Dirty RF

Ilmenau University of Technology
International Graduate School on Mobile Communications

Faculty of Electrical Engineering and Information Technology
Institute for Information Technology
Electronic Measurement Research Lab
Dirty RF Signal Processing

for Mitigation of Receiver Front-end Non-linearity
Dissertation
to obtain a doctoral degree in electrical engineering (Dr.-Ing.)
submitted by
Dipl.-Ing. Michael Grimm
born in Weimar on January 7th, 1986.
Reviewers:
1. Univ.-Prof. Dr.-Ing. habil. Reiner S. Thom (TU Ilmenau)

2. Univ.-Prof. Dr. rer. nat. habil. Matthias Hein (TU Ilmenau)
3. Prof. Dr. Tech. Mikko Valkama (TU Tampere, Finland)
Date of submission:
04/10/2013
Date of defence:
URN:
27/06/2014
urn:nbn:de:gbv:ilm1-2014000134
Acknowledgements
The help of various people have made this thesis possible or complete. First, I would like to
thank my supervisor Prof. Reiner Thom for the continuous support, encouragement, and his
human way throughout the last years. Without him, I would not have chosen this topic and
this thesis would not exist. Second, I would like to thank my master students Diego Dupleich
and Florian Schlembach for their valuable and fruitful cooperation and their contribution
to this thesis through their master theses and student assistant jobs. Third, I thank all
colleagues of the department Electronic Measurement Research Lab for discussions on
all matters and for the friendly working atmosphere. I would like to mention especially
my office colleagues Gerd Sommerkorn, Peter Rauschenbach, Robert Mller, and Stephan
Hfner for their friendly way and support in the everyday work. Throughout the doctoral
studies, feedback is indispensable to successfully complete a PhD. I would like to thank Prof.
Matthias Hein, Wim Kotterman, Ralf Herrmann, and Rajesh Sharma for proof-reading
my papers and thesis. I thank Kurt Blau for fruitful discussions, especially during lab
experiments.
The work presented in this thesis has been carried out within the International Graduate
School on Mobile Communications, that is supported by the German Research Foundation
(DFG GRK1487). I would like to thank Prof. Andreas Mitschele-Thiel for giving me the
chance to work on an interesting research project, and Mirko Kirschkowski for his kind
administrative support throughout my doctoral studies.
The cooperation with external partners has made this thesis complete. I feel special gratitude
for Prof. Mikko Valkama for reviewing my thesis and giving me the opportunity to visit
Tampere University of Technology in Finland for three months. This stay abroad enriched
my social and technical skills significantly. My special thanks go to Markus Alln and Jaakko
Marttila for the greatest cooperation one can think of, even in the time after my visit, where
we prepared a joint journal article for IEEE Trans. on Microwave Theory and Techniques
[GAM+14]. Furthermore, I would like to thank Reda Zemmari from Fraunhofer FKIE for
the great cooperation in the field of GSM and passive radar. Besides, I am very grateful for
our guest researcher Andrey Budilov for his cooperation and for being a great host during a
vacation in Russia in 2011.
iii
During my doctoral studies, I found many new friends who are colleagues in the Graduate
School. I would like to thank Alexander Krah, Andr Puschmann, Noman Murtaza, Anastasia Lavrenko, Tobias Simon, Christopher Schirmer, and Jonas Knig for their cooperation
at work and the nice time at workshops, conferences, and any social event beside work. You
made the last years to an unforgettable time in my life.
Finally, I would like to thank my parents, my brother and sister-in-law, and my grandparents
for their everlasting support and encouragement on my way. Furthermore, I have to thank
my girlfriend Katharina for her love, trust and motivation throughout the last years since
my diploma time.
Ilmenau, October 2013
Michael Grimm
iv
Zusammenfassung
Moderne drahtlose Kommunikationssysteme stellen hohe und teilweise gegenstzliche Anforderungen an die Hardware der Funkmodule, wie z.B. niedriger Energieverbrauch, groe
Bandbreite und hohe Linearitt. Die Gewhrleistung einer ausreichenden Linearitt ist,
neben anderen analogen Parametern, eine Herausforderung im praktischen Design der
Funkmodule. Der Fokus der Dissertation liegt auf breitbandigen HF-Frontends fr Softwarekonfigurierbare Funkmodule, die seit einigen Jahren kommerziell verfgbar sind. Die praktischen Herausforderungen und Grenzen solcher flexiblen Funkmodule offenbaren sich vor
allem im realen Experiment. Eines der Hauptprobleme ist die Sicherstellung einer ausreichenden analogen Performanz ber einen weiten Frequenzbereich. Aus einer Vielzahl an
analogen Streffekten behandelt die Arbeit die Analyse und Minderung von Nichtlinearitten
in Empfngern mit direkt-umsetzender Architektur. Im Vordergrund stehen dabei Signalverarbeitungsstrategien zur Minderung nichtlinear verursachter Interferenz ein Algorithmus,
der besser unter Dirty RF-Techniken bekannt ist. Ein digitales Verfahren nach der Vorwrtskopplung wird durch intensive Simulationen, Messungen und Implementierung in realer
Hardware verifiziert. Um die Lcken zwischen Theorie und praktischer Anwendbarkeit zu
schlieen und das Verfahren in reale Funkmodule zu integrieren, werden verschiedene Untersuchungen durchgefhrt. Hierzu wird ein erweitertes Verhaltensmodell entwickelt, das
die Struktur direkt-umsetzender Empfnger am besten nachbildet und damit alle Verzerrungen im HF- und Basisband erfasst. Darber hinaus wird die Leistungsfhigkeit des Algorithmus unter realen Funkkanal-Bedingungen untersucht. Zustzlich folgt die Vorstellung
einer ressourceneffizienten Echtzeit-Implementierung des Verfahrens auf einem FPGA. Abschlieend diskutiert die Arbeit verschiedene Anwendungsfelder, darunter spektrales Sensing,
robuster GSM-Empfang und GSM-basiertes Passivradar. Es wird gezeigt, dass nichtlineare
Verzerrungen erfolgreich in der digitalen Domne gemindert werden knnen, wodurch die
Bitfehlerrate gestrter modulierter Signale sinkt und der Anteil nichtlinear verursachter Interferenz minimiert wird. Schlielich kann durch das Verfahren die effektive Linearitt des
HF-Frontends stark erhht werden. Damit wird der zuverlssige Betrieb eines einfachen
Funkmoduls unter dem Einfluss der Empfngernichtlinearitt mglich. Aufgrund des flexiblen Designs ist der Algorithmus fr breitbandige Empfnger universal einsetzbar und ist
nicht auf Software-konfigurierbare Funkmodule beschrnkt.
Abstract
Todays wireless communication systems place high requirements on the radios hardware
that are largely mutually exclusive, such as low power consumption, wide bandwidth, and
high linearity. Achieving a sufficient linearity, among other analogue characteristics, is a
challenging issue in practical transceiver design. The focus of this thesis is on wideband
receiver RF front-ends for software defined radio technology, which became commercially
available in the recent years. Practical challenges and limitations are being revealed in realworld experiments with these radios. One of the main problems is to ensure a sufficient
RF performance of the front-end over a wide bandwidth. The thesis covers the analysis
and mitigation of receiver non-linearity of typical direct-conversion receiver architectures,
among other RF impairments. The main focus is on DSP-based algorithms for mitigating
non-linearly induced interference, an approach also known as Dirty RF signal processing
techniques. The conceived digital feedforward mitigation algorithm is verified through extensive simulations, RF measurements, and implementation in real hardware. Various studies
are carried out that bridge the gap between theory and practical applicability of this approach, especially with the aim of integrating that technique into real devices. To this end,
an advanced baseband behavioural model is developed that matches to direct-conversion
receiver architectures as close as possible, and thus considers all generated distortions at RF
and baseband. In addition, the algorithms performance is verified under challenging fading
conditions. Moreover, the thesis presents a resource-efficient real-time implementation of the
proposed solution on an FPGA. Finally, different use cases are covered in the thesis that
includes spectrum monitoring or sensing, GSM downlink reception, and GSM-based passive
radar. It is shown that non-linear distortions can be successfully mitigated at system level
in the digital domain, thereby decreasing the bit error rate of distorted modulated signals
and reducing the amount of non-linearly induced interference. Finally, the effective linearity
of the front-end is increased substantially. Thus, the proper operation of a low-cost radio
under presence of receiver non-linearity is possible. Due to the flexible design, the algorithm is generally applicable for wideband receivers and is not restricted to software defined
radios.
vii
Contents
1 Introduction
1.1 Research Problem . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1.2 Solution Strategies . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

1.3 State-of-the-Art and Research Contribution . . . . . . . . . . . . . . . . . .
3
4
1.4 Organisation of the Thesis . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2 Non-linear Distortions
2.1 Typical Receiver Architectures . . . . . . . . . . . . . . . . . . . . . . . . . .

2.1.1 Heterodyne Architecture . . . . . . . . . . . . . . . . . . . . . . . . .
7
7
2.1.2
2.1.3
Homodyne or Zero-IF Receiver . . . . . . . . . . . . . . . . . . . . .

Low-IF Receivers . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
8
9
2.1.4
Software Radio and Software Defined Radio . . . . . . . . . . . . . .
10
2.2 Sources of Distortions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

2.3 Behaviour of Non-linear Receivers . . . . . . . . . . . . . . . . . . . . . . . .
11
15
2.3.1
2.3.2
Effects . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Metrics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
15
19
2.4 Crucial Signal Scenarios . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

2.5 System-level Modelling of Non-linear Front-ends . . . . . . . . . . . . . . . .
22
23
2.5.1
2.5.2
Behavioural vs. Physical Models . . . . . . . . . . . . . . . . . . . . .

Modelling by Memoryless Polynomials . . . . . . . . . . . . . . . . .
23
25
2.5.3
2.5.4
Modelling of RF Non-linearity . . . . . . . . . . . . . . . . . . . . . .
Modelling of BB Non-linearity . . . . . . . . . . . . . . . . . . . . . .
28
33
2.5.5
Joint Modelling of RF and BB Non-linearity . . . . . . . . . . . . . .
37
2.5.6
2.5.7
Interaction with I/Q Imbalance . . . . . . . . . . . . . . . . . . . . .

Modelling of Memory Effects . . . . . . . . . . . . . . . . . . . . . . .
40
43
2.6 Circuit-level Simulation with SimRF . . . . . . . . . . . . . . . . . . . . . .

2.7 Measurement-based Analysis . . . . . . . . . . . . . . . . . . . . . . . . . . .
48
51
2.7.1
2.7.2
Model Identification Techniques . . . . . . . . . . . . . . . . . . . . .

Two-tone Excitation . . . . . . . . . . . . . . . . . . . . . . . . . . .
51
53
2.7.3
Wideband Test Signals . . . . . . . . . . . . . . . . . . . . . . . . . .
57
ix
Contents
2.8 Summary about Receiver Non-linearity . . . . . . . . . . . . . . . . . . . . .
3 Mitigation Techniques
64
65
3.1 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
65
3.1.1
Transmitter Non-linearity . . . . . . . . . . . . . . . . . . . . . . . .
65
3.1.2
Receiver Non-linearity . . . . . . . . . . . . . . . . . . . . . . . . . .
69
3.2 Dirty RF Techniques State of the Art . . . . . . . . . . . . . . . . . . . . . .
72
3.3 Reasons for Selecting Digital Feedforward Mitigation . . . . . . . . . . . . .
76
3.4 Cancellation of Distortion-Producing Signals . . . . . . . . . . . . . . . . . .
76
3.4.1
Analogue Cancellation . . . . . . . . . . . . . . . . . . . . . . . . . .
77
3.4.2
Spatial Filtering using Antenna Arrays . . . . . . . . . . . . . . . . .
80
3.5 Summary of Possible Mitigation Techniques . . . . . . . . . . . . . . . . . .
81
4 Digital Feedforward Mitigation
83
4.1 Principle . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
83
4.2 Components . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
86
4.2.1
Bandsplit Filtering . . . . . . . . . . . . . . . . . . . . . . . . . . . .
86
4.2.2
Reference Non-linearity Models . . . . . . . . . . . . . . . . . . . . .
88
4.2.3
Adaptive Filter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
95
4.3 Computational Complexity . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107

4.4 Proof of Concept . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108
4.4.1
Performance Simulations . . . . . . . . . . . . . . . . . . . . . . . . . 109
4.4.2
RF Measurements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 115
4.5 Fundamental Limits
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 118
4.6 Inclusion of Memory Effects . . . . . . . . . . . . . . . . . . . . . . . . . . . 122

4.7 Summary of Digital Feedforward Mitigation . . . . . . . . . . . . . . . . . . 122
5 Mitigation under Fading Conditions
125
5.1 Methodology . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 126

5.2 Impact of Fading and AGC . . . . . . . . . . . . . . . . . . . . . . . . . . . 129
5.2.1
Simulation Setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 130
5.2.2
Effects of Fading on the Non-linearities . . . . . . . . . . . . . . . . . 131
5.2.3
Mitigation Performance under Fading Conditions . . . . . . . . . . . 132
5.3 Summary about the Impact of the Radio Channel . . . . . . . . . . . . . . . 137

6 Real-time Implementation
139
6.1 Target Architecture and Basic Scenario . . . . . . . . . . . . . . . . . . . . . 139

6.2 Modules Implementation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 142
Contents
6.3 Experimental Results . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 149
6.4 Summary of Real-time Implementation Issues . . . . . . . . . . . . . . . . . 151
7 Application Scenarios
153
7.1 Spectrum Sensing in Cognitive Radio . . . . . . . . . . . . . . . . . . . . . . 153

7.1.1
Cognitive Radio Principle . . . . . . . . . . . . . . . . . . . . . . . . 153
7.1.2
Sensing Algorithms . . . . . . . . . . . . . . . . . . . . . . . . . . . . 155
7.1.3
Problem Definition . . . . . . . . . . . . . . . . . . . . . . . . . . . . 157
7.1.4
Detection of Mitigated Multi-carrier Signals . . . . . . . . . . . . . . 159
7.2 GSM Cellular Radio . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 162

7.2.1
GSM Physical and Link Layer Overview . . . . . . . . . . . . . . . . 162
7.2.2
Robust Train Radio Reception (GSM-R) . . . . . . . . . . . . . . . . 164
7.2.3
GSM-based Passive Radar . . . . . . . . . . . . . . . . . . . . . . . . 172
7.3 Summary of the Considered Application Scenarios . . . . . . . . . . . . . . . 174

8 Conclusions
177
8.1 Methodology and Main Results of the Thesis . . . . . . . . . . . . . . . . . . 177

8.2 Contributions to the State of the Art . . . . . . . . . . . . . . . . . . . . . . 178
8.3 Future Work . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 179
List of Abbreviations
181
List of Symbols
187
List of Figures
192
List of Tables
193
List of Publications
195
Bibliography
197
A Mathematical Derivations
211
A.1 Complex Memoryless Polynomial . . . . . . . . . . . . . . . . . . . . . . . . 211

B Universal Software Radio Peripheral
213
B.1 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 213

B.2 USRP Motherboard . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 213
B.3 RF Front-end Daughter Boards . . . . . . . . . . . . . . . . . . . . . . . . . 216
B.4 Host Computer Processing and Data Acquisition . . . . . . . . . . . . . . . . 217
xi
Contents
C Source Code
219
C.1 NONLIM Algorithm . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 219

C.2 USRP MATLAB Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . 220
Theses
223
xii
1 Introduction
The demand for broadband wireless communications is increasing every year and continuously accelerates the pace of engineering and science. There is an exponential growth in
data traffic due to the increasing number of users subscribing broadband packages and the
emergence of new devices, data services, and applications. A current trend is to apply multimode and multi-band transceivers based upon simple radio frequency (RF) front-ends that
integrate a variety of wireless standards. Conventional communication systems have fixed
radio parameters such as carrier frequency, bandwidth, and modulation scheme. Instead,
software defined radio (SDR) technology is experiencing increasing popularity where most
of the signal processing is performed in the digital domain. Thereby, simple and flexible
RF hardware can be realised and applied in many different use cases, operating at different
frequencies, power levels, and with different waveforms. Suitable areas of application for
this technology lie not only at the mobile terminal side, where costs have to be kept low
for mass production, but also the base station side that have to accommodate for various
wireless systems in a large frequency range. For instance, cognitive radio (CR) has been
proposed as a novel concept to solve the spectrum scarcity problem by utilising flexible SDR
hardware. Most of the CR research deals with theoretical analyses or is based on simulations that often do not consider the physical limitations of the radio. Since SDR platforms
became commercially available, the community is now focusing on real implementations and
practical evaluation of proposed algorithms and concepts. Thereby, practical limitations of
transceiver electronics in RF front-ends for SDR turned out to cause a performance degradation of the radio. Moreover, future emerging wireless communication systems demand
for higher bandwidth, higher-order constellation diagrams, and multiple antennas. These
increases in modem performance, bandwidth, and carrier frequency provoke dirt effects that
have reached a new problem level. In brief, systems will definitely become more sensitive to
front-end non-idealities. Therefore, it is worthwhile to analyse these effects and to look for
solution strategies.
1 Introduction
1.1 Research Problem

Novel concepts in communications and radar have high demands on the RF transceiver
electronics. Among all, wide bandwidth and frequency-agility play a crucial role. In addition, radios should operate under extreme dynamic range conditions, allowing analysis and
synthesis of different waveforms with different power levels. In fact, many of these requirements reach far beyond the state-of-the-art technology. These demands may be fulfilled
by measurement and prototype devices, but not by practical terminals in mass production.
The analogue stages of a radio front-end are inherently imperfect and are likely to cause
errors that deteriorate the signal quality. Add to this, RF technology is already pushed to
its operation boundaries, making it to a serious problem for future technology [Sch08]. In
general, there are different RF impairments, such as mirror-frequency interference due to
in-phase/quadrature (I/Q) imbalance, non-linear distortions, timing jitter, and phase noise
[VSH10].
The focus in thesis is on non-linear distortions at the receiver side as a subset of the impairments. Nowadays, simple direct-conversion receivers (DCRs) are employed that have
a general lack of selectivity at the intermediate frequency (IF) [Rou09]. Weak and strong
signals may be simultaneously present at the RF input, causing high demands on the receiver dynamic range and linearity. In traditional narrowband RF receivers, strong signals
are greatly suppressed by surface acoustic wave (SAW) filters. Typically for SDRs, there
is no pre-selection filter at the front-end input at all, allowing strong interferers to enter
amplification and mixing stages within the whole receiver bandwidth. Hence, non-linear distortions due to these strong signals are easily generated by non-linear receiver components,
such as the low-noise amplifier (LNA), the mixer, and additional baseband (BB) amplification stages. Finally, distortion products may fall in free frequency bands or hit weak useful
signals.
In contrast to the transmitter side, the perspective on non-linear distortions at the receiver
side is fundamentally different. At the transmitter, there is only one well-known signal
to be transmitted and interference is created in adjacent bands due to transmitter nonlinearity. However, non-linear distortions at the receiver are more challenging due to presence
of multiple unknown signals, with different power levels and dynamics due to specific wave
propagation. In other words, the original input signal is unknown at the receiver, whereas
desired and distorted output can be easily measured at the transmitter. In particular,
highly unbalanced signals appear in frequency-division multiple access (FDMA) systems and
finally sum up in the receiver. Thereby, distortion products are easily generated by different
unknown strong signals. On the other hand, crucial interference conditions seldom appear
1.2 Solution Strategies

in conventional radio systems, such as in cellular networks, but are more probable in CR
or in coexistence scenarios with heterogeneous radio systems. From the mitigation point of
view, handling of non-linear distortions at the receiver always have to start from a guess of
the input compared to pre-distortion techniques at the transmitter [BCG+10].
1.2 Solution Strategies

The main motivation is to consider non-linear distortions, created by the receiver front-end
non-linearity, in the operation of the radio. Finally, mitigating the deteriorating effects
and enhancing the reliability of the radio are desirable. Thereby, only partial mitigation of
non-linear distortions compared to full cancellation or compensation might be achieved.
There are several possible solutions that can be classified into three main categories:
Optimising the front-end design,
Preventing the generation of non-linear distortion, or
Mitigating non-linear distortion.
Following the first approach, highly linear components should be employed in the front-end
design so the receiver can even cope with high-power input signals. By using the second
approach, specific interfering signals are to be modified or suppressed to prevent non-linear
distortion products created by them during reception. The cancellation might be achieved
through attenuation, analogue filtering, analogue subtraction of an interferer replica, or
spatial filtering using an antenna array. Alternatively, waveforms may be adjusted towards
relaxed linearity requirements for the front-end, such as peak-to-average power ratio (PAPR)
reduction techniques. The third approach, being on focus in this thesis, includes all preor post-correction techniques to clean the signal from distortions after they have already
occurred. To this end, digital signal processing is adopted, e.g. by analysing specific training
symbols in the transmit signal, or by using a reference model for distortion regeneration
and subsequent subtraction. This approach is better known as dealing with Dirty RF
effects in the digital domain, a term, that has been first coined in [FLP+05]. However,
several synonyms can be found in literature, such as RF impairment mitigation, or alleviating
RF imperfections. Digital mitigation techniques are preferred, especially due to their easy
integration in the digital back-end. Moreover, digital circuits are growing faster and are less
sensitive to process variations compared to their analogue counterparts [HB08; Chi11]. Up to
now, analogue front-end and digital back-end have been designed independently, assuming
that each block works ideally. Digital mitigation of non-linear distortions can relax the
1 Introduction
linearity and dynamic range requirements of the front-end, enabling the design of low-cost
receivers. That requires a close cooperation between RF and signal processing engineers.
1.3 State-of-the-Art and Research Contribution

Extensive research has been undertaken on pre-distortion techniques for mitigating
power amplifier (PA) non-linearity at the transmitter, such as in [BBC88; BCG+10]. However, there is also some prior work considering non-linear distortions at the receiver side.
Proposed solutions vary in the complexity of implementation and also in the use cases considered. Mitigating non-linear distortions by means of digital signal processing has been
first proposed in [VSHGA+06], and subsequently followed in [KH08b; ZMS09; AMV10b;
GSH+12a], among others. In [VSHGA+06], a reference model of the non-linear receiver is
applied in the digital domain to regenerate distortion products, and to subtract them from
the received signal using adaptive filtering. Custom receiver designs are proposed in [KH08b]
and [ZMS09], to overcome the limitations of the entirely digital approach. Sometimes, pilot signals are integrated in the transmit signal, enabling the subsequent estimation of the
receiver non-linearity at symbol level [DS07].
However, there are some issues in the current state-of-the-art that have not been considered
so far:
Specifics of DCRs,
Application to modulated signals,
Impact of fading conditions, and
Real-time implementation.
The main contributions of this thesis are seen in the consideration of these issues that bridge
the gap between theory and practical applicability of digital mitigation algorithms. Real
front-end signals are analysed and processed throughout the thesis. First, an experimental
characterisation of the non-linear behaviour and the effects of non-linear distortions has
been conducted based on a wideband receiver for SDR. Second, a state-of-the-art mitigation
algorithm has been extended for DCRs and applied to real communication signals. Thereby,
the performance of Dirty RF algorithms was analysed in real-world scenarios. Third, an
advanced reference model for the non-linearity of a DCR has been developed that jointly
considers distortions created at RF and BB. Finally, practical issues have been covered, such
1.4 Organisation of the Thesis

as the impact of fading conditions on the performance of the algorithm, or implementation
issues for embedding the algorithm onto a field-programmable gate array (FPGA).
Three main use cases of the algorithm have been investigated in more detail:
Spectrum sensing in CR,
GSM downlink reception, and
GSM-based passive radar.
1.4 Organisation of the Thesis

The introduction is followed by six main chapters. Chapter 2 deals with characterisation
of receiver non-linearity, including sources, modelling, and measurement-based analysis. In
Chapter 3, a detailed overview of possible mitigation techniques is given. Concepts on
cancelling distortion-producing signals are briefly discussed. Then, the digital feedforward
mitigation algorithm, on focus in this thesis, is discussed in detail in Chapter 4. Practical
aspects of this technique, namely the impact of fading conditions and the challenges of its
real-time implementation, are discussed in the Chapters 5 and 6. Chapter 7 presents three
use cases, underlining the practical relevance of the digital feedforward mitigation. Finally,
conclusions and an outlook on future work are given in Chapter 8.
This chapter deals with typical front-end architectures adopted in state-of-the-art receivers
as well as with sources of non-linearity. In addition, BB models of non-linear front-ends
are developed that are used later in the mitigation part. To end the chapter, results of an
experimental characterisation of a typical SDR platform with narrowband and wideband
test signals are presented.
2.1 Typical Receiver Architectures

There is a variety of receiver architectures that are classified by the number of frequency
conversions to translate the signal from RF to BB. Most common types are heterodyne, homodyne, or low-IF, which have benefits and problems in certain applications [HS11]. Among
all, the homodyne or zero-IF architecture fits best for SDR. Problems of this simplified frontend have been better understood in recent years, thereby enabling the design of a low-cost
front-end [HS11]. First, most common architectures with their pros and cons are presented.
Finally, the concept of SDR is picked up and suitable candidate architectures are discussed.
2.1.1 Heterodyne Architecture

The heterodyne architecture, being the conventional design, first translates the RF signal
to a lower IF by using a down-conversion mixer [GCB08]. Thereby, possible out-of-band
interferers are filtered at IF, hence, a high selectivity is achieved. Therefore, less selective
filters may be adopted before conversion to the digital domain. Add to this, there is a
minimum received signal power level that the receiver can resolve above its noise level. For
example, adding a pre-selection filter in front of an amplifier adds noise and reduces the
sensitivity. Hence, there is always a trade-off between sensitivity and selectivity in designing
a heterodyne receiver [Raz97]. Typically, a dual-IF architecture is utilised as illustrated in
Figure 2.1. A high first IF is used to easily perform image rejection filtering by the bandpass filter BP2 that can have a lower quality factor at high frequencies. A second low IF
LP
VGA
ANT
0
90
BP1
LNA
ADC
ADC
LO2
BP2
LO1
LP
VGA
Figure 2.1: Block scheme of a typical dual-IF heterodyne receiver.

is used to relax requirements for the channel selection filter. At the second IF, quadrature down-conversion is adopted, that separates the signal into in-phase component (I) and
quadrature component (Q) through multiplication with two sinusoidal frequencies that are
exactly 90 out of phase. After digitisation by two analogue-to-digital converters (ADCs), demodulation and detection are performed at BB by means of digital signal processing (DSP).
Alternatively, digitisation may happen already at IF, moving the quadrature mixing and
lowpass filtering into the digital domain.
The heterodyne architecture can be easily adapted to many different standards, achieving
very good sensitivity and selectivity. However, this structure requires a large number of
external components, therefore, making integration for battery-powered terminals difficult.
Moreover, this front-end is less flexible and thereby not particularly well suited for SDR
purposes.
2.1.2 Homodyne or Zero-IF Receiver

The homodyne receiver, shown in Figure 2.2, directly converts the RF signal down to BB,
i.e. direct current (DC) frequency (zero-IF or DCR) [Raz97; GCB08]. Instead of bandpass
filtering at IF, all interferers are suppressed by a lowpass at I and Q branch. This receiver
architecture is simple, cheap, easy to integrate, and causes only a low power consumption.
Therefore, it became very popular in communications. However, the simplicity of this frontend also causes some major drawbacks, such as DC offsets, even-order distortions, I/Q
mismatch, and flicker noise. DC offset is caused by mixing of local oscillator (LO) leakage
with the LO itself, either by the feedthrough from the mixer LO port to the mixer input,
or by a large signal leaking from the LNA to the LO port. Distortions in DCRs will be
discussed in the next section in more detail. Any amplitude or phase mismatch between the
2.1 Typical Receiver Architectures

LP
VGA
ANT
0
90
BP
ADC
ADC
LO
LNA
LP
VGA
Figure 2.2: Block scheme of a homodyne (zero-IF) receiver.

two receiver branches for I and Q causes interference that may fall on top of other signals. In
addition, flicker noise or 1/f noise, typically negligible in heterodyne receivers, increases the
overall noise figure of the receiver, especially due to the low signal frequencies around DC
at the stages after the mixer. However, many of these issues have been solved by integrated
circuit technology in the past years.
2.1.3 Low-IF Receivers

An interesting approach is to combine heterodyne and homodyne architecture, where the
received signal is down-converted to a low non-zero IF in range of several hundred kHz up to
several MHz. After amplification at BB, the signal is digitised by an ADC. The final downconversion to DC is performed digitally by using a numerically controlled oscillator (NCO)
and a complex multiplier, as illustrated in Figure 2.3. As the signal is centred around a low
analogue domain
LP
ADC
ANT
digital domain
VGA
BP
sin
LO
cos
Q
0
90
LNA
ADC
Q
LP
NCO
VGA
Figure 2.3: Block scheme of a low-IF receiver.

IF, DC offset is negligible and can be easily filtered by averaging [LHG+10]. Beside the
preferences as in the zero-IF architecture, a large image rejection ratio (IRR) is required for
this receiver topology.
There are various other receiver types that are out of scope here, such as bandpass sampling
or direct RF sampling, that apply discrete-time analogue signal processing by subsampling
[GCB08]. These concepts can save the mixer through a subsampling operation, but impose
very tough bandpass filter specifications in order to avoid aliasing. Hence, these architectures
are not suited for flexible SDR front-ends that are on focus in this thesis.
2.1.4 Software Radio and Software Defined Radio

In 1995, Mitola proposed a novel concept of a software radio (SR) as shown in Figure 2.4
[Mit95; GCB08]. The idea is to move the digitisation as close as possible to the antenna
Digital Domain
Analogue Domain
ANT
LNA
ADC
Baseband
Processing
Figure 2.4: Ideal SR architecture proposed in [Mit95].

without any frequency conversion. Thus, most of the signal processing tasks are performed
in the digital domain. The remaining analogue components are the LNA with a built-in
automatic gain control (AGC) and the ADC. This receiver design is most flexible as it can
operate over a wide range of frequencies and process arbitrary waveforms. However, the SR
concept is far ahead from practical implementation with the state-of-the-art technology. For
instance, designing an omnidirectional antenna that serves a bandwidth ranging over several
decades of frequencies is challenging. In addition, the requirements for the ADC are very
difficult to achieve, such as the large input bandwidth, high sampling rate, and high dynamic
range, especially when considering direct RF sampling according to the Nyquist theorem.
A practical implementation of the SR concept with todays technology is represented by a
SDR that keeps the frequency conversion stage and some of the analogue signal processing.
Eventually, the analogue RF front-ends components are designed in a way that they are
reconfigurable by software, therefore it is called a software defined radio [Rou09]. In contrast
10
2.2 Sources of Distortions

to traditional single-mode front-ends, a SDR may adapt itself to various air standards and
waveforms at different frequencies, with flexible channel bandwidth, and channel conditions.
To realise the large operating bandwidth, multiple receive chains may be required, each
working in different frequency bands. The best candidate to implement a SDR is the zeroIF or low-IF topology due to their reconfigurability over a broad frequency range. Taking
the heterodyne architecture, no common IF will be found that fits for all input frequency
bands [GCB08]. The reconfigurable parameters of an SDR are, e.g., the carrier frequency,
the channel bandwidth (sampling rate), the amplifier gain, and the filter characteristics.
Thereby, the SDR can fulfil the requirements of different standards.
In contrast to the receiver block schemes illustrated in Figure 2.2 and Figure 2.3, the preselection bandpass between the antenna and the LNA is omitted in typical SDR front-ends
in favour of the flexibility. Typically, a SAW filter is employed as a bandpass, filtering
out most of the harmful out-of-band interference. However, this filter has a fixed centre
frequency and is thereby very application-specific, being a major disadvantage in SDR.
Tunable bandpass filters are very difficult to implement and are not available off the shelf
[VSHGA+06; ZMS09]. The missing pre-selection in such front-ends is a critical issue as
it reduces the receiver immunity to blockers, thereby dramatically increasing the linearity
requirements for the following amplification and mixing stages. Add to this, aliasing effects
in the down-conversion due to signals at mirror frequency are typically accepted in favour of
flexibility. The only filtering remaining in the analogue domain is performed by the lowpass
at BB. However, the analogue front-end components have, in any case, a limited bandwidth.
These are, for example, the input bandwidth of the LNA and the IF bandwidth of the
mixer.
In 1999, Mitola coined the concept of CR that employs the SDR technology [MM99].
Whereas a SDR targets certain standards and their allocated bands, a CR can be seen
as an extension that adds cognition and makes adjustments of the radio based on algorithms
in a cognitive engine [Raz09; Hay05; Law10]. The CR use case is discussed in detail in
Section 7.1.

Non-linear distortions can appear at the transmitter or receiver side of a communication
system, while the receiver side is on focus in this thesis. Electrical systems can be classified into linear and non-linear systems that either exhibit memory or not [VR03]. In
mathematics, a non-linear system is given if the principle of superposition is violated, or
11
if there is no proportional relationship between the input and output. Figure 2.5 depicts
examples of different transfer functions, a linear and two non-linear curves derived from
different input-referred third-order intercept points (IIP3s), a measure for the strength of
non-linearity. The lower the value of IIP3, the stronger is the non-linearity. A linear device
Output voltage (V)
0
Linear
IIP3 = 15 dBm
IIP3 = 12 dBm
1
2
1
0.5
0
Input voltage (V)
0.5
Figure 2.5: Linear and non-linear transfer functions.

outputs a voltage directly proportional to the input (green dashed line in Figure 2.5). The
output voltage is two times higher than the input, indicating an amplification factor of 2
(equals 6 dB gain). A non-linear device, as illustrated by the orange and blue curves, operates as a linear device under small-signal conditions. However, with increasing input signal
strength, the gain decreases due to saturation effects in the electronics. This phenomena is
also called compression and results in a non-linear function [Ken00]. In fact, real circuits
do not exhibit such an exaggerated non-linear behaviour as indicated by the blue curve in
Figure 2.5. It is indeed a result of the limited order of the non-linearity model used for this
simulation. Typically, a non-linear device is going from linear behaviour ([0.25, +0.25] V),
to mildly non-linear ([0.25, 0.5] V), and then to strongly non-linear behaviour (0.5 V onwards). Amplifiers manifesting such a behaviour are called linear, i.e. the amount of distortion can be reduced when the input power is decreased [Sch09]. This is typical for class
A and AB amplifiers, but not for switching-mode amplifiers that are designed for highly
efficient operation.
Non-linear behaviour is caused by parasitic effects in any active analogue circuit, such as
a diode or a transistor. On the contrary, passive elements such as resistors, capacitors,
and inductors are linear components. However, cables, antennas, or connectors may cause
non-linear junctions under special circumstances, better known as passive intermodulation
[Anr12]. From the circuit design point of view, non-linearities are typically annoying as they
12

ANT
Wideband RF Front-end (WBX)
y(t)
Digital Back-end
ADC
IF-A
LNA
y(n)
DSP
(FPGA)
USRP N210
SW
PC
LO
Figure 2.6: Block scheme of a typical SDR receiver, taking the example of USRP N210+WBX
[Ett].
distort electrical signals, but can be sometimes useful for certain applications, such as for
designing a mixer by using generated harmonics of the LO.
Now, sources of distortions in a typical RF front-end for SDR are discussed, taking the
example of Universal Software Radio Peripheral (USRP) version N210 with a wideband RF
front-end (WBX) [Ett]. Details on this receiver, that is used for many of the RF measurements in this thesis, are given in Appendix B. Figure 2.6 depicts a block scheme, where all
components with non-linear characteristics are highlighted. The WBX front-end is built as
a low-IF architecture without SAW pre-selection filter, converting the RF signal down to a
low user definable IF (see also Appendix B). Non-linear components in this front-end are
the LNA and the IF amplifier, the mixer and the LO, as well as the ADC.
The LNA provides substantial gain of the RF signal and fixes the noise floor [HB08]. As a
classical amplifier, it suffers from gain compression under large-signal excitation and causes
non-linear distortions.
The mixer translates the RF signal to BB with a non-linear conversion characteristic. A
single frequency generated by the LO is required for the mixing process. However, a non-ideal
LO may also create harmonics beside the fundamental frequency that can mix with potential
interferers, if the receiver bandwidth is wide enough [Raz97; Raz09]. This effect is negligible
for higher harmonics as their magnitude is inversely proportional to the frequency.
The BB or IF amplifier is used for optimal analogue/digital (A/D) conversion and behaves
in a same non-linear manner as the LNA. However, in terms of linearity performance, the
BB amplifier is typically worse than the RF amplifier (LNA).
The ADC is responsible for digitisation of the analogue BB signal, however, suffers
from non-linearity and other impairments, such as quantisation noise, sampling clock off-
13
1
3
ADC
ANT
2
LNA
LO
IF-A
duplexer
DAC
PA
LO
IF-A
Figure 2.7: Leakage effects in DCR architectures as another source of distortions.

set, and sampling jitter [AMV10a]. Non-linearities are caused by clipping, as well as
integral non-linearity (INL) and differential non-linearity (DNL), denoting unintentional deviations of the ideal quantisation levels. Whereas the DNL denotes the relative difference
between the actual and the ideal code, the INL means the difference between the actual
and ideal transition threshold (integral of DNL). Clipping distortions (CDs) are created due
to improper signal conditioning, i.e. the input signal exceeds the full scale voltage of the
ADC.
Another source of distortions are caused by leakage effects, especially in DCRs. Figure 2.7
depicts three different phenomena: LNA-mixer feed-through ,
1 LO-mixer feed-through ,
2
and duplexer feed-through
3 [KAV13; KAV14; CGH+11; Raz97]. Considering an ideal
mixer, low-frequency even-order distortions produced by the LNA are translated to high
frequencies and become unimportant. However, they can feed through BB without mixing
due to circuit asymmetry [CGH+11; Rou09]. In addition, DC offsets may be generated
due to feed-through between LO and mixer input [Raz97]. Moreover, a strong transmit
signal can leak into the receiver path via the duplexer that typically switches the antenna
between transmitter or receiver. This signal gets further amplified by the LNA and may cause
additional distortions, especially in frequency division duplexing (FDD) operation [Rou09].
Although leakage phenomena are side effects, they are important to know when dealing with
DCR architectures and may contribute to the overall distortion seen in the digitised BB
signal.
14
2.3 Behaviour of Non-linear Receivers

Besides, scaling in technology promotes the non-linear effects by reducing the dimensions of
transistors. Thus, decreasing supply voltages limit the dynamic range and increase the level
of non-linearities [HB08]. Likewise, behaviour of the components is less predictable due to
increased variability in the process.
Above all, non-linear circuits can be either with or without memory. Memory effects of
the receiver components can have a strong impact on the nature of non-linear distortions
[ZWC10; Rou09]. They are generated by inductors and capacitors that store energy, as well
as by their frequency-dependent impedance. As a result of memory, node voltage and current
depend not only on the instantaneous input, but also on historical signals. That is, the output
of the component depends on the current input and the previous input. The higher the
bandwidth of the input signal, the higher the memory effects will raise. Memory effects can
be distinguished into two types: electrothermal memory effects appearing at low modulation
frequencies below 100 kHz, and electrical memory effects appearing at modulation frequencies
of above some MHz [VR03].

