, especially for the feature of bit allocation. For
example, about 33% fixations fall into the regions of top 10% high-valued feature of
bit allocation, whereas the percentage of those hitting the bottom 10% is much less
than 2%. Hence, the HEVC features on splitting depth, bit allocation, and MV, are
explored for video-saliency detection in our method (Section 9.4).

35%

30% Splitting depth

25% Bit allocation

MV
20%
Proportion

15%

10%

0%−10% 10%−20% 20%−30% 30%−40% 40%−50% 50%−60% 60%−70% 70%−80% 80%−90% 90%−100%
Groups

Figure 9.2 The statistical results for fixations belong to different groups of pixels,
in which values of the corresponding HEVC features are sorted in the
descending order. Here, all 392,163 fixations of 33 videos are used for
the analysis. In this figure, the horizontal axis indicates the groups of
pixels. For example, 0%–10% means that the first group of pixels, the
features of which rank top 10%. The vertical axis shows the percentage
of fixations that fall into each group
Machine-learning-based saliency detection 315

9.3.3 Observations from our eye-tracking database

Beyond the analysis of our eye-tracking database, we verify some other factors on
attracting human attention, with the following three observations. These observations
provide insightful guide for developing our saliency-detection method.
Observation 9.1: Human fixations lag behind the moving or new objects in a video
by some microseconds.
In Figure 9.3, we show the frames of videos BasketballDrive and Kimono with the
corresponding heat maps of human fixations. The first row of this figure reveals that
the visual attention falls behind the moving object, as the fixations trail the moving
basketball. In particular, the distance between the basketball and fixations becomes
large, when the basketball moves at high speed. Besides, the second row of Figure
9.3 illustrates that the human fixations lag behind the new appearing objects by a
few frames. It is because the human fixations still stay in the location of the salient
region in previous frames, even when the scene has been changed. This completes
the analysis of Observation 9.1.
Observation 9.2: Human fixations tend to be attracted by the new objects appearing
in a video.
It is intuitive that visual attention is probably to be attracted by the objects newly
emerging in a video. It is thus worth analyzing the influence of the object emergence
on human visual attention. Figure 9.4 shows the heat maps of fixations on several
frames selected from videos vidyo1 and ParkScene. Note that a person appears in the
door from the 553th frame of the video vidyo1, and that a person riding bicycle arises
from the 64th frame of the video ParkScene. From Figure 9.4, one may observe that
once a new object appears in the video, it probably attracts a huge amount of visual

120 frame 125 frame 135 frame 145 frame

138 frame 144 frame 147 frame 187 frame

Time

Figure 9.3 Illustration of Observation 9.1. This figure shows the heat maps of
human fixations of all 32 subjects on several selected frames of videos
BasketballDrive and Kimono. In BasketballDrive, the green box is
drawn to locate the moving basketball
316 Applications of machine learning in wireless communications

431 frame 553 frame 581 frame 599 frame

32 frame 64 frame 78 frame 121 frame

Time

Figure 9.4 Illustration of Observation 9.2. This figure shows the heat maps of
visual attention of all 32 subjects, over several selected frames of
videos vidyo1 and ParkScene

Heatmap

0 1
Counts

Figure 9.5 Illumination of Observation 9.3. This figure shows the map of human
fixations of all 32 subjects, over a selected frame of video
PeopleOnStreet. Note that in the video a lot of visual attention is
attended to the old man, who pushes a trolley and walks in the opposite
direction of the crowd

attention. This completes the analysis of Observation 9.2. Note that there exists the
lag of human fixations, as the door is still fixated on when the person has left. This
also satisfies Observation 9.1.

Observation 9.3: The object, which moves in the opposite direction of the
surrounding objects, is possible to receive extensive fixations.

The previous work [10] has verified that the human fixations on still images are
influenced by the center-surround features of color and intensity. Actually, the center-
surround feature of motions also has an important effect on attracting visual attention.
As seen from Figure 9.5, the old man with a trolley moves in the opposite direction
of the surrounding crowd, and he attracts the majority of visual attention. Therefore,
Machine-learning-based saliency detection 317

this suggests that the object moving in the opposite direction to its surround (i.e., it is
with large center-surround motion) may receive extensive fixations. This completes
the analysis of Observation 9.3.

9.4 HEVC features for saliency detection

In this section, we mainly focus on exploring the features in HEVC domain, which can
be used to efficiently detect video saliency. As analyzed above, three HEVC features,
i.e., splitting depth, bit allocation, and MV, are effective in predicting video saliency.
Therefore, they are worked out as the basic features for video saliency detection, to be
presented in Section 9.4.1. Note that the camera motion has to be removed for the MV
feature, with an efficient algorithm developed in Section 9.4.1. Based on the three
basic HEVC features, the features on temporal and spatial difference are discussed
in Sections 9.4.2 and 9.4.3, respectively.

9.4.1 Basic HEVC features

Splitting depth. The CTU partition structure [29], a new technique introduced by
HEVC, can offer more flexible block sizes in video coding. In HEVC, the block
sizes range from 64 × 64 to 8 × 8. In other words, the splitting depth varies from 0
(=64 × 64 block size) to 3 (=8 × 8 block size). In HEVC, rather than raw pixels, the
residual of each coding block is encoded, which reflects spatial texture in intra-frame
prediction and temporal variation in inter-frame prediction. Consequently, in intra-
frame prediction, splitting depth of each CTU can be considered to model spatial
saliency. In inter-frame prediction, splitting depth of each coding block can be used
to model temporal saliency. Since Section 9.3.2 has demonstrated that most fixations
fall into groups with high-valued splitting depths, the splitting depth of each CU is
applied as a basic HEVC feature in video-saliency detection.
Let dijk be the normalized splitting depth of pixel (i, j) at the kth frame. First, the
splitting depths of all CUs need to be extracted from HEVC bitstreams. Then, we
assume that the splitting depth of each pixel is equivalent to that of its corresponding
CU. Afterwards, all splitting depths should be normalized by the maximal splitting
depth in each video frame. At last, all normalized dijk can be yielded as one basic
feature of our method.
Bit allocation. Since the work of [30] is a state-of-the-art rate control scheme
for HEVC, it has been embedded into the latest HEVC reference software (HM 16.0)
for assigning bits to different CTUs. In the work of [30], the rate-distortion was opti-
mized in each video frame, such that the CTUs with high-information were generally
encoded by more bits. It has been argued in [2] that high-information regions attract
extensive visual attention. Thus, the bits, allocated by [30] in HEVC, are consid-
ered a basic feature, modeling spatial saliency in intra-frame prediction and temporal
saliency in inter frame prediction. Specifically, Section 9.3.2 has shown that visual
attention is highly correlated with the bit allocation of each CTU. Thereby, bit per
pixel (bpp) is extracted from HEVC bitstreams, toward saliency detection. Let bkij
denote the normalized bpp of pixel (i, j) at the kth frame. Here, the bpp is achieved
318 Applications of machine learning in wireless communications

via averaging all consumed bits in the corresponding CTU. Next, the bpp is normal-
ized to be bkij in each video frame, and it is then included as one of basic HEVC
features to detect saliency.
MV. In video coding, MV identifies the location of matching prediction unit
(PU) in the reference frame. In HEVC, MV is sophisticatedly developed to indicate
motion between neighboring frames. Intuitively, MV can be used to detect video
saliency, as motion is an obvious cue [16] of salient regions. This intuition has also
been verified by the statistical analysis of Section 9.3.2. Therefore, MV is extracted
as a basic HEVC feature in our method.
During video coding, MV is accumulated by two factors: the camera motion
and object motion. It has been pointed out in [43] that in a video, moving objects
may receive extensive visual attention, while static background normally draws little
attention. It is thus necessary to distinguish moving objects and static background.
Unfortunately, MVs of static background may be as large as moving objects, due to
the camera motion. On the other hand, although temporal difference of MVs is able
to make camera motion negligible for static background, it may also miss the moving
objects. Therefore, the camera motion has to be removed from calculated MVs to
estimate object motion for saliency detection.
Figure 9.6 shows that the camera motion can be estimated to be the dominant
MVs in a video frame. In this chapter, we therefore develop a voting algorithm to
estimate the motion of camera. Assuming that mijk is the two-dimensional MV of pixel
(i, j) at the kth frame, the dominant camera motion mck in this frame can be determined
in the following way.
First, the static background Skb is roughly extracted to be
⎧ ⎫
⎨ 1 ⎬
Skb = (i, j)|dijk · bkij < k dik j · bki j , (9.1)
⎩ |I | k ⎭
(i , j )∈I

(a) Without camera motion (b) With camera motion

Figure 9.6 An example of MV values of all PUs in (a) a frame with no camera
motion and (b) a frame with right-to-left camera motion. Note that the
MVs are extracted from HEVC bitstreams. In (a) and (b), the dots stand
for the origin of each MV, and the blue lines indicate the intensity and
angle of each MV. It can be seen that in (a) there is no camera motion,
as most MV values are close to zero, whereas the camera motion in (b)
is from right to left according the most MV values
Machine-learning-based saliency detection 319

for the kth frame Ik (with |Ik | pixels). It is because the static background is generally
with less splitting depth and bit allocation than the moving foreground objects. Then,
the azimuth a(mck ) for the dominant camera motion can be calculated via voting all
MV angles in the background Skb as
⎛ ⎞

max hist ⎝ a(mijk )⎠ , (9.2)
i, j∈Skb

where a(mijk ) is the azimuth for MV mijk , and hist( · ) is the azimuth histogram of all
MVs. In this chapter, 16 bins with equal angle width (= 360◦ /16 = 22.5◦ ) are applied
for the histogram. After obtaining a(mck ), radius r(mck ) for the camera motion needs to
be calculated via averaging over all MVs from the selected bin of a(mck ). Finally, the
camera motion of each frame can be achieved upon a(mck ) and r(mck ). For justification,
we show in Figure 9.7 some subjective results of the camera motion estimated by our
voting algorithm (in yellow arrows), as well as the annotated ground truth of camera
motion (in blue arrows). As can be seen from this figure, our algorithm is capable of
accurately estimating the camera motion. See Appendix for more justification on the
estimation of camera motion.
Next, in order to track the motion of objects, all MVs obtained in HEVC domain
need to be processed to remove the estimated camera motion. All processed MVs
should be then normalized in each video frame, denoted as m̂ijk . Since it has been
argued in [16] that visual attention is probably attracted by moving objects, m̂ijk 2 is
utilized as one of the basic HEVC features to predict video saliency.

5 frame 300 frame 50 frame 300 frame 240 frame 144 frame 120 frame

15 frame 350 frame 100 frame 350 frame 300 frame 156 frame 180 frame

25 frame 400 frame 150 frame 400 frame 360 frame 168 frame 240 frame

Figure 9.7 The results of camera motion estimation, yielded by our voting
algorithm. The first six videos are with some extended camera motion,
whereas the last one is without any camera motion. In the frames of the
second row, the yellow and blue arrows represent the estimated and
manually annotated vectors of the camera moving from frames of the
first row to frames of the second row, respectively. Similarly, the yellow
and blue arrows in the frames of the third row show the camera motion
from frames of the second row to the third row. Refer to Appendix for
the way of annotating ground-truth camera motion
320 Applications of machine learning in wireless communications

9.4.2 Temporal difference features in HEVC domain

As revealed in Observation 9.2, humans tend to fixate on the new objects appearing
in a video. In fact, the new appearing or moving objects in the video also leads
to large temporal difference of HEVC features in colocated regions of neighboring
frames. Hence, the temporal difference features, which quantify the dissimilarity of
splitting depth, bit allocation and MV across neighboring frames, are developed as
novel HEVC features in our method. However, the temporal difference in colocated
region across video frames refers to the sum of object motion and camera motion.
It has been figured out in [43] that moving objects attract extensive visual attention,
whereas camera motion receives little attention. Therefore, when developing temporal
difference features, camera motion needs to be removed to compensate object motion
(to be discussed in the following).
Specifically, let us first look at the way on estimating temporal difference of
splitting depths. For pixel (i, j) at the kth frame, t dijk is defined as the difference
value of splitting depth across neighboring frames. It can be calculated by averaging
the weighted difference values of the splitting depths over all previous frames:
k
l=1 exp(−(l /σd ))dij − dij 1
2 2 k k−l
t dij =
k
k 2
, (9.3)
2
l=1 exp(−(l /σd ))

where parameter σd controls the weights on splitting depth difference between two
frames. In (9.3), dijk−l is the splitting depth of pixel (i, j) at the (k − l)th frame. After
considering the camera motion with our voting algorithm, we assume that (ik,l , j k,l ) is
the pixel at the (k − l)th frame matching to pixel (i, j) at the kth frame. To remove
the influence of the camera motion, we replace dijk−l in (9.3) by dik−l k,l j k,l . Then, (9.3) is

rewritten to be
k
l=1 exp(−(l /σd ))dij − dik,l j k,l 1
2 2 k k−l

t dij =
k
k 2
. (9.4)
2
l=1 exp(−(l /σd ))

After calculating (9.4), t dijk needs to be normalized in each video frame, as one of
temporal difference features in HEVC domain.
Furthermore, the bpp difference across neighboring frames is also regarded as a
feature for saliency detection. Let t bkij denote the temporal difference of the bpp at
pixel (i, j) between the currently processed kth frame and its previous frames. Similar
to (9.4), t bkij can be obtained by
k
l=1 exp(−(l /σb ))bij − bik,l j k,l 1
2 2 k k−l

t bij =
k
k 2
, (9.5)
2
l=1 exp(−(l /σb ))

where σb decides the weights of the bpp difference between frames. In (9.5), with
the compensated camera motion, bk−l ik,l j k,l
is the bpp for pixel (ik,l , j k,l ) at the (k − l)th
frame, which matches to pixel (i, j) at the kth frame.
Finally, the temporal difference of MV is also taken into account, by adopting
the similar way presented above. Recall that m̂ijk is the extracted MV of each pixel,
Machine-learning-based saliency detection 321

with the camera motion being removed. Since m̂ijk is a 2D vector, 2 -norm operation
is applied to compute the temporal difference of MVs (denoted by t m̂kij ) as follows:
k
l=1 exp(−(l /σm ))m̂ij − m̂ik,l j k,l 2
2 2 k k−l

t m̂ij =
k
k . (9.6)
2 2
l=1 exp(−(l /σm ))

In (9.6), we can use parameter σm to determine the weights of MV difference between

k,l j k,l is the MV value for pixel (i , j ) at the (k − l)th
two frames. Moreover, m̂ik−l k,l k,l

frame, which is the colocated pixel of (i, j) at the kth frame, after the camera motion
is removed by our voting algorithm.

9.4.3 Spatial difference features in HEVC domain

However, the above features are not sufficient to model saliency in a video, since some
smooth regions may stand out from complicated background for drawing attention
(like a salient smooth ball appearing in grass land). Generally speaking, the basic
features of splitting depth and bit allocation in a smooth region are significantly
different from those in its surrounding background. Thus, we here develop spatial
difference features for saliency detection. In addition, according to Observation 9.3,
the object moving in the opposite direction to the nearby objects may result in extensive
visual attention. Actually, the dissimilarity of object motion can be measured by the
spatial difference of MVs between neighboring PUs. Hence, the spatial difference of
all three basic features is incorporated into our method, as given below.
Recall that Ik is the kth video frame, and that dijk , bkij , and mijk denote the splitting
depth, bit allocation, and MV , respectively, for pixel (i, j) of this video frame. For the
spatial difference of MV, the camera motion has to be removed in each mijk , defined
by m̂ijk . Then, we have
⎧
⎪ (i , j )∈Ik exp(−(((i − i) + ( j − j) )/ξd ))di j − dij 1
2 2 2 k k
⎪
⎪ k
=
⎪
⎪ s d
(i , j )∈Ik exp(−(((i − i) + ( j − j) )/ξd ))
⎪ ij 2 2 2
⎪
⎪
⎪
⎪
⎨
(i , j )∈Ik exp(−(((i − i) + ( j − j) )/ξb ))bi j − bij 1
2 2 2 k k
s bkij = (9.7)
⎪
⎪ (i , j )∈Ik exp(−(((i − i) + ( j − j) )/ξb ))
2 2 2
⎪
⎪
⎪
⎪
⎪
⎪
2
k exp(−(((i − i) + ( j − j) )/ξm ))m̂i j − m̂ij 2
2 2 k k
⎪
⎪ s m̂ij = (i , j )∈I
k
⎩ ,
(i , j )∈Ik exp(−(((i − i) + ( j − j) )/ξm ))
2 2 2

to compute the spatial difference of splitting depth, bit allocation, and MV. As in the
above equations, ξd , ξb , and ξm are the parameters to control the spatial weighting of
each feature.
Finally, all nine features in HEVC domain can be achieved in our saliency
detection method. Since all the proposed HEVC features are block wise, the
block-to-pixel refinement is required to obtain smooth feature maps. For the block-
to-pixel refinement, a 2D Gaussian filter is applied to three basic features. In this
chapter, the dimension and standard deviation of the Gaussian filter are tuned to be
322 Applications of machine learning in wireless communications

(2h/15) × (2h/15) and (h/30), where h is the height of the video. It is worth men-
tioning that the above features on spatial and temporal difference are explored in
compressed domain with the block-to-pixel refinement, while the existing methods
compute contrast features in pixel domain (e.g., in [10,11]). Additionally, unlike the
existing methods, the camera motion is estimated and removed when calculating
the feature contrast in our method. Despite being simple and straightforward, these
features are effective and efficient, as evaluated in experiment section.
Figure 9.8 summarizes the procedure of HEVC feature extraction in our saliency
detection method. As seen from Figure 9.8, the maps of nine features have been
obtained, based on splitting depth, bit allocation, and MV of HEVC bitstreams. We
argue that one single feature is not capable enough [2] but has different impact on
saliency detection. We thus integrate the maps of all nine features with the learned
weights. For more details, refer to the next section.

9.5 Machine-learning-based video-saliency detection

This section mainly concentrates on learning an SVM classifier to detect video
saliency, using the abovementioned nine HEVC features. The framework of our
learning-based method is summarized in Figure 9.9. As shown in this figure, given the
HEVC bitstreams, all HEVC features need to be extracted and calculated. Then, the
saliency map of each single video frame is yielded by combining the HEVC features
with C-support vector classification (C-SVC), which is a kind of nonlinear SVM clas-
sifier. Here, the C-SVC classifier is learned from the ground-truth human fixations
of training videos. At last, a simple forward smoothing filter is applied to the yielded
saliency maps across video frames, outputting the final video-saliency maps. More
details about our learning-based method are to be discussed in the following.

9.5.1 Training algorithm

In our method, the nonlinear C-SVC [44], a kind of SVM, is trained as the binary
classifier to decide if each pixel can attract attention, according to the proposed
HEVC features. First, for the binary classifier, both positive and negative samples
need to be obtained from the training set, in which the positive samples mean the
pixels attracting fixations, and negative samples indicate the pixels without any visual
attention. Next, three basic HEVC features of each training sample are extracted from
the HEVC bitstreams, and then other spatial and temporal features are computed upon
the corresponding basic features. Let {(fn , ln )}Nn=1 be those training samples, where fn
is the vector of the nine HEVC features for the nth training sample, and ln ∈ {−1, 1} is
the class label indicating whether the sample is positive (ln = 1) or negative (ln = −1).
Finally, the C-SVC for saliency detection can be worked out, via solving the following
optimization problem:
1 N
min w22 + C βn
w,b,{βn }N
n=1
2 n=1
(9.8)
s.t. ∀n, ln (wT · φ(fn ) + b) ≥ 1 − βn , βn ≥ 0.
Temporal difference features

Temporal difference Temporal difference Temporal difference

Calculating of splitting depths of bits allocation of motion vectors
HEVC features extraction the temporal
and normalization Three basic features difference

Splitting Gaussian
depths
HEVC bitstreams filter
Basic features
of videos
Splitting depths Bits allocation Motion vectors
Motion Camera Gaussian
vectors motion filter
removal

Bits Gaussian
allocation filter
Spatial difference features
Spatial difference Spatial difference Spatial difference
Calculating of splitting depths of bits allocation of motion vectors
the spatial
difference

Figure 9.8 Framework of our HEVC feature extractor for video-saliency detection
324 Applications of machine learning in wireless communications
HEVC bitstreams
of training videos Human fixations
HEVC feature Linear SVM
Training extractor learning model

HEVC bitstreams
of a test videos Video saliency map
HEVC feature Linear Forward
Test extractor combination smoothing

Figure 9.9 Framework of our learning-based method for video-saliency detection

with HEVC features. For the HEVC feature extractor, refer to Figure 9.8

In (9.8), w and b are the parameters to be learned for maximizing the margin between
positive and negative samples, and βn is a nonnegative slack variable evaluating the
degree of classification error of fn . In addition, C balances the trade-off between
the error and margin. Function φ(·) transforms the training vector of HEVC features
fn to higher dimensional space. Then, w can be seen as the linear combination of
transformed vectors:

N
w= λm lm · φ(fm ), (9.9)
m=1

where λm is the Lagrange multiplier to be learned. Then, the following holds:

N T

w · φ(fn ) =
T
λm lm · φ(fm ) · φ(fn )
m=1

N
= λm lm · φ(fm ), φ(fn ) . (9.10)
m=1

Note that φ(fm ), φ(fn ) indicates the inner product of φ(fm ) and φ(fn ). To calculate
(9.10), a kernel of radial based function (RBF) is introduced:
K(fm , fn ) = φ(fm ), φ(fn ) = exp(−γ fm − fn 22 ), (9.11)
where γ (> 0) stands for the kernel parameter. Here, we utilize the above RBF
kernel due to its simplicity and effectiveness. When training the C-SVC for saliency
detection, the penalty parameter C in (9.8) is set to 2−3 , and γ of the RBF kernel is
tuned to be 2−15 , such that the trained C-SVC is rather efficient in detecting saliency.
Finally, w and b can be worked out in the trained C-SVC as the model of video-saliency
detection, to be discussed below.

9.5.2 Saliency detection

To detect the saliency of test videos, all nine HEVC features are integrated together
using the learned w and b of our C-SVC classifier. Then, the saliency map Sk for each
single video frame can be yielded by
Sk = wT · φ(Fk ) + b, (9.12)
Machine-learning-based saliency detection 325

where Fk defines the pixel-wise matrix of nine HEVC features at the kth video frame.
Note that w in (9.12) is one set of weights for the binary classifier of C-SVC, which
have been obtained using the above training algorithm.
Since Observation 9.1 offers a key insight that visual attention may lag behind
the moving or new appearing objects, a forward smoothing filter is developed in our
method to take into account the saliency maps of previous frames. Mathematically, the
final saliency map Ŝk of the kth video frame is calculated by the forward smoothing
filter as follows:
1
k
Ŝk = Sk , (9.13)

t · fr
k =k−
t·fr+1

where t (> 0) is the time duration1 of the forward smoothing, and fr is the frame
rate of the video. Note that a simple forward smoothing filter of (9.13) is utilized
here, since we mainly concentrate on extracting and integrating features for saliency
detection. Some advanced tracking filters may be applied, instead of the forward
smoothing filter in our method, for further improving the performance on saliency
detection. To model visual attention on video frames, the final saliency maps need to
be smoothed with a 2D Gaussian filter, which is in addition to the one for each single
feature map (as shown in Figure 9.8). Note that the 2D Gaussian filter here shares the
same parameters as those for feature maps.

9.6 Experimental results

In this section, we present the experimental results on video-saliency detection to
validate the performance of our method. Section 9.6.1 shows the settings of our
method, and Section 9.6.2 discusses the parameter selection in our method. Sections
9.6.3 and 9.6.4 compare the saliency detection results by our and other seven methods,
over our and other two public databases, respectively. For comparing the accuracy of
saliency detection, receiver operating characteristic (ROC) curves, the equal error
rate (EER), the area under ROC curve (AUC), normalized scanpath saliency (NSS),
linear correlation coefficient (CC), and KL were measured on the saliency maps
generated by our and other seven methods. Section 9.6.5 evaluates the performance
of our method at different working conditions. In Section 9.6.6, we demonstrate the
effectiveness of each single HEVC feature in saliency detection.

9.6.1 Setting on encoding and training

HEVC configuration. Before saliency detection, the bitstreams of both train-
ing and test videos were generated by the HEVC encoder, for extracting fea-
tures. In our experiments, the HEVC reference software HM 16.0 (JCT-VC;
http://hevc.hhi.fraunhofer.de/) was used as the HEVC encoder. Then, the HEVC

1
We found out through experiments that t = 0.3 s makes the saliency detection accuracy highest. So, time
duration t of our forward smoothing was set to be 0.3 in Section 9.6.
326 Applications of machine learning in wireless communications

bitstreams of all 33 videos in our database were produced for both training and test.
In HM 16.0, the low delay (LD) P main configuration was chosen. In addition, the
latest R–λ rate control scheme [30] was enabled in HM 16.0. Since the test videos are
with diverse content and resolutions, we followed the way of [30] to set the bit rates
the same as those at fixed QPs. The CTU size was set to 64 × 64 and maximum CTU
depth was 3 to allow all possible CTU partition structures for saliency detection. Each
group of pictures (GOP) was composed of 4 P frames. Other encoding parameters
were set by default, using the common encoder_lowdelay_P_main.cfg configuration
file of HM.
Other working conditions. The implementation of our method in random access
(RA) configuration is to be presented in Section 9.6.5. The rate control of RA in HM
16.0 is also enabled. In our experiments, we set all other parameters of RA via the
encoder_randomaccess_main.cfg file. Note that the GOP of RA is 8 B frames for
HM 16.0. Section 9.6.5 further presents the saliency detection results of our method
for the bitstreams of ×265, which is more practical than the HM encoder from the
aspects of encoding and decoding time.2 Here, ×265 v1.8 encoder, embedded in the
latest FFmpeg, was applied. For ×265, both LD and RA were tested. In ×265, the bit
rates were chosen using the same way as we applied for HM 16.0. The GOP structure
is 4 P frames for LD and four frames (BBBP) for RA. Other parameters were all set
by default in the FFmpeg with the ×265 codec. It is worth pointing out that the ×265
codec was used to extract features from the bitstreams encoded by ×265, while the
features of HM 16.0 bitstreams were extracted by the software of HM 16.0.
Training setting. In order to train the C-SVC, our database of Section 9.3.1 was
divided into nonoverlapping sets. For the fair evaluation, 3-fold cross validation was
conducted in our experiments, and the averaged results are reported in Sections 9.6.2
and 9.6.3. Specifically, our database was equally partitioned into three nonoverlapping
sets. Then, two sets were used as training data, and the remaining set was retained for
validating saliency detection. The cross-validation process is repeated by 3-fold, with
each of the three sets being used exactly once as the validation data. In the training set,
3 pixel of each video frame were randomly selected from top 5% salient regions of
ground-truth fixation maps as the positive samples. Similarly, 3 pixel of each video
frame were further chosen from bottom 70% salient regions as negative samples.
Then, both positive and negative samples were available in each cross validation, to
train the C-SVC with (9.8).

9.6.2 Analysis on parameter selection

In HEVC, the bit allocation, splitting depth, and MV of each CTU may change
along with increased or decreased bit rates. Therefore, we analyze the performance
of our method with regard to the videos compressed at different bit rates. Since the
resolutions of test videos vary from 416 × 240 to 1,920 × 1,080, there is an issue on

2
It takes around 100 s for HM to encode a 1080p video frame, in a PC with Intel Core i7-4770 CPU and
16 GB RAM. By contrast, ×265 adopts parallel computing and fast methods to encode videos, such that
real-time 4K HEVC encoding can be achieved by ×265.
Machine-learning-based saliency detection 327

finding bit rates suitable for all videos to ensure proper visual quality. To solve such
an issue, we follow [30] in setting the bit rates of each video for rate control the
same as those of fixed QPs. Then, we report in Figure 9.10 the AUC, CC, and NSS
results of our method at different bit rates. Note that the bit rates averaged over all
33 videos are shown, varying from 2,068 to 100 kbps. Figure 9.10 shows that our
method achieves the best performance in terms of CC and NSS, when the averaged
bit rate of rate control is 430 kbps (equal to those of fixed QP = 37). Therefore, such
bit-rate setting is used for the following evaluation. Figure 9.10 also shows that the
bit rates have slight impact on the overall performance of our method in terms of
AUC, NSS, and CC. The minimum values of AUC, NSS, and CC are above 0.82,
1.52, and 0.41, respectively, at different bit rates, which are superior to all other
methods reported in Section 9.6.3. Besides, one may observe from Figure 9.10 that
the saliency detection accuracy of some HEVC features is fluctuating when the bit
rate is changed. Hence, this figure suggests that our saliency detection should not rely

1
0.85
AUC

0.9
0.8
0 500 1,000 1,500 2,000 0.8
Bit-rates (kbps)

1.6 0.7
AUC
NSS

1.4 MV
0.6 Temporal diff. of MV
Spatial diff. of MV
0 500 1,000 1,500 2,000 Bit allocation
Bit-rates (kbps) 0.5 Temporal diff. of bit allocation
Spatial diff. of bit allocation
Splitting depth
0.4 0.4
CC

Temporal diff. of splitting depth

Spatial diff. of splitting depth

0.3 0.3
0 500 1,000 1,500 2,000 200 400 600 800 1,000 1,200 1,400 1,600 1,800 2,000 2,200
Bit-rates (kbps) Bit-rates (kbps)

1.4 0.35

1.2 0.3

1 0.25
NSS

0.8 0.2
MV MV
Temporal diff. of MV Temporal diff. of MV
0.6 Spatial diff. of MV 0.15 Spatial diff. of MV
Bit allocation Bit allocation
Temporal diff. of bit allocation Temporal diff. of bit allocation
Spatial diff. of bit allocation Spatial diff. of bit allocation
0.4 Splitting depth 0.1
Splitting depth
Temporal diff. of splitting depth Temporal diff. of splitting depth
Spatial diff. of splitting depth Spatial diff. of splitting depth
0.2 0.05
200 400 600 800 1,000 1,200 1,400 1,600 1,800 2,000 2,200 200 400 600 800 1,000 1,200 1,400 1,600 1,800 2,000 2,200
Bit-rates (kbps) Bit-rates (kbps)

Figure 9.10 Performance comparison of our method (first column) and our single
features (second to fourth columns) at different bit rates. The bit rates
of each video in our rate control are the same as those of fixed QPs,
i.e., QP = 27, 32, 35, 37, 39, 42, and 47. Here, the bit rates averaged
over all 33 videos are shown in the horizontal axis
328 Applications of machine learning in wireless communications

0.75 0.73
0.74 0.72
Temporal diff. of MV
0.73 Temporal diff. of bit allocation 0.71
Temporal diff. of splitting depth
0.72 0.7
AUC

AUC
0.71 0.69
0.7 0.68
0.69 0.67 Spatial diff. of splitting depth

0.68 0.66
10 20 30 40 50 60 5 10 15 20 25
σm/σb/σd values ξd values

0.81 0.695
0.8
0.79
0.694
0.78
AUC

AUC

0.77
0.693
0.76
0.75 Spatial diff. of bit allocation Spatial diff. of MV

0.74 0.692
1 2 3 4 5 6 50 60 70 80
ξb values ξm values

Figure 9.11 Saliency detection performance of each single feature at different

parameter settings. Note that only the AUC is utilized here to evaluate
the saliency detection performance. For other metrics (e.g., NSS and
CC), similar results can be found for choosing the optimal values of
parameters

on a single feature. On the contrary, the combination of all features is robust across
various bit rates, implying the benefit of applying the C-SVC in learning to integrate
all HEVC features for saliency detection.
Next, we analyze the parameters of our saliency detection method. When com-
puting the spatial difference features through (9.7), parameters ξd , ξb , and ξm have
been all traversed to find the optimal values. The results are shown in Figure 9.11.
As can be seen in this figure, parameters ξd , ξb , and ξm should be set to 13, 3, and 57
for optimizing saliency detection results. In addition, the saliency detection accuracy
of temporal difference features almost reaches the maximum, when σd , σb , and σm
of (9.4), (9.5), and (9.6) are equivalent to 46, 46, and 26, respectively. Finally, we
achieve the optimal parameter selection for the following evaluation (i.e., ξd = 13,
ξb = 3, ξm = 57, σd = 46, σb = 46, and σm = 26).
The effectiveness of the center bias in saliency detection has been verified in [45],
as humans tend to pay more attention on the center of the image/video than the
Machine-learning-based saliency detection 329

1
0.9
0.8
0.7
True positive rate

0.6
Our
0.5 Fang
0.4 Itti
Judd
0.3 PQFT
0.2 Rudoy
Surprise
0.1 OBDL
0
0 0.2 0.4 0.6 0.8 1
False positive rate

Figure 9.12 ROC curves of saliency detection by our and other state-of-the-art
methods. Note that the results are averaged over frames of all test
videos of 3-fold cross validation

surround. In this chapter, we follow [45] to impose the same center bias map B to both
our and other compared methods, for fair comparison. Specifically, the center bias is
based on the Euclidean distance of each pixel to video frame center (ic , jc ) as follows:

(i − ic )2 + (j − jc )2
B(i, j) = 1 − , (9.14)
ic2 + jc2
where B(i, j) is the center bias value at pixel (i, j). Then, the detected saliency maps
of all methods are weighted by the above center bias maps.

9.6.3 Evaluation on our database

In this section, we evaluate the saliency detection accuracy of our method, in com-
parison with other seven state-of-the-art methods,3 i.e., Itti’s model [10], Bayesian
surprise [14], Judd et al. [19], PQFT [16], Rudoy et al. [23], Fang et al. [27], and
OBDL [28]. Note that 3-fold cross validation was applied in our database for eval-
uation, and the saliency detection accuracy was averaged over the frames of all test
videos of 3-fold cross validation. Furthermore, the saliency maps of some selected
video frames are provided for each cross validation, to show the subjective saliency
detection results of our and other methods.
ROC curves. The ROC curves of our and other seven methods are shown in
Figure 9.12, to evaluate the accuracy of saliency detection in predicting human fix-
ations. As can be seen in this figure, our method generally has higher true positive

3
In our experiments, we directly used the codes by the authors to implement all methods except Fang
et al. [27], which was realized by ourselves as the code is not available online.
330 Applications of machine learning in wireless communications

rates than others at the same false positive rates. In a word, the ROC curves illustrate
the superior performance of our method in saliency detection.
AUC and EER. In order to quantify the ROC curves, we report in Table 9.1
the AUC and EER results of our and other seven state-of-the-art methods. Here,
both mean and standard deviation are provided for the AUC and EER results of
all test video frames of 3-fold cross validation. This table shows that our method
performs better than all other seven methods. Specifically, there are 0.026 and 0.038
enhancement of AUC, over Fang et al. [27] and OBDL [28], respectively, which also
work in compressed domain. The EER of our method has 0.028 and 0.036 decrease,
compared with compressed domain methods of [27,28]. Smaller EER means that there
is a lower misclassifying probability in our method when the false positive rate equals
to the false negative rate. The possible reasons for the improvement of our method
are (1) the new compressed domain features (i.e., CTU structure and bit allocation)
are developed in light of the latest HEVC standard; (2) the camera motion has been
removed in our method; (3) the learning mechanism is incorporated into our method
to bridge the gap between HEVC features and human visual attention. Besides, our
method outperforms uncompressed domain learning-based methods [19,23], with
0.007 and 0.038 improvement in AUC as well as 0.009 and 0.029 reduction in EER.
This verifies the effectiveness of the newly proposed features in compressed domain,
which benefit from the well-developed HEVC standard. However, since extensive
high and middle level features are applied in [19], there is little AUC improvement
(around 0.007) of our method over [19]. Generally speaking, our method outperforms
all other seven methods, which are in compressed or uncompressed domain.
NSS, CC, and KL. Now, we concentrate on the comparison of NSS, CC, and KL
metrics to evaluate the accuracy of saliency detection on all test videos. The averaged
results (with their standard deviation) of NSS, CC, and KL, by our and other seven
state-of-the-art methods, are also reported in Table 9.1. Note that the method with a
higher value of NSS, CC or KL can better predict the human fixations. Again, it can
be seen from Table 9.1 that our method improves the saliency detection accuracy over
all other methods, in the terms of NSS, CC, and KL. Moreover, the improvement of
NSS, CC, and KL, especially CC, is much larger than that of AUC.
Saliency maps. Figure 9.13 shows the saliency maps of four randomly selected
test videos, detected by our and other seven methods, as well as the ground-truth
human fixation maps. Note that the results of only one frame for each video are
shown in these figures. From these figures, one may observe that in comparison with
all other seven methods, our method is capable of well locating the saliency regions
in a video frame, much closer to the maps of human fixations. In summary, the
subjective results here, together with the objective results above, demonstrate that our
method is superior to other state-of-the-art methods in our database.
Computational time. For time-efficiency evaluation, the computational time
of our and other methods has been recorded4 and listed in Table 9.2. We can see
from this table that our method ranks third in terms of computational speed, only

4
All methods were run in the same environment: MATLAB 2012b at a computer with Intel Core i7-4770
[email protected] GHz and 16 GB RAM.
Table 9.1 The averaged accuracy of saliency detection by our and other seven methods, in mean (standard deviation) of all test
videos of 3-fold cross validation over our database

Our Itti [10] Surprise [14] Judd [19] PQFT [16] Rudoy [23] Fang [27] OBDL [28]

AUC 0.823(0.071) 0.688(0.066) 0.752(0.083) 0.816(0.065) 0.750(0.084) 0.785(0.100) 0.797(0.073) 0.785(0.086)

NSS 1.658(0.591) 0.445(0.464) 1.078(0.739) 1.427(0.440) 1.300(0.529) 1.401(0.708) 1.306(0.560) 1.511(0.825)
CC 0.438(0.133) 0.119(0.098) 0.272(0.156) 0.387(0.111) 0.311(0.121) 0.386(0.186) 0.370(0.133) 0.352(0.166)
KL 0.300(0.086) 0.104(0.043) 0.183(0.086) 0.285(0.076) 0.239(0.076) 0.269(0.111) 0.266(0.081) 0.236(0.111)
EER 0.241(0.075) 0.365(0.051) 0.305(0.075) 0.250(0.064) 0.307(0.074) 0.270(0.094) 0.269(0.071) 0.277(0.098)

Note: The bold values indicate the best saliency prediction results in the table.
332 Applications of machine learning in wireless communications

(a) (b) (c) (d) (e) (f) (g) (h) (i) (j)

Figure 9.13 Saliency maps of four videos selected from the first time of our
cross-validation experiments. The maps were yielded by our and other
seven methods as well the ground-truth human fixations. Note that the
results of only one frame are shown for each selected video: (a) input,
(b) human, (c) our, (d) Itti, (e) surprise, (f) Judd, (g) PQFT, (h) Rudoy,
(i) Fang, (j) OBDL

Table 9.2 Computational time per video frame averaged over our database for our
and other seven methods

Our Itti Surprise Judd PQFT Rudoy Fang OBDL

Time (s) 3.1 1.6 40.6 23.9 0.5 98.5 15.4 5.8

slower than Itti [10] and PQFT [16]. However, as discussed above, the performance
of Itti and PQFT is rather inferior compared with other methods, and their saliency
detection accuracy is much lower than that of our method. In summary, our method
has high time efficiency with effective saliency prediction performance. The main
reason is that our method benefits from the modern HEVC encoder and the learning
mechanism, thus not wasting much time on exploiting saliency-detection features. We
further transplanted our method into C++ program on the VS.net platform to figure
out its potential in real-time implementation. After the transplantation, our method
consumes averaged 140 ms per frame over all videos of our database and achieves
real-time detection for 480p videos at 30 frame per second (fps). It is worth pointing
out that some speeding-up techniques, like parallel computing, may further reduce the
computational time of our method for real-time saliency detection of high-resolution
videos.

9.6.4 Evaluation on other databases

For evaluating the generalization of our method, we compared our and other seven
methods on all videos of SFU [41] and DIEM [42], which are two widely used
databases. In DIEM, the first 300 frames of each video were tested for matching the
length of videos in SFU and our databases. Here, all 33 videos of our database were
selected for training the C-SVC classifier. Table 9.3 presents the saliency detection
Table 9.3 Mean (standard deviation) values for saliency detection accuracy of our and other methods over SFU and DIEM databases

SFU

Our Itti [10] Surprise [14] Judd [19] PQFT [16] Rudoy [23] Fang [27] OBDL [28]

AUC 0.83(0.06) 0.70(0.07) 0.65(0.12) 0.77(0.07) 0.72(0.08) 0.79(0.08) 0.80(0.07) 0.80(0.07)

NSS 1.42(0.34) 0.27(0.36) 0.47(0.58) 1.05(0.33) 0.86(0.45) 1.38(0.57) 1.23(0.40) 1.36(0.57)
CC 0.49(0.11) 0.09(0.09) 0.16(0.17) 0.37(0.10) 0.29(0.14) 0.46(0.16) 0.42(0.12) 0.44(0.16)
KL 0.28(0.07) 0.09(0.03) 0.13(0.08) 0.18(0.06) 0.19(0.07) 0.25(0.10) 0.24(0.07) 0.27(0.09)
EER 0.24(0.06) 0.34(0.06) 0.32(0.09) 0.29(0.06) 0.28(0.06) 0.26(0.07) 0.26(0.07) 0.26(0.06)

DIEM

Our Itti [10] Surprise [14] Judd [19] PQFT [16] Rudoy [23] Fang [27] OBDL [28]

AUC 0.86(0.07) 0.77(0.07) 0.75(0.12) 0.75(0.09) 0.79(0.08) 0.80(0.11) 0.80(0.09) 0.79(0.12)

NSS 1.82(0.65) 0.54(0.67) 0.93(0.91) 0.99(0.40) 1.28(0.75) 1.48(0.91) 1.23(0.57) 1.62(1.01)
CC 0.49(0.14) 0.13(0.12) 0.23(0.19) 0.29(0.11) 0.30(0.15) 0.41(0.22) 0.35(0.14) 0.39(0.22)
KL 0.37(0.10) 0.10(0.06) 0.18(0.13) 0.20(0.07) 0.25(0.11) 0.29(0.14) 0.28(0.10) 0.30(0.13)
EER 0.21(0.07) 0.28(0.07) 0.29(0.10) 0.31(0.08) 0.26(0.07) 0.25(0.10) 0.25(0.08) 0.26(0.11)

Note: The bold values indicate the best saliency prediction results in the table.
334 Applications of machine learning in wireless communications

Table 9.4 Comparison to the results reported in [23]

Our PQFT [16] Rudoy [23]

Median shuffled-AUC 0.74 0.68 0.72

Note: The bold values indicate the best saliency prediction results in the table.

accuracy of our and other methods over the SFU and DIEM databases. Again, our
method performs much better than others in terms of all five metrics. Although the
C-SVC was trained on our database, our method still significantly outperforms all
seven conventional methods over other databases.
Although above results were mainly upon the codes by their authors, it is fairer
to compare with the results reported in their literatures. However, it is hard to find
the literatures reporting the results of all seven methods on a same database. Due to
this, we only compare to the reported results of the method with top performance. We
can see from Tables 9.1 and 9.3 that among all methods we compared, Rudoy [23]
generally ranks highest in our, SFU, and DIEM databases. Thus, we implemented our
method on the same database as Rudoy [23] (also the DIEM database), and then we
compared the results of our method to those of seven PQFT [16] and Rudoy [23],
which were reported in [23]. The comparison is provided in Table 9.4. Note that the
comparison is in terms of median shuffled-AUC, as shuffled version of AUC was
measured with median values available in [23]. Note that shuffled-AUC is much
smaller than AUC, due to the removed center bias prior. We can see from Table 9.4
that our method again performs better than [16,23].

9.6.5 Evaluation on other work conditions

For further assessing the generalization of our method, we extended the implemen-
tation of our method at different HEVC working conditions. The working conditions
include HM 16.0 and ×265 v1.8 encoders, at both LD and RA configurations. We
have discussed the parameter settings of these working conditions in Section 9.6.1.
The rate control at these working conditions was also enabled, with the bit rates the
same as above.
Figure 9.14 compares the saliency detection performance of our method applied
to HM and ×265 encoders with LD and RA configurations. The performance is
evaluated in the terms of AUC, CC, NSS, and KL, averaged over all videos of the
three databases, i.e., our, SFU, and DIEM databases. The results of Rudoy [23] and
Fang [27] are also provided in this figure as the reference. As seen from Figure
9.14, although our method in RA performs a bit worse than that in LD, it is much
superior to other state-of-the-art methods. We can further see from Figure 9.14 that the
performance of our method slightly decreases, when using ×265 bitstreams instead
of HM bitstreams. Such a slight decrease is probably due to the simplified process
of ×265 over HM. More importantly, when applied to ×265 bitstreams, our method
still significantly outperforms other methods. In summary, our method is robust to
different working conditions.
Machine-learning-based saliency detection 335

1.8
HM + LD
1.6
1.4 HM + RA
X265 + LD
1.2
X265 + RA
1
Rudoy
0.8
Fang
0.6
0.4
0.2
0
AUC NSS CC KL

Figure 9.14 Performance of our method at different working conditions, compared

with Rudoy [23] and Fang [27]. The performance is assessed in terms
of AUC, NSS, CC, and KL, averaged over all videos of our, SFU, and
DIEM databases

9.6.6 Effectiveness of single features and learning algorithm

It is interesting to investigate the effectiveness of each HEVC feature in our method.
We utilized each single feature of our method to detect saliency of all 33 videos
from our database. Since the learning process is not required when evaluating each
feature of our method, all 33 videos of our database were tested here without any
cross validation. In Table 9.5, we tabulate the saliency detection accuracy of each
single feature, measured by AUC, NSS, CC, KL, and EER. This table shows that the
AUC results of all nine HEVC features in our method are significantly better than
that of random hit, the AUC of which is 0.5. This confirms that the HEVC encoder
can be utilized as an effective feature extractor for saliency detection. Besides, it can
be clearly observed from this table that the accuracy of bit-allocation-related features
ranks the highest among all features. Therefore, we can conclude that the bit allocation
of HEVC is rather effective in saliency detection, compared to other HEVC features.
Furthermore, Figure 9.15 evaluates the robustness of each single feature across
various working conditions (HM+LD, HM+RA, ×265+LD and ×265+RA). Here,
the evaluation is performed on AUC averaged all 33 videos of our database. We can
see that the AUC of each single feature, especially the features of splitting depth,
varies at different working conditions. This implies that each single feature relies on
the working conditions. Benefitting from the machine-learning power of the C-SVC
(presented in Section 9.5), the performance of combining all features is significantly
more robust than a single feature as shown in Figure 9.15. Since the splitting depth is
least robust across various working conditions, we plot in Figure 9.15 the AUC values
of integrating six features (excluding spitting depth related features). It shows that the
integration of six features underperforms the integration of all nine features for all
working conditions. Thus, we can validate that the features of spitting depth are still
able to improve the overall performance of our method at various working conditions.
Table 9.5 Mean (standard deviation) values for saliency detection accuracy by each single feature of our method, averaged over the
frames of all 33 test videos

Basic Temporal difference Spatial difference

Depth Bit MV Depth Bit MV Depth Bit MV

AUC 0.73(0.10) 0.76(0.09) 0.68(0.11) 0.72(0.09) 0.75(0.09) 0.69(0.10) 0.71(0.10) 0.79(0.08) 0.69(0.12)
NSS 0.84(0.49) 1.26(0.72) 0.85(0.67) 0.97(0.55) 1.15(0.63) 0.86(0.61) 0.82(0.50) 1.38(0.70) 0.78(0.62)
CC 0.23(0.12) 0.31(0.15) 0.19(0.15) 0.23(0.12) 0.27(0.14) 0.20(0.15) 0.23(0.13) 0.35(0.15) 0.19(0.15)
KL 0.19(0.09) 0.24(0.10) 0.19(0.09) 0.22(0.08) 0.24(0.09) 0.19(0.08) 0.19(0.08) 0.27(0.09) 0.12(0.09)
EER 0.27(0.08) 0.29(0.09) 0.35(0.09) 0.33(0.08) 0.30(0.08) 0.34(0.09) 0.33(0.09) 0.27(0.09) 0.35(0.02)
Machine-learning-based saliency detection 337

0.82

0.8

0.78

0.76

0.74
AUC

0.72

0.7 MV
Temporal diff. of MV
0.68 Spatial diff. of MV
Bit allocation
Temporal diff. of bit allocation
0.66 Spatial diff. of bit allocation
Splitting depth
Temporal diff. of splitting depth
0.64 Spatial diff. of splitting depth
Six features comb.
Nine features comb.
0.62
H

×2

×2
M

65
+L

+R
D

A
Figure 9.15 AUC curves of saliency detection by each single feature and feature
combination. Six comb. and nine comb. mean the results of saliency
detection by six features (excluding features of splitting depth) and by
all six features, respectively. Similar results can be found for other
metrics, e.g., CC

Table 9.6 The averaged accuracy of saliency detection by our method with C-SVC
and equal weight

AUC NSS CC KL EER

C-SVC 0.823(0.071) 1.658(0.591) 0.438(0.133) 0.300(0.086) 0.241(0.075)

Equal weight 0.775(0.087) 1.268(0.546) 0.330(0.129) 0.247(0.083) 0.279(0.084)

Finally, it is necessary to verify the effectiveness of the C-SVC learning algorithm

in our method, since it bridges the gap between the proposed HEVC features and
saliency. Provided that the learning algorithm is not incorporated, equal weighting
is a common way for feature integration (e.g., in [10]). Table 9.6 compares saliency
detection results of our method with the C-SVC learning algorithm and with equal
weighting. As can be seen in this table, the C-SVC produces significantly better
results in all metrics, compared with the equal weight integration. This indicates the
effectiveness of the learning algorithm applied in our method for saliency detection.
338 Applications of machine learning in wireless communications

9.7 Conclusion
In this chapter, we found out that the state-of-the-art HEVC encoder is not only effi-
cient in video coding but also effective in providing the useful features in saliency
detection. Therefore, this chapter has proposed a novel method for learning to detect
video saliency with several HEVC features. Specifically, to facilitate the study on
video-saliency detection, we first established an eye-tracking database on viewing
33 uncompressed videos from test sets commonly used for HEVC evaluation. The
statistical analysis on our database revealed that human fixations tend to fall into the
regions with the high-valued HEVC features of splitting depth, bit allocation, and
MV. Besides, three observations were also found from our eye-tracking database.
According to the analysis and observations, we proposed to extract and then com-
pute several HEVC features, on the basis of splitting depth, bit allocation, and MV.
Next, we developed the C-SVC, as a nonlinear SVM classifier, to learn the model of
video saliency with regard to the proposed HEVC features. Finally, the experimental
results verified that our method outperforms other state-of-the-art saliency detection
methods, in terms of ROC, EER, AUC, CC, NSS, and KL metrics.
In the reality of wireless multimedia communications, almost all videos exist
in the form of bitstreams, generated by video-coding techniques. Since HEVC is
the latest video-coding standard, there is no doubt that the HEVC bitstreams will be
prevalent in the near future. Accordingly, our method, performed in HEVC domain, is
more practicable over other state-of-the-art uncompressed domain methods, as both
time and storage complexity on decompressing videos can be saved.

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Chapter 10
Deep learning for indoor localization
based on bimodal CSI data
Xuyu Wang1 and Shiwen Mao2

In this chapter, we incorporate deep learning for indoor localization utilizing channel
state information (CSI) with commodity 5 GHz Wi-Fi. We first introduce the state-of-
the-art deep-learning techniques including deep autoencoder network, convolutional
neural network (CNN), and recurrent neural network (RNN). We then present a deep-
learning-based algorithm to leverage bimodal CSI data, i.e., average amplitudes and
estimated angle of arrivals (AOA), for indoor fingerprinting. The proposed scheme
is validated with extensive experiments. Finally, we discuss several open research
problems for indoor localization based on deep-learning techniques.

10.1 Introduction
The proliferation of mobile devices has fostered great interest in indoor-location-based
services, such as indoor navigation, robot tracking in factories, locating workers on
construction sites, and activity recognition [1–8], all requiring accurately identifying
the locations of mobile devices indoors. The indoor environment poses a complex
radio-propagation channel, including multipath propagation, blockage, and shadow
fading, and stimulates great research efforts on indoor localization theory and sys-
tems [9]. Among various indoor-localization schemes, Wi-Fi-based fingerprinting is
probably one of the most widely used. In fingerprinting, a database is first built with
data collected from a thorough measurement of the field in the off-line training stage.
Then, the position of a mobile user can be estimated by comparing the newly received
test data with that in the database. A unique advantage of this approach is that no
extra infrastructure needs to be deployed.
Many existing fingerprinting-based indoor-localization systems use received
signal strength (RSS) as fingerprints, due to its simplicity and low hardware require-
ment [10,11]. For example, radar is one of the first RSS-based fingerprinting systems

1
Department of Computer Science, California State University, United States
2
Department of Electrical and Computer Engineering, Auburn University, United States
344 Applications of machine learning in wireless communications

that incorporate a deterministic method for location estimation [10]. For higher accu-
racy, Horus, another RSS-based fingerprinting scheme, adopts a probabilistic method
based on K-nearest neighbor (KNN) [9] for location estimation [11]. The performance
of RSS-based schemes is usually limited by two inherent shortcomings of RSS. First,
due to the multipath effect and shadow fading, the RSS values are usually highly
diverse, even for consecutively received packets at the same position. Second, RSS
value only reflects the coarse channel information, since it is the sum of the powers
of all received signals.
Unlike RSS, CSI represents fine-grained channel information, which can now
be extracted from several commodity Wi-Fi network interface cards (NIC), e.g., Intel
Wi-Fi Link 5300 NIC [12], the Atheros AR9390 chipset [13], and the Atheros AR9580
chipset [14]. CSI consists of subcarrier-level measurements of orthogonal frequency
division multiplexing (OFDM) channels. It is a more stable representation of chan-
nel characteristics than RSS. Several CSI-based fingerprinting systems have been
proposed and shown to achieve high localization accuracy [15,16]. For example, the
fine-grained indoor fingerprinting system (FIFS) [15] uses a weighted average of
CSI values over multiple antennas. To fully exploit the diversity among the multiple
antennas and subcarriers, DeepFi [16] learns a large amount of CSI data from the
three antennas and 30 subcarriers with an autoencoder. These CSI-based schemes only
use the amplitude information of CSI, since the raw phase information is extremely
random and not directly usable [17].
Recently, for the Intel 5300 NIC in 2.4 GHz, two effective methods have been
proposed to remove the randomness in raw CSI phase data. In [18], the measured
phases from 30 subcarriers are processed with a linear transformation to mitigate
the random phase offsets, which is then employed for passive human-movement
detection. In [17], in addition to the linear transformation, the difference of the
sanitized phases from two antennas is obtained and used for line-of-sight (LOS)
identification. Although both approaches can stabilize the phase information, the
mean value of phase will be zero (i.e., lost) after such processing. This is actually
caused by the firmware design of the Intel 5300 NIC when operating on the 2.4 GHz
band [19]. To address this issue, Phaser [19] proposes to exploit CSI phase in 5 GHz
Wi-Fi. Phaser constructs an AOA pseudospectrum for phase calibration with a single
Intel 5300 NIC. These interesting works motivate us to explore effectively cleansed
phase information for indoor fingerprinting with commodity 5 GHz Wi-Fi.
In this chapter, we investigate the problem of fingerprinting-based indoor local-
ization with commodity 5 GHz Wi-Fi. We first present three hypotheses on CSI
amplitude and phase information for 5 GHz OFDM channels. First, the average
amplitude over two antennas is more stable over time for a fixed location than that
from a single antenna as well as RSS. Second, the phase difference of CSI values
from two antennas in 5 GHz is highly stable. Due to the firmware design of Intel
5300 NIC, the phase differences of consecutively received packets form four clusters
when operating in 2.4 GHz. Such ambiguity makes measured phase difference unus-
able. However, we find this phenomenon does not exist in the 5 GHz band, where
all the phase differences concentrate around one value. We further design a sim-
ple multi-radio hardware for phase calibration which is different from the technique
Deep learning for indoor localization based on bimodal CSI data 345

in [19] that uses AOA pseudospectrum search with a high computation complexity,
to calibrate phase in single Intel 5300 NIC. As a result, the randomness from the
time and frequency difference between the transmitter and receiver, and the unknown
phase offset can all be removed, and stable phase information can be obtained. Third,
the calibrated phase difference in 5 GHz can be translated into AOA with considerable
accuracy when there is a strong LOS component. We validate these hypotheses with
both extensive experiments and simple analysis.
We then design BiLoc, bimodal deep learning for indoor localization with com-
modity 5 GHz Wi-Fi, to utilize the three hypotheses in an indoor fingerprinting
system [20]. In BiLoc, we first extract raw amplitude and phase data from the three
antennas, each with 30 subcarriers. We then obtain bimodal data, including average
amplitudes over pairs of antennas and estimated AOAs, with the calibration procedure
discussed above. In the off-line training stage, we adopt an autoencoder with three
hidden layers to extract the unique channel features hidden in the bimodal data and
propose to use the weights of the deep network to store the extracted features (i.e.,
fingerprints). To reduce the computational complexity, we propose a greedy learn-
ing algorithm to train the deep network in a layer-by-layer manner with a restricted
Boltzmann machine (RBM) model. In the online test stage, bimodal test data is first
collected for a mobile device. Then a Bayesian probability model based on the radial
basis function (RBF) is leveraged for accurate online position estimation.
In the rest of this chapter, preliminaries on deep learning for indoor localization
is introduced Section 10.2. Then, the three hypotheses are given in Section 10.3.
We present the BiLoc system in Section 10.4 and validate its performance in
Section 10.5. Section 10.6 discusses future research problems for indoor localization,
and Section 10.7 concludes this chapter.

10.2 Deep learning for indoor localization

With the rapid growth of computation platforms likeTensorflow, Caffe, andTorch [21],
deep learning has been widely applied in a variety of areas such as object recogni-
tion, natural-language processing, computer vision, robotics, automated vehicles,
and artificial intelligence (AI) games [22]. Compared with shallow machine-learning
algorithms, such as support vector machine (SVM) and KNN, deep learning is a
branch of machine learning, which implements nonlinear transformations with mul-
tiple hidden layers and has high-level data abstractions. In addition, deep learning can
train the weights and bias of the network with a huge quantity of data for improving
classification performance and data-representation capability, which includes unsu-
pervised and supervised learning with different deep-learning models [23]. In this
chapter, three different deep-learning frameworks are discussed below for indoor
localization problems.

10.2.1 Autoencoder neural network

A deep autoencoder neural network is an unsupervised learning, which can produce
output data that is a de-noised input data. Moreover, it is also used to extract data
346 Applications of machine learning in wireless communications

Original Reconstructed
data data

Encode Decode

Figure 10.1 Autoencoder

features or reduce the size of data, which is more powerful than principal component
analysis-based methods because of its nonlinear transformations with multiple hidden
layers. Figure 10.1 shows the architecture of the deep autoencoder neural network. For
training, a deep autoencoder neural network has three stages including pretraining,
unrolling, and fine-tuning [24]. In the pretraining stage, each neighboring set of two
layer is considered as an RBM, is denoted as a bipartite undirected graphical model.
Then, a greedy algorithm is used to train the weights and biases for a stack of RBMs.
In the unrolling stage, the deep autoencoder network is unrolled to obtain the recon-
structed input data. Finally, the fine-tuning phase employs the backpropagation (BP)
algorithm for training the weights in the deep autoencoder network by minimizing
the loss function (i.e., the error).
The first work that applies a deep autoencoder to indoor localization is
DeepFi [16,25], which is a deep autoencoder network-based indoor fingerprinting
method with CSI amplitudes. For every training location, the deep autoencoder net-
work is trained to obtain a set of weights and biases, which are used as fingerprints
for the corresponding locations. For online test, the true location is estimated based
on the Bayesian scheme. The experimental results show that the mean distance error
in a living room environment and a laboratory environment is 1.2 and 2.3 m, respec-
tively. In addition, PhaseFi [26,27] is proposed to use CSI calibrated phase, which
still incorporates a deep autoencoder networks for indoor localization. Moreover, deep
autoencoder networks are used for device-free indoor localization [28,29]. The denois-
ing autoencoder-based indoor localization with Bluetooth Low Energy (BLE) is also
used to provide 3-D localization [30]. In this chapter, we consider deep autoencoder
networks for indoor localization using bimodal CSI data.

10.2.2 Convolutional neural network

CNN is also a useful deep-learning architecture, which has been successfully used
in computer vision and activity recognition [23,31]. In 1998, LeCun proposed
LeNet-5 [32], which is the first architecture of CNN. Figure 10.2 shows the CNN
framework, which includes the convolutional layers, subsampling layers, and fully
connected layers.
Deep learning for indoor localization based on bimodal CSI data 347

Feature Feature Feature Feature

Input Output
maps maps maps maps
24 × 24 20@1 × 1
4@20 × 20 4@10 × 10 8@8 × 8 8@4 × 4

Convolution Subsampling Convolution Subsampling Convolution

Figure 10.2 CNN

The convolutional layer can obtain feature maps within local regions in the pre-
vious layer’s feature maps with linear convolutional filters, which is followed by
nonlinear activation functions. The subsampling layer is to decrease the resolution of
the feature maps by downsampling over a local neighborhood in the feature maps of
the previous layer, which is invariant to distortions in the input data [33]. The feature
maps in the previous layer are pooled over a local temporal neighborhood using the
mean pooling function. Other operations such as the sum or max pooling function
can also be incorporated in the subsampling layer.
After the convolutional and subsampling layers, there is a fully connected layer,
which is a basic neural network with one hidden layer, to train the output data. More-
over, a loss function is used to measure the difference between the true location label
and the output of CNN, where the squared error or cross entropy is used as loss func-
tion for training the weights. Currently, an increasing number of CNN models are
proposed, such as AlexNet [31] and ResNet [34]. AlexNet is a larger and more com-
plex model, where Max pooling and rectified linear unit (ReLU) nonlinear activation
function are used in the model [35]. Moreover, dropout regularization is used to han-
dle the overfitting problem. ResNet was proposed by Microsoft, where the residual
block includes a direct path between the input and output, and the batch normaliza-
tion technique is used to avoid diminishing or exploding of the gradient. ResNet is a
152 layers residual learning framework, which won the ILSVRC 2015 classification
competition [31].
For indoor localization problems, the CiFi [33,36] system leverages the con-
structed images with estimated AOA values with commodity 5 GHz Wi-Fi for indoor
localization. This system demonstrates that the performance of the localization has
outperformed several existing schemes, like FIFS and Horus. Motivated by ResNet,
the ResLoc [37] system uses bimodal CSI tensor data to train a deep residual sharing
learning, which can achieve the best performance among deep-learning-based local-
ization methods using CSI. CSI amplitude is also used to obtain CSI images for indoor
localization [38]. In addition, input images by using received signal strength indicator
(RSSI) of Wi-Fi signals are leveraged to train a CNN model [39,40]. CNN has also
been used for TDoA-based localization systems, which can estimate nonlinearities in
the signal propagation space but also predict the signal for multipath effects [41].
348 Applications of machine learning in wireless communications

ht1 h t2

Ct–1 Ct
tanh tanh

σ σ tanh σ σ σ tanh σ
ht–1 ht

X t1 X t2

Figure 10.3 LSTM

10.2.3 Long short-term memory

To process variable-length sequence inputs, RNNs are proposed, where long range
dependencies can be captured using the feedback loop in the recurrent layer. However,
the dependencies also makes it hard to train an RNN, because of diminishing or
exploding of the gradient of the loss function. Long short-term memory (LSTM) is
proposed to handle the above problem, which has been widely applied for sequence
data processing [42].
For the LSTM algorithm in Figure 10.3, the input gate i decides how much new
information will be exploited in the current memory cell, the forget gate f controls
how much information will be removed from the old memory cell, and the output
gate o determines how much data will be output based on the current memory cell
c. In addition, the sigmoid function σ can control how much information can be
updated and the hyperbolic tangent function tanh can create new candidate values g.
Thus, unlike RNN, LSTM can handle long-term dependency and has better data
representation ability, and has been employed for speech recognition, machine
translation, and time-series problems.
The recently proposed DeepML system uses a two-layer LSTM network for a
higher learning and representation ability on exploiting magnetic and light sensor data
for indoor localization, which can achieve submeter level localization accuracy [43].
LSTM can be used for sequence-based localization problems with other signals. We
have also applied LSTM to wheat moisture level detection [44] and forecasting of
renewable energy generation [45].

10.3 Preliminaries and hypotheses

10.3.1 Channel state information preliminaries

OFDM is widely used in wireless network standards, such as Wi-Fi (e.g., IEEE
802.11a/g/n), where the total spectrum is partitioned into multiple orthogonal subcar-
riers, and wireless data is transmitted over the subcarriers using the same modulation
Deep learning for indoor localization based on bimodal CSI data 349

and coding scheme to mitigate frequency selective fading. Leveraging the device
driver for off-the-shelf NICs, e.g., the Intel 5300 NIC, we can extract CSI for each
received packet, that is a fine-grained physical layer (PHY) information. CSI reveals
the channel characteristics experienced by the received signal such as the multipath
effect, shadow fading, and distortion.
With OFDM, the Wi-Fi channel at the 5 GHz band can be considered as a nar-
rowband flat fading channel. In the frequency domain, the channel model can be
expressed as
Y = CSI · X + N , (10.1)
where Y and X denote the received and transmitted signal vectors, respectively,
N is the additive white Gaussian noise (AWGN), and CSI represents the channel’s
frequency response, which can be computed from Y and X .
Although a Wi-Fi receiver uses an OFDM system with 56 subcarriers for a
20 MHz channel, the Intel 5300 NIC can report 30 out of 56 subcarriers. The channel
frequency response of subcarrier i, CSIi , is a complex value, that is,
CSIi = Ii + jQi = |CSIi | exp( j∠CSIi ), (10.2)
where Ii and Qi are the in-phase component and quadrature component, respectively;
|CSIi | and ∠CSIi are the amplitude response and phase response of subcarrier i,
respectively.

10.3.2 Distribution of amplitude and phase

In general, both Ii and Qi can be modeled as i.i.d. AWGN of variance σ 2 .
The amplitude response is |CSIi | = Ii2 + Qi2 , which follows a Rician distribution
when there is a strong LOS component [46]. The probability distribution function
(PDF) of the amplitude response is given by

|CSIi | |CSIi |2 + |CSI0 |2 |CSIi | · |CSI0 |
f (|CSIi |) = × exp − × I 0 , (10.3)
σ2 2σ 2 σ2
where |CSI0 | is the amplitude response without noise, I0 (·) is the zeroth-
order modified Bessel function of the first kind. When the signal-to-noise ratio
is high, the PDF f (|CSIi |) will converge to the Gaussian distribution as
(SNR)
N ( |CSI0 |2 + σ 2 , σ 2 ) [46].
The phase response of subcarrier i is computed by ∠CSIi = arctan(Qi /Ii ) [46].
The phase PDF is given by

1 |CSI0 |2 |CSI0 | √
f (∠CSIi ) = exp − 1 + 2π cos(∠CSIi )
2π 2σ 2 σ

|CSI0 |2 cos2 (∠CSIi ) |CSI0 | cos(∠CSIi )
× exp 1 − Q ,
2σ 2 σ
where Q(·) is the Q-function. In the high SNR regime, the PDF f (∠CSIi ) also
converges to a Gaussian distribution as N 0, (σ/|CSI0 |)2 [46]. The distribution
of amplitude and phase of the subcarriers would be useful to guide the design of
localization algorithms.
350 Applications of machine learning in wireless communications

10.3.3 Hypotheses
We next present three important hypotheses about the CSI data on 5 GHz OFDM chan-
nels, which are demonstrated and tested with our measurement study and theoretical
analysis.
10.3.3.1 Hypothesis 1
The average CSI amplitude value of two adjacent antennas for the 5 GHz OFDM
channel is highly stable for a fixed location.
We find CSI amplitude values exhibit great stability for continuously received
packets at a given location. Figure 10.4 presents the cumulative distribution functions
(CDF) of the standard deviations (STD) of (i) the normalized CSI amplitude averaged
over two adjacent antennas, (ii) the normalized CSI amplitude from a single antenna,
and (iii) the normalized RSS amplitude from a single antenna, for 90 positions. At
each position, 50 consecutive packets are received by the Intel 5300 NIC operating on
the 5 GHz band. It can be seen that 90% of the testing positions are blow 10% of the
STD in the case of averaged CSI amplitudes, while the percentage is 80% for the case
of single antenna CSI and 70% for the case of single antenna RSS. Thus, averaging
over two adjacent antennas can make CSI amplitude highly stable for a fixed location
with 5 GHz OFDM channels. We conduct the measurements over a long period of
time, including midnight and business hours. No obvious difference in the stability of
CSI is observed over different times, while RSS values exhibit large variations even
for the same position. This finding motivates us to use average CSI amplitudes of two
adjacent antennas as one of the features of deep learning in the BiLoc design.
Recall that the PDF of the amplitude response of a single antenna is Gaussian
in the high SNR regime. Assuming that the CSI values of the two antennas are i.i.d.

0.8

0.6
CDF

0.4

Average CSI amplitude in 5 GHz

0.2
Single CSI amplitude in 5 GHz
RSS in 5 GHz
0
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7
STD of normalized amplitude

Figure 10.4 CDF of the standard deviations of normalized average CSI amplitude,
a single CSI amplitude, and a single RSS in the 5 GHz OFDM channel
for 90 positions
Deep learning for indoor localization based on bimodal CSI data 351

(true when the antennas are more than a half wavelength apart [17]), the average
CSI amplitudes also follow the Gaussian distribution, as N ( |CSI0 |2 + σ 2 , σ 2 /2),
but with a smaller variance. This proves that stability can be improved by averaging
CSI amplitudes over two antennas [47] (as observed in Figure 10.4). We consider the
average CSI amplitudes over two antennas instead of three antennas or CSI amplitudes
from only one antenna, and BiLoc employs bimodal data, including estimated AOAs
and average amplitudes. This requires that we use the same number of nodes as the
input for the deep network.

10.3.3.2 Hypothesis 2
The difference of CSI phase values between two antennas of the 5 GHz OFDM channel
is highly stable, compared to that of the 2.4 GHz OFDM channel.
Although the CSI phase information is also available from the Intel 5300 NIC,
it is highly random and cannot be directly used for localization, due to noise and
the unsynchronized time and frequency of the transmitter and receiver. Recently, two
useful algorithms are used to remove the randomness in CSI phase. The first approach
is to make a linear transform of the phase values measured from the 30 subcarriers [18].
The other one is to exploit the phase difference between two antennas in 2.4 GHz and
then remove the measured average [17]. Although both methods can stabilize the CSI
phase in consecutive packets, the average phase value they produce is always near
zero, which is different from the real phase value of the received signal.
Switching to the 5 GHz band, we find the phase difference becomes highly
stable. In Figure 10.5, we plot the measured phase differences of the 30 subcarriers
between two antennas for 200 consecutively received packets in the 5 GHz (in blue)
and 2.4 GHz (in red) bands. The phase difference of the 5 GHz channel varies between
[0.5, 1.8], which is considerably more stable than that of the 2.4 GHz channel (varies
between [−π, π ]). To further illustrate this finding, we plot the measured phase
differences on the fifth subcarrier between two antennas using polar coordinates in
Figure 10.6. We find that all the 5 GHz measurements concentrate around 30◦ , while
the 2.4 GHz measurements form four clusters around 0◦ , 90◦ , 180◦ , and 270◦ . We
conjecture that this may be caused by the firmware design of the Intel 5300 NIC when
operating on the 2.4 GHz band, which reports the phase of channel modulo π/2 rather
than 2π on the 5 GHz band [19]. Comparing to the ambiguity in the 2.4 GHz band,
the highly stable phase difference in the 5 GHz band could be very useful for indoor
localization.
As in Hypothesis 1, we also provide an analysis to validate the observation from
the experiments. Let ∠CSI i denote the measured phase of subcarrier i, which is given
by [14,48]:

i = ∠CSIi + (λp + λs )mi + λc + β + Z,

∠CSI (10.4)

where ∠CSIi is the true phase; Z is the measurement noise; β is the initial phase
offset because of the phase-locked loop; mi is the subcarrier index of subcarrier i;
λp , λs , and λc are phase errors from the packet boundary detection (PBD); the
352 Applications of machine learning in wireless communications

4
5 GHz
3
Phase difference 2

−1

−2
2.4 GHz
−3

−4
0 50 100 150 200
Number of packets

Figure 10.5 The measured phase differences of the 30 subcarriers between two
antennas for 200 consecutively received packets in the 5 GHz (blue)
and 2.4 GHz (red) bands
90 800
120 60
600

150 400 30

200

180 0

210 330

240 300
270

Figure 10.6 The measured phase differences of the fifth subcarrier between two
antennas for 200 consecutively received packets in the 5 GHz (blue
dots) and 2.4 GHz (red crosses) bands

sampling frequency offset and central frequency offset, respectively [48], which are
expressed by
⎧
⎪ t
⎪
⎪ λp = 2π
⎪
⎪ N
⎨
T − T Ts (10.5)
⎪ s
⎪ λ = 2π n
⎪
⎪ T T
⎪
⎩
u

λc = 2π f Ts n,
Deep learning for indoor localization based on bimodal CSI data 353

where t is the PBD delay, N is the fast Fourier transform (FFT) size, T and T are the
sampling periods from the receiver and the transmitter, respectively, Tu is the length
of the data symbol, Ts is the total length of the data symbol and the guard interval, n
is the sampling time offset for current packet, f is the center frequency difference
between the transmitter and receiver. It is noticed that we cannot obtain the exact
values about t, (T − T )/T , n, f , and β in (10.4) and (10.5). Moreover, λp , λs ,
and λc vary for different packets with different t and n. Thus, the true phase ∠CSIi
cannot be derived from the measured phase value.
However, note that the three antennas of the Intel 5300 NIC use the same clock and
the same down-converter frequency. Consequently, the measured phases of subcarrier
i from two antennas have identical packet detection delay, sampling periods, and
frequency differences (and the same mi ) [19]. Thus the measured phase difference on
subcarrier i between two antennas can be approximated as
i = ∠CSIi + β + Z,
∠CSI (10.6)
where ∠CSIi is the true phase difference of subcarrier i, β is the unknown differ-
ence in phase offsets, which is in fact a constant [19], and Z is the noise difference.
We find that ∠CSI i is stable for different packets because of (10.6) where t and
n are cancelled.
In the high SNR regime, the PDF of the phase response of subcarrier i for each
of the antennas is N (0, (σ/|CSI0 |)2 ). Due to the independent phase responses, the
measured phase difference of subcarrier i is also Gaussian with N (β, 2σ 2 (1 +
1/|CSI0 |2 )). Note that although the variance is higher comparing to the true-phase
response, the uncertainty from the time and frequency differences is removed, leading
to much more stable measurements (as shown in Figure (10.6)).
10.3.3.3 Hypothesis 3
The calibrated phase difference in 5 GHz can be translated into the AOA with
considerable accuracy when there is a strong LOS component.
The measured phase difference on subscriber i can be translated into an estimation
of AOA, as

iλ
∠CSI
θ = arcsin , (10.7)
2πd
where λ is the wavelength and d is the distance between the two antennas (set to
d = 0.5λ in our experiments). Although the measured phase difference ∠CSI i is
highly stable, we still wish to remove the unknown phase offset difference β to
further reduce the error of AOA estimation. For commodity Wi-Fi devices, the only
existing approach for a single NIC, to the best of our knowledge, is to search for β
within an AOA pseudospectrum in the range of [−π , π ], which, however, has a high
time complexity [19].
In this chapter, we design a simple method to remove the unknown phase offset
difference β using two Intel 5300 NICs. As in Figure 10.7, we use one Intel 5300 NIC
as transmitter and the other as receiver, while a signal splitter is used to route signal
from antenna 1 of the transmitter to antennas 1 and 2 of the receiver through cables
354 Applications of machine learning in wireless communications

Antenna 1 Antenna 1

Antenna 2 Signal Antenna 2

splitter

Antenna 3 Antenna 3

NIC 1 NIC 2

Transmitter Receiver

Figure 10.7 The multi-radio hardware design for calibrating the unknown phase
offset difference β

−10
Magnitude (dB)

−20

−30

−40

−50
−100 −50 0 50 100
Angle (degree)

Figure 10.8 The estimated AOAs from the 30 subcarriers using the MUSIC
algorithm, while the real AOA is 14◦

of the same length. Since the two antennas receive the same signal, the true phase
difference ∠CSIi of subcarrier i is zero. We can thus obtain β as the measured
phase offset difference between antennas 1 and 2 of the receiver. We also use the
same method to calibrate antennas 2 and 3 of the receiver, to obtain the unknown
phase offset difference between them as well. We find that the unknown phase offset
difference is relatively stable over time.
Having calibrated the unknown phase offset differences for the three antennas,
we then use the MUSIC algorithm for AOA estimation [49]. In Figure 10.8, the AOA
estimation using MUSIC with the calibrated phase information for the 30 subcarriers
is plotted for a high SNR signal with a known incoming direction of 14◦ . We can see
that the peak occurs at around 20◦ in Figure 10.8, indicating an AOA estimation error
of about 6◦ .
Deep learning for indoor localization based on bimodal CSI data 355

We can obtain the true incoming angle with MUSIC when the LOS component is
strong. To deal with the case with strong NLOS paths (typical in indoor environments),
we adopt a deep network with three hidden layers to learn the estimated AOAs and the
average amplitudes of adjacent antenna pairs as fingerprints for indoor localization.
As input to the deep network, the estimated AOA is obtained as follows:

θ = arcsin ∠CSI i − β λ π
+ , (10.8)
2πd 2
where β is measured with the proposed multi-radio hardware experiment. The
estimated AOA is in the range of [0, π].