This section discusses the effects of receiver non-linearity and their metrics for describing
the strength of non-linear distortion.
2.3.1 Effects
General results of circuit non-linearity are twofold. First, new frequency components are
generated by the non-linear device, thereby adding unwanted signal energy to the received
signal [Ken00]. Second, there is an amplitude-dependency of the fundamental signal gain
[VR03]. Phenomena of non-linear systems can be classified in order and degree [VR03].
The higher the order of non-linearity, the more non-linear signal content will be generated.
Degree is a property of the non-linear device and defines the shape of the non-linearity.
Order is related to the amplitude and frequency of the distortion terms, that depend on the
degree of non-linear device and the amplitude and frequency of the input. For instance, the
order of a distorted signal can be higher than the degree of the device in case the input was
already distorted, e.g. in cascaded non-linear devices in a DCR chain (see Section 2.5.5).
Typically, non-linear devices are classified into linear, mildly non-linear, and strongly nonlinear according to their degree of non-linearity (see also Figure 2.5) [Sch09].
15
7
4
5
6
even-order
odd-order
HD
IMD
7th
6th
4th
3rd
2nd
DC
Fund
2nd
4th
5th
6th
f /fc
Figure 2.8: Spectral components induced by non-linear device with degree 7 and two-tone
input.
16

Five different types of non-linear distortions at the receiver can be distinguished and are now
discussed in detail. Figure 2.8 illustrates most of the different distortions induced by nonlinear device of degree 7 and with two-tone input. In contrast to typical figures in textbooks,
even and odd-order distortions are illustrated at complex BB after passing a low-IF receiver
with quadrature mixer. The two-tone input signal, the blue arrows at f /fc = 1, are assumed
to be ideally converted to BB and then distorted by a BB amplifier of degree 7 in the I and
Q branch, respectively. In addition, DC, fundamental zone (Fund), and harmonic zones up
to order 7 are sketched in Figure 2.8 [Ken00].
Compression or in-band distortion denotes the impact of the receiver non-linearity on the
original input signal that basically results in gain reduction as illustrated in Figure 2.5.
Harmonic distortion (HD) denotes integer multiples of the input frequency fc . That is,
harmonic signals are generated at DC, 2fc , 3fc , up to N fc , where the factor N is the order.
HDs are already generated with a single input tone, which is especially a crucial effect for
non-linear LOs. In Figure 2.8, HDs are depicted with dashed lines and squared markers.
Intermodulation distortion (IMD) or non-harmonic distortion describe linear combinations of the input frequency components. They can be interpreted as the response of the
non-linear system to multiple signals at the input that comprises at least two frequencies
[Rou09]. Odd-order IMD are the most annoying effects and describe distortion terms falling
into the desired frequency band nearby the original input. Multiple odd-order IMD terms
cause a spreading of the original input signal, also called spectral regrowth or out-of-band
distortion, especially in the field of transmitter non-linearity. It is a special term for IMD
caused by complex-envelope or non-constant envelope modulated signals and is referred to
the bandwidth of the input signal [Sch09]. Even-order IMD shows up around DC and plays
a key role in DCRs, where even-order distortion induced by the LNA easily leak through BB
via the mixer (see Section 2.2), or directly fall on top of the desired signal if the receiver has
a high bandwidth over several decades. The IMD products show up in each harmonic zone,
illustrated with straight lines and circled markers in Figure 2.8, surrounding the harmonic
frequencies of the pure tones. For an input of two tones at f1 and f2 , the new frequencies
generated due to HD and IMD will be of the form
f = m f1 n f2 ,
17
(2.1)
where m and n are positive integers and m + n equals the order of distortion [Ken00]. If
either m or n is zero, the resulting frequency denotes HD, otherwise IMD. Following the
example of a non-linear device with degree 7, as illustrated in Figure 2.8, m and n will run
from 0 to 7 in (2.1), while obeying the condition m + n 7.
Cross-modulation distortion (XMD) is a specific type of intermodulation between two

relatively strong signals. Here, the stronger signal given to any odd-order non-linearity is
transferring its amplitude modulation to a weaker signal, even though the opposite case
is also possible. The amplitude modulation can be a result of a filtered digital modulation
scheme or a result of fast fading of the radio channel [Ken00]. The generation of XMD effects
depends on the power of all signals at hand [VSHGA+06]. Figure 2.9 illustrates this problem
for a Gaussian shaped modulated strong interferer and a weak continuous wave (CW) wanted
signal.
Strong unwanted
signal
IMD
Weak wanted
signal
XMD
Non-linear system
Figure 2.9: Impact of cross-modulation distortion (XMD) onto a weak carrier.
Phase Distortion or amplitude-to-phase (AM/PM) distortion denotes any modification of

the phase of the input according to the non-linear transfer function [Ken00]. The preceding
types just denote amplitude-to-amplitude (AM/AM) distortion, i.e. modification of the
amplitude of the signal according to a non-linear transfer function. Indeed, AM/AM and
AM/PM distortion superimpose in a real receiver, i.e. both amplitude and phase of the
input are shaped according to the non-linear transfer function. For this reason, upper and
lower IMD products may differ if AM/PM is out of phase for one of the products (e.g. 180
would cause disappearance of the distortion product).
18

Clipping distortion (CD) is caused due to limiting amplifiers and ADCs. These are severe
distortions that cover the whole BB signal and appear almost randomly due to modulated
signals [VSH10; AMV10b]. Whereas the first four types describe mild distortions that can
be mitigated at reasonable complexity level, mitigating CD can be more challenging and is
out of focus in this thesis.
Memory effects cannot directly be measured from the input and/or output signal of the
non-linear circuit [ZWC10]. Memory itself does not modify the steady-state signal waveform,
but introduces a phase shift between the input and the output. Thereby, the largest time
delay of the output determines the depth or order of the memory. Indicators for memory
effects in practical measurements are, e.g. imbalances between the upper and lower IMD
products in case of multi-tone excitation, or spread of AM/AM and AM/PM [Sch09]. In the
latter case, the conversion is broadened, i.e. the input/output values are not identical to the
mean amplification. In general, memory effects appear if the bandwidth of the excitation
signal is comparable with the inherent bandwidth of the analogue receiver components.
To sum up, non-linearities limit the receiver sensitivity and dynamic range through aforementioned distortion phenomena.
2.3.2 Metrics
There are several practical metrics describing the non-linear behaviour that differentiate
between narrowband and wideband modulated signals [VR03; Ken00]. Figure 2.10 is illustrating, in logarithmic scale, the output vs. the input power of a non-linear device of third
degree and supports the derivation of typical metrics for circuit non-linearity.
1dB compression point (P1dB) denotes the input power level where the large signal gain
has dropped by 1 dB due to the compressive and saturating nature of the circuit [Rou09].
The input/output relationship of the device is no longer linear, hence, the real output drops
off from the linear output power curve. Non-linear devices operating around the P1dB are
already heavily non-linear and cause severe distortions. Therefore, receivers should typically
operate below the P1dB with a certain back-off in order to avoid spectral regrowth and
signal distortion.
19
Intercept Points are one of the most common metrics describing the strength of circuit
non-linearity and became standard figures of merit in the RF and microwave community.
These are fictitious points, e.g. IP2 and IP3 in the power level diagram Figure 2.10, where
extrapolated linear and distortion products cross [VR03]. That is, the IMD product equals
that of the fundamental signal amplitude. In Figure 2.10, power of IMD products of second
and third order are illustrated in red and blue colour, respectively. The slope of these curves
depends on the order of the distortion term. Hence, the slope of the fundamental term is
1 : 1 (linear curve), and 2 : 1 and 3 : 1 for the IMD2 and IMD3 terms, respectively. In
other words, with increasing input power, the IMD products increase 2 (or 3) times faster
than the power of the fundamental frequency. Intercept points can be defined according to
the input or output power. For instance, the IIP3 denotes the input-referred third-order
intercept point, the OIP2 denotes the output-referred second-order intercept point. It is
noteworthy that the system power would compress before the input power reaches the power
of the intercept point. In case of memoryless non-linearity under weak signal conditions, i.e.
with a high back-off to the P1dB, there are simple relationships between the IMD power and
the corresponding IIP, as given in (2.2) and (2.3).
PIMD2 = 2Pin PIIP2
(2.2)
PIMD3 = 3Pin 2PIIP3
(2.3)
For memoryless systems, IIP3 is typically 10 dB higher than the P1dB [Rou09]. Also, a
system intercept point can be defined, which depends on the receiver line-up and describes
the overall intermodulation characteristics, or in general sense, the linearity of the system.
It is determined by various blocks linearity and gain, as well as their contribution in the
receiver line-up [Rou09].
Basic Performance Specifications are typically found in data sheets of non-linear devices
and base on raw-level numbers. They are either expressed in dBc, if there is some dependency
on the LO or interferer power level, otherwise in dB, dBm, or % [Rou09]. For instance, HD2
given in unit dBc means the power difference between the fundamental (or carrier) frequency
to the second harmonic. Similarly, IMD3 in dB denotes the power difference between the
fundamental and the one of the third-order IMD products. Moreover, there are other figures,
such as the total harmonic distortion, often used in acoustics, that describes the harmonic
performance. It is defined as the ratio of the sum of powers of all harmonic components to
the power of the fundamental frequency in %.
20

Output power (dB)
Linear curve
OIP2
OIP3
IP3
IP2
Real
P1dB
1 dB
3
2
1
1
Input power (dB)
P1dB IIP3 IIP2
Figure 2.10: Power level diagram illustrating typical metrics for non-linear circuits.
AM/AM and AM/PM are transfer functions modelling the amplitude and phase of the
fundamental signal with increasing input [VR03]. AM/AM measures the non-linearity on
basis of the fundamental signal that comprises a strong linear term. However, very small
non-linear effects will be seen on top of it due to common good linearity of systems. Similarly,
the AM/PM is typically very small and at the order of 1 to 2 at full power and approaches
zero with decreasing power. Therefore, both metrics are highly sensitive to measurement
errors and seldom used in practice. It is easier and more robust to measure metrics based
on the generated distortion terms, such as the intercept points.
Metrics for Wideband Signals are, e.g., adjacent channel power ratio (ACPR),
noise power ratio (NPR), and multi-tone intermodulation ratio (M-IMR) [Ken00]. Previous metrics were originally defined based on narrowband signals, therefore, representing
only a narrowband approximation of real bandwidth-dependent systems. However, metrics describing the effects of non-linearity for wideband modulated signals are available.
ACPR denotes the degree of signal spreading, i.e. spectral regrowth by any odd-order
IMD, into an adjacent channel. NPR is a measure of unwanted in-channel distortion
power, found by examining the level of distortion filling a spectral gap in the input
signal, that has been generated with a notch filter before. The M-IMR, often used in
21
orthogonal frequency-division multiplexing (OFDM) systems, is defined as the ratio of the
wanted tone power to the highest IMD power.
Error vector magnitude (EVM) is a performance metric of a digital radio demodulator

in general. It measures how far the received constellation points are from the ideal locations
[ZMB12]. Sometimes, EVM is mentioned in the context of non-linear distortions. However,
the radio channel and any RF impairment may degrade the EVM performance. Thereby,
different signal space constellations are subject to different EVM values, a ratio of the power
of the error vector to the root mean square (RMS) power of the reference (typically the
outermost constellation point or average power).
2.4 Crucial Signal Scenarios

Beside the non-linear behaviour of the receiver circuits, signal power at the receiver input
must be sufficiently high such that the non-linearity gets excited and essentially generates
distortion. Hence, the amount of distortion depends on the strength of non-linearity and the
input power, and therefore, on the actual signal configuration at hand.
Due to the missing pre-selection filters before the LNA in typical SDRs (see Section 2.1.4),
strong out-of-band blocking signals may enter the front-end amplification and mixing stages
[VSH10]. Coexistence scenarios with weak wanted and strong unwanted signals in heterogeneous radio systems are of major concern. For instance, these strong signals may cause
signal desensitisation and blocking of the weak signal [Rou09], and are often referred to as
blockers. With increasing signal power, the compressive behaviour of the circuit appears
and the desired signal gets attenuated. This effect is known as desensitisation [WSW+10].
The power dynamics between the two signals may be up to 6080 dB, as illustrated in Figure 2.11 [Raz10]. These conditions are quickly achieved due to near-far effects caused by
Blocker signal
up to
80 dB
Desired signal
Frequency
Figure 2.11: Typical signal configuration with weak desired and strong blocker signal.
22
2.5 System-level Modelling of Non-linear Front-ends

wave propagation in a mobile environment, especially at the mobile terminal in cellular networks [Ken00]. Here, weak links to a serving base transceiver station (BTS) in presence of
strong interfering BTSs are susceptible to non-linear distortions (see Section 7.2.2). In contrast, there may be also very high linearity requirements for the BTS receiver hardware that
has to accommodate for various wireless system standards in a large frequency range [Etsa;
Etse]. Further scenarios on focus in this thesis, where such conditions occur, are spectrum
sensing in CR and passive radar (see Chapter 7).
However, in general, receivers need to operate under such crucial signal configurations only
for a small fraction of time of roughly 10 % [KH08a; KH09b]. Though, in many applications,
even such short-time problematic blocker conditions may cause a failure of the radio and
should therefore be handled properly.

This section deals with behavioural modelling of non-linear devices that is required later on
for the mitigation processing. The use of memoryless polynomials is justified and modelling
of a typical DCR topology is discussed in detail. The section is closed with some remarks
on modelling of non-linear circuits with memory.
2.5.1 Behavioural vs. Physical Models

From the system identification point of view, physical and behavioural models can be distinguished [Sch09].
Physical models, based on equations related to voltage and current, require significant knowledge about the front-ends analogue components, as well as the relationship and theoretical
rules describing their interaction. They are used in circuit-level simulations to simulate the
behaviour under all kinds of excitations and environmental conditions, but are very timeconsuming as they need to handle true RF (bandpass) modulated signals. Thereby, circuitlevel simulations account for possible harmonic content and distinct time scales for RF carrier
and BB information. For example, according to the Nyquist theorem, a quasi-analogue signal representation of a 20 MHz wide signal in a 2 GHz RF band will require sampling with at
least 4 GHz, causing a huge amount of samples to be processed in the circuit-level simulation.
Two types of physical models can be further distinguished: (i) equivalent circuit models and
(ii) circuit-level behavioural models, both working with voltages and currents or incident and
23
reflected waves. The latter one also includes possible interstage mismatches in the input and
output ports. The most recent circuit-level model is the X-parameter model as a non-linear
extension of S-parameters, invented by Agilent Technologies Inc. [Vye10; VRW+05].
Behavioural models are extracted from input-output observations and enable complete
system-level simulations. They are preferred in most cases, especially when no equivalent
circuit description is available [Sch09]. However, simulation results are sensitive to the model
structure and the parameter extraction method, thus, behavioural models have a drawback
of generalisation. In other words, they are accurate for the data set used for its extraction
and most likely for the same excitation class, but are not guaranteed to provide useful results
for a different data set [TGT+96]. For example, applying a broadband input to a model
that has been extracted with narrowband excitation intuitively produces wrong results and
is equivalent to assume the superposition principle for non-linear systems which is of course
wrong [Sch09]. It is likely that there are unknown aspects of response-based behavioural
models as technological attributes are omitted in the model. The closer the model matches
the real front-end architecture, the more robust and accurate will be the prediction of the
behavioural activity under all kinds of excitations and environmental conditions. A detailed
classification of behavioural models according to application, model structure, and amplifier
physics is given in [Sch09].
A major distinction of behavioural models is made between lowpass or bandpass equivalent
models [Sch09]. Physical (circuit) models work directly with the actual RF signal which
contains the full RF circuits bandpass nature. Lowpass or BB equivalent modelling processes
only the complex envelope of the actual information signal, its carrier frequency is basically
neglected. This type of system-level modelling is computationally efficient and matches also
to the application of mitigating non-linear distortion at system level.
A further classification of behavioural models is conducted based upon the memory characteristics of the underlying non-linear component into:
memoryless (only AM/AM),
quasi-memory (AM/AM and AM/PM), and
memory models [Sch09].
Memoryless models consider only static and frequency-independent AM/AM distortion, thus,
no previous values are considered. Quasi-memory models additionally include AM/PM distortion that manifest instantaneous memoryless non-linear phase distortion effects. Examples for memoryless models are complex power series (polynomials), Saleh model and
modifications of it, Fourier series model, and Bessel-Fourier model, among others [Sch09].
24

Memory models consider also the dependency of current output by previous input and are
often applied for wideband excitation signals. Typical candidates for models with memory
are Volterra series, memory polynomial, and the Wiener/Hammerstein model [Sch09].
2.5.2 Modelling by Memoryless Polynomials

The memoryless polynomial is applicable for many non-linearity related issues [AMV10a]
and often provides an acceptable level of accuracy in many applications. It allows for easy
computational implementation and very effective system simulations, thereby it found good
use in engineering design and analysis. An extension to model strong or higher-order nonlinearities is easily achieved.
However, the polynomial model is subject to various limitations. First, it is only valid for
a small fitting range due to the Taylor series development around a specific operating point
with small-signal excitation. Thus, the front-end should operate in the linear or mildly
non-linear regions. Applying the model for strongly non-linear regions will result in a high
modelling error [Sch09]. In fact, the limitation for large-signal operation is very problematic,
as complex input signals with a significant envelope PAPR are often to be processed in
modern RF air interfaces. Second, the memoryless assumption is valid only for one frequency
at a time, i.e. for narrowband systems where the RF carrier frequency is much larger than
the desired signal bandwidth. Although the bandwidth dependency may be negligible, longterm memory effects can still appear [MKH+10; Sch09]. By strict definition, the memoryless
assumption holds only for circuits without inductors or capacitors, so that voltages and
currents are not depending on previous values. In practice, all active circuits manifest
memory, e.g. due to transistors equivalent schemes and matching and bias networks [Sch09].
The memoryless polynomial approximation is acceptable under the assumption that the
front-end manifests no or few memory effects, whose duration is close to the period of the
RF carrier [Sch09]. In addition, the input signal should be narrowband and comprise a
constant envelope in the band of interest.
The choice of memoryless polynomials for modelling weakly non-linear systems is mainly
application-driven. From the mitigation point of view, a simple model is required that
identifies the origin of new frequency components. After the source and process of the
unwanted signal is known, a mitigation algorithm can be formulated. That is, reproducing
the most important non-linear behaviour of the front-end is sufficient for the algorithm, thus,
a simple power series is chosen.
25
From the system-level point of view on memoryless non-linear models, the output envelope
simultaneously reacts to the input envelope, represented by amplitude and phase. That is,
there is a static non-linear relation between the input x(t) and the output y(t) given by
(2.4)
y(t) = G [x(t)] .
Here, G denotes the complex transfer function of the non-linear device and can be written
as
G(A) = g(A)ej(A) = P (A) + jQ(A),
(2.5)
where g(A) is the AM/AM distortion and (A) represents the AM/PM distortion. The
complex BB signal is represented by
x(t) = A(t)ej(t) = xI (t) + j xQ (t),
(2.6)
where A(t) and (t) are the general envelope amplitude and phase components, and xI and
xQ are the I and Q-component of the excitation signal. Equation (2.5) is also referred to as
quadrature model, constituting a general form of a memoryless non-linear model [Sch09]. A
functional schematic of such a quadrature model is depicted in Figure 2.12.
P (A)
x(t)
y(t)
90
Q(A)
Figure 2.12: Quadrature model of a memoryless non-linear model according to [Sch09].
An Nth order complex power series memoryless model is expressed as

y(t) =
N
X
cn xn (t) = c1 x(t) + c2 x2 (t) + c3 x3 (t) + . . . + cN xN (t),
(2.7)
n=1
where cn are complex-valued coefficients and x(t) being a real-valued input (e.g. I- or Q
component of complex envelope x(t)). Equation (2.7) represents a memoryless polynomial
at RF passband. Here, the output signal as polynomial expansion of the instantaneous input
has been truncated after N coefficients. With complex coefficients cn in (2.7), phase shift
in narrowband systems is also modelled, thus, also memory effects are considered to some
26

extent (quasi-memory model) [VR03]. If these coefficients are real, only AM/AM distortion
would be considered.
Equation (2.7) can be transformed to its equivalent BB representation with the complex
envelope input x(t) to [KK01; Teh09]
y(t) =
N
X
n=1
cn x(t) |x(t)|n1 ,
(2.8)
by assuming that some zonal filtering is adopted, hence, only odd-order distortion terms
around the carrier frequency fc contribute to the output signal [MMK+06]. The coefficients
are then given by

1
n
cn = n1 n+1 cn .
(2.9)
2
2
A detailed derivation is given in Section A.1.
Typically, real-valued coefficients are assumed for the polynomial (2.7), where relations with
the intercept points can be easily deduced [ZMS09; Che11]. The coefficients can be derived
by applying the definition of the intercept points, namely the equality between the linear
term and the IMD2 and IMD3 term, respectively. The following derivation is based on a
two-tone excitation of a non-linear device of degree 3. Assuming a linear gain of
G|dB = 20 log10 c1 ,
(2.10)
the corresponding relations of input-referred second-order intercept point (IIP2) and IIP3
are
c1 AIIP2 = c2 A2IIP2 ,
3
c1 AIIP3 = c3 A3IIP3 ,
4
(2.11)
(2.12)
AIIP2 and AIIP3 are the peak voltage levels at the corresponding power levels of the intercept
points that cannot be reached in practice. The coefficients on the right result from the square
and cubic operation (cp. Table 2.2). Converting IIP2 and IIP3 powers in dBm to voltages
and inserting (2.11) and (2.10) yields the desired relation between the intercept points and
27
the polynomial coefficients
IIP2|dBm
IIP3|dBm

A2IIP2
c21
= 10 log10
= 10 log10 2
,
2 R 103
c2 2 R 103

4 c1
A2IIP3
= 10 log10
,
= 10 log10
2 R 103
c3 3 2 R 103
(2.13)
(2.14)
where R is the reference resistance for converting W to V and is assumed to be 1 for sake
of simplicity (system-theoretic resistance). For extracting the model coefficients out of the
intercept points, inverse formulas are desirable
G
c1 = 10 20 ,
s
c2 =
(2.15)
c21
IIP2 ,
2 R 103 10 10
4 c1
c3 = ()
IIP3 .
3 2 R 103 10 10
(2.16)
(2.17)
Note that the third-order coefficient c3 is always negative. This kind of model extraction
based on intercept points will be discussed in Section 2.7.2.
In the following subsections, BB equivalent modelling with memoryless polynomials for individual receiver components as well as the total RF chain of a typical DCR, as illustrated in
Figure 2.6, is discussed. This approach has not been considered in the state-of-the-art literature so far. Indeed, the RF front-end is generally modelled to be ideal when evaluating the
performance of a communication system, and the receiver non-linearity is often just modelled
as an additive white Gaussian noise (AWGN) source [Sch08]. Moreover, all the models are
originally introduced for PAs, because the PA non-linearity at the transmitter is considered
to be the major source of non-linearity in the whole transmission chain [Sch08]. As seen
in the next sections, modelling of receiver RF front-ends can be fairly different to a single
transmitter amplifier. For instance, the impact of frequency conversion by the mixer is considered therein for developing appropriate BB representations of the distortion terms. This
novel approach of modelling the complete DCR chain has been published in [GAM+14].
2.5.3 Modelling of RF Non-linearity

Next, a BB representation of the distortion caused by the RF amplifier is developed, i.e.
distortion generated by the LNA of a DCR with a low-IF architecture as introduced in
Section 2.1.3. The LNA constitutes the only non-linear component in such an RF chain that
28

operates with the true RF bandpass signal and is highlighted in blue colour in Figure 2.13.
The real-valued RF bandpass signal can be written as
LP
VGA
ANT
LNA
LP
yRF
(t)
xRF (t)
RF non-linearity
ADC
LO
90
BP
ADC
VGA
yRF (t)
ejc t
Figure 2.13: Modelling of a non-linear LNA adopted in a low-IF receiver.

xRF (t) = 2 Re x(t)ejc t = x (t) ejc t + x (t)ejc t ,
(2.18)
where c = 2fc is the angular RF carrier frequency and (.) denotes complex conjugate.
Note that x(t) may contain multiple carriers of multiple co-existing radio access technologies
at different complex IFs, especially the desired signal among other strong interferers (see
Section 2.4). The RF non-linearity is then modelled by a memoryless polynomial according
to (2.7), outputting the signal
yRF (t) = c1 xRF (t) + c2 x2RF (t) + c3 x3RF (t) + . . . + cN xN
RF (t)
(2.19)
that is still including HD and IMD at passband [Sch09]. Now, some significant simplifications
are made that simplify the derivation of a BB representation of (2.19).
29
First, the RF carrier frequency fc is typically significantly higher than the maximum envelope
frequency of x(t) [Sch09]. That is, harmonics and even-order distortion are likely to be filtered
out when translated to BB, obtained by the so called zonal filters (typically lowpass)
adopted in the analogue or digital domain at BB [Sch09]. Moreover, signals may also be out
of the down-conversion bandwidth of the mixer.
Second, due to typically low IIP2 values, even-order distortion is very weak and generates
signal components near DC or at frequencies very far from fc [Raz97; Raz09]. They are
not harmful, except if the RF front-end is extremely wideband and xRF consists of multiple
strong signals that occupy a large bandwidth. Even-order distortion in the DC zone may
leak through the mixer to BB without frequency translation, as discussed in Section 2.2.
However, these issues have been reduced significantly due to integrated circuit design and
differential signalling [Kar01]. They can also be effectively filtered out by alternating current
coupling or filtering. Instead, odd-order distortion products can fall into the original signal
band (fundamental zone) as depicted in Figure 2.8. In practice, the third-order non-linearity
is usually the strongest and the only one that generates distortion that clearly appears above
the noise level. Thus, (2.19) can be simplified to
yRF
(t) = a1 xRF (t) + a2 x3RF (t),
(2.20)
where a1 = c1 denotes the linear gain and a2 = c3 the third-order coefficient of the RF
amplifier. In fact, operating at large-signal conditions demand for larger number of terms,
however, higher-order coefficients are difficult to extract as they cannot be separated out
from lower-order coefficients. In other words, many terms of the complex power series (2.19)
give simultaneous spectral contribution. The coefficients a1 and a2 are chosen to be complex
to model also static AM/PM distortion. By inserting (2.18) into (2.20), a further analysis
of the new frequency components can be performed:

yRF
(t) = a1 x(t) + 3 a2 x2 (t)x (t) ejc t

+ a1 x (t) + 3 a2 x (t) x2 (t) ejc t
+ a2 x3 (t)ej 3c t
+ a2 x3 (t)ej 3c t .
(2.21)
Only the first term ejc t will remain after mixing and zonal filtering at BB, i.e. by multiplying
(2.21) with ejc t as illustrated in Figure 2.13. The BB representation of (2.20) after zonal
30

filtering is therefore
yRF (t) = a1 x(t) + 3a2 x2 (t)x (t)
= a1 x(t) + 3a2 A2 (t)x(t)
(2.22)
= a1 x(t) + 3a2 z(t).
A block scheme of the RF non-linearity model, derived from (2.22), is depicted in Figure 2.14.
x(t)
a1
(.)2 (.)
y(t)
3a2
Figure 2.14: Block scheme of the RF non-linearity model derived from (2.22).
The second form comprises a notation with the envelope x(t)x (t) = |x(t)|2 = A2 (t). This
exactly matches to the complex polynomial notation of (2.8) for n = 1, 3. Indeed, same
assumptions about filtering have been assumed in the derivation (cp. Section A.1). In the
last form, z(t) is introduced to enhance readability for the following joint modelling of RF
and BB non-linearity. Expanding (2.22) with (2.6) and separating in-phase and quadrature
component leads to

yRF,I (t) = a1 xI (t) + 3a2 x3I (t) + xI (t)x2Q (t)

yRF,Q (t) = a1 xQ (t) + 3a2 x3Q (t) + xQ (t)x2I (t) .
(2.23)
(2.24)
Next, a two-tone simulation has been conducted to illustrate the derivation. The complexvalued BB signal x(t) according to (2.6) is
x(t) = ej1 t + ej2 t ,
(2.25)
with the tones frequencies f1 = 6.4 MHz and f2 = 7.9 MHz relative to the carrier frequency
fc = 50 MHz. The RF spectrum is depicted by Figure 2.15(a), where only the RF carrier
of the two-tone signal is marked. Both tones have an equal power of 23 dBm, resulting in
a total power of 20 dBm. The noise power has been set to a particularly small value of
about 200 dBm in order to make also weak IMD after subsequent distortion at BB visible.
31
fc
PSD (dBm/Hz)
PSD (dBm/Hz)
40
20
0
fc
20
40
60
80
100
120
140
160
180
100 80 60 40 20
20
40
0
RF Frequency (MHz)
(a) Real-valued RF passband signal with two-tone signal at
trum.
40
20
0
20
40
60
80
100
120
140
160
180
200
(3fc + 3f1 )
80
100
both sides of the spec-
3fc + 3f1
100
Fundamental
3rd order IMD
15
2f1 f2
f1
f2
f1 + 2f2
PSD (dBm/Hz)
50
0
50
100
150
RF Frequency (MHz)
(b) RF passband distortion with zonal filter marked in green colour.
40
20
0
20
40
60
80
100
120
140
160
180
20
150
60
10
0
5
10
15
20
5
Baseband Frequency (MHz)
(c) BB equivalent representation of RF distortion after zonal filtering.
Figure 2.15: RF distortion caused by two-tone input.
32

Note that the noise power distributes over the full BB bandwidth. By substituting (2.25) in
(2.22) and sorting the terms yields the signal components in Table 2.1.
Table 2.1: Frequency components generated by RF non-linearity.
Coefficient
Frequency
Type
9 a2 + a1
+1
Fundamental
9 a2 + a1
+2
Fundamental
3 a2
22 1
3rd order IMD
3 a2
21 2
3rd order IMD
Note that the amplitude of the fundamental frequencies is, beside linear gain a1 , weighted by
the third-order coefficient a2 which is always negative and causes the compression behaviour.
Distortion on top of the actual input signal is also denoted as in-band distortion from the
perspective/bandwidth of the input signal. The RF distortion at passband is illustrated
in Figure 2.15(b), assuming a1 = 2 and a2 = 1000, that corresponds to GRF = 6 dB
and IIP3RF = +6 dBm according to Equations (2.10) and (2.14). A bit exaggerated value
is chosen for the third-order coefficient a2 of the RF non-linearity for the sake of better
visualisation. The band of interest after zonal filtering (green frame in Figure 2.15(b)) is
illustrated in Figure 2.15(c). Hence, only the third-order IMD around the fundamental
frequencies is left from the distortion introduced by a single RF amplifier (LNA).
2.5.4 Modelling of BB Non-linearity

Now, modelling of sole BB non-linearity is discussed. Contrary to RF non-linearity, there are
physically separated I and Q branches at BB that manifest a different non-linear behaviour.
Physical sources of BB non-linearity in both I and Q branches are the mixer, BB amplifier
(variable-gain amplifier (VGA)) or ADC driver, and the ADC , as illustrated in Figure 2.16.
The total distorted BB signal
yBB (t) = yBB,I(t) + jyBB,Q (t)
(2.26)
can be modelled by two individual memoryless polynomial models

2
3
N
yBB,I(t) = c1,I yRF,I(t) + c2,I yRF,I
(t) + c3,I yRF,I
(t) + . . . + cN ,I yRF,I
(t)
2
3
N
yBB,Q (t) = c1,Q yRF,Q (t) + c2,Q yRF,Q
(t) + c3,Q yRF,Q
(t) + . . . + cN ,Q yRF,Q
(t),
33
(2.27)
(2.28)
where yRF,I = a1 xI and yRF,Q = a1 xQ , i.e. fully linear amplification at the RF amplifier is
assumed for a moment. Indeed, the non-linear characteristics of the I and Q channels can
be modelled as a sum of two bandpass non-linearities [LHG+10]. Following the modelling
approach for the RF non-linearity, similar simplifications are made. First, even-order distortion is assumed to be very weak due to efficient circuit design techniques such as differential
signalling [Kar01]. In typical DCRs, symmetrical electric wiring is adopted behind the LNA,
i.e. by using two signals V+ and V that are ideally 180 out of phase, i.e. V+ = V .
Following [Kar01], the output or differential voltage can be written as
Vdiff = V+ V ,
(2.29)
where the time-dependency of the voltages is omitted for a moment. Then, assuming perfectly balanced signals, both voltages can be written as
V+ = c1 V+ + c2 V+2 + c3 V+3 + . . . + cN V+N
(2.30)
V = c1 (V+ ) + c2 (V+ )2 + c3 (V+ )3 + . . . + cN (V+ )N ,
(2.31)
according to (2.27) or (2.28). Note that the pair (2.30) and (2.31) have the same form for
both I and Q component, so appropriate subscripts are omitted here. Eventually, the odd
order terms keep their sign, but the even-order distortion is always positive. Computing Vdiff
according to (2.29) yields
Vdiff
= V+ V = 2c1 V+ + 2c3 V+3 + . . . + 2c2n+1 V+2n+1 .
LP
VGA
ANT
ADC
ADC
LO
90
BP
(2.32)
LNA
LP
VGA
Figure 2.16: Modelling of non-linear BB components in a low-IF receiver.
34

That is, the even-order terms cancel out, but the odd-order terms increase by a factor
two [Kar01]. However, perfect even-order cancellation by differential signalling requires
symmetrical layout and matched amplifiers to ensure perfect balance among the signals.
The minor importance is also justified by measurements presented in Section 2.7. Moreover,
higher-order terms are typically masked by noise. Therefore, it is assumed that third-order
non-linearity is describing the most significant distortion, hence, (2.27) and (2.28) can be
simplified to:
3
yBB,I
(t) = a3,I yRF,I (t) + a4,I yRF,I
(t)
(2.33)
3
yBB,Q
(t) = a3,Q yRF,Q (t) + a4,Q yRF,Q
(t),
(2.34)
where a3,I = c1,I , a3,Q = c1,Q , a4,I = c3,I , and a4,Q = c3,Q for notational convenience in
the following derivation including both RF and BB distortion. A block scheme of the BB
non-linearity model is depicted in Figure 2.17.
yRF
yBB
a3,I
Re
(.)3
Im
a4,I
a3,Q
(.)3
a4,Q
Figure 2.17: Block scheme of the BB non-linearity model employing a cubic term separately
for the I and Q part.
Assuming a two-tone down-converted signal according to (2.25), the output components
after passing BB non-linearity are as listed in Table 2.2.
The BB distortion is illustrated in Figure 2.18, assuming a3,I = a3,Q = 2 and a4,I = a4,Q =
1000 similar to the RF distortion case. Beside IMD, the BB non-linearity generates also
HD directly in the BB on the left side of the spectrum at 31,2 , if the strong complex
carriers fall onto low (non-zero) IFs. Thus, HD cannot be neglected in the discussion as for
the RF non-linearity. Moreover, it is noteworthy that the BB non-linearity causes third-
35
Table 2.2: Frequency components generated by BB non-linearity.
Coefficient
Frequency
Type
9
4
9
4
a3 + a4
+1
Fundamental
a3 + a4
+2
Fundamental
a4
31
3rd order HD
21 2
3rd order IMD
(22 + 1 )
3rd order IMD
1
4
1
4
3
4
3
4
3
4
3
4
a4
a4
a4
a4
a4
32
22 1
(21 + 2 )
3rd order HD
3rd order IMD
3rd order IMD
50
2 f1 f2
f1
f2
f1 + 2 f2
PSD (dBm/Hz)
3 f2
(f1 + 2 f2 )
(2 f1 + f2 )
3 f1
50
Fundamental
3rd HD
3rd order IMD
100
150
200
30
20
10
0
10
20
30
Figure 2.18: BB distortion caused by two-tone input generates additional frequency components in the fundamental and third harmonic zone.
order IMD in the fundamental zone, at exactly the same frequencies of the RF distortion.
Considering the total receiver line-up of a DCR, both RF and BB distortion will sum up in
the fundamental zone, as will be shown later.
Considering only the third-order of the received signal is the main approach followed in the
state-of-the-art literature, such as in [GSH+12a; VSHGA+06]. However, it is restricted for
modelling non-linear systems at equivalent BB. Down-conversion by the mixer and subsequent distortion of RF distortions at BB are not considered therein that is one of the research
gaps tackled in this thesis and being on focus in the next section.
36
2.5.5 Joint Modelling of RF and BB Non-linearity

Now, a cascade of RF and BB models is considered, i.e. both models depicted in Figure 2.14
and Figure 2.17 are connected in series. In practice, the non-linearities of the RF stage are
transferred to the BB receiver stages, thereby the order of the final distortion and the number
of IMD products is increased. In fact, adopting a cascade of two third-order polynomials
matches to the physical architecture of a DCR. By splitting (2.22) into I and Q component
and substituting them in (2.33) and (2.34) yields
yBB,I
(t) = a3,I a1 xI (t) + 3a3,I a2 zI (t)
+ a4,I a1 3 x3I (t) + 9a4,I a1 2 a2 x2I (t)zI (t)

+ 27a4,I a1 a2 2 xI (t)zI2 (t) + 27a4,I a2 3 zI3 (t),
(2.35)
and
yBB,Q
(t) = a3,Q a1 xQ (t) + 3a3,Q a2 zQ (t)
+ a4Q a1 3 x3Q (t) + 9a4Q a1 2 a2 x2Q (t)zQ (t)

2
3
+ 27a4Q a1 a2 2 xQ (t)zQ
(t) + 27a4Q a2 3 zQ
(t).
(2.36)
Equations (2.35) and (2.36) can be further opened by applying (2.23) and (2.24). Hence, a
BB representation of the total RF and BB distortion is obtained with relation to the input
signal x(t) (2.6):
yBB,I
(t) = a3,I a1 xI (t) + 3 a3,I a2 x3I (t) + 3 a3,I a2 xI (t)x2Q (t)
+ 27 a4,I a32 x9I (t) + 81 a4,I a32 x7I (t)x2Q (t)

+ 27 a4,I a22 a1 x7I (t) + 81 a4a32 x5I (t)x4Q (t)
+ 54 a4,I a22 a1 x5I (t)x2Q (t) + 9 a4,I a2 a21 x5I (t)
+ 27 a4,I a32 x3I (t)x6Q (t) + 27 a4,I a22 a1 x3I (t)x4Q (t)
+ 9 a4,I a2 a21 x3I (t)x2Q (t) + a4 a31 x3I (t),
(2.37)
and
yBB,Q
(t) = a3,Q a1 xQ (t) + 3 a3,Q a2 x3Q (t) + 3 a3,Q a2 xQ (t)x2I
+ 27 a4,Q a32 x9Q (t) + 81 a4,Q a32 x7Q (t)x2I (t)

+ 27 a4,Q a22 a1 x7Q (t) + 81 a4,Q a32 x5Q (t)x4I (t)
+ 54 a4,Q a22 a1 x5Q (t)x2I (t) + 9 a4 a2 a21 x5Q (t)
+ 27 a4 a32 x3Q (t)x6I (t) + 27 a4,Q a22 a1 x3Q (t)x4I (t)
+ 9 a4,Q a2 a21 x3Q (t)x2I (t) + a4,Q a31 x3Q (t).
37
(2.38)
Note that due to the cascade of two third-order polynomials, i.e. two non-linear devices
of degree three in series, up to ninth order IMD is considered. The equations (2.37) and
(2.38) clearly show the interaction between the RF and BB stages. Figure 2.19 depicts a
simplified block scheme of the joint model considering RF and BB distortion as a cascade of
two third-order polynomials.
ejc t
LNA
xRF (t)
(.)
BB
yRF
(t)
yRF (t)
(.)
yBB
(t)
Figure 2.19: Model of joint RF and BB distortion using a cascade of two third-order
polynomials.
Again, a two-tone excitation is assumed and signal components generated by the cascaded
non-linearity model are analysed. The signal yBB

(t) after passing the cascaded non-linearity
in the complex BB are listed in Table 2.3.
Figure 2.20 illustrates the complex BB spectrum with a two-tone input, where a =
[a1 , a2 , a3 , a4 ] = [2, 1000, 2, 1000]. In fact, there are 20 signal components in total al-
50
100
150
200
30
20
4 f2 + 5 f1
3 f2 + 4 f1
2 f2 + 3 f1
2 f1 f2
f1
f2
f1 + 2 f2
(2 f1 3 f2 )
(3 f1 4 f2 )
(4 f1 5 f2 )
PSD (dBm/Hz)
6 f2 + 3 f1
5 f2 + 2 f1
f1 4 f2
3 f2
(f1 + 2 f2 )
(2 f1 + f2 )
3 f1
f2 4 f1
(5 f1 2 f2 )
(6 f1 3 f2 )
50
10
0
10
Fundamental
3rd HD
3rd order IMD
5th order IMD
7th order IMD
9th order IMD
20
30
Figure 2.20: Signal components of cascaded non-linearity with two-tone input.

though there was only a two-tone (two signal components) input. However, the magnitude of the IMD products decrease with increasing order of non-linearity. Noise has
38

Table 2.3: Frequency components generated by cascaded RF and BB non-linearity.
Coefficient
9
a a 3 + 5103
a4 a2 3 +a3 a1 +9 a3 a2 + 135
a4 a1 2 a2 + 2835
a4 a1 a2 2
4 4 1
2
2
4
9
a a 3 + 5103
a4 a2 3 +a3 a1 +9 a3 a2 + 135
a4 a1 2 a2 + 2835
a4 a1 a2 2
4 4 1
2
2
4
1
a a 3 + 567 a4 a2 3 + 567
a4 a1 a2 2 + 45
a a 2a
4 4 1
4
4 4 1 2
1
a a 3 + 567 a4 a2 3 + 567
a4 a1 a2 2 + 45
a a 2a
4 4 1
4
4 4 1 2
3
a a 3 + 1701 a4a2 3 + 135
a4 a1 2 a2 + 1701
a4 a1 a2 2 + 3 a3 a2
4 4 1
4
4
3
a a 3 + 1701 a4a2 3 + 135
a4 a21 a2 + 1701
a4 a1 a2 2 + 3 a3 a2
4 4 1
4
4
1701
a4 a2 3 + 43 a4 a1 3 + 45
a a 2 a + 945
a4 a1 a2 2
2
2 4 1 2
4
1701
, a4 a2 3 + 43 a4 a1 3 + 45
a a 2 a + 945
a4 a1 a2 2
2
2 4 1 2
4
9
a a 2 a + 189
a4 a1 a22 + 243 a4 a2 3
4 4 1 2
4
9
a a 2 a + 189
a4 a1 a2 2 + 243 a4a2 3
4 4 1 2
4
27
a a 2 a + 567
a4 a1 a2 2 + 729 a4a2 3
4 4 1 2
4
27
a a 2 a + 567
a4 a1 a2 2 + 729 a4a2 3
4 4 1 2
4
81
, a4 a1 a2 2 + 729
a4 a2 3
4
4
81
4
27
4
27
4
81
4
81
4
27
4
27
4
a4 a1 a2 2 +
a4 a1 a2 2 +
a4 a1 a2 2 +
729
4
243
4
243
4
Frequency
Type
+1
Fundamental
+2
Fundamental
31
3rd order HD
21 2
3rd order IMD
32
22 1
3rd order IMD
(21 + 2 )
3rd order IMD
(22 + 1 )
a4 a2 3
a4 a2 3
a4 a2 3
a4 a2 3
a4 a2 3
a4 a2 3
3rd order IMD
1 42
5th order IMD
21 + 32
5th order IMD
31 + 42
7th order IMD
2 41
a4 a2 3
3rd order HD
5th order IMD
22 + 31
5th order IMD
32 + 41
7th order IMD
51 + 22
7th order IMD
42 + 51
9th order IMD
62 + 31
9th order IMD
52 + 21
7th order IMD
41 + 52
9th order IMD
61 + 32
9th order IMD
been excluded in this particular simulation to make also high-order distortion visible. In
practice, 7th- and 9th-order non-linearity are below the receiver noise level or beyond the
signal-to-noise ratio (SNR) of the ADC. The SNR of a typical ADC is approx. 6080 dB,
hence, making all signal components below 80 dBFS to disappear due to the quantisation
noise.
Moreover, it is visible that the level of third-order IMD in the fundamental and third harmonic zone do not match. That is, RF non-linearity initially generated some additional
third-order distortion power around the fundamental frequencies as already seen in Figure 2.15(c).
39
2.5.6 Interaction with I/Q Imbalance

Beside RF and BB non-linearity, I/Q imbalance of the mixer and the individual BB branches
cause mirror signals that may interfere with other useful signals. Especially the interaction
between non-linear distortion and the I/Q imbalance effects plays a special role when dealing
with DCRs [GAM+14]. More details on I/Q imbalance effects and appropriate mitigation
techniques can be found in [Ant11; AVR08].
Up to now, ideal down-conversion at the mixer has been assumed. In practice, a quadrature
mixer cannot provide exactly 90 phase shift, but sustains a phase mismatch of m (rad).
In addition, the I and Q branches manifest a certain amplitude mismatch gm . The I/Q
imbalance of the down-conversion stage can be modelled as
yRF (t) = k1 yRF (t) + k2 yRF

(t),
(2.39)
where the complex mismatch coefficients are

k1 = (1 + gm ejm )/2,
(2.40)
k2 = (1 gm e+jm )/2.
(2.41)
Figure 2.21 depicts the mixer I/Q imbalance model. In the previous derivations, perfect I/Q
yRF
k1
(.)
yRF
k2
Figure 2.21: Block scheme of the mixer I/Q imbalance model.

balance has been assumed, i.e. gm = 1, m = 0, hence k1 = 1, k2 = 0 and yRF (t) = yRF (t).
Next, a complex representation of the BB distortion under presence of I/Q imbalance is
derived [GAM+14].
Equations (2.33) and (2.34) can be combined to
yBB
(t) = yBB,I
(t) + j yBB,Q
(t)

3
3
= a3,I yRF,I (t) + a4,I yRF,I
(t) + j a3,Q yRF,Q (t) + a4,Q yRF,Q
(t) ,
40
(2.42)

with yRF,I/Q including RF non-linearity and I/Q imbalance. Real and imaginary part of
yRF,I/Q can be written as
yRF (t) + yRF

(t)
,
2
yRF (t) yRF

(t)
,
=
2j
yRF,I =
yRF,Q
yielding the BB distortion in complex notation:

a3,I + a3,Q
a3,I a3,Q
yRF (t) +
yRF (t)
2
2
a4,I + a4,Q 3
3a4,I 3a4,Q 2
+
yRF (t) +
yRF (t)
yRF
(t)
8
8
a4,I + a4,Q
3a4,I 3a4,Q
yRF (t) [
yRF
(t)]2 +
[
yRF (t)]3 .
+
8
8
yBB
(t) =
(2.43)
This equation needs to be further expanded with (2.39) to get the complete representation
of the BB distortion including mixer I/Q imbalance (k1 6= 1, k2 6= 0) and further imbalance
between the I and Q receiver chains (a3,I 6= a3,Q and a4,I 6= a4,Q ). The full form is given by
(t)
yBB

a3,I a3,Q
a3,I + a3,Q
=
k1 +
k2 yRF (t)
2
2

a3,I a3,Q
a3,I + a3,Q
k2 +
k1 yRF (t)
+
2
2

i
i
a4,I a4,Q h 3
a4,I + a4,Q h 3
2
2
3
+
yRF
(t)
k1 + 3k1 (k2 ) +
(k2 ) + 3k1 k2
8
8

i
i a +a h
a4,I a4,Q h 3
4,I
4,Q
2
3
2
+
[yRF
(t)]3
k2 + 3(k1 ) k2 +
(k1 ) + 3k1 k2
8
8

i
a4,I a4,Q h 2
3 k1 k2 + |k2 |2 k2 + 2|k1 |2 k2
+
8
i
a4,I + a4,Q h 2
2
2
2
+
3 k1 (k2 ) + |k1 | k1 + 2|k2 | k1 yRF
(t)yRF
(t)
8

i
a4,I a4,Q h
3 k1 k2 2 + |k1 |2 k1 + 2|k2 |2 k1
+
8
i
a4,I + a4,Q h 2
2
2
+
3 (k1 ) k2 + |k2 | k2 + 2|k1 | k2 yRF (t)[yRF
(t)]2 .
(2.44)
8
According to (2.22), yRF (t) contains the original signal x(t) and its RF distortion, hence, both
components have also mirror components stemming from yRF