10.4 The BiLoc system

10.4.1 BiLoc system architecture

The overall architecture of BiLoc is illustrated in Figure 10.9. The BiLoc design uses
only one access point and one mobile device, each equipped with an Intel 5300 NIC,
servicing as receiver and transmitter, respectively. All the communications are on the

CSI data collection

Access point Mobile device

Bimodal data extraction

Location 1
bimodal data
... Location N
bimodal data

Deep learning

Bimodal fingerprint database

Off-line
Online
Test location X
Data fusion
Bimodal data

Estimated location

Figure 10.9 The BiLoc system architecture

356 Applications of machine learning in wireless communications

5 GHz band. The Intel 5300 NIC has three antennas; at each antenna, we can read
CSI data from 30 subcarriers. Thus we can collect 90 CSI data for every received
packet. We then calibrate the phase information of the received CSI data using our
multi-radio hardware design (see Figure 10.7). Both the estimated AOAs and average
amplitudes of two adjacent antennas are used as location features for building the
fingerprint database.
A unique feature of BiLoc is its bimodal design. With the three receiving antennas,
we can obtain two groups of data: (i) 30 estimated AOAs and 30 average amplitudes
from antennas 1 and 2 and (ii) that from antennas 2 and 3. BiLoc utilizes estimated
AOAs and average amplitudes for indoor fingerprinting for two main reasons. First,
these two types of CSI data are highly stable for any given position. Second, they are
usually complementary to each other under some indoor circumstances. For example,
when a signal is blocked, the average amplitude of the signal will be significantly
weakened, but the estimated AOA becomes more effective. On the other hand, when
the NLOS components are stronger than the LOS component, the average amplitude
will help to improve the localization accuracy.
Another unique characteristic of BiLoc is the use of deep learning to produce
feature-based fingerprints from the bimodal data in the off-line training stage, which
is quite different from the traditional approach of storing the measured data as fin-
gerprints. Specifically, we use the weights in the deep network to represent the
features-based fingerprints for every position. By obtaining the optimal weights with
the bimodal data on estimated AOAs and average amplitudes, we can establish a
bimodal fingerprint database for the training positions. The third feature of BiLoc
is the probabilistic data fusion approach for location estimation based on received
bimodal data in the online test stage.

10.4.2 Off-line training for bimodal fingerprint database

In the off-line stage, BiLoc leverages deep learning to train and store the weights to
build a bimodal fingerprint database, which is a deep autoencoder that involves three
phases: pretraining, unrolling, and fine-tuning [24]. In the pretraining phase, a deep
network with three hidden layers and one input layer is used to learn the bimodal
data. We denote hi as the hidden variable with Ki nodes at layer i, i = 1, 2, 3, and h0
as the input data with K0 nodes at the input layer. Let the average amplitude data be
v1 and the estimated AOA data be v2 . To build the bimodal fingerprint database, we
set h0 = v1 and h0 = v2 for databases 1 and 2, respectively, each of which is a set of
optimal weights. We denote W1 , W2 , and W3 as the weights between input data and
the first hidden layer, the first, and second hidden layers, and the second and third
hidden layers, respectively.
We define Pr(h0 , h1 , h2 , h3 ) as the probabilistic generative model for the deep
network. To derive the optimal weights, we maximize the marginal distribution of the
input data for the deep network, which is given by

max Pr(h0 , h1 , h2 , h3 ). (10.9)
{W1 ,W2 ,W3 }
h1 h2 h3
Deep learning for indoor localization based on bimodal CSI data 357

Because of the large number of nodes and the complex model structure, it is
difficult to find the optimal weights for the input data with the maximum likelihood
method. To reduce the computational complexity, BiLoc utilizes a greedy learning
algorithm to train the weights layer by layer based on a stack of RBMs [50]. We
consider an RBM as a bipartite undirected graphical model [50] with joint distribution
Pr(hi−1 , hi ), as
exp(−E(hi−1 , hi ))
Pr(hi−1 , hi ) = i−1 , hi ))
, (10.10)
hi−1 hi exp(−E(h

where E(hi−1 , hi ) denotes the free energy between layer (i − 1) and layer i, which is
given by
E(hi−1 , hi ) = −bi−1 hi−1 − bi hi − hi−1 Wi hi , (10.11)
where b and b are the biases for the units of layer (i − 1) and that of layer i,
i−1 i

respectively. To obtain the joint distribution Pr(hi−1 , hi ), the CD-1 algorithm is used
to approximate it as [50]:
⎧ i−1
⎨Pr(hi−1 |hi ) = Kj=1 j |h )
Pr(hi−1 i

(10.12)
⎩Pr(hi |hi−1 ) = Ki Pr(hi |hi−1 ),
j=1 j

j |h ) and Pr(hj |h
where Pr(hi−1 i i i−1
) are given by the sigmoid belief network as follows:
⎧ −1
⎪ Ki
⎨Pr(hi−1
j |h i
) = 1 + exp(−b i−1
j − t=1 W i
j,t i
h t )
Ki−1 j,t i−1 −1 (10.13)
⎪
⎩Pr(h |h ) = 1 + exp(−b − t=1 Wi ht )
i i−1 i
j j .

We propose a greedy algorithm to train the weights and biases for a stack of
RBMs. First, with the CD-1 method, we use the input data to train the parameters
{b0 , b1 , W1 } of the first layer RBM. Then, the parameters {b0 , W1 } are frozen, and we
sample from the conditional probability Pr(h1 |h0 ) to train the parameters {b1 , b2 , W2 }
of the second layer RBM. Next, we freeze the parameters {b0 , b1 , W1 , W2 } of the
first and second layers and then sample from the conditional probability Pr(h2 |h1 ) to
train the parameters {b2 , b3 , W3 } of the third layer RBM. In order to train the weights
and biases of each RBM, we use the CD-1 method to approximate them. For the
layer i RBM model, we estimate ĥi−1 by sampling from the conditional probability
Pr(hi−1 |hi ); by sampling from the conditional probability Pr(hi |ĥi−1 ), we can estimate
ĥi . Thus, the parameters are updated as follows:
⎧
⎨Wi = ε(h h − ĥ ĥ )
⎪ i−1 i i−1 i

bi = ε(hi − ĥi ) (10.14)

⎪
⎩ i−1
b = ε(h − ĥ ),
i−1 i−1

where ε is the step size.

After the pretraining phase, we obtain the near-optimal weights for the deep
network. We then unroll the deep network with forward propagation to obtain the
reconstructed input data in the unrolling phase. Finally, in the fine-tuning phase, the
358 Applications of machine learning in wireless communications

BP algorithm is used to train the weights in the deep network according to the error
between the input data and the reconstructed input data. The optimal weights are
obtained by minimizing the error. In BiLoc, we use estimated AOAs and average
amplitudes as input data and obtain two sets of optimal weights for the bimodal
fingerprint database.

10.4.3 Online data fusion for position estimation

In the online phase, we adopt a probabilistic approach to location estimation based on
the bimodal fingerprint database and the bimodal test data. We derive the posteriori
probability Pr(li |v1 , v2 ) using Bayes’ law as

Pr(li ) Pr(v1 , v2 |li )

Pr(li |v1 , v2 ) = N , (10.15)
j=1 Pr(lj ) Pr(v , v |lj )
1 2

where N is the number of reference locations, li is the ith reference location in the
bimodal fingerprint database, and Pr(li ) is the prior probability that the mobile device
is considered to be at the reference location li . Without loss of generality, we assume
that Pr(li ) is uniformly distributed. The posteriori probability Pr(li |v1 , v2 ) becomes:

Pr(v1 , v2 |li )
Pr(li |v1 , v2 ) = N . (10.16)
j=1 Pr(v , v |lj )
1 2

In BiLoc, we approximate Pr(v1 , v2 |li ) with an RBF in the similar form of a

Gaussian function, to measure the degree of similarity between the reconstructed
bimodal data and the test bimodal data, given by

v1 − v̂1 v2 − v̂2
Pr(v1 , v2 |li ) = exp −(1 − ρ) −ρ , (10.17)
η1 σ 1 η2 σ 2

where v̂1 and v̂2 are the reconstructed average amplitude and reconstructed AOA,
respectively; σ1 and σ2 are the variance of the average amplitude and estimated AOA,
respectively; η1 and η2 are the parameters of the variance of the average amplitude
and estimated AOA, respectively; and ρ is the ratio for the bimodal data.
For the (10.17), the average amplitudes v̂1 and the estimated AOAs v̂2 are as
the input of deep network, where the different nodes of the input can express the
different CSI channels. Then, by employing the test data v̂1 and v̂2 , we compute the
reconstructed average amplitude v̂1 and reconstructed AOA v̂2 based on databases 1
and 2, respectively, which is used to compute the likelihood function Pr(v1 , v2 |li ).
The location of the mobile device can be finally estimated as a weighted average
of all the reference locations, which is given by

N
l̂ = Pr(li |v1 , v2 ) · li . (10.18)
i=1
Deep learning for indoor localization based on bimodal CSI data 359

10.5 Experimental study

10.5.1 Test configuration

We present our experimental study with BiLoc in the 5 GHz band in this section. In
the experiments, we use a desktop computer as an access point and a Dell laptop as
a mobile device, both equipped with an Intel 5300 NIC. In fact, we use the desktop
computer instead of the commodity routers that are not equipped with the Intel 5300
NIC nowadays. Our implementation of BiLoc is executed on the Ubuntu desktop
14.04 LTS OS for both the access point and mobile device. We use Quadrature Phase
Shift Keying (QPSK) modulation and a 1/2 coding rate for the OFDM system. For
the access point, it is set in monitor model, and the distance between two adjacent
antennas is d = 2.68 cm, which is half of a wavelength for the 5 GHz band. For the
mobile device, it transmits packets at 100 packets per second using only one antenna
in injection mode. By using packet-injection technique based on LORCON version
1, 5 GHz CSI data can be obtained. Then, we extract bimodal data for training and
test stages as described in Section 10.4.2.
We also implement three representative schemes from the literature, i.e.,
Horus [11], FIFS [15], and DeepFi [16]. For a fair comparison, all the schemes
use the same measured dataset captured in the 5 GHz band to estimate the location
of the mobile device. We conduct extensive experiments with the schemes in the
following two representative indoor environments:
Computer laboratory: This is a 6 × 9 m2 computer laboratory, a cluttered environment
with metal tables, chairs, and desktop computers, blocking most of the LOS paths.
The floor plan is shown in Figure 10.10, with 15 chosen training positions (marked
as red squares) and 15 chosen test positions (marked as green dots). The distance

Table Table Table Table

Table Air conditioner

Figure 10.10 Layout of the computer laboratory: training positions are marked as
red squares and testing positions are marked as green dots
360 Applications of machine learning in wireless communications

between two adjacent training positions is 1.8 m. The single access point is put close
to the center of the room. We collect bimodal data from 1,000 packet receptions for
each training position, and from 25 packet receptions for each test position. The deep
network used for this scenario is configured as {K1 = 150, K2 = 100, K3 = 50}. Also,
the ratio ρ for the bimodal data is set as 0.5.
Corridor: This is a 2.4 × 24 m2 corridor, as shown in Figure 10.11. In this scenario,
the AP is placed at one end of the corridor, and there are plenty of LOS paths. Ten
training positions (red squares) and ten test positions (green dots) are arranged along
a straight line. The distance between two adjacent training positions is also 1.8 m. We
also collect bimodal data from 1,000 packets for each training position and from 25
packets for each test position. The deep network used for this scenario is configured
as {K1 = 150, K2 = 100, K3 = 50}. Also, the ratio ρ for the bimodal data is set as 0.1.

10.5.2 Accuracy of location estimation

Tables 10.1 and 10.2 present the mean and STD of localization errors, and the exe-
cution time of the four schemes for the two scenarios, respectively. In the laboratory

312 311 308

313
AP 8m

314 318 310 309

24 m

Figure 10.11 Layout of the corridor: training positions are marked as red squares
and testing positions are marked as green dots

Table 10.1 Mean/STD error and execution time of the laboratory experiment

Algorithm Mean error (m) Std. dev. (m) Mean execution time (s)

BiLoc 1.5743 0.8312 0.6653

DeepFi 2.0411 1.3804 0.3340
FIFS 2.7151 1.0805 0.2918
Horus 3.0537 1.0623 0.2849

Table 10.2 Mean/STD errors and execution time of the corridor experiment

Algorithm Mean error (m) Std. dev. (m) Mean execution time (s)

BiLoc 2.1501 1.5420 0.5440

DeepFi 2.8953 2.5665 0.3707
FIFS 4.4296 3.4256 0.2535
Horus 4.8000 3.5242 0.2505
Deep learning for indoor localization based on bimodal CSI data 361

environment, BiLoc achieves a mean error of 1.5743 m and an STD error of 0.8312 m
across the 15 test points. In the corridor experiment, because only one access point is
used for this larger space, BiLoc achieves a mean error of 2.1501 m and an STD error
of 1.5420 m across the ten test points. BiLoc outperforms the other three benchmark
schemes with the smallest mean error, as well as with the smallest STD error, i.e.,
being the most stable scheme in both scenarios. We also compare the online test time
of all the schemes. Due to the use of bimodal data and the deep network, the mean
executing time of BiLoc is the highest among the four schemes. However, the mean
execution time is 0.6653 s for the laboratory case and 0.5440 s for the corridor case,
which are sufficient for most indoor localization applications.
Figure 10.12 presents the CDF of distance errors of the four schemes in the
laboratory environment. In this complex propagation environment, BiLoc has 100%
of the test positions with an error under 2.8 m, while DeepFi, FIFS, and Horus
have about 72%, 52%, and 45% of the test positions with an error under 2.8 m,
respectively. For a much smaller error of 1.5 m, the percentage of test positions having
a smaller error are 60%, 45%, 15%, and 5% for BiLoc, DeepFi, FIFS, and Horus,
respectively. BiLoc achieves the highest precision among the four schemes, due to
the use of bimodal CSI data (i.e., average amplitudes and estimated AOAs). In fact,
when the amplitude of a signal is strongly influenced in the laboratory environment,
the estimated AOA can be utilized to mitigate this effect by BiLoc. However, the other
schemes-based solely on CSI or RSS amplitudes will be affected.
Figure 10.13 presents the CDF of distance errors of the four schemes for the
corridor scenario. Only one access point is used at one end for this 24 m long corridor,
making it hard to estimate the location of the mobile device. For BiLoc, more than
90% of the test positions have an error under 4 m, while DeepFi, FIFS, and Horus have
about 70%, 60%, and 50% of the test positions with an error under 4 m, respectively.
For a tighter 2 m error threshold, BiLoc has 60% of the test positions with an error
below this threshold, while it is 40% for the other three schemes. For the corridor

0.8

0.6
CDF

0.4
BiLoc
DeepFi
0.2
FIFS
Horus
0
0 1 2 3 4 5
Distance error (m)

Figure 10.12 CDF of localization errors in 5 GHz for the laboratory experiment
362 Applications of machine learning in wireless communications

0.8

0.6
CDF

0.4
BiLoc
DeepFi
0.2 FIFS
Horus
0
0 2 4 6 8 10 12
Distance error (m)

Figure 10.13 CDF of localization errors in 5 GHz for the corridor experiment

scenario, BiLoc mainly utilizes the average amplitudes of CSI data, because the
estimated AOAs are similar for all the training/test positions (recall that they are
aligned along a straight line with the access point at one end). This is a challenging
scenario for differentiating different test points and the BiLoc mean error is 0.5758 m
higher than that of the laboratory scenario.

10.5.3 2.4 versus 5 GHz

We also compare the 2.4 and 5 GHz channels with the BiLoc scheme. For a fair
comparison, we conduct the experiments at night, because the 2.4 GHz band is much
more crowded than the 5 GHz band during the day.
Figure 10.14 presents the CDF of localization errors in the 2.4 and 5 GHz band
in the laboratory environment, where both average amplitudes and estimated AOAs
are effectively used by BiLoc for indoor localization. We can see that for BiLoc,
about 70% of the test positions have an error under 2 m in 5 GHz, while 50% of
the test positions have an error under 2 m in 2.4 GHz. In addition, the maximum
errors in 2.4 and 5 GHz are 6.4 and 2.8 m, respectively. Therefore, the proposed
BiLoc scheme achieves much better performance in 5 than 2.4 GHz. In fact, the
phase difference between two antennas in 2.4 GHz exhibits great variations, which
lead to lower localization accuracy. This experiment also validates our Hypothesis 2.

10.5.4 Impact of parameter ρ

Recall that the parameter ρ is used to trade-off the impacts of average amplitudes
and estimated AOAs in location estimation as in (10.17). We examine the impact
of ρ on localization accuracy under the two environments. With BiLoc, we use
bimodal data for online testing, and ρ directly influences the likelihood probability
Pr (v1 , v2 |li ) (10.17), which in turn influences the localization accuracy.
Deep learning for indoor localization based on bimodal CSI data 363

0.8

CDF 0.6

0.4

0.2
BiLoc in 5 GHz
BiLoc in 2.4 GHz
0
0 1 2 3 4 5 6 7
Distance error (m)

Figure 10.14 CDF of localization errors in 5 and 2.4 GHz for the laboratory
experiment

4.5
4
3.5
Distance error (m)

3
2.5
2
1.5
1
Laboratory experiment
0.5 Corridor experiment
0
0 0.2 0.4 0.6 0.8 1
Parameter, ρ

Figure 10.15 Mean localization errors versus parameter, ρ, for the laboratory and
corridor experiments

Figure 10.15 presents the mean localization errors for increasing ρ for the lab-
oratory and corridor experiments. In the laboratory experiment, when ρ is increased
from 0 to 0.3, the mean error decreases from 2.6 to 1.5 m. Furthermore, the mean
error remains around 1.5 m for ρ ∈ [0.3, 0.7], and then increases from 1.5 to 2 m
when ρ is increased from 0.6 to 1. Therefore, BiLoc achieves its minimum mean
error for ρ ∈ [0.3, 0.7], indicating that both average amplitudes and estimated AOAs
are useful for accurate location estimation. Moreover, BiLoc has higher localization
accuracy with the mean error of 1.5 m, compared with individual modality such as
the estimated AOAs with that of 2.6 m or the average amplitudes with that of 2.0 m.
364 Applications of machine learning in wireless communications

In the corridor experiment, we can see that the mean error remains around 2.1 m
when ρ is increased from 0 to 0.1. When ρ is further increased from 0.1 to 1, the
mean error keeps on increasing from 2.1 to about 4.3 m. Clearly, in the corridor
experiment, the estimated AOAs provide similar characteristics for deep learning and
are not useful for distinguishing the positions. Therefore, BiLoc should mainly use
the average amplitudes of CSI data for better accuracy. These experiments provide
some useful guidelines on setting the ρ value for different indoor environments.

10.6 Future directions and challenges

10.6.1 New deep-learning methods for indoor localization

This chapter has discussed three deep-learning technologies including autoencoder,
CNN, LSTM for fingerprinting-based indoor localization. With the rapid growth in the
AI field, new deep-learning approaches are proposed for mainly handling computer
vision problems, such as robust object recognition and detection, data generation, as
well as the Go game. For example, generative adversarial network (GAN) can be used
for generating new data samples; deep reinforcement learning has been leveraged for
AlphaGo; deep Gaussian process can be utilized for improving the robustness of object
detection. In fact, these new deep-learning methods can be also used for solving basic
indoor localization problems such as radio map constructions, environment change,
and devices calibration. For example, deep reinforcement learning [51] can be used
for improving localization performance and reduce cost. Moreover, Bayesian deep
learning such as deep Gaussian process [52,53] has high robustness for environment
noise, which can be exploited for radio map construction, and mitigating environment
changes and devices calibration. Moreover, GAN can be incorporated for building
radio map and increasing the number of training data samples. In addition, com-
pressed deep learning [54] by using pruning and quantization can be considered for
resource limited mobile devices. Thus, we can implement deep-learning models on
smartphones in addition to servers for indoor localization.

10.6.2 Sensor fusion for indoor localization using deep learning

In this chapter, we have proposed bimodal CSI data for indoor localization. In fact,
multiple sensor data sources can be fused for improving indoor localization perfor-
mance. Traditionally, sequence models such as Kalman filter, particle filter, hidden
Markov model, and conditional random field can fuse Wi-Fi and inertial sensor data
on smartphones for indoor localization, which requires for obtaining the sequence
data from continuing smartphone movement. Deep-learning techniques can improve
the performance of indoor localization using multimodal sequence data. For exam-
ple, LSTM method can be leveraged for indoor localization using sequence RSS or
CSI data, which also fuse multimodal data for improving the localization accuracy.
Considering Wi-Fi and magnetic sensor data from smartphone, we can integrate them
into a large data matrix as input to LSTM for indoor localization. In fact, Wi-Fi and
magnetic sensor data are complementary to each other. For example, because of lower
Deep learning for indoor localization based on bimodal CSI data 365

resolution of Wi-Fi signals, only using Wi-Fi RSS values cannot obtain better per-
formance at close locations, while magnetic sensor data at such positions is greatly
different. LSTM can effectively fuse them for indoor localization [43]. In addition,
an integrated CNN and LSTM model can be used for Wi-Fi RSS or CSI images data,
which can be easily created from different access points or different subcarriers. In
fact, the LSTM model can be combined with other deep-learning models such as
autoencoder, GAN, deep reinforcement learning, Bayesian model for different local-
ization problems such as radio map construction, device calibration, and environment
change. For sensor data fusion for indoor localization, different sensor data sources
should be normalized and aligned [23].

10.6.3 Secure indoor localization using deep learning

For wireless-fingerprinting-based indoor localization, security becomes increasingly
important, where wireless signals are susceptible to eavesdropping, distributed denial-
of-service (DDoS) attacks, and bad data injection [55]. Specially, for crowd-sourcing-
based indoor localization, fingerprints are from different devices at different times,
which greatly exposes the security problem. For attacker models, there are three
general scenarios for RSS fingerprinting-based localization [56]. First, the attacker
does not know the true RSS fingerprints and injects fake RSS data at random. Second,
the attacker knows legitimate RSS fingerprints and add noise to them. Third, the
attacker can change the mapping between RSS fingerprints and positions. For defense
models, they can consider the temporal correlation and spatial correlation within
RSS Traces against different attackers. In fact, deep learning can study the feature
of Localization signals to address the above security problems. Deep learning can
consider different data features from multiple paths of wireless signals to classify
eavesdropping, DDoS attack, or bad data injection for fingerprinting-based indoor
localization.
On the other hand, deep-learning security is also an important problem, which
mainly focuses on how to recognize adversarial data and clean RSS data. Deep learn-
ing will have bad performance with adversarial data, which can be obtained by adding
small noise into clear RSS data. Thus, adversarial data should be recognized before
implementing indoor localization systems based on deep learning, thus guaranteeing
good localization performance. In addition, privacy persevering deep learning can
be used for indoor localization problems, which can protect user location privacy
information.

10.7 Conclusions
In this chapter, we proposed a bimodal deep-learning system for fingerprinting-based
indoor localization with 5 GHz commodity Wi-Fi NICs. First, the state-of-the-art
deep-learning techniques including deep autoencoder network, CNN, and LSTM
were introduced. We then extracted and calibrated CSI data to obtain bimodal CSI
data, including average amplitudes and estimated AOAs, which were used in both
366 Applications of machine learning in wireless communications

the off-line and online stages. The proposed scheme was validated with extensive
experiments. We concluded this chapter with a discussion of future directions and
challenges for indoor localization problems using deep learning.

Acknowledgments
This work is supported in part by the US NSF under Grants ACI-1642133 and CNS-
1702957, and by the Wireless Engineering Research and Education Center (WEREC)
at Auburn University.

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Chapter 11
Reinforcement-learning-based wireless
resource allocation
Rui Wang1

In wireless systems, radio resource management (RRM) is a necessary approach to

improve the transmission efficiency. For example, the base stations (BSs) of cellular
networks can optimize the selection of downlink receiving users in each frame and
the transmission power for them according to channel state information (CSI), such
that the total throughput is maximized. The RRM of various kinds are usually rep-
resented by mathematical optimization problems, with an objective to be optimized
(e.g., throughput or delay) and some constraints on limited resources (e.g., transmis-
sion time, frequency or power). The mathematical modeling procedure from the needs
of resource allocation to optimization problems is usually referred to as “problem
formulation.”
In this chapter, we shall focus on the formulation of RRM via Markov decision
process (MDP). Convex optimization has been widely used in the RRM within a
short-time duration, where the wireless channel is assumed to be quasi-static. These
problems are usually referred to as deterministic optimization problems. On the other
hand, MDP is an elegant and powerful tool to handle the resource optimization of wire-
less systems in a longer timescale, where the random transitions of system and channel
status are considered. These problems are usually referred to as stochastic optimization
problems. Particularly, MDP is suitable for the joint optimization between physical
and media-access control (MAC) layers. Based on MDP, reinforcement learning is a
practical method to address the optimization without a priori knowledge of system
statistics. In this chapter, we shall first introduce some basics on stochastic approxi-
mation, which serves as one basis of reinforcement learning, and then demonstrate the
MDP formulations of RRM via some case studies, which require the knowledge of sys-
tem statistics. Finally, some approaches of reinforcement learning (e.g., Q-learning)
are introduced to address the practical issue of unknown system statistics.

11.1 Basics of stochastic approximation

Stochastic approximation is a general iterative method to solve some stochastic fixed-
point problems or optimization problems without the knowledge of statistics in the

1
Department of Electrical and Electronic Engineering, The Southern University of Science and Technology,
China
372 Applications of machine learning in wireless communications

problem. Mathematically, these kinds of problems are not well defined. They may refer
to some systems with unknown random behavior. For example, the wireless trans-
mitter wants to make sure that the average receiving signal-to-interference-plus-ratio
(SINR) is above certain quality level; however, the interference level at the receiver
is hard to predict without its statistics. Clearly, this problem cannot be solved unless
more information can be collected. In the transmission protocol design, the receiver
can estimate the receiving interference level and report it to the transmitter periodi-
cally, so that the transmitter can adjust its power and guarantee an acceptable average
SINR level. Hence, the procedure of problem-solving includes not only calculation
but also system observation. Stochastic approximation is such an online learning and
adapting procedure, which collects the information from each observation and finally
converges to the solution.

11.1.1 Iterative algorithm

In this section, we use one example of deterministic fixed-point problem to demon-
strate the structure of iterative solution, which is widely used in solving the problems
without close-form solutions. The exemplary fixed-point problem is provided below.

Problem 11.1 (Deterministic fixed-point problem). Find an x such that

f (x) = 0, (11.1)

where f (x) is a monotonically increasing function.

Providing the expression of f (x), this problem may be solved analytically. For
example, x = log10 a when f (x) = 10 x − a and a is a positive constant. Nevertheless,
the following iterative algorithm is useful when the explicit expression of x cannot be
derived.

Iterative algorithm for Problem 11.1

Let n be the index of iteration and xn be the median value of x in the nth iteration,
the solution of (11.1) can be achieved as follows:
1. Initialize the iteration index n as n = 0, and the value of x as x0 .
2. Update the median value of x as
xn+1 = xn − γn+1 f (xn ), (11.2)
where γn is the step size of iteration. Let n = n + 1.
3. Let ε be a threshold for terminating the iteration. The algorithm stops if
|xn+1 − xn | < ε or goes to Step 2 otherwise.

Strictly speaking, the solution for Problem 11.1 may not be unique; the above
algorithm is to find one feasible solution if it exists. There are a number of choices
Reinforcement-learning-based wireless resource allocation 373

Controller
xn+1 = xn – γn+1 f(xn)

System
f(x)
Input: xn Output: f(xn)

Figure 11.1 Block diagram for the iterative algorithm of Problem 11.1

on the step size {γn |∀n = 1, 2, . . .}. For the Newton’s method (also known as the
Newton–Raphson method), the step size is
1
γn = ,
f (xn )

where f (xn ) is the first-order derivative of f (x) at x = xn . For the case that f (xn )
cannot be obtained, one more general choice of step size is the harmonic series
1
γn = .
n
An intuitive explanation on using the harmonic series as iteration step size is
provided below:
● Note that this series is monotonically decreasing. When xn is close to the solution,
smaller step
size is better for fine adjustment.
● Note that +∞ n=1 (1/n) = +∞, the incremental update of −γn+1 f (x) is not neg-
ligible as long as f (x) = 0. Hence the algorithm could drive xn to the solution
of f (x) = 0.
A block diagram of the iterative algorithm is illustrated in Figure 11.1, where f (x)
may be an observation of certain system with input x. A controller, with the objective
of f (x) = 0, collects the observation of f (x) and update the value of x.

11.1.2 Stochastic fixed-point problem

The problem introduced in the previous section assumes that the function f (x) can
be accurately estimated when an input value of x is provided. However, this may not
be the case in some applications. For example, suppose that Y is a random variable
whose distribution is unknown, and we want to find an appropriate x to satisfy the
following equation:
E[x − Y ] = 0. (11.3)
374 Applications of machine learning in wireless communications

This is not a well-defined problem if there is no further information of Y , and its

solution can only be obtained by learning its statistics. This section will provide a
general iterative solution for such stochastic fixed-point problem, which is referred
to as stochastic approximation.
Consider a general expression of stochastic fixed-point problem. Let f (x, Y ) be
a function of variable x and random variable Y . f (x, Y ) can be treated as the output
of a system, which depends on the input parameter x and random internal statue Y .
In some applications, people may be interested in the expected output of the system.
Hence, it is defined that

f (x) = E f (x, Y ) = f (x, y)p(y)dy, (11.4)

where p(y) is the PDF of random variable Y . The fixed-point problem to be solved
becomes.

Problem 11.2 (stochastic fixed-point problem). Find x such that

f (x) = 0, (11.5)
where f (x) and its realization f (x, Y ) satisfy the following conditions:
● Suppose f (θ ) = 0. Given x, there exists a finite positive constant δ such that
f (x) ≤ −δ for x < θ, f (x) ≥ δ for x > θ. (11.6)
● There exists a finite constant c such that
Pr[| f (x, Y )| ≤ c] = 1. (11.7)

The condition (11.6) guarantees that the value of f (x) can be used to update
the variable x: x should be decreased when f (x) is positive, and vice versa. The
condition (11.7) assures that each realization f (x, Y ) can be adopted to evaluate its
expectation f (x). According to the method of stochastic approximation, the solution
of Problem 11.2 is described below.

Iterative algorithm for Problem 11.2

Let xn be the nth input value of x, and Yn be the nth realization of random variable
Y . The solution of (11.5) can be achieved as follows:
1. Initialize the iteration index n as n = 0, and the value of x as x0 .
2. Update the input value of x as
xn+1 = xn − γn+1 f (xn , Yn ), (11.8)
where γn is the step size of iteration. Let n = n + 1.
3. Let ε be a threshold for terminating the iteration. The algorithm stops if
|xn+1 − xn | < ε or goes to Step 2 otherwise.
Reinforcement-learning-based wireless resource allocation 375

The convergence of the above algorithm is summarized below.

Theorem 11.1. Let en = E(xn − θ)2 . If γn = 1/n, the conditions of (11.6) and (11.7)
are satisfied, then
lim en = 0.
n→+∞

Please refer to the Theorem 11.1 of [1] for the rigorous proof. Some insights on
the convergence property are provided here. In fact, f (xn , Yn ) can be treated as one
estimation of f (xn ) with estimation error Zn . Thus:
f (xn , Yn ) = f (xn ) + Zn , ∀n. (11.9)
Since E f (xn , Yn ) = f (xn ), we know that E[Zn ] = 0. From the iterative update equation
(11.8), it can be derived that
x1 = x0 − γ1 f (x0 , Y0 ) = x0 − [ f (x0 ) + Z0 ]
1
x2 = x1 − γ2 f (x1 , Y1 ) = x0 − [ f (x0 ) + Z0 ] − [ f (x1 ) + Z1 ]
2
···
n−1
1
n−1
1
xn = xn−1 − γn f (xn−1 , Yn−1 ) = x0 − f (xi ) − Zi
i=0
i+1 i=0
i+1

k−1
1 1
n−1 1
n−1
= x0 − [Zi + f (xi )] − f (xi ) − Zi . (11.10)
i=0
i+1 i=k
i+1 i=k
i+1

xk
Hence, the convergence of {xn } is discussed as follows:
n−1
1. The last term of (11.10), i=k (1/(i + 1))Zi , can be treated as the noise of
iteration. Because without it, the iteration becomes:

n−1
1
xn = xk − f (xi ), (11.11)
i=k
i+1

which is the solution algorithm for the deterministic fixed

point f (x) = 0. The
n−1
expectation of the noise is zero, i.e., E i=k (1/(i + 1))Zi = 0.
n−1
2. The variance of noise i=k (1/(i + 1))Zi are analyzed as follows:
n−1 n−1
1 1 n−1
1
Var Zi = Var(Zi ) ≤ σ 2, (11.12)
i=k
i + 1 i=k
(i + 1) 2
i=k
(i + 1) 2 Z

where σZ2 = max Var(Zi ).

∞ i
3. i=k (1/(i + 2 2
∞ )σZ is finite.2Moreover,
1) for arbitrary ε > 0, there exists an integer
K such that i=K (1/(i + 1) )σZ2 < ε.
376 Applications of machine learning in wireless communications

4. Let n → ∞ on both sides of (11.10), we have

∞
1 1
∞
x∞ = xk − f (xi ) − Zi . (11.13)
i=k
i+1 i=k
i+1

Note that ∞ i=k (1/(i + 1))Zi is generally finite accordingto the above discus-
sion. If f (xi ) is always greater than certain positive value, ∞ i=k (1/(i + 1)) f (xi )
will drive x∞ to negative infinity, which will lead to negative f (xi ). Thus, the
convergence can be intuitively understood according to the above contradiction.

11.2 Markov decision process: basic theory and applications

In wireless systems, the transmission time and spectrum are usually organized as
frames. For example, every 1 ms in the long-term evolution (LTE) system is orga-
nized as one subframe, and every ten subframes constitutes a frame. The wireless
channel is usually assumed to be quasi-static within one subframe or even one frame.
In LTE systems, the CSI can be estimated in every subframe, which is used to decode
the current subframe or determine the transmission parameters of the following sub-
frames. The transmission resource allocation in one subframe can be formulated as
various optimization problems. For example, one possible problem formulation is to
jointly optimize the uplink or downlink transmission time and power among multi-
ple mobile users, such that the overall throughput of the subframe is maximized. In
fact, many resource optimization problems share the similar structure, which usually
consists of the following:
● System state: A number of parameters specifying the system status, which can
be estimated and notified to the scheduler, e.g., coefficients of large-scale and
small-scale fading.
● Scheduling action: A number of transmission or receiving parameters can be
adjusted, e.g., transmission time and power.
● Objective: A function measuring the utility or cost within a time slot, where the
system state is assumed to be quasi-static. For example, the overall throughput
in an LTE subframe. The objective is usually a function of system state and
scheduling action.
● Constraints: A number of limitations on the transmission resources within the
aforementioned time slot, e.g., the total resource elements (symbols) for data
transmission in an LTE subframe. There might be a number of constraints on
different types of transmission resource. Each constraint is usually a function of
scheduling action; some of them may also depend on system state.
As a result, the resource allocation problems can be generally formulated by the
following optimization problem:
max or min Objective (state, action)
Action Action

subject to Constraints(state, action) ∈ System affordable region.

Reinforcement-learning-based wireless resource allocation 377

These type of problems aim at finding the optimal values of some transmission or
receiving parameters for each time slot (i.e., action). It implies that there is a sched-
uler, who observes the system state in each time slot, solves the above optimization
problem, and uses the solution in transmission. In many of such optimization prob-
lems, the optimization action in one time slot does not affect that of the followings.
Hence, we shall refer to this type of problems as the “single stage” optimization in the
remaining of this chapter. The following is an example of single stage optimization
formulation.

Example 11.1 (Multi-carrier power allocation). Suppose that there is one point-
to-point OFDM link with NF subcarriers, and their channel gains in one certain
time slot are denoted by {hi |i = 1, 2, . . . , NF }. Let pi (i = 1, 2, . . . , NF ) be the
transmission power on the ith subcarrier. One typical power allocation problem
is to determine the transmission power on each subcarrier { pi |i = 1, 2, . . . , NF }
such that the overall throughput is maximized, which can be formulated as follows:
● System state: {hi |i = 1, 2, . . . , NF }.
● Action: { pi
|i = 1, 2, . . . , NF }.
● Objective: Ni=1 F
log 1 + ((pi
hi
2 )/σz2 ) .
NF 2
● Constraint: i=1 pi ≤ P, where P is the peak transmission power.
Hence, the overall optimization problem can be written as

NF
pi
hi
2
max log2 1 +
{pi |i=1,2,...,NF }
i=1
σz2

NF
subject to pi ≤ P,
i=1

where σz2 is the power of noise. This problem can be solved by the well-known
water-filling algorithm. Note that this is the single stage optimization, since there
is no connection between the optimization in different time slots.

However, the above single stage formulation is not powerful enough to address
all the wireless-resource-allocation problems, especially when the scope of opti-
mization is extended to the MAC layer and larger timescale. From the MAC layer
point of view, the BS maintains one queue for each active downlink mobile user.
If the BS schedules more transmission resource to one user in certain time slot,
the traffic load for this user in the following time slots can be relieved. Hence, the
scheduling action in one time slot can affect that of the following ones, and a joint
optimization along multiple time slots becomes necessary. The difficulty of such joint
optimization is that the wireless channel is time varying. Hence, the scheduler can-
not predict the channel of the following time slots and, of course, cannot determine
378 Applications of machine learning in wireless communications

their scheduling parameters in advance. We shall refer to this type of problems as the
“multistage” optimization in the remaining of this chapter. Due to the uncertainty on
future system state, its differences from the single-stage optimization problem are as
follows:
● Instead of calculating the values for scheduling action, we should provide a map-
ping from arbitrarily possible system state to the corresponding scheduling action,
so that the system can work properly in all possible situations. This mapping is
called policy, thus:
Policy : System State → Scheduling Action. (11.14)
● The expectation should be taken on the objective and constraints (if any) since
these functions depend on random system state.
In this section, the MDP is introduced to formulate and solve this kind of multi-
stage optimization problem. In order to bring up the basic principle without struggling
with the mathematical details, this section is only about the discrete-time MDP with
finite state and action spaces, and some mathematical proof is neglected. For the read-
ers who are interested in a overwhelm and rigorous discussion on MDP optimization
theory, please refer to [2,3].

11.2.1 Basic components of MDP

As illustrated in Figure 11.2, there are three basic components in an MDP, namely,
system state, control or scheduling action, and cost function. Sometimes, people may
want to maximize some utility function, this is equivalent to minimize the inverse
of utility function, which can be treated as cost function. System state, denoted as
s, consists of the set of parameters, which uniquely specify the system at any time
instance. The set of all the possible values of system state is called the state space,
denoted as S . In this chapter, we consider the case that the cardinality of state space

Transition kernel: Pr(sn+1 | sn, an)

System
State: sn
Cost: g(sn, an)

Action: an = Ωn(sn)

Figure 11.2 Block diagram for Markov decision process

Reinforcement-learning-based wireless resource allocation 379

S , denoted as |S |, is finite, or it can be compressed to finite. Given the state s, there

is a controller, which is able to manipulate the system by adjusting a set of control
parameters a. This set of control parameters is called the control action, and the set of
all the possible choices of control actions are named as action space, denoted as A . In
Example 11.1 (it can be treated as a trivial case of MDP), the system state and control
action are the CSI and the transmission powers of all the subcarriers, respectively.
Thus,

s = {hi |i = 1, 2, . . . , NF }

and

a = {pi |i = 1, 2, . . . , NF }.

We focus on discrete-time MDP, where subscript of notations is used to indicate

the stage index. For example, at the tth stage, the system state and control action are
denoted as st and at , respectively. The control action at is uniquely determined by the
system state st , the mapping from system state to control action is named as control
policy. Let t : S → A be the control policy at the tth stage, i.e.:

t (st ) = at , ∀st ∈ S . (11.15)

In MDP, the state of a system evolves with time in a Markovian way: providing
the current system state and control action, the distribution of next system state is
independent of other historical states or actions. In other words, the evolving of system
state is a Markov chain, providing the control policy at each stage. Given the current
(say the tth stage) system state st and control action at , the distribution of next system
state, Pr(st+1 |st , at ), is called the state transition probability or transition kernel.
The expense of control action is measured by the cost function. The cost of the
system at the tth stage is a function of st and at , which is denoted as gt (st , at ). In fact,
gt can be a random variable given st and at , i.e., gt (st , at , ξt ) where ξt for different
t are independent variables. To simplify the elaboration, we focus on the form of
gt (st , at ) in the following discussion. Note that the cost function can be homogeneous
or heterogeneous along the time line. Particularly, gt can be different with respect to
stage index t for an MDP with finite number of stages. However, when it is extended
to the optimization over infinite number of stages, gt should usually be homogeneous
and subscript of stage index can be removed.
What are optimized in an MDP are not some parameters but a policy, which
maps from the system state to control (scheduling) action. Thus, the solution of an
MDP is a “function” rather than the values for some parameters. In Example 11.1,
the water-filling algorithm can be used to figure out the values of the transmission
powers on all the subcarriers. This is a physical-layer point of view. The following
example shows that the if the scope of resource allocation is extended to MAC layer,
the value optimization will transfer to policy optimization, which can be formulated
as an MDP.
380 Applications of machine learning in wireless communications

Example 11.2 (Multi-carrier power allocation: view from MAC-layer). Similar

to Example 11.1, a point-to-point OFDM link with NF subcarriers is considered. In
MAC layer, a transmission queue is maintained, which accepts packets from upper
layer and delivers them via physical layer. Suppose that each packet consists of
B information bits, and the number of arrival packets in each frame follows the
Poisson distribution with expectation λ, i.e.:

λn e−λ
Pr [The number of arrival packets in one frame = n] = . (11.16)
n!

The packet departure in the MAC layer is determined by the physical layer trans-
mission. Let q(t) be the number of packets waiting to be transmitted in the tth
frame, the queue dynamics can be represented by

q(t + 1) = max{0, q(t) − d(t)} + c(t), (11.17)

where d(t) and c(t) are the numbers of departure and arrival packets in the tth
frame.
In physical layer, let {hi (t)|i = 1, 2, . . . , NF } be the CSI of all the subcarriers in
the tth frame, and { pi (t)|i = 1, 2, . . . , NF } be the corresponding power allocation.
The number of packets can be delivered in the tth frame is

NF log 1 + ((pi (t)

hi (t)
2 )/σ 2 )
2 z
d(t) = i=1
. (11.18)
B

Clearly, larger transmission power will lead to larger departure rate of the
transmission queue. However, some systems may have the following concerns
on the average power consumption, which is particularly for battery-powered
device:

1
T NF
lim E pi (t) ≤ P, (11.19)
T →+∞ T t=1 i=1

where P is the average power constraint.

When the queue in the MAC layer is considered, the maximum throughput
used in the Example 11.1 may not be suitable as the scheduling objective. One
reasonable objective is the minimum average delay, which measures the average
time duration for one packet from arrival to departure. Let:

1
T
Q = lim E q(t) (11.20)
T →+∞ T t=1
Reinforcement-learning-based wireless resource allocation 381

be the average queue length at the transmitter, the average delay W is given below
according to the Little’s Law [4]:

1 q(t)
T
Q
W = = lim E . (11.21)
λ T →+∞ T t=1 λ

Therefore, one possible problem formulation is to minimize the average packet

delay while satisfying the average power constraint. Thus,

min W
{pi (t)|∀i,t}

subject to (11.19).

In Example 11.1, the scheduling in different frames is independent. In other

words, one does not need to worry about the impact of current frame scheduling on
the future frames. However, when the average transmission delay W is the objective
with the average power constraint, the scheduling in the current frame will affect
that of the future frames. For example, the current frame may be scheduled with
a power level greater than P, which consumes the power budget of the following
frames. Hence, it becomes meaningless to consider the resource optimization in
one single frame (as Example 11.1), and the scope of optimization is the whole
time line. As a result, what should be optimized is a mapping from CSI and queue
length (also called queue state information or QSI) to the power allocation. Thus
it is a “function,” rather than some “variables.”

Three forms of MDP formulation will be discussed in the following: first of all, we
introduce the finite-horizon MDP, where the number of stages for joint optimization
is finite. Then, we move to the infinite-horizon MDP, where two cost functions are
considered: namely, average cost and discounted cost.

11.2.2 Finite-horizon MDP

In this section, we focus on the optimization along a fixed number of stages, say T
stages. The overall cost function, denoted as G, can be written as
T

G {n |n = 1, 2, . . . , T } = E gt (st , at ) , (11.22)
t=1

where at = t (st ). The expectation in the above equation is with respect to the ran-
domness of the system state at the first stage and the state transition given the control
action. Note that with the expectation on random system state, the overall cost func-
tion G depends on the control policies used in all the stages. With the objective of
minimizing G, the problem of finite-horizon MDP is described below.
382 Applications of machine learning in wireless communications

Problem 11.3 (Finite-horizon MDP). Find the optimal control polices for each
stage, denoted as {∗t |t = 1, 2, . . . , T }, such that the overall cost G is minimized, i.e.:

{∗t |t = 1, 2, . . . , T } = arg min G {t |t = 1, 2, . . . , T } . (11.23)
{t |t=1,2,...,T }

It is worth to highlight that in finite-horizon MDP, the optimal scheduling policies

in different stages are usually different. For example, the policy design of the first
stage should jointly consider the cost of the current stage and the potential cost of the
following N − 1 stages, whereas the policy design of the last stage only needs to care
the cost of the current stage.
In order to elaborate the solution structure of Problem 11.3, we first define the
following cost-to-go function:
T

Vt (st ) = min E gk (sk , ak ) , ∀t, st , (11.24)
{k |k=t,t+1,...,T }
k=t
which is the average cost from the tth stage to the last one given the system state in
the tth time slot st . The cost-to-go function Vt is usually named as value function.
It is straightforward to see that they satisfy the following iterative expressions:
⎡ ⎤

Vt (st ) = min⎣gt (st , at ) + Pr(st+1 |st , at )Vt+1 (st+1 )⎦ , ∀t, st . (11.25)
at
st+1

Equation (11.25) is usually referred to as the Bellman equation. It provides impor-

tant insights on the solution of Problem 11.3. Given the system state at the tth stage
st , the optimal control action minimizing the right-hand-side of (11.25), denoted as
∗t (st ), is obviously the optimal control action for system state st at the tth stage, i.e.:
⎡ ⎤

∗t (st ) = arg min ⎣gt (st , at ) + Pr(st+1 |st , at )Vt+1 (st+1 )⎦ , ∀t, st . (11.26)
at
st+1

Hence, in order to obtain the optimal control policy at the tth stage, it is neces-
sary to first figure out the value function Vt+1 for all possible next state. It implies
that before calculating the optimal policy, a backward recursion for evaluating VT ,
VT −1 , …, V1 sequentially is required, which is usually referred to as value iteration
(VI). The VI algorithm for finite-horizon MDP is elaborated below.

VI algorithm for finite-horizon MDP

The value functions for finite-horizon MDP, as defined in (11.24), can be evaluated
by the following steps:
1. Calculate the value function VT for the last stage by
VT (sT ) = min gT (sT , aT ), ∀sT . (11.27)
aT

2. For t = T − 1, T − 2, . . . , 1, calculate the value function Vt according to

(11.25) sequentially.
Reinforcement-learning-based wireless resource allocation 383

Note that the value functions are calculated from the last stage to the first one.
This is because of iterative structure as depicted in the Bellman equation (11.25). As
a summary, the procedure to obtain the optimal control policy for the finite-horizon
MDP can be described below.

● Off-line VI: Before running the system, the controller should evaluate the value
functions for all the possible system states and all the stages. Their values can be
stored in a table.
● Online scheduling: When the system is running, the controller should identify
the system state, solve the corresponding Bellman equation, and apply the optimal
action.

Hence, the solution raises both computation and memory requirements to the con-
troller, whose complexities are proportional to the size of state space |S | and
the number of stages T . In the following, we shall demonstrate the application of
finite-horizon MDP via the multi-carrier power allocation problem.

11.2.2.1 Case study: multi-carrier power allocation via finite-horizon

MDP
Suppose that there is one OFDM transmitter which wants to deliver a file of B bits
to the receiver within T frames. The number of subcarriers is NF . The transmitter
is a battery-powered device, and it tries to save the transmission energy as much as
possible by exploiting the channel temporal diversity in the T frames. One possible
formulation for transmission scheduling at the transmitter is provided below:

● System state: In the tth frame (t = 1, 2, . . . , T ), the system state st is uniquely

specified by the CSI of all the subcarriers {hi (t)|i = 1, 2, . . . , NF } and the number
of remaining bits at the transmitter q(t), which is usually called QSI. Thus:

st = {hi (t)|i = 1, 2, . . . , NF }, q(t) . (11.28)

● Control policy: The control action in the tth frame (t = 1, 2, . . . , T ) is the power
allocation on all the subcarriers, i.e.,

at = {pi (t)|i = 1, 2, . . . , NF }.

Then the control policy in the tth frame, denoted as t , can be written as
t (st ) = at , ∀t, st . (11.29)
● Transition kernel: The block fading channel model is considered, and the CSIs
in different frames are i.i.d. distributed. Therefore, the transition kernel can be
rewritten as

Pr(st+1 |st , at ) = Pr {hi (t + 1)|i = 1, 2, . . . , NF } Pr q(t + 1)|st , at ,
(11.30)
384 Applications of machine learning in wireless communications

where

q(t + 1) = max{0, q(t) − d(t)}, (11.31)

and

pi (t)
hi (t)
2
NF
d(t) = log2 1 + (11.32)
i=1
σz2

is the number of bits transmitted in the tth frame. Hence, given st and at , q(t + 1)
is uniquely determined.
● Cost: In the tth frame (t = 1, 2, . . . , T ), the cost of the system is the total power
consumption, i.e.:

NF
gt (st , at ) = pi (t), ∀t = 1, 2, . . . , T . (11.33)
i=1

Due to the randomness of the channel fading, a penalty is added in case there are
some remaining bits after T frames transmission (penalty on the remaining bits
in the (T + 1)th frame). Hence, the following cost is introduced for the (T + 1)th
frame:

gT +1 (sT +1 , aT +1 ) = w q(T + 1), (11.34)

where w is the weight for the penalty and q(T + 1) is the number of remaining
bits after T frames. Note that there is no control action in the (T + 1)th frame and
aT +1 is introduced simply for notation consistency.

As a result, the optimization of transmission resource allocation can be written

as the following finite-horizon MDP:
T +1

T
NF
min E gt (st , at ) = min E pi (t) + w q(T + 1) .
{t |t=1,2,...,T } {t |t=1,2,...,T }
t=1 t=1 i=1

(11.35)

The expectation is because that pi (t) (∀i, t) and q(T + 1) are random due to channel
fading. It can be observed that the choice of weight w may have strong impact on the
scheduling policy: small weight leads to conservative strategy (try to save energy)
and large weight makes the transmitter aggressive.
The Bellman equation for the above MDP is given in (11.25), where VT +1 (sT +1 ) =
w q(T + 1) can be calculated directly. However, because the space of CSI is continuous
and infinite, it is actually impossible to evaluate the other value functions. Note that
Reinforcement-learning-based wireless resource allocation 385

the CSI is i.i.d. distributed among different frames, the expectation on CSI can be
taken on both sides of the Bellman equation, which can be written as

V t (q(t)) = Eh Vt (st )
⎡ ⎤

= Eh min⎣gt (st , at ) + Pr(st+1 |st , at )Vt+1 (st+1 )⎦
at
st+1
⎡ ⎤

= Eh min⎣gt (st , at ) + Pr {hi (t + 1)|∀i} Vt+1 (st+1 ) Pr(q(t + 1)|st , at )⎦
at
st+1

= Eh min gt (st , at ) + V t+1 (q(t + 1)) , (11.36)

where Eh denotes the expectation over CSI. Therefore, an equivalent Bellman equation
with compressed system state is obtained, whose value function V t (t = 1, 2, . . . , T )
depends only on the QSI. The dependence of CSI is removed from the value function,
which is mainly due to the nature of i.i.d. distribution. As a result, the state space is
reduced from infinite to finite, and a practical solution becomes feasible.
The off-line VI can be applied to compute the new value function V t for all states
and stages (off-line VI), which is given below:

● Initialize the value function of the (T + 1)th stage as

V T +1 (q(T + 1)) = w q(T + 1). (11.37)

● For t = T , T − 1, . . . , 1, evaluate the value function according to (11.36). Note

that there is an expectation with respect to channel fading, the Monte Carlo method
can be used to calculate the value function numerically by generating sufficient
number of CSI realizations according to its distribution (e.g., Rayleigh fading).

With the value functions, the optimal online scheduling when the system is running
can be derived in each stage according to

∗t (st ) = at∗ = arg min gt (st , at ) + V t+1 (q(t + 1)) . (11.38)
at

Note that in both off-line VI and online scheduling, we always need to find the
optimal solution for (11.38), which can be solved as follows. From the principle of
water-filling method, it can be derived that with a given total transmission power in
the tth frame, the optimal power allocation on each subcarrier can be written as

1 σz2
pi (t) = max 0, − , ∀i = 1, 2, . . . , NF , (11.39)
βt
hi (t)
2

where βt is determined by the total transmission power on all the subcarriers. There-
fore, the key of solution is to find the optimal total transmission power (or βt ) for the
386 Applications of machine learning in wireless communications

tth frame such that the right-hand-side of (11.38) is minimized. Notice that the num-
ber of information bits delivered in the tth frame is given by (11.32), the optimization
problem on the right-hand-side of (11.38) can be rewritten as

NF
1 σz2
min max 0, − + V t+1 (max{0, q(t) − d(t)}), (11.40)
βt
i=1
βt
hi (t)
2

which can be solved by one-dimensional search on βt .

From the above solution, the difference between this problem and Example 11.1
can be observed. In both problems, the power allocation follows the expression of
(11.39). Their difference lies in the choice of βt . In Example 11.1, βt is determined by
the peak transmission power constraint, whereas in this problem, βt should be opti-
mized via (11.40). In other words, if the MAC layer queue dynamics are considered
in the power allocation, different system states or stages will result in different water
levels. This insight is intuitive. For example, if there are still a lot of bits waiting to
be delivered, the transmitter tends to use high-transmission power (small βt ), and vice
versa. It is worth to mention that the value function V 1 (B) is the minimum average
system cost to deliver all the B information bits within T frames.
Finally, a numerical simulation result is provided in Figure 11.3 to demonstrate
the performance gain of the finite-horizon MDP formulation, where the baseline is the
power allocation via conventional physical layer water-filling method with constant
power constraint in each frame. It can be observed that the MDP approach always has

650
Baseline [peak power(30 W) water-filling]
600 Baseline [peak power(25 W) water-filling]
Baseline [peak power(20 W) water-filling]
550 Baseline [peak power(15 W) water-filling]
Proposed (MDP)

500
Average cost

450

400

350

300

250

200
6 7 8 9 10 11
T (frames)

Figure 11.3 Performance comparison between conventional water-filling

algorithm (baseline algorithm) and finite-horizon MDP algorithm
(proposed algorithm)
Reinforcement-learning-based wireless resource allocation 387

less cost than the physical layer approach with various peak power levels. This gain
mainly comes from the cross-frame power scheduling, which exploits the channel
temporal diversity.

11.2.3 Infinite-horizon MDP with discounted cost

When the scope of optimization is extended to infinite time horizon, the MDP formu-
lation and solution would become quite different from the case of finite time horizon.
In the example of Section 11.2.2.1, the system cost is the summation of all transmis-
sion powers in all the subcarriers and frames. It can be imagined that if it is extended
to infinite time horizon with the possibility of new packet arrival at the transmitter,
the system cost will tend to infinity, i.e., it cannot be measured. In order to handle this
issue, in infinite-horizon MDP, two measurements on the system cost are considered,
namely, discounted cost and average cost. This section will introduce the formulation
and solution for discounted system cost, and the case of average cost is left to the
next one.
In infinite-horizon MDP, it is usually assumed that the cost function and the
control policy are the same for all stages. Hence, let s and a be the system state and
control action in certain stage, the corresponding cost can be denoted as g(s, a) and
the overall discounted cost for one certain control policy and initial system state at
the first stage s1 can be written as
T

G() = lim E γ g(st , at )s1 ,
t−1
(11.41)
T →+∞
t=1

where st and at = (st ) are the system state and control action of the tth stage,
respectively. The expectation is taken over all possible state transition, and the infinite
summation usually converges due to the discount factor γ ∈ (0, 1). As a result, the
infinite-horizon MDP can be mathematically described as follows:

Problem 11.4 (Infinite-horizon MDP with discounted cost). Find the optimal
control polices, denoted as ∗ , such that the overall cost G is minimized, i.e.:
T

∗
= arg min G() = arg min lim E t−1
γ g st , (st ) . (11.42)
T →+∞
t=1

In order to derive the solution of the above problem, the following cost-to-go
function (value function) is first defined for one arbitrary system state s1 at the first
stage:
T

V (s1 ) = min lim E γ g st , (st ) s1
t−1
T →+∞
t=1

T

= min g(s1 , a1 ) + lim E γ t−1 g(st , at )s1 . (11.43)
T →+∞
t=2
388 Applications of machine learning in wireless communications

With the definition of V , the system cost starting from the tth stage given the state st
at the tth stage can be written as
T

min lim E γ g(sn , an )st
n−1
T →+∞
n=t

T

=γ t−1
min lim E γ n−t
g(sn , an )st
T →+∞
n=t
⎡
⎤

T

= γ t−1 min lim E ⎣ γ k−1 g(sk+t−1 , ak+t−1 )st⎦ , (11.44)
T →+∞
k=1

where the second equality is due to k = n − t + 1 and T = T − t + 1. If we define the

new notation for system state by letting sk = sk+t−1 and ak = ak+t−1 , the minimization
of the above equation can be written as
⎡ ⎤
T

min lim E ⎣ γ k−1 g(sk+t−1 , ak+t−1 )st⎦
T →+∞
k=1
⎡
⎤

T

= min lim E ⎣ γ k−1 g(sk , ak )s1⎦
T →+∞
k=1

= V (s1 ) = V (st ). (11.45)
Hence, it can be derived that
T

min lim E γ g(sn , an )st = γ t−1 V (st ).
n−1
(11.46)
T →+∞
n=t

Since the time horizon is infinite, the optimal policy minimizing the system cost
since the first stage also minimizes the system cost since any arbitrary stage. Hence,
(11.43) can be written as
T

V (s1 ) = min g(s1 , a1 ) + Es2 lim E{si |i=3,4,...} γ t−1 g(st , at )s2
T →+∞
t=2

T

= min g(s1 , a1 ) + Es2 min lim E{si |i=3,4,...} γ t−1
g(st , at )s2
(s1 ) T →+∞
t=2
!
= min g(s1 , a1 ) + γ Es2 V (s2 ) , (11.47)
(s1 )

where the last equality is due to (11.46). Similarly, for arbitrary system state at the
arbitrary tth stage st , the Bellman equation for infinite-horizon MDP with discounted
cost can be written as follows:

V (st ) = min g(st , at ) + γ Est+1 V (st+1 ) . (11.48)
(st )
Reinforcement-learning-based wireless resource allocation 389

Regarding to the solution, if the value function has already been calculated, it is
straightforward to see that the optimal control action for arbitrary stage is

∗ (st ) = at∗ = arg min g(st , at ) + γ Est+1 V (st+1 ) , ∀t, st . (11.49)
at

On the other hand, the value function should satisfy the Bellman equation in (11.48).
This is a fixed-point problem with minimization on the right-hand-side, and we have
to rely on the iterative algorithm, which is named as VI. The detail steps of VI is
elaborated below, and please refer to [3] for the proof of convergence.

VI algorithm for infinite-horizon MDP with discounted cost

The value functions defined in (11.48) can be evaluated by the following steps:
1. Let i = 0 and initialize the value function V (s) for all possible s ∈ S , which
is denoted as V i (s).
2. In the ith iteration, update the value function as

V i+1 (st ) = min g(st , at ) + γ Est+1 V i (st+1 ) , (11.50)

for all possible st ∈ S .

3. If the update from V i to V i+1 for any system state is negligible (or less than one
predetermined threshold), the iteration terminates. Otherwise, let i = i + 1
and jump to Step 2.

In the following section, we still use the case of multi-carrier power allocation
to demonstrate the formulation via infinite-horizon MDP with the discounted cost. It
takes the packet arrival at the MAC layer into considered, which is not addressed in
Section 11.2.2.1.

11.2.3.1 Case study: multi-carrier power allocation with random

packet arrival
The resource allocation problem introduced in Example 11.2 can also be addressed
with discounted cost, which will be elaborated in this example. Specifically, a point-
to-point OFDM communication link with NF subcarriers and random packet arrival at
the transmitter is considered. It is assumed that each packet consists of B information
bits, and one packet should be transmitted within one frame. The key elements of
MDP formulation are elaborated below:

● System state: In the tth frame (t = 1, 2, 3, . . .), the system state st is uniquely
specified by the CSI of all the subcarriers {hi (t)|i = 1, 2, . . . , NF } and the QSI
q(t). The latter denotes the number of remaining packets waiting at the transmitter.
Thus:

st = {hi (t)|i = 1, 2, . . . , NF }, q(t) . (11.51)
390 Applications of machine learning in wireless communications

● Control policy: The control action in the tth frame (t = 1, 2, 3, . . .) is the power
allocation on all the subcarriers, i.e., at = {pi (t)|i = 1, 2, . . . , NF }. Then the
control policy in the tth frame, denoted as , can be written as
(st ) = at , ∀t, st . (11.52)
● Transition kernel: The block fading channel model is considered, and the CSI in
each frame is i.i.d. distributed. Therefore, the transition kernel can be written as

Pr(st+1 |st , at ) = Pr {hi (t + 1)|i = 1, 2, . . . , NF } Pr q(t + 1)st , at ,
(11.53)
where
q(t + 1) = max{0, q(t) − d(t)} + c(t), (11.54)
and
1
pi (t)
hi (t)
2
NF
d(t) = log2 1 + (11.55)
B i=1
σz2

are the number of packets transmitted in the tth frame, c(t) is the number of arrival
packets in the tth frame. It is usually assumed that c(t) follows the Poisson arrival
with expectation λ, as in Example 11.2. Thus, there are λ arrival packets in one
frame on average.
● Cost: The average power consumption at the transmitter is

1
T NF
P = lim E pi (t) . (11.56)
T →+∞ T t=1 i=1

According to the Little’s Law, the average transmission delay of one packet is

1 q(t)
T
Q
W = = lim E , (11.57)
λ T →+∞ T t=1 λ

where Q is the average number of packets waiting at the transmitter. The weighted
sum of average power and delay is

1 q(t)
T NF
P + ηW = lim E η + pi (t) , (11.58)
T →+∞ T t=1 λ i=1

where η is the weight on the average transmission delay. The problem of minimiz-
ing P + ηW is an infinite horizon MDP with average cost, whose solution will
be introduced in the next section. Usually, people prefer to consider the discount
approximation of P + ηW as follows:
T
q(t) NF

G = lim E γ t−1 η + pi (t) . (11.59)

T →+∞ λ
t=1 i=1
Reinforcement-learning-based wireless resource allocation 391

The main reason for approximating average cost via discounted cost is that it has
better converge rate in VI.
Hence, the resource allocation problem can be formulated as
⎡ ⎤
⎢ T ⎥
⎢ t−1 q(t) NF
⎥
⎢
min G = min lim E ⎢ + pi (t) ⎥
T →+∞
γ η
λ ⎥, (11.60)

⎣ t=1 i=1 ⎦

g(st ,at )

which is an infinite-horizon MDP with discounted cost. The Bellman equation for
this problem is

q(t)
NF
V (st ) = min η + pi (t) + γ Est+1 V (st+1 ) . (11.61)
λ i=1

Note that the space of the system state includes all possible values of CSI, and it is
actually impossible to evaluate value function. Similar to Section 11.2.2.1, since the
CSI is i.i.d. distributed in each frame, the expectation with respect to the CSI can be
taken on both side of the above Bellman equation, i.e.:

q(t)
NF
V q(t) = Eh min η + pi (t) + γ Est+1 V (st+1 )
λ i=1
⎧ ⎫
⎨ q(t) NF
⎬
= Eh min η + pi (t) + γ Pr q(t + 1)|st , at V q(t + 1)
⎩ λ ⎭
i=1 q(t+1)
⎧ ⎫
q(t) ⎨NF
λc(t) e−λ ⎬
=η + Eh min pi (t) + γ V q(t + 1)
λ ⎩ c(t)! ⎭
i=1 c(t)
N
q(t) F
=η + Eh,c min pi (t) + γ V q(t + 1) , (11.62)
λ
i=1

where Eh is the expectation over CSI, Est+1 is the expectation over next system state,
and Eh,c is the expectation over random packet arrival.
The off-line VI can be applied to compute the value function V for all possible
queue length. In order to avoid infinite transmission queue, we can set a buffer size.
Thus, the overflow packets will be dropped. With the value function, the optimal
scheduling action can be calculated via:
⎧ ⎫
⎨ NF
λc(t) e−λ ⎬
∗ (st ) = at∗ = arg min pi (t) + γ V q(t + 1) . (11.63)
⎩ c(t)! ⎭
i=1 c(t)

Note that in both off-line VI and online scheduling, we always need to find the optimal
solution for (11.63), which can be solved with the approach introduced in Section
11.2.2.1 (i.e., water-filling with optimized water level).
392 Applications of machine learning in wireless communications

45
Baseline [peak power(25 W) water-filling]
Baseline [peak power(20 W) water-filling]
Baseline [peak power(15 W) water-filling]
40 Baseline [peak power(10 W) water-filling]
Proposed (MDP)

35
Average cost

20
0.6 0.65 0.7 0.75 0.8 0.85 0.9 0.95 1
λ

Figure 11.4 Performance comparison between conventional water-filling

algorithm (baseline algorithm) and infinite-horizon MDP algorithm
(proposed algorithm)

Finally, a numerical simulation result is provided in Figure 11.4 to demonstrate

the performance gain of the infinite-horizon MDP formulation, where the baseline
is the power allocation via conventional physical layer water-filling method with
constant power constraint in each frame. It can be observed that the MDP approach
always has less cost than the physical layer approach with various peak power levels.
Particularly, the performance gain of the MDP formulation is more significant in the
region of heavier traffic (larger λ).

11.2.4 Infinite-horizon MDP with average cost

In the problem formulation of Section 11.2.3.1, the average cost, a weighted summa-
tion of average power and average delay, is approximated as discounted cost so that
the solution for infinite-horizon MDP with discounted cost can be applied. In this
section, we shall show how to handle the exact average cost via infinite-horizon MDP.
Let s and a be the system state and control action in certain stage, the corre-
sponding cost is g(s, a) and the overall average cost for one certain control policy
can be written as

1
T
G() = lim E g(st , at ) , (11.64)
T →+∞ T t=1
where st and at = (st ) are the system state and control action of the tth stage,
respectively. The expectation is taken over all possible state transition. Therefore,
Reinforcement-learning-based wireless resource allocation 393

the infinite-horizon MDP with average cost can be mathematically described as

follows:
Problem 11.5 (Infinite-horizon MDP with discounted cost). Find the optimal
control policy, denoted as ∗ , such that the overall cost G is minimized, i.e.:

1
T
∗
= arg min G() = arg min lim E g st , (st ) . (11.65)
T →+∞ T t=1

Comparing with Problem 11.4, it can be observed that the discounted cost MDP
values the current cost more than the future cost (due to discount factor γ ), but the
average cost MDP values them equally. Moreover, when the discount factor γ of
Problem 11.4 is close to 1, the discounted cost MDP becomes closer to the average
cost MDP.
Unlike the case of discounted cost, the value function for the case of average cost
does not have straightforward meaning. Instead, the value function is defined via the
following Bellman equation:

θ + V (st ) = min g(st , at ) + Est+1 V (st+1 ) , ∀st , at , (11.66)
(st )
where V (s) is the value function for system state s. As proved in [3], this Bellman
equation could bring the following insights on Problem 11.5:
● θ is the minimized average system cost, i.e.:

1
T
θ = min lim E g st , (st ) . (11.67)
T →+∞ T t=1
● The optimal control action for arbitrary system state st at arbitrary tth stage can
be obtained by solving the right-hand-side of (11.66), i.e.:

∗ (st ) = at∗ = arg min g(st , at ) + Est+1 V (st+1 ) . (11.68)
at

Moreover, the VI to calculate the value function is elaborated below.

VI algorithm for infinite-horizon MDP with average cost

The value functions for infinite-horizon MDP can be evaluated by the following
steps:
1. Let i = 0 and initialize the value function V i (s) for all possible s ∈ S . More-
over, arbitrarily choose one system state as the reference state, which is
denoted as sref .
2. In the ith iteration, update the value function for all system states as follows:

V i+1 (st ) = min g(st , at ) + Est+1 V i (st+1 ) − V i (sref ), ∀st . (11.69)
(st )

3. If the update from V i to V i+1 for all system states is negligible, the iteration
terminates. Otherwise, let i = i + 1 and jump to Step 2.
394 Applications of machine learning in wireless communications

11.2.4.1 Case study: multi-carrier power allocation with average cost

Following the definition of system state, transition kernel and control policy in
Section 11.2.3.1, the average cost minimization of the point-to-point OFDM
transmission is given by
⎡ ⎤
⎢ ⎥
⎢ T 1 q(t) NF
⎥
⎢
min G = min lim E ⎢ η + pi (t) ⎥ (11.70)
T →+∞ ⎥.

⎣ t=1 T λ i=1 ⎦

g(st ,at )

Its Bellman equation after taking expectation on the CSI can be written as
⎧ ⎫
q(t) ⎨NF
λc(t) e−λ ⎬
θ + V q(t) = η + E min pi (t) + V q(t + 1) ,
λ ⎩ c(t)! ⎭
i=1 c(t)

(11.71)

where V (q(t)) is the value function with q(t) packets at the transmitter. The VI can
be used to evaluate the value function for all the possible queue lengths (a maximum
queue length can be assumed to avoid infinite queue). Moreover, with the value
function, the optimal scheduling action at arbitrary one stage (say the tth stage) with
arbitrary system state st can be derived via:
⎧ ⎫
⎨NF
λc(t) e−λ ⎬
∗ (st ) = at∗ = arg min pi (t) + V (q(t + 1)) . (11.72)
at ⎩ c(t)! ⎭
i=1 c(t)

Note that q(t + 1) depends on both pi (t) (∀i) and c(t). This problem can be solved
with the approach introduced in Section 11.2.2.1 (i.e., water-filling with optimized
water level).

11.3 Reinforcement learning

In the previous section, when introducing the solution of MDP, we actually assume
that the state transition kernel and the system cost function of each stage are precisely
known. If this knowledge is unknown, which might happen in practice, the methods
of reinforcement learning can be used to collect the information in an online way and
finally yield the desired solution.
In this section, we shall use the case of infinite-horizon MDP with discounted cost
as an example to explain some methods of reinforcement learning. The approaches
can be similarly applied on the other forms of MDPs. It has been introduced in
the previous section that the solution of MDP includes the off-line VI and online
Reinforcement-learning-based wireless resource allocation 395

scheduling. Now, consider the following VI, which is supposed to be finished before
running the system:

!
V i+1 (st ) = min g(st , at ) + γ Est+1 V i (st+1 )
at
⎧ ⎫
⎨ ⎬
= min g(st , at ) + γ Pr(st+1 |st , at )V i (st+1 ) . (11.73)
at ⎩ ⎭
st+1

It can be observed that the VI relies on the knowledge of cost function g(st , at ) and
the state transition probability Pr(st+1 |st , at ). In other words, the VI is infeasible if
they are unknown. In the following example, we extend the power allocation example
of Section 11.2.3.1 from ideal mathematical model to practical implementation and
show that the cost function or the transition kernel (state transition probability) may
be unknown in some situation.