(t), [yRF
(t)]2 , and [yRF
(t)]3 . The
2
other terms yRF

(t)yRF
(t) and yRF (t)[yRF
(t)]2 cause further IMD around the already existing
RF distortion. Note that (2.44) can be simplified to (2.33) and (2.34) assuming k1 = 1, k2 = 0
41
and separating in I and Q component. Assuming perfectly aligned I and Q paths (a3,I = a3,Q
and a4,I = a4,Q ) reduces (2.44) to
3
1
(t)]3 + a4 yRF
(t)yRF
(t).
yBB
(t) = a3 yRF (t) + a4 [yRF
4
4
(2.45)
In the following, two-tone simulations are performed to demonstrate the impact of the imbalance effects and their interaction with the RF and BB non-linearity. The used parameters
are summarised in Table 2.4. One important parameter is the IRR, the receivers ability to
suppress mirrors of the RF signal to be I/Q down-converted. The value is set to 40 dB, which
is a realistic value for a typical DCR as will be seen later in measurements. Moreover, a small
Table 2.4: I/Q imbalance simulation parameters
IRR 40 dB
a3,I
gm
0.99
a3,Q
2.2
0.0174 1
a4,I
1000
k1
k2
0.9949 j0.0086
a4,Q
0.0051 j0.0086
900
phase mismatch of the mixer has been assumed that results in complex mismatch coefficients
k1 and k2 according to (2.40) and (2.41). Slightly different third-order coefficients a3,I/Q and
a4,I/Q take also I/Q imbalance at the two physically separated I/Q paths into account.
After yRF (t) passes the mixer, mirror products of the fundamental tones and RF distortion
appear, as shown in Figure 2.22(a). Compared to Figure 2.15(c), only additional components due to the imbalance effects are highlighted. Feeding yRF (t) to the BB stages yields
the total RF and BB distortions under I/Q imbalance conditions according to (2.44). In
Figure 2.22(b), additional components compared to those in Figure 2.20 are highlighted.
Due to the imbalance effects, basically same frequency components are generated as listed
in Section 2.5.5, but all with negative sign (complex conjugates) and with reduced power
levels dictated by the IRR. In other words, the spectral gaps in Figure 2.8 are filled due
to the mirror effects. Note that also mixer non-linearity and further BB non-linearities can
be combined in the non-linear coefficients a3,I/Q and a4,I/Q . Different third-order coefficients
between the I and Q path result in different power levels of distortions, but same frequency
components are generated. It just denotes different scaling of the real and imaginary parts
that sum up in the complex-valued representation.
42
50
Mirror Fundamental
Mirror 3rd order IMD
f1 2f2
f2
f1
f2 2f1
PSD (dBm/Hz)
50
100
150
10
0
10
20
(a) RF distortion after passing a non-ideal mixer manifesting I/Q imbalance.
30
20
30
PSD (dBm/Hz)
0
50
100
150
200
250
30
20
6 f1 3 f2
5 f1 2 f2
f2 + 4 f1
3 f1
2 f1 + f2
f1 + 2 f2
3 f2
f1 + 4 f2
5 f2 2 f1
6 f2 3 f1
4 f1 5 f2
3 f1 4 f2
2 f1 + 3 f2
f1 2 f2
f2
f1
2 f1 + f2
2 f2 3 f1
3 f2 4 f1
4 f2 5 f1
50
10
0
10
30
20
Frequency (MHz)
(b) Further distortion at BB with I and Q paths incorporating different gain and third-order coefficients.
Figure 2.22: RF and BB non-linear distortion under I/Q imbalance conditions.
2.5.7 Modelling of Memory Effects

In case of wideband signals, the frequency-dependent behaviour of the front-end cannot be
neglected any more [Sch09]. That is, the limitations of memoryless polynomials reported
in Section 2.5.2 become problematic, if the signal bandwidth becomes comparable with the
bandwidth of the non-linear front-end. Memory effects can be itself linear or non-linear, and
short- or long-term, respectively [Sch09].
43
The most general model for describing both non-linearities and memory effects is the Volterra
series [VR03; Ken00; Pin01]. It provides a complete and reliable description of the non-linear
systems function by using multidimensional convolutions [Pin01]. A finite discrete-time
Volterra series model is given by [MMK+06]
y(s) =
yn (s) =
N
X
(2.46)
yn (s)
n=1
M
1
X
m1 =0
M
1
X
mn =0
hn (m1 , , mn )
n
Y
l=1
x(s ml ),
(2.47)
yn (s) being the n-dimensional convolution of the input with Volterra kernel hn . It is obtained
by combining the convolution sum of a linear time invariant (LTI) system, modelling the
memory, and the power series according to (2.7), modelling a memoryless non-linearity. Thus,
the set of Equations (2.46) and (2.47) is also called a power series with finite memory of length
M [Rou09]. The Volterra kernels denote different transfer functions (impulse responses) for
each order n. For n = 1, the kernel h1 (m1 ) coincides with a linear impulse response. Nonlinearity is characterised by any higher-order impulse response (n > 1), e.g. h2 (m1 , m2 ) and
h3 (m1 , m2 , m3 ) describe the second- and third-order non-linearity, respectively. However,
the Volterra model also has some drawbacks. First, the parameters increase with respect to
non-linear order N and memory depth M significantly, essentially complicating its practical
implementation. Second, high-order Volterra kernels are difficult to measure and convergence
of the Volterra series is also not guaranteed. Third, the Volterra model shows, due to
the Taylor series approximation, only good approximation at the point where it has been
extended, i.e. it suits for modelling weakly non-linear systems (mildly non-linear regimes)
[Sch09].
Next, some simpler memory structures are considered that model linear memory and constitute special cases of the general Volterra formulation.
Modelling of Linear Memory

In general, two-box, three-box, and parallel-cascade models are distinguished [Sch09]. Most
common simplified two-box structures of the Volterra approach are the Wiener and Hammerstein model [Sch09]. These are cascades of a static non-linear system (memoryless polynomial) and a memory system, implemented by a linear filter (LTI subsystem). If the linear
filter comes before the non-linearity, it is called Wiener model, otherwise Hammerstein. Both
structures are depicted in Figure 2.23(a) and Figure 2.23(b), respectively. The Wiener model
44
x(s)
yW (s)
LTI
FIR filter
Polynomial
(a) The Wiener model (linear non-linear).
x(s)
yH (s)
LTI
FIR filter
Polynomial
(b) The Hammerstein model (non-linear linear).
Figure 2.23: Two-box models as simplifications of the Volterra series model [Sch09].
is equivalent to taking just

hn (m1 , , mn ) = cn h(m1 ) h(mn ),
n = 1, 2, . . . N
(2.48)
in (2.47), hence, the model can be written as [MMK+06]

yW (s) =
N
X
cn
n=1
"M 1
X
m=0
h(m)x(s m)
#n
(2.49)
On the contrary, the Hammerstein model, which is formed by a static non-linearity followed
by a common linear filter h(m), can be written as
yH (s) =
M
1
X
m=0
h(m)
N
X
n=1
cn xn (s m).
(2.50)
m1 = m2 = = mn ,
otherwise,
(2.51)
Here, the equivalent Volterra kernels are given by

hn (m1 , , mn ) =
cn h(m)
0
i.e. the kernels are only non-zero along their diagonals [Sch09].
45
The Hammerstein model acquires special significance since this structure is used later in
the mitigation architecture (Chapter 4). Finally, all modelling of non-linearity is still
based on memoryless polynomials, but memory effects are considered by a cascade with
an adaptive filter (AF) manifesting the linear subsystem.
The model memory depth M determines the length of the finite impulse response (FIR) filter
adopted in these models. It can be estimated from the first-order Volterra kernel. There are
several approaches to obtain the linear and non-linear block coefficients, the most common
being the least squares (LS) method from broadband time-domain measurements. Details
on that and of further structures, such as the three-box Wiener-Hammerstein model or the
parallel-cascade models, can be found in [Sch09].
Modelling of Non-Linear Memory

The aforementioned Wiener and Hammerstein models assume that the qualitative shape of
the conversion does not change with the frequency. Hence, no interaction between the tones
of a wideband signal is considered, even though such a behaviour occurs in real devices with
finite memory [Sch09]. There are more accurate non-linear models that consider both linear
and non-linear memory effects, such as the memory polynomial [MMK+06; Dup12; ZWC10]
or the time-delay neural network (TDNN) [Sch09], among others.
The Hammerstein model can be generalized by choosing different filters for each non-linearity
order n [MMK+06]
yGH (s) =
N M
1
X
X
n=1 m=0
an,m xn (s m),
(2.52)
where {an,m } denotes a combination of different filter and polynomial coefficients cn into a
2D-array. By assuming that the signals bandwidth is small compared to the centre frequency
fc , the real input signal can be expressed by its complex BB representation x and yields the
memory polynomial, similar to (2.8),
yM P (s) =
1
N M
X
X
n=1 m=0
an,m x(s m) |x(s m)|n1 .
(2.53)
Note that (2.53) ends up in a memoryless polynomial model according to (2.8) for M =
0. The memory polynomial can be seen as a parallel structure of different Hammerstein
subsystems. Equation (2.53) can be simplified further by taking only the odd-order terms
2(n 1) as the even-order terms fall at frequencies very far from fc . A block scheme of a
memory polynomial is shown in Figure 2.24. The unit delay tap is depicted by the symbol
46
x(s)
a1,0 aN,0
z 1
x(s 1)
a1,1 aN,1
z 1
y(s)
z 1
x(s M)
Linear dynamic
subsystem
a1,M aN,M
Non-linear static
subsystem
Figure 2.24: Block scheme of a memory polynomial model [Sch09].
z 1 and introduces a delay of one sampling interval m (unit time delay). The total number of
parameters is N M, hence, easing the implementation of this model significantly compared
to other memory models [Dup12].
The complex polynomial coefficients an,m can be estimated by the LS method from L samples
of the complex envelope BB signal. Assuming that input and output signal are collected in
two vectors x = [x(0), x(1), . . . , x(L 1)]T and y = [y(0), y(1), . . . , y(L 1)]T , the matrix R
is given by
R = [R0 , R1 , . . . , Rm ] ,
h
i
Rm = xm , xm |xm | , xm |xm |2 , . . . , xm |xm |N 1 ,
xm = [x(0 m), x(1 m), . . . , x(L 1 m)]T ,
with m = 0 . . . M 1. Then, (2.53) can be written in vectorial notation as

y M P = Ra,
47
(2.54)
where a is the vector with the complex polynomial coefficients an,m arranged as a =
[a1,0 , . . . , aN,0 , a1,1 , . . . , aN,1 , . . . , a1,M . . . , aN,M ]T . The LS solution for (2.54) is
a = RH R
1
RH y,
(2.55)
where (.)H denotes the Hermitian transpose of the matrix R [Sch09; Dup12]. A development
of a memory polynomial model based on measured data is discussed in Section 2.7.3.
In the TDNN approach, the memoryless non-linear subsystem is modelled by an artificial
neural network. However, as it is trained to a given data set, its usability for other data is
questionable. The interested reader is referred to [Sch09].
2.6 Circuit-level Simulation with SimRF

Although mathematical behavioural modelling is on focus in this thesis, some abstract
circuit-level simulations have been conducted for the purpose of comparisons. In the following, a modern RF circuit simulation approach is presented that can be easily integrated
in a system-level simulation framework.
SimRF is an extension of MATrix LABoratory (MATLAB)s system-level simulation
tool Simulink and provides a dedicated library for modelling of RF systems including its impairments [Sim; Zuc13].
With traditional circuit models based on
simulation program with integrated circuit emphasis (SPICE), that process true RF passband signals, it is fairly impossible to assess the complete RF chain. Instead, a smarter way
is desirable to deal with the RF signals, i.e. to apply models at higher levels of abstraction
and solvers with a larger time steps. Circuit-envelope simulation, being one simulation mode
in SimRF beside traditional BB equivalent modelling, is a possible solution to reduce the
simulation time while taking into account multiple RF carriers at the input. SimRF enables
design and simulation of wireless transceivers with at higher level of abstraction by basic
blocks, such as mixers, amplifiers, and S-parameter blocks. There are input and output
ports that behave as gateways and allow to translate Simulink signals to voltages that have
a certain power and are centred around a certain carrier frequency. Thereby, SimRF blocks
can be easily connected to other Simulink blocks, such as signal generation or spectrum
indication (fast Fourier transform (FFT) sink).
Figure 2.25 shows an example of a SimRF model for the USRP WBX front-end. It is based
on the schematic [Wbx] and data sheets of its discrete components, for inputting realistic
parameters into the blocks. The total model needs to be refined into a chain of components
48
From workspace
input_signal
49
Solver
Config
f (x) = 0
SimRF
Import
SL RF
SimRF Blocks
LNA
Out
SimRF
Parameters
In
Gnd
Out Q
Mixer
RF In
Out I
Out
Out
IF-Amp
Quadrature
In
R2
IF-Amp
In-Phase R1
In
SimRF
Outport
Quadrature
RF SL
Gnd1
SimRF
Outport
In-Phase
RF SL
Re-Im to
Complex
Im
Re
To workspace
output_signal
2.6 Circuit-level Simulation with SimRF
Figure 2.25: Top-level Simulink model of the WBX front-end.
1
RF input
In
Out
LO
2
I
Mixer I
In
Out
In
Out
LO
3
Q
Mixer Q
Phase
Shift
LO
Figure 2.26: Simulink model of the mixer subsystem in Figure 2.25.

that actually build up the receiver. Essentially, the model represents the low-IF receiver
architecture from the RF down to the BB amplifier similar to Figure 2.6. However, ADC
and analogue BB filters are neglected in this simulation, as well as LO phase noise or mixer
leakage effects. Figure 2.26 depicts the mixer subsystem that is part of the top level model
Figure 2.25. It implements a quadrature I/Q mixer based on the mixer core blocks of
SimRF.
The SimRF simulation provides a realistic representation of the HD and IMD product levels.
I/Q imbalance effects as well as BB distortions can be easily tracked and match the output of
the mathematical cascaded model perfectly, as depicted in Figure 2.27 for a two-tone excitation. Compared to the fully BB equivalent mathematical model, the circuit-level simulation
has been simulated at fc = 920 MHz. Same input power, noise level, IRR and intercept
points for RF and BB amplifier have been assumed in both simulations. However, memory
effects have not been simulated and seem not to be included in the SimRF blocks itself.
In [Zuc13] it is said that purely static ninth order polynomial expressions are behind the
SimRF amplifier blocks, hence producing same distortion profiles. However, own blocks can
50
2.7 Measurement-based Analysis
Power (dBm)
20
40
TT Input
Cascaded Model
SimRF
RF+BB Distortions
Mirror-Frequency
BB Distortions
BB Distortions
(Mirror+TT)
60
80
100
120
10
5
0
5
10
Figure 2.27: Comparison between behavioural and circuit model.

be built by using the SimScape language. On the other hand, much more different RF impairments can be included in the SimRF simulation, e.g. the LO mixer feed-through by a
resistor between LO and RF input port of the mixer, or impedance mismatches between the
several stages. Thereby, severe imperfections can be modelled that go beyond the simple BB
equivalent mathematical model or are at least difficult to model at this level.
2.7.1 Model Identification Techniques
Beside physical and behavioural modelling, real RF device measurements with realistic excitation signals can be conducted. There are two approaches to derive a mathematical model
out of measured data, namely mathematical modelling and system identification [Sch09;
Pin01]. In mathematical modelling, basic laws of physics are used to describe the systems
dynamic behaviour. However, the few physical insights are unlikely to work even with simple
single-stage receivers that are on focus in this thesis. Instead, the DCR should be modelled
as a whole. The idea of system identification is to fit a model to the data extracted from
empirical measurements, therefore, it is also called direct empirical modelling [Sch09]. It
requires less physical insight and eases the construction, but is limited in validity.
51
System Identification Procedure
First, available a-priori knowledge concerning the RF chain and the desired model application is used to determine the type of excitation signal, the measurement setup including the
measurement procedure and the operating range. Here, the main goal is to extract a simple
model that describes the non-linear behaviour of the receiver to optimise the performance of
the radio by RF impairment mitigation techniques. In general, the test signal should excite
all possible states of the non-linear front-end (persistent excitation) [Sch09]. The numbers
of coefficients should be chosen in a way that enough degrees of freedom are available that
describe the full range of the front-end behaviour. After parameter extraction from the
measured data, the model is typically validated by using other data than in the extraction
process. There are some factors that may lead to a deficient model [Sch09]. The measurements may not provide sufficient information or are too noisy, e.g. if the ADC quantisation
noise covers higher-order distortion products. Next, the model may be inappropriate to represent the front-end behaviour, e.g. if the order is too low or too high. Finally, the model
identification algorithm itself may fail to extract the parameters correctly.
Excitation Signal
There are several different types of excitation signals that can be used for system identification techniques [Sch09]. These are single-tone, two-tone, multi-sine, band-limited noise,
pseudo-noise, or digitally modulated signals. In addition, the excitation signal is typically
swept over the input frequency and power range of the device under test (DUT) to capture
its complete behaviour.
The excitation signal class plays a key role for extracting a valid model. For instance, the
IMD caused by a two-tone interferer will differ from the IMD caused by a wideband digitally
modulated signal. The distortions, and hence the frequency response of the DUT, depend
on the amplitude distribution of the excitation signal [Pin01]. In order to get the best linear
approximation, it is important to use the same kind of excitation in terms of power spectrum
and amplitude distribution as will be applied later during operation of the radio [TGT+96].
Otherwise, the approximation can become invalid. Assuming a fully memoryless behaviour,
the excitation signal does not need to match the application signal, however, swept-frequency
response should be measured to establish the band limits of the memoryless property.
52
2.7.2 Two-tone Excitation

Performing a two-tone (TT) analysis to characterise the extent of circuit non-linearity is a
classical approach especially in RF and microwave engineering. In general, measuring the
generated spectra, i.e. HD and IMD, is easier than evaluating the variations of the fundamental signal, which is done for obtaining the AM/AM and AM/PM characteristics [VR03].
However, the analysis can become very complex if three or more tones are present at the input
of the non-linear DUT. Therefore, it is a common practice to limit the analysis to two tones
[Anr00]. A two-tone excitation signal is simple to create and shows a non-constant envelope
with a PAPR = 6 dB. On the other hand, two tones are not representative for wideband
communication signals in strict sense, as the amplitude distribution and power spectrum
do not match common communication signals [TGT+96]. Although memory effects remain
hidden due to the narrowband nature of the two-tone signal, bandwidth-dependent effects
can be recognized if the phase of the IMD product differs at low and high tone spacing
[VR03; Sch09].
Model Extraction
A memoryless polynomial model according to (2.7) can be directly extracted by the intercept
points obtained by prior RF measurements. Intercept points provide a good indication of the
degree of non-linearity under simple signal conditions [Ken00], whereas digitally modulated
signals allow only for little insight in the operation of the analysed system. The points are
typically found by extrapolation of the IMD power against different input powers. That is,
fundamental and IMD product are plotted in a power level diagram and intercept points are
found by graphical intersection of their extrapolated curves.
Measurement Setup
The two-tone measurement setup is sketched in Figure 2.28. The two-tone signal is generated by two individual signal generators, whose outputs are combined by a power splitter in reverse mode. A controllable attenuator is used to set the desired signal power
and to provide enough isolation of the individual tones. Finally, the signal is fed to
the USRP2+WBX, the non-linear DUT. Generators and attenuator are connected via
general purpose interface bus (GPIB), a control bus for instrumentation and measurement
that enables full-automated operation. Moreover, the setup enables coherent sampling by
53
R&S SMIQ06
f1 = fc + 570 kHz
at 14 dBm
attenuator
020 dB
power
combiner
RX
TX USRP2+
WBX
phase-locked
REF
R&S SMHU
ETH
PC
f2 = fc + 770 kHz at 14 dBm
GPIB
Figure 2.28: Sketch of the two-tone measurement setup.
phase synchronisation (10 MHz reference) between the generators and USRP. This is essential for avoiding spectral leakage and for obtaining absolute power levels. Note that the total
RF front-end non-linearity is captured as a whole and determined by the FFT spectrum of
the digitised BB signal. Thus, the sinusoidal frequencies are chosen as integer multiplies of
an FFT bin that represents the sampling grid in the frequency domain with
fbin =
fs
,
NFFT
(2.56)
where fs and NFFT denote the sampling frequency and the FFT length, respectively. Indeed, the FFT can be interpreted as a filter bank with the bandwidth fbin in each channel.
The higher the FFT length, the lower the channel bandwidth of the FFT a behaviour like
the resolution bandwidth of a spectrum analyser. By choosing the tone frequencies within
the FFT grid, its powers are concentrated on single FFT bins and allow for proper power
detection. Moreover, the frequencies should be kept small so all distortions, especially those
in the third harmonic zone, remain within the BB bandwidth. Finally, the distortion products at the sum- and difference frequencies of the two tones will also fall onto FFT bins.
In order to avoid spectral leakage, integer number of signal periods should be obtained, in
this case N tp = N 1/fbeat = 2N/(f1 + f2 ), where fbeat denotes the beat frequency of
the two-tone complex envelope. The spectral leakage can be further reduced by adopting a
54

window function, such as the Hann window. However, when applying the aforementioned
rules, almost no leakage will occur and windowing does not provide further improvement.
When aiming for characterizing the non-linearity of the DUT, all the measurement equipment
should be spurious-free [Anr00]. That is, any residual IMD by signal generator should be
avoided and appropriate filtering of the two-tone signal should be applied, if necessary. Next,
signal combiners with a good input-to-input isolation are required, as well as attenuators
for decoupling the generators. Before running measurements with the DUT, the spectral
purity of the generated signal is to be checked, e.g. with a conventional spectrum analyser.
In practice, the spurious-free dynamic range (SFDR) of the generators should exceed the
receiver dynamic range.
A general drawback of this two-tone measurement setup is the inability to characterise
AM/PM distortion. Extensions with vector network analysers and reference IMD generators,
as well as broadband setups similar to modern communication systems have been reported
in [Sch09].
For the actual RF measurements, the tone spacing was 200 kHz. A swept-tone measurement
has been conducted at frequencies across the entire operating band of the analogue front-end
WBX from 70 MHz to 2.2 GHz in steps of 67 MHz in order to check for potential frequencydependent effects. The input power was solely set by the controllable attenuator. Gain losses
of the DUT at higher frequencies were compensated by the attenuator.
Results
Figure 2.29 illustrates an example of the received BB spectrum at 472 MHz and 13 dBm input power. Beside second and third-order IMD in the fundamental zone, also BB distortions
in the third harmonic zone are visible. In addition to that, there is a mirror frequency due
to mixer I/Q imbalance, as well as a DC offset that results from the ADC offset and the LOmixer feed-through (see Section 2.2). Due to the dominating RF distortion, third-order IMD
products in the fundamental zone have been used for extracting the model coefficients. A
gain loss of the RF front-end of 16 dB from the lower to the higher end of the frequency range
has been observed and compensated by a reduced attenuation. A little imbalance between
the upper and higher IMD3 products is visible, that indicate some memory effect.
The corresponding power level diagram at 472 MHz is illustrated in Figure 2.30. The measured fundamental and IMD powers are depicted in straight lines, whereas its extrapolated
curves have dashed lines. The extrapolation is done by simple polynomial fitting with the
assumed slopes of 1 : 1, 1 : 2, and 1 : 3 for the fundamental, IMD2, and IMD3, respectively
55
PSD (dBm/Hz)
20
40
60
TT Response
IMD3
IMD2
TT Input
LO Leakage
Mirror-Frequency
BB Distortions
80
100
120
140
4
1
0
1
2
Figure 2.29: BB spectrum of a two-tone excitation measured at 472 MHz and 13 dBm with
USRP2+WBX [Ett].
(see also Section 2.3.2). The measurement of the IMD power levels is challenging as the RF
front-end is quite linear in general. Thus, clipping distortion at the ADC occur before the
IMD products reach a measurable level. That is, the high distortions at around the P1dB
are mainly induced by clipping. The resulting power level for P1dB, IIP2, and IIP3 in this
case are: 11 dBm, 64 dBm, and 11 dBm, respectively.
The results obtained for the intercept points over the total receiver operating bandwidth
are illustrated in Figure 2.31. Apart from the small notch for IIP2 at around 1.7 GHz, the
non-linear behaviour is approximately frequency-independent and static over the input signal
band. The average values for the intercept points are IIP2 59.7 dBm and IIP3 13.0 dBm.
Assuming an RF receiver gain of G = 10 dB, the real coefficients cn in Equation (2.7) can
be computed by (2.15), (2.16), and (2.17) to the values in Table 2.5.
Table 2.5: Extracted coefficients of the memoryless polynomial.
Parameter
Value
10 dB
IIP2
59.7 dBm
IIP3
13.0 dBm
c1
3.1623
c2
0.0734
c3
106.0839
56
Output Power (dBm)
80
60
40
20
0
20
40
60
80
100
120
140
60
Clipping
Ideal Fund
Real Fund
IMD3
IMD3 Polyfit
IMD2
IMD2 Polyfit
P1dB
IP3
IP2
40
20
0
20
Input Power (dBm)
40
60
80
Figure 2.30: Power level diagram at 472 MHz highlighting the power of the IMD products
including their extrapolation.
80
IIP2
IIP3
70
60
(dBm)
50
40
30
20
10
0
200
400
600
800 1000 1200 1400 1600 1800 2000 2200

RF Frequency (MHz)
Figure 2.31: IIP2 and IIP3 measured with USRP2+WBX [Ett].
2.7.3 Wideband Test Signals

Multi-tone Signal
The previous two-tone signal is unrealistic in terms of its amplitude statistics and
power spectrum, though it provides a rough description of the extent of receiver
non-linearity and eases the extraction of a simple model. Figure 2.32 depicts the
probability density functions (PDFs) of various CW signals. The single sinusoid has an Ushaped amplitude statistic, also known as bath tube curve, as illustrated in Figure 2.32(a).
The peak envelope of the tone and the PAPR are 0.14 V and 3 dB, respectively. Combining
two sinusoidal functions of different frequencies with 100 kHz spacing yields a superposition
57
16
14
Probability
12
10
8
6
4
2
Probability
0
0.2
5
4.5
4
3.5
3
2.5
2
1.5
1
0.5
0
0.15
0.1
0.05
0.05
0.1
0
Amplitude (V)
(a) Single-tone signal (PAPR = 3 dB).
0.15
0.2
0.250.20.15 0.10.05 0 0.05 0.1 0.15 0.2 0.25

Amplitude (V)
(b) Two-tone signal (PAPR = 6 dB).
8
7
Gaussian
Distribution
Probability
6
5
4
3
2
1
0
0.4 0.3 0.2 0.1

0.1
0.2
0.3
0
Amplitude (V)
(c) Multi-tone signal (5 tones, PAPR = 10 dB).
0.4
Figure 2.32: Amplitude statistics of various CW signals.
58

of two bath tube curves, as depicted in Figure 2.32(b). Note that the overall power is kept
constant, by weighting the sum signal with its total power. The peak power adds up by
factor 2 to 0.28 V, the resulting PAPR = 6 dB. Next, a multi-tone signal with 5 sinusoids
is considered in Figure 2.32(c) to come closer to a realistic distribution of complex digitally
modulated signals. Note that the phases of each tone are constant zero in all cases, hence,
resulting in a high PAPR. The peak power and the PAPR are now 0.45 V and 10 dB, respectively. In addition, also the Gaussian normal PDF is plotted in Figure 2.32(c) that is
typical for modern OFDM-like communication signals. According to the central limit theorem in probability theory, the sum of independent and identically distributed amplitude
distributions of sine wine waves will tend to a Gaussian distribution for a large number of
sine waves with random phases [TGT+96]. However, for this low number of tones, both the
high and low voltages occur with higher probability than in the normal PDF. On the other
hand, the signals PAPR is significantly high to provoke severe distortion.
Periodic multi-tone signals are effective in reproducing the statistics of a desired realistic
communication signal, as they produce similar levels of distortions [LHG+10; TGT+96]. In
fact, they can be used to design a specific amplitude statistic by modifying the amplitude
and phase of each tone. Compared to other wideband signals, such as pseudo random noise,
it is limited in bandwidth. Details on the design of arbitrary signal statistics by multi-tone
signals are given in [TGT+96].
Next, an RF measurement with the USRP N210+WBX is discussed [GSH+12c; GSH+12a].
A signal configuration of a strong 5-tone blocker at 5 MHz and a weak 5-tone desired signal
at 2 MHz, having a higher tone spacing of 400 kHz, has been considered. The sampling rate
and the centre frequency of the device were 16.67 MHz and 487 MHz, respectively. Both
signals were generated by two arbitrary waveform generators (AWGs) and combined using a
power splitter, similar to the setup in Figure 2.28. The distorted output BB spectrum as well
as the ideal input signal are illustrated in Figure 2.33. The blocker signal causes severe IMD
distortion in the fundamental and the mirror zone. As the distortions at the third harmonic
fall beyond the sampling range, they appear in the DC zone. Some distortion products are
even hitting the desired multi-tone signal, which can be distinguished by the different tone
spacings adopted for the two multi-tone signals.
WCDMA Signal
In [Dup12], a memory polynomial model has been developed for the USRP with WBX
front-end. Here, the explicit goal was to check the DUT for memory effects by inputting
a wideband test signal with fast varying amplitude (high PAPR). For that purpose, a
59
Distorted Signal
Input Signal
PSD (dBm/Hz)
20
40
Mirror Frequency
Blocker Signal
IMD
Desired Signal
60
80
100
120
8
2
0
2
4
Figure 2.33: Distorted BB spectrum with multi-tone input, measured with USRP
N210+WBX [Ett].
wideband code division multiple access (WCDMA) signal has been used that is adopted in
the Universal Mobile Telecommunications System (UMTS) standard, better known as the
third generation mobile cellular system.
The measurement setup is sketched in Figure 2.34. The WCDMA signal is generated in
MATLAB by modulating random binary data with an offset-quadrature phase-shift keying
and subsequent raised-cosine filtering. In WCDMA, the user data at relatively low rate is
spread over a much wider bandwidth of 5 MHz, by using a sequence of pseudo-random units
called chips (3.84 Mcps).
Next, the generated signal is uploaded to the signal generator R&S SMU200A that has
an integrated AWG for replaying and up-converting the provided I/Q sample data stream.
Signal generator and the non-linear DUT, USRP+WBX, are synchronized by a 10 MHz
Input signal (AWG)
PC
ETH
R&S
SMU200A
WBX
USRP
N210
10 MHz reference
Figure 2.34: WCDMA Measurement setup.
60

reference clock. As the actual receiver input signal cannot be measured by this setup, it
is assumed that all non-linear distortion is generated in the receiver itself. The SFDR of
the signal generator is sufficiently high and supports this assumption. A generator power of
16 dBm has been used to stay within the mildly non-linear regime. The centre frequency
and the sampling rate of the DUT were 120 MHz and 25 MHz, respectively. A complete
WCDMA frame length of 10 ms has been captured (250 000 samples in total). This high
number of samples has been used for reasons of accuracy. After reception, the output data is
aligned according to the input data by correlation. The amplitude distribution (histogram)
is depicted in Figure 2.35. As just the modulation (waveform) is generated without taking
2.5
Probability
Gaussian
Distribution
1.5
1
0.5
0
1
0
0.8 0.6 0.4 0.2
0.2
Amplitude (V)
0.4
0.6
0.8
Figure 2.35: Amplitude distribution of the measured WCDMA signal (PAPR = 5.1 dB).
into account the specific WCDMA frame structure, the output signal is not Gaussian. It has
a moderate PAPR of 5.1 dB in order to avoid CD at the receiver. Figure 2.36 illustrates the
ideal input and measured output WCDMA spectrum. It clearly shows the generated IMD
in the fundamental zone.
Next, a memory polynomial (2.53) is extracted out of this WCDMA measurement.
First, the memory depth M and the polynomial order N need to be estimated [FB11;
WLR+06]. This is done by the method of false nearest neighbours [Dup12]. That is, the
normalised mean square error (NMSE) between the measured output y and the model estimation y
M P is evaluated until a preset criterion is reached. The NMSE is defined as
NMSE|dB = 10 log10
kek22
kyk22
(2.57)
where kek2 = ky y
M P k2 . Figure 2.37 depicts the error obtained for different polynomial
orders and memory depths. With a memory order M = 1, an improvement of approx.
61
20
PSD (dBm/Hz)
30
Ideal Input
Measured Output
40
50
60
IMD
70 (Spectral Regrowth)
80
90
100
10
5
0
5
10
Figure 2.36: BB spectrum of the input and output WCDMA signal at 120 MHz and
16 dBm.
25 dB is achieved. The NMSE decreases only slightly further for M = 2 and onwards. In
order to compare the results with the memoryless polynomial extracted in Section 2.7.2,
N = 3 and M = 1 are chosen. The complex coefficients are computed by the LS method
according to (2.55). The only critical point is the matrix inversion in (2.55), whose accuracy
is expressed in terms of the condition number. Orthogonal memory polynomials have been
proposed in [Dup12] that have lower conditions numbers. However, the NMSE performance
of conventional and orthogonal memory polynomial are almost identical for the obtained
data. For the aforementioned WCDMA data set, the computed coefficients of the memory
polynomial are
0.1031 + j0.0868
0.7569 + j0.4423
an,m = 0.0045 j0.0045 0.0015 j0.0031 .

0.0022 + j0.0028 0.0100 + j0.0046
(2.58)
Note that the second-order coefficients are very small, thereby validating the low significance
of the even-order distortion studied in earlier measurements. In Figure 2.38, both memory
and memoryless polynomial models are visually compared based on the captured BB spectrum. The NMSE of the memoryless and the memory model are 4.5 dB and 26.2 dB,
respectively, indicating a significant improvement of the memory model for wideband excitation, even though the magnitude response looks identical. There is only a significant
small difference outside the triple bandwidth of the signal. However, this is due to the realvalued coefficients of the memoryless polynomial that cannot mimic the phase distortion
of the receiver. A validation with the time domain signal yields that there is a significant
phase mismatch of the waveform predicted by the memoryless polynomial, compared to the
62
15
MP N=3
MP N=5
MP N=7
NMSE (dB)
20
25
30
35
40
45
50
0
8
10
12
Memory Order M
14
16
18
20
Figure 2.37: Error of memory polynomial modelling for different polynomial orders N and
increasing memory order M obtained with a measured WCDMA signal.
20
PSD (dBm/Hz)
0
20
Measured Output
Memoryless Polynomial
Memory Polynomial
40
60
80
100
120
140
10
5
0
5
10
Figure 2.38: Comparison between memory and memoryless polynomial modelling.

memory model which follows almost ideally the measured curve. That is, validating the
model predictions based on the power spectrum may lead to wrong conclusions. Finally, the
memory polynomial output better estimates the distortion caused by wideband input.
This exposes that memory effects are present in the DUT and should not be neglected.
From the system identification point of view, the memory model performs better than the
memoryless polynomial in case of wideband excitation. However, for the purpose of distortion
mitigation, memoryless polynomials are adopted due to the ease of implementation and
sufficient level of accuracy. In Chapter 4, it will be discussed how memory effects can be
considered in distortion mitigation by still employing a static and memoryless polynomial
as a reference non-linearity.
63
2.8 Summary about Receiver Non-linearity

This chapter has shown that odd-order IMD is of major concern in wideband receivers with
limited pre-selection at RF. Sources, effects, and metrics of receiver non-linearity have been
discussed in detail. Crucial scenarios occur, if weak desired and strong unwanted (blocking)
signals are simultaneously present at the RF antenna input. In such cases, strong blocking
signals can enter the front-end amplification and mixing stages and may cause non-linear
distortions. For the purpose of distortion mitigation, it has been demonstrated that nonlinearities at RF and/or BB can be efficiently modelled by static memoryless polynomials.
Thereby, a novel cascaded model is developed that considers the total receiver line-up of a
DCR. Finally, a measurement-based characterisation of a typical SDR under test (USRP)
has been conducted to derive simple polynomial models with and without memory. To this
end, two-tone and wideband WCDMA signals have been used as test signals.
64
This chapter presents various techniques to handle front-end non-linearity and its effects.
Thereby, linearisation techniques for transmitter non-linearity are also briefly discussed, to
justify that they are inadequate for alleviating receiver-induced distortion. Next, suitable
approaches to mitigate receiver non-linearity are classified and evaluated regarding their
efficiency and limitations. Afterwards, state-of-the-art Dirty RF techniques, that denote
system-level techniques to alleviate RF impairments, are presented briefly. In general, this
chapter studies two main techniques to handle receiver non-linearity in detail. These are
preventing non-linear distortions by cancelling the blocker signal and mitigating the distortion products. Although the digital feedforward mitigation is on focus in the thesis and
detailed in Chapter 4, analogue cancellation and spatial filtering of strong blocker signals
are discussed as alternative solutions.
3.1 Overview
3.1.1 Transmitter Non-linearity
Mitigating PA non-linearity at the transmitter side has been tackled for decades. Although
it does not match to the research problem of mitigating receiver non-linearity, it is worth
studying the principles that are used therein. Finally, mitigation at transmitter and receiver
side can be clearly distinguished.
There is a huge number of circuit-level techniques for RF PA linearisation that can be
classified into three main categories [Ken00]
Feedback techniques,
Feedforward techniques, and
Predistortion.
The following paragraphs briefly discuss these techniques from an abstract level. For implementation details, the interested reader is referred to [Ken00].
65
Feedback systems intuitively cancel the error by feeding back a reference signal formed
upon the output, as illustrated in Figure 3.1. The distortion d(t) is mitigated by the fraction
d(t)
Input
e(t)
Output
Non-linear amplifier
Feedback
path
Figure 3.1: Feedback applied around a non-linear PA generating distortion [Ken00].
K/G, the ratio between the voltage division factor of the feedback path K and the gain
G of the original amplifier. However, linearity is achieved at the expense of overall gain
of the linear signal that is reduced to the factor K, constituting an inherent drawback of
feedback systems [Ken00]. Moreover, high G is typically required to allow using high K.
High gain is fairly difficult to achieve at RF frequencies and justifies the wide application
of feedback systems for audio amplifiers, where gain is easily achieved at low frequencies.
Instead, RF amplification is much more demanding, especially due to the higher frequencies,
higher bandwidth, and more tight linearity requirements.
There are several variants of feedback linearisation techniques, such as passive feedback,
envelope (modulation) feedback, polar feedback, and cartesian feedback [Ken00]. Passive
feedback denotes feedback around the transistor circuit of the amplifier and is typically
applied for gain levelling, as it improves only the linear distortion. Thus, it does not improve
any IMD or HD. There are also approaches feeding back only new generated frequency
components (IMD and HD) that in turn does not improve linear distortion. Envelope or
modulation feedback a predistorted version of the BB input before actual up-conversion. The
polar feedback adds a phase correction loop and is able to compensate for amplitude and
phase distortion. The cartesian loop, being the most common feedback technique, is similar
to the polar feedback, but the BB signal is now processed in I and Q. As the cancellation or
error signal e(t) is generated at BB, it requires up- and down-conversion.
Feedforward linearisation, as depicted in Figure 3.2, is less addressed in literature, most
likely due to its high complexity and tight amplitude and phase matching requirements for
66
3.1 Overview
Main amplifier
Input
Output
Correction loop
Error loop
Error amplifier
f
f
Figure 3.2: Configuration of a basic feedforward PA, sketching the signal flow by two-tone
input [Ken00].
all its components. However, it is discussed here as its digital counterpart is implemented
in this thesis (see Chapter 4). The input signal is split into two paths, whereas the top path
constitutes the main amplifier that generates HD and IMD. Next, a part of the non-linear
output is taken by a directional coupler and subtracted from a time-delayed version of the
input. The signal flow is sketched with two-tone signals in Figure 3.2, but only IMD of up to
5th order is illustrated here due to reasons of simplicity. As a result, an error signal remains
that contains only the distortion in the ideal case. Finally, the error signal is amplified by
the error amplifier and fed to the output coupler to cancel the distortion in the main path.
Both main and error amplifier introduce some delay that need to be compensated by delay
elements, denoted by , in the mutual paths. Beside exact amplitude and phase matching
at RF frequency and full signal bandwidth, the error amplifier needs to be distortion-free.
Otherwise, new distortion by the error amplifier is generated at the output. These issues have
a strong impact on the overall mitigation performance. Another drawback of the feedforward
system is its inability to measure its own performance due to the lack of intrinsic feedback.
Predistortion techniques constitute the simplest form of linearisation of an RF PA [Ken00].

Here, the key idea is to introduce a purposeful function prior the PA that is complementary to the PA non-linearity, so called non-linear inverse [FB11]. When cascading both
transfer functions, the output will have little or no distortion and approaches the linear
67
Output
Input
Predistorter
Non-linear
PA
Figure 3.3: Principle of BB predistortion, indicating that expansive and compressive curve
result in a linear characteristic.
characteristic. Variants of predistortion (RF, IF, or BB predistortion) are classified according to the placement of the predistortion block in the transmitter chain. BB predistortion
or digital predistortion (DPD) is the most common approach and is illustrated in Figure 3.3
[FB11; BBC88; BCG+10; ZWC10]. On the contrary to the RF or IF predistortion, it relaxes
the burden on the non-linear inverse block, because the BB signal is manipulated before upconversion. Furthermore, feedback can be adopted to provide updating of this block. In
general, predistortion has the ability to linearise the entire bandwidth, however, finding a
circuit with expansive (third-order inverse) characteristic may be difficult. But, shifting the
model of a non-linear inverse to BB is much more beneficial and modern radio transceivers
anyway employ some form of DSP in their BB processing. Following this approach also enables adaptive predistortion to handle updating and accuracy problems that are in common
for all the linearisation techniques. Successful DPD applications have been reported, e.g. in
[BCG+10; ZWC10]. In [BCG+10], a Hammerstein model with static polynomial coefficients
has been developed, whose coefficients stem from a co-simulation between RF and BB-DSP.
An adaptive DPD approach has been followed in [ZWC10], that employs adaptive filtering
for finding the coefficients.
A main advantage of mitigating transmitter non-linearity is that the actual input and respective wanted output is always perfectly known. This holds for all aforementioned techniques.
Attempts of postdistortion in the receiver fail or perform worse due to the lack of knowledge of the original transmitted signal. Therefore, mitigating distortion in the receiver is
much more challenging, as the original input is to be estimated first. That might be one
reason why the receiver side is much less considered in the state of the art. In [Ken00], the
possibility of postdistortion in the receiver is also briefly discussed, but worse performance
has been reported due to the aforementioned reason. Also, the postdistortion model need to
be trained according to some known signal configurations, such as a vacant channel that is
68
3.1 Overview
filled by IMD of two adjacent channels. Finally, postdistortion at the receiver can only assist
prior predistortion at the transmitter and relax the performance of the mutual linearisation
techniques. Another problem is that only a single non-linearity is taken into account here,
namely the PA non-linearity. Modelling frequency-conversion effects and cascaded non-linear
systems with higher-order distortion terms, such as contained in a typical DCR front-end
line-up, seems to be impossible with these techniques. Beside adaptive BB distortion, most
of the approaches are not taking into account temperature drifts, component ageing, or any
change in the non-linear characteristic. Due to these reasons, an online-adaptive approach is
highly desirable. Finally, most of the approaches have constraints on the signal bandwidth
or modulation the feedback or feedforward path can handle.
3.1.2 Receiver Non-linearity

There is a number of analogue and digital approaches for mitigating the specific receiver
non-linearity that can be classified into three main categories (see also Chapter 1). These
are discussed in the following paragraphs.
Design Optimisation includes all circuit-level techniques to generally improve the linearity
of the front-end. This is achieved, e.g. by utilising highly linear components in the RF frontend chain. Thus, the level of HD and IMD is reduced and the front-end can tolerate higher
input power. But, there are strict limitations due to ever increasing demands for lower
voltages, scaling, etc. [Sch08]. In addition, using highly linear amplifiers can be problematic
for battery-powered terminals, as such amplifiers are typically very energy-inefficient [DF12].
There is always a trade-off between linearity and power consumption. For mobile terminals,
low cost and low power consumption is typically desirable.
Another example for improving the linearity of receiver front-end by circuit design is the use
of 3 dB hybrids and a balanced amplifier, as illustrated in Figure 3.4 [Sha]. Here, the input
power is split into two paths that are 90 out of phase, amplified by two separated identical
amplifiers, and finally combined again. By applying this technique, the intercept point and
the P1dB are bootstrapped by 3 dB. The 3 dB hybrids can be also replaced by conventional
power splitters. The advantage of 90 hybrids is that reflections by the amplifiers due to
impedance mismatch are passed on to the matched input/output of the hybrids. Thus,
perfect matching is achieved outside the balanced amplifier, that can be beneficial for the
following receiver components. However, 90 hybrids are typically frequency-selective and
not suitable for all applications.
69
In
90
90
Out
Figure 3.4: Principle of a balanced amplifier according to [Sha].