Example 11.3 (Multi-carrier power allocation with unknown statistics). In the

example of Section 11.2.3.1, after taking the expectation on CSI, the equivalent
Bellman equation is given by
N
q(t) F
V q(t) = η + Eh,c min pi (t) + γ V q(t + 1) .
λ
i=1

The expectation on the right-hand-side is with respect to the distributions of CSI

and packet arrival. It is usually to assume that they are Rayleigh fading and Poisson
arrival, respectively. In practice, the BS may be lack of their statistics, e.g., mean
or variance, or they may even not follow the assumed distributions. Hence, the VI
based on the above equation cannot be carried on off-line.
In order to match the above equations with the standard form of Bellman
equation (11.73), we can define the following control policy with respect to the
queue length:

q(t) = (st )|∀hi (t), i = 1, 2, . . . , NF = at . (11.74)

Thus, is a mapping from the queue length to the power allocations for all
possible CSI. With the definition of , the Bellman equation for the example of
Section 11.2.3.1 can be written as
N
q(t) F

V q(t) = min η + Eh pi (t) + γ Eh,c V q(t + 1) ,

(q(t)) λ i=1
396 Applications of machine learning in wireless communications

where cost function and state transition probability of (11.73) are given by
N
q(t) F

g(st , at ) = η +E pi (t) , (11.75)

λ i=1

and

Pr(st+1 |st , at ) = Eh Pr q(t + 1)q(t), (q(t)), {hi (t)} , (11.76)

respectively. Hence, the cost function requires the knowledge of CSI distribu-
tion, and the state transition probability depends on the distributions of both CSI
and packet arrival. Without the knowledge of both distribution, the off-line VI is
infeasible.

In order to find the optimal policy without the a priori knowledge on the statistics
of the system, we have to perform VI in an online way, which is usually referred to
as reinforcement learning. In the remaining of this section, we shall introduce two
learning approaches. The first approach can be applied on the example of Section
11.2.3.1 without any knowledge on CSI distribution, and the second one, which is
call as Q-learning, is more general to handle unknown statistics of packet arrival.

11.3.1 Online solution via stochastic approximation

In this section, we shall focus on the particular type of MDPs as elaborated in Example
11.3. For the elaboration convenience, we extend the formulation of infinite-horizon
MDP with discounted cost in Section 11.2.3 by including an independent random
variable in each stage. Specifically, suppose that ξt is a random variable (or a set
of random variables) at the tth stage, and its distribution is i.i.d. with respect to t.
Now, consider an infinite-horizon MDP with discounted cost, where the cost function
at the tth stage is g(st , at , ξt ) and the transition kernel is given by the distribution
Pr(st+1 |st , at , ξt ). st is the system state at the tth stage, and ξt (which is not included
in st ) can also be observed at the beginning of the tth stage. In a policy , the control
action at is determined according to both st and ξt , i.e., at = (st , ξt ). The MDP
problem can be described as

T

min lim E γ t−1
g(st , at , ξt )s1 , (11.77)
T →+∞
t=1

where the value function is defined as

T

V (s) = min lim E γ t−1 g(st , at , ξt )s1 = s . (11.78)
T →+∞
t=1
Reinforcement-learning-based wireless resource allocation 397

Note that this is the exact MDP problem discussed in Section 11.2.3.1, where ξt and
system state st refer to the CSI and queue length in the tth frame, respectively. Its
Bellman equation can be written as

V (st ) = Eξt min g(st , at , ξt ) + γ Pr(st+1 |st , at , ξt )V (st+1 ) . (11.79)
at

In this section, we assume that ξt (∀t), g(st , at , ξt ), and Pr(st+1 |st , at , ξt ) can be
observed or measured at each stage, but the distribution of ξt is unknown. This
refers to the situation that the CSI distribution in the example of Section 11.2.3.1
is unknown (the distribution of packet arrival is known). Hence, the off-line VI is
infeasible as the right-hand-side of (11.79) cannot be calculated. Instead, we can
first initialize a control policy, evaluate the value function corresponding to this
policy via stochastic approximation in an online way, and then update the policy
and reevaluate the value function again. By such iteration, it can be proved that
the Bellman equation of (11.79) can be finally solved. The algorithm is elaborated
below.

Online value and policy iteration

1. Let i = 0, and initialize a control policy i .
2. Run the system with the policy i and evaluate the corresponding value
function V i which satisfies:
* +
V i (st ) = Eξt g(st , at , ξt ) + γ Pr(st+1 |st , at , ξt )V i (st+1 ) , ∀st , (11.80)
where st+1 denotes the next stage system state given the current stage system
state st .
3. Update the control policy from i to i+1 via:

i+1 (st , ξt ) = arg min g(st , at , ξt ) + γ Pr(st+1 |st , at , ξt )V i (st+1 ) .
at

(11.81)
Note that ξt can be observed at the tth stage (e.g., the CSI can be estimated
at the beginning of each frame in the example of Section 11.2.3.1), the above
optimization problem can be solved.
4. If the update on the control policy is negligible, terminate the algorithm.
Otherwise, let i = i + 1 and jump to Step 2.

It can be proved that the policy and value function obtained by the above itera-
tive algorithm, denoted as V ∞ and ∞ , can satisfy the Bellman equation in (11.79).
Thus, ∞ is the optimal control policy and V ∞ represents the minimum discounted
cost for each initial system state. Notice that in the second step of the above algo-
rithm, we should solve a fixed-point problem with unknown statistics. The stochastic
398 Applications of machine learning in wireless communications

approximation introduced in Section 11.1 can be applied. Specifically, the procedure

is elaborated below.

Stochastic approximation algorithm for value function

1. Let j = 1. Initialize the value function V i , and denote it as V i, j .
2. At the jth stage, denote sj as the system state and update the value function
as follows:
j 1 * +
V i, j+1 (sj ) = V i, j (sj ) + g(sj , aj , ξj ) + γ V i, j (sj+1 ) , (11.82)
j+1 j+1
and

V i, j+1 (s) = V i, j (s), ∀s = sj . (11.83)

Thus, the value for the current system state sj is updated, and others remain
the same. As a remark,
* notice that g(sj , aj , ξj ) + γ V i, j (sj+1 + ) is an unbiased
estimation of Eξt g(st , at , ξt ) + γ Pr(st+1 |st , at , ξt )V i (st+1 ) . Moreover, since
the knowledge on sj+1 is required, the above update should be calculated after
observing the next system state.
3. If the update on the value function is negligible, terminate the algorithm.
Otherwise, let j = j + 1 and jump to Step 2.

11.3.1.1 Case study: multi-carrier power allocation without channel

statistics
In this example, we shall continue the optimization of power allocation without chan-
nel statistics, as initiated in Example 11.3, by the online value and policy iteration
described in this section (Section 11.3.1). The notations will follow the definitions in
Example 11.3 and Section 11.2.3.1. In order to match the MDP formulation of this
section, we shall treat the QSI only as the system state (i.e., st ), and the CSI as the
independent variables in each optimization stage (i.e., ξt ). Specifically, the problem
formulation of Example 11.3 is established below.

● System state: Since the distribution of CSI is i.i.d. in each frame, we can treat
the CSI as the independent random variables ξt , instead of the system state. Thus:

ξt = {hi (t)|∀i}, (11.84)

whose distribution is unknown. The system state becomes:

st = {q(t)}. (11.85)
Reinforcement-learning-based wireless resource allocation 399

● Control policy: As elaborated in Example 11.3, when the CSI is removed from
the system state, the control action becomes the power allocation for all possible
CSI given the QSI. Thus, the control action of the tth frame is

at = (st ) = q(t), {hi (t)} ∀i, hi (t) . (11.86)

As a remark note that and represent the same scheduling behavior, however
their mathematical meanings are different: is a policy with respect to QSI and
CSI, and is a policy with respect to QSI only. Hence, one action in consists
of a number of actions in with the same QSI.
● Transition kernel: Given the system state, CSI and control action of the tth frame,
the transition kernel can be written as

Pr(st+1 |st , at , ξt ) = Pr q(t + 1)q(t), {hi (t)|∀i}, {pi (t)|∀i} , (11.87)

where
q(t + 1) = max{0, q(t) − d(t)} + c(t), (11.88)
1
pi (t)
hi (t)
2
NF
d(t) = log2 1 + (11.89)
B i=1 σz2
is the number of packets transmitted in the tth frame, c(t) is the number of arrival
packets in the tth frame. Note that the randomness of q(t + 1) comes from random
packet arrival c(t).
● Cost: The overall cost function as defined in Section 11.2.3.1 is
T
q(t) NF

G = lim E γ t−1
η + pi (t) . (11.90)
T →+∞ λ
t=1 i=1

As elaborated in Example 11.3, the Bellman equation for the above MDP
problem is
N
q(t) F

V q(t) = min η + Eξt pi (t) + γ Eξt ,c(t) V q(t + 1) ,

(q(t)) λ i=1

(11.91)
or equivalently:
⎧ ⎫
⎨ q(t) NF
⎬
V q(t) = Eξt min η + pi (t) + γ Pr c(t) V q(t + 1) .
(q(t)) ⎩ λ ⎭
i=1 c(t)

(11.92)
Note that without the distribution knowledge of CSI ξt , the expectations in the above
Bellman equation cannot be calculated directly. Hence, we have to rely on the online
value and policy iteration introduced in this section, which consists of two levels of
iteration. The outer iteration is for updating the policy, and the inner one is to find the
value function corresponding to the policy. The procedure is elaborated below.
400 Applications of machine learning in wireless communications

Step 1 (Initialize a policy): In order to obtain an initial policy, we can first

0
initialize the value function, which is denoted as V , and derive a power allocation
policy by solving the right-hand-side of the above Bellman equation (11.92) with the
initialized value function. Specifically, from the principle of water-filling method, it
is known that given a total transmission power in the tth frame, the optimal power
allocation on each subcarrier can be written as

1 σz2
pi (t) = max 0, − , ∀i = 1, 2, . . . , NF , (11.93)
βt
hi (t)
2

where βt depends on the total transmission power of the tth frame. βt is usually
referred to as the Lagrange multiplier as the above power allocation is derived via
convex optimization [5]. Moreover, given st and ξt , the βt with respect to the initialized
0
value function V , denoted as βt1 , can be calculated according to the right-hand-side
of (11.92), i.e.,

⎧ ⎫
⎨ q(t) NF ⎬
1 σz 2 0
βt1 = arg min η + 0, − +γ Pr c(t) V q(t + 1) .
⎩ λ βt
hi (t)
2 ⎭
i=1 c(t)

As a result, the initial power allocation policy is then given by

1 σz2
p1i (t) = max 0, 1 − , ∀i = 1, 2, . . . , NF . (11.94)
βt
hi (t)
2

Step 2 (Value function evaluation): Given the power allocation policy derived
based on βti (i = 1, 2, . . .), the corresponding value function can be calculated as
follows:
i, j i−1
● Let j = 1. Initialize the value function by V = V .
● At the jth stage, denote sj as the system state and update the value function as
follows:

i, j+1 j i, j 1 * +
V (sj ) = V (sj ) + g(sj , aj , ξj ) + γ V i, j (sj+1 ) , (11.95)
j+1 j+1

and
i, j+1 i, j
V (s) = V (s), ∀s = sj . (11.96)

i
● Let j = j + 1 and repeat the above step until the iteration converged. Let V be
the converged value function.
Reinforcement-learning-based wireless resource allocation 401

Step 3 (Policy evaluation): Update the βt as

NF
q(t) 1 σz2
βt = arg min η
i+1
+ 0, −
λ i=1
βt
hi (t)
2
⎫
i ⎬
+γ Pr c(t) V q(t + 1) .
⎭
c(t)

As a result, the updated power allocation policy is then given by

1 σz2
pi (t) = max 0, i+1 −
i+1
, ∀i = 1, 2, . . . , NF . (11.97)
βt
hi (t)
2
If the update on the power allocation policy is negligible, terminate the algorithm;
otherwise, let i = i + 1 and jump to the Step 2.
The above online algorithm will converge to the optimal power allocation as in
Section 11.2.3.1, and the performance illustrated in Figure 11.4 also applies on it.
Moreover, it can be observed from the above algorithm that the calculation of βt in
each iteration requires the distribution knowledge to packet arrival. We may learn
it from the history of packet arrival, if it is unknown at the very beginning. In fact,
Q-learning is one more elegant way to handle this situation, which is elaborated in
the below section.

11.3.2 Q-learning
The stochastic-approximation-based learning approach in the previous section is able
to handle the situation that the controller knows the transition kernel Pr(st+1 |st , at , ξt )
but does not know its expectation with respect to ξt , i.e., Pr(st+1 |st , at ) =
Eξt [ Pr(st+1 |st , at , ξt )]. Regarding to the example in Section 11.3.1.1, it refers to the
circumstance that the transmitter knows the distribution of packet arrival in each
frame, but not the CSI distribution. Q-learning is a more powerful tool to solve MDP
problems with unknown transition kernel Pr(st+1 |st , at ). In other words, it can handle
the power allocation even without the statistics of packet arrival.
We use the infinite-horizon MDP with discounted cost in Problem 11.4 as the
example to demonstrate the method of Q-learning. First of all, the Q function is
defined as
T

Q(s, a) = min lim E γ g(st , at )s1 = s, a1 = a .
t−1
(11.98)
T →+∞
t=1

Hence, the relation between value function and Q function is

V (s) = min Q(s, a), (11.99)
a

and the optimal control action for system state s is

∗ (s) = arg min Q(s, a). (11.100)
a
402 Applications of machine learning in wireless communications

In other words, the optimal control policy can be easily obtained with the Q function
of the MDP. Moreover, the Bellman equation in (11.48) can be written in the form of
Q function, i.e.:
Q(st , at ) = g(st , at ) + γ Pr(st+1 |st , at )V (st+1 ), (11.101)

V (st ) = min g(st , at ) + γ Pr(st+1 |st , at ) min Q(st+1 , at+1 ) , (11.102)
at at+1

or
Q(st , at ) = g(st , at ) + γ Pr(st+1 |st , at ) min Q(st+1 , at+1 ). (11.103)
at+1

In order to compute and store the values of Q function, it is actually required that
both the state and action spaces should be finite. Regarding the example of Section
11.3.1.1, we should quantize the transmission power into finite levels.
The Bellman equation (11.103) provides an iterative way to evaluate the
Q function. The procedure is described below.

VI algorithm for Q function

The Q function defined in (11.103) can be evaluated by the following steps:
1. Let i = 0 and initialize the Q function Q(s, a) for all possible s ∈ S and
a ∈ A , which is denoted as Qi (s, a).
2. In the ith iteration, update the Q function as
Qi+1 (st , at ) = g(st , at ) + γ Pr(st+1 |st , at ) min Qi (st+1 , at+1 ),
at+1

for all possible st ∈ S and at ∈ A .

3. If the update from Qi to Qi+1 for any system state and action is negligible (or
less than one predetermined threshold), the iteration terminates. Otherwise,
let i = i + 1 and jump to Step 2.

The above VI require the knowledge on the distribution of Pr(st+1 |st , at ). If they
are not available at the controller, the Q-learning algorithm is provided below.

Q-learning algorithm
1. Let j = 1, initialize the Q function, denoted as Qj .
2. At the jth stage, denote si as the system state and ai as the action, update the
value function as follows:
, -
j 1
Q (si , ai ) =
i+1
Q (si , ai ) +
i
g(sj , aj ) + γ min Q (si+1 , ai+1 ) ,
i
j+1 j+1 ai+1

and
Qi+1 (s, a) = Qi (s, a), ∀(s, a) = (sj , aj ). (11.104)
Reinforcement-learning-based wireless resource allocation 403

Since the knowledge on sj+1 is required, the above update should be calculated
after observing the next system state.
3. If the update on the Q function is negligible, terminate the algorithm.
Otherwise, let j = j + 1 and jump to Step 2.

In the above algorithm, the control action for each stage should be chosen to
guarantee that the Q function for all pairs of system state and control action can be
well trained.

11.3.2.1 Case study: multi-carrier power allocation via Q-learning

In this example, we shall still consider the power-allocation problem introduced
in Section 11.3.1.1, however, with more practical assumption that not only the
distribution of CSI but also the distribution of packet arrival is unknown. We shall
show that the Q-learning method could provide an online optimization algorithm. The
definitions of system state, control policy, transition kernel, and cost function follow
that in Section 11.3.1.1. Note that Q function is defined in terms of system state
and control action both with finite space, we can quantize the choice of Lagrange
multiplier βt into a finite space B. Thus, the Q function can be written as

Q q(t), βt , where βt ∈ B. (11.105)
Hence, the Bellman equation in terms of Q function becomes:
N
q(t) F
Q q(t), βt = η + Eξt pi (t) + γ Eξt Pr q(t + 1)|q(t), βt , ξt
λ i=1

min Q q(t + 1), β . (11.106)
β

Without the statistics knowledge of ξt and packet arrival c(t), the Q-learning algorithm
is provided below:
1. Let i = 1. Initialize the Q function, denoted as Qi .
2. In the ith frame, let q(i) and βi be the system state (QSI) and the Lagrange
multiplier for power allocation, update the value function as follows:
i
Qi+1 q(i), βi = Qi q(i), βi
i+1
⎡ ⎤
1 ⎣ q(i)
NF
+ η + pj (i) + γ min Q q(i + 1), β ⎦ ,
i+1 λ j=1
β

and

Qi+1 (s, β) = Qi (s, β), ∀(s, β) = q(i), βi . (11.107)
Since the knowledge on q(i + 1) is required, the above update should be
calculated after observing the next system state.
404 Applications of machine learning in wireless communications

3. If the update on the Q function is negligible, terminate the algorithm. Otherwise,

let i = i + 1 and jump to Step 2.
With the Q function, the power allocation for the tth frame can be written as
1 σz2
pi (t) = max 0, − , ∀i = 1, 2, . . . , NF , (11.108)
βt
hi (t)
2

where βt = minβ Q(q(t), β). Note that in this solution, the Lagrange multiplier βt
is determined according to the QSI only. A better solution may be obtained if we
treat both CSI and QSI as the system state in the Q-learning algorithm (βt is then
determined according to both CSI and QSI). However, it requires the quantization of
CSI and larger system complexity.
Comparing with the method introduced in Section 11.3.1, it can be observed that
the Q-learning approach is more general in the sense that it can be applied on the
situation without knowledge of transition kernel. However, the price to pay is that the
Q function depends on both system state and control action. Thus, the storage and
computation complexities for evaluating the Q function is higher.

11.4 Summary and discussion

In this chapter, we focus on the wireless resource allocation along a number of frames,
where the MDP is used to formulate the scheduling as an stochastic optimization
problem. As the foundation of stochastic learning, we first elaborate on the basics
of stochastic approximation. Then we introduce the MDP with three different for-
mulations, and one example of power allocation is provided for each formulation.
As we can see, MDP is powerful to handle the multistage optimization problem with
random future. Moreover, it is common that some system statistics are unknown
before running; we continue to introduce the reinforcement learning to construct
online algorithms, which collect the system information and drive the scheduling to
optimal.
In order to simplify the elaboration, we have ignored the proofs of some math-
ematical statements in this chapter. For the readers who are interested in a more
rigorous treatment on mathematical derivation, please refer to [2,3] for the discus-
sions on the MDP and [6] for the discussions on the reinforcement learning. Moreover,
the application of MDP and reinforcement learning in wireless resource allocation
has drawn a number of research interests. For example, the infinite-horizon MDP has
been used to optimize the point-to-point link [7], cellular uplink [8,9], cellular down-
link [10], relay networks [11], and wireless cache systems [12], where the average
transmission delay is either minimized or constrained. Moreover, the low-complexity
algorithm design via approximate MDP can be considered to avoid the curse of
dimensionality [13].
Reinforcement-learning-based wireless resource allocation 405

References
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[5] Boyd S, and Vandenberghe L. Convex Optimization. 1st ed. Cambridge:
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age Delay: The Single-User Case. IEEE Transactions on Information Theory.
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[8] Moghadari M, Hossain E, and Le LB. Delay-Optimal Distributed Scheduling
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[9] Cui Y, and Lau VKN. Distributive Stochastic Learning for Delay-Optimal
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Chapter 12
Q-learning-based power control in small-cell
networks
Zhicai Zhang1 , Zhengfu Li2 , Jianmin Zhang3 ,
and Haijun Zhang3

Because of the time-varying nature of wireless channels, it is difficult to guarantee

the deterministic quality of service (QoS) in wireless networks. In this chapter, by
combining information theory with the effective capacity (EC) principle, the energy-
efficiency optimization problem with statistical QoS guarantee is formulated in the
uplink of a two-tier femtocell network. To solve the problem, we introduce a Q-
learning mechanism based on Stackelberg game framework. The macro users act as
leaders and know the emission power strategy of all femtocell users (FUS). The femto-
cell user is the follower and only communicates with the macrocell base station (MBS)
without communicating with other femtocell base stations (FBSs). In Stackelberg
game studying procedure, the macro user chooses the transmit power level first
according to the best response of the femtocell, and the micro users interact directly
with the environment, i.e., leader’s transmit power strategies, and find their best
responses. Then, the optimization problem is modeled as a noncooperative game,
and the existence of Nash equilibriums (NEs) is studied. Finally, in order to improve
the self-organizing ability of femtocell, we adopt Q-learning framework based on
noncooperative game, in which all the FBS are regarded as agents to achieve power
allocation. Numerical results show that the algorithm cannot only meet the delay
requirements of delay-sensitive traffic but also has good convergence.

12.1 Introduction
In recent years, most voice and data services have occurred in indoor environments.
However, due to long-distance transmission and high penetration loss, the indoor
coverage of macrocell may not be so good. As a result, FBS has gained wide attention
in wireless industry [1,2]. With the exponential growth of mobile data traffics, wireless

1
College of Physics and Electronic Engineering, Shanxi University, China
2
Beijing Key Laboratory of Network System Architecture and Convergence, Beijing University of Posts
and Telecommunications, China
3
School of Computer & Communication Engineering, University of Science and Technology Beijing,
China
408 Applications of machine learning in wireless communications

communication networks play a more and more important role in the global emissions
of carbon dioxide [3]. Obviously, the increasing energy cost will bring significant
operational cost to mobile operators. On the other hand, limited battery resources
cannot meet the requirement of mass data rate. In this chapter, the concept of green
communication is proposed to develop environmentally friendly and energy saving
technologies for future wireless communications. Therefore, the use of energy aware
communication technology is the trend of the next generation wireless network design.
In a two-tier network with shared spectrum, due to cross-layer interference,
the target user and the femtocell user of each signal-to-interference-plus-noise
ratio (SINR) sampling macrocell are coupled. The SINR target concept estab-
lishes application-related minimum QoS requirements for each user. It is reasonably
expected that since home users deploy femtocell for their own benefit and because
they are close to their BS, femtocell users and cellular users seek different SINRs
(data rates)—usually higher data rates using femtocell. However, QoS improvements
from femtocell should be at the expense of reducing cellular coverage.
In practice, a reliable delay guarantee is provided for delay sensitivity. High
data rate services, such as video calling and video conferencing, are the key issues
of wireless communication network. However, due to the time-varying nature of
wireless channel, it is difficult and unrealistic to apply the traditional fixed delay
QoS guarantee. To solve the problem, the statistical QoS metric with delay-bound
violation probability have been widely adopted to guarantee the statistical delay QoS
[4–6]. In [5], for delay-sensitive traffic in single-cell downlink Orthogonal Frequency
Division Multiple Access (OFDMA) networks, the effective spectrum design based
on EC delay allocation is studied. In [6], a joint power and subchannel allocation
algorithm in vehicular infrastructure communication network is proposed. It has the
requirement of delayed QoS. However, as far as we know, EC-based delay provision
in two-tier femtocell cellular networks has not been widely studied.
In addition, due to the scarcity of spectrum, the microcell and macrocell usu-
ally share the same frequency band. However, in the case of co-channel operation,
intensive and unplanned deployment will lead to serious cross-tier and co-tier inter-
ference, which will greatly limit the performance of the network. Microcell base
stations are low-power, low-cost, and user-deployed wireless access points that use
local broadband connections as backhauls. Not only users but also operators ben-
efit from femtocell. On the one hand, users enjoy high-quality links; on the other
hand, operators reduce operating expenses and capital expenditure due to service
uninstallation and user deployment of FBS.
Therefore, it is necessary to design effective interference suppression mecha-
nism in the two-tier femtocell networks to reduce cross-tier and co-tier interference.
In [7,8], the author reviews the interference management in two-level microcellular
networks and small cellular networks. In [9], the authors have proposed a novel inter-
ference coordination scheme using downlink multicell chunk allocation with dynamic
inter-cell coordination to reduce co-tier interference. In [10], based on cooperative
Nash bargaining game theory, this chapter proposes a cognitive cell joint uplink sub-
channel and power-allocation algorithm to reduce cross-layer interference. In [11],
in order to maximize the total capacity of all femtocell users under the constraints of
Q-learning-based power control in small-cell networks 409

co-layer/cross-layer interference and given minimum capacity QoS, a resource allo-

cation scheme for cognitive nano-cellular is proposed. However, the delay QoS
provisioning was not taken into consideration in [9–11]. Also, as a randomly deployed
base station, the traditional centralized network scheduling is difficult to optimize its
network performance. Therefore, the reinforcement learning which can provide agents
with self-organization capability has attracted considerable interest in academy and
industry [12].
In [13], the author has studied the self-optimization, self-configuration and self-
optimization of small cell network. In [14], aiming at the power control problem in ad-
hoc networks, an enhanced learning algorithm based on random virtual game theory is
proposed. In [15,16], aiming at the utility maximization problem of two-tier femtocell
networks, an enhanced learning algorithm based on hierarchical Stackelberg game is
proposed. However, the algorithms in [15,16] require frequent routing information
exchanges between macrocells and micro cells, which greatly increases network load.
In recent years, there have been many researches on energy-efficient resource
management [17,18]. Energy efficiency was first proposed by Goodman et al., which
is defined as the number of error-free delivered bits for each energy-unit used in trans-
mission and is measured in bit/joule [19]. FBS is a low-power, low-cost base station
that can enhance indoor environment coverage and unload traffic from macrocell.
A low complexity energy-efficient subchannel allocation scheme is proposed in [17],
but the method does not consider interference caused by neighbors. In [18], joint
subchannel allocation and power control are modeled as a potential game to maxi-
mize energy efficiency of multicell uplink OFDMA systems, but QoS guarantees are
without consideration.
In addition to energy saving management of radio resources, femtocell network is
another promising technology for energy saving. Because of this type of deployment
strategy, the transmitter is closer to the receiver and reduces penetration and path loss.
As we know, FBS is installed by end users, who have not enough professional skills to
configure parameters of FBS. On this account, FBS should have self-learning ability to
automatically configure and optimize its operating information, e.g., transmit power
assignment. In recent years, reinforcement learning mechanism, such as Q-learning,
is widely used in radio resource allocation of wireless network [20–22]; however,
most of the existing works are focusing in cognitive radio networks.
In addition, providing delay QoS guarantees while minimizing energy consump-
tion is a key problem in green communication systems. For example, in real-time
services, such as multimedia video conferencing and live sports events, latency time
is a key QoS metric. Since the time-varying channel, deterministic delay QoS guaran-
tee mechanisms used in wired networks cannot take affect in wireless networks [4]. To
address this issue, statistical QoS provisioning, in terms of delay exponent and EC, has
become an effective method to support real-time service in wireless networks [23–25].
Machine learning can be widely used in modeling various technical prob-
lems of next-generation systems, such as large-scale Multiple-Input Multiple-Output
(MIMO), device-to-device networks, heterogeneous networks constituted by fem-
tocells and small cells [26]. Therefore there are some existing works about the
application of machine learning to small cell networks. In [27], a heterogeneous
410 Applications of machine learning in wireless communications

fully distributed multi-objective strategy based on a reinforcement learning model are

proposed to be built for self-configuration and optimization of femtocells. In [28],
the state of the system consists of the user’s specific allocation of small cell resource
blocks and channel quality, and the action consists of downlink power control actions.
The reward is quantified based on an improvement in SINR. The results show that
the compensation strategy based on the reinforcement learning model has achieved
excellent performance improvement.
Machine learning is a discipline that specializes in algorithms that can be learned
from data. In general, these algorithms are run by generating models built from
observational data and then using the generated models to predict and make decisions.
Most problems in machine learning can be translated into multi-objective optimization
problems where multiple targets must be optimized simultaneously in the presence
of two or more conflicting targets. Mapping multiple optimization problems to game
theory can provide a stable solution [29]. Game theory focuses on the nature of
equilibrium states. For example, an in-depth study of the concept of algorithmic
game theory is the concept of anarchy price. The anarchy of certain problems (such as
routing in a crowded network) is the biggest difference between the NE configuration
(the best way each participant routes in the case of other people’s behavior) and the
global optimal solution. However, NE is a subtle object. In large systems with multiple
entities with limited information, it is more natural to assume that each entity self-
adjusts its behavior based on past experience, producing results that may be stable
or unstable. Therefore, it is desirable to use the understanding of the characteristics
of such an adaptive algorithm to draw conclusions about the behavior of the overall
system [30].
In this chapter, we will study energy-efficient power control in uplink two-tier
femtocell networks with delayed QoS guarantees. Based on the concept of EC, we
formulate an energy-efficiency optimization problem with statistical QoS guarantee.
To solve the problem, a transmit power learning mechanism based on Stackelberg
game is proposed. In the learning process, macro users are leaders and can communi-
cate with micro users. Femto-users act as followers and only know the power strategy
of leader rather than other followers. Besides, leader knows followers’ best responses
of transmit power and selects strategy first; followers move subsequently. We use EC
as a network performance metric to provide statistical delay QoS.
Then we adopt pricing mechanism to protect macrocell users (MU) from severe
cross-layer interference. The optimization problem is modeled as a noncooperative
game. Then we study the existence of NEs. Specifically, considering that femtocells
are deployed by end users who have not enough professional skills to configure
and optimize FBSs’s parameters, such as transmit power, we use Q-learning theory
to enable femtocells to achieve self-organizing capability in terms of transmission
power and other parameters. And we propose a distributed Q-learning procedure based
on Stackelberg game. Simulation results show the proposed algorithm has a better
performance in terms of convergence compared with a conjecture-based multi-agent
Q-learning (CMAQL) algorithm with no information exchange between each player
[31]. Based on the noncooperative game framework, a Boltzmann distribution-based
weighted filter Q-learning algorithm (BDb-WFQA) is proposed to realize power
allocation. The simulation results show the proposed BDb-WFQA algorithm can
Q-learning-based power control in small-cell networks 411

meets the large-scale delay requirements, has a better convergence performance,

and a small EC loss compared with the Noncooperative Game-based Power Control
Algorithm (NGb-PCA). This algorithm can meet the large-scale delay requirements,
and has better convergence performance, and has a small EC loss.
The rest of the chapter is organized as follows. In Section 12.2, we briefly discuss
EC and formulate an energy-efficiency optimization problem with statistical delay
provisioning. A noncooperative game theoretic solution is proposed in Section 12.3.
A Q-learning mechanism based on Stackelberg game framework and a WFQA based
on Boltzmann distribution are proposed in Section 12.4. Simulation results are shown
in Section 12.5. In Section 12.6, we conclude the chapter.

12.2 System model

12.2.1 System description

The scenario considered in this chapter is shown in Figure 12.1, where N femtocells
are overlaid in a macrocell, which constitutes a two-tier femtocell network. FBSs are
in closed subscriber group (CSG) mode, i.e., mobile stations (MSs) that are not the
members of the CSG, are not allowed to access the CSG FBSs.
As Figure 12.2, the representative macrocell is covered by several femtocell. In
each femtocell, FBS provides services for its FUS. To analyze traceability, we assume
that only one active MU/FU is scheduled in each MBS/FBS in each signaling slot. It
is worth pointing out that the algorithm obtained under this assumption can be easily
extended to each MBS/FBS scenario with multiple active users.
However, in the two-tier network, cross-layer interference significantly hinders
the performance of traditional power-control schemes. For example, signal-strength-
based power control (channel inversion) adopted by cellular users results in unaccept-
able deterioration of cellular SINR. Because users carry out high-power transmissions

Queue 0 MS 0 B0
h00
h01

Queue 1 MS 1 h0N
B1

hN0
Queue N MS N BN
hN1 hNN

Figure 12.1 System model of two-tier femtocell networks

412 Applications of machine learning in wireless communications

Macrocell

Femtocell

Figure 12.2 The scenario of two-tier femtocell networks

at the edges of their cell to meet their received power targets and cause excessive cross-
layer interference in nearby microcellular networks. Due to scalability, security, and
limited availability of backhaul bandwidth, base station (BS) and femtocells base
station APS.
Let i ∈ N = {0, 1, . . . , N } denote the index of active users, where i = 0 indicates
the scheduled user in macrocell B0 and i ∈ {1, 2, . . . , N } denotes the scheduled users
in femtocell Bi .
Let Bi (i ∈ N ) denote the base station (BS), where N = {0, 1, 2, . . . , N }. B0
denotes the MBS, and Bi (i ∈ N , i = 0) is FBS. We assume that each MS will be
allocated only a subchannel, and in order to avoid intra cell interference during each
frame time slot, only one active MS in each cell can occupy the same frequency.
Let i ∈ N denote the index of scheduled user in Bi .
The received SINR of MS i in Bi can be expressed as
pi hii
γi (pi , p−i ) = , ∀i ∈ N , (12.1)
j =i pj hij + σi
2

where pi denotes the transmit power of MS i, and p−i , (−i ∈ N ) denotes the transmit
power of other MSs except MS i. hii and hij are the channel gains from MS i to BS
Bi , Bj respectively, σi2 is the variance of additive white Gaussian noise (AWGN) of
MS i.
Similarly, the received SINR of MU is
h0,0 p0
γ0 = N , (12.2)
i=1 hi,0 pi + σ02
where hi,0 is the channel gain from FBS Bi to the active MU and h0,0 denotes the
channel gain from MBS to its active MU.
According to the Shannon’s capacity formula, the ideal achievable data rate of
MS i is
Ri (pi , p−i ) = wlog2 (1 + γi (pi , p−i )), (12.3)
where w is the bandwidth of each subchannel.
Q-learning-based power control in small-cell networks 413

12.2.2 Effective capacity

The concept of statistical delay guarantee has been extensively studied in the effective
bandwidth theory [32]. Delay provisioning is an important and challenging problem in
wireless networks for delay-sensitive services such as video calls, video conferencing,
and online games. However, owing to the time-varying nature of wireless channels, it
is difficult and unrealistic to have deterministic delay guarantees for mobile services.
According to Shannon’s law, the capacity potential of a femtocell can be quickly
verified by associating the wireless link capacity (in bits per second) in the bandwidth
with the SINR. SINR is a function of the desired transmitter’s transmit power, path
loss, and shadows. Path losses cause the transmitted signal to decay as Ad −α , where
A is a fixed loss, d is the distance between the transmitter and receiver, and α is the
path loss exponent. The key to increasing capacity is to enhance reception between
intended transmitter receiver pairs by minimizing d and α.
To solve this problem, the concept of EC is proposed in [4], which is defined as
the maximum constant arrival rate guaranteed by a statistical delay specified by the
QoS index of θ on a time-varying channel.
Based on large deviation principle, Chang [32] has pointed out that with sufficient
condition, for a dynamic queueing system with stationary ergodic arrival and service
processes, the queue length process Q(t) converges to a random variable Q(∞):

log(Pr{Q(∞) > Qth })

lim = −θ , (12.4)
Qth →∞ Qth

where Qth is queue length bound and θ > 0 is the decay rate of the tail distribution of
the queue length Q(∞).
If Qth → ∞, we get the approximation of the buffer violation probability,
Pr{Q(∞) > Qth } ≈ e−θ Qth .
We can find that the larger θ corresponds to the faster fading rate, which means
more stringent QoS constraints, while the smaller θ leads to a slower fading rate,
which means a looser QoS requirement. Similarly, the delay-outage probability can
be approximated by [4], Pr{Delay > Dth } ≈ ξ e−θ δDth , where Dth is the maximum
tolerable delay, ξ is the probability of a non-empty buffer, and δ is the maximum
constant arrival rate.
The concept of EC is proposed by Wu et al., in [4], it is defined as the maxi-
mum constant arrival rate that can be supported by the time-varying channel, while
ensuring the statistical delay requirement specified by the QoS exponent θ. The EC
is formulated as

1 K
E c (θ) = − lim ln(E{e−θ k=1 S[k] }), (12.5)
K→∞ Kθ

where {S[k]|k = 1, 2, . . . , K} denotes the discrete-time, stationary, and ergodic

stochastic service process. E{·} is the expectation over the channel state.
414 Applications of machine learning in wireless communications

We assume that the channel fading coefficients remain unchanged over the frame
duration T and vary independently for each frame and each MS. From (12.5), Si [k] =
T Ri [k] is obtained. Based on the above analysis, the EC of MS i can be simplified as
1
Eic (θi ) = − ln(E{e−θi T Ri [k] }). (12.6)
θi T

12.2.3 Problem formulation

The energy efficiency under statistical delay guarantees of MS i is defined as the ratio
of the EC to the totally consumed energy as follows:
Eic (θi )
ηi (pi , p−i ) = . (12.7)
pi + p c
In (12.7), pc represents the average energy consumption of device electronics,
including mixers, filters, and digital-to-analog converters, and excludes that of the
power amplifier. Femtocell is deployed randomly by end users, so cross-layer inter-
ference against MU is uncertain. When the cross-tier interference exceeds MUs’
threshold, the communication of MUs is seriously affected or even interrupted.
Given the minimum SINR guarantee γi∗ , FU i, s utility function can be
expressed as
C
Ei (θi ), if γi (pi , p−i ) ≥ γi∗
ui (pi , p−i ) = , (12.8)
0, otherwise
where p−i denotes the transmit power of other FBSs except FBS i.
Our goal is to maximize the energy efficiency of each MS while meeting the
delay QoS guarantee. Therefore, the corresponding problem is
−ln(E{e−θi T Ri [k] })
max , (12.9a)
θi T (pi + pc )
pi ≥ pmin , ∀i ∈ N , (12.9b)
pi ≤ pmax , ∀i ∈ N , (12.9c)
θi > 0, ∀i ∈ N , (12.9d)
where pmin and pmax are the lower and upper bounds of each MS’s transmit power,
respectively.

12.3 Noncooperative game theoretic solution

In this section, we formulate the FBSs’selfish behavior as a noncooperative game. Let
G = {N , {Pi }, ui (pi , p−i )} denote the noncooperative power control game (NPCG),
where N = {1, 2, . . . N } is the set of FBSs, {Pi } is the strategy set of all players,
and ui (pi , p−i ) is the utility function. It is obvious that the level of each FU’s utility
depends on its FBS’s transmit power and other FBSs’ strategies. We assume that each
Q-learning-based power control in small-cell networks 415

FBS is rational. Each player pursues the maximization of its own utility, which can
be denoted as
max ui (pi ∗ ,p∗ −i ), ∀i ∈ N , (12.10a)
pi ∈Pi

subjectto : pi ≥ pi min , (12.10b)

pi ≤ pi max
, (12.10c)
θi > 0. (12.10d)
Definition 12.1. A given power control strategy (pi ∗ ,p∗ −i ) is an NE point of NPCG,
if for ∀i ∈ N ,∀pi ∈ Pi , the following inequality is satisfied:
ui (pi ∗ ,p∗ −i ) ≥ ui (pi ,p∗ −i ). (12.11)
On the NE point, no player can improve their utility by changing its strategy uni-
laterally [33]. Generally speaking, we can prove the existence of NE by the following
Theorem 12.1.

Theorem 12.1. An NE exits in the NPCG G = {N , {Pi }, ui (pi ,p−i )}, if for all i ∈ N ,
the following two conditions are satisfied:
1. In Euclidean space RN , the strategy set {Pi } is a non-empty, convex, and compact
subset.
2. The utility function ui (pi ,p−i ) is continuous in (pi ,p−i ) and quasi-concave in pi .

Proof. For condition (1), it is obvious that {Pi } is a non-empty, convex, and compact
subset. We prove condition (2) inthe following:
For fixed p−i , let hi = (gi,i / j=0,i gj,i pj + g0,i p0 + σ 2 ) denote the channel gain-
to-interference-plus-noise ratio of FU i and f (hi ) is the probability density of hi . For
almost all practical environment, we assume f (hi ) is continuous and differentiable
in hi :
∞
∼ 1 −θi Ri (pi )
(θ
E Ci i ) = − ln e f (h i )dh i − ugi,0 pi . (12.12)
θi 0

It is apparent that ui (pi ,p−i ) is continuous in (pi ,p−i ). In addition, ugi,0 pi is

linear about pi , which does not affect the concavity of the equation. Based on
b b ∼
( a f (p, h)dh)p = a f p (p, h)dh, it is easy to prove that (∂ 2 E Ci (θi )/∂pi 2 ) ≤ 0. Thus,
∼
E Ci (θi ) is concave and the condition (2) is proved.
Therefore, the NPCG G = {N , {Pi }, ui (pi ,p−i )} admits an NE point.

12.4 Q-learning algorithm

As far as we know, FBS is installed by end users, who have not enough professional
skill to configure parameters of FBS. On this account, FBS should have self-learning
416 Applications of machine learning in wireless communications

ability to automatically configure and optimize the FBS’s operating information. In

Stackelberg learning game, every user in the network behaves as an intelligent agent,
whose goal is to maximize its expected utility. The game is repeated to achieve the
best strategy. Stackelberg learning framework has two hierarchies: (1) MU maximizes
its expected utility by knowing the response of all FUs to each possible game and
(2) given an MU strategy, FU performs a noncooperative game.
In this section, we will adopt the reinforcement learning mechanism based on
the Stackelberg game framework to achieve the energy-saving transmission power
allocation while ensuring the delay of QoS requirements.
To be compatible with reinforcement learning mechanism [13], the transmit
power of MS i is discretized as Pi = (pi,vi |vi = 1, 2, . . . , Vi ). The probability of MS
i choosing transmit power pi,vi at time slot t is πi,v t t
(πi,v ∈ πit ), and πit = (πi,v
t
|vi =
Vi i i i
1, 2, . . . , Vi ), which satisfies vi =1 πi,vi = 1.
t

Then, the expected utility of MS i is given by

ui (πit , π t−i )=E{ηi (p)|πit , π t−i }= ηi (p) t
πj,vj
, (12.13)
p∈P j∈N

where p = (p0,v0 , . . . pi,vi . . . , pN ,vN ) ∈ P is the actions of all MSs at time slot t, and
P = ×i∈N Pi .

12.4.1 Stackelberg game framework

The Stackelberg game model [33] is very suitable for two-tier femtocell networks,
where MS 0 is formulated as a leader, and MSs i (i ∈ N , i = 0) are modeled as
followers. In Stackelberg game framework, the leader can first know the strategy
information of all followers, then choose the action, and followers can receive the
leader’s strategy and then act.
Based on above analysis, it is easy to find that the goal of MS 0 is to maximize
its revenue as
max u0 (π0 , π −0 ), (12.14)
and the objective of MS i, (i ∈ N , i = 0) is
max ui (πi , π −i ). (12.15)
Because of this fact, FBS is deployed by end users randomly. There is no com-
munication or coordination between femtocells. They pursue their profits selfishly.
Equation (12.10a)–(12.10d) can be modeled as a noncooperative power allocation
sub-game G = [{i}, {Pi }, {ui }] (i ∈ N , i = 0).

Theorem 12.2. Given MS 0’s strategy π0 , there exists a mixed strategy {πi∗ , π ∗−i }
satisfies:
ui (πi∗ , π ∗−i ) ≥ ui (πi , π ∗−i ), (12.16)
which is an NE point.
Q-learning-based power control in small-cell networks 417

Proof. As it has been shown in [33], every limited strategic game has a mixed strategy
equilibrium, i.e., there exists NE(π0 ) for given π0 .

Lemma 12.1. The problem exists a Stackelberg equilibrium (SE) point {π0∗ , πi∗ , π ∗−i }
(∀i ∈ N , i = 0), which is a mixed strategy.

The proof of the existence of SE point is omitted here for brevity. We will employ
reinforcement learning mechanism, called Q-learning, to find SE point.

12.4.2 Q-learning
Based on reinforcement learning, each femtocell can be an intelligent agent with
self-organization and self-learning ability, and its operation parameters can be opti-
mized according to the environment. Q-learning is a common reinforcement learning
method, which is widely used in self-organizing femtocell networks. It does not need
teachers’ signals. It can optimize its operation parameters through experiments and
errors. Each BS acts as an intelligent agent, maximizing its profit by interacting
directly with the environment.
We define pi,vi ∈ Pi (∀i ∈ N ) as actions of Q-learning model, and π t−i (−i ∈ N )
are environment states. In a standard Q-learning model, an agent interacts with its
environment to optimize its operation parameters. First, the agent perceives the envi-
ronment and observes its current state s ∈ S. Then, the agent selects and performs
an action a ∈ A according to a decision policy π : s → a and the environment will
change to the next state s + 1. Meanwhile, the agent receives a reward W from the
environment.
In each state, there is a Q-value associated with each action. The definition of a
Q-value is the sum of the received reward (possibly discounted) when an agent per-
forms an associated action and then follows a given policy thereafter [34]. Similarly,
the optimal Q-value is the sum of the received reward when the optimal strategy is
followed. Therefore, the Q-value can be expressed as
Qπt (a, s) = W t (a, s) + λ max Qπt−1 (a, s + 1), (12.17)
a∈A

where W t (a, s) is the received reward when an agent performs an action a at the
state s in the time slot t and λ denotes a discount factor, 0 ≤ λ < 1. However, at the
beginning of the learning, the (12.17) has not been established. The deviation between
the optimal value and the realistic value is
Qπt (a, s) = W t (a, s) + λ max Qπt−1 (a, s + 1) − Qπt−1 (a, s), (12.18)
a∈A

Therefore, the Q-value is updated as the following rule:

Qπt (a, s) = Qπt−1 (a, s) + ρt Qπt (a, s)
= (1−ρt )Qπt−1 (a, s) + ρt [W t (a, s) + λ max Qπt−1 (a, s + 1)], (12.19)
a∈A

where ρt is a learning factor.

418 Applications of machine learning in wireless communications

Q-learning represents the knowledge by means of a Q-function, whose Q-value

is defined as Qit+1 (pi,vi , π t+1
−i ) and is updated according to

−i ) = Qi (pi,vi , π −i ) + α (ri (pi,vi , π −i ) − Qi (pi,vi , π −i )), (12.20)

Qit+1 (pi,vi , π t+1 t t+1 t t+1 t t+1

where α t ∈ [0, 1) is the learning rate. In (12.20), ri (pi,vi , π t+1

−i ) is the reward function of
MS i when selecting pi,vi and other MSs’ strategies are π t+1 −i . The relationship between
reward and utility function of MS i is

Vi
ui (πit , π t−i ) = πi,vi ri (pi,vi , π t−i ). (12.21)
vi =1

Each BS updates its strategy based on Boltzmann distribution [14], which is

formally described as
exp(Qit (pi,vi , π t+1
−i )/τ )
t
πi,vi
= Vi t t+1
, (12.22)
vi =1 exp(Qi (pi,vi , π −i )/τ )

where τ (τ > 0) is temperature parameter. Higher value of τ causes the probabilities

of all actions of MS i to be nearly equal; lower value of τ leads to the probability of
actions’ bigger difference with respect to their Q-values.

12.4.3 Q-learning procedure

In this section, we will study the QoS aware power allocation in sparse and dense
deployment of femtocell networks. The Q-learning mechanism based on Stackelberg
game framework is adopted.

12.4.3.1 Sparsely deployed scenario

In sparsely deployed femtocell networks, for example, in rural areas, the interference
between FBS is negligible due to path loss and penetration loss.
As we have assumed before, MBS knows complete strategies of all FBSs and
updates its Q-value by (12.20). The reward function of MS 0 is the following:

−0 ) = {η0 (p)δ−(0,v },
t+1
r0 (p0,v0 , π t+1 0)
(12.23)
p∈P

where t+1
δ−(0,v0)
= t+1
πj,vj
denotes the probability of actions vector p−(0,v0 ) =
j∈N , j =0
(p1,v1 , . . . pi,vi . . . , pN ,vN ).
For MS i (∀i ∈ N , i = 0), due to the fact that FBSs can receive MBS’s transmit
power strategy, and there is no interference between FBSs, the reward function of
MS i is

V0
ri (pi,vi , π0t+1 ) = t+1
δ−(i,v η (pi,vi , p0,v0 ),
i) i
(12.24)
v0 =1

t+1
where δ−(i,vi)
= π0,v
t+1
0
.
Q-learning-based power control in small-cell networks 419

12.4.3.2 Densely deployed scenario

In densely deployed femtocell networks, such as in urban areas, FBS is close to each
other and the interference between with each other cannot be ignored.
In this scenario, the reward function of MS 0 is according to (12.13). The reward
function of MS i (∀i ∈ N , i = 0) in this scenario is

V0
ri (pi,vi , π0t+1 ) = t+1
δ−(i,v η̂ (pi,vi , p0,v0 ).
i) i
(12.25)
v0 =1

Since there is no communication or cooperation between FBSs, if the selected

power level at time shot t + 1 satisfies pt+1 i,vi = pi,vi , η̂i (pi,vi , p0,v0 ) is estimated by
t+1

(12.20), else η̂it+1 (pi,vi , p0,v0 ) = η̂it (pi,vi , p0,v0 ):

ηi (pi,vi ,p−i )−η̂it (pi,vi , p0,v0 )
η̂it+1 (pi,vi , p0,v0 ) = ρ t (pi,vi ,p0,v0 )+1
+ η̂it (pi,vi , p0,v0 ). (12.26)

i,vi = pi,vi , which can be calculated

In (12.20), ηi (pi,vi , p−i ) is the real value when pt+1
by the feedback information from FBS Bi . ρ t (pi,vi , p0,v0 ) is the times number when
the MS 0s transmit power is p0,v0 , and MS i selects power level pi,vi until time shot
t [14].

12.4.3.3 Distributed Q-learning algorithm

Theorem 12.3. The proposed algorithm can discover a SE mixed strategy.

Due to the limited space, the convergence of the proposed algorithm can be found
in [35]. As the Algorithm 12.1, a distributed Q-learning algorithm is proposed.

Algorithm 12.1: Distributed Q-learning algorithm

Step 1: Initialization: for t = 0, Qit (pi,vi , π t−i ), ∀i ∈ N ;

power discretization: pi = (pi,1 , . . . , pi,vi , . . . , pi,Vi );
Learning:
Step 2: Update t = t + 1;
Step 3: Update πit according to (12.22);
Step 4: Update MS 0s transmit power according to p0,v0∗ = arg max (Q0t (p0,v0 , π t−0 )),
and send the value of π0t to FBS.
Step 5: Update MS i’s (i = 0) transmit power according to
pi,vi∗ = arg max (Qit (pi,vi , π t−i )), and send the value of πit to MBS.
Step 6: Calculate MS 0s reward according to (12.23), calculate MS i’s (i = 0) reward
by (12.25).
Step 7: Update MS i’s Q-value by (12.20).
Step 8: Back to Step 2.
End learning
420 Applications of machine learning in wireless communications

12.4.4 The proposed BDb-WFQA based on NPCG

In the NPCG G = {N , {Pi }, ui (pi ,p−i )}, the strategy set Pi = [pmin max
i , pi ] is continu-
ous, which is not applicable in the Q-learning method. To be compatible with the
Q-learning method, we discretize the continuous power set pi ∈ Pi = [pmin i , pi
max
] [14]
as following:

ai ai max
pi (ai ) = 1 − pmin
i + p , (12.27)
Mi Mi i
where ai ∈ Ai = {0, 1, . . . , Mi } and Ai is the set of FBS Bi’s action space. The number
of action space is Mi + 1.
Thus, the NPCG G = {N , {Pi }, ui (pi ,p−i )} transforms to the discrete game Gd =
{N , {Ai }, ui (pi (ai ),p−i )}. Based on the discrete game Gd , we design an appropriate
Q-learning algorithm to achieve the EC-based power allocation for FBSs.
According to the Q-learning theory, agent, state, and action can be defined as
follows:
Agent: All of the FBSs Bi . As stated in Section 12.2, there is only one sched-
uled active FU in each FBS during each signaling slot. Therefore, i ∈ N =
{1, 2, . . . N }.
State: t−1
FBS Bi’s policy 2πi t−1and thet−1receivedt−1interference of FU i Iit =
j =i gj,i pj + g0,i p0 + σ . πi = (πi,0 , . . . , πi,ai , . . . , πi,Mi ) is a probability
t−1

t−1
vector, where πi,a i
is the probability with which FBS Bi chooses action ai at
time t − 1.
Action: Each discrete transmit power can be denoted by each action ai . Therefore,
we use action ai ∈ Ai to replace the FBS Bi’s transmit power. According to
policy πit−1 , FBS Bi selects transmit power ai with probability πi,a t−1
i
.
The Q-value can be formulated according to the utility function of discrete
game Gd :
Qπt i (ai , si ) = W t (ai , si ) + λ max Qπt−1
i
(ai , si + 1) = πi,a
t−1 t
i
ui (pi (ai ), p−i ). (12.28)
a∈A

Therefore, we adopt the following rule to update Q-value:

Qπt i (ai , si ) = Qπt−1
i
(ai , si ) + ρt (A)[πi,ai
ui (pi (ai ), p−i ) − Qπt−1
t−1 t
i
(ai , si )], (12.29)
where ρt (A) is the learning factor. In practice, FBS Bi knows neither the opponents’
t−1
strategy π−i nor the true utility before running the action ai . But the FBS Bi can
compute the attainable utility uit (pi (ai ), p−i ) through the feedback information of the
receiver; thus, we design the following learning factor to estimate the utility:
⎧
⎪ 1
⎪
⎨ t−1 if A = ai ,
tπi,ai
ρt (A) = (12.30)
⎪
⎪ α
⎩ otherwise.
t+α
t−1
where α is the filter parameter. Notice that the tπi,ai
is approximately equal to the
frequency of FBS Bi selecting the action ai until time t. Therefore, the Qπt i (ai , si )
Q-learning-based power control in small-cell networks 421

is the approximation of FU i s expected utility when FBS Bi adopts the action ai .

Additionally, α/(t + α) decreases with the increase of the time slot and

α ≥ 0.5 if t ≤ α,
= (12.31)
t+α < 0.5 if t >α.

Therefore, we can believe that the α represents the weight of the historical learning
process and can speed up learning. Moreover, in order to ensure fast convergence, we
propose a weighted filter algorithm-based Boltzmann distribution [31] to update the
policy πit :
t−1
α2 exp(Qπt i (ai , si )/T ) t 2 πi,a
t
πi,a = + i
, (12.32)
i
t2 + α2 M i t
j=0 exp(Qπ ( j, si )/T )
t2 + α2
i

where T is the temperature parameter.

The convergence of the proposed algorithm is proved as follows. Because the NEs
exit in the NPCG and the action set of discrete game Gd is the discretized strategy
set of NPCG, there is at least one action ai ∗ at which the maximum Q-value Qπ∗i is
∗
attained [13]. Although there may be more than one optimal∞action ai , the maximum
∗
Q-value Qπi is unique. Additionally, we can prove the t=1 (α/(t + α)) = ∞ and
∞ ∗ ∗
t=1 (α/(t + α)) <∞ easily. According to [16], we achieve Qπi (ai , si ) → Qπi (ai , si )
2 t
∗ ∗
as t → ∞ with probability 1, where Qπi (ai , si ) denotes the optimal Q-value for
optimal action ai ∗ at state si .
The proposed BDb-WFQA algorithm is given in Algorithm 12.2.

Algorithm 12.2: The proposed BDb-WFQA algorithm

Step 1: Initialization: for t = 0;
Step 2: Select a0i = rand(0, Mi );
Step 3: Compute pi (a0i ) using (12.27);
Step 4: Compute the received interference Ii0 ;
Step 5: Compute ui 0 (pi (ai 0 ),p−i ) using (12.8);
Step 6: Initialize Qπ0 i (ai 0 , si ) = ui 0 (pi (ai 0 ),p−i );
exp(Qπ0 i ( j,si )/T )
Step 7: Initialize policy πi0 ,for the action j ∈ Ai ,πi,j0 = Mi 0
k=0 exp(Qπi (k,si )/T )
End Initialization
Step 8: Learning: for t = t + 1.
Step 9: Select ati = l according to πit−1 , l ∈ Ai ;
Step 10: Compute pi (ati ) using (12.27);
Step 11: Compute the received interference Iit of FU i;
Step 12: Compute uit (pi (ai t ), p−i ) using (12.8);
Step 13: Update the Q-value using (12.19);
Step 14: Update the πit using (12.22);
End learning
422 Applications of machine learning in wireless communications

12.5 Simulation and analysis

12.5.1 Simulation for Q-learning based on Stackelberg game

In this section, we will introduce the simulation of the proposed algorithm and simulate
a CMAQL algorithm to compare with the proposed algorithm [31]. Macro-users and
micro-users are distributed randomly in the two-tier femtocell networks and share
the same spectrum with w = 200 kHz. The channel-fading is modeled as Rayleigh
block-fading channels, the fading-block duration T = 1 ms. Noise spectral density
is N0 = −174 dBm/Hz. The channel gain for macro-user and femto-users are λL−3
and λL−4 , respectively, where L is the transmitter–receiver separation in meters, and
λ = 2 × 10−4 [36].
The additional circuit power pc is 10 dBm for all users, the lower bound of
transmit power for each user is pmin = 10 dB m, and upper bounds for femto-users
and macro-user are pmax = 20 dB m and pmax = 30 dB m, respectively. The transmit
power region [pmin , pmax ] is divided into d parts equally in the Q-learning procedure,
and we consider d = 3, 10, 20, respectively, in the simulation.
Figure 12.3 shows expected utilities with respect to the QoS exponent. When the
value of θ is small, i.e., θ ≤ 10−4 , there is no significant expected utility change. This
is because the smaller the QoS index, the looser the delay requirements, and the EC
is close to Shannon capacity, regardless of the arrival rate and delay requirements.
Instead, when the value of θ is larger, and the delay requirement is tighter, EC and
expected utility decrease correspondingly. On the other hand, the discretization of
transmit power results in the best transmit power error, while the smaller of d will
lead to a higher expected utility loss.

τ = 0.001, α = 0.5

7×107 d=3
d = 10
6×107 d = 20
Expected utilities (bit/J)

5×107

4×107

3×107

2×107

1×107

0
10–6 10–5 10–4 10–3 10–2 10–1 100 101
Delay exponent, θ

Figure 12.3 Expected utilities versus different QoS exponent

Q-learning-based power control in small-cell networks 423

Figures 12.4 and 12.5 show the convergence of the proposed algorithm. From
these figures, we can see that the proposed algorithm has faster convergence speed
than CMAQL algorithm. The reason is that micro-users in the proposed Q-learning
mechanism can share transmit power strategy with macro-user, while the value of
t+1
δ−(i,vi)
is estimated by only the past experiences in CMAQL algorithm.

θ = 0.001, τ = 0.001, α = 0.5, d = 10

7×107

6×107
Proposed-algorithm
Maximal Q-value (bits/J)

CMAQL-algorithm
5×107

4×107

3×107

2×107

40 50 60 70 80 90 100
Time shot, t

Figure 12.4 The convergence of Q-learning mechanism

θ = 0.001, τ = 0.001, α = 0.5, d = 10

Proposed algorithm MS2

Proposed algorithm MS3
CMAQL algorithm MS2
CMAQL algorithm MS3
Expected utility (bit/J)

108

107
20 40 60 80 100 120
Time shot, t

Figure 12.5 The convergence of expected utilities

424 Applications of machine learning in wireless communications

12.5.2 Simulation for BDb-WFQA algorithm

This section shows the performance of the proposed algorithm through numerical
simulations. We consider a two-tier femtocell network in which a macrocell is overlaid
by three cochannel deployed femtocells, which is similar to the scenario in [15]. The
related simulation parameters, as shown in Table 12.1, note that channel fading is
considered to be independent. And the Rayleigh block fading is also allocated. The
path loss of MUs and FUs is kd −3 and kd −4 , respectively, where d is the distance
from the transmitter to receiver and k = 2 × 10−4 [36].
The convergence of the proposed BDb-WFQA is shown in Figure 12.6. For
the comparison purpose, two other algorithms are also simulated. The first one is
the NGb-PCA. The second one is the hierarchical reinforcement learning algorithm
(HRLA) in [15], which employs the discrete power as action profile and chooses

Table 12.1 Simulation parameters

Parameter Value

×105
8

7
Average Q-value of FUs (bit/s)

6 HRLA
NGb-PCA
5 BDb-WFQA

0
0 5 10 15 20 25 30 35 40 45 50 55 60
Time slot, t

Figure 12.6 The convergence of algorithms

Q-learning-based power control in small-cell networks 425

action through the Boltzmann distribution. From Figure 12.6, after 20 iterations, the
BDb-WFQA Algorithm is stable, which guarantees the convergence of the algorithm.
In addition, we find that compared with NGB-PCA and HLA, the proposed BDB-
WFQA has faster convergence. This is because the proposed BDb-WFQA employs
the discrete power as action profile and uses the weighted filter way to update the
policy where the filter parameter α can be considered as a believable parameter to
accelerate learning.
The average EC of FUs is shown in Figure 12.7. It is can be observed that the
average EC of FUs reduces with the increase of delay QoS exponent θ for both the
NGb-PCA and the proposed BDb-WFQA. This is because a larger θ means a more
stringent delay requirement. In addition, we find that the performance of the proposed
BDb-WFQA is slightly lower than that of NGB-PCA. This is because the proposed
BDb-WFQA uses discrete action contours, but it may lose correct power values.
However, as mentioned earlier, we know that the proposed BDb-WFQA converges
faster than NGb-PCA.
The average EC of MUs is shown in Figure 12.8. From the five curves in the
Figure 12.8, it can be observed that the average EC of MUs increases with the increase
of μ. Besides, we can see that when the pricing factor μ = 0, the average EC of
MUs is the smallest. This is because that μ = 0 means there is not interference
constraint at FBSs’ side, the FBSs will choose the optimal transmit power to self-
ishly increase their EC, which will cause severe cross-tier interference to macrocell.
When the μ ≥ 170 dB, MUs gain the largest average EC. This is because the suffi-
ciently large pricing will make the FBSs choose the smallest transmit power; thus,
the cross-tier interference each MU received is smallest, and the achievable average

× 105
14
NGb-PCA
BDb-WFQA
12
Average EC of FUs (bit/s)

0
10–7 10–6 10–5 10–4 10–3 10–2 10–1
Delay QoS exponent, θ

Figure 12.7 The average EC of FUs versus delay QoS parameter θ

426 Applications of machine learning in wireless communications

× 105
2.3

2.2
Average EC of MUs (bit/s)

2.1
μ=0
μ = 150 dB
2
μ = 160 dB
μ = 170 dB
1.9 μ = 175 dB

1.8

1.7

1.6
0 5 10 15 20 25 30 35 40
Time slot, t

Figure 12.8 The average EC of MUs

EC is largest. Therefore, we can choose a pricing factor which can guarantee that
the received cross-tier interference of MUs is acceptable, and the FBSs can achieve a
good EC performance.

12.6 Conclusion

We investigate the energy efficient power control in two-tier femtocell networks

with considering delay-QoS guarantee. In order to enhance FBS’s ability of self-
configuration and self-optimization, we propose a Q-learning mechanism based on
Stackelberg game framework. In the learning procedure, macro-user is a leader, who
knows transmit power strategies of all femto-users and chooses power level first;
while femto-users acting as followers can communicate with only the leader and
move subsequently. Finally, a distributed Q-learning algorithm based on Stackelberg
game is proposed to study the downlink power control problem in two-layer femto-
cellular networks with statistical delay QoS constraints and interlayer interference
constraints. We also design a network performance measure with the statistical delay
QoS provisioning based on the concept of EC. Then we model the power allocation
problem as a noncooperative game and verify the existence of NEs. In particular,
we adopt a Q-learning theory to achieve self-organizing ability of femtocells and
propose BDb-WFQA to realize power allocation of FBS. Simulation results show
that compared with CMAQL algorithm, the Q-learning mechanism based on Stack-
elberg game framework has faster convergence speed. Simulation results also show
that the proposed BDb-WFQA increases the achievable EC of MUs through the pric-
ing method and provides a delay QoS provision for MUs and FUs. Furthermore, the
Q-learning-based power control in small-cell networks 427

BDb-WFQA has a better performance in the convergence compared with NGb-PCA

and HRLA. In the future, we will continue to study wireless resource optimization
issues and further use game theory to ensure the QoS in wireless networks.

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Chapter 13
Data-driven vehicular mobility modeling
and prediction
Yong Li1 , Fengli Xu1 , and Manzoor Ahmed2

Vehicular networks have been recently attracting an increasing attention from both the
industry and research communities. One of the challenges in this area is the under-
standing of vehicular mobility and further propose accurate and realistic mobility
models to aid the vehicular communication and networks design and evaluation. In
this chapter, different from the current works focusing on designing microscopic level
models that are describing the individual mobility behaviors, we are exploring the use
of open Jackson queuing network frameworks to model the macroscopic level vehic-
ular mobility. The proposed intuitive model can accurately describe the vehicular
mobility, and further predict various measures of network-level performance. These
measures include the vehicular distribution and vehicular-level performance, such as
average sojourn time in each area and the number of sojourned areas in the vehicular
networks. Model validation based on two large-scale urban vehicular motion traces
reveals that such a simple model can accurately predict a number of system measure
concerned with the vehicular network performance. Moreover, we develop two appli-
cations to illustrate the proposed model’s effectiveness in the analysis of system-level
performance and dimensioning of vehicular networks.

13.1 Introduction
Recently, as more and more vehicles are equipped with multiple sensors and hetero-
geneous communication access devices to enable wireless connectivity, interests on
vehicular communications and networks have grown tremendously [1]. It is seen as the
key technology for improving road safety and building intelligent transportation sys-
tem (ITS) [2]. Many applications of vehicular networks are also emerging, including
automatic collision warning, remote vehicle diagnostics, emergency management and
assistance for safely driving, vehicle tracking, automobile high speed Internet access,
and multimedia content sharing. In USA, Federal Communications Commission has

1
Department of Electronic Engineering, Tsinghua University, China
2
Department of Computer Science, Qingdao University, China
432 Applications of machine learning in wireless communications

allocated 75 MHz of spectrum for dedicated short-range communications in vehicular

networks, and IEEE is also working on the related standard specifications. The aim
of these leading consortia and standardization bodies is to develop technologies and
protocols for information transmission between vehicles and roadside units (RSU),
infrastructures equipment, known as vehicles to infrastructures (V2I), and between
vehicles, known as vehicles to vehicles (V2V).
Urban vehicular ad hoc networks (VANETs) [3] are recognized as an important
component of the next generation ITS to alleviate serious problems, such as traffic
jams and accidents, as well as to enable new mobile applications to the public [1].
Since urban VANETs are highly mobile, it is difficult to maintain a connected and
stable network for communication. Thus, they are usually distributed, self-organized
by the mobile vehicles, characterized by very high velocity, and limited degrees of
freedom in nodes mobility patterns. This brings a strong interaction between the
vehicular mobility and network protocol design, which is the main focus of current
development of VANETs [3]. First, mobility in the macroscopic means the flows of
vehicular traffic directed from one region to another, which influences the spatial
distribution of vehicles, and the data traffic may also be altered by mobility. Thus,
a specific relationship between the mobility and wireless communication exists in
VANETs. Second, mobility in the microscope means the individual vehicular mobility,
which influences the position of each vehicle. Then, the communication rate changes
when the vehicles communicate with the RSUs [4] or vehicles [5] via V2I or V2V.
In terms of VANETs’ design, since the development of VANETs’ technologies
has huge impact on the automotive market, we should put a growing effort in the
development of communication protocols and mobility models by efficiently utilizing
their relationship and the influences of mobility on the communications, specific to the
vehicular networks. In terms of protocol and vehicular network system performance
evaluation, economic issues and technology limitations make theoretical analysis
and simulation as the prime choices in the validation of VANETs, and also as the
widely adopted first step in the development of real world technologies [5]. A critical
aspect in the theoretical analysis and simulation of VANETs is the need for a realistic
mobility model reflecting the real behaviors of vehicles in terms of both large-scale
vehicular traffic and microscope level of individual mobility. In conclusion, mobility
models are significant to the development of vehicular networks and related works
have become an important part on vehicular networks [5].
After a few years of exciting work, a large variety of mobility models are available,
which can be categorized in three different classes known as synthetic, survey-
based model and trace-based models. The synthetic models as their name implies
are obtained by mathematical models, while the survey models extract mobility pat-
terns through surveys, and finally the trace-based models generate mobility patterns
from real mobility traces [5]. These models vary from the most trivial to the most
realistic ones, or from freely available models to commercial vehicular simulator.
However, these models consider each vehicle as a distinct entity, and they are in the
microscopic level [6]. Although microscopic level models describe the individual
Data-driven vehicular mobility modeling and prediction 433

mobility behaviors precisely, unfortunately, they fail to capture the overall mobility
in the whole network. In contrast, macroscopic level description can lead to gross
quantities of metrics like vehicular distribution, density and means of velocity, by
treating vehicular traffic according to fluid dynamics, and then large-scale overall
vehicular behaviors and traffic can be easily revealed. Further, such models are indis-
pensable for network dimensioning, answering the “what if ” questions like how the
network performance changes or the deployed network evolves as the number of
vehicles or communication demands scale-up [7]. Thus, macroscopic level vehicular
mobility models are crucial for the development of vehicular networking protocols
and algorithms.
In this chapter, against this background, we consider the problem of modeling
the macroscopic level vehicular mobility. Specifically, we explore the use of an open
Jackson queueing network to model the vehicular mobility among areas divided by the
intersections of the city road. In the model, vehicles arrive in the system according
to a random process, move from one area to another area by making independent
probabilistic transitions, and finally depart the system. The question we address is
can this simple queueing network model accurately describe the vehicular mobility
and further predict various measures of network-level performance like the vehicular
distribution, and vehicular-level performance like average sojourn time in each area
and the number of sojourned areas in the vehicular networks. Our novel contributions
are summarized as follows:

1. We model the macroscopic level vehicular mobility as an open Jackson queueing

network. Under this model, we obtain three important metrics related to vehicular
mobility and system performance, which are vehicular area distribution, average
sojourn time in each area, and average mobility length.
2. Using two large-scale urban vehicular motion traces, we validate the accuracy of
the proposed queueing network model by comparing the model-predicted results
with the observations in the traces. The results reveal that such a simple model
can accurately predict a number of system metrics concerned with the vehicular
network performance.
3. Under the proposed open Jackson queueing network for vehicular mobility, we
introduce two specific applications. One is the decision of how much the capacity
of the RSUs should provide with the increase of the communication demand
coming with the increasing of city vehicles. The second is investigating the
performance of the combined communications of V2I and V2V. The applied
applications illustrate that the proposed model is effective in the analysis of
system-level performance and dimensioning of vehicular networks.

The rest of this chapter is organized as follows. After introducing related work
in Section 13.2, we give the model motivation and describe the system model in
Section 13.3. While in Section 13.4, we derive related system performance metrics
based on the proposed model. Moreover, in Section 13.5, we introduce the vehicular
434 Applications of machine learning in wireless communications

mobility trace for model simulation and provide the validation results, and followed by
two specific applications of vehicular network performance analysis in Section 13.6.
Finally, we conclude the chapter in Section 13.7.