Preventing Non-linear Distortions can be also achieved by cancelling or modifying the
actual signals that caused distortions. The most intuitive approach is to filter these signals,
either by analogue filters, filter banks [Mar09], or tunable bandpass filters. The latter one,
the tunable preselection bandpass between antenna and LNA, is very difficult to implement
and not available as an off-the-shelf component [ZMS09]. Although placing the filter before
the LNA will result in a higher noise figure (NF), strong blocker signals need to be suppressed
before they are fed to any non-linear component in the RF chain. Otherwise, distortions
are generated that may already fall into the desired band and that cannot be simply filtered
out afterwards. Also, spatial filtering by an antenna array or analogue feedback cancellation
with a replica is thinkable [WSW+10]. However, the most common practice is to attenuate
the input so the operating point is always within a certain back-off to the saturation point
(P1dB) and to be far away from the heavily non-linear region. Hence, the signals experience
the linear part of the transfer function of the components [Sch08]. A significant drawback
of this simple approach is that the sensitivity for weak signals is simultaneously reduced.
An interesting idea to solve that problem is proposed in [OAKS+10], by cross-correlating
the output of two duplicated front-ends. Adopting this approach and assuming uncorrelated noise in both sources can enable the detection of such weak signals while adopting a
higher back-off. Finally, variation of the signal envelope can be reduced (PAPR reduction
techniques) or by using constant-envelope modulation. For instance, constant-envelope is
adopted in Global System for Mobile Communications (GSM) to be resistant to any nonlinear distortion in the BTS or mobile station (MS) as far as possible.
Mitigating Non-linear Distortions can be realised by pre- or postcorrecting techniques,

although postcorrection by means of DSP is on focus here. Examples of such techniques are
70
3.1 Overview
just mentioned here selected approaches are discussed in the next section in more detail.
The received signal can be equalized at symbol level with, e.g., embedded training or pilot
symbols [DS07; Chi11; Sch08], but also blind estimation may work [DKF10; DF11; DF12].
The performance may be also improved by an iterative detection at symbol level using the
Turbo equalisation principle [DSA09]. Also, digital postdistortion with a non-linear inverse,
i.e. a DPD counterpart, can mitigate distortion, but seems to be less reasonable with an
inaccurate model. The approach followed in this thesis depicts a feedforward architecture
and regenerates distortion products with the help of a reference model and subsequently subtracts them from the received signal [VSHGA+06; KH08b; AMV10b; GSH+12b; GSH+12a;
GAM+14].
Efficiency and Limitations Not all of the aforementioned techniques are reasonable and
really effective in practice. First of all, any modification of the front-end circuit is typically
not desirable due to reasons of cost, complexity, and flexibility. Indeed, it is also not in line
with the SDR concept. Although analogue approaches have a great potential for avoiding
blocking conditions, circuit non-linearity is more efficiently improved by system-level techniques [Chi11]. In addition, the technological development in the digital domain still rises
according to Moores law, hence, more and more signal processing tasks are shifted into
the digital domain. However, analogue circuitry is still much faster than digital circuitry
[KH08a]. Moreover, implementation of mitigation techniques by digital circuitry has the
advantage of insensitivity to process variations [KH08a].
Simple filtering of out-of-band blocking signals may work as long as the distortion products
are not in the band of interest. In that case, filtering should be employed directly at the
RF input prior any non-linear component. This becomes difficult for multi-carrier or multichannel receivers, where HD and IMD can hit the target band. If the distortion products
have fallen onto the band of interest, more sophisticated signal processing becomes necessary
rather than ordinary linear filtering [VSHGA+06]. Furthermore, tunable selective analogue
filters are difficult to implement.
The proposed digital feedforward technique in its purely digital implementation also has some
fundamental limits (see also Section 4.5). It requires that all distortion-producing carriers
are available in the digital domain. Accordingly, the ADC needs to pass the full spectrum of
potential problematic blockers within a frequency range much larger than the original signal
bandwidth [KH08b; KH09b]. Hence, multiple RF chains or wideband digitisers are required.
Although most of the approaches are blind mitigation techniques that do not require a
training sequence or detailed knowledge of the interfering signal, they will fail under overload
71
conditions. Moreover, the distortion regeneration is affected by a non-ideal reference signal
that is subjected to in-band distortion. Nevertheless, these techniques are most attractive as
no exact knowledge of the non-linear distortion profile, the input waveforms, or the desired
signal is needed.
Finally, only mixed-signal solutions will complement each other and provide the best mitigation performance. For instance, the purely digital feedforward correction can be assisted
by some relaxed filtering at the input in order to avoid overload conditions.
3.2 Dirty RF Techniques State of the Art

The general idea of Dirty RF, a term first coined in [FLP+05], is to tolerate a certain
amount of RF imperfections and to mitigate them mainly by DSP at another part of the
signal chain [Sch08]. Improving the system linearity by system-level DSP is a popular alternative to analogue design optimisation with the potential for power-saving and improved
scalability [Chi11]. In the following paragraphs, some selected approaches from the state of
the art for mitigating receiver non-linearity are briefly presented and evaluated according to
digital feedforward technique, that is chosen in this thesis (see Section 3.3).
Blind Estimation of transmitted symbols over non-linear channels based on

maximum likelihood (ML) method has been proposed in [DKF10; DF11; DF12]. Above
those, an ML estimator can produce all possible transmit symbols, compute the distance
to the actual received symbols, and finally choose the symbol with the minimal distance.
The idea followed in aforementioned references is to estimate the parameters of the receiver
amplifier model by ML estimation. The underlying model reproduces only AM/AM distortion and is found using the PDF after non-linearity. It is finally used in a decision feedback
detector depicted in Figure 3.5, that remodulates the detected signals s, feed that through
the estimated non-linearity model, isolates the distortion d and subtracts it from the input
y. The architecture is based on Bussgangs theorem [Sch08],
x =
s + d,
(3.1)
which states that non-linearity causes a scaling of the wanted signal and introduces an
additive error term that is typically treated like additional noise. Different variants for
calculating the convolution of the PDFs are discussed with regard to its computational
complexity, aiming for a real-time implementation in LabView based on USRP [DF12].
72
Detector
Modulator
x =
s + d
Figure 3.5: Decision feedback detector with estimated receiver non-linearity [DF11].
A good mitigation performance in terms of SNR and bit error rate (BER) is reported
therein, especially in case the real-world behaviour matches to the amplifier model considered in the detector. In real RF measurements, approx. 36 dB improvement in the
signal-to-interference ratio including noise and distortion (SINAD) and up to two orders of
magnitude in terms of the BER have been obtained. It is likely that the low SINAD improvement is due to the mismatch between the underlying amplifier model and the physical RF
front-end architecture. This issue is exactly one of the gaps that is treated in this thesis.
Two-channel receiver is proposed in [ZMS09] to jointly estimate the channel response and
non-linearity parameters by applying a-priori knowledge of the desired and blocker signal.
Figure 3.6 illustrates the proposed architecture, where the RF is split after the LNA into
two receiver chains. Alternatively, the signal may be split already at the antenna, to get
amplified by two individual LNAs. Thereby, cross-modulation distortion (see Section 2.3.1)
of a strong blocker on a weak desired carrier is explicitly addressed in context of SDR.
The custom receiver with two RF chains is employed for separated reception and downconversion of the blocker, however, only one blocker at a time can be considered. Finally,
the distortion term can be subtracted via LS and minimum mean square error estimation
with the knowledge of the blocker amplitude that can be tracked by a low speed and low
resolution auxiliary ADC in the second RF chain. Although this approach relaxes the burden
on a single wideband ADC to capture both weak and strong signals, it requires a separate RF
path for each blocker and increases the front-end complexity. The BER of an OFDM system
can be reduced by up to one order of magnitude, as shown by simulations in [ZMS09].
73
LO1
Desired Signal Path
ANT
ADC
BB
Processing
ADC
LNA
LO2
Blocker Signal Path
Figure 3.6: Two-channel wideband SDR receiver for acquiring wanted and blocker signal
[ZMS09].
Mixed-Signal Feedforward Mitigation by using a custom-designed RF front-end is proposed in [KH08b] and related references [KH08a; KH09b; KH09a]. The proposed architecture
is illustrated in Figure 3.7. Here, the main idea is to regenerate the third-order IMD products
Main Path
ADC
ANT
LNA
(.)3
ADC
Adaptive
Filter
Alternate Path
Figure 3.7: Mixed-signal feedforward mitigation with RF cubic-term generator highlighted
[KH08b].
at the analogue RF domain in an alternate path and to pass only problematic IMD products
down to BB in the same manner as the desired signal (using same LO). The processing and
adaptive filtering takes place at digital BB. Thus, digital BB and ADC requirements are not
greater than that of the main path. It generally solves the main limitations of purely digital
feedforward mitigation followed in this thesis, but also requires a lot of additional circuitry to
be implemented on chip. The main advantage of this approach is that all distortion products,
74

appearing at the full receiver RF bandwidth, are represented by the cubic-term generator in
the alternate path [KH09a], and only those falling into the desired BB bandwidth are further
processed in the digital domain. Following this approach, only RF distortion is considered.
Further BB distortion by mixer or BB amplifiers are not considered herein, although it could
be mitigated by adaptive filtering in the digital domain as well. It is also noteworthy that
RF distortions are implicitly modelled by physically down-converting these distortion products to BB. In [KH08b], a custom RF front-end for UMTS application has been designed to
meet the linearity specifications imposed by an out-of-band blocker test. The effective IIP3
has been improved by 12 dB, and an SNR gain of 20 dB and BERs of down to 104 for the
UMTS scenario have been reported. These are very impressive results that are achieved at
the expense of circuit complexity.
ADC Distortions are specifically addressed in [AMV10a; AMV10b; ALM+12; Chi11].

Here, the main goal is to mitigate both mild non-linear distortion and also clipping distortion,
i.e. to remove any kind of distortion regardless of its source. Beside aforementioned distortion
phenomena, ADC-specific non-idealities are DNL, INL, and clipping (see Section 2.2). In
[AMV10a], an alternate path with an additional ADC is proposed to obtain a non-clipped
reference signal. That auxiliary ADC can be a low cost and low resolution type as in the
two-channel receiver approach proposed in [ZMS09]. Finally, clipping distortion is mitigated
by recovering the clipped waveforms using interpolation. In [AMV10b], a mathematical
model of zero-symmetric clipping based on Fourier analysis is developed for understanding
and modelling the impact of the ADC and essential IMD distortion on top of weak signals.
This model is used for post-correcting the ADC non-linearity by using only one RF path.
Clipping-induced IMD is regenerated and subtracted from the received signal using adaptive
filtering.
Digital Feedforward Mitigation has been first proposed in [VSHGA+06; gha11] and
is detailed in Chapter 4. This algorithm, illustrated in Figure 4.1, is working purely in
the digital domain. The main idea is to regenerate distortion products by specific strong
distortion-producing signals and to subtract them from the desired signal observation in an
adaptive manner. Since the error signal and reference model are never precisely known in
practice, an adaptive filter is used to subtract the reference signal from the main receiver
path. There is a gap in the state of the art with regard to the mitigation architecture. The
specifics in modelling of a complete RF chain of a DCR have not been considered so far. Up
to now, the non-linearity of the total RF front-end is modelled by a single polynomial. This
approach does not distinguish different types of RF and BB distortion and cannot capture
75
all distortion products generated by such front-ends. Therefore, the model is extended in a
way that it matches to the physical front-end architecture.
3.3 Reasons for Selecting Digital Feedforward

Mitigation
There are several advantages why the digital feedforward mitigation has been chosen for
this work, among other analogue or digital techniques. First, the algorithm is suitable for
universal application and can be simply integrated, especially due to the fully DSP-based
approach. Thereby, a very flexible implementation of the algorithm on an FPGA or DSP
can be achieved without employing any additional hardware. It is efficiently correcting the
I/Q signals in a way that the amount of non-linear distortion is reduced. As it is solely
processing the distortion-producing carriers, it is independent of the desired signal and its
specific modulation.
Second, non-constant characteristics, such as production tolerances in the front-ends components, thermal influences, or device ageing can be easily taken into account by the adaptive
filtering stage. This is a clear advantage over all analogue linearisation techniques discussed
in Section 3.1.1 in context of transmitter non-linearity mitigation.
Last, it is noteworthy that no detailed knowledge of the non-linearity profile and no exact
polynomial coefficients are necessary [VSHGA+06], only basic knowledge about the physical
receiver front-end architecture is required.
Due to the aforementioned advantages, the approach has been followed in this thesis for
alleviating non-linear distortions in the receiver. Beside general limits and requirements of
this technique, it perfectly fits to the presented research problem and represents an effective
solution for mitigating receiver non-linearity at system level.
3.4 Cancellation of Distortion-Producing Signals

In the following, blocker or interference cancellation techniques are discussed as an alternative to alleviate non-linear distortion at the receiver. Although leaving the scope of Dirty
RF digital post-correction, such techniques constitute a promising solution to prevent the
generation of non-linear distortion at all.
76
3.4.1 Analogue Cancellation

An active feedback interference cancellation technique has been proposed in [WSW+10]
that bases on an analogue cancellation filter core, as illustrated in Figure 3.8. Here, the
LNA
:2
2LO
Active Cancellation Filter Core

Figure 3.8: Block diagram of the active feedback interference cancellation proposed in
[WSW+10].
received signal including the wanted and blocker signal are down-converted by the receiver
LO signal. Then the wanted signal is eliminated by highpass filtering, thus, the remaining
signal constitutes a blocker replica. Finally, the blocker replica is again up-converted to
RF and subtracted from the received RF signal, resulting in a partial cancellation of the
blocker.
This intuitive approach has some major drawbacks. First, the highpass corner frequencies are
determined by the bandwidth of the actual wanted signal at DC, assuming a zero-IF downconversion. The remainder of the signal after highpass filtering is considered as interference.
It may comprise multiple strong blockers, but also other useful signals, that are finally fed to
the subtraction node. That is, the filter core is fix in its cut-off frequency and not selective
to specific blocker signals. Second, as the blocker subtraction is conducted after the nonlinear LNA, distortion can be already on top of the wanted carrier and cannot be simply
filtered out by this approach. Ideally, the subtraction should be performed prior the LNA,
even though the overall NF is affected. Filter core noise may be added in the main path
also in the proposed configuration, especially due to the LO phase noise. Third, there are
also some implementation challenges reported in [WSW+10]. For instance, the filter core
IF bandwidth determines the loop stability. The filter implementation is also challenging
when the blocker comes close to the wanted signal. In addition, I/Q imbalance in the up-
77
ANT
f
ADC
LNA
BB
Processing
+
Replica
Generator
IF-A
LO
DAC
PA
LO
Figure 3.9: Block diagram of the analogue blocker interference cancellation.

and down-converter paths may cause blocker images in the main LNA path. Although the
problem of amplifier non-linearity is not mentioned in [WSW+10] and references therein,
the proposed cancellation technique is an interesting solution to avoid non-linear distortion
at the receiver by simply cancelling crucial signals. However, the solution is not very flexible
and rather unsuitable for cleaning the whole BB signal from distortions, as it is needed for
application in CR. It can rather provide a performance improvement for a single wanted
signal of interest.
A similar approach has been followed at the beginning of this work, but has not been
pursued further due to major drawbacks like those mentioned in the aforementioned article.
The proposed architecture is depicted in Figure 3.9. The key idea is to cancel the blocker
signal at RF before entering the LNA or any other non-linear component in the receiver
front-end. Thereby, non-linear distortion provoked by a specific blocker signal is prevented.
As the blocker replica is generated at digital BB, it is much more flexible than the previous
approach. First, the blocker needs to be located by using a coarse sensing algorithm such
as energy detection (see Section 7.1). Next, a replica of the blocker is generated using the
complete transmitter chain of the SDR architecture. Finally, the blocker is subtracted from
the input at RF level.
This approach has several advantages when assuming ideal operating conditions. It is likely
that the blocker can be successfully cancelled if the replica has a perfect match in amplitude
78

and phase. It requires no demodulation of the blocker, although a remodulation based on
the detected symbols might be a possible solution. A major advantage is that no model of
the non-linear receiver is required at all, hence, the approach is fully independent from the
specific RF front-end design. Even if the blocker is only partially cancelled due to imperfect
match of the replica, the linearity requirements of the front-end are reduced significantly.
Except than the subtraction node, that can be a differential LNA in practice, no hardware
modification is necessary, as the transmitter stage is typically available in an SDR.
However, there are some practical challenges and drawbacks that make this kind of blocker
cancellation not feasible in practice. There is an unavoidable delay of the replica generation
and the feedback loop as the complete transmitter is included in the feedback loop. That
is, it will only work for deterministic or at least slow-changing blocker signals that can be
predicted. In theory, the RF signal should be delayed to provide sufficient time for generating
the replica. In order to establish an exact amplitude and phase match, a closed loop is
required. Also, in case of blocker modulation and channel fading, amplitude and phase of
the blocker will change over time. Beside these conceptual challenges, there are also some
implementation issues that need to be looked at closely. For instance, significant noise may
be introduced prior the LNA due to the feedback path. Also, the PA at the transmitter may
add new undesired components. However, the blocker replica needs to be only amplified to
relatively small power levels at the order of approx. 80 dBm. Thus, leakage effects through
the receiver antenna should be significantly small.
Proof-of-Principle Experiment
Nevertheless, some real RF measurements have been carried out in order to pursue the idea
and to discover the challenges of this technique in more detail. The setup is depicted in
Figure 3.10. For reasons of simplicity, only one blocker has been assumed that is static
over time. The actual blocker and its replica are generated by two individual USRP SDR
platforms, equipped with WBX daughterboard [Ett]. Random static phase offsets at USRP
generating the blocker replica have been adapted by a delay line. These offsets are caused
due to the FPGA-based processing on the USRP. A desired signal is generated by an AWG
and is coupled into the main receiver path. In this setup, the blockers amplitude and phase
are assumed to be perfectly known and both transmitting USRPs are fully synchronised by a
10 MHz reference clock and a pulse per second (PPS) signal. Finally, desired signal, blocker,
and replica are fed to a third USRP that acts as the actual receiver. In the final application,
the receiving USRP should be able to create the blocker replica by itself.
Figure 3.11 illustrates the measured BB spectrum before and after cancellation. Desired
79
Power
Combiner
Blocker Signal
RX
WBX+
USRP2
TX
WBX+
USRP2
PPS, REF_CLK
RX
RX
TX
Desired Signal
REF
Blocker Replica
Receiver
f
WBX+
USRP2
REF
AWG7102
f
TX
REF
Delay Line
Figure 3.10: Experimental setup for the analogue blocker cancellation.

and blocker signal are CW 5-tone sinusoidal signals with different tone spacing and carrier
frequencies. Up to 40 dB blocker suppression have been achieved, depending on the resolution
of the delay line. It is also visible that the IMD distortion caused by the 5-tone blocker almost
vanished. As the LO leakage is different in the blocker and replica transmitter, it could not
be compensated. To sum up, the general concept of blocker cancellation has been proven to
be functional, however, many of the aforementioned drawbacks make it not practical. For
these reasons, digital feedforward mitigation has been improved further instead of pursuing
the analogue blocker cancellation.
3.4.2 Spatial Filtering using Antenna Arrays

Another interesting approach to increase the dynamic range and to avoid blocking conditions
is to adopt the filtering of strong blockers in the spatial domain by an antenna array instead of
the frequency domain [CB05]. A SDR with a large number of receiver antennas is illustrated
in Figure 3.12 [Sch08]. Basically, desired and blocker signals may impinge from different
directions. Therefore, the main idea is to achieve a selective reception and suppression of
specific impinging waves with antenna arrays and beam-forming techniques, that combine
different antenna signals before digitisation. Here, closely placed antenna elements are fed in
a way that a desired radiation pattern is generated. This is accomplished by controlling the
power levels and phases of each path, indicated by antenna array coefficients ai in Figure 3.12.
These coefficients are computed in the digital domain. Finally, nulls in the antenna diagram
80
PSD (dBm/Hz)
3.5 Summary of Possible Mitigation Techniques
20
30
40
50
60
70
80
90
100
110
120
130
6
Uncompensated
Compensated
Blocker Signal
Desired Signal
LO Leakage
2
0
2
Figure 3.11: BB spectrum of the received signal before and after blocker cancellation.
in the direction of strong blockers can be generated after prior measurement of their angle
of arrival by sweeping the coefficients. Simultaneously, weak signals can be amplified by
imposing some higher gain in their angle of arrival.
Beside beamforming techniques, also a simple opportunistic choice of the best antenna is
thinkable. That is, all possible antenna diagrams are scanned and the optimal diagram with
the most suitable signal configuration of desired and blocker signal is just switched. In any
case, spatial filtering will support other digital mitigation algorithms by avoiding overload
conditions and relaxing the burden on the A/D interface.
Such interferences scenarios can be emulated using the setup for over-the-air testing in a virtual electromagnetic environment (OTAinVEE) build at Ilmenau University of Technology
[KGM+11; MKG+11; GKM+11a]. This testbed allows for emulating a realistic radio environment with reproducible signal scenarios for a single centred radio node, surrounded by
an antenna array in a circular arrangement. For details on this test approach, the interested
reader is referred to the aforementioned references.
Though, spatial filtering is likely to be effective, the approach is out of the scope of this
thesis and has not been pursued further.
3.5 Summary of Possible Mitigation Techniques

This chapter has presented a broad range of analogue and digital techniques to alleviate
circuit non-linearity. Receiver non-linearity can be handled by design optimisation, cancelling distortion-producing signals, or by pre- and post-correction techniques. The latter
81
a1
a2
LNA
IF-A
ADC
+
.
.
.
BB
Processing
+
Adaptive
Nulling
LO
aM
Figure 3.12: Wideband SDR front-end with antenna array for spatial filtering [CB05].
one includes Dirty RF signal processing, that represents a trendy system-level technique
to alleviate non-linear distortions. Beside receiver non-linearity, techniques for mitigating
transmitter non-linearity have been also discussed, as some principles are in common. Finally, a digital feedforward mitigation algorithm has been selected, especially due to its
simple integration and efficient post-correction of distorted BB signals in the receiver.
82

This chapter discusses the selection mitigation technique in detail. First, the main architecture and its individual components are explained. The actual mitigation architectures
are developed based on the appropriate BB models discussed in Chapter 2. Then, the effectiveness of the algorithm is demonstrated by extensive simulations and RF measurements.
Finally, fundamental limits are discussed that need to be considered for employing this
technique into real-world devices.
4.1 Principle
The key idea of the feedforward mitigation algorithm is taken from [VSHGA+06]. Figure 4.1
illustrates the algorithm in its general form. The signal flow is hereby illustrated by sketched
f
RF
xRF (t)
A/D
Digital Domain
d(n)
yBB (t)
Non-linear
Front-end
Band
Splitting
Delay
x
(n)
Reference
Model
Adaptive
Filter
e(n)
BB
Processing
Figure 4.1: Block scheme of the generalized digital feedforward mitigation algorithm.
BB equivalent spectra at different points along the signal processing chain.
83

The RF antenna signal xRF (t) passes the non-linear front-end where the signal gets amplified,
down-converted, and digitised. This analogue processing introduces noise and distortions in
the signal, resulting in the distorted version yBB

(t). In most of the applications, the input
signal consists of a weak desired signal (green-coloured signal in the sketches) and a close
strong blocker (black-coloured). At BB, additional distortions around the blocker (orangecoloured area) show up and can fall on top of desired signals or onto free frequency ranges.
Now, the main idea is to mitigate these distortions by introducing some signal post-processing
in the digital domain that is placed in between the A/D-conversion and the system-specific
BB processing for the desired signal or spectrum sensing. That is, the algorithm is working
at waveform level with the I/Q signal. It consists of the following components:
Band splitting including interference detection,
Reference non-linearity model, and
Adaptive filter.
The key idea is to extract crucial interferers contained in the BB and to reproduce their
distortion products. First, a filter pair with inverse characteristics, e.g. bandpass/bandstop
or lowpass/highpass configurations, splits the distorted signal yBB

(t) into a desired signal
d(n) and the reference signal x(n). The upper path signal, d(n), consists of the signal of
interest and distortions products caused by the blocker, while the reference path signal x(n)
constitutes an estimate of the originally transmitted blocker. Second, the reference signal in
the lower branch is fed into a reference model of the non-linear front-end reproducing RF
and BB non-linearity as well as I/Q imbalance, to regenerate the distortion (HD, IMD) that
was generated during non-linear reception. The reference model output at this stage does
not match the amplitude and phase of the distortions in the upper branch (illustrated by
lower amplitude of the orange-coloured area) due to model inaccuracy or incorrect estimate
of the interferer signal. Therefore, the reference distortions are further processed by an AF
to adjust them to exact values. Thus, by subtracting the AF output from d(n), the effect
of non-linearly induced interference is diminished. The cleansed signal is the common error
signal e(n) of the AF.
In general, two different use cases need to be distinguished. These are referred to as
1. Single signal of interest (SOI), and
2. Sensing (or multi-carrier BTS).
Examples for these use cases are given in Chapter 7. In the first case, a certain narrow band
is to be cleaned from distortions that have been generated by strong neighbouring blockers.
84
4.1 Principle
This small band typically holds a weak carrier and does not occupy the whole BB bandwidth.
Then, in the upper branch, this single SOI is isolated using a bandpass. In contrast, in the
lower branch, the desired signal is filtered out and the remainder of the BB signal is used for
modelling the distortion falling into the desired band. In the last sketched spectrum, only
the desired signal remains (upper right spectrum in Figure 4.1).
On the other hand, the whole BB bandwidth is to be cleaned from distortions in the second
use case, e.g. to enhance the reliability of spectrum sensing techniques or for wideband BTS
receivers that simultaneously demodulate all down-converted carriers. A bandstop filter in
the upper branch filters out the strong blocker to provide a good desired signal for the
AF. Conversely, a bandpass filter in the lower branch isolates the strong blocker that is
used for distortion modelling. For applications like spectrum sensing, it is desirable to have
the strong blockers present after mitigation, because they can be also interpreted as useful
signals. These scenarios aim for an undistorted BB representation of xRF (t) at the antenna
input. In such a case, the blocker estimate x(n) needs to be added back after mitigation to
the error signal e(n), as it is indicated in Figure 4.1 by a dashed arrow. Finally, a cleaned
version of yBB
(t) is obtained, i.e. without the orange-coloured distortion components.
XMD between strong blockers and weak desired signals, as discussed in Section 2.3.1, are not
taken into account by this architecture. However, this specific kind of distortion is typically
weak compared to the IMD caused by dominating blockers [VSHGA+06]. Furthermore, it
is sufficient to isolate only the strongest blockers as they cause the most harmful distortion.
The amount of generated distortion always depends on the peak signal power present at
the input. By selecting specific distortion-producing signals, the distortion estimates will be
also more accurate, as no additional distortions are generated that are actually not present
in the received signal. Feeding a large band of signals through the reference path likely
causes unwanted distortions, depending on the actual power levels and frequency separations
between the carriers.
The mitigation processing in the lower branch, especially the filtering, introduces some delay.
To compensate for that effect, the same delay can be easily added in the upper branch. This
is one of the major advantages of performing this processing digitally compared to the main
drawback of the analogue blocker cancellation techniques discussed in Section 3.4.
For the simulation testbed, the three main components have been implemented in
MATLAB using an object-oriented design. This simulation framework is referred to as
non-linearly induced interference mitigation (NONLIM) algorithm and is detailed in Section C.1. In the subsequent sections, all components are discussed from the system-theoretic
point of view in detail.
85
4.2 Components
4.2.1 Bandsplit Filtering
The derivation of the desired and reference signal plays an important role for the effectiveness
of the adaptive mitigation processing, as their stopband attenuation dictate the spectral
purity of the input for the AF. That is, the filters should have a high slope to clearly separate
the blocker from the IMD and the remainder of the BB signal. The exact filter characteristics
particularly depend on the RF signal configuration and the underlying receiver front-end at
hand. If the scenario is known in advance, fixed filter settings can be used. Otherwise, the
filters need to be tuned accordingly by changing their coefficients during runtime. The first
case is valid for applications with one SOI, while the latter one should be considered for
highly dynamic scenarios like for spectrum sensing applications in CR.
For the spectrum sensing application, strong blockers near to desired signals or bands for
secondary transmissions can be identified in the first step by a coarse energy detector (Section 7.1). A threshold of 20 dB below the maximum input signal level has been found through
experiments and is assumed to be reasonable to detect all strong blockers. Thereby, the filter pair may consider a single blocker at a time or even multiple blockers by a multi-band
filter.
By filtering the blocker out of the input signal, an estimate x(n) of the originally transmitted
signal x(t) causing the distortions is obtained. However, the strong signal (or blocker) itself
suffers from non-linear distortion, so called in-band distortion. Therefore, the reference
distortions provided by the following reference model will differ from those in the upper
branch. This problem is discussed in detail in section 4.5.
The computational complexity of the filtering should not be underestimated. The filter pair
should have linear phase (or equivalently constant group delay), high stopband attenuation,
and unity amplitude response in the passband in order to not distort the signal further. These
characteristics are required for phase-sensitive communications and are fulfilled by FIRs with
symmetric coefficient sequences. Because FIR filters do not require any feedback, they are
inherently stable compared to their counterparts, the infinite impulse response filters.
Following Figure 4.1, a pair of a complex bandpass and bandstop filter is typically required,
especially due to the mirror-frequency interference occurring in any practical receiver. The
classical way of designing such filters is to frequency-shift a real-valued lowpass and highpass
FIR prototype, respectively. The following section will give a brief overview on FIRs. More
details can be found in literature, e.g. in [MB04; Hay02; Spt].
86
4.2 Components
FIR Filters are the most common LTI filters that act on the input through linear convolution [MB04]. Hence, the output is given by
H
y(n) = w (n)u(n) =
M
X
k=0
wk u(n k),
(4.1)
where M is referred to as filter order with the filter coefficients wk and corresponding input
u(n). The FIR filter architecture is depicted in Figure 4.2 in a transposed transversal form
[MB04]. This structure is also known as tapped delay line and consists only of the three
u(n)
w1
w0
z 1
z 1
w2
wM
y(n)
Figure 4.2: FIR filter in transposed structure [MB04].

elements delays, adders, and multipliers. The task of the multiplier is to multiply each tap
input u(n) by a filter coefficient wk that is also referred to as tap weight [Hay02]. The
transposed form is preferred here as no shift register is necessary for holding the input
samples [MB04].
A main advantage of FIR filters is their inherent stability as they do not employ any feedback.
Hence, they only show zeros in the pole-zero plot. The number of the delays represents the
finite duration of its impulse response. Linear-phase FIRs have a constant group delay
() =
d
= const.
d
(4.2)
over a range of frequencies, if the filter is even- or odd-symmetric. Symmetric FIRs also
require only half the number of multipliers, constituting an important advantage for efficient
FPGA implementation.
FIRs are typically designed using direct computer specifications and algorithms, such as the
MATLAB Signal Processing Toolbox [Spt]. For this purpose, different algorithms are available, such as equiripple, Kaiser window design, or LS FIR method. Throughout the thesis,
the equiripple method following the Parks-McClellan optimal FIR filter design algorithm
87

[Spt] based on the Chebyshev approximation theory is adopted. This and other methods
require the desired filter characteristics as input, i.e. the transfer function in the frequency
domain represented by passband, transition band, and stopband characteristics. Then, a
mathematical algorithm optimises the filter coefficients according to the desired specifications. There is also the possibility to design and perform the filtering in the frequency
domain, by using the FFT [MRS91]. However, this approach just provides less computational complexity for high filter orders M or is preferred if subsequent signal processing is
anyhow conducted in the frequency domain.
4.2.2 Reference Non-linearity Models

The mitigation algorithm requires a signal model of the front-ends imperfections to regenerate the distortions caused by strong signals (blockers) in the BB. The model output basically
tells where distortions are located in the BB by distorting the input in the (ideally) same way
like the actual physical front-end. The reference model has to reproduce the behaviour of
the original front-end as accurate as possible (behavioural modelling). That is, the structure
of the reference model should match the physical origins of the distortion in the front-end.
Depending on the receiver architecture, there are different types of models or subsets of
the full cascaded model can be used. Three types of reference models are distinguished,
single RF non-linearity, single BB non-linearity, and combined/cascaded non-linearity. The
total cascaded model is adopted first to derive a complete mitigation architecture. Then, it
can be decided from the use case and the underlying receiver which parts of the mitigation
architecture are finally employed for the algorithm. This novel kind of modelling has been
published in [GAM+14].
Total Cascaded Non-linearity Model

The theoretical architecture for joint mitigation of RF and BB non-linearity as well as mixer
and BB I/Q imbalance is depicted in Figure 4.3. The complex bandpass and bandstop filters
C
are denoted by hC
BP and hBS , respectively. New symbolic blocks have been used for the digital
bandpass and bandstop filters to differentiate them from their analogue counterparts used in
the receiver block diagrams. The most intuitive way to design the reference model is to simply
cascade RF, mixer, and BB model that have been developed in Section 2.5. Hence, the total
model decomposes into several blocks. However, this structure will only work if the exact
coefficients are known a-priori. In fact, an online adaptation of all the model coefficients is
practically required, hence, AFs are included and depicted by gray-coloured arrows and the
88

yBB
(n)
hC
BP
hC
BS
3a2
a
1
+
RF Non-linearity Model
(.)2 (.)
x(n)
d(n)
k2
Mixer I/Q Imbalance Model
(.)
k1
Im
Re
89
a
4,Q
a
3,Q
a
4,I
BB Non-linearity Model
(.)3
(.)3
a
3,I
e(n)
4.2 Components
Figure 4.3: Theoretical architecture for mitigating cascaded non-linearity [GAM+14].

tilde coefficients in Figure 4.3, indicating estimates of the exact coefficients. The challenging
issue is that RF and BB distortion generate same spectral content, especially the IMD in the
fundamental zone, and thereby cause severe difficulties for the convergence behaviour of the
individual AFs. By using the sequential structure, the AF adapting the coefficients of the RF
non-linearity model will try to eliminate the IMD in the fundamental zone at once. If this
distortion is already vanished, adapting the BB non-linearity coefficients becomes difficult,
as some of the BB distortion components are not any more in the signal. The same happens
if the BB distortion is removed before the RF distortion. In other words, there is a certain
time-domain correlation between the distortion components that originate from the same
blocker signal. Difficulties of the adaptation process with correlated input has also been
discussed in [Hay02]. These convergence problems are relaxed if the AF could be assisted by
telling the basic relations between the distortion terms. For instance, if the RF non-linearity
coefficient a3 is exactly known a-priori, the BB distortion term can be easily adapted. Since
a3 is raised up to the third power while passing the BB non-linearity, the accuracy of a3
must be high to avoid severe deviation. In order to solve these problems in practice, the
coefficients should be updated simultaneously and not sequentially [GAM+14].
Next, the terms produced by the cascaded model developed in Section 2.5.5 and Section 2.5.6
are analysed and then a simplified parallel mitigation architecture is derived that enables
simultaneous adaptation of the distortion components. The I/Q imbalance is neglected
for extracting the distortion terms, but is considered again in the final architecture. The
distortion terms produced by the total cascaded model can be extracted from the complex
representation (2.45) and are summarised in Table 4.1. The terms are listed with regards
Table 4.1: Distortion terms produced by the cascaded non-linearity model without I/Q
imbalance.
No. Coefficient
2
a3 a1 + 27
a a 2 a |z|2
2 4 2 1
3a3 a2 + 43 a4 a1 3 + 18
aaa2
4 4 2 1
1
aa3
4 4 1
4
5
6
7
8
9
9
aaa2
4 4 2 1
27
a a 2a
4 4 2 1
81
aa3
4 4 2
27
aa3
4 4 2
|x|
Term
I/Q Representation
xI + jxQ
z
3
(x )
x2 z
2
(x ) z
x z 2
x (z )
z2z
3
(z )
90
x3I + jx3Q
x2I zI + jx2Q zQ
2
xI zI2 + jxQ zQ
3
zI3 + jzQ
4.2 Components
to their power contribution in descending order, while the time variable t is omitted to
enhance readability. The last column in Table 4.1 provides the I/Q branch counterparts to
the complex terms, indicating that each term in I/Q form corresponds to two complex terms.
Doing the same analysis with I/Q imbalance included results in another nine terms that are
the complex conjugates of the terms listed in Table 4.1. In general, all distortion terms can
be considered if all of them have separate parallel branches. However, not all terms have a
significant contribution in practice, hence, the architecture can be simplified further.
Only the most significant distortion terms plus their complex conjugates should be considered in order to reduce the computational complexity and to overcome difficulties in the
adaptation process for weak terms having contribution below the noise power level. In addition, the terms should be carefully selected so they do not have much spectral overlap. The
final terms that have been selected for the parallel mitigation architecture are listed in Table 4.2. Hats () and the discrete variable n are used in the term notation to indicate that an
Table 4.2: Selected terms for parallel mitigation architecture [GAM+14].
No.
Term
Filtering
x (n)
SL
z(n)
WL
WL-RC
x (n)
estimate of the distortion-producing blocker signal x(t) has been obtained by filtering in the
digital domain. In fact, the blocker estimate x(n) also contains in-band distortion, a problem that is further detailed in Section 4.5. The third column indicates if the terms should
have strictly-linear (SL), widely-linear (WL), or reduced-complexity widely-linear (WL-RC)
filtering [GAM+14; ALA09]. This is related to the adaptive filtering implementation and
is discussed later in this section. They mainly differ in the number of complex terms to be
processed in the AF stage.
Term 1 holds the mirror of the blocker and has typically the strongest contribution that
can be easily assessed from the IRR of the receiver. Term 2 has contributions from RF and
BB non-linearity, and Term 3 is the BB distortion in the fundamental and third harmonic
zone. It is likely that the remaining terms are negligible due to 5th , 7th , and 9th order
terms. The previous terms have a maximum order of three and therefore represent the
most significant distortion that finally need to be subtracted from the desired signal with a
least-mean square (LMS) AF.
91

In order to demonstrate the relevance of the distortion terms, power levels of the distortion
Terms 1-6 of Table 4.1 are studied by means of a two-tone simulation. Noise and I/Q
imbalance (mirror-frequency interference) are not considered in this simulation for the sake
of simplicity. Figure 4.4(a) depicts the distortion estimates for dominating RF non-linearity,
assuming a vector a = [a1 , a2 , a3 , a4 ] = [5.62, (84351 + j74391), 3.16, 1588.7], i.e. having
|a2 | > |a4 |. These parameters correspond to the IIP3 and gain values listed in Table 4.3. It
Relative Power (dB)
50
0
50
80 dBFS
1
2
3
4
5
6
100
150
200
10
5
0
5
(a) Dominating RF non-linearity.
50
Relative Power (dB)
Term
Term
Term
Term
Term
Term
0
50
80 dBFS
10
Term
Term
Term
Term
Term
Term
1
2
3
4
5
6
100
150
200
10
5
0
5
(b) Dominating BB non-linearity.
10
Figure 4.4: Power levels of distortion estimates Terms 1-6 of Table 4.1 [GAM+14].
is apparent that the Terms 2 and 3 are the most significant values in this case. Furthermore,
it is sufficient to consider the common spectral content of multiple terms by only one term,
as the AF adapts them to the total distortion at these frequencies. That is, only distortions
of higher-order terms need to be considered that have not been covered already by the
lower-order terms.
92
4.2 Components
Table 4.3: Parameters of the Two-Tone Simulation
Parameter
Value
10 dBm
IIP3RF
15 dB
GRF
IIP3BB
+6 dBm
10 dB
GBB
30 dBm
Pinput
In addition, SNR of the ADC is an important limiting factor for resolving weak distortion
terms. The SNR of a typical ADC is approx. 6080 dB depending on its resolution. Thus,
all terms below 80 dBFS are covered by the quantisation noise. Consequently, Terms 4-6
in Figure 4.4 are likely to be masked by noise in practice. Typically, the ADC quantisation
noise is stronger than noise introduced by the analogue RF front-end.
Figure 4.4(b) illustrates the case for dominating BB non-linearity with a
[3.16, 1588.7, 5.62, (84351 + j74391)], i.e. having |a2 | < |a4 |. These values result
from exchanging the IIP3s and gains in Table 4.3 of RF and BB stage to make the BB stage
the dominating non-linearity. Again, Terms 2 and 3 are the most significant ones and IMD
in the fundamental and third harmonic zone are almost equally strong, indicating that only
little RF distortion is present. Indeed, the order of the meaningful distortion terms can
slightly vary.
Finally, it is assumed that the selected distortion terms of Table 4.2 are sufficient for any
practical values of RF and BB non-linearities. However, the mitigation architecture can be
easily extended for front-ends manifesting a higher sensitivity, severe non-linearity, or both
that make the inclusion of higher-order terms necessary.
Another limiting factor for the practical non-linearity modelling is the ADC sample rate fs
used for the mitigation processing. The blocker should have a frequency less than fs /6 to
model third-order non-linearity without aliasing. The reason is that any third-order nonlinearity occupies three times the bandwidth of the distortion-producing blocker band.
As a conclusion of the aforementioned findings, a practical mitigation architecture is proposed
that is a carefully found compromise between the computational complexity and the achievable mitigation performance [GAM+14]. This architecture is proposed in Figure 4.5. Here,
Terms 1 to 3 are moved in parallel and manifest separate branches that are adapted simultaneously by multitap AFs to consider also frequency-selective nature of the non-linearities
(memory effects). Some branches employ a real bandstop, depicted by hR
BS in Figure 4.5,
93
yBB
(n)
hC
BS
hC
BP
e(n)
d(n)
x(n)
(.)
SL
w x
I/Q Imbalance
hR
BS
(.)2 (.)
wz
wz
wx3I
WL
(.)
RF Non-linearity
BB Non-linearity (I)
Re
(.)3
hR
BS
WL-RC
Im
hR
BS
w x3Q
BB Non-linearity (Q)
Figure 4.5: Parallel mitigation architecture for cascaded non-linearity [GAM+14].
that filters out the spectral content from the original blocker band. Otherwise, there is a
risk of misadjustment by the AF stages as the desired path d(n) does not hold any content
from the original blocker band. Different types of filtering have been implemented to find
a good trade-off between computational complexity and the additional complex conjugate
terms. For Term 1, x (n), SL filtering is deployed which means that one set of complex AF
coefficients w x is found. For Term 2, z(n), also the complex conjugate need to be included
to consider mixer I/Q imbalance effects. Therefore, WL is used to find two separate complex
sets of AF coefficients w z and w z . In addition z(n) need to be filtered to cancel the blocker
band content, whereas no filtering is necessary (or at least not harmful) for z (n) as it contains no signal content at the original blocker band. WL filtering is also used for Term 3, i.e.
x3 (n) and [
x (n)]3 . As both terms need to be filtered by hR
BS , the WL-RC approach can be
exploited, which means that finding two complex sets of coefficients for the complex signal
and its conjugate is equivalent to finding two complex set of coefficients for the real and
94
4.2 Components
imaginary part of that signal [ALA09; NNS10; GAM+14]. Hence, the complexity of WL is
reduced to that of SL. In other words, finding AFs wx3 and wx3c for
w x3 (n) x3 (n) + wx3c (n) [
x (n)]3
(4.3)
is equivalent to finding filters w x3I (n) and w x3Q (n) for real and imaginary parts of x3 (n), i.e.,
x3 (n)] + w x3Q (n) Im[
x3 (n)].
wx3I (n) Re[
(4.4)
Altogether, the computational complexity is reduced by factor two, because complex AFs are
applied for real-valued signals in (4.4) instead of complex-valued signals as in (4.3). Due to
the selected filtering approaches, there are in total five distortion branches in the mitigation
architecture of Figure 4.5. Note that due to the complex implementation of the AFs also
AM/PM distortion at the RF non-linearity, amplitude and phase mismatch at the mixer,
and I/Q imbalance in the physical I and Q BB branches are taken into account.
Single RF or BB Non-linearity
If the non-linearity of either the RF amplifier (LNA) or the BB amplifier is dominating, the
single type can be considered by just using a subset of the branches of the proposed parallel
mitigation architecture in Figure 4.5. Adopting just the BB non-linearity model, i.e. the
separated I/Q processing of the blocker raised to its third power, is exactly the procedure
followed in the prior work, such as in [VSHGA+06; gha11; GSH+12a]. RF-only modelling
has been adopted in [GSH+13; SGT13], assuming severe LNA non-linearity. In fact, the
BB-only model sufficiently mitigates the non-linear distortion if the receiver manifests only
very mild RF distortion. In Section 4.4, it will be shown that the proposed cascaded model
clearly outperforms the RF-only and BB-only model mitigation architectures.
4.2.3 Adaptive Filter