13.2 Related work

In terms of individual mobility models in the microscopic level, after a few years
of exciting developments, a large variety of models are available. Different from
the synthetic models [8,9] and survey-based models [10,11], the trace-based model
try to extract mobility patterns from existing mobility trace by approximating the
movements based on observed movement patterns [12,13]. Even both the synthetic
and survey-based models are very complex, still they are not able to come close to
realistic modeling of motion patterns. All these microscopic level individual mobility
modeling approaches have the limitation of obtaining global mobility patterns instead
of precise movements; and they also sometimes are too complex to solve by math-
ematical equations. On the contrary to modeling the individual mobility, our work
focuses on the macroscopic mobility modeling. To our knowledge, this is the first
work that gives a simple queueing model of mobility with large-scale urban vehicular
mobility empirical data validation.
Recent works [14] focus on studying the metric of inter-contact time, which
denotes the time between two successive communication contacts of two vehicles,
and it finds that the inter-contact time exhibits the exponential distribution over a
large range of timescales. Poisson distributed contact rate has been validated to fit
well to real vehicular traces and is widely used to model opportunistic vehicular
systems [15,16]. Instead of studying inter-contact time, Li et al. [17] puts forward
another key metric known as contact duration, which is how long a contact lasts. In
contrast to these works, revealing the vehicular contact patterns that indirectly reflect
the macroscopic mobility, we directly model the vehicular mobility among areas and
reveal the vehicular mobility flow and its spatial distribution in direct manner.
Previous works on modeling and performance analysis with queueing network
model studied mostly the wired network and applications like peer-to-peer live stream-
ing systems [18,19]. The most closely related works are theoretical analysis for cellular
and Wi-Fi networks [7,20,21]. Ashtiani et al. [20] used a closed queueing network
with fixed nodes to model the users and traffic in the cellular network, while Kim
et al. [21] utilized M/M/c/c queues to model cellular network mobile users. Simi-
larly, Chen et al. [7] proposed a mixed queueing network model to describe the user
mobility among access points in the campus wireless network environment. All these
models for wireless networks are proposed under different assumptions of mathe-
matical properties. In contrast, our work focuses on modeling the large-scale urban
vehicular mobility. Rather than giving complex mathematical deviation, we justify
that using the simplest open Jackson queueing model can capture the essential prop-
erties for vehicular mobility which are validated by two empirical traces. Moreover,
we introduce two typical applications, which show our proposed model is useful in
the vehicular network performance analysis and design.
Data-driven vehicular mobility modeling and prediction 435

13.3 Model

13.3.1 Data sets and preprocessing

Shanghai trace [14] was collected in SG project [22], in which 2,019 operational taxis
continuously covered the whole month of February 2007 without any interruptions
in Shanghai city. In this trace, a taxi sends its location coordinates by GPRS to the
central database every 1 min when it has passengers onboard but every 15 s when it
is vacant for the reason of real-time scheduling. However, the different intervals of
reporting may distort the records of the physical movements of the taxis, since most
of the taxis are not vacant most of the time. Another drawback of this trace is that
the number of taxis is limited. Indeed, 2,000 taxis and 1 min duration may not be
sufficient to record the statistical features of contact duration in a high-speed large
urban environment.
In collecting Beijing trace, we used the mobility track logs obtained from 27,000
participating Beijing taxis carrying GPS receivers during May 2010. The reason
behind choosing taxis as vehicular devices is that taxis are more sensitive to urban
environments in terms of underlying road topology, traffic control, and urban plan-
ning, and they have broader coverage in terms of space and operation time than that
of buses and private cars. Specifically, we utilized the GPS devices to collect the
taxis’ locations and timestamps, and further GPRS modules report the records every
15 s for moving taxis. The specific information contained in such a report includes
the taxi’s ID, the longitude and latitude coordinates of the taxi’s location, timestamps,
instant speed, and heading direction. Beijing trace is the largest vehicular data trace
available.
By collecting the GPS information of longitude and latitude coordinates, we
obtain the taxi’s moving traces that are locations varying with the time. Since these
locations are measured by the GPS devices, the noise may exist in the collected data
due to the inaccuracy of the GPS device. At the same time, since the taxies may not
report their location time at the same timeslots at the fixed frequency like in Shanghai
trace. Therefore, we need to process the data trace to get accurate locations of all the
taxies in the same frequency and timeslots. In order to achieve these goals, we first
use the city map of Shanghai and Beijing to correct the taxi’s locations based on the
coordinates of city road. Then, we use the method of linear interpolation (LI) to insert
location points to let all the taxies have a location information at every 15 s. For the LI
method, we first select any location that is near to each other in the original trace at two
time points. If their interval is larger than 15 s, we use the selected two time points and
map the information to estimate the unknown ones. For example, suppose we have
the location information of one taxi in the original trace at time t1 < t2 < · · · < tn ,
and their corresponding locations are l1 , l2 , . . . , ln . If we want to insert the location
information of time t that are calculated according to the time interval, we find m that
satisfies tm ≤ t < tm+1 . Then, we calculate the location by the following expression
through LI:
tm+1 − t t − tm
lt = lm · + lm+1 · . (13.1)
tm+1 − tm tm+1 − tm
436 Applications of machine learning in wireless communications
40
31.27

39.98
31.26

39.96
31.25

39.94 31.24

39.92 31.23

39.9 31.22

39.88 31.21

39.86 31.22

39.84 31.19

39.82 31.18
116.25 116.3 116.35 116.4 116.45 116.5 121.4 121.42 121.44 121.46 121.48 121.5

(a) (b)

Figure 13.1 City maps recovered from one data taxis mobility trace of (a) Beijing
and (b) Shanghai

In order to verify the above data preprocess approach does not introduce inaccurate
information to the original data trace, we use the obtained data of locations of 1 day to
plot the trajectory of all taxies, which are shown in Figure 13.1(a) and (b) for Beijing
and Shanghai data, respectively. From these two figures, we can see that our data set
is fine-grained that even using 1 day’s data can recover the map of the whole city. In
order to further show the accuracy of our data processing, we compare the obtained
figures with the original Beijing and Shanghai Map and find that all the trajectories
are in the city road and thus demonstrate that the map drawn by these trajectories are
very similar with the original city map.

13.3.2 Model motivation

Consider a vehicle moving in the city roads. It will travel along a road and then come
across an intersection. It may then wait for traffic signal for a while in the intersection
to choose the direction and travel to another road to drive on. In the downtown of a
city, the road is usually very crowded, and the intersections may be very dense, which
lead to very long waiting time at the intersections and short driving time along the
road. Therefore, the intersection is an important factor to model the urban vehicular
mobility. From the viewpoint of the whole city, we observe a large group of vehicles
waiting at the area of each intersection, and the streams of traffic moving from one
area to another area. That is to say, in order to describe the vehicular distribution, we
need to pay attention to the areas around the intersections and understand vehicular
behaviors of transition from one area to another from the system viewpoint.
Thus, if we divide the whole urban area into different areas that include at least one
intersection, we can model the vehicles moving from one area to another adjacent area
Data-driven vehicular mobility modeling and prediction 437

31.212
31.21
31.208
31.206
31.204
Latitude

31.202
31.2
31.198
31.196
31.194
31.192
121.44 121.445 121.45 121.455 121.46 121.465
Longitude

Figure 13.2 Illustration of area partition algorithm for a part of Shanghai city

and model the vehicular traffic transiting from one area to another. Taking Figure 13.2
as an example. We first marked the important intersections surrounded by a large
number of vehicles as the red point and then divide the whole area around the selected
intersections. The methods for the area partition can be chosen flexibly by the specific
applications. For example, if we want to adapt the model to the use in the vehicular
network design, i.e., deploying the RSU system, then the Voronoi diagram can be
used, where each point is divided into the intersection that it is nearest to and then
obtain the boundary of each area.
Let us consider the two-dimensional vehicular mobility defined by a sequence
of steps that a vehicle travels in the city road, which is modeled by the above areas
formed by the intersections. A step is denoted by a tuple (t 1 , t 2 , A), where A is the area,
t 1 is the time entering area A, and t 2 is the time that it departs the area. In the first step,
the vehicle enters the modeled region from the entering area, and after some step,
it moves out of the modeled region. Every vehicle moves in this way by transiting
from one area to another area. In this way, we can depict one vehicle’s mobility and
overall describe the traffic flows of the whole system by combining all the vehicles
and intersections together as a system. Now, we are ready to introduce our queue
network to model the above vehicle-mobility scenario.

13.3.3 Queue modeling

By the method of area dividing described above, the whole vehicular system can be
described by the partitioned areas. The number of areas is denoted by N , across which
438 Applications of machine learning in wireless communications

the vehicles transit, and the vehicles move into the system, move from one area to
another, and finally moving out the system. We use a queueing network to model
the above system, which is shown in Figure 13.3. It includes N server nodes with
infinite queue size, which models the N partitioned areas in the system. The servers
are denoted by set N = {A1 , A2 , AN }. The vehicular movement into the system and
moving from one area to another are modeled by the entrance into the queueing
network and the transition from one server to another.
Now, we describe the dynamic behaviors of vehicles moving in different areas
of the system. In such a vehicular mobility system with multi-areas, the vehicle
dynamic behaviors occur on two different timescales. One is on the long timescale, in
which the vehicle may enter and depart the system. The other is on the short timescale,
where a vehicle changes areas, which means it switches from one area to another. In
the viewpoint of queueing network model, the vehicles enter into the system with
certain rate, stay in the server’s queue, and then transfer into another server. For
the long timescale dynamics, we assume that vehicles arrive server n, n ∈ N with
rates of λn . When a vehicle moves to area n, it will continually stay in this area for
a period of time. We assume that the average amount of time that a vehicle stay in
area n is μn . The distribution of staying time is arbitrary. For the short timescale
dynamics, we consider that after the vehicle staying in area n for a random period
of time, it switches to another area m with probability pnm or leaves the system with
probability pn0 . In this way, the vehicles move from one area to another, depart or enter
the system.
We have modeled the vehicular mobility system including N areas as an open
network of N servers with infinite queues. In such an open system, vehicles freely
join and leave the system. The exogenous arrival rate for server n is λn . After staying
in the queue of server n for time period of 1/μn , it will leave the queueing network
with probability pn0 or switch to other server like m with probability pnm and denote
the switching matrix as P. Therefore, for server n, its load is denoted by ρn = λn /μn .

A2
p2n
p12

pn2
p21
p1n

A1 An

pn1

λn pn0

Figure 13.3 Illustration of the queue model for vehicular mobility

Data-driven vehicular mobility modeling and prediction 439

Table 13.1 Parameters and notations used in the queueing

network model

Notations Meaning

N The set of areas, or the servers set

N The number of servers in the system
λn The exogenous arrival rate for server n
1/μn The expected vehicle residence time at area n
ρn The load of server n
M The number of vehicles in the system
pnm Probability of the vehicles moving from area n to area m
Wn The number of vehicles in area n

Since each server’s queue is considered as an infinite queue, and each user is served
immediately if we assume there are an infinite number of servers, and each vehicle is
independent of each other. Now, we summarize the key parameters and their notations
in our model in Table 13.1.
As we model the vehicular mobility system as the aforementioned open queueing
system, it would be very easy to get further results if it is an open Jackson network,
which has well-known results about the user distribution and waiting time. Related
to the Jackson network, we need to demonstrate that the exogenous arrival to each
server follows Poisson process. If it holds, the queueing network can be modeled
by a network with infinite server queue (i.e., M /G/∞). Thus, we need to study
the property of exogenous arrival rate in the system. By leveraging the Beijing and
Shanghai traces, we find that the actual exogenous arrival process of the vehicular
mobility matches well with the exponential distribution. Thus, the vehicular mobility
system can be modeled as an open Jackson network. Based on this model, we will
derive some important metrics to depict the system performance.

13.4 Performance derivation

Based on the open Jackson-queueing network model for the vehicular mobility, we
will investigate three important metrics related to the system performance, which are
vehicular area distribution, average sojourn time, and average mobility length. First,
we give formal definitions of above three metrics:
● Vehicular area distribution: The vehicular area distribution is defined as the steady
population probability distribution of all areas in the system. That is how the
vehicles of whole system are distributed in the areas. Basically, we can define it
as the separate distribution of each area and joint area distribution of all the areas.
Specifically, the vehicular distribution of separated area n is the probability that
there are wn vehicles in area n, which can be expressed as P(Wn = wn ), where
440 Applications of machine learning in wireless communications

wn = 0, 1, . . . , M , 1 ≤ n ≤ N , and Wn is the random variable denoting the number

of vehicles in an area n. Then, the joint area vehicular distribution, denoted by
→
π( wn ), can be expressed as follows:
→
π( wn ) = P(W1 = w1 , . . . , WN = wN ). (13.2)
● Average sojourn time: Average sojourn time is defined in the viewpoint of vehi-
cles, which is how long a vehicle will stay in the system. That is the time period
between that the vehicles entering into the region by one of the areas and that it
finally leaves the region. By averaging when all the vehicles in the system is in
a steady state, the average sojourn time. This parameter is related to the session
time of a vehicle visiting in the system.
● Average mobility length: Average mobility length is also defined in terms of
vehicles. That is the average number of areas that a vehicle will travel along when
it is in the system during the sojourn time. This parameter is related to the average
number of transitions of a vehicle during a session.

13.4.1 Vehicular distribution

Considering area n, the exogenous arrival rate for area n is λn . The vehicles switch
from one area to another area according to the matrix P. We treat N areas as an
open Jackson network of N nodes with infinite servers, and view each vehicle as a
customer that sojourns at node n for a random period of time with mean 1/μn , which
is the server time in node n. Then, we let γ = (γ1 , γ2 , . . . , γN ) be the effective arrival
rate vector for all areas in N . According to Figure 13.4, for any area, say n, we have

γn = λn + γn pjn , 1 ≤ n ≤ N . (13.3)
j =n

We express above expression in matrix form accordingly as

γ= λ + γ P, (13.4)
where P is the N × N area-switching matrix.
For area n, let ρn = πn /μn , and ρn is the load of area n. Actually, ρn is the average
number of vehicles in area n. That is to say, ρn is the expected number of vehicles in

pnj

λn
An Aj

pjn

pn0

Figure 13.4 Effective arrival rate of Ai

Data-driven vehicular mobility modeling and prediction 441

area n. Furthermore, we get the separated area vehicular distribution and joint area
vehicular distribution expressions by the following lemma:

Lemma 13.1. The vehicular distribution of the separated area n is

ρn wn e−ρn
P(Wn = wn ) = , (13.5)
wn !
and the joint area vehicular distribution is

→
N
ρj wj e−ρj
π ( wn ) = , (13.6)
j=1
wj !

and the expected number of vehicles that stay in the area n in dynamic vehicular
mobility system is ρn .

Proof. We consider one area, say n. The user arrival rate is γn and the serve time is
1/μn . We view this area as an infinite serve node. Therefore, according to the Jackson
networks, for the vehicular mobility system, we have

→
N
ρj wj e−ρj
π ( wn ) = P(W1 = w1 , . . . , WN = wN ) = . (13.7)
j=1
wj !

The marginal distribution for individual area is expressed as

ρn wn e−ρn
P(Wn = wn ) = . (13.8)
wn !

We note that the distribution of vehicles at the queue of node n follows Poisson
with mean ρn . Therefore, the expected number of vehicles that stay in area n is ρn ,
which proves the lemma.

13.4.2 Average sojourn time

Using the queueing model for the vehicular mobility, we can further analyze important
system metrics related to the mobility behaviors and properties. As introduced above,
now, we derive two metrics of average sojourn time and mobility length based on the
proposed queueing model of mobility.
First, recall the queueing server set is N = {A1 , A2 , AN }, which are the system
states that vehicles transfer from one to another, and P = {pnm , 1 ≤ n ≤ N , 1 ≤ m ≤
N } is the switching matrix of the probability that a vehicle moves into area m when
it is in area n. Note that we already define the probability that a vehicle leaves the
modeled region from area n with probability pn0 . In order to define a complete system
state, we add another system state A0 into N , namely, the state when the vehicle is
out of the region and denote the new system states set as M , which is M = N A0 .
442 Applications of machine learning in wireless communications

We define p0n as the probability that a vehicle move into the system through area n.
From the definition of pn0 and p0n , we can obtain the following expression for them:

N
pn0 = 1 − pnm , n = 1, . . . , N , (13.9)
m=1

γn
p0n = N , n = 1, . . . , N . (13.10)
m=1 γm
In order to better distinguish the transitions on states N and M , we refer P =
{pnm }, 0 ≤ n, m ≤ N as the system transition on states M and denote the sub-matrix
R as the transition among the areas on state N , where R = P{pnm }, 1 ≤ n, m ≤ N .
Now, we obtain the average vehicular sojourn time by the following theorem:

Theorem 13.1. Denoting the vehicular sojourn time in the system by S, we can obtain
the average sojourn time designated as E[S] by the following expression:

E [S] = p0n E [Tn ] , (13.11)
n∈N

where E [Tn ] is obtained by [E [T1 ] , . . . , E [TN ]] = T = (I − R)−1 U, I is an all-one

vector, and U = [1/μ1 , . . . , 1/μN ].

Proof. We denote Tn as the vehicular sojourn time in the system on the condition that
it is always in the system. That is to say, the vehicle will not move out of the region
when it stays in any area of n, n ∈ N . Considering the staying duration in area n, and
using the Jackson queue network model, we have

1
E [Tn ] = + pnm E [Tm ] . (13.12)
μn
m∈N

Define U = [1/μ1 , . . . , 1/μN ] and T = [E [T1 ] , . . . , E [TN ]], we can derive as

the following matrix form:

T = U + RT. (13.13)

By [23], we can obtain that (I − R) is reversible. Thus, we can have

T = (I − R)−1 U. (13.14)

Therefore, the average sojourn time E[s] is

E [S] = p0i E [Tn ] , (13.15)
n∈N

which proves the theorem.

Data-driven vehicular mobility modeling and prediction 443

13.4.3 Average mobility length

Based on the average sojourn time of Theorem 13.1, we can obtain the average
vehicular mobility length in the following theorem:

Theorem 13.2. The average vehicular mobility length, denoted by E [L], can be
expressed by

E [L] = p0n E [Tn ] , (13.16)
n∈N

where E [Tn ] is obtained by [E [T1 ] , . . . , E [TN ]] = T = (I − R)−1 I

Proof. Note that in the obtained sojourn time that a vehicle stays in the system when
it is already in the system. In (13.13), we can change the sojourn time to mobility
length just by setting the staying time in each area by 1. That is, U = [1, . . . , 1]. Then,
we obtain:

T = (I − R)−1 1. (13.17)

Therefore, the average sojourn time E[L] is

E [L] = p0n E [Tn ] , (13.18)
n∈N

which proves the theorem.

13.5 Model validation

By leveraging the two most largest urban vehicular mobility traces as introduced
earlier, we validate our proposed open Jackson queueing network model. The traces
record the location information, i.e., longitude and latitude of the vehicles during the
trace-collection period. Therefore, we first need to per-process the traces to fit our
model. Then, we validate our model by using the empirical trace data in terms of
following metrics: vehicular arrival rate, vehicular area distribution, average sojourn
time, and mobility length.

13.5.1 Time selection and area partition

In order to use the GPS vehicular trace to validate our model, we need to preprocess the
data on two different dimensions, i.e., time and location. In terms of time dimension,
since the traces record the continuous vehicular mobility trajectory of the whole day,
we need to select the period of time that is more stable in terms of numbers of the
vehicles in the system to observe. While in the location dimension, we need to partition
the urban map into areas according to the intersections and further decide which
vehicles belong to which area considering their longitude and latitude information.
444 Applications of machine learning in wireless communications

13.5.1.1 Area partition

In order to divide the vehicular mobility system into areas that they transit on, we
need to take the road and city structure into consideration. As mentioned before,
the intersection is the most important factor to model the urban vehicular mobility
and distributions. Therefore, we divide the system according to the position of the
intersections in the city roads. More specifically, we take the intersections as the
center of each area and then partition the roads into area according to the distance to
the intersections. We use Voronoi diagram to achieve this, which is a frequently used
method of decomposition of a given space [24]. In Voronoi diagram, we are given a
finite set of sites {p1 , . . . , pn } in the Euclidean plane. The site of pi ’s corresponding
Voronoi cell, denoted by Vi , consists of all points whose distance to pi is not greater
than their distance to any other sites. While using the Voronoi diagram to partition
the system region, we take all the intersections as the set of sites, denoted by I =
{p1 , p2 , . . . , pN }, and refer to all the points in the system region as L . We denote the
dli as the distance between point l, l ∈ L and site i, i ∈ I , which is the geometric
distance between point l and the point of intersection in site i. Then, the area for
site, i, is designated as Vi and expressed as Vi = {l|dli ≤ dlj , ∀j ∈ I \ {i}, ∀l ∈ L }.
According to the regulation introduced above, we obtain the boundary and partition
of the desired system region into different areas.
For example, we consider a part of Shanghai city, the area partitioned results
depicted via Voronoi diagram is shown in Figure 13.2. The blue points in the figure
are the records of vehicular trajectory, the marked red points are the intersections, and
the cells are the partition areas distinguished by the yellow curves. We observe that
the region is divided by different areas, and according to the coordinate information
of each vehicle, we can decide which area it belongs to. Consequently, the vehicular
mobility is modeled by the transitions from one area to another.

13.5.1.2 Observation period selection

Since we are interested in the time period when vehicles are most active and stable,
we look into the trace to investigate the time period when there are enough vehicles
in the system, and the system is relatively stable and stationary. In the urban city,
the vehicle traffic appears almost the same patterns varying from one day to another,
except the little difference between the working day and weekend. Thus, we can select
the suitable observation period by just investigating the aggregate vehicular arrival
rate in the timescale of 1 day.
We plot the aggregated vehicular arrival rate into the system of both Beijing and
Shanghai trace in Figure 13.5. By observing the curve of Beijing trace, we find the
arrival rate is very low from the midnight to 5 am, and it increases quickly during
6–9 am. Then, the arrival rate almost keeps in the same level during the daytime from
9 am to 7 pm. Following comes a rate-decreasing period. Similar arrival rate patterns
can be observed in the Shanghai trace, except the traffic also keeps in the very active
state during the period of 7 pm to the midnight. Combining the Shanghai and Beijing
traces’ results, our goals of finding the most active period for the vehicular mobility
system, we select the period of 9 am to 7 pm as the observation period as the data to
validate our proposed model. When processing the vehicular mobility trajectory, if a
Data-driven vehicular mobility modeling and prediction 445

30
Shanghai
25 Beijing

Vehicular arrival rate λ (No./s)

0
0 5 10 15 20
Time (h)

Figure 13.5 Average vehicular arrival rate to the system in timescale of 1 day for
Shanghai and Beijing trace

vehicle does not have records in any of the areas in the period of 10 min, we assume
that it has departed from the system and will take it as a new vehicle moving into the
system when it appears.

13.5.2 Arrival rate validation

We note that in the open Jackson network model, an important requirement is that
the exogenous arrival to each area should follow Poisson process. Therefore, we
need to validate this. In the open Jackson model, the arrival to each server is the
aggregated customer arrival process. Consequently, the aggregated arrivals is the
metric to validate the rationality of our proposed model. We validate the vehicu-
lar arrival rate in the timescales of not only 1 day but also all the days. First, we
investigate the aggregated arrivals of all vehicles in the system. Based on the intro-
duced method of area partition, we count the exogenous arrivals to each area. To
validate that the aggregated arrival rate follows Poisson process, we need to jus-
tify the arrival time that can be fitted well by the exponential distribution. We plot
the Complementary Cumulative Distribution Function (CCDF) of the arrival time
distribution obtained from Shanghai and Beijing traces in Figure 13.6. Since we
plot the results in linear-log scale, where the curves of exponential distribution will
become a straight line, we can observe that the arrival time may match the exponen-
tial distribution well. Furthermore, we use exponential distribution to fit the 90% of
the distribution, where the red curves are the exponential distribution and the blue
curves are the empirical curves. The goodness of fit is measured quantitatively by the
R-square statistics [25], which is defined as the percentage of the variation between
the empirical CCDF and the fitted distribution. We obtain that the average adjusted
R-square statistics is over 98% for both Shanghai and Beijing traces. This confirms
446 Applications of machine learning in wireless communications

100
Empirical curves
CCDF Exponential fittings

Beijing

Shanghai

10−1
0 2 4 6 8 10
Arrival time (s)

Figure 13.6 Exponential fittings for aggregated distribution of vehicular arrival

time of Shanghai and Beijing trace

the accuracy of the exponential distribution of the aggregated arrival time, which
indicates the aggregated arrivals follows Poisson process.
Second, in order to further validate the exogenous arrivals to each area following
Poisson distribution, we investigate the distribution of the exogenous arrival time
of each area in the timescale of 1 day and select the first 15 days for studying. To
measure the closeness of the Poisson distribution and empirical ones, we use the
Kolmogorov–Smirnov test (KS test) instead of CCDF fitting due to the large amount
of curves in each area of 15 days. The KS statistic can quantify the distance between the
empirical distribution function of the sample and the cumulative distribution function
of the theoretical distribution [26]. The smaller the KS statistic, the closer the two
distributions are. In our study, we set the significance level [26] of KS test to 0.01,
which means the confidence level is 99%. Figure 13.7 shows the goodness-of-fit
measured by the acceptance ratio of KS tests of each day by averaging the results of
all areas. From the results, we can observe that the acceptance ratio of Beijing trace
is above 90% except the second day, which have a relatively smaller vehicle mobility
records. With regards to the Shanghai trace, we note a good match between the model
distribution and empirical results, and the average acceptance ratios are around 80%,
which means the overall accuracy of the Poisson model is about 80%. Combing the
results of Shanghai and Beijing, we come to the conclusion that the exogenous arrivals
to each area can be accurately modeled by the Poisson distribution.
Now, we have completed the validation of Poisson-based exogenous arrivals’
accuracy in our model. Next, we focus on validating the results of vehicular distribu-
tion, sojourn time, and mobility length, which are important metrics obtained from
the open Jackson queuing network-based vehicular mobility model.
Data-driven vehicular mobility modeling and prediction 447

0.8
Ratio of passing KS tests

0.6

0.4

0.2
Beijing
Shanghai
0
5 10 15
Day

Figure 13.7 Results of KS tests passing ratio for the vehicular arrival rate in
timescale of 1 day for Shanghai and Beijing traces

13.5.3 Vehicular distribution

At the first glance, we would like to know how well the proposed queueing network
model matches the empirical results. We note that the direct metrics obtained from
the model is the vehicular distribution at each area. Therefore, we first investigate the
vehicular distribution by comparing the results obtained from the theoretical model
and empirical data.
By selecting the six most busiest areas from Beijing and Shanghai traces, we plot
the distribution of the number of vehicles in these areas by the empirical results and
also plot the model results obtained from Lemma 13.1. The results of Beijing trace are
shown in Figure 13.8, and that of Shanghai trace are shown in Figure 13.9, where the
blue solid lines are the model results and the red dotted lines are the empirical results.
From the results, it is quite obvious that our model results match the empirical results
well, which shows the accuracy of our proposed queuing network-based vehicular
mobility model.
In order to measure the closeness of the predicted results obtained from the
model and the empirical results of all areas, we use the distribution obtained in
Lemma 13.1 to fit the empirical curves of all the areas in both Shanghai and Beijing
traces. The goodness of fit is also measured quantitatively by the R-square statistics
[25], which is defined as the percentage of the variation between the empirical CCDF
and the fitted distribution. Figure 13.10 shows the adjusted R-square statistics of the
model distribution fittings, where the adjusted R-square statistics are computed with
448 Applications of machine learning in wireless communications

0.2
0.25
0.18
0.2
0.16
0.2
0.14
0.15
0.12
0.15
PDF

PDF

PDF
0.1
0.1
0.1 0.08

0.06
0.05 0.04
0.05
0.02

0 0 0
0 10 20 0 10 20 0 10 20
No. of vehicles No. of vehicles No. of vehicles

0.25
0.25

0.2
0.2
0.2

0.15
0.15 0.15
PDF

PDF

0.1
0.1 0.1

0.05 0.05 0.05

0 0 0
0 10 20 0 10 20 0 10 20
No. of vehicles No. of vehicles No. of vehicles

Figure 13.8 Vehicular distribution of six intersections in Beijing trace, where the
red and dotted curves are the empirical results obtained from the
trace, and the blue and solid curves are the theoretical results
obtained by our proposed model

MATLAB® Curve Fitting Toolbox. It can be seen from Figure 13.10 that the average
adjusted R-square statistics of over 90% areas for the Shanghai trace is larger than
98%, and that of over 90% areas for the Shanghai trace is larger than 95%. This
confirms the accuracy of the model-based prediction for vehicular distribution.
Data-driven vehicular mobility modeling and prediction 449

0.18 0.16

0.16 0.14 0.2

0.14
0.12
0.12 0.15
0.1
0.1
PDF

PDF

PDF
0.08
0.08 0.1
0.06
0.06

0.04 0.04
0.05

0.02 0.02

0 0 0
0 10 20 0 10 20 0 10 20
No. of vehicles No. of vehicles No. of vehicles

0.16
0.2 0.25
0.14 0.18

0.12 0.16 0.2

0.14
0.1
PDF

PDF

0.12 0.15
0.08 0.1

0.06 0.08 0.1

0.06
0.04
0.04 0.05
0.02
0.02
0 0 0
0 10 20 0 10 20 0 10 20
No. of vehicles No. of vehicles No. of vehicles

Figure 13.9 Vehicular distribution of six intersections in Shanghai trace, where the
red and dotted curves are the empirical results obtained from the
trace, and the blue and solid curves are the theoretical results
obtained by our proposed model

13.5.4 Average sojourn time and mobility length

In this subsection, we compare the theoretical results of mean sojourn time and mobil-
ity length predicted from our proposed model against the empirical results obtained
by the mobility traces, where the theoretical results is calculate by Theorems 13.1
and 13.2.
450 Applications of machine learning in wireless communications

0.8

0.6
CCDF

0.4

0.2
Beijing
Shanghai
0
0.8 0.85 0.9 0.95 1
R2

Figure 13.10 Distribution of the adjusted R-square statistics of the theoretical

results and empirical results

Table 13.2 Predicted and empirical results of average sojourn time and mobility
length of Shanghai trace

Mobility Average sojourn time Average mobility length

Number of vehicles Predicted and empirical
Predicted results Empirical results results

1,000 1,498.0 1,412.6 16.69

3,000 1,462.6 1,386.3 16.28
4,441 1,475.4 1,374.2 16.10

For testing the scalability of the accuracy of the proposed model, we vary the
number of evolved vehicles. Both for the Shanghai and Beijing trace, we sort the
vehicles according to their number of positions recorded in the GPS trace. We first
select vehicles that have at least 80% of record during the trace collection time, and
then put more and more vehicles into the system for testing. For Shanghai trace, we
selected different number of vehicles as 1,000, which have 80% of record, 3,000 and
4,441, which is the total number of vehicles in the trace. For Beijing trace, the set of
the number of vehicles as 3,000, 6,000, 10,000, and 28,590.
The results of average sojourn time and mobility length of predicted and empir-
ical results under the Shanghai and Beijing traces are shown in Tables 13.2 and
13.3, respectively. From the results of average sojourn time, we find that the pre-
dicted results match the empirical results very well for both Shanghai and Beijing
traces. Especially, when we only use vehicles with more completed record, the pre-
dicted results are very near to the empirical results. For example, in Shanghai trace,
Data-driven vehicular mobility modeling and prediction 451

Table 13.3 Predicted and empirical results of average sojourn time and mobility
length of Beijing trace

Mobility Average sojourn time Average mobility length

Number of vehicles Predicted and empirical
Predicted results Empirical results results

3,000 1,689.8 1,720.5 9.86

6,000 1,662.5 1,699.7 9.62
10,000 1,613.4 1,658.1 9.23
28,590 1,450.4 1,500.1 8.37

the average deviation between the predicted and empirical results is only 5.7% when
the number of vehicular is 1, 000; while in Beijing trace, it is only 1.8% when the
number of vehicular is 3, 000. With the increasing number of vehicles, although more
and more vehicles are with imperfect records, which will induce some errors into the
system and model, the accuracy of our model is also accepted. For example, the pre-
diction accuracy are higher than 6.9% and 3.4% when the number of vehicles are
4, 441 and 28, 590 in Shanghai and Beijing trace, respectively. In terms of average
mobility length, the predicted results comply with the theoretical results completely.
Consequently, we validate that our model is accurate enough to model the vehicular
mobility to obtain the average and stable system performance.

13.6 Applications of networking

In vehicular networks, regarding with random and bursty data traffic initiated from
vehicles, RSUs play as the gateways to the internet and to the infrastructure of other
systems, such as ITS. Vehicles transmit their Internet access requests and information
to RSUs, and RSUs then send responses to the Internet for querying the data and
information needed by vehicles. Therefore, deploying RSUs appropriately is signif-
icant to the performance of vehicular networks. On one hand, the capacity and the
number of deployed RSUs determine the capacity and service that can be provided to
the vehicular network. On the other hand, a large number of RSUs deployed with large
capacity mean more infrastructure cost. Therefore, the decision for RSU deployment
should depend on the demands of vehicles. Basically, in the large urban city, it is very
difficult to make such decisions due to the dynamics of vehicular traffic and ran-
domness of vehicle mobility. However, based on our proposed vehicular model, we
can obtain some fundamental results of the relationship with the RSUs capacity and
network performance. Using the proposed queuing network-based vehicular mobility
model, we will analyze how much RSUs’ capacity should be provided with the rise
in communication demands resulting from the increase of urban vehicles.
In reality, it is difficult to cover roads with enough RSUs so that each vehicle
on the road can always be connected to nearby RSU in terms of infrastructure cost.
452 Applications of machine learning in wireless communications

Instead, vehicle-to-vehicle communications by opportunistic contacts offer higher

bandwidth communication capacity for data transmission, which can be utilized
to form what is known as the opportunistic vehicular network or vehicular delay
tolerant network (VDTN) [27]. By exploiting the delay-tolerant nature of non-real-
time applications, service providers can delay and even shift the data transmission to
VDTN. Therefore, in vehicular networking, vehicle-to-RSUs communications (V2I)
and vehicle-to-vehicle communications (V2V) usually are combined to offer ser-
vices. In this situation, investigating the performance of the networks considering
the vehicular mobility and different approaches of communications is a very difficult
problem. As the second part of this section, we will study the application of using the
queueing model to investigate the performance of the combined communications of
V2I and V2V.

13.6.1 RSU capacity decision

Now, we investigate the problem that if the number of vehicles increases with the
development of economy and human demands, can the deployed RSUs handle the
emerging growth of the communication demand. In this case, we define the capacity
of the RSU as the maximum number of vehicles that can be served. Since the vehicular
system in terms of both vehicular traffic and communication traffic is dynamic, we
further define the RSU is overloaded if the time period, in which the number of served
vehicles is more than its capacity, exceeds 95% of the whole measurement time. That
is if the probability that the served vehicles, more than the maximum number, is below
95%, the RSU is not overloaded. By increasing the exogenous arrival rate λ, we look
at the case that what fraction of RSUs will be overloaded.
From both the Shanghai and Beijing traces, we select those vehicles having almost
completely recorded during the whole trace collection time, which is 3,000 taxis, and
obtain the original λ from the trace, increase it from λ to 5λ, and plot the results
in Figures 13.11 and 13.12 for Beijing and Shanghai traces, respectively. From the
results, we can observe that with the increase in RSUs’ capacity, the fraction of the
overloaded RSU decrease very fast, while with the increase of the vehicular exogenous
arrival rate, it also increases. For quantitative analysis, we examine the requirements
of the capacity of RSU needed when at least 95% RSUs are not overloaded. In Beijing
trace, we note that when the vehicular arrival rate is λ, the required capacity is about
10, on the other hand, when the arrival rate is increased by five times, i.e., 5λ, the
required RSU capacity is about 35, which is only 3.5 times compared with the case
when the arrival rate is λ. With regard to the Shanghai trace, we can obtain similar
results when we increase the arrival rate by five times from λ to 5λ, the required RSU
capacity increases about three times. This result is not obvious since the capacity does
not need to increase linearly with the arrival rate, which means the number of vehicles
in the system. Moreover, according to the obtained results shown in the figures, we can
decide the RSU deploying policy according to the network performance requirements
and investigation of deploying RSU equipment in terms of the cost.
Data-driven vehicular mobility modeling and prediction 453

1
5λ
4λ

Percentage of overloaded RSUs

0.8 3λ
2λ
λ
0.6

0.4

0.2

0
5 10 15 20 25 30 35 40
RSU capacity (C)

Figure 13.11 Fraction of overloaded RSU according to its capacity when

scaling-up the vehicular arrival rate for Beijing trace

1
5λ
4λ
Percentage of overloaded RSUs

0.8 3λ
2λ
λ
0.6

0.4

0.2

0
5 10 15 20 25 30 35 40
RSU capacity (C)

Figure 13.12 Fraction of overloaded RSU according to its capacity when

scaling-up the vehicular arrival rate for Shanghai trace

13.6.2 V2I and V2V combined performance analysis

Based on the queueing network model, we derive closed-form expressions for per-
formance metrics, such as the probability of network-wide satisfied areas and the
454 Applications of machine learning in wireless communications

average number of areas that satisfy the communication demands with V2I and V2V
communications.
We say satisfied area occurs when every vehicle in this area can receive the
demanded data rate. When the network is in the state of satisfying communication,
all vehicles in the network are satisfied. For a given vehicular network, it is hard
for the network to enjoy the satisfying communication all the time. Then, the metric
of the steady-state probability that the network is in satisfying communication is an
important parameter to evaluate the performance of the vehicular network. Another
important metric is the expected number of areas that are enjoying the satisfying
communication. Now we give more precise definitions for these two metrics.
The communication capacity index for area n, denoted by n (Wn ), is defined as
c n + pn
n (Wn ) = , (13.19)
dn (Wn )
where Wn is the number of vehicles in the area of n, cn is the communication capacity
of the deployed RSU in area n, pn is the capacity of the V2V communications in this
area, and dn (Wn ) is the demand of communication of vehicles in the area. Based on
n (Wn ), the probability that area n is enjoying the satisfying communication can be
defined as

AS n = P(n (Wn ) ≥ 1) . (13.20)

The probability of all the areas that are satisfying is defined as

PS = P(n (Wn ) ≥ 1, n = 1, . . . , N ) (13.21)

Similarly, the average number of satisfied areas is defined as

N
NS = P(n (Wn ) ≥ 1). (13.22)
n=1

Now, we consider area n and calculate the area satisfying probability ASn . In this case,
cn is the capacity of the deployed RSU located at the center of this area, pn is the V2V
communication capacity, which depends on the number of vehicles in this area. We
assume each vehicle can offer capacity of ui , then pn = i∈Wn ui . Assume vehicles
in area n need communication capacity of rn , then dn (Wn ) = Wn rn . Hence, ASn can
be expressed as

ASn = P cn + u i ≥ Wn rn . (13.23)
i∈Wn

In the reality vehicular system, the V2V communication capacity depends on

the wireless interface. Suppose there are two classes of vehicles, i.e., one class uses
Bluetooth and the other uses Wi-Fi to achieve the short range peer-to-peer commu-
nications. Therefore, one class of vehicles has low capacity of u j , and the other class
Data-driven vehicular mobility modeling and prediction 455

has large capacity of uk ; then, according to Theorem 13.1, the satisfying probability
of area n, ASn , is given by

ASn = P cn + u i ≥ Wn r n

i∈Wn

= P cn + u j Wnj + uk Wnk ≥ (Wnj + Wnk )rn

P Wnj = wnj P Wnk = wnk ×
∞ ∞

= j j
P cn +u wj n +uk wn ≥ rn Wnj = wnj , Wnk = wnk
k k

wn =0 wn =0 (wn +wn )
j k

j
k
n = wn P Wn = wn ×
j k
∞ ∞ P W (13.24)

= 1 cn + u wn + u wn ≥ (wn + wn )rn
j j k k j k
wn =0 wn =0
j k

∞ ∞ P(Wnj = wnj )P(Wnk = wnk )×
=

k =0 1 (rn − u )wn + (rn − u )mj ≤ cn
j j k k
j
wn =0 n
w
j j k
(ρn )wn (ρnl )wn −ρnj −ρnk
= j w !k e
j wn ! n
0≤(rn −u j )wn +(rn −uk )mkj ≤cn

Consequently, we can obtain the probability of the network-wide satisfying PS

and the number of satisfying areas NS as follows:
PS = P(n (Wn ) ≥ 1, n = 1, . . . , N )

N
N
= P(n (Wn ) ≥ 1) = ASn ; (13.25)
n=1 n=1

N
N
NS = P(n (Wn ) ≥ 1) = ASn . (13.26)
n=1 n=1

Based on the above derivation, we set a vehicular network environment to observe

the performance. For the considered system, we assume there are 20 areas. The RSU’s
communication capacity of each area is set to be uniformly distributed, i.e., [500,
12,000] bps by considering the capacity provided by usually deployed Wi-Fi or 3G/4G
base stations. The capacity of V2V communication of each vehicle is divide into two
class, one class having [750, 850] bps, and [50, 150] bps of low capacity vehicles.
Related to the load of each area, we use the data obtained from Beijing trace, select
most 20 areas with largest number of vehicles in the system, let half of vehicles are with
high capacity, and the other half are with low capacity. We set this area load settings
as ρ and increase the load by 3, 5, 7, and 9 times. In the vehicular side, we set their
communication by an exponential distribution with parameter of ϑ. Under the above
settings, we can obtain the vehicular networking system performance of area satisfying
probability, network-wide satisfying probability, and the number of satisfying areas.
By varying the mean of the communication demand of vehicles 1/ϑ, the results of
satisfying probability of the most loaded areas are shown in Figure 13.13. From the
results, we can see that the satisfying probability is near 100% when the average
demand is less than 300 bps. With the increase in demand, the satisfying probability
456 Applications of machine learning in wireless communications

0.95

0.9
ASn

0.85
ρ
0.8 3ρ
5ρ
0.75 7ρ
9ρ
0.7
200 250 300 350 400 450 500
Demand of communication rate

Figure 13.13 Area satisfying probability according to the demand of

communication rate

1
ρ
3ρ
0.8 5ρ
7ρ
9ρ
0.6
PS

0.4

0.2

0
200 400 600 800 1,000
Demand of communication rate

Figure 13.14 Network-wide satisfying probability according to the demand of

communication rate

decreases. The larger the load, the sharper the decreasing rate. Under these results,
we can decide how to deploy the RSU equipment according to the performance curve
and specific requirements. In terms of the network-wide performance, Figures 13.14
and 13.15 show the results of PS and NS. With the increase in average demand and
Data-driven vehicular mobility modeling and prediction 457

15
NS

ρ
10 3ρ
5ρ
7ρ
9ρ
5
200 400 600 800 1,000
Demand of communication rate

Figure 13.15 The number of satisfying areas according to the demand of

communication rate

area load, we can see both PS and NS decrease. In this case, we can use the related
results to design the network system according to the requirements and decide how
to deploy infrastructure and RSU devices supporting V2V communications.