The reference distortions are further processed by an AF to adjust amplitude and phase to
the actual distortions in the desired signal branch so that they can be finally subtracted (mitigated). As such parameter learning algorithms are considered as advanced signal processing
techniques, a short survey on AF theory is given in the following paragraphs. At the end of
the section, a compact notation of the AF is given for mitigating cascaded non-linearity.
95

AFs have been successfully applied in many different technologies, such as communications,
radar, sonar, seismology, and biomedical engineering [Hay02]. They are used in all cases,
where a system should respond to a changing input during runtime. That is, an AF adjusts
its free parameters (coefficients) in accordance with the input data. Four main classes of
applications are distinguished: identification, inverse modelling, prediction, and interference
cancellation. The latter one is of interest for mitigating non-linear distortions in a received
signal.
Figure 4.6 illustrates the basic architecture of an AF applied for interference cancellation. A
Desired Signal
d(n)
Adaptive Filter
Delay 0
Input Signal
u(n)
w0
Delay 1
z 1
y(n)
Error Signal
e(n)
w1
Delay M 1
z 1
Algorithm for
adaptive
or optimal design
wM 1
Figure 4.6: Basic AF architecture for interference cancellation

classical application of this AF configuration is the removal of the power-line hum of 50 Hz
from an information-bearing desired signal. In the distortion mitigation application, the
AF is used to cancel unknown interference contained in the primary (desired) signal. A
reference signal, containing an estimate of the distortions generated by the non-linear frontend, is employed as an input to the filter. For this application, the reference signal is first to
be derived by passing the distortion-producing blocker(s) through the reference non-linearity
model. Then, the main task of the AF is to adjust these estimates so they match those in the
desired signal. Finally, a cleaned desired signal is obtained by subtraction of the adjusted
96
4.2 Components
reference distortions. In the following, it is discussed how the algorithms for an optimal or
adaptive design of the filter parameters wk look like.
Optimal Design: Wiener Filter

First, the optimal design, so called Wiener filter, is considered. In the following discussion,
FIRs are considered due to reasons of stability and practical considerations. The output
signal of the filter is given by (4.1)
y(n) =
X
k=0
wk (n)u(n k), n = 0, 1, 2, . . . .
(4.5)
The algorithm should find the parameters {w0 , w1 , w2 , . . . } such as to minimize the
mean-square error (MSE) defined as

J = E {e(n)e (n)} = E |e(n)|2 ,
(4.6)
i.e. the expectation value E {.} of the square of the error signal with
e(n) = d(n) y(n).
(4.7)
The criterion for the cost function J to attain its minimum is that the kth entry of its
gradient must be identically zero, that is
J
J
+j
ak
bk

e(n)
e (n)
e (n)
e(n)
=E
e (n) +
e(n) + j
e (n) +
e(n)
ak
ak
bk
bk
k J =
= 2E {u(n k)e (n)} 0,
(4.8)
where the kth filter parameter wk is denoted in terms of its real and imaginary part wk =
ak + jbk . Equation (4.8) yields the fundamental principle of orthogonality, which states that
the error signal eopt (n) of the filter in optimum condition is orthogonal to the samples u(n)
of the AF input:

E u(n k)eopt (n) = 0, k = 0, 1, 2, . . . .
(4.9)
It can be also used to test if a given filter is operating in its optimum condition. As a
corollary of the principle of orthogonality, it can be shown that also the filter output yopt (n)
97

and the error eopt (n) are orthogonal to each other. Expanding equation (4.9) with equations
4.5 and 4.7 yields
E
X
i=0
u(n k) d (n)
X
i=0
wopti u (n i)
!)
=0
wopti E {u(n k)u (n i)} = E {u(n k)d (n)}
(4.10)
In fact, the left side of equation (4.10) contains the autocorrelation function r(i k) of the
input at lag i k, and the right the cross-correlation p(k) between the filter input u(n k)
and the desired signal d(n). Replacing the expectations with the correlation functions yields
the famous Wiener-Hopf equations in its general form
X
i=0
wopti r(i k) = p(k).
(4.11)
Assuming now an FIR with M parameters (length of its impulse response) reduces equation (4.11) to a system of M equations. Then, matrix representation of the Wiener-Hopf
equations, now in complex notation, can be formulated with
u = [u(n) u(n 1) . . . u(n M + 1)]T
r(0)

r (1)
R = E u(n)uH (n) =
..
r (M 1)
r(1)
r(0)
..
.
..
.
r (M 2)
p = E {u(n)d (n)} = [p(0) p(1) . . . p(1 M)]T

T

w opt = wopt0 wopt1 . . . woptM 1 ,
r(M 1)
r(M 2)
..
r(0)
where R is the autocorrelation matrix of the AF input in Toeplitz structure, i.e. it can be
built from the first row and first column vector only (symmetry along the diagonal). The
Wiener-Hopf equations in its compact form are then given by
Rwopt = p.
(4.12)
Assuming that the inverse R1 exists, the optimum tap-weight vector of the FIR is
w opt = R1 p.
98
(4.13)
4.2 Components
This solution can be obtained if the underlying statistics of the stationary process, meaning
the autocorrelation of the input and the cross-correlation of the input with the desired signal,
are known. The cost function in equation (4.6) can be written in matrix-vector representation
as
Jw = d2 2pH w + wH Rw,
(4.14)
thus, yielding the minimum MSE with the obtained Wiener-Hopf equations
Jwopt = d2 pH R1p.
(4.15)
Adaptive Design: LMS Algorithm

Second, the adaptive design of the filter parameters is considered, which approaches iteratively to the Wiener solution. The adaptation of the filter parameters can be achieved by
evaluating the gradient of the performance criterion and determining its steepest descent.
The gradient of J(n) (equation (4.14)) with respect to the parameters w(n) is
w(n) J(n) = 2p + 2Rw(n) = 2E {u(n)e (n)} .
(4.16)
From this equation, the adaptation mechanism can be formulated. The idea is to find a new
set of filter parameters by taking the old ones and adding something. By moving a small step
to the opposite site of the gradient vector ( in front of ), the parameters will approach
to those of the optimal Wiener filter, no matter where the iterations are initialized [Hay02].
Hence, the steepest descent algorithm can be written as
1
w(n + 1) = w w(n) J(n)
2
= w(n) + E {u(n)e (n)} .
(4.17)
This algorithm will track the time variations of the signals statistics without solving the
Wiener-Hopf equations every time the statistics change. Applying the steepest descent
method to the Wiener filter yields

w(n + 1) = w(n) + p Rw(n) .
(4.18)
Since the terms E {u(n)e (n)} and p Rw(n) are equivalent, an error-driven adaptation
process can be formulated. This is the basis for the common LMS algorithm, named by
its originators Widrow and Hoff (1960, [Hay02]). To simplify the algorithm further, the
99

expectation operator in equation (4.16) can be neglected, that is, an immediately available
approximation is used to get an estimate of the gradient
w(n) J(n) = 2u(n)e (n).
(4.19)
Hence, the tap-weight adaptation in the LMS algorithm is done with

w(n + 1) = w(n) + u(n)e (n).
(4.20)
In practice, exact measurements of the gradient vector are not possible, because this would
require prior knowledge of the correlation matrix of the inputs and the cross-correlation
between the input and the desired response.
Note that due to the efficient method of the steepest decent, actually no a-priori knowledge
of the non-linearity coefficients is necessary. Even if intentionally wrong values are used as
initial parameters, the LMS algorithm will converge to the true values very fast. That is, the
effort for detailed characterisations of the non-linearity is not necessary for this application.
It is more important to choose the core of the model in a way that it matches the physical
architecture of the front-end, as already mentioned.
Applying this simplification, computing correlation functions and matrix inversion are not
required any more. Algorithms based on 4.20 are simple to design and highly effective in
performance. As a consequence of equation (4.19), the adaptation suffers from gradient noise,
that is, there will be a random motion around the minimum MSE Jmin . A general result
and limit for LMS-AFs using only first order information is the slow rate of convergence,
however, they can react immediately to changes in the input signal statistics.
With properly set step size the parameters w and the MSE will converge to the Wiener
optimal solution, that is
w(n) w opt
J(w(n)) J(w opt ).
From the analysis of the convergence of the average parameter vector E {w(n)}, a stability
condition for the LMS algorithm can be obtained

0<<
100
2
max
(4.21)
4.2 Components

where max is the largest eigenvalue of the autocorrelation matrix R = E u(n)uH (n) .
Indeed, the eigenvalues are not known in practice. If obeys this condition, the LMS
converges in mean to the Wiener solution. Another criterion for can be derived from the
convergence analysis of the MSE which yields
<
2
Power of the input
for J Jopt ,
(4.22)
where J is the steady-state value of the MSE. This relationship for can be used in
practical implementations as the power of the input can be easily determined. The choice
of a small is accompanied with a trade-off between convergence in speed and MSE. Using
a small , the adaptation progresses slowly while achieving a small misadjustment between
the steady-state MSE J and the minimum MSE Jopt . The reciprocal of can be viewed
as the memory of the LMS filter. Low causes long memory, that is, a large number of
samples need to be processed to reach a significant small MSE. Choosing a large results
in fast adaptation, but with increased misadjustment for the MSE.
There are several variants of the LMS algorithm, among others, the normalised LMS algorithm with the updating equation
w(n + 1) = w(n) + (n)u(n)e (n) with
=
(n) =
,
PM 1
2
+ ME {u(n)2 }
+ ku(n)k
+ i=0 u(n i)2
(4.23)
and a small constant > 0 (overcome numerical difficulties when ku(n)k is close to zero) as
well as the adaptation constant
. In the normalised least-mean square (NLMS) algorithm,
the step size changes every adaptation while taking the power of the input into account, that
is, the adaptation becomes data-dependent. The power is computed as the sum of squares
of the past M samples. Thereby, large fluctuations in the power levels and the length N of
the input vector by reducing the step size are considered. The stability condition for NLMS
reduced to the simple relationship
0<
< 2.
(4.24)
In fact, the small constant a also depends on the input signal power, that is, the NLMS
algorithm has 2 degrees of freedom, the numerical constant a and the adaptation constant
.
As a result of a direct comparison between LMS and NLMS, the normalised version achieves a
similar steady-state MSE but with faster convergence. Assuming similar convergence speed,
NLMS provides lower steady-state MSE. In other words, the NLMS offers better trade-
101

offs between convergence speed and stationary MSE than conventional LMS at the cost of
slightly higher computational complexity. In practice, (4.24) eases the choice of the step
size dramatically compared to the LMS. This is especially important for mitigating cascaded
non-linearity with multiple distortion estimates, that can have very different powers varying
over several orders of magnitudes.
Alternative algorithms try to monotonically reduce the step size in such a way that a large
is used at the beginning for fast adaptation and a very small is applied at the end to achieve
a small MSE. However, these algorithms cannot react to changes of a non-stationary input
any more. Sign algorithms achieve a very fast adaptation by modifying the estimate of the
through replacing the multiplications with the sign of the error signal, the input,
gradient J
or both. Beside a fast convergence, the steady-state error will be higher. Further approaches
base on smoothing the gradient estimate by filtering, double updating by computing the filter
output with the old and new set of parameters, or the extension with quadratic combinations
of the input for filtering (LMS Volterra algorithm). For details on these algorithms, the
reader is referred to [Hay02].
Linear Least Squares Filtering
The method of the LS can be used as an alternative approach to develop a linear AF to

adjust the distortion estimates. It provides a prediction for the optimal weights for the
distortion estimates so they fit the desired signal. The advantage of the LS approach is the
existence of closed-form expressions for the estimate as well as for its properties. The result
depends on the number of samples N used for the computation. That is, the LS solution is
computed block-wise. The LS can be seen as an alternative to the Wiener-Hopf solution in
equation (4.13). While the Wiener solution is derived from ensemble averages (set of multiple
realisations) and the parameters are the optimal ones for all realisations of the underlying
physical process, the LS finds the optimum parameters for a single realisation, i.e. for the
given data. In other words, the Wiener solution is the average of the block-wise LS solution
and requires the full statistics of the signal. Although both are fully theoretical approaches
and cannot be implemented in practice, they provide a kind of upper bound for the best
achievable mitigation performance.
102
4.2 Components
Assuming a set of distortion estimates samples ui and the desired signal vector d, the task
is to find the parameters wki of the linear filter with the output
yi (n) =
M
1
X
k=0
wki ui (n k) = w H
i (n)ui (n).
(4.25)
The estimation error is defined as the difference between the desired response and the filter
output [Hay02]
e(n) = d(n) y(n).
(4.26)
In the LS method, the goal is to minimize the cost function

E(w0 , w1, . . . , wM 1 ) =
i2
X
i=i1
|e(i)|2 ,
(4.27)
that is, the sum of error squares, where i1 and i2 are the index limits where the minimisation
is done. While the error criterion of the Wiener filter in equation (4.6) used expectations, this
cost function is now closer to the actual data. To find the filter parameters w for the minimal
error emin when the cost function is minimized, a so called normal system of equations is
used [Hay02]. It can be derived by the gradient vector E or the principle of orthogonality.
The matrix formulation of these equations is
w = ,
(4.28)
where and are the correlation matrix of the input u and the cross-correlation vector
between the input and the desired signal, respectively. The tap-weights of the LS filter are
then
w = []1 ,
(4.29)
assuming that the inverse of exists. Equation (4.29) is the counterpart to the solution of
the Wiener-Hopf equations in equation (4.13). A compact form of equation (4.28) with the
actual data matrices is derived with the help of the following redefinitions:
= AH A
= AH d,
103

where A is a rectangular matrix constructed from the input samples by

u(i1 )
u(i1 1) u(i1 2) u(i1 M + 1)
u(i1 )H

u(i1 + 1)
u(i1 )
u(i1 1) u(i1 M + 2) u(i1 + 1)H
,
=
A=
..
..
..
..
..
..

.
.
.
.
.
.

H
u(i2 )
u(i2 )
u(i2 1) u(i2 2) u(i2 M + 1)
(4.30)
where the limits i1 and i2 belong to a used data window. Equation (4.29) can then be
rewritten as
w = AH A
1
AH d,
(4.31)
containing only the known input and desired signal. This solution is equal to LS approximation of the general overdetermined system of equations in linear algebra [Str09]
Ax = b,
(4.32)
where A is a K M matrix and b a column vector of length K (K 1). The matrix A has
more rows than columns (K > M), which is the common case in this application. There are
many samples of input data, but only M unknown coefficients. Then there is a best x
which
minimizes the sum of squares of the errors in the K equations
kAx bk2
(4.33)
with
x
= AH A
1
AH b = A+ b.
(4.34)
x
yields the combination of the normal equations with the smallest MSE. In other words,
the estimate can be computed by the pseudoinverse (or Moore-Penrose generalized inverse)
1 H
A+ = AH A
A . Equation (4.31) and (4.34) are equivalent. In other words, x
fits the
distortion estimates to the sample vector in the LS sense, without having an AF taken into
consideration. The filter coefficients became here the unknown coefficients of a model, which
fits to the data.
Finally, the adjusted estimates can be subtracted from the desired signal, to get the cleaned
signal
x.
y = b A
104
(4.35)
4.2 Components
Attention has to be paid on the construction of the data matrix A. In the simplest case, it
contains the distortion estimate sample vectors in the columns. It is important to mention
that the condition number of A, indicating the accuracy for the matrix inversion, determines
the quality of the results for the found coefficients. A condition number of 1 indicates a
well-conditioned matrix. If it raises significantly, e.g. by adding more correlated distortion
terms in the columns, the LS solution differs a lot from the LMS solution. High condition
numbers basically mean that there are multiple solutions for solving the data fitting problem.
Therefore, also different coefficients by LMS and LS will be found.
Finally, the LMS algorithm is chosen instead of the LS method. The LS filtering requires a
long number of samples to provide useful results. For practical implementation, it is difficult
to store and process a high amount of data, and to subsequently correct the cached desired
signal with the found coefficients. Instead, the LMS algorithm with a continuous adaptation
at every sample instant is much easier to implement. However, the LS is computed in
simulations to compare it with the LMS solution.
LMS Notation for Cascaded Non-linearity
Finally, a vector notation for mitigating cascaded non-linearity based on LMS-based learning
algorithm is given [GAM+14]. The special aspect for the non-linearity mitigation is that
multiple reference signals, the distortion estimates, need to be subtracted from the desired
signal at once. Furthermore, multi-tap AFs with a filter length M > 1 are typically required
to compensate also for memory effects in the non-linear receiver front-end.
Following Figure 4.5, there are in total five distortion estimate signals that can be combined
in one vector s(n) of size 5M 1, namely
s
x
sz
s(n) = sz
sx 3
I
sx3Q
(4.36)
where the subscripts indicate the distortion branches. For instance, Term 1 vector is
x (n), x (n 1), . . . , x (n M + 1)]T .
sx (n) = [
105
(4.37)

Note that the vectors sz (n), sx3I (n), and sx3Q (n) also include filtering with hR
BS to remove the
original blocker band content from the distortion estimate. Then, sz (n), e.g., look like
zfilt (n), zfilt (n 1), . . . , zfilt (n M + 1)]T ,
sz (n) = [
(4.38)
where zfilt (n) denotes the bandstop-filtered version of z(n). Next, also the AFs are combined
in one vector w(n) of size 5M 1 given by
w
x
wz
w(n) = w z
w x3
I
wx3Q
(4.39)
where the subscript indicates the corresponding distortion branch. Note that each individual
AF has a length M. Also different AF lengths for different distortion estimates can be employed, if e.g. RF and BB non-linearities manifest different levels of frequency-selectivity.
The combined AF output is then computed by
x(n) = w H (n)s(n),
(4.40)
e(n) = d(n) x(n).
(4.41)
and the resulting error signal
Finally, the tap-weight update equation according to (4.20) is given by

(4.42)
w(n + 1) = w(n) + diag()s(n)

x(n),
where diag(.) denotes conversion of a vector to a diagonal matrix. Thereby, every distortion
estimate is processed by a different step size. They can be collected in a step size vector
i
h
= x , z , z , x3I , x3Q ,
(4.43)
where the subscripts indicate the corresponding distortion branches. Following the NLMS
algorithm according to (4.23) yields step-size vector
NLMS =
x
,...,
+ ksx (n)k2
106
x3Q
x3Q
2 ,

+ sx3Q (n)
(4.44)
4.3 Computational Complexity

i
h
with the additional selectable numerical constants = x , z , z , x3I , x3Q . NLMS is
chosen in Section 4.4 due to its preferences in convergence and choosing the step size.
4.3 Computational Complexity

With the system-level implementation of all three main components (FIR filter, reference
non-linearity, and AF), the complexity of the digital feedforward mitigation algorithm can be
computed. The overall complexity is summarised in Table 4.4. The complexity for creating
Table 4.4: Computational complexity for mitigating cascaded non-linearity [GAM+14].
Operation
# Ops.
Ref. Modeling
Mutiplications
Summations
Divisions
16
12P 8
8M
Static Filtering
12P
SL LMS
8M + 2
WL-RC LMS
8M + 2
8M
WL LMS
16M + 2
16M
NLMS Scaling
10M + 5
5M + 5
Overall
12P + 42M + 27 12P + 37M + 5
the distortion estimates is 16 real multiplications and 8 real summations according to the
complex-valued operations in Figure 4.5.
Add to this, there is the cost of the five filters applied in the mitigation architecture, namely
C
R
the two complex filters hC
BP and hBS and three instances of the real-valued filter hBS . If P
denotes the order of the direct-form FIR filters, P multiplications and P 1 summations
are necessary for real-valued filters, processing a real-valued signal [MRS91]. Same orders
are assumed for all filters, as same requirements in transition bandwidth and stopband
attenuation hold for all of them. Two instances of hR
BS process real-valued signals (BB nonlinearity branches), whereas all other filters operate on complex-valued signals. Multiplying
complex input with a complex tap weight requires 4P real multiplications and 2P + 2(P 1)
summations, where the 2P comes from the complex multiplications and 2(P 1) summations
from the parallel filtering of I and Q signals. Hence, the computational load of the two
C
complex filters hC
BP and hBS processing complex signals is twice that amount, i.e. 8P real
multiplications and 8P 4 summations. Then, there is one filter hR
BS operating on complex
signals, requiring 2P real multiplications and 2(P 1) summations for filtering the I and Q
branches. The remaining two real filters on real signals need 2P multiplications and 2(P 1)
107

summations. Overall, the complexity of the filtering sums up to 12P multiplications and
12P 8 summations.
Finally, there is the cost of the adaptation by the complex LMS. The complexity of the
conventional LMS is 2M + 1 complex multiplications and 2M complex summations per
iteration [Hay02]. The SL and WL-RC cases require 8M + 2 real multiplications, 4M each
AF filtering and tap-weight-update plus two for the multiplication of real and imaginary
part with the step size. The number of summations accounts for 4M 1 real summations
for the tap-weight updating and the AF filtering, respectively, plus two summations in
the tap-weight update equation and the error signal equation. Hence, the total amount
of real summations for SL and WL-RC is 8M. The complexity for WL LMS is twice the
amount of SL LMS as two complex signals (strict and conjugate) need to be processed,
introducing 16M + 2 real multiplications and 16M summations. Including also the NLMS
update introduces additional 2M real multiplications and M real summations for computing
the euclidean norm plus 1 real division (Divs) and 1 real summation (with the numerical
constant) for finally scaling the step size. Thus, the additional amount for NLMS scaling
of five distortion estimates is 10M + 5 real multiplications and 5(M + 1) = 5M + 5 real
summations.
The final numbers of multiplications, summations, and divisions are summed up in the last
row of Table 4.4. The introduced processing delay of the digital feedforward mitigation is
P + M that is practically dictated by the filter order P , as P M.
In particular, the computational load is important for real-time implementation, as discussed
in Chapter 6, and for performance comparisons with other linearisation techniques.
4.4 Proof of Concept

This sections provides simulation and RF measurement results which evaluate the performance of the proposed digital feedforward mitigation. In particular, explicit comparisons between the novel joint processing of RF and BB distortion and the RF-only
and BB-only approaches are conducted. These investigations base on two-tone and
binary phase shift keying (BPSK) excitation signals that are used for both simulations and
measurements. These results have been published in [GAM+14].
108
4.4.1 Performance Simulations

The basic simulation architecture is depicted in Figure 4.7. The complex BB signal passes
an AWGN source, the RF non-linearity, the I/Q imbalance block, and the BB non-linearity.
All the blocks within the dashed frame mimic a DCR front-end at equivalent BB. In the
simulations, the SNR of the blocker is 61 dB, related to the entire BB bandwidth of 25 MHz.
The IRR is set to 30 dB, a value that is realistic for an integrated DCR [Raz97]. Finally,
the composite signal is processed by the digital feedforward mitigation algorithm presented
in the previous sections.
Two-tone and BPSK signals have been generated as excitation signals to cover two signal
classes, CW and modulated signals, for performance evaluation of the algorithm. The following investigations are evaluated from the sensing point of view (use case 2), i.e. there
is no specific signal of interest. Instead, the processing aims for cleaning the whole BB
spectrum from distortions meaning that only the strong input signal should be left after
mitigation. For this purpose, the blocker is added back to the output signal e(n) of the
mitigation structure, as described in Section 4.1.
As there is a perfect match between the model mimicking the DCR in the simulator and
the reference non-linearity model in the mitigation architecture, single-tap complex AFs are
sufficient in simulations to accomplish full cancellation of the distortion. In addition, both
models are memoryless. After mitigation, block-wise 1024-point FFT is applied to analyse
and illustrate the remaining non-linear distortion. Always the last one of 29 FFT blocks is
shown in which the convergence of the AF coefficients is guaranteed.
The mitigation gain is computed as a figure of merit in dB that describes the reduction
of average signal power outside the original input (blocker) frequency band. The blocker
mirror band has been also excluded from the mitigation gain calculations as the focus is
purely on non-linear distortion mitigation. The mitigation gain computation is established
RF3
I/Q
Imbal.
BB3I
BB3Q
yBB
Mitigation
Algorithm
e
FFT
AWGN
Figure 4.7: Simulation architecture for performance evaluation of the mitigation algorithm
in MATLAB [GAM+14].
109

in the frequency domain by the following rule

Yin(blocker mirror)
MG = 10 log10 mean
,
Ymit (blocker mirror)
(4.45)
where Yin and Ymit denote a matrix with the set of block-wise FFTs of the original input
and the mitigated output, respectively. The logical indices contain all bins of the FFT block
that do not belong to the blocker or mirror band. Compared to the BPSK case, definition
of the mitigation gain for the two-tone case is a bit different. Here, the gain is computed
explicitly at the actual frequency components containing the non-linear distortion, according
to the rule

Yin (spurs(s))
,
(4.46)
MGTT = 10 log10 mean
Ymit(spurs(s))
where spurs(s) is a vector holding discrete indices of the spurious frequency components
caused by the two-tone input. In addition, mirrors of the distortion products are considered
in simulations due to exaggerated coefficients used for the models. However, in measurements, these components do not appear above the noise level. Consequently, the absolute
mitigation gain obtained from simulations is a bit higher.
The parameters of the performance simulations are summarised in Table 4.5, extending the
values of Table 4.3. It lists all coefficients of the simulator model including the non-linearity
Table 4.5: Parameters of the performance simulations for mitigating cascaded non-linearity.
Parameter
Value
IIP3RF
10 dBm
GRF
15 dB
IIP3BB
+6 dBm
GBB
10 dB
Pinput
30 dBm
a
gm
m
[5.62, (84351 + j74391), 3.16, 1588.7]

0.99
IRR 30 dB
0.0628 =
b 3.6
[1, 1, 0.01, 1, 1]
[109 , 108 , 104, 109 , 108]
and I/Q imbalance coefficients. In addition, phase distortions created by the LNA and the
110

mixer are considered by complex coefficients a2 , k1 , and k2 . The coefficients a1 . . . a4 have
been computed based on the listed gain and IIP3 values of the RF and BB amplifiers by
utilising the conversion formulas (2.15), (2.16), and (2.17).
Two-tone Blocker Input

Figure 4.8(a) illustrates the BB spectrum with the two-tone input and after mitigation with
the cascaded model. In total, 16 components are created due to the receiver non-linearities
and I/Q imbalance. The mirror-frequency appears 30 dBc (below the original tones), whereas
the strongest distortion in the fundamental zone is 36 dBc. First, the two-tone blocker is
distorted by the RF amplifier, generating IMD in the fundamental zone at around +2.6 MHz
(center of the two tones). Second, mirror-frequency interference of the blocker and the
RF distortion appear due to down-converting mixer I/Q mismatches. Third, blocker, RF
distortion and their respective mirror components are all further distorted by the BB I
and Q non-linearities. Hence, distortion in the third harmonic zone at around 7.8 MHz
and in the fundamental zone are added. In addition, the BB non-linearity causes HD and
IMD originating from the mirror components which then appear as low-power tones around
+7.8 MHz. Furthermore, the BB distortions can have respective mirror components, if there
is I/Q imbalance in the BB branches, although these components are very likely to remain
below the noise floor in practice. In fact, the mirror components of the main RF and BB
distortions are mitigated with the WL filters in the proposed structure if they do appear.
Furthermore, AM/PM distortion caused by the RF amplifier [Ken00] and phase mismatch
introduced by the mixer are taken into account by the complex implementation of the AFs.
In Figure 4.8(a), the average suppression of non-linear distortion with the total cascaded
model is 32.6 dB, indicating full cancellation of all distortion and mirror components. The
convergence of the AF coefficients by using the NLMS algorithm is illustrated in Figure 4.9.
The deployed NLMS parameters are also listed in Table 4.5. In Figure 4.8(a), magnitudes
of the adapted complex coefficients are shown together with the corresponding LS solutions
for the coefficients, depicted by dashed lines. In general, there is a good match between the
steady-state solutions of the adapted coefficients and the LS solutions. About 3000 samples
(iterations of the AF) are required to achieve convergence.
Beside the total cascaded model, simulations with the RF-only and BB-only model are also
shown in Figure 4.8. However, the same simulator model mimicking the DCR front-end has
been used for all cases. For the RF-only results in Figure 4.8(b), only the blocker mirror
and the RF distortion in the fundamental zone are mitigated (Terms x and z). Since
no distortion estimate for the components in the third harmonic zone are provided by the
111
Relative Power (dB)
Relative Power (dB)
Relative Power (dB)
replacemen
0
TT Input
Before Mitigation
10
After Mitigation
20
RF+BB Distortions
Mirror-Frequency
30
BB Distortions
40
BB Distortions
50
(Mirror+TT)
60
70
80
90
100
10
5
0
5
10
(a) Total cascaded non-linearity model.
0
TT Input
Before Mitigation
10
After Mitigation
20
RF+BB Distortions
Mirror-Frequency
30
BB Distortions
40
BB Distortions
50
(Mirror+TT)
60
70
80
90
100
10
0
5
10
5
(b) RF-only model.
0
TT Input
Before Mitigation
10
After Mitigation
20
RF+BB Distortions
Mirror-Frequency
30
BB Distortions
40
BB Distortions
50
(Mirror+TT)
60
70
80
90
100
10
5
0
5
10
(c) BB-only model.
Figure 4.8: Mitigation results with two-tone excitation obtained with different reference models [GAM+14].
112

wx3I
wz
150
15
100
10
50
0
w x
0.04
wz
60
0.03
40
0.02
20
0.01
0
wx3Q
15
NLMS
5
LS
9 10
103
10
5
0
4 5 6
Samples
9 10
103
Figure 4.9: Adaptation of the AF coefficients with two-tone input (magnitude of complex
coefficients) [GAM+14].
model, no distortion mitigation is achieved here. On the contrary, in Figure 4.8(c), only the
blocker mirror and the third-order non-linearity at BB are considered (Terms x , Re [
x3 ],
and Im [
x3 ]). There is a poor performance of the BB-only model as the RF distortion is
not taken into account. Consequently, distortion products in the fundamental zone, but also
HD and IMD of the mirror image cannot be fully cancelled. However, if the RF and mixer
distortion is mild and does not manifest severe phase distortion (AM/PM), the BB-only
model can totally remove all occuring BB distortion. Finally, the joint processing of RF and
BB distortion clearly outperforms the RF-only and BB-only model considered in the state
of the art.
Additional simulations have been conducted which analyse the mitigation gain vs. the power
of the two-tone input obtained with the different models, to further proof the efficiency of the
total cascaded model. The achieved gain of the three models as well as the ideal suppression,
indicating how much spurious power has been added due to the receiver non-linearity, are
113

illustrated in Figure 4.10. With the deployed IIP3 values of Table 4.5, almost no IMD is
Mitigation Gain (dB)
35
30
RF-only
BB-only
Cascaded
Ideal
25
20
15
10
5
0
50 48 46 44 42 40 38 36 34 32 30
Input Power (dBm)
Figure 4.10: Average suppression of non-linear distortion components for different mitigation
models obtained with simulated data [GAM+14].
generated up to the input power of 46 dBm. Here, only the blocker mirror x is dominating.
As all models include the blocker mirror mitigation, the blocker mirror band is excluded from
the mitigation gain calculations to enhance visibility of non-linear distortion mitigation. RF
and BB distortions raise with increasing input power and different gains are achieved with
different models. The BB-only model performs poorly due to the strong RF non-linearity,
whereas the total cascaded model follows the ideal suppression. That is, all non-linear
distortion products and mirror terms are essentially cancelled. In brief, the total cascaded
model provides much better performance than the RF-only or BB-only non-linearity models
due to their fundamental shortcomings. These results show that proper modelling of the
underlying receiver architecture is essential to achieve improved mitigation performance in
a broad variety of applications.
BPSK Blocker Input
Next, a BPSK-modulated blocker is assumed to show that the digital feedforward mitigation
algorithm is totally independent from the actual blocker waveform. The BPSK signal is
generated with a raised-cosine pulse-shaping filter, with a roll-off factor of 0.5, and a symbol
rate of approx. 788 ksym/s. These values have been chosen arbitrarily to generate a simple
modulated blocker that covers approx. 1 MHz bandwidth. Beside that, the parameters
for the BPSK simulation are exactly the same as listed in Table 4.5. The RMS power
has been chosen to be equal to that in the two-tone case. Due to the modulation, the
PAPR of the BPSK is with PAPRBPSK = 4.1 dB higher than that in the two-tone case
114

(PAPRTT = 3.01 dB). This is important as the peak values of the time-domain signal
determine the level of distortion.
A mitigation gain of 20.4 dB is achieved for the BPSK case, as indicated in the cleaned
Relative Power (dB)
BB spectrum shown in Figure 4.11. The AF coefficients are converged after approx. 4000
0
10
20
30
40
50
60
70
80
90
100
BPSK Input
Before Mitigation
After Mitigation
Mirror-Frequency
BB Distortions
RF+BB Distortions
BB Distortions
(Mirror+BPSK)
10
5
0
5
10
Figure 4.11: Mitigation results with BPSK excitation [GAM+14].

samples, by using the NLMS algorithm. The adaptation of the five distortion branches of the
total cascaded model is illustrated in Figure 4.12. The results with BPSK input verify that
the feedforward mitigation algorithm is able to mitigate distortions induced by modulated
blocker waveforms.
4.4.2 RF Measurements
In order to verify the efficiency of the digital feedforward mitigation with true RF signals and
components, real-world measurements with the USRP+WBX front-end have been conducted
[Ett; GAM+14]. The automated setup for the experiments with the aforementioned signal
scenarios, similar to that employed in Section 2.7.2, is illustrated in Figure 4.13. The signals
were generated by the vector signal generator of type Rohde & Schwarz SMU200A. Before
running the actual measurements, the spectral purity (SFDR) of the generator was checked
with a conventional spectrum analyser, as they suffer from a limited linearity. The generator
itself has a SFDR > 80 dB, i.e. it can be guaranteed that all remarkable non-linear distortions
are solely generated in the non-linear DUT. Signal generator and receiver are synchronized
using a 10 MHz reference signal, enabling coherent sampling for precise power measurements.
The IIP3 of the total receiver RF front-end including RF, BB, and ADC non-linearity has
been determined already in Section 2.7.2 and are listed in Table 2.5.
115
wz
100
wx3I
15
80
10
60
40
20
0
0
w x
0.04
0.03
60
0.02
40
0.01
20
0
w x3Q
15
wz
80
NLMS
5
LS
9 10
103
10
5
0
4 5 6
Samples
9 10
103
Figure 4.12: Adaptation of the AF coefficients with BPSK input (magnitude of complex
coefficients) [GAM+14].
Figure 4.14(a) illustrates the mitigation performance achieved with a two-tone blocker input.
The tone frequencies are the same as in simulations and have been chosen within the grid
of the 1024-point FFT. The generator power was 39 dBm. The centre frequency of the
receiver is 200 MHz, the BB bandwidth being 25 MHz. By using two taps for the AFs, the
average suppression of non-linear distortion components is 25.4 dB in Figure 4.14(a). The
additional taps have been introduced, as the real receiver suffers from memory effects that
are excluded in the simulations. The memory depth of M = 1 found in Section 2.7.3, that
corresponds to two AF coefficients, has thereby been validated through measured data that
has been processed by the feedforward mitigation algorithm.
The obtained suppression at different input power levels is illustrated in Figure 4.15.
Compared to simulations, RF distortion is milder in measurements which explains the relatively high suppression obtained by the BB-only model. Add to this, CD will occur long
before the RF distortion products become remarkable. This is due to the gain settings of
116
R&S SMU200A
Control Bus (GPIB)
Two-tone/BPSK at
58 . . . 38 dBm
USRP N210
10 MHz
Reference
Ethernet
Figure 4.13: Sketch of the measurement setup [GAM+14].
the WBX front-end that employs fixed gain at the RF and BB amplifier. Only the overall
gain can be controlled through a variable attenuator at the RF input (see Appendix B).
Unfortunately, the BB gain of 9 dB is much too high and cannot be easily changed to provoke more RF distortion. However, the cascaded model provides still the best mitigation
performance.
Similar performance has been achieved with the BPSK-modulated signal, as shown in Figure 4.14(b). Here, the centre frequency and the BB bandwidth of the USRP were chosen
to 570 MHz and 25 MHz, respectively. The BPSK signal has been generated with the same
parameters as used in simulations, and the generator power and PAPR were 44 dBm and
3.35 dB, respectively. A mitigation gain of 7.1 dB is achieved using two taps for each AF.
To sum up, the co-existence of RF and BB distortions has been verified through real-world
measurements, demonstrating the effectiveness of the total cascaded model. The suppression of the different distortion components is significantly better with the joint mitigation
of the RF and BB stage non-linearities, compared to the previous solutions presented in the
literature, which employ a model for only either of the stages. Moreover, joint mitigation
of non-linear distortions and corresponding mirror components due to I/Q imbalance has
been demonstrated by measurements. Thus, a complete model has been found capturing
all essential distortion in the RF receiver front-end. Consequently, the digital feedforward
mitigation method provides a high-performance linearisation solution for complete wideband
DCR chains, enabling flexible sensing and processing of the RF spectrum in radio communication and radar devices.
117
Relative Power (dB)
Relative Power (dB)
0
10
20
30
40
50
60
70
80
90
100
0
10
20
30
40
50
60
70
80
90
100
TT Input
Before Mitigation
After Mitigation
Mirror-Frequency
RF+BB Distortions
BB Distortions
10
5
0
5
10
(a) Two-tone scenario with 39 dBm input power.
BPSK Input
Before Mitigation
After Mitigation
Mirror-Frequency
RF+BB Distortions
BB Distortions
10
5
0
10
5
(b) BPSK scenario with 44 dBm input power.
Figure 4.14: Mitigation results obtained with measured data [GAM+14].
The digital feedforward mitigation principle is accompanied by some fundamental limits

and challenges, which have to be considered when aiming for employing this algorithm into
real devices. Basically, these limitations are imposed not only by the available technology,
but also by the mitigation principle itself. These issues as well as some solution ideas are
discussed in the following paragraphs. In addition, a summary like that presented in this
section is not available in the state of the art, hence, it has been included in the publication
[DGS+13].
118
Mitigation Gain (dB)
35
30
RF-only
BB-only
Cascaded
Ideal
25
20
15
10
5
0
50
48
46
44
42
Input Power (dBm)
40
38
Figure 4.15: Average suppression of non-linear distortion components for different mitigation
models obtained with measured data [GAM+14].
Model Mismatch significantly affects the achievable mitigation performance. First, only
distortions created by the reference model can be mitigated by the AF. That is, the core
distortion terms should match the physical structure of the RF front-end as close as possible
to address all possible distortion. This has been shown by the joint modelling of RF and BB
distortion generated in typical DCR front-ends. Otherwise, poor mitigation performance is
achieved by the RF-only and BB-only models aiming for mitigating cascaded non-linearity
in DCR front-ends. In such a case, adding more taps to the AF cannot compensate for severe
model mismatch, since only the provided reference signal content is filtered.
Bandsplit Filters are adopted to derive the desired and reference signal for the AF stage
and are therefore very important for the mitigation performance. Due to the high dynamic
range conditions with weak signals and strong blockers, the filters should provide a high
stopband attenuation. Thus, high filter orders and a significant amount of DSP resources may
be required to implement them. However, achieving a good selectivity in the digital domain
is still much easier compared to analogue filtering. For instance, their analogue counterparts
suffer from component tolerances, design size and cost challenges, and thermal effects such
as temperature drifts. Hence, the challenge of FIR filter is rather an implementation issue
than a fundamental limit of the algorithm.
In-Band Distortion is a severe problem for the purely digital implementation of the feedforward mitigation. It solely processes the blocker signals and subtracts them from the
received desired signal. As the original transmitted blocker x(t) is not known a priori at
119

the receiver, an estimate x(t) is filtered out of the received BB signal. However, the blocker
itself does not only cause out-of-band distortion, but suffers also from in-band distortion,
from the perspective of the blocker band. Thus, also the distortion estimates will differ from
the original distortions contained in the desired signal branch.
The intuitive solution to that problem is to somehow obtain an ideal reference signal
x(t) x(t) and to generate the distortion estimates by using that signal. Finally, significant
better mitigation performance and convergence characteristics can be expected. Several
techniques are reasonable. For instance, the blocker waveform can be re-modulated after
prior demodulation. For this, exact knowledge of the blocker waveform including its modulation, coding etc. or at least certain signal sequences need to be known. Likewise, a
non-distorted version of the blocker can be gathered from a second linear RF path that employs higher attenuation to keep within the linear range. Also, digitisation requirements are
relaxed as a reduced resolution auxiliary ADC may be sufficient to obtain a clean blocker
signal. Similarly, an AF length can be increased to alleviate the in-band distortion problem.
That is, the provided non-ideal distortion estimate is filtered until it matches the original
distortion. High AF orders can shape the reference signal until it matches the desired signal.
However, the other solution strategies are preferred compared to increasing the AF length,
as it can result in high unnecessary filter lengths and unpredictable behaviour concerning
the convergence characteristics and the step size setting.
A/D Interface is a significant bottleneck of the whole digital feedforward mitigation algorithm as it should provide a digitised version of the total BB signal including the strong
distortion-producing blockers. If the distortion stems from a blocker that is outside the digitisation bandwidth, no reference signal is available for distortion regeneration. Add to this,
there is the dynamic range problem when digitising a BB signal comprising several weak
and strong carriers. In brief, there is a huge burden on the A/D interface of the receiver
front-end concerning the ADC resolution and bandwidth. These requirements also depend
on the selectivity employed at the RF input, e.g. by means of a preselection bandpass, and
the actual target system specifications.
To overcome that problem, a large frequency bandwidth should by digitised as a whole by a
single ADC or piecewise by parallel ADCs, each digitising a subset of the total bandwidth.
However, the first approach is quite unrealistic with todays ADC technologies, especially
due to the high power consumption [VSHGA+06]. However, a single wideband and high
resolution ADC will likely become feasible in future, especially due to the fast growing
digital circuit technology. In [KH08b; ZMS09], mixed-signal/hybrid techniques have been
120

proposed, that are discussed in Section 3.2. A clever solution is presented in [KH08b], where
all possible distortion, that appears within the full receiver RF bandwidth, is included and
only those distortion components falling into the desired bandwidth are further processed in
the digital domain. However, beside the general benefits of the digital feedforward mitigation
discussed in Section 3.2, no additional mismatch between the desired and reference signal is
generated, compared to the mitigation architecture used in [KH08b]. Furthermore, a purely
digital implementation also enables runtime reconfigurability of the mitigation algorithm,
such as its FIR filter parameters or the AF step size.
Clipping Distortion (CD) cannot be mitigated by the presented algorithm due to the
different nature of the distortion process. The feedforward algorithm at hand can only mitigate mild non-linear distortion where the distortion-producing blocker can be still recovered.
Clipping causes severe distortion falling into the entire BB. Handling CD would require
a non-clipped input signal, as discussed in Section 3.2. Such algorithms typically require
hardware modifications of the RF front-end, e.g. a second RF path, and are out of scope in
this thesis. Therefore, many practical RF measurements employ modulated signals with a
reduced PAPR to strictly avoid CDs.
Multiple Blockers may be simultaneously present at the RF input, thus, parallelism of

the digital feedforward architecture should be possible to exploit. That is, one blocker is
processed by five parallel branches when employing the total cascaded model. Another
blocker would require a further bunch of five branches that is arranged in parallel to the
others. However, several blockers and their distortion products should not coincide to avoid
the aforementioned time-domain correlation of the distortion branches. Otherwise, it results
in severe convergence problems and difficult setting of the step sizes for the AFs. According
to Figure 4.16, there should be a minimum frequency separation of strong blockers of at least
the sum of the two blockers bandwidth B1 +B2 assuming dominant third-order non-linearity.
Cross-modulation effects are also neglected throughout the thesis and considered to be
low, due to their dependence on the relative strength of the underlying signals [PC03;
VSHGA+06]. The presented feedforward algorithm focuses on mitigation of the dominant
IMD products.
121
3B1
3B2
B1
B2
IMD
f
B1 + B2
Figure 4.16: Minimum frequency separation for multiple blockers.
4.6 Inclusion of Memory Effects

As already mentioned in Section 2.5.7, the digital feedforward mitigation represents an adaptive Hammerstein model, comprising of a static reference non-linearity model followed by
an adaptive FIR filter. In simulations, single-tap AFs are typically sufficient, due to the
perfect match of the artificial model, mimicking the non-linear front-end, and the reference
model, as well as due to the absence of memory. In measurements, at least two taps are
necessary to obtain similar mitigation performance. In Section 2.7.3, it has been proven
that the SDR under test, the USRP+WBX, manifests little memory. Hence, the AF stage
can accomplish for linear memory effects by adding more taps. In that sense, the digital
feedforward mitigation is able to cope with frequency-dependent characteristics in the RF
front-end.
4.7 Summary of Digital Feedforward Mitigation

This chapter has presented the selected mitigation technique in detail. Starting from the basic concept of regenerating distortions and their adaptive subtraction in the digital domain,
all necessary building blocks have been examined. In particular, a parallel mitigation architecture has been developed to mitigate cascaded non-linearities being present in DCRs with
non-linear components at RF and BB. It has been shown that all generated distortion products can be suppressed significantly. The overall mitigation gain achieved with the proposed
architecture outperforms state-of-the-art techniques. It is noteworthy that linear memory
122
4.7 Summary of Digital Feedforward Mitigation

effects can be considered due to the adaptive Hammerstein structure, composed of a static
reference non-linearity model and the adaptive filtering stage. The effectiveness of this approach has been verified through extensive simulations and RF measurements with two-tone
and modulated BPSK signals. Real-world measurements with the USRP confirm that RF
and BB distortions co-exist. Finally, the proposed mitigation architecture enables a total
clean-up of the BB signal. The chapter has been finished with an overview of fundamental
(non-technological) limits including some solution ideas.
123

Up to now, the efficiency of the digital feedforward mitigation has been successfully verified in AWGN channels through simulations and RF laboratory measurements with cable
connections. In principle, nothing changes in the mitigation architecture if the input signal
level is varying due to a realistic radio channel manifesting all wave propagation phenomena,
namely reflection, diffraction, scattering, and shadowing. In general, any fluctuation in the
input signal level, also known as fading, is seen in the reference signal. However, the AF
stage needs to adapt to these changes quickly. It is likely that there are bad channel conditions where the convergence characteristics and the mitigation performance of the algorithm
are affected. Typically, an AGC is in charge of relaxing the fading effects and to provide a
stable power level to the subsequent receiver stages. However, weak desired signals may get
cut, while the AGC reduces the overall signal power depending on the blocker power level
in order to prevent clipping. Employing digital feedforward mitigation has the potential to
maintain and even increase the dynamic range for weak carriers while non-linear distortion
effects are also alleviated. Finally, it is worth studying the impact of different fading phenomena on the level of the distortion products and the mitigation processing, in order to
prove functionality of the algorithm in real-world scenarios, where an MS moves through a
certain radio environment.
This problem is not considered in detail in the state of the art. Prior works in [VSHGA+06;
AMV10b] just mention the role of the communication channel and suggest exploring dynamic
channel effects in more detail. Time-varying waveforms occur due to channel conditions, MS
movement, and user allocations. [KH08b] mentions the problem of the AF convergence time
and concludes that the convergence should be faster than the minimum channel coherence
time of a frame/block of the underlying standard of the desired signal.
Therefore, the influences of fading conditions and AGC on the distortion power levels and
mitigation processing are investigated in more detail in the following sections. For this
purpose, system-level simulations have been conducted including radio channel, receiver
front-end simulation and BB processing. This work has been carried out within a master
thesis [Dup13], where the interested reader is referred to for details. In addition, the results
125

have been published in [DGS+13]. This chapter will provide a brief overview about the
methodology and the key findings.
5.1 Methodology
A typical scenario of interest is depicted in Figure 5.1, where an MS receives two adjacent
FDMA channels belonging to different BTSs, a desired signal to be demodulated and a
blocker signal. It is likely that there are conditions where the blocker is received with higher
BTS1:
Desired Channel
MS
BTS1
Non-linear
Behaviour
BTS2:
Blocker Channel
BTS2
Figure 5.1: Typical fading scenario with two adjacent FDMA channels from different BTSs
[Dup13].
power than the desired one, e.g. due to relative distances or unequal channel fading conditions caused by movement of the MS. Then, the blocker easily causes non-linear distortion
that may fall on top of the adjacent weak desired signal, as sketched in Figure 5.1.
Such a scenario practically appears in GSM for railways (GSM-R) at the transition band between the public GSM extended band (GSM-E) and the GSM-R band at GSM 900 downlink
frequencies. Details are given in Section 7.2.2. Following that practical use case, a signal
configuration with a weak desired GSM carrier and a strong adjacent BPSK blocker has
been assumed for the simulations with the parameters summarised in Table 5.1. The exact
geometrical layout of the scenario under consideration is sketched in Figure 5.2. Assuming
126
5.1 Methodology
Table 5.1: Detailed characteristics of the simulated signals [Dup13].
Parameter
Desired
Blocker
Modulation
GSM
BPSK
Symbol rate
256 kbps
2 Mbps
Bandwidth
200 kHz
2.7 MHz
0.35
Carrier frequency
0 Hz
1.55 MHz
PAPR
0 dB
5 dB
SNR
15 dB
67 dB
Roll-off factor
BTS Desired
20
MS
20
0m
20
0m
00
BTS Blocker
Figure 5.2: Geometrical layout of the fading scenario [Dup13].

Cartesian coordinates with the origin in the centre of the MS, the locations of the desired
and blocker BTSs are (2000, 200) m and (0, 200) m, respectively. The MS is moving in
direction x and thereby experiences different channel conditions and receives unbalanced
power levels from both BTSs. Omnidirectional antennas have been adopted for all nodes as
spatial effects are not of interest in the evaluation.
For the radio channel simulation, the wireless world initiative for new radio (WINNER)
channel model [BESJ+05] is used.
A tapped-delay line model mimics the channel in the simulator, based on coefficients extracted out of WINNER. There are different modelling techniques that can be classified into deterministic, stochastic, and
geometry-based stochastic channel models (GSCMs). The deterministic model bases on
Maxwells equations or ray tracing models, which mimic reflection, scattering, and diffraction
based on information of the shape and electromagnetic properties of surrounding objects.
In stochastic models, the received power at different delays, Doppler shifts, and angles of
arrival follow distributions derived from measurements. It provides a random number of
multipath components with a random amplitude, delay, and phase. The GSCM technique
127

is adopted in WINNER, that randomly places scatterers and sums up all signal contributions of each scatterer. The GSCM technique is less expensive in terms of computation
than conventional ray tracing and is widely accepted in industry. The model outputs the
channel impulse response (CIR) at the desired sampling frequency.
WINNER offers a broad variety of propagation scenarios. The three scenarios summarised
in Table 5.2 have been selected for studying the channel effects. These scenarios are mainly
Table 5.2: Selected WINNER propagation scenarios [Dup13].
Scenario
C3: Bad urban macro-cell
LOS/NLOS
NLOS
Speed
(km/h)
20
60
Characteristic
Long delays, high
frequency-selectivity
20
D1: Rural macro-cell
NLOS
60
Frequency-flat
120
20
D2a: Rural moving networks
LOS
60
120
Large Doppler
variability,
frequency-selective
300
distinguished by their line of sight (LOS) or non-line of sight (NLOS) characteristic, i.e. if
there is dominant multipath propagation or not. Scenario C3 causes high time variation
and is assumed to stress the algorithm due to rapid changing regrowth from snapshot to
snapshot. Scenario D1 possess a low selectivity in frequency and low time variation and is
likely to work without trouble. It constitutes a counterpart to the other scenarios. Large
Doppler variability and high frequency-selectivity is caused in scenario D2a for representing
moving networks, that should stress the convergence time of the AF stages significantly.
The DCR front-end is modelled by a static memoryless polynomial for modelling RF nonlinearity that is assumed to be dominating. Memory effects are not considered in this
simulation, hence, single-tap AFs are sufficient. First, the impact of the channel on the
spectral regrowth of the blocker is studied. Then, the convergence characteristics of the
AF stage is investigated under varying blocker power conditions from snapshot to snapshot.
Finally, the performance of the algorithm is verified and reasons for failures are given.
128
5.2 Impact of Fading and AGC

In general, the typical wave propagation phenomena lead to multipath (NLOS) propagation,
Power
where different delayed versions of the received signals superimpose. In addition, there is a
frequency-dependent behaviour of the channel effects. Hence, multiple types of fading are
distinguished, that are sketched in Figure 5.3. Large-scale fading denotes the path loss due
Path loss
Path loss +
Shadowing
Path loss +
Shadowing +
Fast fading
Distance
Figure 5.3: Qualitative behaviour of different fading phenomena [Dup13].
to the relative distance and shadowing of large objects like buildings. The MS moves through
a large distance in relation to the BTS. Small-scale fading comprises fast variations on the
amplitude of the received signal, mainly due to constructive and destructive superposition
of multipath components. Here, the MS moves in small distances. Furthermore, frequencyflat and frequency-selective fading is distinguished, depending on the number of resolvable
delays. In that context, the coherence bandwidth denotes the width of frequencies of the
signal that are similarly affected by the channel. Finally, also Doppler spread occurs due to
the movement of the MS. Depending on the speed, the Doppler spread causes slow or fast
fading. In particular, in fast fading, the coherence time of the channel is smaller than the
symbol period and causes a broadening of the original signal in time domain.
Due to these fading conditions, the non-linear distortion will vary from snapshot to snapshot
in a non-linear manner. Also, signals from different BTSs experience different channel effects
and sum up in the receiver. Thereby, highly unbalanced signal configurations with weak and
strong signals may occur, especially in high frequency-selective channels, where signals can
be even temporarily cancelled. Finally, the algorithm has to track the changes in these signals
introduced by the channel characteristics, which requires a certain convergence time of the
129

AF stage. However, large scale fading is omitted from the analysis as it is obviously not a
challenge for the algorithm to track these slow variations. Thus, the impact of small-scale
fading is on focus in the simulations.
Reducing the channel effects and smoothing deep fades is typically the task of an AGC.
It adapts the input power level to predefined levels to keep the components within their
linear region, by comparing the average signal power to a reference power and adjusting
a voltage-controlled amplifier at the AGC loop front. There are feedback and feedforward
implementations of AGCs similar to the linearisation techniques discussed in Section 3.1.1.
Basically, the feedback implementation is preferred due to its closed-loop regulation, although
there is a longer delay introduced due to the decision. Receivers typically employ a stepped
AGC at RF and a second AGC at IF or BB. A general drawback of the AGC is that the whole
signal gets attenuated with regard to the high power signals. Hence, the overall dynamic
range is reduced and weak signals may get cut, and thus covered by noise.
5.2.1 Simulation Setup

The simulation setup is depicted in Figure 5.4. The channel changes every snapshot having a
GSM
Channel 1
BPSK
Channel 2
Tx signal
WINNER
AGSM
KAGC
+
ABPSK
SNR
Adjustment
BERlinear
GSMlinear
LP
BERnon-linear
GSMnon-linear
LP
BERNONLIM
GSMNONLIM
LP
Interferer
Adjustment
AGC
+
a1 , a3
Demodulator
a
1 , a
3
NONLIM
Digital domain
AWGN
Noise
Artificial Ref
Model
Analogue domain
Figure 5.4: Block diagram of the fading simulation setup [Dup13].

length of a single GSM burst. Then, both blocker and desired GSM carrier are convolved with
the individual CIRs (). For this purpose, the channel coefficients are normalised to eliminate
the influence of path loss and shadowing. Referring to Table 5.1, the SNR of the GSM is
kept fixed at 15 dB to avoid bit errors due to channel effects and noise. Bit errors should only
130

occur due to non-linearly induced interference. The power of the BPSK is set 52 dB higher
than that of the GSM. The AGC averages the power over 75 samples which corresponds to
1 bit of a GSM burst (oversampling rate of 75). Then, the composite signal is fed through
the artificial reference model with a1 = 1, a3 = 200 (equals to an IIP3 = 7 dBm) and an
AWGN source that jointly mimic the non-linear DCR in the simulator. Before demodulation,
the signal runs through a lowpass (LP) filter. The burst-wise BER analysis of the desired
GSM carrier is conducted with the GSMSim simulation framework [EM97], that is also used
and detailed in Section 7.2.2. In brief, it is an easy to use implementation of a transmitter and
receiver front-end of a standard compatible GSM system. In total, 200 channel realisations
are simulated, with 100 GSM frames per realisation.
5.2.2 Effects of Fading on the Non-linearities

Frequency-selective and fast fading conditions are most crucial operating conditions for
the mitigation algorithm. In a frequency-selective channel, both blocker and desired
signal are shaped from snapshot to snapshot in a different way, thereby changing the
signal-to-interference ratio (SIR), here being defined as the ratio between the desired signal power level and the level of spectral regrowth of the blocker falling into the desired
channel. The speed of the fast fading will be important for the convergence time of the
algorithm as it determines the SIR variation rate. The algorithm should follow and mitigate
these variations as fast as possible while keeping stability and providing sufficient mitigation
gain.
Figure 5.5 illustrates an example of two consecutive snapshots that experience very different
frequency responses due to a heavily frequency-selective channel. The wider the bandwidth
of the blocker, the more harmful is its regrowth. From Figure 5.5(a) it is clearly visible that
the weak GSM carrier is completely covered by the spectral regrowth induced by the BPSK
blocker, resulting in a high BER. In addition, the channel shapes the blocker spectrum in a
non-symmetric way, resulting in different levels for the upper and lower IMD.
Next, the statistics of the spectral regrowth are analysed as part of non-linear distortion
is falling on top of the desired band and that is being processed by NONLIM. Especially
the difference from snapshot to snapshot is expected to stress the convergence of the AF
stages. In [Dup13], it has been shown that the spectral regrowth follows a generalized
extreme value distribution, whereas the Spectral_Regrowth is fitted best by a t locationscale distribution. By taking the desired signal power and the amount of spectral regrowth,
the SIR and the corresponding SIR from snapshot to snapshot are obtained. Figure 5.6
131

replacemen
20
PSD (dBm/Hz)
30
Linear
Non-linear
40
50
SIR
60
Spectral Regrowth
80
GSM Channel
70
90
BPSK Blocker
100
110
10
20
PSD (dBm/Hz)
30
4 2
0
2
4
(a) Snapshot n.
10
10
Linear
Non-linear
40
50
60
70
80
90
100
110
10
4 2
0
4
2
(b) Snapshot n + 1.
Figure 5.5: Impact of frequency-selective channel on SIR [Dup13].

illustrates the distribution of SIR for the three different scenarios for a speed of 60 km/h.
D1 shows only little variance of the SIR and looks almost impulse-like, mainly because of
the low frequency-selectivity of the channel. On the contrary, D2a and C3 cause the high
variance due to the frequency-selectivity. For D2a, this variance will increase with the speed
of the MS.
5.2.3 Mitigation Performance under Fading Conditions

Now, the performance of the feedforward mitigation is evaluated by means of the SIR and
BER improvement. Figure 5.7 shows the mitigation gain for the scenario C3, where each
point represents one snapshot. In total, 20 000 snapshots have been simulated.
132
Basically,
0.12
0.1
C3
D1
D2a
PDF
0.08
0.06
0.04
0.02
0
60
40
20
0
SIR (dB)
20
40
60
SIR After Mitigation (dB)
Figure 5.6: PDF of SIR in different fading scenarios [Dup13].

70
60
50
Highest Mitigation
40
30
Bad Mitigation/
20
35 dB
Mitigation Error
10
0
No Mitigation
10
20
30
30 20 10
0
20
30
40
50
60
10
SIR Before Mitigation (dB)
70
SINR After Mitigation (dB)
Figure 5.7: SIR enhancement analysis for the scenario C3 at 60 km/h [Dup13].
20
SNR = 15 dB
15
10
5
0
5
10
15
20
25
30
30 25 20 15 10 5
0
5
SINR Before Mitigation (dB)
10
15
20
Figure 5.8: SINR enhancement analysis for the scenario C3 at 60 km/h [Dup13].
133

the mitigation gain is not constant and varies from snapshot to snapshot. The variation is
mainly caused by the interference power, i.e. the amount of spectral regrowth in the desired
band, as the desired signal power has been kept constant. The highest mitigation gain is
approx. 35 dB with regard to the noise floor as the algorithm will not mitigate noise. There
are cases without improvement of the SIR (no mitigation) due to convergence problems of
the AF. Bad mitigation occurs for high SIR conditions, i.e. the power of the interference is
even higher after applying the algorithm. Here, the algorithm introduces noise to the desired
signal output, thereby increasing the level of interference. In such a case, the AF cannot
find the correlating reference distortions in the desired signal branch, as the blocker has been
received without distortions. Thus, the AF will not converge and adds mitigation noise to
the output. To overcome this problem, the algorithm should be bypassed under high SIR
conditions, when there is no need for mitigation in general.
The interference mitigation results also in an SINR improvement as illustrated in Figure 5.8.
The upper limit is determined by the SNR = 15 dB. Hence, the ideal mitigation is denoted by
a straight line at 15 dB, however, the algorithm does not always fully cancel the interference,
which causes some variance.
Finally, the resulting BER after applying the mitigation algorithm is analysed. Three main
performance cases can be distinguished
(i) Mitigation BERNONLIM < BERnon-linear ,
(ii) No need for/ no mitigation BERNONLIM = BERnon-linear = BERlinear or BERNONLIM =
BERnon-linear > BERlinear ,
(iii) Bad mitigation BERNONLIM > BERnon-linear .
Following Figure 5.6, scenario D1 is likely to provide the highest mitigation performance
due to the low variation of the SIR. On the other hand, C3 and D2a often do not require
mitigation due to the high SIR variation.
A figure of merit, the bit error mitigation performance (BEMP) is introduced with
BEMP = 1
ENONLIM Elinear
,
Enon-linear Elinear
(5.1)
where E denotes the amount of errors per burst in the corresponding cases indicated by the
subscripts. Note that there is a certain minimum error Elinear due to the noise and fading,
the mitigation algorithm is not supposed to mitigate. A BEMP value near 1 indicates the
best performance, whereas BEMP 1 denotes a bad performance. From Figure 5.9 it
can be concluded that in 90 % of all simulated snapshots, the BEMP is higher than 90 %,
134
BEMP
1
0.9
0.8
0.7
0.6
0.5
0.4
0.3
0.2
0.1
0
C3 w/o AGC
D1 w/o AGC
D2a w/o AGC
C3 w/ AGC
D1 w/ AGC
D2a w/ AGC
0
10
20
30
40
50
60
% Realisations
70
80
90
100
Figure 5.9: BEMP analysis for different scenarios at 60 km/h [Dup13].

independently of the scenario. When having the AGC enabled, the algorithm operates at
high SIR regime and the AF coefficients converge more slowly. Finally, in only 75 % of the
realisations the BEMP is higher than 90 %. In [Dup13] it is shown that the bad mitigation
cases occur due to convergence problems caused by high Spectral_Regrowth and where
the algorithm is working in high SIR regime.
More interestingly is the actual BER enhancement evaluated on a burst-based analysis. For
that purpose, the SINRs have been extracted from the simulations, re-ordered, and displayed
with their corresponding BER. Finally, the mean BER per burst is computed, even though,
there is not an equal number of realisations per SINR value available. Figure 5.10 shows
a significant improvement of the BER performance independently from the scenario for
SINR < 0. The lowest mitigation is achieved for C3 in the high SINR regime. Figure 5.11
100
BER
101
102
103
104
25
C3 Before Mitigation
D1 Before Mitigation
D2a Before Mitigation
C3 After Mitigation
D1 After Mitigation
D2a After Mitigation
20
15
10
5
0
SINR (dB)
10
15
Figure 5.10: BER vs. SINR at different scenarios [Dup13].
135

illustrates the performance with AGC inclusion at the example of scenario C3. If the AGC
100
Gain w/ AGC
BER
101
Gain w/o AGC
102
103
104
25
C3
C3
C3
C3
20
Before w/ AGC
Before w/o AGC
After w/ AGC
After w/o AGC
15
10
5
0
SINR (dB)
10
15
Figure 5.11: BER vs. SINR for scenario C3 with and without AGC [Dup13].
is enabled, the mitigation performance is reduced significantly. Although the AGC provides
approx. 10 dB of SINR gain, the BER is higher compared to the case without AGC. This is
due to the attenuation of the desired signal, which is affected significantly by noise.
To sum up, the radio channel causes a certain shaping of the blocker signal according to the
characteristics of the scenario. In the statistical analysis of the behaviour of the NONLIM
algorithm under random channels, it has been shown that there is not a significant performance degradation among the scenarios, although there is a dependence on the speed
[Dup13]. In brief, the algorithm is applicable in receivers that operate in a real-world radio environment. However, the best performance is achieved in scenario D1 due to the less
frequency-selective, low-variance interference. In addition, it has been found that the AGC
reduces the overall system performance, although it prevents IMD generation. By disabling
or relaxing the AGC under the presence of high-power signals/blockers, the dynamic range
can be improved with the help of the feedforward algorithm. Furthermore, the algorithm
should be bypassed under high SIR conditions.
The bypassing needs to be switched according to the interference power at hand. For this
purpose, the input power can be gathered from the desired branch to estimate the significance
of non-linear distortion. Another solution can be to perform a cross-correlation between the
desired and reference branch to determine the amount of non-linear distortion present. Then,
the algorithm can be enabled or disabled according to a threshold.
136
5.3 Summary about the Impact of the Radio Channel
5.3 Summary about the Impact of the Radio Channel

This chapter has discussed an important practical issue when aiming for integration of the
digital feedforward mitigation into real devices. It has been confirmed that the algorithm
can be applied even under frequency-selective and fast fading conditions imposed by wave
propagation in a real radio environment. In particular, it has been shown that the BER
of distorted signals, as well as the energy of non-linearly induced interference decrease significantly after applying the algorithm. However, bypassing the algorithm under high SIR
conditions seems to be inevitable to overcome convergence difficulties and the addition of
noise. Eventually, the dynamic range and effective linearity of the RF front-end can be
increased solely by the feedforward techniques with disabled or relaxed AGC.
137
In the prior chapters, the feedforward mitigation capabilities have been demonstrated based
upon offline post-processing of simulated and measured data in MATLAB. Practical aspects
of embedding the DSP-based algorithm into the devices digital back-end are rarely discussed
in the state of the art. Although the complexity of the algorithm has been analysed on a
high level in Section 4.3, it is desirable to build an FPGA prototype of the purely digital
feedforward mitigation to obtain further insights into practical challenges. An FPGA is well
suited for this application due to its rapid prototyping.
Embedding the algorithm into the device offers tremendous advantages. The receiver selfregulates the induced non-linear distortion and thereby improves its effective linearity. In
addition, the host PC is offloaded and can solely perform the specific BB processing since
the digital back-end transfers the corrected I/Q data directly to the host.
There is no implementation of the purely digital feedforward mitigation algorithm available
in the state of the art. The only prior implementation of the feedforward mitigation algorithm
is reported in [KH08b], where the distortion regeneration takes place at RF level. This kind
of mixed-signal implementation, that has been discussed in Section 3.2, requires a custom
hardware design with considerable effort. General implementation of an LMS AF on FPGAs
can be found in [MB04]. An optimisation of the LMS tap-weight update by simple bit-shift
operations is proposed in [OKA06].
In this chapter, a brief overview of a purely digital implementation on the FPGA of the
USRP N210 [Ett] is given. This work has been carried out within a master thesis [Sch13],
to which the interested reader is referred for more details. In addition, the results have been
published in [SGT13].
6.1 Target Architecture and Basic Scenario

The design is implemented on the FPGA Xilinx Spartan-3A DSP (XC3SD3400A) of
the USRP N210 motherboard [Ett; Xilb].
139
An FPGA is an integrated circuit that
holds programmable logic.
For implementing digital circuits on an FPGA, a high-
level description of the circuit is mapped to the FPGA architecture with the help of
hardware description languages (HDLs) (very high speed integrated circuit HDL and Verilog) and software tools for electronic design automation (EDA) [Sch13]. FPGAs are very
popular due to their flexibility and very high speed supporting parallelism of algorithms. The
digital circuit is implemented based on configurable logic blocks (CLBs), which comprise
look-up tables (LUTs), flip-flops (FFs), multiplexers (MUXs), and carry/arithmetic logic.
Beside this general FPGA fabric, the target architecture offers dedicated memory (Block
random-access memory (RAM)) and DSP48A slices that are well suited for DSP applications [Sch13]. A DSP48A slice is composed of a pre-adder, a signed 18 18 bit multiplier,
and a post-adder. These slices can implement multiply-accumulate (MAC) architectures
such as FIR filters, where a multiplication is followed by an addition.
The number representation is a challenge for implementing DSP-based algorithms on an
FPGA. A fixed-point or floating-point arithmetic with a certain bit width can be implemented. However, a fixed-point implementation is more common due to its increased efficiency. Nevertheless, the designer has to deal with bit growth, overflow, and scaling/
rounding methods. For example, if two numbers are added, the word length needs to be
increased by 1 bit in order to avoid potential overflow. Likewise, multiplying N and P wide
numbers result in a bit width of N + P for maintaining full precision. Overflows need to be
strictly avoided, as the number can wrap around from positive to negative or vice versa and
may cause abrupt jumps in the waveform. Consequently, the result of any arithmetic operation needs to be rounded and truncated. Beside other techniques, non-symmetric rounding
to positive is applied for this design, that adds a binary 1 to the rightmost bit position after
truncation and right-shifting by k bit positions (scaling by 2k ) [Sch13].
Basic Scenario and Simplifications

The basic scenario and the block diagram for the real-time implementation is illustrated in
Figure 6.1, similar to Figure 4.1. In the sketched spectrum ,
1 a simple two-tone blocker
is assumed, whose IMD products may hit an adjacent target frequency band. Hence, a
bandstop and a bandpass filter form a bandsplit filtering stage assuming fixed filter characteristics. Thereby, the runtime reconfigurability of the filters on the FPGA is omitted
for the sake of simplicity. As a further simplification, only a third-order RF non-linearity
model is implemented as the RF non-linearity, or the distortion in the fundamental zone,
is assumed to be dominating. Further distortion at BB is not taken into account to keep
the model as simple as possible. Moreover, a separated I/Q processing of the distortion
140
xQ (t)
xI (t)
ADC
ADC
xQ (n)
xI (n)
wBP (n)
bpr_filter
wBS (n)
bsd_filter
Target Spectrum
Strong Blockers
IMD products
141
uQ (n)
uI (n)
dQ (n)
dI (n)
refmodel
x
2 x
, x C
u3I + uI u2Q
u3Q + u2I uQ
d_delay
delay
wBS (n)
bsr_filter
rQ (n)
rI (n)
r_delay
delay
lms_filter
L=1
w0,Q
w0,I
eQ (n)
eI (n)
x
Q (n)
x
I (n)
6.1 Target Architecture and Basic Scenario
Figure 6.1: Block diagram of the NONLIM FPGA implementation by means of a two-tone
example [Sch13].
estimates is performed in the reference branch, that shows already sufficient mitigation in
practice [GSH+12a].
The architecture is composed of the FPGA module instances bandstop-desired filter
(bsd_filter ), bandpass-reference filter (bpr_filter ), bandstop-reference filter (bsr_filter ), reference model (refmodel), and the LMS AF (lms_filter). The band-split filter separates
the blocker and the useful signal to
2 and ,
3 whereas a complex implementation for the
bpr_filter is chosen due to the mirror-frequency interference being present in real-world measured signals (cp. Section 4.4). The reference model computes the RF non-linearity term
x2 x according to (2.22) and outputs a signal such as .
4 For separated I/Q processing, this
term is decomposed into (2.23) and (2.24). The exact coefficient 3a2 in (2.22) is omitted
without affecting the performance of the algorithm, as the AF stage will find the correct
value any way. Next, the distortion estimates are fed through a real-valued bandstop filter
(bsr_filter ) to suppress the blocker and to obtain only the distortion. An equivalent filter is employed in the desired branch, which has an inverse characteristic to the bpr_filter.
Then, the output signal
5 enters the AF stage that is depicted with a filter order M = 0
(L = M + 1 = 1). That is, only scaling of the estimates is conducted in the ideal case having
a perfect match of the reference non-linearity and the real-world non-linear behaviour of the
front-end. Here, a real-valued LMS implementation is chosen for reasons of simplification.
A complex-valued LMS implementation might outperform the real-valued one due to the
capability to mitigate AM/PM distortion and I/Q imbalance effects that are present in realworld signals. Alternatively, the AF length could be increased to compensate for the missing
cross-products between I and Q. Finally, as depicted in ,
6 the blocker may be added back
in order to satisfy a spectral sensing use case (Section 4.1).
There are certain processing delays introduced by NONLIM that need to be compensated by
delay blocks (d_delay and r_delay). There is a pure processing delay of the processing chain
as well as the settling time of the FIR filters (length of M/2). The signals of the desired
and reference branch need to be aligned to provide the AF with corresponding samples.
The delay block d_delay accounts for the relative delay between the desired and reference
branch. Another delay block r_delay compensates for the absolute delay of the reference
branch to synchronously add the blocker to the AF error signal.
6.2 Modules Implementation

Figure 6.2 illustrates the existing direct-down conversion (DDC) chain of the USRP with the
integrated NONLIM FPGA module. Basically, even in a modern SDR, all the high-speed
142

rdecim = 8
fs = 12.5 MHz
fs,ADC = 100 MHz

ADC
CORDIC
CIC
Decimator
Small
Halfband
Decimator
Halfband
Decimator
fine downconversion
decimation
decimation
decimation
ADC
Existing DDC Chain
Delay
bsd_filter
bpr_filter
d_delay
u3I + uI u2Q
u3Q + u2I uQ
bsr_filter
refmodel
Custom NONLIM Module
Ethernet
Interface
VITA
Protocol
lms_filter
Figure 6.2: Existing DDC chain with the integrated NONLIM module [Sch13].
processing tasks such as decimation, filtering, and interpolation still remain on the FPGA.
The main task of the existing DDC chain is to perform frequency conversion to zero-IF
and decimation, such that the samples can get transferred to the PC. The ADC clock with
100 MHz is much too high to directly stream the data to a PC in real-time, i.e. a decimation
by at least 4 needs to be performed in order to transfer them via gigabit ethernet (GigE)
[Ett; Sch13].
The coordinate rotation digital computer (CORDIC) denotes an iterative algorithm performing approximated computations of trigonometric functions to form a NCO that conducts the final down-conversion. The cascaded integrator comb (CIC) is a hardware-efficient
implementation for decimation and interpolation filters by using only additions and subtractions. However, the CIC causes aliasing components and a roll-off in the passband that
makes subsequent filtering essential. In order to compensate for the CIC roll-off, a two-stage
halfband filter is employed with symmetric FIRs with every second coefficient being zero.
This allows for using the hardware multipliers in a time-multiplexed manner. In order to
efficiently compensate for the CIC roll-off, a decimation rate as a multiple of 4 should be
chosen [Sch13]. Only in the case this constraint is met, both halfband filters are activated.
To relax the requirements of the real-time processing of the samples in the custom NONLIM
module, a decimation rate of rdecim = 8 is used. Hence, the custom module is placed after
the decimation as indicated in Figure 6.2. This allows for a processing time of 8 clock cycles
until the next sample appears at the input. Also, BB distortion is likely to be filtered out if
143
the blocker IF frequency is chosen to be slightly greater than fs /6 (cp. Section 4.2). Thus,
distortion in the fundamental zone are on focus in this implementation.
The HDL Verilog has been used for the implementation, as the existing USRP FPGA
firmware is already written in Verilog [Ett]. To verify the correct functionality of each submodule, a top-down design flow is adopted [Sch13]. Starting from a fixed-point high-level
simulation in MATLAB, an HDL-MATLAB co-simulation is performed by using a custom
file interface and the ISim behavioural simulator that is included in the Xilinx EDA tool
ISE. Finally, the modules are validated running on real hardware. The following subsections
provide further details of the implementation of the main sub-modules.
FIR Filter
There are three instances of FIR filters, namely bsd_filter, bpr_filter and bsr_filter, as
summarised in Table 6.1. These are designed with the given characteristics using the HDLTable 6.1: FIR filter implementation for a two-tone blocker centred at fIF = 3 MHz [Sch13].
Filter instance
BW
fcutoff1
fcutoff2
bsd_filter
bpr_filter
640 kHz
320 kHz
600 kHz
bsr_filter
Astop /Apass1/2
Filter order M
85/0.2 dB
161 (Mbsd )
40/0.2 dB
95 (Mbpr )
95 (Mbsr )
MATLAB interface and the Parks-McClellan algorithm (Section 4.2). The implementation
is established using the intellectual property core generator of the Xilinx EDA tool, that
efficiently map a pre-defined module with a specific function to the target device.
Basically, there are two main possible implementation structures: MAC (symmetric systolic
FIR) and distributed arithmetic (DA). The MAC structure uses multiplications and additions, as illustrated in Figure 6.3, that are mainly implemented by concatenated DSP48A
slices. This type of implementation is used for the bsd_filter and bsr_filter. On the contrary, the DA approach substitutes the multiplications by bit shifts, additions, subtractions,
and LUTs. It is used for the bpr_filter to save DSP48A slices at the expense of general
FPGA fabric (CLBs). A complex-valued implementation has been chosen for the bpr_filter.
Although resulting in twice as much resources, a non-symmetric filtering characteristic at
both positive and negative frequencies cannot be tolerated with focus to the mitigation performance. Basically, a very pure reference signal is desirable for the reference branch to
144

x
18
w0
18
0
DSP48A Slice
w1
w2
w3
y
DSP48A Slice
DSP48A Slice
DSP48A Slice
37
Figure 6.3: Block diagram of a symmetric systolic FIR implementation [Sch13].

make the AF stage work efficiently. Symmetric (real-valued) filtering is, on the other hand,
acceptable for other bandstop filters with regard to the more efficient resource utilisation.
A general challenge of FIR filter implementation is the quantisation of the filter coefficients
using a fixed-point number format. It results in small deviations in the frequency response
characteristics compared to their floating-point counterparts. In [Sch13], these deviations
have been analysed and reported to be negligible for the chosen number format and filter
characteristics.
Reference Model
The block diagram for implementing the reference non-linearity, illustrated in Figure 6.4,
exactly follows (2.23) and (2.24), where the initial value 3a2 is omitted to save another multiplier. Basically, the implementation performs the multiplications and additions sequentially.
In order to keep the results within the given bit width, a non-symmetric rounding to positive
and a subsequent bit truncation of the least significant bits (LSBs) is established. In total,
the implementation of the reference model requires 6 DSP48A slices.
LMS Filter
Regarding the LMS filter, a real-valued implementation is chosen that is decomposed into
an FIR and coefficient update part, as depicted in Figure 6.5. The FIR is implemented by
L parallel multipliers and is followed by a hard truncation of SF bits. Here, no rounding
is performed as it is expected that the AF stage accounts for the imposed bias. A similar
145
x
2I
x
I
16
round
and
truncate
32
x
3I + x
I x
2Q
x
3I
33
32
16
round
and
truncate 16
x
I x
2Q
32
x
Q
16
x
2Q
round
and
truncate
32
Q
2I x
x
3Q + x
x
3Q
33
32
16
round
and
truncate 16
x
2I x
Q
32
Figure 6.4: Block diagram of the pipelined implementation of refmodel [Sch13].

structure is adopted for the coefficient update part with a bit truncation factor MU. For
that purpose, the approach of [OKA06] is followed that approximates the multiplication with
the step size by a simple bit shift operation. In [Sch13], rules for choosing the truncation
factors SF and MU are given. They need to be adapted during runtime and may cause
overflow conditions in case they have been chosen to be too low. For example, both values
SF and MU need to be as small as possible and have to obey the rule SF + MU 26
[OKA06], in order to achieve a low residual error. For the scenario of interest, a setup with
L = 4 provided the best mitigation performance. That is, the DSP48A slices are used in a
parallel manner to perform the entire algorithm within 8 clock cycles.
On the other hand, a complex-valued LMS architecture can be used to account for I/Q
imbalance and AM/PM distortion. That is, lower AF order and therefore less resource
utilisation can be achieved. Table 6.2 compares the real- and complex-valued structure in
terms of their resource utilisation. Indeed, when setting the AF order L = 2, 4 DSP slices
Table 6.2: DSP48A Utilisation of the real- and complex-valued LMS implementation [Sch13].
L
Real-valued
Complex-valued
In general
2L + 2L = 4L
4L + 2L = 6L
12
12
18
16
24
146
Truncate
Factor
18
MU
w
18
32
16
r(n L 1)
32
16
r(n j 1)
32
r(n)
16
j = 0, . . . , L 1
18
r(n L 1)
r(n j 1)
16
16
34
34
Coefficient Update
y(n)
r(n)
16
34
34 + L
1
Truncate
Factor
16
SF
16
16
d(n)
j = 0, . . . , L 1
FIR Filter
e(n)
Figure 6.5: Implementation structure of the LMS filter [Sch13].

could be saved with a complex-valued implementation.
Delay Blocks
There are two instances of delay blocks, d_delay in the desired branch and r_delay for
adding back the blocker to the final AF error signal. Both are implemented by block RAM
following the first in first out principle.
Design Report
The resource utilisation of each FPGA module is detailed in Table 6.3, though fabric for
interconnection of the modules is not considered here. Table 6.4 summarises the amount of
used resources with and without the custom NONLIM module. Although there is a large
amount of resources available, timing violations turn out to be the challenging problem in
practical FPGA implementations when approaching a high resource utilisation of the total
available resources. Fortunately, all timing constraints have been met with the aforementioned design.
Another figure of merit is the introduced processing delay of the NONLIM module, which is
of high interest in time division multiple access (TDMA)-based communication systems like
147
Table 6.3: Resource utilisation of the individual sub-modules [Sch13].
Type\Module
bsd_filter bpr_filter
refmodel
bsr_filter
r/d_delay lms_filter
LUTs
1158
6796
160
697
562
FFs
1478
7294
352
883
878
DSP48A
22
13
16
Block RAM
1 (each)
Filter order
161
95
95
Architecture
MAC
DA
MAC
MAC
MAC
Table 6.4: Relative resource utilisation of stock and custom FPGA firmware [Sch13].
Type\Design
USRP N210 (stock)
NONLIM
USRP N210 + NONLIM
LUTs
31649/47744 (66 %)
9783 (20 %)
40578/47744 (85 %)
FFs
20007/47744 (41 %)
11446 (24 %)
31179/47744 (65 %)
DSP48A slices
31/126 (24 %)
57 (45 %)
88/126 (70 %)
Block RAM
41/126 (32 %)
2 (2 %)
43/126 (34 %)
GSM. The delay of the individual modules in FPGA clock cycles is detailed in Table 6.5.
The total delay accounts for the sum of all sub-modules in the reference branch plus the
Table 6.5: Delay of the invidual FPGA sub-modules [Sch13].
Module
Delay (cycles)
bsd_filter
33
bpr_filter
16
bsr_filter
24
lms_filter
refmodel
10
r_delay
49 8
16 8
d_delay
settling time of the FIR filters, delays by pipeline stages, and rounding operations. The
total delay amounts for 201 clock cycles, which corresponds to 2.01 s with a master clock
being fCLK = 100 MHz. Regarding this consideration, the convergence time of the AF is not
included, as it depends on the signal configuration and chosen step size at hand. Basically, the
processing delay of the FPGA prototype is likely to be negligible in modern communication
systems.
148
6.3 Experimental Results

Another challenge is to enable runtime reconfigurability of the algorithm for changing the
algorithms parameters, which depend on the respective signal scenario. The presented
FPGA prototype employs a rather static design. In practice, all filters need to be adjusted
to the centre frequency and bandwidth of the (modulated) blocker. Furthermore, the LMS
filter parameters L, SF , and MU are to be properly chosen to obtain convergence of the
coefficients and finally sufficient mitigation performance. The truncation factors SF and
MU can be easily adjusted by a control interface in the USRP architecture (settings bus
interface). Besides, flags for adding back the interferer and bypassing the algorithm can be
defined. For that purpose, a user interface has been designed in GNURadio [BCB+]. On
the other hand, the AF length and the FIR characteristics cannot be easily adjusted. While
the FIR coefficients may be changed using a block RAM, the maximum length is fixed in
the synthesized design. Also, the filter design should be established on a PC, as it would
not be efficient to run it directly on the FPGA. Then, the new set of filter coefficients can
be transferred via the Ethernet interface to update those in the FPGA design. Eventually,
the general filter structure may be changed during runtime by partial reconfiguration of the
FPGA [Xila].
6.3 Experimental Results

This section analyses the real-time performance of the algorithm with real communication
signals that are fed to the device and processed with the custom FPGA firmware. For that
purpose, a two-tone and a BPSK-modulated blocker have been generated, similar to earlier
post-processing-based experiments. In both cases, the AF stage comprises L = 4 taps that
are fixed in the FPGA design.
The setup for the two-tone scenario is depicted in Figure 6.6. Two sinusoidal signals with
a spacing of 600 kHz are fed to the USRP+WBX device using a power combiner [Ett].
A 10 MHz reference clock is adopted for coherent interaction between the devices. The
power spectral density (PSD) with the stock and custom firmware is illustrated in Figure 6.7.
Each curve depicts the average of 15 FFT blocks with Hann windowing and length 1024.
The corresponding truncation factors are SF = 16 and MU = 11. It is apparent that
the upper and lower third-order IMD products (IMD3) have been mitigated. However, the
algorithm is also affecting the negative side of the BB due to the real-valued implementation
of the bsd_filter. A complex implementation would not affect the mirror frequency range,
but results in a larger amount of required logic resources. Unfortunately, there are notches
appearing at the locations of the IMD products after mitigation. A detailed analysis revealed
149
Power
Combiner
TX
10 MHz phase-locked
REF
R&S SMHU
93.3 MHz at 11 dBm
WBX
RX
USRP N210
R&S SMY02
92.7 MHz at 11 dBm
fcentre = 90 MHz
Figure 6.6: Experimental setup of the two-tone scenario.