13.7 Conclusions
In this chapter, we used the open Jackson queueing network to model the macro-
scopic level vehicular mobility. Our proposed simple model can accurately describe
the vehicular mobility and predict various measures of network-level and vehicular-
level performance. Based on two large-scale urban city vehicular motion traces, we
validated the accuracy of our proposed model. Finally, we proposed two applica-
tions as an example to illustrate our proposed model effectiveness in the analysis of
system-level performance and dimensioning of vehicular networks.

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Index

A3C algorithm: see actor–critic (AC) BP de-noising (BPDN) 201

algorithm batch algorithms 111, 130
action-value function: see Q-value matrix completion 116–24
function neural networks 113–16
active learning 126, 131 support vector machine (SVM)
query by committee (QbC) 126 111–13
side information 127 Baum–Welch (BW) algorithm 170
actor–critic (AC) algorithm 53–6 -based modulation classifier 174–5
adaptive backoff algorithms 240–1 hidden Markov model (HMM)
adaptive projected subgradient method description for CPM signals
(APSM) 110, 119, 124–5, 131 173
additive white Gaussian noise (AWGN) numerical results 175
channels 99, 101, 170, 176,
comparison with approximate
189–93, 349, 412
entropy-based approach 176–7
alternating least squares (ALSs)
impact of initialization of
algorithm 117, 119
unknowns 177–8
alternating projection (AP) methods
performance with simulated
121–4
annealing initialization 178
angular spread of arrival (ASA) 87–8
problem statement 171–3
area partition 437, 444–5
arrival rate validation 445–6 Bayesian network method 312
artificial intelligence (AI) 68–9, 101, Bellman equations 43–5, 48–9, 51,
135–6, 160 53–4, 382–5, 388–9, 391,
artificial neural networks (ANN) 95, 393–5, 397, 399–400, 402–3
113–14 big data 1, 261
autoencoder 37–41 BiLoc system 355
autoencoder neural network 345–6 architecture 355–6
automotive sensors 228 off-line training for bimodal
average sojourn time 440–3 fingerprint database 356–8
and mobility length 449–51 online data fusion for position
estimation 358
back propagation (BP) algorithm 16, binary exponential backoff (BEB)
69, 97, 141, 346 algorithm 236–7
base station (BS) 145–6, 148, 371, 377, Bluetooth Low Energy (BLE) 346
408, 412, 417–18 BMCD (balanced multipath component
basis pursuit (BP) 201 distance) 85–7
462 Applications of machine learning in wireless communications

Boltzmann distribution-based weighted development of

filter Q-learning algorithm neural-network-based channel
(BDb-WFQA) 410 modeling 96–9
RBF-based neural network for
Caffe 18, 345 wireless channel modeling
Calinski–Harabasz (CH) index 74 99–101
cameras 228, 318–22 machine-learning-based MPC
carrier sense multiple access with clustering 68, 72
collision avoidance (CSMA/CA) improved subtraction for
protocol 226–7, 235–6, 238, cluster-centroid initialization
240 84–6
channel estimation 135 Kernel-power-density
channel model 137–9 (KPD)-based clustering 78–81
deep-learning-based channel KPowerMeans-based clustering
estimation 140 73–6
for massive MIMO CSI feedback MR-DMS (multi-reference
145–9 detection of maximum
for orthogonal frequency division separation) 86–8
multiplexing (OFDM) systems sparsity-based clustering 76–8
142–4 target-recognition-based clustering
EM-based channel estimator 149 82–4
time-cluster-spatial-lobe (TCSL)
basic principles of EM algorithm
clustering 82
149–52
propagation scenario classification
example of channel estimation
68–9
with EM algorithm 152–6
design of input vector 70–1
open problems 156
training and adjustment 71–2
in point-to-point systems 139
channel prediction based on
estimation of frequency-selective machine-learning algorithms
channels 139–40 109
channel impulse responses (CIRs) 76 channel measurements 110–11
channel modeling 67 learning-based reconstruction
automatic MPC tracking 69, 89 algorithms 111
extended Kalman filter-based batch algorithms 111–24
parameters estimation and online algorithms 124–5
tracking 92–3 optimized sampling 126
Kalman filter-based tracking 91–2 active learning 126–7
MCD-based tracking 89 with path-loss measurements
probability-based tracking 93–5 127–30
two-way matching tracking 90–1 channel state information (CSI) 138–9,
deep-learning-based 69, 95 145–6, 343, 371
algorithm improvement based on amplitude and phase, distribution of
physical interpretation 101–3 349
BP-based neural network for BiLoc system 355
amplitude modeling 96 architecture 355–6
Index 463

off-line training for bimodal relative core merge clustering

fingerprint database 356–8 algorithm 27–9
online data fusion for position sparsity-based 76–8
estimation 358 target-recognition-based 82–4
deep learning for indoor localization time-cluster-spatial-lobe
345 (TCSL)-based 82
autoencoder neural network 345–6 cluster pruning 74–5
convolutional neural network clusters, defined 20, 68
346–7 cognitive radios, signal identification in
long short-term memory 348 159
experimental study 359 continuous phase modulation
2.4 versus 5 GHz 362 classification in fading channels
impact of parameter ρ 362–4 via Baum–Welch algorithm 170
location estimation, accuracy of BW-based modulation classifier
360–2 174–5
test configuration 359–60 comparison with approximate
future directions and challenges entropy-based approach 176–7
new deep-learning methods for HMM description for CPM signals
indoor localization 364 173
impact of initialization of
secure indoor localization using
unknowns 177–8
deep learning 365
performance with simulated
sensor fusion for indoor
annealing initialization 178
localization using deep learning
problem statement 171–3
364–5
modulation classification in
hypotheses 350–5
multipath fading channels via
preliminaries 348–9
expectation–maximization 162
Chen et al.’s method 210–11
modulation classification via EM
classification and regression tree 163–7
(CART) 5–7, 19 numerical results 167–70
classification task 2, 19, 56, 111, 179 problem statement 162–3
CLEAN 83, 89 specific emitter identification via
closed subscriber group (CSG) mode machine learning 178
411 feature extraction 181–5
cluster-centroid initialization, improved numerical results 189–94
subtraction for 84–6 support vector machine (SVM),
clustering 20–1, 40, 73–4 identification procedure via
cluster-centroid initialization 84–6 185–9
density-based spatial clustering 23–4 system model 179–81
by fast search and find of density communication, types of 229
peaks 24–7 Complementary Cumulative
Kernel-power-density-based 78–81 Distribution Function (CCDF)
KPowerMeans-based 73–6 445–7
MR-DMS clustering 86–8 compressed sensing (CS)-based
RECOME clustering 30 methods 145
464 Applications of machine learning in wireless communications

compressive data gathering (CDG) continuous phase modulation

scheme 214 classification in fading channels
compressive sensing (CS) 197, 200–1 via Baum–Welch algorithm 170
-based wireless sensor networks Baum–Welch (BW)-based
(WSNs) 211 modulation classifier 174–5
compressive data gathering hidden Markov model (HMM)
213–14 description for CPM signals
localization 217–18 173
reduced-dimension multiple access numerical results 175
216–17 comparison with approximate
robust data transmission 211–13 entropy-based approach 176–7
impact of initialization of
sparse events detection 214–16
unknowns 177–8
equivalent sensing matrix, conditions
performance with simulated
for 202
annealing initialization 178
mutual coherence 203–4
problem statement 171–3
null space property 202 control channel behaviour (CCH) 233
restricted isometry property 202–3 conventional drive test 110
optimized sensing matrix design for convex optimization algorithms 204–5
CS 206 convolutional neural network (CNN)
Chen et al.’s method 210–11 16–18, 343, 346–7
Duarte-Carvajalino and Sapiro’s convolution layer 17–18
method 208 cooperative awareness messages
Elad’s method 206–8 (CAMs) 234
Xu et al.’s method 209–10 correlation-based (CB) algorithm 181,
representation error 199–200 184
signal representation 198–9 coverage map 109–11, 114, 116–20,
sparse recovery, numerical 124
algorithms for 204 Cramér–Rao Lower Bound (CRLB)
convex optimization algorithms 154, 156, 168
204–5 crowdsourcing applications 110
greedy pursuit algorithms 205–6 CsiNet 146–8
conjecture-based multi-agent C-support vector classification
Q-learning (CMAQL) algorithm (C-SVC) 322
410, 422–3 cumulative distribution functions (CDF)
350
connected vehicles architecture 227
curse of dimensionality 114
automotive sensors 228
cyclic prefix (CP) 142
electronic control units (ECUs) 227
intra-vehicle communications 228 data-driven vehicular mobility modeling
types of communication 229 and prediction 431
vehicular ad hoc networks 228–9 data sets and preprocessing 435
contact duration 434–5 model motivation 436–7
continuous phase modulation (CPM) model validation 443
161, 172 arrival rate validation 445–6
Index 465

average sojourn time and mobility development of neural-network-based

length 449–51 channel modeling 96–9
time selection and area partition RBF-based neural network for
443–5 wireless channel modeling
vehicular distribution 447–8 99–101
networking, applications of 451–2 deep learning for indoor localization
RSU capacity decision 452 345
V2I and V2V combined autoencoder neural network 345–6
performance analysis 453–7 convolutional neural network 346–7
performance derivation 439–40 long short-term memory 348
average mobility length 443 deep neural networks (DNNs) 13–14,
average sojourn time 441–2 114, 142
vehicular distribution 440–1 deep reinforcement learning (DRL) 18,
queue modeling 437–9 53–4, 364–5
related work 434 deep residual sharing learning 347
data-driven video-saliency detection delay spread 138
311–12 density-based spatial clustering of
Davies–Bouldin criterion (DB) 74 applications with noise
decision tree 4–9, 19 (DBSCAN) 23–4, 40–1
dedicated short range communication density estimation 20–1, 41
231
deterministic optimization problems
control channel behaviour (CCH)
371
233
dimensionality-reduction techniques
IEEE 802.11p 231–2
112, 114–15
message types 234
dimension reduction 20–1, 34, 41
WAVE Short Message Protocol
discrete cosine transform (DCT) 309
(WSMP) 232–3
deep autoencoder network 343, 346 discriminant ratio (FDR)-based
DeepFi 346 algorithm 181, 191
deep learning 19–20, 68, 136 distributed coordination function (DCF)
history of 141–2 234–5, 238
multilayer perceptron and 13–18 distributed denial-of-service (DDoS)
deep-learning-based channel estimation attacks 365
140 Doppler shift 138
for massive MIMO CSI feedback Duarte-Carvajalino and Sapiro’s method
145–9 208
for orthogonal frequency division
multiplexing (OFDM) systems Eckart–Young theorem 122
142–4 Elad’s method 206–8
deep-learning-based channel modeling electronic control units (ECUs) 227
approach 69, 95 empirical mode decomposition (EMD)
algorithm improvement based on 181
physical interpretation 101–3 energy entropy 183
BP-based neural network for enhanced distributed channel access
amplitude modeling 96 238–9
466 Applications of machine learning in wireless communications

Euclidean distance 4, 95, 103, Gaussian mixture models (GMM)

114, 329 29–32, 41
event-triggered messages 234 EM algorithm for 33–4
expectation–maximization (EM) general expression 137
algorithm 32, 146 generative adversarial network (GAN)
basic principles of 149–52 364
example of channel estimation with global positioning system (GPS) 217
152–6 gradient boosting decision tree (GBDT)
for GMM 33–4 7–9, 19
modulation classification in greedy pursuit algorithms 205–6
multipath fading channels via
162 heuristic video-saliency detection
modulation classification via EM 310–11
163–7 hidden Markov model (HMM) 161,
numerical results 167–70 171–4
problem statement 162–3 hierarchical reinforcement learning
extended Kalman filter-based algorithm (HRLA) 424, 427
parameters estimation and high-efficiency video-coding (HEVC)
tracking 92–3 standard 261–2, 267–8
eye tracking database 307, 312 for video-saliency detection: see
video-saliency detection
analysis on 313–17
higher density nearest neighbour (HDN)
on raw videos 312–13
25–7, 40
eye-tracking-weighted peak
high-resolution parameter estimation
signal-to-noise ratio (EWPSNR)
(HRPE) techniques 67, 82–3
267, 283
Hilbert–Huang transform (HHT) 181
empirical mode decomposition
facial-recognition technology 1
181–2
Fast search-and-find of Density Peaks Hilbert spectrum 181–4
(FDP) 24–6, 40–1 human visual system (HVS) 262, 307
femtocell base stations (FBSs) 407, 409
femtocell users (FUS) 407–8 IEEE 802.11p 225–6, 231–2, 241–3,
fifth generation (5G) 251–5
cellular-communication systems IEEE 802.11p medium access control
159 234
fine-grained indoor fingerprinting basic access mechanism 235–6
system (FIFS) 344 binary exponential backoff (BEB)
first- and second-order moments 183–4 algorithm 236–7
Fisher’s discriminant ratio-based DCF for broadcasting 238
algorithm 184–5 distributed coordination function
flat fading 138 (DCF) 234–5
frequency selective channel 138 enhanced distributed channel access
estimation of 139–40 238–9
fully connected (FC) layer 16 RTS/CTS handshake 237
fusion center (FC) 197 ImageNet 141
Index 467

Intel 5300 NIC 344–5, 349–51, 353, learning channel access control
355–6, 359 protocols 241–2
intelligent transportation system (ITS) least square (LS) 139
228, 240, 431–2, 451 Levenberg–Marquardt algorithm 116
inter-contact time 434 LightGBM 9
intra-vehicle communications 228 linear minimum mean square error
intrinsic mode functions (IMFs) 181 (LMMSE) estimator 140
in-vehicle communication 229 line-of-sight (LOS)/non-line-of-sight
inverse discrete Fourier transform (NLOS) scenarios 68, 70–2
(IDFT) 142 location estimation 344, 360–2
iterative algorithm 372–3 logistic regression 12–14, 19
iterative hard thresholding (IHT) 124 long short-term memory (LSTM) 348

Jackson queueing network model 433, macrocell base station (MBS) 407,
439, 443, 457 412, 418
Jensen’s inequality 151 macrocell users (MU) 410, 412, 414,
425–6
Kalman filter-based tracking 91–2
Manhattan distance 4
Kalman filters 69
Markov decision process (MDP) 42–5,
Keras 148
242, 371, 376, 378
kernel k-means 22–3
basic components of 378–81
Kernel-power-density (KPD)-based
finite-horizon MDP 381–2
clustering 78–81
infinite-horizon MDP
k-means 21–3, 40–1
with average cost 392–4
k-nearest neighbours method 2–4, 19,
119, 344 with discounted cost 387–9
Kolmogorov–Smirnov test (KS test) multi-carrier power allocation with
446 random packet arrival 389–92
KPowerMeans-based clustering 68, 73, matching pursuit (MP) 205–6
84, 102 matrix completion 116
clustering 73–4 alternating projection (AP) methods
cluster pruning 74–5 121–4
development 75–6 nuclear norm minimization-based
validation 74 methods 117–21
kriging-based techniques 110 maximum likelihood estimates (MLEs)
Kuhn–Munkres algorithm 33, 69, 95 149, 161–3, 171, 173
Kullback–Leibler (KL) divergence 310 max-pooling 17–18
McCulloch–Pitts (MP) model 135
Lagrange multiplier 123, 266, 269, MP neuron model 140–1
294, 324, 400, 403–4 mean squarederror (MSE) 148
Laplacian Kernel density 78 medium access control (MAC) 216–17,
learning-based reconstruction 226–7, 234, 238–41, 254, 371,
algorithms 109, 111 377
batch algorithms 111–24 MicaZ sensor platform 212
online algorithms 124–5 Middleton Class A model 193
468 Applications of machine learning in wireless communications

minimization of drive test (MDT) 110, extended Kalman filter-based

130 parameters estimation and
mobile stations (MSs) 229, 237, 411 tracking 92–3
model-free algorithms 54, 56 Kalman filter-based tracking 91–2
modulation classification in multipath MCD-based tracking 89
fading channels 162 probability-based tracking 93–5
modulation classification via two-way matching tracking 90–1
expectation–maximization multiple-input–multiple-output
163–7 (MIMO) systems 67–8, 145,
numerical results 167–70 409
problem statement 162–3 mutual coherence 203–4
Monte Carlo methods 46–8
MPCs distance (MCD) 74, 85 Nash equilibriums (NEs) 407
-based tracking 89–90 natural language processing (NLP) 1
MR-DMS (multi-reference detection of neighborhood radius 86
maximum separation) 86–8 network traffic congestion in wireless
vehicular networks 239
multi-hop communications, WSNs with
adaptive backoff algorithms 240–1
214
transmission power control 240
multi-kernel algorithm 125
transmission rate control 240
multilayer neural network: see
neural network (NN) 113–16, 141
multilayer perceptron
common architecture of 98
multilayer perceptron 13
see also specific entries
and deep learning 13–18
Noncooperative Game-based Power
multilayer perceptron (MLP) neural Control Algorithm (NGb-PCA)
networks 96,99, 114–15 411, 424–5
multipath component (MPC) clustering, noncooperative game theoretic solution
machine-learning-based 68, 72 414–15
improved subtraction for noncooperative power control game
cluster-centroid initialization (NPCG) 414–15, 420–1
84–6 non-safety communications 234
Kernel-power-density (KPD)-based nuclear norm minimization-based
clustering 78–81 methods 117–21
KPowerMeans-based clustering null space property 198, 202
73–6 Nyquist–Shannon sampling theorem
MR-DMS(multi-reference detection 197
of maximum separation) 86–8
sparsity-based clustering 76–8 observation period selection 444–5
target-recognition-based clustering Okumura–Hata’s model 113
82–4 on-board units (OBUs) 226, 229
time-cluster (TC)-spatial-lobe (SL) online algorithms 111, 124–5
(TCSL) clustering 82 open-source deep-learning frameworks
multipath components (MPCs) 67–9, 18
72, 75–82, 85–95, 103 Open Systems Interconnection model
automatic MPC tracking 69, 89 (OSI model) 226
Index 469

opportunistic vehicular network 452 learning-based approaches 266

optical character recognition model-based approaches 265–6
technology 1 minimizing perceptual distortion
OPTICS algorithm 24 with the RTE method 267
optimized sampling 126 bit reallocation for maintaining
active learning 126–7 optimization 274–5
channel prediction results with optimization formulation on
path-loss measurements 127–30 perceptual distortion 269–70
optimum cluster number 87 rate control implementation on
Orthogonal Frequency Division HEVC-MSP 267–8
Multiple Access (OFDMA) RTE method for solving the
networks 408–9 optimization formulation 270–4
orthogonal frequency division perceptual models 264
multiplexing (OFDM) channels automatic identification 264–5
344 manual identification 264
deep-learning-based channel single image coding, experimental
estimator for 142–4 results on 279
BD-rate savings, assessment of
packet delivery ratio (PDR) 239, 287–9
253–5 control accuracy, assessment of
parameter sharing 16 289–90
Parseval tight frame 210 generalization test 290–2
perceptron 9–18, 141 rate–distortion performance,
perceptual video coding 262, 264 assessment on 281–7
perceptual models 264 test and parameter settings 279–81
automatic identification 264–5 periodic safety messages 234
manual identification 264 phase spectrum of quaternion Fourier
video coding, incorporation in 265 transform (PQFT) 311, 331–2
learning-based approaches 266 pilot symbols 139, 143
model-based approaches 265–6 point-to-point systems, channel
perceptual video coding, estimation in 139–40
machine-learning-based 261 policy gradient methods 51–3
background 261–4 policy iteration 55
computational complexity analysis power angle spectrum(PAS) 70–2
275 PAS-based clustering and tracking
numerical analysis 278–9 algorithm (PASCT) 83–4
theoretical analysis 276–7 power delay profile (PDP) vector 76
experimental results on video coding principal component analysis (PCA)
292 34–7, 41, 111, 115
RC accuracy, evaluation on probability-based tracking algorithm
297–300 93–5
R–D performance, evaluation on propagation scenario classification
296–7 68–9
settings 296 design of input vector 70–1
incorporation in video coding 265 training and adjustment 71–2
470 Applications of machine learning in wireless communications

protocol performance 251 Q-value function 43, 47, 49, 51, 53–5
effect of data rate 254–5 approximation 55–6
effect of increased network density
252–4 radar sensors 228
effect of multi-hop 255–6 radial basis function (RBF) 72, 114,
simulation setup 251–2 345
Python environment 144 RBF-based neural network 99–101
RBF kernel function 72
Q-learning 49–50, 55, 242–3, 401–4, radio resource management (RRM) 371
417–18 radio waves 67, 126
MAC protocol 243 random forest (RF) 7–8, 20
action selection dilemma 243 rate–quantization (R–Q) model 265
a priori approximate controller Rayleigh fading 216, 385
244–6 received signal strength (RSS) 217, 343
convergence requirements 244 received signal strength indicator
implementation details 247–8 (RSSI) 347
online controller augmentation rectified linear unit (ReLU) activation
246–7 function 14, 143,347
procedure 418 recurrent neural network (RNN) 343
densely deployed scenario 419 recursive Taylor expansion (RTE)
distributed Q-learning algorithm method, minimizing perceptual
419 distortion with 267
sparsely deployed scenario 418 bit reallocation for maintaining
Q-learning-based power control in optimization 274–5
small-cell networks 407 optimization formulation on
noncooperative game theoretic perceptual distortion 269–70
solution 414–15 rate control implementation on
proposed BDb-WFQA based on HEVC-MSP 267–8
NPCG 420–1 for solving the optimization
simulation and analysis 422 formulation 270–4
simulation for BDb-WFQA reduced-dimension multiple access
algorithm 424–6 216–17
simulation for Q-learning based on compressive data gathering 213
Stackelberg game 422–3 multi-hop communications, WSNs
Stackelberg game framework with 214
416–17 single hop communications, WSNs
system model 411 with 213–14
effective capacity 413–14 robust data transmission 211–13
problem formulation 414 RefineNet 147–8
system description 411–12 region-of-interest (ROI) 262
Quadrature Phase Shift Keying (QPSK) regression task 2, 19
modulation 359 reinforcement learning (RL) 41, 53–6,
quality of service (QoS) 238, 407 394–6
query by committee (QbC) 117–20, deep reinforcement learning 50
126, 131 policy gradient methods 51–3
Index 471

value function approximation protocol performance 251

50–1 effect of data rate 254–5
Markov decision process 42–4 effect of increased network density
model-based methods 44–5 252–4
model-free methods 45 effect of multi-hop 255–6
Monte Carlo methods 45–8 simulation setup 251–2
temporal-difference (TD) learning Q-learning MAC protocol 243
48–50 action selection dilemma 243
online solution via stochastic
a priori approximate controller
approximation 396–401
244–6
Q-learning 401–4
convergence requirements 244
reinforcement learning-based channel
implementation details 247–8
sharing, in wireless vehicular
networks 225 online controller augmentation
connected vehicles architecture 227 246–7
automotive sensors 228 reinforcement learning-based channel
electronic control units (ECUs) access control 241
227 Markov decision processes 242
intra-vehicle communications 228 Q-learning 242–3
types of communication 229 review of learning channel access
vehicular ad hoc networks 228–9 control protocols 241–2
dedicated short range communication VANET simulation modelling 248
231 implementation 249–51
control channel behaviour (CCH) mobility simulator 249
233 network simulator 248–9
IEEE 802.11p 231–2 reinforcement-learning-based wireless
message types 234 resource allocation 371
WAVE Short Message Protocol Markov decision process (MDP) 376
(WSMP) 232–3 basic components of 378–81
IEEE 802.11p medium access control
finite-horizon MDP 381–2
234
infinite-horizon MDP with average
basic access mechanism 235–6
cost 392–4
binary exponential backoff (BEB)
infinite-horizon MDP with
algorithm 236–7
discounted cost 387–9
DCF for broadcasting 238
distributed coordination function multi-carrier power allocation with
(DCF) 234–5 random packet arrival 389–92
enhanced distributed channel reinforcement learning 394–6
access 238–9 online solution via stochastic
RTS/CTS handshake 237 approximation 396–401
motivation 226–7 Q-learning 401–4
network traffic congestion 239 stochastic approximation 371–2
adaptivebackoff algorithms 240–1 iterative algorithm 372–3
transmission power control 240 stochastic fixed-point problem
transmission rate control 240 373–6
472 Applications of machine learning in wireless communications

RElative COre MErge (RECOME) signal-to-noise ratio (SNR) 240, 349

clustering algorithm 27–30, simulated annealing (SA) 166, 178
40–1 single hop communications, WSNs with
relative core merge clustering algorithm 213–14
27–9 single-snapshot evaluation algorithms
relative k-NN kernel density (NKD) 89
(RNKD) 27–8 singular value decomposition (SVD)
request-to-send (RTS)/clear-to-send 122–3, 207
(CTS) handshake 237 singular value thresholding (SVT)
restricted Boltzmann machine (RBM) 117–19, 128–30
model 345–6 sparse events detection 214–16
restricted isometry property 202–3 sparse recovery, numerical algorithms
RiMAX algorithm 87, 93 for 204
roadside units (RSU) capacity decision convex optimization algorithms
452 204–5
root mean square error (RMSE) greedy pursuit algorithms 205–6
129–30 sparsity-based clustering 76–8
round-trip time (RTT) 254 spatial difference features in HEVC
domain 321–2
SAGE 67, 83, 89, 93 specific emitter identification (SEI)
saliency weighted PSNR (SWPSNR) 160
267, 269, 274, 279, 282–3 specific emitter identification via
Sarsa algorithm 48–9, 55 machine learning 178
semidefinite programming (SDP) 118 correlation-based algorithm 184
Shannon’s law 413 energy entropy 183
ShapePrune 74–5 first- and second-order moments
Shengjin formula 271 183–4
signal identification in cognitive radios: Fisher’s discriminant ratio-based
see cognitive radios, signal algorithm 184–5
identification in Hilbert–Huang transform 181
signal recovery, compressive sensing empirical mode decomposition
(CS) and 181–2
CS model 200–1 Hilbert spectrum analysis 182–3
equivalent sensing matrix, conditions system model 179
for 202 relaying scenario 180–1
mutual coherence 203–4 single-hop scenario 179–80
null space property 202 Stackelberg game
restricted isometry property 202–3 framework 416–17
sparse recovery, numerical simulation for Q-learning based on
algorithms for 204 422–3
convex optimization algorithms stochastic approximation 371
204–5 iterative algorithm 372–3
greedy pursuit algorithms 205–6 online solution via 396–401
signal-to-interference-plus-noise ratio stochastic fixed-point problem
(SINR) 372, 408, 412–13 373–6
Index 473

stochastic fixed-point problem 373–6 Tobii TX300 eye tracker 313

stochastic optimization problems 371 Torch 345
sum of the absolute transformed trace-based models 432, 434
differences (SATD) 268, 270 training symbols 139
supervised learning 1, 19–20 Transmission Control Protocol
decision tree 4 (TCP)/UDP 232–3
classification and regression tree transmission power control 240
(CART) 5–7 transmission rate control 240
gradient boosting decision tree two-way matching tracking 90–1
(GBDT) 7–9
random forest (RF) 7–8 unified R–Q (URQ) rate control scheme
k-nearest neighbours (k-NNs) 266
method 2–4 unsupervised learning 20, 40–1
perceptron 9 autoencoder 37–40
logistic regression 12–13 clustering by fast search and find of
multilayer perceptron and deep density peaks 24–7
learning 13–18
density-based spatial clustering of
support vector machine (SVM)
applications with noise
10–12
(DBSCAN) 23–4
support vector machine (SVM) 10–12,
EM algorithm 32
19, 68–70, 111–13, 141, 179,
Gaussian mixture models (GMM)
307
29–32
identification procedure via 185–9
EM algorithm for 33–4
linear SVM 185–6
k-means 21–3
multi-class SVM 187
principal component analysis (PCA)
nonlinear SVM 186–7
34–7
support vector regression 12
support vectors 113 relative core merge clustering
survey-based model 432 algorithm 27–9
synthetic model 432, 434 User Datagram Protocol (UDP)
transactions 232
target-recognition-based clustering user of interest (UOI) 127
82–4
Taylor expansion 271 value function approximation 50–1
Taylor polynomial 189 value iteration 45–6, 55
Taylor series 179, 181 vanishing gradient 96, 99
temporal-difference (TD) learning vehicles to infrastructures (V2I) 229,
48–50, 55 432
temporal difference features in HEVC and vehicles to vehicles (V2V)
domain 320–1 combined performance analysis
TensorFlow 18, 135, 144, 148, 345 453–7
time-cluster-spatial-lobe (SL) (TCSL) vehicle-to-broadband cloud
clustering 82 communication 229
time-invariant channels 137–9 vehicle-to-vehicle (V2V) technology
time-varying flat-fading channel 137 225, 452
474 Applications of machine learning in wireless communications

vehicular ad hoc networks (VANETs) evaluation on our database 329–32

225–9, 432 setting on encoding and training
simulation modelling 248 325–6
implementation 249–51 heuristic video-saliency detection
mobility simulator 249 310–11
network simulator 248–9 machine-learning-based
vehicular delay tolerant network video-saliency detection 322
(VDTN) 452 saliency detection 324–5
vehicular distribution 440–1, 447–9 training algorithm 322–4
video coding, experimental results on spatial difference features in HEVC
292 domain 321–2
RC accuracy, evaluation on 297–300 temporal difference features in
R–D performance, evaluation on 296 HEVC domain 320–1
BD-PSNR and BD-rate 297 virtual carrier sensing 237
R–D curves 296–7 visual inspection 68, 72
subjective quality 297
settings 296 WAVE Service Advertisement (WSA)
video coding, incorporation in 265 messages 233
learning-based approaches 266 WAVE Short Message Protocol
model-based approaches 265–6 (WSMP) 232–3, 240
video-saliency detection 307 Wi-Fi-based fingerprinting 343
analysis on eye-tracking database WINNER model 87, 143
313–17 wireless access in vehicular
basic HEVC features 317–19 environments (WAVE) stack
bit allocation 317–18 232
motion vector (MV) 318–19 wireless channel model 137
splitting depth 317 wireless sensor networks (WSNs) 197
database of eye tracking on raw compressive sensing (CS)-based 211
videos 312–13 compressive data gathering
data-driven video-saliency detection 213–14
311–12 localization 217–18
experimental results 325 reduced-dimension multiple access
analysis on parameter selection 216–17
326–9 robust data transmission 211–13
effectiveness of single features and sparse events detection 214–16
learning algorithm 335–7
evaluation on other databases XGboost 9
332–4 Xu et al.’s method 209–10
evaluation on other work
conditions 334–5

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IET - Applications of Machine Learning in Wireless Communications

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IET - Applications of Machine Learning in Wireless Communications

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IET TELECOMMUNICATIONS SERIES 81

The Institution of Engineering and Technology

© The Institution of Engineering and Technology 2019

First published 2019

The Institution of Engineering and Technology

British Library Cataloguing in Publication Data

ISBN 978-1-78561-657-0 (hardback)

Typeset in India by MPS Limited

1 Introduction of machine learning 1

2 Machine-learning-enabled channel modeling 67

2.3.3 Kernel-power-density-based clustering 78

3 Channel prediction based on machine-learning algorithms 109

4 Machine-learning-based channel estimation 135

4.3.3 Deep learning for massive MIMO CSI feedback 145

5 Signal identification in cognitive radios using machine learning 159

6 Compressive sensing for wireless sensor networks 197

7 Reinforcement learning-based channel sharing in wireless

7.9.2 Effect of increased network density 252

8 Machine-learning-based perceptual video coding in wireless

9 Machine-learning-based saliency detection and its video decoding

9.4.2 Temporal difference features in HEVC domain 320

11 Reinforcement-learning-based wireless resource allocation 371

12 Q-learning-based power control in small-cell networks 407

13 Data-driven vehicular mobility modeling and prediction 431

13.4 Performance derivation 439

The technologies of wireless communications have been changed drastically in recent

RL-based adaptation and insight of the various challenges appearing when

● In Chapter 12, by integrating information theory with the principle of effective

Machine learning, as a subfield of artificial intelligence, is a category of algorithms

1.1 Supervised learning

1.1.1 k-Nearest neighbours method

(a) (b) (c)

Algorithm 1.1: k-NN method

where I (yi = c) equals 1 if yi = c and 0 otherwise;

Another useful distance is the angular distance, which is defined as

1.1.2 Decision tree

Figure 1.2 Decision tree structure

1.1.2.1 Classification and regression tree

Algorithm 1.2: CART constructing tree

Input: training set T = {(xi , yi ) ∈ Rd+1 }ni=1 , stop number s

Algorithm 1.3: CART predicting

1.1.2.2 Random forest

1.1.2.3 Gradient boosting decision tree

Algorithm 1.4: Random forest

Input: training set T = {(xi , yi ) ∈ Rd+1 }ni=1 , number of trees m, number of

9 if the task is regression then

fj (x) = fj−1 (x) + Treej (x; j ) ( j = 1, . . . , m), (1.9)

2 Hyperplane Positive sample

Algorithm 1.5: Perceptron learning

where wT x + b = 0 is the classification hyperplane, and yi (wT xi + b) > 0 implies

1.1.3.1 Support vector machine

1.1.3.2 Logistic regression

Figure 1.5 (a) A decision hyperplane wT x + b = 0 and three samples: x1 (blue

g(wT x + b) represents the probability of x belonging to the positive class. Thus, by

Because there is no closed-form solution to the problem of maximizing (1.16), the

1.1.3.3 Multilayer perceptron and deep learning

Table 1.1 Three commonly used activation functions

Name Abbreviation Formula Range

g (x (k1 , . . . , km )) = (g (x k1 ) , . . . , g (x km )) = (y1 , . . . , ym ) . (1.25)

Similarly, we can continue to transform (y1 , . . . , ym ) by using more convolution layers.

Fully connected layer (output)

Fully connected layer

Figure 1.8 (a) An example of max-pooling to x ∈ R6 with a window of size 2 and

1.1.4 Summary of supervised learning

Decision tree Random forest

Supervised Gradient boosting

Perceptron Logistic regression

Table 1.2 Summary of applications of supervised learning in wireless

Method Function Application in

k-Nearest Classification/Regression Fall detection [10]

cost huge amount of computing resources for training. As a reward, it is state-of-the-

1.2 Unsupervised learning

Algorithm 1.6: K-means algorithm

in collaborative signal processing, wireless surveillance systems and wireless hybrid

1.2.2 Density-based spatial clustering of applications with noise

Nε (x) = {xi |xi ∈ X , dist(x, xi ) ≤ ε} , (1.29)

Gk = (X , E), E = {x, z|x ∈ Nk (z) ∧ z ∈ Nk (x)}. (1.37)

9 construct directed graph G = (X , A), where A = {x, π (x)|x ∈ X \O};

where i 0 means that i should be a positive semi-definite matrix. Unfortunately,