that these are only visible with measured two-tone signals and if averaging is applied over
several FFT blocks. It is likely to be caused by the phase noise of the signal generators and
the finite stopband attenuation and finite slope steepness of the FIR filters. Hence, good
filter characteristics are essential for providing a clean reference signal to the AF stage. In
practice, the reference distortions contain also other components than the IMD products.
An increased filter order is desirable to obtain a higher stopband attenuation and a lower
transition bandwidth. However, the effect is accepted in this case, due to the large amount
of required resources. In the subsequent paragraph, it will be seen that such effects do not
appear with modulated blockers.
Next, a BPSK-modulated blocker is combined with a weak desired GSM channel and fed to
the USRP+WBX configured with the custom firmware. The measurement setup, depicted
in Figure 6.8, consists of two vector signal generators. The BPSK signal has a symbol rate of
450 ksym/s and is filtered with a raised-cosine, which has a roll-off factor of 0.5. In addition,
a GSM channel is generated by a second vector signal generator with all 8 time slots being
active. As seen in Figure 6.9, the weak GSM channel is completely masked by spectral
regrowth of the BPSK signal. After mitigation, the correlating non-linear distortion on top
of the weak GSM channel is cancelled. It is worth noticing that the cleansed spectrum
perfectly encloses the GSM channel and no notch-like effect is occurring. The corresponding
truncation factors are SF = 18 and MU = 9. In Chapter 7, it will be demonstrated that the
BER of a weak GSM channel, which is affected by non-linear distortion induced by nearby
stronger channels, can be improved significantly.
150
6.4 Summary of Real-time Implementation Issues
PSD (dBm/Hz)
20
40
Distorted Signal
Cleansed Signal
TT Input
Mirror Frequency
60
IMD3
IMD3
LO Leakage
80
100
Undesired Notches
120
0
1
2
3
6 5 4 3 2 1
Figure 6.7: Two-tone mitigation performance with the custom FPGA firmware [Sch13].
Power
Combiner
TX
10 MHz phase-locked
REF
R&S SMIQ06B
122.3 MHz at 65 dBm
WBX
RX
USRP N210
R&S SMBV
123 MHz at 8 dBm
fcentre = 120 MHz
Figure 6.8: Experimental setup of the BPSK scenario.
6.4 Summary of Real-time Implementation Issues

This chapter has presented a resource-efficient implementation of the digital feedforward
mitigation on the FPGA of the USRP N210 [Ett]. By embedding the mitigation algorithm into the radio, the cleaned signal is directly forwarded to the host PC for further
BB processing. Thus, the receiver self-regulates its non-linearity, whereas the host PC is
offloaded. The proposed custom FPGA module adds negligibly small processing delays,
hence, TDMA-like communication systems are not disturbed by the mitigation processing.
Even the simplified FPGA prototype shows a promising mitigation performance through RF
151
PSD (dBm/Hz)
20
40
60
Distorted Signal
Cleansed Signal
BPSK Input
Single GSM Channel
Mirror Frequency
Spectral Regrowth
LO Leakage
80
100
120
6 5 4 3 2 1
0
1
2
3
Figure 6.9: BPSK mitigation performance with the custom FPGA firmware [Sch13].
device measurements with two-tone and BPSK blocker signals. The mitigation performance
may be improved further by employing the extended cascaded model with a complex-valued
LMS implementation, as it has been proposed in Chapter 4.
152
The main scope of this chapter is to present some use cases of the digital feedforward mitigation algorithm. Spectrum sensing in CR, GSM downlink reception, and GSM-based passive
radar are discussed as examples. The results have been published in [GSH+12c; GSH+12b;
GSH+12a; GSH+13].
7.1 Spectrum Sensing in Cognitive Radio

7.1.1 Cognitive Radio Principle
The CR was first coined by Joseph Mitola in his famous article [MM99] following his previous
article on SDR [Mit95]. It is a promising concept to improve the spectrum utilisation in
communication systems that have a fixed frequency allocation. Typically, many frequency
bands in the spectrum are largely unoccupied most of the time, whereas a few are heavily
used [Hay05]. A CR is aware of its radio environment and is able to adapt its internal
parameters to use available radio resources. Thereby, a secondary use of licensed bands is
done on a basis of non-interference with the primary users (PUs) that own the the licensed
bands [Hay05; YMC+08].
Whereas a SDR supports multiple standards, a CR adds cognition to it and arranges the
adjustments of its parameters [Law10]. Thus, a CR can be considered as an intelligent
extension of a typical SDR. To make this concept work, the radio needs to fulfil several
capabilities that can be depicted in the cognition cycle [Hay05] illustrated in Figure 7.1.
First, the radio needs to perform spectrum sensing to be aware of the radio environment,
reflecting the capability observe. Then, an intelligent processing is performed on the observed
data, reflecting the capabilities orient and plan. Based on this processing, the CR decides
on the spectrum opportunities and adjusts its physical transmission parameters, such as
the transmission power, frequency, and modulation. Finally, the CR nodes exploit these
opportunities under the constraint to avoid interference to the PUs, that corresponds to the
capability act. The reasoning and learning is a central part of a CR node and is therefore in
153
Radio Environment
Observe
Act
Learn
Orient/Plan
Decide
Figure 7.1: CR cognition cycle [Hay05; PKM+11].

the centre of the circle. It is often referred to as cognitive engine that takes decisions based
on a knowledge base, a radio environment map, and a policy engine.
There are many challenges of the CR concept, but those related to the RF front-end are on
focus here. Basically, CRs impose very tough requirements on the transceivers electronics
[CB05; Raz09; MKH+10]. They need to work under huge dynamic range conditions of up
to 100 dB and require a high resolution A/D interface as well as highly linear front-end
components. Even more, a CR is supposed to operate at any frequency under a wide range
from tenths of MHz to about 10 GHz. Due to the wide bandwidth, also down-conversion of
interferers at LO harmonics may be crucial for CRs. Moreover, a CR requires fast digital
radio processors that enable fast sensing and adaptation without causing harmful interference
to other users. Eventually, a CR need to tolerate interferers at any frequency band, causing
high requirements to the mixing purity.
There are certain applications of CRs from TV white spaces, public safety networks, and
cellular networks [WGC11]. TV white spaces denote the frequency bands allocated to broadcasting services, but which are not used most of the time at a certain location. They offer a
good playground for researchers to verify the CR principle in real-world scenarios until they
154

move to heavily used licensed bands. In fact, the CR concept is far from being mature and
requires additional experimental-driven research to convince regulators to allow CR in future.
However, there is also a first CR standard, the IEEE 802.22 wireless regional area network, a
broadband internet service designed for use in rural areas. In public safety networks, CR can
assist emergency responders of a disaster scenario to overcome the incompatibility of their
radios, by serving a facilitator between different devices. In general, CR is applicable in
cellular networks to cope with the growing number of services and demands while remaining
within the given spectrum mask.
For details on the CR principle, the interested reader is referred to [Fet09; Ars07].
7.1.2 Sensing Algorithms
It is obvious that a CR requires reliable spectrum sensing algorithms to detect the PU

activity and to minimize the interference to existing communication systems [CGG+10].
There are three main sensing algorithms known from literature: energy detection, matched
filter detection, and cyclostationary feature detection [Ars07]. Energy detection is a simple
common technique that does not require a-priori knowledge about the signals to be detected.
Instead, matched filter or autocorrelation-based detection is more complex and requires the
signal characteristics, i.e. for every PU system. The feature detection seeks signatures
produced by the modulation schemes, e.g. periodic signal features that are visible only in
higher order statistics. Basically, the more knowledge of the signals to be sensed is available,
the weaker signals can be detected [WM10].
However, there are several challenges of sensing. First, the effects of the radio channel need
to be considered, especially fading conditions and the so called hidden terminal problem
due to shadowing effects. Second, the time to arrive at a reliable decision without causing
interference should be significantly small. In practice, interference is almost unavoidable, as
the PU can be only detected after it already transmits. Eventually, future available radio
resources in time, frequency, and space need to be predicted properly. There are several
techniques to improve the sensing results, e.g. by multi-antenna-based sensing techniques
[CB05] or by collaborative spectrum sensing among multiple nodes [CB05; QCP+08]. In the
latter approach, the spatial diversity of the nodes is used and data fusion is subsequently
performed to overcome the hidden terminal problem.
155
Energy-based Detection
Here, the main focus is on energy-based detection because it is the most common approach
for detecting unknown signals in CR applications especially due to its ease of implementation
[Urk67; DAS03; MGC10]. A major drawback of an energy detector (ED) is its large sensing
time when operating in low SNR conditions. Also, an ED is not able to sense weak signals
that are covered by noise, however, they have to detect SNRs as low as 20 to 30 dB
[Raz09].
The ED, illustrated in Figure 7.2, measures the energy of the input signal over a specific time
interval T and compares it with a threshold [MGC10; GKM+11a; GKM+11b; Urk67]. The
||2
ADC
W
1
T
H0,1
Figure 7.2: Block diagram of a discrete energy detector.

detector usually compares between two hypotheses signal present (H1 ) and signal absent
(H0 ):
H0 : x(t) = n(t)
H1 : x(t) = n(t) + s(t),
where x(t) denotes the received signal, s(t) the band-limited (W ) signal of the PU, and n(t)
AWGN. According to Figure 7.2, a discrete-time test statistic [MGC10]
N
H1
1 X
V =
b 2
|xi |2
0 i=1
H0
(7.1)
can be deduced, where N = T W is the total number of samples acquired in the sensing
time T and 02 is the noise power of the detector. There are closed-form expressions for the
distribution of both hypotheses depending on the type of modelling for the input signal. The
probability of detection PD and the probability of false alarm PF A are the most common
performance figures for an ED and can be expressed by
PF A = P (V > | H0 ) and
PD = P (V > | H1 ).
156

Setting the threshold for a desired performance (PF A , PD ) requires knowledge about the
noise power 02 of the ED [YMC+08]. In theory, the noise power is assumed to be perfectly
known but it should be estimated in real systems based on recorded samples from a noiseonly observation H0 . In general, low SNR signals can only be detected by sensing a large
T like in a spectrum analyser [Ars07]. For long sensing duration T , the aforementioned
Gaussian distribution of the noise is valid, and hence the (7.1) in both hypotheses can be
approximated as Gaussian [Sha10]:
N ( , 2 ) H
0
0
0
V
N ( , 2 ) H
1
1
1
with 0 = 2N, 02 = 4N, 1 = 2N(1 + SNR) and 12 = 4N(1 + 2SNR). In this case, PD
and PF A can be calculated with
PF A

1
( 0 )
( 1 )
1
= erfc
and PD = erfc
,
2
2
0 2
1 2
(7.2)
where erfc() denotes the complementary error function. In fact, assuming Gaussian random
processes provides closed-form expressions for computing the threshold for a given performance (PF A , PD ).
7.1.3 Problem Definition

Compared to the use cases considered in the state of the art, the focus of this work is on
analysis and mitigation of receiver non-linearity for reliable detection of the radio spectrum
in CR, such that weak signals are properly detected as well as transmit opportunities are
not missed. Consequently, distortions are mitigated in the whole BB spectrum to make
it suitable for spectrum sensing, instead of eliminating distortions only created on a single
information-bearing signal of interest for correct demodulation. In fact, contrary to the
signals that have been assumed in the previously mentioned references, this work aims at
mitigation of interference caused by realistic modulated signals. In keeping with this objective, realistic constellations with wideband CW user and blocker signals are considered for
the interference analysis and mitigation. A feedforward approach with time- and frequency
domain implementation of an AF stage is investigated in order to cleanse the BB of distortions. For this purpose, the spectral sensing information about level and spectral location of
strong interferers is used as an input for the mitigation algorithm to handle the most critical
distortions.
157
The main objectives for considering non-linear distortion in the operation of a CR are
Reliable spectrum sensing under huge dynamic range conditions and
Proper demodulation of weak desired signals for communication.
Signal constellations with weak user and strong blocker signals are on focus in the following
discussion. However, from the CR perspective, strong interferers can be PU signals and the
weak user signals might be associated with secondary users (SUs).
First, IMD products can show up as unwanted signals in free frequency bands, where the
CR may operate as SU. As a consequence, simple energy-based spectrum sensing would
detect this band as occupied and the radio would miss its transmit opportunity [GKM+11a].
Figure 7.3 depicts the distorted BB spectrum ye(t) of Figure 2.33 with 1000 averages and the
thresholds in 1024 sub-bands for energy detection. For threshold calculation, a noise-only
measurement and a false alarm probability PF A = 0.01 have been used. The sub-bands
are filtered from the input signal by a 1024-point FFT. With respect to the lower IMD
region in Figure 7.3, signal present was detected in all realisations (PD = 1). These are
actually the false alarms due to interference induced by the non-linear wideband receiver,
decreasing the spectrum sensing reliability and causing the loss of opportunity for secondary
PSD (dBm/Hz)
communication.
10
20
30
40
50
60
70
80
90
100
110
Distorted Signal
(PF A = 0.01)
Lower IMD Products
False alarm
due to interference
479
481
483
487
489
491
485
493
495
Figure 7.3: Averaged BB spectrum and energy detection thresholds [GSH+12a].
Second, IMD- and XMD-products may coincide with a desired SU signal, causing an increase
of the BER and hence difficulties during demodulation.
To alleviate these effects, either interferers themselves have to be cancelled or their distortion
products need to be reduced. The interferer itself can only be fully compensated if it is sub-
158

tracted in the analogue domain at the front-end input, i.e. before the signal passes through
any non-linear component. In theory, the analogue cancellation with digital generation of
an interferer replica is possible, but not feasible as the interferer must be known in advance
(Section 3.4). This approach might work for deterministic or slow-changing (modulated)
signals, which can be predicted with high accuracy. Mitigating the distortions is desirable
and takes also the interference to other weak users into account. Nevertheless, information
about the spectral location of strong interferers could be considered in the sensing algorithm
or exploited for secondary transmissions where adjacent frequency ranges being affected by
the distortion products are not used [MKH+10]. Secondary transmissions very close to a
strong incoming signal can cause harmful intermodulation in the receiver of the mutual secondary communication partner. Instead, frequency ranges that are further away can be used
for transmissions.
Adopting digital feedforward mitigation especially for cyclostationary feature detectors is reported in [RGV+13; AMV+13]. Alternative ideas are presented in [MKH+10; Mar09]. The
idea in [MKH+10] is to assist the sensing by determining the spectral location of distortion
products, so the CR can choose an IMD-free spot for its operation. A similar approach is presented in [Mar09], that proposes pre-selection filters and linear circuits to prevent non-linear
distortion and to avoid affected frequency ranges like being occupied by other users.
7.1.4 Detection of Mitigated Multi-carrier Signals

In order to verify the sensing algorithm under conditions of receiver non-linearity, measurements have been conducted with the USRP N210+WBX [Ett]. A multi-carrier CW signal
configuration as sketched in Figure 2.33 is assumed with a
Weak signal: 5-tone sine at 2 MHz with 68 dBm (400 kHz tone spacing), and a
Blocker: 5-tone sine at 5 MHz with 20 dBm (100 kHz tone spacing).
The signal and carrier frequencies were chosen arbitrarily. The large difference in power
of 48 dB was used to provoke non-linearly induced interference at the receiver. The centre
frequency and the sample rate were 487 MHz and 16.67 MHz, respectively
The data acquisition and the processing were implemented in MATLAB using NONLIM.
Finally, the suppression of the IMD products are examined on the basis of different filter
orders and step sizes of the AF stage.
Figure 7.4 illustrates the achieved mitigation performance with conventional LMS filtering
after 10 240 iterations. Because odd-order IMDs around the blocker are superimposed on
159
PSD (dBm/Hz)
PSD (dBm/Hz)
20
30
40
50
60
70
80
90
100
110
120
8
20
30
40
50
60
70
80
90
100
110
120
4
Distorted Signal
Cleaned Signal
NONLIM Filter Range
4.2
2
0
2
4
(a) Entire baseband.
4.4
4.6 4.8
5
5.2 5.4
(b) Fundamental zone enlarged.
5.6
5.8
Figure 7.4: Mitigation performance with measured multi-carrier signals.
each other, they are jointly denoted by spectral regrowth in the remaining discussion. The
suppression of the spectral regrowth in the fundamental zone is on average 23 dB, as seen in
Figure 7.4(b), indicating the possibility to cleanse the BB of regrowth for reliable spectrum
sensing. The mitigation has been performed with a simple BB non-linearity model manifesting 2nd , 3rd , and 5th -order distortion (Section 2.5). The operating range of the mitigation
algorithm is depicted by the orange-coloured surface in Figure 7.4(a). For the AF stages,
different step sizes have been used: 2 = 20, 3 = 30, and 5 = 10 (subscript n in n
denoting the order of non-linearity). The chosen step sizes and the filter order are relatively
large because the distortion products had very low amplitudes in general. A substantially
long AF length of 128 was necessary to mitigate the distortions, attributed to the simplified
I/Q-based processing and the model mismatch.
160

Now, the ED performance is to be compared before and after application of the imple-
PSD (dBm/Hz)
mented mitigation algorithm on a specific bandwidth of interest (260 kHz) in the fundamental zone. Figure 7.5 depicts the original and cleaned baseband spectrum, the band
10
20
30
40
50
60
70
80
90
100
110
Oiginal Signal
Cleaned Signal
(PF A = 0.01)
Band of Interest
479
481
483
485
489
491
487
493
495
Figure 7.5: Original and cleaned baseband spectrum with energy detection thresholds.
of interest, as well as its ED threshold for PF A = 0.01. Figure 7.6 illustrates the
receiver operating characteristic (ROC) for both cases, showing the detection of interfer-
Probability of Detection (PD )
ence [GKM+11a]. Note that the ROC is typically presented for the desired detections. In
1
0.9
0.8
0.7
0.6
0.5
0.4
0.3
0.2
0.1
0
Original Signal
Cleaned Signal
Noise-only Characteristic
0
0.1
0.2 0.3 0.4 0.5 0.6 0.7 0.8

False Alarm Probability (PF A )
0.9
Figure 7.6: ROC curves before and after mitigation.

this case, it shows the detection of non-linearly induced interference with the aim to minimise
these detections. While the interference detection probability PD , or equivalently the false
alarm probability due to interference, is always one for the original signal, the ROC curve
of the cleaned signal is approaching the ideal noise-only characteristic (PD PF A ), indicat-
ing that there is no more interference signal present in that band. In fact, the mitigation
161
aims for avoiding any detection in the band of interest, as only non-linear distortions appear
there. After mitigation, the effect of the non-linearly induced interference has been fully
compensated, thus improving the reliability of spectrum sensing. In some cases PD is even
below PF A , attributed to the long AFs that suppressed all reference signal content being
present in the desired signal. Figure 7.4(a) clearly shows that the algorithm in the used
implementation only works locally compared to the wideband impact of the distortions.
7.2 GSM Cellular Radio

7.2.1 GSM Physical and Link Layer Overview
The GSM is the most common mobile communication system worldwide and denotes the
second generation (2G) of the digital cellular networks [EVB+09]. GSM implements a
Gaussian minimum-shift keying (GMSK) with a modulation index of h = 0.5 [Etsd]. It is a
variant of the minimum-shift keying that encodes each symbol as a half sinusoid resulting in
a constant envelope modulation. In the GMSK case, the data stream is additionally filtered
with a Gaussian pulse-shape filter, having a filter length bit period product BT = 0.3, to reduce the side-lobe power. This modulation has been chosen to be resistant to any non-linear
distortion in the transmitter or receiver, enabling the deployment of low-cost amplifiers in
the MS RF front-end.
The symbol rate accounts for rb = 1625/6 103 bps = 270.833 kbps. There are four different
GSM bursts, namely normal burst (NB), frequency correction burst, synchronisation burst,
and dummy burst (DB) [Etsb]. In total, 148 bits are grouped together in one burst, and a
guard period of 8.25 bits is left in between the bursts to avoid collisions. The burst duration
is approx. 577 s. Figure 7.7 illustrates the structure of a NB, where 2 57 = 114 are pure
data bits. Regarding the access to the network, the GSM implements a TDMA scheme,
3
57
Data bits block I
26
Training sequence
57
Data bits block II
Figure 7.7: GSM NB structure.

as illustrated in Figure 7.8. Eight time slots (TSs) are grouped in frames that are again
integrated into higher hierarchy frames. A single user can assign one or more TSs.
162
Hyper frame
0
2047
Super frame, 6120 ms
0 1
50
Multi-frame, 120 ms
0 1
25
0 1 2 3
5 6 7
TDMA frame, 4.615 ms
Transmission burst, 577 s

Figure 7.8: GSM frame structure.
In terms of duplexing, FDD is widely used, i.e. there are separated frequency bands for upand downlink. These bands are divided into 200 kHz wide channels where the GSM carriers
are placed.
There are two different types of RF channels that are distinguished in GSM:
traffic channels (TCHs) and control channels [Etsb]. The complete control channel, containing the broadcast control channel (BCCH) and other channels, is here referred to as
BCCH. These are beacon channels which transmit continuously at full power. For example,
they are used by the MS for synchronisation, network registration, and paging. Every cell in
the GSM network has one BCCH and several TCHs. Table 7.1 shows its TDMA structure,
which is clearly defined in the standard [Etsb]. In the Table only the first 10 TDMA frames,
Table 7.1: Excerpt of TDMA frame mapping of the BCCH, cp. [Etsb]
0
1
2
3
4
5
6
7
8
9
TS0
FCCH
SCH
BCCH
BCCH
BCCH
BCCH
CCCH
CCCH
CCCH
CCCH
TS1
DB
DB
DB
DB
DB
DB
DB
DB
DB
DB
TS2
DB
DB
DB
DB
DB
DB
DB
DB
DB
DB
TS3
DB
DB
DB
DB
DB
DB
DB
DB
DB
DB
163
TS4
DB
DB
DB
DB
DB
DB
DB
DB
DB
DB
TS5
DB
DB
DB
DB
DB
DB
DB
DB
DB
DB
TS6
DB
DB
DB
DB
DB
DB
DB
DB
DB
DB
TS7
DB
DB
DB
DB
DB
DB
DB
DB
DB
DB
each consisting of 8 TSs or bursts, are illustrated. TS 0 is always used for control information, e.g. the frequency correction channel (FCCH), or the synchronization channel (SCH).
The remaining seven TSs are reserved for TCHs, but are usually not used. In the idle state,
DBs are transmitted, which have a predefined bit sequence.
For more details on the GSM system, the interested reader is referred to [EVB+09] and
references therein.
7.2.2 Robust Train Radio Reception (GSM-R)

The work reported in this section provides insight into the degradation of performance of the
GSM system by RF impairments in the receiver under realistic interference conditions. The
feedforward mitigation algorithm has been applied to real measured GSM signals, distorted
due to receiver non-linearity. A significant reduction in the BER for the desired channel
has been achieved by reducing non-linear distortions in the receiver caused by two strong
neighbouring interferer signals. Thereby, the principle of DSP-based mitigation of receiver
non-linearity was successfully demonstrated for real communication signals. The result of
these studies have been published in [GSH+13].
Although the GSM system is very mature, interference problems in GSM can still occur. Typically, co-channel interference due to bad network planning, coexisting cellular systems, and
radio channel effects are of major concern [Sch94]. The focus here is on downlink interference
in GSM, caused by adjacent channels, channels located further away, or even out-of-band
signals. All these types of interference signals, herinafter referred to as neighbour channels,
can create harmful interference on top of weak desired channels. The interference is induced
by the spectral leackage of the neighbour channel or by non-linear distortion in the receiver
front-end of the MS. Although the GSM signal itself is not sensitive to non-linearities due
to its constant-envelope modulation, IMD may occur in unfavourable signal configurations
with at least two strong GSM neighbour channels. Receiver non-linearities seem not to be
critical if two adjacent channels belong to the same BTS, since power control mechanisms
avoid co-channel interference. However, a crucial coexistence scenario practically appears
in the adjacent downlink bands of GSM-R, containing a weak desired channel, and public
GSM (GSM-E) that may include various strong channels. Thus, distortions are easily generated in wideband direct-conversion receivers without filtering as used in SDR, but also
in narrow-band superheterodyne receivers with limited selectivity and linearity. Distortions
may fall top of the desired channel and thus increase its BER, causing a poor speech quality,
low data rate, or even dropped calls and a loss of connectivity to the serving BTS. The
164

signal quality becomes unacceptable if the BER exceeds a certain threshold, or equivalently,
a certain receiver quality (RXQUAL) level [Etsc]. Because distortions on top of the desired
channel cannot simply be filtered out, clever signal processing is required. The aim is on
design and analysis of DSP algorithms to mitigate non-linear distortions in GSM in order to
improve the BER performance for maintaining reliable GSM communication.
The main principle of the DSP-based mitigation algorithm has been already discussed in
Chapter 4. Up to now, DSP-based mitigation algorithms alleviating receiver distortions
have been applied to CW signals (sinusoid, multi-carrier) and simple amplitude modulated
signals. There is no work in the state-of-the-art literature applying this kind of DSP-based
mitigation to real communication signals. It will be discussed, how this algorithm can be
used for GSM signals, aiming for a future employment into real devices.
The GSM communication scenario is sketched in Figure 7.9. The MS is communicating with
Serving
BTS
Interferer
MS
Figure 7.9: Considered interference scenario.

its serving BTS, while it is interfered by another BTS operating in a neighbour channel.
The amount of non-linearly induced interference depends on the signal level S of the serving
BTS, and the signal level I of the interferer. The SIR is a widely used parameter to describe
the amount of interference. These signals vary over time, if the MS is moving (fading).
Different types of interfering signals may be present: other GSM signals, UMTS, or even
Long Term Evolution (LTE) signals. Figure 7.10 shows a simulated baseband spectrum
with a weak desired GSM channel and two strong GSM neighbouring channels. Considering
GSM interferers, at least two strong signals need to be present to cause harmful IMD on
top of the desired channel. Conversely, single neighbouring UMTS or LTE signals with high
PAPR would cause severe regrowth that may mask the narrow GSM signal completely. The
spectrum has been computed by a 8192-point FFT with Hann window. In Figure 7.10, the
165
PSD (dBm/Hz)
20
30
40
50
60
70
80
90
100
110
120
4
Desired GSM channel

Strong neighbour channels
Sum signal + noise + IMD
3.2 2.4 1.6 0.8

0
0.8 1.6
2.4
3.2
Figure 7.10: Complex baseband spectrum with a weak channel and two strong neighbouring
GSM channels, creating IMD on top of the weak channel.
Neighbour Interference
GSM
-R
921
925
GSM 900
GSM-E
935
960
f in MHz
Figure 7.11: Radio spectrum frequency allocation for GSM 900 downlink in Europe.
SIR was set 0 dB, resulting in a BER for the desired channel of 0.38 (103 600 total bits) that
makes the signal already unusable.
The scenario illustrated in Figure 7.10, for example, practically appears in the GSM-R
[KBN+06; BFM+10]. GSM-R is based on the common GSM standard and uses reserved
frequencies for railway applications, e.g. 876-880 MHz in uplink, and 921-925 MHz in downlink in Europe [Etsd]. The frequency allocation for GSM 900 downlink band in Europe is
illustrated in Figure 7.11, where interference in the downlink from public GSM-E band to the
GSM-R band is of major concern. The problem became more serious since the refarming of
the GSM 900 band, since operators are allowed to employ UMTS or LTE systems in GSM-E
band. The consequences of this close frequency allocation for the MS in downlink are less
considered by regulators or network planners. Although operators keep themselves within
their assigned frequency ranges and power levels, it is likely that non-linear distortions in
the MS can easily cause harmful interference. A detailed description of GSM-R sensitivity
to external wireless interference is given in [BFM+10].
166

Application of Feedforward Mitigation
The main idea in order to decrease the BER of GSM signals in aforementioned interference
scenarios is to mitigate the effects of receiver non-linearity. This is done by DSP at I/Q
signal level, which is inserted in between the ADC of the receiver and the GSM specific
demodulation. The basic system architecture is shown in Figure 7.12. The pure RF antenna
RF
x(t)
digital
analogue/digital
Non-linear
Receiver
y(n)
Feedforward
Mitigation
Processing
y(n)
GSM
Demodulator
{
s}
BER
Figure 7.12: Basic system architecture, treating with non-linear distortions in a GSM
receiver.
signal x(t) is amplified, down-converted, and digitised by a non-linear receiver to y(n). It
does not only contain an amplified baseband version of x(t), but also distortion products
added by the receiver. In the very simple case, the non-linear behaviour of the receiver can be
modelled by a memoryless polynomial, as discussed in Section 2.5. IMD products generated
by mixing of the desired and the interference signal are particularly annoying [Ken00]. The
mitigation DSP block tries to eliminate these distortions, outputting the signal y(n). The
signal is then further processed by the GSM demodulator including synchronisation and
channel equalisation. Finally, the BER can be computed with the received bits {
s}, if the
transmitted bits are known.
Referring to Chapter 4, a feedforward mitigation algorithm has been applied to GSM signal
in order to handle third-order distortions of the LNA at RF, falling on top of the desired
signal. Figure 7.13 illustrates the main architecture, where the basic idea is to isolate the
C
interferers and to regenerate the distortions they cause. A lowpass/bandpass (hR
LP /hBP ) pair
splits the input signal into a reference signal yref , containing the interferer only, and the
desired signal ydes , including the desired signal and the distortions caused by the interferer.
The reference signal is then fed into a reference non-linear model, in this case a simple memoryless polynomial. The term x2 x denotes the baseband representation of the third-order
distortions induced by the real-valued RF passband signal at the RF amplifier (Section 2.5).
The distortion products are then further processed by an AF to adjust their magnitudes
and phases to exact values [Hay02]. The common error signal e(n) is used for adapting the
167
hR
LP
ydes
hC
BP
y(n)
Adaptive Filter
yref
hR
LP
x2 x
y3
FIR
e(n) = y(n)
f
Reference
Non-linearity
Adaptation
Algorithm
f
Figure 7.13: Block scheme of the implemented mitigation algorithm for GSM.
FIR filter taps and represents the main output of the mitigation processing. Finally, the
reference distortions are subtracted from the desired signal, hence reducing the non-linear
distortions falling onto the desired signal. Both I and Q components of the complex BB
signal are processed separately for reasons of simplification.
Due to the sample-by-sample operation of the adaptive filtering and the constant-envelope
modulation, the TDMA structure does not affect the convergence behaviour of the AF. Instead, the characteristics of the lowpass/bandpass pair as well as the quality of the reference
signal yref dictate the mitigation performance. Table 7.2 summarizes the used filter parameters for a best case estimation. The filter order depends on the dynamic range between
Table 7.2: Filter parameters for the considered GSM interference scenario.
Parameter
Passband hR
LP (real)
Passband hC
BP (complex)
Value
0.25 MHz . . . +0.25 MHz
2.6 MHz . . . 0.8 MHz
Transition bandwidth f
50 kHz
Stopband attenuation
80 dB
FIR order
1200
interfering and desired channel at hand. In this example, it is chosen sufficiently high to
obtain high stopband attenuation, and thus pure desired and reference signals. In general,
sufficient knowledge about the interferer signal is required, such as its modulation and fre-
168

quency location. The reference distortions are imperfect as the interferer itself suffers from
non-linear distortion (in-band distortion). For exact distortion regeneration, an undistorted
version of the interferer, like at the antenna input, is desirable. Suitable approaches for
obtaining a better reference signal would be using a second receive chain for the interferer
with additional attenuation for linear reception [ZMS09], or re-modulating the interferer
after demodulating its bits.
Demodulation and BER Detection

In this work, DBs of the GSM BCCH are used as a desired signal. That is, the transmitted bit
sequence is known and used to compute the BER at the receiver. The mitigation processing
as well as the GSM demodulation have been implemented in MATLAB using the existing
GSMsim toolbox that is fully compliant with the GSM standard [EM97]. Figure 7.14 shows
a block diagram of the GSMsim framework that enables the implementation of a complete
ADC
Speech
encoder
Channel encoder
interleaver
MUX
RF
Tx
GMSK
modulator
Transmitter
GSM communication system. Random bits are generated and passed through the channel
DAC
Speech
decoder
Channel decoder
deinterleaver
DeMUX
GMSK
demodulator
RF
Rx
Receiver
Mobile
Channel
Figure 7.14: Block diagram of the GSMSim framework [EM97].

encoder/ interleaver and are eventually compared to the received bits delivered by the channel
decoder/ de-interleaver. The voice interfaces are not included in the toolbox and are just
depicted to complete the communication system.
The transmitter adds channel encoding to assist the receiver for detecting transmission errors.
The task of the interleaver is to shuffle the bits and to distribute them over a number of
bursts, so the errors appearing in a received data block are uncorrelated. Also, a training
sequence is added by the MUX to enable channel estimation at the receiver. Finally, the
169
bit stream is modulated with GMSK, performing differential encoding of the incoming burst
with a non-return to zero sequence.
The detailed block diagram of the GSM receiver is illustrated in Figure 7.15. It represents
yb(n)
Matched filter
MLSE
Demux
Channel decoder/
de-interleaver
{b
s}
Channel estimator/
synchronization
Figure 7.15: Block diagram of the receiver implementation used in GSMsim [EM97].
not a complete front-end as, e.g., filtering of the desired channel has to be applied manually.
The demodulator estimates the most probable received symbol sequence with the help of the
training sequence. First, channel estimation, synchronisation, and matched filtering of the
received GSM bursts are performed. The delay and the channel coefficients are determined
by cross-correlating the known training sequence with the actual received one. The sequences
provided by the matched filter, containing one sample for each transmitted symbol, are then
further processed by the minimum least square error (MLSE) detector. The MLSE detector
is implemented as a Viterbi equalizer which provides an estimate of the most probable
sequence of transmitted binary symbols. Finally, the estimated bits are extracted by demultiplexing, de-interleaving, and channel decoding.
Simulation Setup
In order to evaluate the performance of the mitigation algorithm, a complete GSM link
including receiver non-linearity has been simulated. The non-linear receiver is represented by
a third-order polynomial and an AWGN source, whereas the radio channel is not considered
herein. The signal configuration as sketched in Figure 7.10 has been simulated with different
SIRs for the desired channel, taking into account fixed SNR of 15 dB (constant signal and
noise power for the desired channel). In total, 700 DBs (103 600 bits) have been analysed.
The adaptive filtering was performed using conventional time domain LMS filtering with
a single tap, that has been found to provide sufficient convergence speed and low residual
error.
170

R&S
SMU200A
Control bus (GPIB)
GSM interferers
919.7 MHz@
0.6 . . . 8.4 dBm
PC
f
BCCH ch. 955

921.2 MHz@
98 dBm
ETH
R&S
CMU200
WBX
USRP
N210
Figure 7.16: Laboratory setup reproducing the considered interference scenario.

Measurement Setup
An experiment with same parameters as in the simulations has been conducted by using
an automated setup, as illustrated in Figure 7.16. Fully standard compliant BCCH bursts
were generated with a BTS emulator. The interferer signal with two strong neighbouring
GSM signals (1.5 MHz offset) was coupled into the RF path using a directional coupler.
Both signals were received by the SDR USRP N210, equipped with the WBX wideband
front-end [Ett]. The linearity of WBX with a third-order input-referred intercept point
IIP3 = +13.8 dBm is considerably good [GSH+12a] and required high power levels for the
interferer. The centre frequency and sample rate of the device were 925 MHz (LO offset of
3.8 MHz) and 25 MHz, respectively. The I/Q data has been transferred to a PC via GigE
and processed offline as described in Section 7.2.2.
Results
The BER performance before and after mitigation (BM/AM), as well as the ideal BER level
without interference (linear receiver), are shown in Figure 7.17 for simulated and measured
data. Basically, the BER is saturating at around 0.007 (RXQUAL = 2), which is the lower
bound dictated by the fixed SNR of the desired channel. The simulated BER performance
after mitigation is approaching to the ideal BER level, indicating that almost all IMD on
top of the desired signal has been mitigated. The BER has been reduced significantly to less
than 0.03 (RXQUAL = 4) for all SIR conditions. That is, the GSM receiver provides a better
detection of the transmitted symbols after applying the mitigation processing. However, the
mitigation gain for the measured data is a bit less in low SIR conditions, mainly due to the
inaccurate distortion regeneration. In contrast to simulations, an AF of order 6 has been
171
BER
101
102
103
BM-Sim
BM-Meas
0
AM-Sim
AM-Meas
5
SIR (dB)
Ideal BER level

10
7
6
5
4
3
2
1
0
RXQUAL
100
15
Figure 7.17: BER performance vs. SIR for the desired channel before (BM) and after mitigation (AM).
used for processing the measurement data. Thus, inaccuracies are compensated, that have
not been considered in the reference model, such as memory effects and I/Q imbalance.
It has been shown that non-linear distortions can be the predominant interference phenomena, causing serious problems in scenarios with strong neighbouring channels. Due to the
presence of non-linear distortions in almost every receiver, this algorithm can be applied
in many scenarios. The effectiveness of this approach has been demonstrated here on real
communication signals.
In general, the performance is limited by the basic assumptions for DSP-based mitigation
processing [GSH+12a]. Assuming high input signal power, the receiver amplifier and mixing
stages might be overloaded or even clipping might occur in the ADC. In such conditions, the
generated distortions are almost irreparable. Moreover, detecting and isolating the interferer
might need an additional RF path, making the receiver architecture more complex. In
addition, the effect of the radio channel on the mitigation processing need to be looked at
closely.
7.2.3 GSM-based Passive Radar

Beside GSM communication, similar problems with receiver non-linearities exist in GSM
passive radar. A GSM based passive radar system, depicted in Figure 7.18, is a bistatic
system with no dedicated transmitter [ZNW09; ZDB+12]. It uses GSM BTSs as illuminators
of opportunity to detect and track moving targets. At the receiver, the direct signal of the
172
x(t)
xref (t)
xtar (t)
BTS
Figure 7.18: Sketch of the passive radar system.
BTS and the reflections from the target are obtained. Finally, the target is located by
computing the cross-correlation of both signals, yielding the so called range-doppler map.
The performance of such a system depends critically on the used waveform, which is not
under the control of the radar designer. A limiting factor in using GSM as an illumination
waveform is given by the low bandwidth which results in a fairly low range resolution (i.
e. the ability of separating two targets in range). To overcome this inconvenience, fusion
of multiple bistatic configurations (several illuminators) is required. That is, simultaneous
reception of all potential illuminations (BTSs) is necessary.
A multi-channel receiver system with an uniform linear antenna array is used to extract the
direct path signal (reference path) and target echos (surveillance path), by taking advantage
of spatial and time filtering methods. The challenge in passive radar is to detect the weak
target echos in continuous presence of the strong direct path signal. Any possible small
distortion or neighbouring channel cross-talk will not only affect the extraction of the direct
signal, but also mask the weak target echos. The performance of clutter suppression and
weak target detection depends, therefore, critically on a clean desired channel.
Next, a simulation is conducted to demonstrate that weak target echos are masked due
to clutter induced by non-linearly induced interference. The signal scenario of interest is
illustrated in Figure 7.19. It depicts the 8192-point FFT spectrum (with Hann window)
averaged over 100 bursts, for the ideal and distorted signal, respectively. Two strong GSM
carriers are constantly present at 0.8 MHz and 1.6 MHz, causing third-order IMD around
DC and 2.4 MHz. In the DC zone, there is a target echo 70 dB below the blocker, having
a Doppler shift fd = 0.2 kHz and a time shift tshift = 40 ms (360 samples). Thus, it is a
delayed version (reflection) of the desired GSM carrier. In addition, there is a clutter signal
being 60 dB below blocker level with fd = 1.7 103 kHz and tshift = 4 ms. The desired
173
PSD (dBm/Hz)
40
50
60
70
80
90
100
110
120
130
Strong Blockers
IMD3
Ideal signal
Distorted signal
IMD3
Clutter and Target Echo
Desired Signal
2
1
0
1
2
Figure 7.19: BB spectrum of a GSM-based passive radar scenario with a weak desired carrier
and two strong blockers.
signal is masked by IMD due to RF non-linearity with a third-order coefficient of 12 that
corresponds to an IIP3 = 17.4 dBm following (2.14). After filtering the desired channel by
a 200 kHz lowpass, the target is located in range and Doppler by assuming an ideal direct
(reference) signal.
The diagrams in Figure 7.20(a) and Figure 7.20(b) illustrate Doppler-cuts of the RangeDoppler map before and after applying the feedforward mitigation algorithm. In Figure 7.20(a), the weak target echo at fd = 0.2 kHz is covered by an increased noise floor
due to the RF distortion. The target cannot be distinguished from the nearby clutter due
to the receiver non-linearity. After applying the algorithm, all interference on top of the
desired channel has been mitigated, thereby making that weak target echo visible again.
More information on GSM-based passive radar for medium range surveillance can be found
in [ZNW09; ZDB+12] and references therein.
7.3 Summary of the Considered Application Scenarios

This chapter has presented three use cases that show the benefit of mitigating receiver
non-linearity. In CR spectrum sensing, the false alarm rate due to non-linearly induced
interference has been reduced significantly after applying the algorithm. Second, cellular
mobile communication based on GSM has been investigated as an example for a real-world
communication system. It has been shown through simulations and measurements, how the
174
(dB)
(dB)
7.3 Summary of the Considered Application Scenarios
0
5
10
15
20
25
30
35
40
45
50
1.5
0
5
10
15
20
25
30
35
40
45
50
1.5
Weak Target Echo
0.5
0
0.5
Doppler (kHz)
(a) Before mitigation
1.5
Weak Target Echo
0.5
0.5
0
Doppler (kHz)
(b) After mitigation
1.5
Figure 7.20: Doppler-cuts of the 3D-shaped Range-Doppler maps.

BER performance of weak distorted GSM channels can be improved for reliable communication in GSM-R. Finally, GSM-based passive radar needs to handle weak target reflections
and strong direct signals from the GSM BTSs. It has been shown how weak targets can
be made visible after applying the feedforward algorithm. The presented use cases are just
examples of the broad range of applications of the feedforward mitigation. Basically, the
mitigation algorithm can be beneficial in all use cases with weak and strong signals being
present at wideband receivers.
175
8 Conclusions
8.1 Methodology and Main Results of the Thesis
This thesis provides a closed representation of receiver non-linearities, BB modelling, digital
mitigation, and practical aspects including implementation issues. Non-linear distortions,
being a subset of RF impairments, may cause a significant degradation in performance of
sensing, communication, and radar.
IMD in signal configurations of weak and strong signals are of major concern, especially in
coexistence scenarios of heterogeneous wireless systems. Circuit non-linearity is even more
crucial in wideband receivers with limited selectivity, such as those used in SDRs. Simple
behavioural modelling by means of memoryless polynomials, found by system identification techniques, has been shown to be sufficient for the application of distortion mitigation.
Thereby, the behavioural model should match the target receiver topology as close as possible.
A purely digital feedforward mitigation algorithm, among other Dirty RF techniques, has
been followed to enhance the RF performance of the receiver, especially due to its easy
integration and efficient mitigation capabilities at system level. It has been shown through
extensive simulations and real-world RF measurements that a considerable mitigation of
non-linearly induced interference in low-cost RF front-ends can be achieved by applying
that algorithm. As a result, a significant increase of the effective linearity of the RF frontend can be obtained. Beside suppression of the newly generated frequency components,
the BER of modulated distorted signals has been reduced substantially. RF measurements,
including device characterisation and mitigation verification, have been carried out with a
typical wideband SDR, the USRP [Ett].
Moreover, it has been found through extensive simulations that the algorithm is able to
cope with dynamic radio channel conditions, manifesting severe small-scale and frequencyselective fading. Finally, a first FPGA prototype of the purely digital algorithm has been
successfully implemented on the SDR platform USRP, thereby demonstrating the real-time
operation of the algorithm on an SDR.
177
8 Conclusions
In conclusion, the thesis demonstrates the algorithms potential of mitigating RF and/or
BB distortion created by typical DCR architectures. System-level DSP-based algorithms to
handle Dirty RF effects represent a promising alternative to conventional analogue design
optimisation. A benefit of the digital solution is its versatile range of applications due to
its independence to the actual waveform of the desired input, as only the strong blocker
signal is processed. The proposed solution is a cost efficient technique that can be easily
implemented in hardware, right in between the digitiser and the BB-specific processing.
Finally, the pure digital implementation totally meets the concept of SDR. Even more, the
mitigation algorithm is generally applicable for wideband DCRs and is not restricted to SDR
adopted in the majority of the presented use cases.
8.2 Contributions to the State of the Art

In brief, various studies have been carried out that bridge the gap between theory an practical
applicability of the digital feedforward mitigation algorithm (NONLIM), especially with the
aim of integrating that algorithm into real devices.
First, an experimental characterisation of the SDR under test (USRP) has been conducted
to obtain simple memoryless and memory polynomial models for the purpose of distortion
mitigation. Beside the common approach of analysing CW signals, wideband modulated and
real communication signals have been used as test signals.
Second, a novel cascaded model has been developed that mimics the DCR receiver topology
as close as possible, by joint consideration of non-linearities at RF and BB, as well as I/Q
imbalance effects. A feedforward mitigation architecture based on this model clearly outperforms state-of-the-art techniques in terms of the achievable mitigation gain. Add to this, the
effectiveness of NONLIM has been verified by extensive simulations and offline-processing
of measured RF front-end signals. Above all, limitations of the NONLIM algorithm are
summarised together with proposed solutions.
Third, novel studies have been conducted on the impact of fading on the nature of nonlinear distortions at the receiver. These results provide general useful information of the
interference behaviour in real-world radio environments. The algorithms performance has
been verified through simulations including realistic AGC functionality. The results show a
good mitigation and promising BER improvement.
Next, a first FPGA prototype of the NONLIM algorithm has been built to explore the
challenges of a real-time implementation. The results show that the algorithm can be easily
178
8.3 Future Work

integrated in a resource-efficient manner, introducing a negligible overall processing delay,
which is of concern in TDMA-like communication systems.
Finally, the algorithm has been applied to spectrum sensing for CR and GSM for reliable
communication and weak target detection in passive radar. These use cases have not been
considered so far and clearly show the benefit of the Dirty RF signal processing technique
in real-world scenarios.
8.3 Future Work

Future work will address in more detail, in which scenarios the algorithm will fail or behave
counter-productive, especially detecting the need for bypassing the algorithm under high
SIR conditions. Next, some limitations of the approach need to be looked at closely, e.g.
the burden on the A/D interface. A trade-off between analogue distortion regeneration
and cancellation vs. purely digital feed-forward algorithm is to be found. Obtaining ideal
reference signals without in-band distortion, either by additional linear RF paths or by remodulating the blockers, is also an attractive approach to further enhance the mitigation
performance. The use of larger FPGAs would allow, for example, to implement the total
cascaded model or to enable the mitigation processing at the full ADC sampling rate. Finally,
the real-time implementation can be ported to DSP processors, in order to demonstrate the
practical applicability of NONLIM on various real-world digital back-ends.
179
A/D . . . . . . . analogue/digital
ACPR
. . . . . adjacent channel power ratio
ADC . . . . . . analogue-to-digital converter

AF . . . . . . . . adaptive filter
AGC . . . . . . automatic gain control
AM/AM . . . . amplitude-to-amplitude
AM/PM . . . . amplitude-to-phase
ANT . . . . . . antenna
API . . . . . . . application programming interface
AWG . . . . . . arbitrary waveform generator
AWGN . . . . . additive white Gaussian noise
BB
. . . . . . . baseband
BCCH
. . . . . broadcast control channel
BEMP . . . . . bit error mitigation performance

BER
. . . . . . bit error rate
BP . . . . . . . . bandpass
BPSK . . . . . . binary phase shift keying
BTS . . . . . . . base transceiver station
CD
. . . . . . . clipping distortion
CIC . . . . . . . cascaded integrator comb
181
CIR . . . . . . . channel impulse response
CLB . . . . . . . configurable logic block
CORDIC . . . . coordinate rotation digital computer
CPLD . . . . . . complex programmable logic device
CR
. . . . . . . cognitive radio
CW . . . . . . . continuous wave
DA
. . . . . . . distributed arithmetic
DAC
. . . . . . digital-to-analogue converter
DB
. . . . . . . dummy burst
DC
. . . . . . . direct current
DCR
. . . . . . direct-conversion receiver
DDC . . . . . . direct-down conversion

DNL
. . . . . . differential non-linearity
DPD
. . . . . . digital predistortion
DSP . . . . . . . digital signal processing

DUT . . . . . . device under test
ED
. . . . . . . energy detector
EDA
. . . . . . electronic design automation
EVM . . . . . . error vector magnitude

FCCH
FDD
. . . . . frequency correction channel

. . . . . . frequency division duplexing
FDMA . . . . . frequency-division multiple access

FF . . . . . . . . flip-flop
FFT . . . . . . . fast Fourier transform
FIR . . . . . . . finite impulse response
182
FPGA
. . . . . field-programmable gate array
GigE . . . . . . gigabit ethernet

GMSK . . . . . Gaussian minimum-shift keying
GPIB . . . . . . general purpose interface bus
GSCM . . . . . geometry-based stochastic channel model
GSM . . . . . . Global System for Mobile Communications
GSM-E . . . . . GSM extended band
GSM-R . . . . . GSM for railways
HD . . . . . . . harmonic distortion
HDL
. . . . . . hardware description language
I . . . . . . . . . in-phase component
I/Q . . . . . . . in-phase/quadrature
IF . . . . . . . . intermediate frequency
IIP2 . . . . . . . input-referred second-order intercept point
IIP3 . . . . . . . input-referred third-order intercept point
IMD . . . . . . . intermodulation distortion
INL . . . . . . . integral non-linearity
IRR . . . . . . . image rejection ratio
LMS . . . . . . . least-mean square
LNA
. . . . . . low-noise amplifier
LO . . . . . . . . local oscillator
LOS . . . . . . . line of sight
LP . . . . . . . . lowpass
LS . . . . . . . . least squares
LSB . . . . . . . least significant bit
183
LTE . . . . . . . Long Term Evolution
LTI . . . . . . . linear time invariant
LUT . . . . . . . look-up table
M-IMR . . . . . multi-tone intermodulation ratio
MAC . . . . . . multiply-accumulate
MATLAB . . . MATrix LABoratory
MIMO . . . . . multiple input multiple output
ML . . . . . . . maximum likelihood
MLSE . . . . . . minimum least square error
MS
. . . . . . . mobile station
MSE
. . . . . . mean-square error
MUX . . . . . . multiplexer
NB
. . . . . . . normal burst
NCO . . . . . . numerically controlled oscillator

NF . . . . . . . . noise figure
NLMS
. . . . . normalised least-mean square
NLOS . . . . . . non-line of sight

NMSE
. . . . . normalised mean square error
NONLIM . . . . non-linearly induced interference mitigation

NPR
. . . . . . noise power ratio
OFDM . . . . . orthogonal frequency-division multiplexing

P1dB . . . . . . 1dB compression point
PA . . . . . . . . power amplifier
PAPR . . . . . . peak-to-average power ratio
PCB
. . . . . . printed circuit board
184
PDF . . . . . . . probability density function
PPS . . . . . . . pulse per second
PSD . . . . . . . power spectral density
PU
Q
. . . . . . . primary user
. . . . . . . . quadrature component
RAM . . . . . . random-access memory

RF . . . . . . . . radio frequency
RMS
. . . . . . root mean square
ROC
. . . . . . receiver operating characteristic
RSSI
. . . . . . received signal strength indicator
RXQUAL . . . receiver quality

SAW
. . . . . . surface acoustic wave
SCH . . . . . . . synchronization channel

SDR . . . . . . . software defined radio
SFDR . . . . . . spurious-free dynamic range
SINAD . . . . . signal-to-interference ratio including noise and distortion
SINR . . . . . . signal-to-interference-plus-noise ratio
SIR . . . . . . . signal-to-interference ratio
SL . . . . . . . . strictly-linear
SNR . . . . . . . signal-to-noise ratio
SOI . . . . . . . signal of interest
SPICE . . . . . simulation program with integrated circuit emphasis
SR . . . . . . . . software radio
SU . . . . . . . . secondary user
TCH
. . . . . . traffic channel
185
TDMA . . . . . time division multiple access
TDNN . . . . . time-delay neural network
TS . . . . . . . . time slot
TT
. . . . . . . two-tone
UHD . . . . . . universal hardware driver

UMTS . . . . . Universal Mobile Telecommunications System
USRP . . . . . . Universal Software Radio Peripheral
VGA . . . . . . variable-gain amplifier
WCDMA . . . . wideband code division multiple access
WINNER
. . . wireless world initiative for new radio
WL . . . . . . . widely-linear
WL-RC . . . . . reduced-complexity widely-linear
XMD . . . . . . cross-modulation distortion
186
List of Symbols
x . . . . . . . . . column vector
X . . . . . . . . matrix
[X]1 . . . . . . matrix inversion
(.)
. . . . . . . complex conjugate
(.)T
. . . . . . . transpose
(.)H . . . . . . . Hermitian transpose (complex conjugate transpose)

a1 . . . . . . . . linear gain of the RF amplifier
a2 . . . . . . . . complex third-order coefficient of the RF amplifier
a3 . . . . . . . . linear gain of the BB amplifier
a4 . . . . . . . . complex third-order coefficient of the BB amplifier
E {.} . . . . . . expectation operator
J . . . . . . . . . mean square error
k1 . . . . . . . . complex mismatch coefficient for modelling I/Q imbalance
k2 . . . . . . . . complex mismatch coefficient for modelling I/Q imbalance
x(t) . . . . . . . complex (transmitted) BB signal
xI (t) . . . . . . . in-phase component of complex BB signal x(t)
xQ (t) . . . . . . quadrature component of complex BB signal x(t)
xRF (t) . . . . . . received antenna RF signal
yBB
(t) . . . . . . BB signal including RF and/or BB distortion
187
List of Symbols
yRF
(t) . . . . . . RF signal including LNA distortion
yRF (t) . . . . . . BB representation of the distorted RF signal
188
List of Figures
2.1 Block scheme of a typical dual-IF heterodyne receiver.
. . . . . . . . . . . .
2.2 Block scheme of a homodyne (zero-IF) receiver. . . . . . . . . . . . . . . . .

2.3 Block scheme of a low-IF receiver. . . . . . . . . . . . . . . . . . . . . . . . .
9
9
2.4 Ideal SR architecture proposed in [Mit95]. . . . . . . . . . . . . . . . . . . .

2.5 Linear and non-linear transfer functions. . . . . . . . . . . . . . . . . . . . .
2.6 Block scheme of a typical SDR receiver . . . . . . . . . . . . . . . . . . . . .
10
12
13
2.7 Leakage effects in DCR architectures as another source of distortions. . . . .

2.8 Spectral components induced by non-linear device with degree 7 . . . . . . .
14
16
2.9 Impact of cross-modulation distortion (XMD) onto a weak carrier. . . . . . .

2.10 Power level diagram illustrating typical metrics for non-linear circuits. . . . .
2.11 Typical signal configuration with weak desired and strong blocker signal. . .
18
21
22
2.12 Quadrature model of a memoryless non-linear model according to [Sch09]. .

2.13 Modelling of a non-linear LNA adopted in a low-IF receiver. . . . . . . . . .
26
29
2.14 Block scheme of the RF non-linearity model derived from (2.22). . . . . . . .

2.15 RF distortion caused by two-tone input. . . . . . . . . . . . . . . . . . . . .
2.16 Modelling of non-linear BB components in a low-IF receiver. . . . . . . . . .
31
32
34
2.17 Block scheme of the BB non-linearity model . . . . . . . . . . . . . . . . . .

2.18 BB distortion caused by two-tone input . . . . . . . . . . . . . . . . . . . . .
35
36
2.19 Model of joint RF and BB distortion. . . . . . . . . . . . . . . . . . . . . . .

2.20 Signal components of cascaded non-linearity with two-tone input. . . . . . .
2.21 Block scheme of the mixer I/Q imbalance model. . . . . . . . . . . . . . . .
38
38
40
2.22 RF and BB non-linear distortion under I/Q imbalance conditions. . . . . . .

2.23 Two-box models as simplifications of the Volterra series model [Sch09]. . . .
43
45
2.24 Block scheme of a memory polynomial model [Sch09]. . . . . . . . . . . . . .

2.25 Top-level Simulink model of the WBX front-end. . . . . . . . . . . . . . . . .
2.26 Simulink model of the mixer subsystem in Figure 2.25. . . . . . . . . . . . .
47
49
50
2.27 Comparison between behavioural and circuit model. . . . . . . . . . . . . . .

2.28 Sketch of the two-tone measurement setup. . . . . . . . . . . . . . . . . . . .
2.29 BB spectrum of the two-tone excitation measured with USRP2+WBX . . .
51
54
56
2.30 Power level diagram at 472 MHz . . . . . . . . . . . . . . . . . . . . . . . . .
57
189
List of Figures
2.31 IIP2 and IIP3 measured with USRP2+WBX [Ett]. . . . . . . . . . . . . . .
57
2.32 Amplitude statistics of various CW signals. . . . . . . . . . . . . . . . . . . .
58
2.33 Distorted BB spectrum with multi-tone input. . . . . . . . . . . . . . . . . .
60
2.34 WCDMA Measurement setup. . . . . . . . . . . . . . . . . . . . . . . . . . .
60
2.35 Amplitude distribution of the measured WCDMA signal. . . . . . . . . . . .
61
2.36 BB spectrum of the input and out WCDMA signal. . . . . . . . . . . . . . .
62
2.37 Parameter estimation of the memory polnyomial . . . . . . . . . . . . . . . .
63
2.38 Comparison between memory and memoryless polynomial modelling. . . . .
63
3.1 Feedback applied around a non-linear PA generating distortion [Ken00]. . . .
66
3.2 Configuration of a basic feedforward PA. . . . . . . . . . . . . . . . . . . . .
67
3.3 Principle of BB predistortion . . . . . . . . . . . . . . . . . . . . . . . . . . .
68
3.4 Principle of a balanced amplifier according to [Sha]. . . . . . . . . . . . . . .
70
3.5 Decision feedback detector with estimated receiver non-linearity [DF11]. . . .
73
3.6 Two-channel wideband SDR receiver proposed in [ZMS09]. . . . . . . . . . .
74
3.7 Mixed-signal feedforward mitigation proposed in [KH08b]. . . . . . . . . . .
74
3.8 Block diagram of the active feedback interference cancellation. . . . . . . . .
77
3.9 Block diagram of the analogue blocker interference cancellation. . . . . . . .
78
3.10 Experimental setup for the analogue blocker cancellation. . . . . . . . . . . .
80
3.11 BB spectrum of the received signal before and after blocker cancellation. . .
81
3.12 Wideband SDR front-end with antenna array for spatial filtering [CB05]. . .
82
4.1 Block scheme of the generalized digital feedforward mitigation algorithm. . .
83
4.2 FIR filter in transposed structure [MB04]. . . . . . . . . . . . . . . . . . . .
87
4.3 Theoretical architecture for mitigating cascaded non-linearity [GAM+14]. . .
89
4.4 Power levels of distortion estimates Terms 1-6 of Table 4.1 [GAM+14]. . . .
92
4.5 Parallel mitigation architecture for cascaded non-linearity [GAM+14]. . . . .
94
4.6 Basic AF architecture for interference cancellation . . . . . . . . . . . . . . .
96
4.7 Simulation architecture for performance evaluation of the mitigation . . . . . 109

4.8 Mitigation results with two-tone excitation. . . . . . . . . . . . . . . . . . . . 112
4.9 Adaptation of the AF coefficients with two-tone input [GAM+14]. . . . . . . 113
4.10 Non-linear distortion mitigation with simulated data. . . . . . . . . . . . . . 114
4.11 Mitigation results with BPSK excitation [GAM+14]. . . . . . . . . . . . . . 115
4.12 Adaptation of the AF coefficients with BPSK input [GAM+14]. . . . . . . . 116
4.13 Sketch of the measurement setup [GAM+14]. . . . . . . . . . . . . . . . . . 117
4.14 Mitigation results obtained with measured data [GAM+14]. . . . . . . . . . 118
4.15 Non-linear distortion mitigation with measured data. . . . . . . . . . . . . . 119
190
List of Figures
4.16 Minimum frequency separation for multiple blockers. . . . . . . . . . . . . . 122
5.1 Typical fading scenario with two adjacent FDMA channels. . . . . . . . . . . 126
5.2 Geometrical layout of the fading scenario [Dup13]. . . . . . . . . . . . . . . . 127
5.3 Qualitative behaviour of different fading phenomena [Dup13].
5.4 Block diagram of the fading simulation setup [Dup13].
. . . . . . . . 129
. . . . . . . . . . . . 130
5.5 Impact of frequency-selective channel on SIR [Dup13]. . . . . . . . . . . . . . 132

5.6 PDF of SIR in different fading scenarios [Dup13]. . . . . . . . . . . . . . . 133
5.7 SIR enhancement analysis for the scenario C3 at 60 km/h [Dup13]. . . . . . . 133
5.8 SINR enhancement analysis for the scenario C3 at 60 km/h [Dup13]. . . . . . 133
5.9 BEMP analysis for different scenarios at 60 km/h [Dup13]. . . . . . . . . . . 135
5.10 BER vs. SINR at different scenarios [Dup13]. . . . . . . . . . . . . . . . . . 135
5.11 BER vs. SINR for scenario C3 with and without AGC [Dup13]. . . . . . . . 136
6.1 Block diagram of the NONLIM FPGA implementation [Sch13].
. . . . . . . 141
6.2 Existing DDC chain with the integrated NONLIM module [Sch13]. . . . . . . 143
6.3 Block diagram of a symmetric systolic FIR implementation [Sch13]. . . . . . 145
6.4 Block diagram of the pipelined implementation of refmodel [Sch13]. . . . . . 146
6.5 Implementation structure of the LMS filter [Sch13]. . . . . . . . . . . . . . . 147
6.6 Experimental setup of the two-tone scenario. . . . . . . . . . . . . . . . . . . 150
6.7 Two-tone mitigation performance with the custom FPGA firmware [Sch13]. . 151
6.8 Experimental setup of the BPSK scenario. . . . . . . . . . . . . . . . . . . . 151
6.9 BPSK mitigation performance with the custom FPGA firmware [Sch13]. . . 152
7.1 CR cognition cycle [Hay05; PKM+11]. . . . . . . . . . . . . . . . . . . . . . 154
7.2 Block diagram of a discrete energy detector. . . . . . . . . . . . . . . . . . . 156
7.3 Averaged BB spectrum and energy detection thresholds [GSH+12a]. . . . . . 158
7.4 Mitigation performance with measured multi-carrier signals. . . . . . . . . . 160
7.5 Original and cleaned baseband spectrum with energy detection thresholds. . 161
7.6 ROC curves before and after mitigation. . . . . . . . . . . . . . . . . . . . . 161
7.7 GSM NB structure. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 162
7.8 GSM frame structure.
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 163
7.9 Considered interference scenario.
. . . . . . . . . . . . . . . . . . . . . . . . 165
7.10 Baseband spectrum with a weak and 2 strong GSM channels . . . . . . . . . 166
7.11 Radio spectrum frequency allocation for GSM 900 downlink in Europe. . . . 166
7.12 Basis system architecture for GSM scenario
. . . . . . . . . . . . . . . . . . 167
7.13 Block scheme of the implemented mitigation algorithm for GSM. . . . . . . . 168
7.14 Block diagram of the GSMSim framework [EM97]. . . . . . . . . . . . . . . . 169
191
List of Figures
7.15 Block diagram of the receiver implementation used in GSMsim [EM97]. . . . 170
7.16 Laboratory setup reproducing the considered interference scenario. . . . . . . 171
7.17 BER performance vs. SIR for the desired channel. . . . . . . . . . . . . . . . 172
7.18 Sketch of the passive radar system. . . . . . . . . . . . . . . . . . . . . . . . 173
7.19 BB spectrum of a GSM-based passive radar scenario. . . . . . . . . . . . . . 174
7.20 Doppler-cuts of the 3D-shaped Range-Doppler maps. . . . . . . . . . . . . . 175
B.1 Block scheme of the complete USRP SDR platform. . . . . . . . . . . . . . . 214
B.2 USRP2 motherboard, Rev. 1.0 (2007), cp. [Ett] . . . . . . . . . . . . . . . . 215
B.3 Wi-Fi transceiver board RFX2400 [Ett] . . . . . . . . . . . . . . . . . . . . . 217
B.4 Wideband transceiver board WBX [Ett] . . . . . . . . . . . . . . . . . . . . 218
C.1 Flowgraph of the MATLAB simulator NONLIM. . . . . . . . . . . . . . . . . 219
C.2 Software framework of the USRP MATLAB interface. . . . . . . . . . . . . . 221
192
List of Tables
2.1 Frequency components generated by RF non-linearity. . . . . . . . . . . . . .
33
2.2 Frequency components generated by BB non-linearity. . . . . . . . . . . . . .

2.3 Frequency components generated by cascaded RF and BB non-linearity. . . .
2.4 I/Q imbalance simulation parameters . . . . . . . . . . . . . . . . . . . . . .
36
39
42
2.5 Extracted coefficients of the memoryless polynomial. . . . . . . . . . . . . .
56
4.1 Distortion terms produced by the cascaded non-linearity model . . . . . . . .
90
4.2 Selected terms for parallel mitigation architecture [GAM+14]. . . . . . . . . 91

4.3 Parameters of the Two-Tone Simulation . . . . . . . . . . . . . . . . . . . . . 93
4.4 Computational complexity for mitigating cascaded non-linearity [GAM+14]. 107
4.5 Parameters of the performance simulations. . . . . . . . . . . . . . . . . . . . 110
5.1 Detailed characteristics of the simulated signals [Dup13]. . . . . . . . . . . . 127
5.2 Selected WINNER propagation scenarios [Dup13]. . . . . . . . . . . . . . . . 128
6.1 FIR filter implementation for a two-tone blocker [Sch13]. . . . . . . . . . . . 144
6.2 DSP48A Utilisation of the LMS implementation [Sch13]. . . . . . . . . . . . 146
6.3 Resource utilisation of the individual sub-modules [Sch13]. . . . . . . . . . . 148
6.4 Relative resource utilisation of stock and custom FPGA firmware [Sch13]. . . 148
6.5 Delay of the invidual FPGA sub-modules [Sch13]. . . . . . . . . . . . . . . . 148
7.1 Excerpt of TDMA frame mapping of the BCCH, cp. [Etsb] . . . . . . . . . . 163
7.2 Filter parameters for the considered GSM interference scenario. . . . . . . . 168
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209
A.1 Complex Memoryless Polynomial
The common polynomial equation (2.7) stemming from a Taylor series expansion holds only
for real-valued input x(t) and suits particularly well for modelling RF distortion. However,
a notation with complex input x(t) can be derived out of it.
Starting from (2.7)
y(t) =
N
X
cn xn (t) = c1 x(t) + c2 x2 (t) + c3 x3 (t) + . . . + cN xN (t),
n=1
and writing the real part of x(t) = Re [x(t)] as

x(t) =
yields
y(t) =
N
X
n=1
N
X
cn

1
x(t)ej2fc t + x (t)ej2fc t
2

1
j2fc t
j2fc t n
x(t)e
+
x
(t)e
2n
n
x (t) + x(t)ej22fc t
|
{z
}
n=1
Pn
n
(t)]nk xk (t)ej22kfc t
[x
)
(
k=0 k

N
n
X 1 X n
xk (t) [x (t)]nk ej2(2kn)fc t ,
=
cn n
2 k=0 k
n=1
cn
(A.1)
1 j2nfc t
e
2n
(A.2)
by using the binomial identity. Next, an important assumption is made to simplify (A.2).
It is assumed that y(t) passes a zonal filter within the receiver line-up, thus, all frequency
components other than those centred at fc will be removed [MMK+06; BB99; KK01; Teh09].
Hence, only terms with 2kn = 1 k = (n1)/2 remain and only odd n will provide non-
zero output. Therefore, the binomial terms in (A.2) reduce to two and yield the simplified
211
output
N
X
1
y(t) =
cn n 2
2
n=1
N
X

n+1
n n+1
2
2 (t) [x (t)]
n+1 x
1 1
=
cn n 1
2 2
n=1
=
N
X
n=1
cn n1
2
n=1 |
{z
N
X
cn

n1
2 x1 (t)
n+1 [x(t)x (t)]
{z
}
|
2
n
n+1
2
|x(t)|2
x(t) |x(t)|n1
cn x(t) |x(t)|n1 .
Thus, (2.8) and (2.9) have been proven.
212
(A.3)

B.1 Overview
The USRP is an inexpensive and versatile platform for building SDRs, developed by Ettus
Research LLC, California [Ett]. It is a frequency-agile RF front-end that can transmit and
receive arbitrary signals on a reconfigurable frequency. With its completely open design,
it has become a very popular experimental platform for education, research projects, and
industrial applications. It consists of a motherboard and a variety of daughter boards and
needs a host computer for the main signal processing and visualisation of the results. USRP
was developed from the open-source software project GNU Radio by Matt Ettus. The
current version USRP N210 is on sale since February 2012 and is on main focus in the
following sections.
B.2 USRP Motherboard

The main USRP board acts as an interface between the host PC and the actual RF
front-ends. A block scheme of the whole USRP system is illustrated in figure B.1. Figure B.2 shows a picture of the USRP motherboard with an accentuation of its most important circuit components.
Its task is the digital/analogue and analogue/digital conversion to and from the daughter boards, signal preprocessing, and data transmission
to the PC. The printed circuit board (PCB) provides two channels, one receiver and one
transmitter. It holds a dual-channel ADC with 14 bit and 100 Msps and a dual-channel
digital-to-analogue converter (DAC) with 16 bit and 100 Msps for the I and Q component of
the receiver and transmitter path [Ett].
Because the major part of the signal processing is performed on the host PC, high data
rates have to be achieved by the interface between USRP and the host. This is a well-known
bottleneck of the USRP system. Compared to the former Bus-based USRP, the networked
USRP N210 is connected to the PC via a faster GigE instead of universal series bus 2.0
by an extra Ethernet controller chip. However, the data rate still needs to be decreased by
213

FPGA
SD
card
CPLD
Host
REF
CLK
ETH
HR
HBF
LR
HBF
CIC
HR
HBF
LR
HBF
CIC
I
Q
Mux/
Demux
CIC
LR
HBF
HR
HBF
CLK
PLL
CIC
LR
HBF
HR
HBF
JTAG
Debug
MIMO exp
ADC
Daughter
Board
RAM
PPS
ADC
DAC
DAC
Serial
Figure B.1: Block scheme of the complete USRP SDR platform.

decimation and interpolation, respectively. The maximum transferable RF bandwidth for
USRP N210 is 25 MHz in each direction, i.e. a total of 50 MHz in full-duplex mode.
The heart of the PCB is an FPGA, the Xilinx Spartan 3A DSP 3400 (XC3SD3400A), which
performs high sample rate preprocessing [Ett]. Here, the digital up- and down-conversion to
and from an IF stage is done by an NCO and CIC filters [BR06]. The USRP system has a
low-IF receiver architecture, i.e. the up- and down-conversion of the BB to RF and vice versa
is done in two steps. First, there is a coarse tuning with the programmable synthesizer at the
RF daughter board. Then, a digital fine tuning is executed, if the analogue LO frequency is
apart from the desired centre frequency. There is a decimation/interpolation filter chain of
each I and Q component: a low-rate half-band filter, the CIC and a high-rate half-band filter.
The rate of decimation/interpolation can be changed in a range [4, 512], although a factor
as multiple of 4 is recommended to enabled both filters. For the transmission of the sampled/generated data via the Ethernet connection, the data is multiplexed and demultiplexed,
respectively. There are free logic resources to shift further steps of signal processing directly
onto the device by developing an own FPGA design [Lef10; SGT13]. This would allow a
higher data rate due to further decimation or a standalone mode of the USRP. The FPGA
design uses a lot of open cores, e.g. the Wishbone bus, interrupt controller, GPIO, SPI,
214
B.2 USRP Motherboard

Bootstrap CPLD
Debug Port
Xilinx FPGA Daughter Board Tx Port
SD Card Slot
DAC
MIMO
Expansion Port
Gigabit
Ethernet Port
ADC
DC Power
Ethernet PHY Chip
Daughter Board Rx Port
Figure B.2: USRP2 motherboard, Rev. 1.0 (2007), cp. [Ett]

I2C, UART, Gigabit Ethernet MAC and the complex programmable logic device (CPLD)
bootstrap [Ett09]. Furthermore, it holds a 32 bit 50 MHz RISC soft-core micro processor,
the aeMB, for control of the daughter boards, Ethernet etc. The FPGA bitfile is loaded
from a CPLD after power-up to the actual FPGA.
There a lot of further parts and features on the USRP, but most of them are currently not
supported by the prebuild design and need a custom design. There are
1 MB of extra SRAM onboard,
A multiple input multiple output (MIMO) expansion to run multiple USRPs fully coherent in a multi antenna system [LW10],
A reference clock input for synchronisation,
A 1 PPS input for precise timing applications,
215

Auxiliary
input
and
output
channels
(auxiliary
ADCs
and
DACs)
for
received signal strength indicator (RSSI) level measurement, temperature, bias levels
[Ham08],
I/O control interface for daughter boards (e.g. selection of RF port, synthesizer lock
detection [Ham08]),
Debug interface for common logic analyzers,
High-speed serial interface, and a
JTAG interface for reprogramming the CPLD or FPGA debugging purposes using the
Xilinx ChipScope tool.
B.3 RF Front-end Daughter Boards

Ettus Research LLC offer a variety of RF front-ends for designated frequency ranges from
DC up to 6 GHz. The transmitter and receiver are either in single daughter boards or
combined in one transceiver board. The daughter boards hold an analogue circuitry to
up-convert the signal to passband frequency or to down-convert it via synthesizers and I/Q
mixers, which are software-controllable [BR06]. The daughter board is the direct interface for
attaching antennas. Furthermore, it provides amplification or attenuation of the IF signal.
All parameters can be controlled from software or directly through the FPGA [Ett]. In
addition, most of the daughter cards have a built-in transmit/receive switching to configure
either one RF port as Tx/Rx or to have separated Tx and Rx on two RF ports for a fullduplex operation. They also hold an analogue RSSI measurement and a digital I/O to control
external devices like antenna switches. Because of the USRPs open design, a development
of an own custom daughter board is possible to fulfil specific design requirements.
The daughter boards used at Ilmenau University of Technology are [Ett]:
RFX2400, full-duplex TRx for Wi-Fi frequency from 2.3 GHz . . . 2.9 GHz and
WBX, full-duplex wideband TRx for 50 MHz . . . 2.2 GHz.
The RFX2400 has a transmit power of 50 mW (17 dBm) and the WBX up to 100 mW
(20 dBm), which is frequency-dependent. Figure B.3 and figure B.4 show a picture of these
PCBs.
The TRx boards have independent LOs/RF synthesizers for the Tx and Rx path
216
B.4 Host Computer Processing and Data Acquisition
Figure B.3: Wi-Fi transceiver board RFX2400 [Ett]

to allow a split-frequency operation [Ham08]. Moreover, each daughter board has an EEPROM, which identifies the board with its parameters to the system (e.g. range of carrier
frequency and gain) and stores calibration data like DC offsets or I/Q imbalances.
B.4 Host Computer Processing and Data Acquisition

The main part of the BB signal processing takes place at the host computer. However,
the main idea of SDR is to perform all the waveform-specific processing (modulation and
demodulation) in software to be most flexible [Ham08]. There are some existing software
frameworks for own SDR implementations like GNU Radio, IRIS, and OSSIE [Ett]. Furthermore, some groups achieved to connect USRP with MATLAB/Simulink and LabView. The
most famous SDR framework is GNU Radio and the origin for the idea to develop USRP.
GNU Radio is an open-source toolkit and started in early 2000 by Eric Blossom [DGM+06].
It includes drivers for USRP and a lot of pre-assembled libraries for modulators, filters,
channel coders, and equalizers. With the GNU Radio libraries, a flow graph (written in
Python) with sources and sinks and a connection of signal processing blocks (written in
C++) can be realised. The flow graph is a direct realisation of radio block diagrams and
allows signal processing of continuous data streams. Furthermore, there is a control interface
in Python for the software-defined parameters, e.g. centre frequency, gain/attenuation,
interpolation/decimation factor, Tx and Rx multiplex options, whereas the range for certain
parameters depend on the attached daughter board [Wen09]. One disadvantageous aspect
of GNU Radio is its continuous transmission of data streams without concepts for packets or
217
Figure B.4: Wideband transceiver board WBX [Ett]

frames as well as timers [DGM+06]. Therefore, a direct MAC implementation is difficult. A
control for the MIMO expansion, external clocks and antenna selection for USRP is currently
not supported by GNU Radio and needs a custom FPGA and firmware design [LW10].
After USRP is powered on, the onboard CPLD uploads the bit file to the FPGA and the
firmware for the soft-core micro processor. In the transmit case, the host generates samples
and sends it to the USRP device. Here, the data is demultiplexed, passed through filter
chains and converted to analogue by the DAC. After a further analogue filtering, the signal
is up-converted to RF frequency at the RF daughter card. The higher the interpolation rate
is, the fewer samples have to be generated by the host.
The signal chain for the receive case is completely inverse. First, a down-conversion of the
received signal from RF to BB is done a the RF daughter card. Then, the signal is converted
to digital by the ADC, filtered and decimated, multiplexed and transmitted to the host PC.
According to the transmit case, the higher the decimation rate is, the lower the data rate
for the transmission to the host PC.
218
C Source Code
C.1 NONLIM Algorithm
The NONLIM simulator is an object-oriented implementation of the DSP-based digital feedforward mitigation algorithm in MATLAB. It comprises a group of objects, in the following
referred to as blocks, that are connected with each other to form a flowgraph that realises
the feedforward mitigation processing. This flowgraph is illustrated in Figure C.1 and is
similar to a MATLAB Simulink or GNURadio environment [BCB+]. The main idea for this
Source.m
RefModel.m
BandSplit.m
AFilt.m
NonLIM.m
Block.m
User Requests
MATLAB
Workspace
Application
Figure C.1: Flowgraph of the MATLAB simulator NONLIM.
unitised structure is to represent the block diagram Figure 4.1 and to assign all block-related
properties and methods in an own object. All blocks are connected with their input and
219
C Source Code
output matrices and a few control signals. Thereby, all individual settings of each block can
be easily accessed and controlled in the MATLAB workspace by invoking the related handles.
Another great advantage of NONLIM is that it can be easily extended by new methods. That
was a great benefit during the development and research, especially for investigations with
the different reference models and AF variants including parameter variations. Finally, the
algorithm can be applied to very different kind of data, either synthetic simulated samples
or samples from real-world RF measurements.
Next, the functionality of each block is briefly described.
NonLIM.m is the top class that instantiates other blocks and connects them into the
flowgraph. It holds some methods for plotting and calculation of figure of merits, such as
the mitigation gain.
BandSplit.m does the detection of the spectral location of the strong blocker signal and
splits the input into a desired and a reference signal.
RefModel.m computes the distortion estimates by using a specific reference signal and
performs the subsequent filtering to provide only the added distortion products in the band
of interest to the AF, without the actual blocker signal.
AFilt.m holds the AF and adjusts the distortion estimates in amplitude and phase to those
in the desired signal. It is the actual core of the algorithm and allows for choosing different
implementations, such as LMS, NLMS, and recursive least square [Hay02].
Block.m is a parent class for all blocks and holds very general properties, such as the FFT
size and axis limits for plotting. All other blocks are derived from this (inheritance).
C.2 USRP MATLAB Interface

This USRP MATLAB interface was developed at the department Electronic Measurement
Research Lab and is used for all offline measurements presented in this thesis. It is composed
of a MATLAB class and a console application (arg-interface), as illustrated in Figure C.2.
The interface follows the idea of a USRP recording service, i.e. it receives a finite number
of samples from the USRP and stores them on the host PC together with some other files
220
MATLAB
class USRP
UHD API
arg-interface
USRP
folder <timestamp>_<name>
.XML capabilities
.DAT data_output
.XML measurement
.DAT time_output
Figure C.2: Software framework of the USRP MATLAB interface.

containing meta information about the measurement. Thereby, all device parameters can
be directly set from MATLAB. Eventually, the measurement data including the devices
configuration during measurement time can be read back to the MATLAB workspace at any
time for further processing of the data.
The arg-interface mainly calls methods of the universal hardware driver (UHD), which denotes an application programming interface (API) for USRP devices [Ett]. They are both
written in C++. The arg-interface implements all low-level functions of the interface and
can be parametrised by command line options. On the application level, MATLAB employs
an abstract class that generates these command line parameters and calls the arg-interface.
All intelligence, especially the error handling, is moved into this class. At user level, all
parameters of the USRP can be simply invoked by the object properties.
During data acquisition, the arg-interface stores a set of files to the hard disk. First, there
are two binary .dat files containing the I/Q samples (data_output.dat) and the timestamp
information provided by the UHD API (time_output.dat). Second, there are two .xml files
holding meta information about the general device parameters (capabilities.xml) and the
measurement configuration (measurement.xml).
The software project has been continuously developed and successfully applied also in other
physical layer-related projects that base on USRP devices.
221
Theses
1. Modern communication systems impose tough requirements on the transceivers electronics that reach far beyond the state-of-the-art technology.
2. The analogue stages of a radios RF front-end are inherently imperfect. Achieving a
sufficient linearity, among other RF impairments, is a challenging issue in the practical
receiver design.
3. Circuit non-linearity is even more crucial in wideband receivers with limited selectivity,
such as those used as software defined radios (SDRs).
4. Coexisting heterogeneous wireless systems lead to scenarios with weak desired and
strong unwanted (blocking) signals. These blocking signals can easily enter front-end
amplification and mixing stages.
5. Receiver non-linearity generates additional frequency components that may fall
into free frequency bands or hit weak desired signals.
Thereby, odd-order
intermodulation distortions is of major concern.
6. Receiver non-linearity can be handled by design optimisation, cancelling distortionproducing signals, or by pre- or post-correction (mitigation) techniques.
7. System-level Dirty RF signal processing in the digital domain is a popular alternative
to alleviate non-linear distortions in the receiver.
8. Mitigating non-linear distortions at the receiver is more challenging than at the transmitter, especially due to the presence of multiple unknown signals with different power
levels and dynamics due to the radio channel.
9. A digital feedforward mitigation algorithm is employed that splits the received signal
in a desired and a reference signal, re-generates the distortion products, and adaptively
subtracts them from the desired signal.
10. Feedforward mitigation decreases the bit error rate (BER) of distorted signals, and
minimizes the energy of non-linearly induced interference. Finally, the effective linearity of the front-end and the dynamic range is increased.
223
11. Behavioural modelling of non-linear RF front-ends is preferred to physical modelling.

However, behavioural models need to match the physical front-end architecture and
are to be verified with different data sets and measurements.
12. A cascaded non-linear model is developed that fully matches to the topology of
a direct-conversion receiver and addresses all generated distortions at the RF and
baseband (BB) stages.
13. No detailed knowledge of the non-linearity profile and no exact polynomial coefficients
are necessary. The adaptive filter (AF) stage adjusts the reference distortions in amplitude and phase, in order to finally subtract them from the desired signal.
14. The static reference non-linearity model and the adaptive filtering stage form a kind
of adaptive Hammerstein model that takes also linear memory effects into account.
15. A typical low-cost SDR under test, the Universal Software Radio Peripheral (USRP),
is modelled by simple memoryless polynomials derived by two-tone measurements.
This type of modelling is sufficient for distortion mitigation application.
16. Measurements with wideband WCDMA signals have revealed that the USRP manifests
little memory effects of order M = 2, thus, same AF length is utilised in the distortion
mitigation architecture.
17. The feedforward mitigation principle works independently of the desired signal waveform and processes only the strong blocker signal. From theory, the mitigation architecture works for arbitrary blocker signals.
18. Extensive simulations and real-world RF measurements demonstrate the effectiveness
of the algorithm based upon two-tone and modulated BPSK signals.
19. The algorithm has been proven to work even under frequency-selective and fast fading
conditions imposed by wave propagation in a real radio environment.
20. The purely digital algorithm can be easily integrated on an FPGA with enhanced
DSP features. Thereby, the receiver self-regulates its non-linearity in real-time and
immediately provides the corrected I/Q samples for the following BB processing.
21. The feedforward technique has been successfully applied in cognitive radio (CR) for
improving the reliability of spectrum sensing, in GSM communication for improving
the BER performance of distorted GSM channels, and in GSM-based passive radar for
enabling weak target detection.
224

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Ilmenau University of Technology

International Graduate School on Mobile Communications

Dirty RF Signal Processing

1. Univ.-Prof. Dr.-Ing. habil. Reiner S. Thom (TU Ilmenau)

Ilmenau, October 2013

1.1 Research Problem . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

1.2 Solution Strategies . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

1.4 Organisation of the Thesis . . . . . . . . . . . . . . . . . . . . . . . . . . . .

2.1 Typical Receiver Architectures . . . . . . . . . . . . . . . . . . . . . . . . . .

Homodyne or Zero-IF Receiver . . . . . . . . . . . . . . . . . . . . .

Software Radio and Software Defined Radio . . . . . . . . . . . . . .

2.2 Sources of Distortions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

2.4 Crucial Signal Scenarios . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

Behavioural vs. Physical Models . . . . . . . . . . . . . . . . . . . . .

Joint Modelling of RF and BB Non-linearity . . . . . . . . . . . . . .

Interaction with I/Q Imbalance . . . . . . . . . . . . . . . . . . . . .

2.6 Circuit-level Simulation with SimRF . . . . . . . . . . . . . . . . . . . . . .

Model Identification Techniques . . . . . . . . . . . . . . . . . . . . .

Wideband Test Signals . . . . . . . . . . . . . . . . . . . . . . . . . .

3.2 Dirty RF Techniques State of the Art . . . . . . . . . . . . . . . . . . . . . .

3.3 Reasons for Selecting Digital Feedforward Mitigation . . . . . . . . . . . . .

3.4 Cancellation of Distortion-Producing Signals . . . . . . . . . . . . . . . . . .

Spatial Filtering using Antenna Arrays . . . . . . . . . . . . . . . . .

3.5 Summary of Possible Mitigation Techniques . . . . . . . . . . . . . . . . . .

4 Digital Feedforward Mitigation

Reference Non-linearity Models . . . . . . . . . . . . . . . . . . . . .

4.3 Computational Complexity . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107

Performance Simulations . . . . . . . . . . . . . . . . . . . . . . . . . 109

4.5 Fundamental Limits

4.6 Inclusion of Memory Effects . . . . . . . . . . . . . . . . . . . . . . . . . . . 122

5.1 Methodology . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 126

Simulation Setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 130

Effects of Fading on the Non-linearities . . . . . . . . . . . . . . . . . 131

Mitigation Performance under Fading Conditions . . . . . . . . . . . 132

5.3 Summary about the Impact of the Radio Channel . . . . . . . . . . . . . . . 137

6.1 Target Architecture and Basic Scenario . . . . . . . . . . . . . . . . . . . . . 139

7.1 Spectrum Sensing in Cognitive Radio . . . . . . . . . . . . . . . . . . . . . . 153

Cognitive Radio Principle . . . . . . . . . . . . . . . . . . . . . . . . 153

Sensing Algorithms . . . . . . . . . . . . . . . . . . . . . . . . . . . . 155

Problem Definition . . . . . . . . . . . . . . . . . . . . . . . . . . . . 157

Detection of Mitigated Multi-carrier Signals . . . . . . . . . . . . . . 159

7.2 GSM Cellular Radio . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 162

GSM Physical and Link Layer Overview . . . . . . . . . . . . . . . . 162

Robust Train Radio Reception (GSM-R) . . . . . . . . . . . . . . . . 164

GSM-based Passive Radar . . . . . . . . . . . . . . . . . . . . . . . . 172

7.3 Summary of the Considered Application Scenarios . . . . . . . . . . . . . . . 174

8.1 Methodology and Main Results of the Thesis . . . . . . . . . . . . . . . . . . 177

A.1 Complex Memoryless Polynomial . . . . . . . . . . . . . . . . . . . . . . . . 211

B.1 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 213

C.1 NONLIM Algorithm . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 219

1.1 Research Problem

1.2 Solution Strategies

1.2 Solution Strategies

1.3 State-of-the-Art and Research Contribution

1.4 Organisation of the Thesis

1.4 Organisation of the Thesis

2.1 Typical Receiver Architectures

2.1.1 Heterodyne Architecture